| 00:00.34 | joobie | ManxPower-work, what do you think the prob is? |
| 00:00.48 | ManxPower-work | joobie: that is with PRI debug on? |
| 00:01.03 | joobie | yaa |
| 00:01.23 | joobie | as I turned on pri debug, i am getting spammed with "Sending Set Asynchronous Balanced Mode Extended" in console every second |
| 00:01.58 | bmoraca_work | joobie: i suspect your system.conf or dahdi-channels.conf are incorrect. do you have any alarms? what is the output of "pri show span 1"? |
| 00:02.02 | ManxPower-work | joobie: contact your telco and say "I have no alarms, but I am not receiving any data on the D-channel.". If they say the line is fine, then insist a tech comes out with a T-Byrd or similar PRI test set. |
| 00:02.37 | joobie | ManxPower-work, http://pastebin.com/m4f586c8f .. output from 'pri show span 1' |
| 00:02.48 | joobie | this was working yesterday btw.. and no changes on our end |
| 00:02.58 | bmoraca_work | joobie: you're in alarm. d-channel is down. |
| 00:03.10 | joobie | does that mean it's the uplink |
| 00:03.11 | joobie | or me |
| 00:03.11 | *** join/#asterisk jks (i=jks@193.189.93.254) |
| 00:03.14 | ManxPower-work | bmoraca_work: dahdi_cfg shows no alarms |
| 00:03.25 | ManxPower-work | joobie: your line needs to be repaired |
| 00:03.38 | bmoraca_work | ManxPower-work: could be administratively down (can asterisk do that?) if it's yellow, it's them. red, it's you. |
| 00:03.39 | Docteh | jax55: xlite has some other wideband codecs in its list |
| 00:03.42 | ManxPower-work | (more technically they must fix the setup of your PRI in the telco switch) |
| 00:03.47 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
| 00:03.50 | bmoraca_work | either way, the d-channel isn't up |
| 00:03.52 | ManxPower-work | bmoraca_work: Asterisk can't do that. |
| 00:03.59 | joobie | ahh k |
| 00:04.02 | ManxPower-work | bmoraca_work: hence what I told him to tel the telco |
| 00:04.06 | bmoraca_work | yep |
| 00:04.09 | joobie | i logged a bug with them already |
| 00:04.18 | joobie | should i just call them and say "i have no alarms and the dchannel is down?" |
| 00:04.22 | bmoraca_work | didn't think asterisk could administratively take it down, though that'd be useful |
| 00:04.24 | jax55 | Docteh what other wideband codecs can deliver HD voip? |
| 00:04.26 | ManxPower-work | joobie: telling them what I told you will help them diagnose the problem quickly |
| 00:04.39 | *** join/#asterisk Akiraaa (n=Akiraaaa@79.112.35.151) |
| 00:05.00 | joobie | OK thanks a heap |
| 00:05.04 | bmoraca_work | jax55: any wideband codec is technically "HD voice" |
| 00:05.11 | joobie | btw, what should i be looking for in the output for when they fix the problem? |
| 00:05.32 | jax55 | bmoraca_work got it |
| 00:05.33 | jax55 | thx |
| 00:05.39 | Docteh | "BroadVoice-32" "Speex Wideband" |
| 00:05.43 | bmoraca_work | jax55: the term "HD Voice" is a polycom marketing term. it's their way of saying that they support wideband codecs |
| 00:05.55 | jax55 | ah ha |
| 00:05.56 | jax55 | i c |
| 00:06.19 | jax55 | ok, good to know guys...now asterisk supports all this right? i see thse codedcs if i do core show codecs |
| 00:06.30 | bmoraca_work | joobie: asterisk will tell you the D-channel is up and you should see yellow or red alarm cleared on all channels, followed by all channels informing you that they're up (depending on your console verbosity) |
| 00:06.32 | Docteh | gigaset also uses the term HD voice :) |
| 00:07.04 | jax55 | got it |
| 00:07.25 | jax55 | is there like a list of these wideband codecs that i can find? |
| 00:07.32 | Docteh | wikipedia might have something |
| 00:07.36 | bmoraca_work | jax55: whether or not asterisk can transcode certain codecs depends on whether or not you've installed those codecs. "core show translation" will tell you which codecs asterisk can transcode between and how "long" it takes to transcode |
| 00:07.46 | Docteh | or could check some softphones |
| 00:08.14 | jax55 | cool guys, thats great info...i will go ahead and do the testing now, thanks |
| 00:11.20 | joobie | bmoraca_work, what piece of data there shows you the alarm is yelllow or red though? not seeing this bit |
| 00:11.35 | joobie | in dahdi show status it says "alarms ok" |
| 00:11.58 | bmoraca_work | joobie: that means it's not in alarm, which means that you need to call your telco and tell them their switch is fucked. |
| 00:12.37 | bmoraca_work | joobie: ManxPower-work is right. call them and tell them your PRI is not in alarm, yet the d-channel is down. that'll help them figure out what's actually going on. |
| 00:13.26 | joobie | ok.. yea just called them to tell them that |
| 00:13.33 | joobie | they are saying up to 3 hours to fix it - ergh :/ |
| 00:13.39 | bmoraca_work | ick |
| 00:13.46 | bmoraca_work | have you tried rebooting yet? |
| 00:13.54 | joobie | nod |
| 00:14.15 | bmoraca_work | how many times? :P |
| 00:14.20 | joobie | i started by destroying dahdi channels, restarting dahdi channels, then restarting asterisk.. then the box itself |
| 00:14.23 | joobie | once |
| 00:14.26 | bmoraca_work | it was a joke |
| 00:14.29 | joobie | :P |
| 00:14.39 | bmoraca_work | go watch www.thewebsiteisdown.com for more info |
| 00:14.49 | joobie | there's a possibility this could be my hardward tho ya? |
| 00:15.23 | bmoraca_work | joobie: no, probably not. but i've seen cases where flapping the interface has fixed this type of thing |
| 00:15.41 | joobie | ahh |
| 00:15.53 | jax55 | Docteh , bmoraca_work i just set my softphone to G.722 ad called an extension that i know is offline, and asterisk couldn't do it...is there a special setting to set voicemail to use wideband ? |
| 00:15.59 | joobie | sucks that they are gona take 3 hours to resolve this |
| 00:16.41 | bmoraca_work | jax55: like i said, you need to have g722 support installed in asterisk. "core show translation" will tell you whether or not you can transcode between g722 and other codecs. if there's no number listed, then you don't have g722 support. |
| 00:16.55 | jax55 | i do |
| 00:17.07 | jax55 | it shows |
| 00:17.23 | bmoraca_work | pastebin the output of that command and a sip debug of the failed call |
| 00:18.38 | jax55 | ok, i think i see the setting...there is line in voicemail.conf, format |
| 00:18.44 | jax55 | and G.722 is not there |
| 00:18.58 | jax55 | only wav49|gsm|wav |
| 00:19.05 | bmoraca_work | jax55: that's the codec that the voicemail file is saved in |
| 00:19.06 | jax55 | let me try to add and reload |
| 00:19.13 | jax55 | ah |
| 00:19.37 | jax55 | ok, let me pastebin then |
| 00:21.22 | jax55 | http://pastebin.ca/1770091 |
| 00:21.50 | bmoraca_work | jax55: that's not exactly what i asked for. |
| 00:22.00 | bmoraca_work | jax55: until you get me what i asked for, i cannot help you |
| 00:22.21 | Docteh | what softphone are you using? mine doesn't do g722 |
| 00:22.47 | jax55 | using Blink |
| 00:23.31 | Docteh | oh for osx, dang |
| 00:23.47 | jax55 | yeah |
| 00:23.48 | jax55 | http://pastebin.ca/1770095 |
| 00:24.14 | jax55 | bmoraca_work is that what your looking for? |
| 00:24.41 | bmoraca_work | jax55: what format are your prompts saved in? |
| 00:24.56 | *** part/#asterisk Yedidya (n=Yedidya@nat67.mia.three.co.uk) |
| 00:25.02 | jax55 | prompts? |
| 00:25.03 | bmoraca_work | jax55: also, pastebin the sip.conf entry for this peer. |
| 00:26.05 | bmoraca_work | jax55: the prompts...the files that get played when asterisk plays something |
| 00:27.10 | jax55 | bmoraca_work the prompts is what i recorded using the voicemail recording system *97 ... and the sip.conf peer, not sure how to get that...i think this server is trixbox |
| 00:27.46 | bmoraca_work | jax55: then chances are you have not allowed this peer to be able to use g722. check with #trixbox or #freepbx on how to do that |
| 00:28.47 | jax55 | ok, so its something i need to allow per extenstion...ok, i think i can see that if i do sip show peer <ext no'> |
| 00:29.06 | p3nguin | per extension, no. per device, yes. |
| 00:29.30 | bmoraca_work | p3nguin: that shit is so played out |
| 00:29.39 | Pan3D | heh |
| 00:29.43 | jax55 | <PROTECTED> |
| 00:29.59 | jax55 | is that what we are looking for? |
| 00:30.10 | bmoraca_work | jax55: you cannot use g722 on that device then. you need to go and allow g722 on that device |
| 00:30.24 | jax55 | got it...ok, let me try and do that then |
| 00:30.27 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
| 00:30.47 | p3nguin | bmoraca_work: As soon as people learn that end-point devices are NOT EXTENSIONS, you'll never read it again. |
| 00:31.20 | Pan3D | ice fishing in hell? |
| 00:32.03 | bmoraca_work | p3nguin: Asterisk is the ONLY telephony platform that makes that distinction. get over it already. to the rest of the telephony world, extension = phone. |
| 00:32.10 | joobie | trying to implement a temporary work around for hte ISDN being down.. ive just setup a standard PSTN number to forward to my pennytel service.. previously i used to use pennytel just for outbound calls.. now im trying to setup pennytel in sip.conf so that it can receive and make calls.. here's the config i have so far http://pastebin.com/m1d9bc982 ... |
| 00:32.23 | jax55 | bmoraca_work is there a way to set this up system wide, where all peers can use the wideband codecs? |
| 00:32.23 | joobie | just changed it from peer to friend, but still not working.. anything im missing? |
| 00:32.26 | Pan3D | mbranca: that's not true at at all |
| 00:32.29 | p3nguin | I don't know if you think foul words will compel me to stop making the correction, but it won't. |
| 00:32.29 | joobie | been so long since i've reviewed this config |
| 00:32.32 | *** join/#asterisk youngproguru (n=youngpro@76.180.188.78) |
| 00:32.34 | bmoraca_work | jax55: in sip.conf, yes. |
| 00:32.45 | Pan3D | there are plenty of institutions that have modern phone (VoIP) setups which treat extensions as portable and not the device |
| 00:32.49 | jax55 | bmoraca_work, ok, trying that then |
| 00:33.01 | Pan3D | whoops |
| 00:33.06 | Pan3D | bmoraca_work: |
| 00:33.23 | bmoraca_work | joobie: what's "not working" about it? |
| 00:33.36 | jax55 | bmoraca_work can you tell me what the setting is called so i can grep for it |
| 00:33.47 | bmoraca_work | jax55: "allow" |
| 00:34.06 | *** join/#asterisk mattwj2002 (n=Matt@wikisource/pdpc.active.mattwj2002) |
| 00:34.07 | bmoraca_work | jax55: sip.conf.sample is a great resource for sip settings |
| 00:34.11 | mattwj2002 | hey guys |
| 00:34.13 | mattwj2002 | question |
| 00:34.19 | jax55 | cool! thx, checking |
| 00:34.22 | joobie | bmoraca_work, when i dial the number i get a pennytel voice recording saying the person im trying to reach is unavailable.. could be their end - but im not seeing anything come through in the asterisk console as i dial the number.. i can dial out of pennytel fine too |
| 00:34.34 | mattwj2002 | how hard is it to hook up a Cisco phone to an asterisk server? in particular a 7960 |
| 00:34.47 | p3nguin | mattwj2002: It's very simple. |
| 00:34.56 | bmoraca_work | joobie: i didn't see a "register" entry for that peer. could be what you're missing. |
| 00:35.01 | p3nguin | mattwj2002: Will you be using SCCP, SIP, or MGCP firmware on the phone? |
| 00:35.22 | mattwj2002 | I would probably go with SIP |
| 00:35.53 | mattwj2002 | SIP would have the most features correct? |
| 00:35.55 | joobie | thanks bmoraca_work - checking out the registrer syntax.. btw is ther ea way i can force that sip channel to reauthenticate? |
| 00:36.05 | mattwj2002 | or would SCCP be better? |
| 00:36.16 | joobie | it may even be that.. ive reloaded the sip module, but duno if that is sufficient to force reconnection |
| 00:36.22 | Pan3D | if you're using *, SIP |
| 00:36.27 | bmoraca_work | joobie: your peer entry is enough to tell asterisk how to get to pennytel, but pennytel still needs to know how to get back to you (that's what the register is for) |
| 00:36.27 | p3nguin | mattwj2002: SCCP loses features, and you'll have softkeys that do nothing. |
| 00:36.35 | mattwj2002 | oh |
| 00:36.39 | mattwj2002 | okay sip it is |
| 00:36.39 | p3nguin | mattwj2002: Go with SIP 8.11 |
| 00:36.40 | mattwj2002 | :) |
| 00:36.41 | bmoraca_work | joobie: reloading sip will cause your asterisk box to reregister |
| 00:36.51 | joobie | kk thanks :) |
| 00:36.52 | p3nguin | mattwj2002: SIP 8.12 has a CallerID bug. |
| 00:37.18 | bmoraca_work | 8.14 is latest, i believe...haven't had any problems with that one |
| 00:37.22 | p3nguin | oh |
| 00:37.35 | p3nguin | 8.12 was the latest when I got my firmware. |
| 00:37.41 | bmoraca_work | i'm using whatever the latest is, and it works fine |
| 00:37.48 | p3nguin | I haven't looked for an upgrade beyond that. |
| 00:38.00 | bmoraca_work | p3nguin: why fix what isn't broke? |
| 00:38.05 | mattwj2002 | so.... |
| 00:38.06 | p3nguin | pretty much |
| 00:38.32 | mattwj2002 | you lose a lot of features with a sip firmware? |
| 00:38.36 | bmoraca_work | mattwj2002: read this entire document before you ask any questions: http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7960g_7940g/sip/8_0/english/administration/guide/sipaxd80.html |
| 00:38.41 | p3nguin | I would be interested to know operation differences between 8.11 and 8.14, though. Not differences on paper, but actual differences. |
| 00:38.46 | bmoraca_work | mattwj2002: no, you just need to manyally configure them. |
| 00:38.50 | bmoraca_work | p3nguin: i doubt there are any |
| 00:38.50 | p3nguin | mattwj2002: Depends on what you mean. |
| 00:39.29 | p3nguin | mattwj2002: SIP does everything it is supposed to do, minus line presence (but Cisco SIP wasn't supposed to do it anyway). |
| 00:39.40 | mattwj2002 | ok |
| 00:40.32 | p3nguin | Using SCCP, the transfer button does nothing, the park button does nothing... there was probably something else, but I don't remember. |
| 00:42.00 | jax55 | bmoraca_work cool! it works! i did allow=all and now i got HD Voice! |
| 00:42.08 | mattwj2002 | you guys are never going to believe what I am going to use for an asterisk server |
| 00:42.15 | mattwj2002 | this is a hobby server by the way |
| 00:42.20 | Pan3D | amiga? |
| 00:42.22 | mattwj2002 | *hobbyist |
| 00:42.27 | mattwj2002 | nope |
| 00:42.40 | mattwj2002 | Aspire Aspire One netbook :) |
| 00:42.45 | *** join/#asterisk Alagar (n=Administ@122.164.103.38) |
| 00:42.57 | bmoraca_work | jax55: it's generally best to be as explicit as possible in your peer configurations. i don't really recommend "all" as a setting for anything. but, as always, YMMV. |
| 00:43.01 | Pan3D | uhhh... that's going to run really hot |
| 00:43.16 | Pan3D | you won't be able to let it sleep if you're expecting incoming |
| 00:43.20 | p3nguin | My SCCP test was with 8.1(2) on a 7940G. |
| 00:43.28 | jax55 | bmoraca_work is there security concerns or just best practice concerns? |
| 00:43.34 | mattwj2002 | I won't let it sleep |
| 00:43.39 | Pan3D | security |
| 00:43.45 | mattwj2002 | I am only going to have a few phones on it |
| 00:43.46 | Pan3D | mattwj2002: yeah, that will run hot |
| 00:44.03 | bmoraca_work | jax55: in IT, it's ALWAYS best practice to be as explicit as possible. best practice = secure, 99% of the time. |
| 00:44.08 | mattwj2002 | it has a 1.6 Ghz Atom processor |
| 00:44.18 | Pan3D | proceed with caution, but it's doable. I had * running on a PowerBook G4 laptop briefly. |
| 00:44.36 | mattwj2002 | ok |
| 00:44.50 | jax55 | Pan3D bmoraca_works are there vulnerabilities in the codecs themselves? |
| 00:45.12 | mattwj2002 | see I thought it would be good because of power outages.... |
| 00:45.19 | mattwj2002 | the machine won't drop |
| 00:45.29 | p3nguin | built in UPS |
| 00:45.32 | mattwj2002 | yup |
| 00:45.41 | Pan3D | yes, but that isn't really the concern. It's that the are variables in the config which, when not explicit, could lead to access/manipulation by unintended sources. |
| 00:45.53 | bmoraca_work | jax55: maybe. the point is that you don't just want your users doing anything. then again, you are running trixbox, so you may as well blow it all out of the water. |
| 00:45.53 | p3nguin | Put it on a cooling pad. |
| 00:45.57 | Pan3D | hehe |
| 00:46.07 | jax55 | hehehe |
| 00:46.16 | mattwj2002 | :P |
| 00:46.23 | jax55 | thanks guys, i will take that in consideration...thanks again for all the help |
| 00:46.41 | p3nguin | The bad thing is that leaving your AC power plugged in at all times will shorten the battery's life. |
| 00:46.53 | mattwj2002 | true |
| 00:47.26 | mattwj2002 | this is a spare machine anyways |
| 00:47.54 | mattwj2002 | it was a door stop until last night |
| 00:49.22 | Docteh | odd i just tried out "PhonerLite" on windows and the gui shows it can reorder codecs but that actually doesn't happen. |
| 00:51.11 | Docteh | or is codec priority not something able to be done in SIP? |
| 00:53.14 | Pan3D | jax55: as an side, I found the asterisk book to be really useful. The authors point out places in the * configuration where there could be possible security leaks. |
| 00:53.26 | Pan3D | aside* |
| 00:53.40 | Pan3D | also, http://www.amazon.com/Asterisk-Hacking-Ben-Jackson/dp/1597491519 |
| 00:53.50 | p3nguin | ~book |
| 00:53.50 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 00:54.03 | jax55 | Pan3D excellent thanks! I will get it |
| 00:56.17 | jax55 | when 2 peers are talking, how can I see in what codec the conversation is being handeled on the server side? |
| 00:57.17 | bmoraca_work | jax55: sip show channels |
| 00:57.34 | jax55 | excellent! thanks |
| 00:58.25 | bmoraca_work | time to go home |
| 01:00.50 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 01:02.10 | *** join/#asterisk fibres (n=no@cpc2-nfds1-0-0-cust1021.lei3.cable.ntl.com) |
| 01:04.38 | Docteh | odd, if i enable g722 on a softphone and then call a ulaw only device, asterisk transcodes, shouldn't it go ulaw? |
| 01:05.18 | p3nguin | Is ulaw preferred over g722? |
| 01:05.46 | Docteh | well i told it to prefer g722 but im not sure why that results in transcoding |
| 01:06.00 | Docteh | is there a way to prefer it with less force? |
| 01:08.18 | p3nguin | * is transcoding because you told the phone to prefer g722, I would guess. |
| 01:09.09 | p3nguin | If one device prefers ulaw and another prefers something else, the result should be transcoding. |
| 01:11.11 | Docteh | dang, i guess to avoid that I'd have to force g722 when desired via the dialplan? |
| 01:13.24 | *** join/#asterisk ShadowHntr (n=sentinel@wikipedia/Shadowhntr) |
| 01:15.36 | *** part/#asterisk paulc (n=paulc@unaffiliated/paulc) |
| 01:18.24 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:19.40 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.243) |
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| 01:21.44 | *** join/#asterisk dkirker-openmobl (n=dkirker@openmobl/ceo/dkirker) |
| 01:21.54 | ruben23 | hi any suggestion on this error: http://pastebin.com/m1fad1e6a |
| 01:23.12 | *** join/#asterisk Kumbang (n=kumbang@rusnas.paume.itb.ac.id) |
| 01:27.56 | *** join/#asterisk em_pleh (n=em_plehh@cpe-98-149-154-239.socal.res.rr.com) |
| 01:27.58 | em_pleh | hello there |
| 01:28.11 | em_pleh | is it possible for me to use my modem as a fxo? for my fax line? |
| 01:28.58 | *** join/#asterisk coppice (n=chatzill@234.157.17.210.dyn.pacific.net.hk) |
| 01:30.18 | carrar | no |
| 01:30.22 | em_pleh | lol |
| 01:30.32 | em_pleh | that was not the answer i was looking for |
| 01:31.13 | carrar | Your modem would need to be a FXS |
| 01:31.29 | em_pleh | ah thats what i need |
| 01:31.30 | em_pleh | lol |
| 01:31.39 | em_pleh | can I fax using my asterisk box? |
| 01:31.43 | carrar | yes |
| 01:31.59 | carrar | and receive |
| 01:32.13 | em_pleh | ah |
| 01:32.17 | em_pleh | how do i do that? |
| 01:32.28 | carrar | read up! |
| 01:32.36 | carrar | 1.6 supports faxing |
| 01:32.44 | carrar | or you can use Hylafax |
| 01:32.50 | carrar | and iaxmodem |
| 01:33.14 | em_pleh | i cannot find any good connectors to hylafax for windows 7 |
| 01:33.24 | carrar | there are lots |
| 01:33.29 | carrar | and they work great |
| 01:33.30 | em_pleh | really |
| 01:33.34 | *** join/#asterisk Linuturk (n=linuturk@unaffiliated/linuturk) |
| 01:33.35 | em_pleh | where in the world did you find it |
| 01:33.39 | carrar | they are printer drivers |
| 01:33.39 | em_pleh | i googled all over |
| 01:33.50 | carrar | well for windows |
| 01:33.56 | carrar | no idea about 7 |
| 01:34.03 | carrar | time to stop using windows |
| 01:34.09 | em_pleh | lol |
| 01:34.35 | *** join/#asterisk neurosys (n=neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 01:35.45 | carrar | You might be able to use hyafax via Email and a PDF attachement |
| 01:35.48 | carrar | to send faxes |
| 01:36.31 | carrar | defaintely for receiving faxes |
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| 02:36.59 | jmcdowell | Sup ma shizznizzles? |
| 02:37.00 | jmcdowell | ;) |
| 02:37.18 | jmcdowell | Anyone have any experience with say 3 line freepbx setups? |
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| 02:37.47 | jmcdowell | I haven't even done it yet, but now I am getting curious and may setup a couple more lines up so I know how it works. |
| 02:38.06 | jmcdowell | I am guessing it would trying, and if busy, try 2 and if busy try 3 and if busy fail. |
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| 03:26.13 | ectospasm | I need to write a script, either in AMI, AGI, or bash/Perl which generates one-time or infrequent-use dialplan. This script needs to set multiple Skype account properties for many Skype user accounts, and I don't know how best to start. |
| 03:27.13 | ectospasm | It seems that to use the command Set(SKYPE_ACCOUNT_PROPERTY(account,property=value)), I need to have an active channel open to run this application |
| 03:28.37 | ectospasm | since this is a batch operation, I don't know which channle to use. I'm trying a limited AMI test, with Action: Originate, and setting a custom Local channel (created exten 1234 in [ami], so the channel I'm trying is Local/1234@ami) |
| 03:29.49 | ectospasm | it doesn't seem to be executing the Application I specify (for testing, just a NoOp). It seems to hit the Answer I've put in dialplan, and then doesn't do anything |
| 03:31.31 | *** join/#asterisk mattems (n=matt@203.217.56.72) |
| 03:31.36 | mattems | hey all |
| 03:32.04 | mattems | im wanting to change the language to AU for my phone system, any helo would be appreciated |
| 03:32.11 | mattems | help* |
| 03:33.18 | *** part/#asterisk mattems (n=matt@203.217.56.72) |
| 03:39.10 | ectospasm | I think what I'll do is create a custom Macro, and generate special dialplan to execute the script. Then, create an AMI script which iterates through all the accounts and calls the macro. |
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| 04:21.23 | normaldotcom | hello, I'm trying to connect a softphone to my asterisk box, but I'm getting "Registration from <sip:username@domain> failed for '<ipaddr>' - no matching peer found |
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| 04:44.08 | voipmonk | pastebin the peer in sip.conf , normaldotcom |
| 04:48.19 | normaldotcom | thanks, I got it fixed |
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| 04:59.51 | securevoip | Polycom phones upset me... They remove themselves from queues after 1 minute of ringing... -- Got SIP response 603 "Decline" back from 10.10.11.142 |
| 04:59.52 | securevoip | <PROTECTED> |
| 04:59.52 | securevoip | <PROTECTED> |
| 05:01.06 | securevoip | Any idea what param to change to fix this? |
| 05:18.30 | p3nguin | I would bet there is a setting in the config for that. |
| 05:20.38 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat/x-termutqoocokcjuo) |
| 05:21.05 | securevoip | yes, but I know Asterisk (not Poly). Wondering if anyone out there is a Poly config expert?!? |
| 05:24.02 | carrar | call.offeringTimeOut |
| 05:24.12 | carrar | defaults to 60 seconds |
| 05:24.38 | carrar | in your sip{MAC}.cfg |
| 05:25.04 | carrar | You really should read the Polycom Admin guide if you are gonna use Polycom Phones |
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| 05:35.31 | securevoip | thx! |
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| 06:54.03 | kruemeltee | good mornin' |
| 07:02.28 | *** join/#asterisk benngard (n=benngard@213.88.138.230) |
| 07:04.04 | kruemeltee | is there any way to tell a SIP Client (telephone) it's always an agent for a specific queue? The manual in front of me is telling about a AgentLogin, but I don't want the agents to login. There telephone has to be an active agent the whole day, the whole week, the whole year ;-) |
| 07:04.40 | kruemeltee | :%s/There/Theire/g :-) |
| 07:06.30 | hardwire | For occasionals you add them in via Agent login or as a dynamic queue member. |
| 07:06.40 | hardwire | for weeklys they make sure they are logged in every day. |
| 07:06.50 | hardwire | for yearlys you get up off yer tookie and plug them into a static list. |
| 07:08.10 | hardwire | I prefer to flush, via script, all agent logins at the end of the day. |
| 07:08.50 | p3nguin | kruemeltee: Use "member => SIP/yourdevice" in the queue config. |
| 07:09.05 | hardwire | That sort of forces them to log in which is part of our time system, but also gets them in the habit of doing so every day they need to be part of a queue. |
| 07:10.03 | hardwire | -> *zonk* |
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| 07:10.46 | p3nguin | kruemeltee: I don't like logins either, so I just use the device directly. |
| 07:11.57 | kruemeltee | thanks hardwire and p3nguin ... I'll try at first p3nguin's hint ... but I'll reflect on hardwire's hints too :-) |
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| 07:13.35 | p3nguin | In a small office where you'll always have the same handful of phones with the same users answering calls, adding the devices as members is perfectly acceptable. |
| 07:14.43 | p3nguin | If you intend to do "hot desking" where people move around to different phones, you'll have to go a different route. |
| 07:16.43 | kruemeltee | p3nguin, we have a small Call Center here ... there are currently 7 devices within the Call Center ... 4 of them are always agents ... the rest of them are office phones that don't have to ring if anybod's in the queue ;-) So it's fixed |
| 07:18.21 | p3nguin | You can also use Local channels as members. That gives you flexibility to work additional dialplan magic between the queue and the phone that answers. |
| 07:19.33 | kruemeltee | that's one item on my "to-do" list ... getting to know, what a "local channel" is ;-) |
| 07:20.23 | p3nguin | You're familiar with SIP/somedevice... |
| 07:20.42 | kruemeltee | jep |
| 07:21.00 | kruemeltee | (I hope so) |
| 07:21.18 | p3nguin | You can use Local/exten123@randomcontext to call to another extension in some other context. |
| 07:21.38 | fiddur | But using "local" interfaces makes the queue magic much harder at the same time... The queue is not as aware of the "inuse" state of a local interface |
| 07:21.50 | p3nguin | That's just a crude example. |
| 07:23.00 | p3nguin | Yeah, local channels as queue members does have caveats -- this is why I just use the SIP device as a member. |
| 07:23.12 | *** part/#asterisk GameGamer43|Mac (n=GameGame@CPE-65-27-76-78.new.res.rr.com) |
| 07:23.32 | kruemeltee | I've currently got a simple dialplan ... 2 Phones registered ... 601 (agent for Queue "Call Center 1") within context "Call_Center" and 2000, normal phone within context "default" |
| 07:23.43 | kaldemar | there's always /n with local channels to make them behave just like any other channel. |
| 07:24.08 | kruemeltee | if I'm dialling the 600 from 2000er if use this extension |
| 07:24.30 | kruemeltee | exten => 600,1,Goto(Call_Center,_0351896911X.,1) |
| 07:24.53 | kruemeltee | and that's the "normal" extension for dropping somebody in the queue ... |
| 07:25.34 | kruemeltee | _0351896911X.,1 is the extension within context "Call-Center" that drops the call into the queue ... |
| 07:25.42 | p3nguin | I guess if that works for you. |
| 07:25.45 | kruemeltee | so this has to be right, isn't it? |
| 07:25.54 | kruemeltee | yea ... great ... |
| 07:26.05 | p3nguin | I would have to see the entire thing to know what's going on. |
| 07:26.16 | kruemeltee | wait a second ... |
| 07:27.25 | p3nguin | On my system, call comes in, some sound files are played, callers have the opportunity to dial extensions to listen to other sound files and/or call phones. If they don't dial any extensions, they go into Queue(). |
| 07:27.42 | kruemeltee | http://kubuntu.pastebin.ca/1770452 |
| 07:28.02 | kruemeltee | that's a simple dialplan ... just for testing my queue ;-) |
| 07:28.18 | p3nguin | I haven't been able to load pastebin.ca for two days, so we'll see if I can view that. |
| 07:28.23 | *** join/#asterisk e-jones (n=jkastner@nat/redhat/x-skjootvngixlhgaa) |
| 07:28.29 | kruemeltee | good luck ;-) |
| 07:28.53 | p3nguin | It hasn't loaded yet, so I'm sure it will timeout. |
| 07:29.35 | p3nguin | Maybe you could duplicate it in pastebin.com. |
| 07:29.56 | kruemeltee | sure |
| 07:30.55 | kruemeltee | http://www.archlinux.pastebin.com/d389e5fcb |
| 07:31.32 | carrar | exten => _0351896911X.,n,Queue(Call Center 1) |
| 07:31.40 | carrar | You have a space in the queue name |
| 07:31.50 | kruemeltee | is this wrong? |
| 07:32.06 | kruemeltee | I thought I can use names without any restriction |
| 07:32.33 | *** join/#asterisk joepunk (i=4c1d535e@gateway/web/freenode/x-clgyiufokpveafjs) |
| 07:33.02 | joepunk | anyone interested in seeing asterisk perform an old parlor trick called "The Wizard" ? |
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| 07:36.03 | p3nguin | kruemeltee: Looks pretty good. I didn't funny critique it, but I did look somewhat closely, and I didn't see anything really wrong. |
| 07:36.33 | kruemeltee | lokk like the example of the asterisk manual :-) |
| 07:36.34 | p3nguin | kruemeltee: err... fully critique |
| 07:36.43 | kruemeltee | okay ... go on |
| 07:36.58 | kruemeltee | (if it's the space within the quene Name ... thats fixed |
| 07:38.05 | p3nguin | I would replace spaces with hyphens or underscores just to not create bad habits, but I can't see that a space in a queue name would cause failure unless I tested it. |
| 07:38.19 | p3nguin | can't say |
| 07:38.30 | kruemeltee | yea ... but I already fixed that ... ;-) |
| 07:38.45 | p3nguin | I should go to bed, with all the wrong words I'm typing. |
| 07:38.53 | kruemeltee | it's just a convention for me ... as I'm a beginner ... |
| 07:39.00 | kruemeltee | go to bed? where are you from? |
| 07:39.06 | p3nguin | central USA |
| 07:39.14 | kruemeltee | okay ... :-) |
| 07:39.21 | p3nguin | 01:39 here |
| 07:39.51 | kruemeltee | P.S. I'm from germany (hope nobody's agry 'bout that) ... here it's 08:39 in the morning :-) |
| 07:40.21 | p3nguin | By 8:39, I'll be dead tired. |
| 07:40.40 | kruemeltee | *gg* |
| 07:41.00 | p3nguin | We have people from all over the world in this channel, so make yourself at home. |
| 07:41.01 | kruemeltee | I used to be too a few weeks ago |
| 07:41.16 | kruemeltee | I'll do so ... thanks ... |
| 07:42.00 | *** join/#asterisk joepunk_ (n=anonymou@c-76-29-83-94.hsd1.il.comcast.net) |
| 07:43.05 | joepunk | I've been dying to get some feedback on "The Wizard" |
| 07:43.22 | kruemeltee | so go on with "The wizard" |
| 07:43.31 | joepunk | hehe |
| 07:43.34 | *** join/#asterisk kerx (n=Peter@76-240-161-68.lightspeed.irvnca.sbcglobal.net) |
| 07:43.43 | kerx | Hi, anyone have a recommendation for an Open Source SIP Phone? |
| 07:43.48 | joepunk | well, I made a set of extensions to eliminate the need for a human when doing this trick: http://www.wikihow.com/Play-Mr.-Wizard |
| 07:44.04 | joepunk | 630-422-7390 (it's live now) |
| 07:44.46 | joepunk | my script is it answers, I say "The wizard please" then when he hits the suit I say "Sure, I'll hold" then when he says the number I say "hello?" and hit speaker phone. |
| 07:45.10 | joepunk | then when he says "this is the wizard how can i help you" I say "please tell ___ his/her card" |
| 07:45.12 | joepunk | and he does :) |
| 07:45.27 | joepunk | i posted my source on the asterisk forum under support |
| 07:45.43 | p3nguin | You know it just keeps saying the suits, right? |
| 07:45.49 | kerx | anyone use linphone before? |
| 07:45.52 | joepunk | you have to say "Sure i'll hold" |
| 07:45.58 | joepunk | right after he says the suit you need |
| 07:46.06 | joepunk | then when he says the number you want, you say "HI" |
| 07:46.18 | joepunk | not familiar with linphone |
| 07:46.20 | p3nguin | Seems like a bother. |
| 07:46.30 | kruemeltee | *gg* ... already read the how to ... great deal ;-) |
| 07:49.05 | p3nguin | A few more hours and I'll be up a full day. |
| 07:53.14 | kerx | anyone know how to make a sip soft phone? |
| 07:53.15 | kerx | ;) |
| 07:56.05 | kerx | guess not |
| 07:56.40 | ChannelZ | I think it requires some eggs. |
| 07:57.06 | joepunk | http://stackoverflow.com/questions/1067692/how-to-build-a-softphone-using-sip-protocol-using-c |
| 07:57.15 | kerx | joepunk, I read that. Not really helpful |
| 07:57.21 | joepunk | hehe |
| 07:57.25 | kerx | I'm digging through linphone though |
| 07:57.28 | kerx | http://www.linphone.org/index.php/eng/code_review/liblinphone |
| 07:57.34 | joepunk | i just use zoiper :P |
| 07:57.35 | kerx | It may be the best answer I got from Google |
| 07:57.43 | kerx | joepunk, is zoiper open source? |
| 07:57.45 | joepunk | and siax for iphone |
| 07:57.52 | joepunk | no clue, probably not |
| 07:58.09 | kerx | Hrm. It's nto free, but it has an API |
| 07:58.10 | kerx | Interesting |
| 07:58.48 | joepunk | yeah, linphone and siphon. I like zoiper a lot |
| 07:58.59 | joepunk | been using it for years now |
| 07:59.01 | kerx | Have you used Linphone by any chance? |
| 07:59.06 | joepunk | no i have not |
| 07:59.47 | *** join/#asterisk maagic (i=maagic@fsck.fi) |
| 07:59.59 | joepunk | looks pretty easy to setup & use |
| 08:00.30 | joepunk | installing now |
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| 08:35.05 | *** join/#asterisk _abc_ (n=no@unaffiliated/ccbbaa) |
| 08:35.50 | _abc_ | hello. it seems that lg lip7000 phones are not supported by asterisk. true? also is vodavi ip 7000 a direct drop in replacement? is vodavi supported? |
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| 08:40.18 | kaldemar | _abc_: they're proprietary. |
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| 08:42.39 | _abc_ | key systems? kaldemar documents indicate that LIP7000 phones know how to dial out through soho adsl connections. this strongly suggests a popular protocol |
| 08:42.49 | _abc_ | kaldemar: are you sure they are all proprietary? |
| 08:43.10 | _abc_ | and is vovida an oem or something like that? their phones are almost identical to the lg ones |
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| 08:45.36 | kaldemar | _abc_: that only indicates that they work over IP. nothing more. LG-nortel lists them as proprietary themselves, so it's pretty obvious. |
| 08:46.05 | _abc_ | i see. well mgcp and unistim were also proprietary once upon a time. thanks for the info |
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| 09:23.27 | reptiles | anyone mind answering a debian linux question? |
| 09:25.03 | m0t3jl | Hi, what is the name of the feature I need to use when I would like to have three outgoing SIP connections to three PSTN to SIP modems and when someone from my network wants to make a call Asterisk would determine which of these outgoing connections to use for the call? Thanks a lot |
| 09:28.04 | kaldemar | m0t3jl: you have the modems defined as devices in sip.conf and then do the selection in your dialplan. |
| 09:28.42 | m0t3jl | kaldemar, I can determine a line is being used? Wow ; |
| 09:28.43 | m0t3jl | ;) |
| 09:28.51 | m0t3jl | kaldemar, how? ;) |
| 09:30.26 | c0rnoTa | m0t3jl, once a solved this task via global variable, where was defined witch channel was used last time and then GOTOIF application told me what channel i must use. Ofcourse, it's busy, it makes circle looking for free line. |
| 09:30.48 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 09:30.49 | c0rnoTa | or i didn't understand your question? |
| 09:32.43 | c0rnoTa | PBX -> ATA (PSTN TO SIP) -> Asterisk is it right? |
| 09:32.58 | m0t3jl | c0rnoTa, opt out the PBX ;) |
| 09:33.19 | m0t3jl | c0rnoTa, I will blow the PBX up when Asterisk finally takes over :D |
| 09:35.25 | c0rnoTa | :) i couldn't understant direction of your call. From Asterisk via SIP to PSTN over ATA, right? :)) Asterisk -> SIP -> Modem -> PSTN |
| 09:35.31 | m0t3jl | c0rnoTa, I am basically asking if there is a way to tell Asterisk to determine which line is free to use, since there can only be one conversation held on one line... |
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| 09:37.37 | m0t3jl | Äau |
| 09:37.46 | kaldemar | m0t3jl: you need to learn some dialplan coding for that. keywords are extensions, SIP devices, core show functions like GROUP, core show function DEVICE_STATE. two last ones are asterisk CLI commands that print functions and their usage. there's many ways to do this. |
| 09:38.34 | m0t3jl | kaldemar, wow, I'd think there would be working solution since this is not so rare thing ;) |
| 09:38.52 | kaldemar | those are the working solution. |
| 09:39.02 | m0t3jl | kaldemar, ;) |
| 09:39.20 | m0t3jl | kaldemar, what is this "use case" called? Trunking? |
| 09:39.47 | kaldemar | ~trunk |
| 09:39.48 | infobot | it has been said that trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 09:40.19 | m0t3jl | kaldemar, so what is it then? :) |
| 09:40.23 | kaldemar | no special name for that use case, it's just a selection you do in your dialplan. |
| 09:40.32 | m0t3jl | kaldemar, damn :( |
| 09:40.49 | *** join/#asterisk defswork (n=andy@mx2.3gcomms.co.uk) |
| 09:41.04 | m0t3jl | kaldemar, I was hoping I could find some working examples to steal from :) |
| 09:41.06 | kaldemar | were you expecting a google search term to find a ready made dialplan? |
| 09:41.48 | m0t3jl | kaldemar, basically yes :) |
| 09:42.03 | m0t3jl | kaldemar, at least something decent enought to start with ;) |
| 09:43.16 | kaldemar | i already have you some. |
| 09:43.49 | kaldemar | s/have/gave/ |
| 09:44.05 | kaldemar | ~thebook |
| 09:44.06 | infobot | methinks thebook is 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://astbook.asteriskdocs.org |
| 09:44.16 | kaldemar | for dialplan basics, that's a good read. |
| 09:48.31 | m0t3jl | methinks I've already read the Book ;) |
| 09:48.48 | *** part/#asterisk Tech_Travis (n=Administ@cpe-76-168-191-127.socal.res.rr.com) |
| 09:49.26 | c0rnoTa | i've used only voip-info.org for my asterisk knowlage |
| 09:50.26 | kaldemar | voip-info.org is dangerous in the sense that the examples may be based on any version. lots of the information there is usually outdated. |
| 09:50.31 | *** join/#asterisk aidinb (n=Aidin@32.169.86.34) |
| 09:51.07 | m0t3jl | kaldemar, that I agree with... |
| 09:51.08 | kaldemar | primary source for app and function documentation is asterisk itself. core show applications and core show functions always prints the docs for your version. |
| 09:51.24 | kaldemar | and there's the doc directory in the source package. |
| 09:53.12 | c0rnoTa | kaldemar, you are right. voip-info has use google search engine, that's way i can print some of my thoughts and it gives me direction of research |
| 09:53.52 | c0rnoTa | i'm never search 'ready made dialplan' |
| 09:55.19 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat/x-wllkflukiusjyxgv) |
| 10:03.13 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 10:03.29 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 10:04.12 | maxagaz | Is is better to buy Dialogic cards or Digium cards ? |
| 10:04.16 | maxagaz | Is it |
| 10:04.25 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 10:05.55 | m0t3jl | maxagaz, I hear there are some problems when you have multiple Digium cards in one machine... |
| 10:07.48 | kaldemar | maxagaz: which cards are you referring to? |
| 10:08.05 | maxagaz | kaldemar, no card in particular |
| 10:08.12 | maxagaz | kaldemar, I'm just wondering |
| 10:09.46 | kaldemar | maxagaz: i'd go for ones with best support in asterisk, i.e. digium's cards. digium and sangoma are used most. |
| 10:12.12 | *** join/#asterisk aidinb (n=Aidin@32.169.86.34) |
| 10:16.27 | Akiraa | What about digium clones, anything worthwhile there? |
| 10:19.57 | *** join/#asterisk tw56 (n=tw56@host86-178-171-207.range86-178.btcentralplus.com) |
| 10:20.47 | kaldemar | Akiraa: i haven't tried any, but haven't heard any praise for them either. |
| 10:21.34 | tw56 | Hi People. I'm getting the following errors on misdnportinfo but i haven't got a clue what's wrong - can anyone help. http://pastebin.com/m101069a4 |
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| 10:54.16 | Falle4 | I have a problem with incoming sip calls. The prompts played to external incoming calls are in language "en" even though the language setting in sip.conf is "se". If i call from a local IP-phone it works fine. This only happens on external calls from our provider. There is only one occurance of the language setting in sip.conf and it is set to "se". Anyone that can shed some light on this issue? |
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| 11:29.54 | garymc | Hey anyone know what card i need to buy to go with the sangoma A101D |
| 11:30.14 | garymc | to do faxes? |
| 11:30.30 | garymc | and how hard would it be for me to configure now? |
| 11:30.56 | kaldemar | what do you mean by go with the A101D? |
| 11:32.29 | garymc | I heard there was a card that goes with the A101D or plugs in the server so i can send faxes down my isdn? |
| 11:32.51 | garymc | PRI |
| 11:33.21 | kaldemar | you can't plug a fax machine into a PRI. you need a channel bank in between if that's the setup you're aiming at. |
| 11:34.37 | kaldemar | an analog interface card is probably what you're looking for. |
| 11:36.37 | kaldemar | if you only have one A101D that's interfacing PSTN, then an analog card or an ATA is your only choice. might be an unreliable one though. i'd connect the fax machine directly to an analog line. |
| 11:40.01 | garymc | hmmm |
| 11:40.29 | garymc | im hearing conflicting advice im sure a few people said the only way to get 100% fax success was to install the card in the server |
| 11:44.18 | TommyBotten | Hmm.. the card itself does not help your fax syste |
| 11:44.18 | TommyBotten | m |
| 11:44.31 | TommyBotten | But it can be used to signal whatver analog/T.38 format you'd like |
| 11:44.39 | *** join/#asterisk soman (n=somnath@stargate.starnet.fi) |
| 11:45.06 | TommyBotten | Faxes may be achieved without analog or PRI for that matter. Using Sendfax() from spandsp and converting files to .tiff-files. |
| 11:45.28 | TommyBotten | And for receiving, using RecieveFax() |
| 11:55.53 | plundra | Is app_queue.so not really good at be reloading after you've changed queues.conf? |
| 11:56.01 | plundra | I mean, there is no "queue reload" :-) |
| 11:56.27 | plundra | Everything is working, but the stats isn't updated when doing "queue show" |
| 11:57.06 | kaldemar | there is a queue reload, but it requires a parameter. |
| 11:57.20 | kaldemar | core show help queue reload |
| 11:57.31 | plundra | Then I must be using an old version :-) |
| 11:57.54 | plundra | Was it added in 1.6.1.x or 1.6.2.x? |
| 12:00.03 | kaldemar | you can reload the module itself with module reload. |
| 12:00.24 | plundra | That is what I do now. |
| 12:00.43 | plundra | But I don't see the stats being updated, on number of calls processed or last call for each member. |
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| 12:31.24 | awk | hi guys... hmmm I have something i'm not understanding correctly', i'm trying to send out a call to a voip provider how do I change this WW-Authenticate: Digest realm="" ... the realm = part |
| 12:31.37 | awk | at the moment its showing my internal IP and thats the reason they saying its not working. |
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| 12:53.36 | ariel_ | Morning eveyone |
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| 13:23.39 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-40-42.acanac.net) |
| 13:26.32 | Pan3D | morning kids |
| 13:28.41 | razu | anyone using chan_ss7 here ? I have strange problem that when call goes through to a SIP account then call will be dropped the same second phone starts ringing. Seems like some RTP issue ? |
| 13:29.04 | *** join/#asterisk Jaipal (n=jai@122.169.251.179) |
| 13:30.06 | [TK]D-Fender | razu: Got a call with full debug for us to look at? |
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| 13:30.31 | *** part/#asterisk Jaipal (n=jai@122.169.251.179) |
| 13:30.35 | [TK]D-Fender | razu: Maybe some complete details as well... |
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| 13:30.52 | razu | [TK]D-Fender, a sec |
| 13:35.25 | razu | [TK]D-Fender, http://pastebin.com/d6b68a69b |
| 13:40.26 | [TK]D-Fender | razu: remove the core debug and jsut go with SIP debug & verbose 10 |
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| 13:46.54 | angryuser | Good day, i got asterisk 1.2 and 4 FXS port analog card, i just dont remember one stuff, in /etc/zaptel.conf i declared them as :fxoks and in zapata.conf as fxs_ks ? |
| 13:47.02 | angryuser | i declare* |
| 13:47.39 | angryuser | or ot is the same for zaptel and zapata :fxoks and fxo_ks ? |
| 13:48.35 | [TK]D-Fender | should be "fxoks" in zaptel, and "fxo_ks" in Zaata |
| 13:48.39 | [TK]D-Fender | Zapata |
| 13:49.24 | angryuser | kk |
| 13:49.38 | angryuser | Thank you, memory leak |
| 13:51.23 | [TK]D-Fender | angryuser: Better plug that up fast, our mop is already out on loan. |
| 13:52.45 | angryuser | i suppose its a ca humor |
| 13:53.09 | *** join/#asterisk squeeb (n=squeeb@office.slelectrotech.co.uk) |
| 13:53.57 | razu | [TK]D-Fender, http://web.razu.pri.ee/tmp/ss7.sipdebug.txt |
| 13:54.10 | squeeb | Allo, Having a small problem. I have a section of a dialplan that takes keypresses from customers, however I'm trying to use the 'i' extension to return them to the top of the section but instead it's dropping out the bottom of the section and then hanging up |
| 13:54.51 | razu | [TK]D-Fender, looking at the log there seems to be problem on SIP side ? |
| 13:56.56 | [TK]D-Fender | razu: [Jan 29 15:50:50] Reliably Transmitting (NAT) to 10.30.30.107:5060: CANCEL sip:6661022@10.30.30.107 SIP/2.0 |
| 13:57.26 | [TK]D-Fender | razu: caling end is aborting, but you dont have a timeout, the codec are agreed upon and included by default..... |
| 13:57.35 | [TK]D-Fender | razu: I didn't see an SS& error... |
| 13:57.45 | squeeb | also, do 'i' extensions work within included sections of a dialplan? |
| 13:57.46 | *** join/#asterisk moy (n=moy@74.12.123.169) |
| 13:57.56 | [TK]D-Fender | squeeb: yes |
| 13:58.18 | squeeb | because from doing 'core set verbose 10', I can see that the button pressed is outside of the include unless it's specifically assigned within the include |
| 13:58.38 | squeeb | IE, if 1 2 and 3 are defined in the included section, they work, if not, it drops out of the include onto the next in the default queue. |
| 13:58.43 | squeeb | not queue |
| 13:58.45 | squeeb | section * |
| 13:59.26 | *** part/#asterisk benngard (n=benngard@213.88.138.230) |
| 14:01.20 | squeeb | It won't do it |
| 14:01.26 | squeeb | no matter what I do it just falls through to the next include |
| 14:02.26 | squeeb | http://pastebin.com/m5801a3a6 |
| 14:02.32 | squeeb | This is correct for the include yea? |
| 14:02.44 | voipmonk | autofallthrough=no ? |
| 14:02.47 | squeeb | yep |
| 14:02.51 | squeeb | already checked that |
| 14:03.10 | voipmonk | ok can u show me some debug - If u sent the pastebin , i missed it -back from dog walk |
| 14:03.16 | razu | [TK]D-Fender, yes well ... even if I add timeout it doesn't change anything. also if I move the sip account to the same box |
| 14:03.42 | voipmonk | <PROTECTED> |
| 14:03.52 | squeeb | http://pastebin.com/m1c0258fc |
| 14:04.00 | squeeb | that's the debug |
| 14:04.01 | [TK]D-Fender | squeeb: that isn't even a whole context. I don't see REAL CODE, witha REAL CALL to debug |
| 14:04.45 | [TK]D-Fender | squeeb: there is no input shown in that CLI output. |
| 14:05.26 | [TK]D-Fender | squeeb: I see "0" called... how is that invlice? |
| 14:05.33 | squeeb | http://pastebin.com/m65ecc7b8 |
| 14:05.39 | *** join/#asterisk chilicuil (n=sistemas@unaffiliated/chilicuil) |
| 14:05.39 | squeeb | invlice? |
| 14:05.49 | razu | [TK]D-Fender, funny thing is if I answer the channel in some other box (or same box) ... everything works :) |
| 14:06.13 | *** join/#asterisk Da-Geek (n=Da-Geek@62.189.17.99) |
| 14:06.54 | [TK]D-Fender | squeeb: invalid |
| 14:07.46 | [TK]D-Fender | razu: I don't really get it myself.... |
| 14:08.12 | razu | [TK]D-Fender, ok ... I'll debug it further :) |
| 14:08.17 | [TK]D-Fender | razu: Idea : Answer the SS7 channel before calling out... |
| 14:08.36 | razu | that would be a workaround ... not a solution :) |
| 14:09.10 | squeeb | ok |
| 14:09.13 | squeeb | massive debug alert |
| 14:09.14 | squeeb | http://pastebin.com/m5d235b24 |
| 14:09.26 | squeeb | I called, it went to IVR, I pressed an invalid key, it dropped to DID_default |
| 14:10.48 | squeeb | line 371 looks a bit strange |
| 14:10.50 | squeeb | [Jan 29 14:07:52] DEBUG[25166]: pbx.c:2423 __ast_pbx_run: Oooh, got something to jump out with ('0') |
| 14:10.52 | [TK]D-Fender | squeeb: -- Executing [0@DID_trunks:1] Answer("SIP/7674126-f7a6ace8", "") in new stack <-- CLEARLY NOT INVALID |
| 14:11.04 | squeeb | yea but, I have exten => i set |
| 14:11.05 | squeeb | :) |
| 14:11.05 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:11.06 | [TK]D-Fender | squeeb: you HAVE a match. Please caffeinate |
| 14:11.19 | squeeb | where it matching if i haven't defined it anywhere? |
| 14:11.39 | [TK]D-Fender | squeeb: YOU DO |
| 14:11.54 | squeeb | I do what? |
| 14:12.12 | [TK]D-Fender | squeeb: THERE IS A GOD DAMNED MATCH LOOK AT THE DIALPLAN LINE ITS EXECUTING |
| 14:12.25 | [TK]D-Fender | squeeb: squeeb Clear now? |
| 14:12.34 | squeeb | No, and capitals aren't helping really |
| 14:12.58 | *** join/#asterisk ktwilight (n=keliew@51.183-241-81.adsl-dyn.isp.belgacom.be) |
| 14:13.04 | [TK]D-Fender | squeeb: You seem to be missing every other blatant statement, i was wondering it print size was a challenge |
| 14:13.23 | [TK]D-Fender | [09:11]<[TK]D-Fender>squeeb: you HAVE a match. Please caffeinate <- |
| 14:13.28 | [TK]D-Fender | [09:10]<[TK]D-Fender>squeeb: -- Executing [0@DID_trunks:1] Answer("SIP/7674126-f7a6ace8", "") in new stack <-- CLEARLY NOT INVALID |
| 14:13.35 | [TK]D-Fender | [09:11]<[TK]D-Fender>squeeb: YOU DO |
| 14:13.45 | squeeb | I appreciate you're trying to help me |
| 14:13.57 | squeeb | but please don't be an ass about it |
| 14:15.27 | Katty | dlynes_laptop: i got some reflective whiteness behind the camera today |
| 14:15.42 | m0t3jl | Is there a way to tell Asterisk not to store the voicmail messages, but only to send them via e-mail? |
| 14:16.01 | [TK]D-Fender | m0t3jl: delete=yes on the box definition. |
| 14:16.15 | m0t3jl | [TK]D-Fender, thx |
| 14:22.11 | [TK]D-Fender | razu: [Jan 29 15:31:21] DEBUG[15426]: channel.c:1462 ast_softhangup_nolock: Soft-Hanging up channel 'SS7/siuc/6' <-- yeah its clearly aborting the SS7 channel first |
| 14:23.02 | razu | [TK]D-Fender, mkay ... what could initiate that ? ... |
| 14:23.16 | [TK]D-Fender | razu: I really don't know enough about SS7 to venture a guess... |
| 14:23.18 | razu | [TK]D-Fender, as I see asterisk code is 111 which is protocol error ? |
| 14:23.42 | razu | possibly just linkset conf is bad |
| 14:24.29 | [TK]D-Fender | razu: "misc error" indeed. Not very helpful. |
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| 14:30.30 | Katty | oh man, it's gettin snowy outside. |
| 14:30.45 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:30.45 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:31.03 | Katty | dlynes_laptop: ping. |
| 14:31.28 | *** join/#asterisk TimeRider (n=steve@78.32.26.1) |
| 14:31.34 | Katty | hi TimeRider |
| 14:31.47 | TimeRider | hi |
| 14:31.53 | TimeRider | .. just passing by... |
| 14:31.58 | [TK]D-Fender | katyWe got ass-raped last night. We went from 45F to 0F f-ing fast |
| 14:32.07 | [TK]D-Fender | Katty: rather |
| 14:32.29 | TimeRider | omg, did I speak to a bot? |
| 14:32.41 | TimeRider | .. ya never know these days! |
| 14:32.58 | Katty | yes, i'm a bot. |
| 14:33.04 | TimeRider | now I don't believe ya, lol |
| 14:33.10 | Katty | [TK]D-Fender: did you get any snow dumped on you? |
| 14:33.31 | [TK]D-Fender | Katty: Yup, and driver's spines went out the door immediately... |
| 14:33.48 | Katty | photo! |
| 14:34.47 | [TK]D-Fender | Katty: maybe later... |
| 14:35.29 | Katty | :< |
| 14:37.00 | Katty | http://i.imgur.com/BQofF.jpg <- wow. |
| 14:37.12 | Katty | and the cat seems oddly... calm |
| 14:41.11 | dlynes_laptop | Katty, ? |
| 14:41.42 | Katty | dlynes_laptop: i added a white background to the camera (= |
| 14:41.52 | Katty | dlynes_laptop: do you think it looks less washed out? |
| 14:42.09 | dlynes_laptop | Katty, holy cow....all those badgers...is that your backyard? |
| 14:42.25 | Katty | no, that was on reddit this morning |
| 14:42.30 | dlynes_laptop | oh |
| 14:42.36 | Katty | but i do have four very handsome squirrels in the yard. |
| 14:42.40 | dlynes_laptop | ~crittercam |
| 14:42.41 | infobot | methinks crittercam is Katty's broadcast of The Nut House Critter Cam @ http://ustre.am/8H5d and The Nut House Bird Bath @ http://ustre.am/bEBU |
| 14:43.05 | Katty | i think the wireless adaptor must have fallen behind the couch because the fps are rather poor |
| 14:43.27 | Katty | ah, 6 squirrels this morning |
| 14:43.34 | dlynes_laptop | I really hate whatever ustream did to their website |
| 14:43.49 | Katty | well |
| 14:43.52 | Katty | i can embedd it into my blog |
| 14:43.53 | dlynes_laptop | guaranteed within 2 seconds, my browser completely locks up on their website now |
| 14:43.54 | coppice | Katty: is that your usual breakfast? |
| 14:44.18 | Katty | dlynes_laptop: let me in embed it, sec. |
| 14:44.40 | eppigy | NEIN |
| 14:45.39 | Katty | dlynes_laptop: 42ndgeekstreet.blogspot.com |
| 14:46.25 | Katty | hi eppigy |
| 14:46.34 | dlynes_laptop | Katty, yeah...not locking up now |
| 14:46.38 | dlynes_laptop | Katty, must be their website |
| 14:47.10 | Katty | bummer. |
| 14:47.14 | dlynes_laptop | Katty, but yeah...the fps really sucks, but with that white background, it really fixes up the contrast of the background so you can see the tree better |
| 14:47.29 | Katty | yeah i'm guessing my little usb thing fell behind the couch |
| 14:47.48 | *** join/#asterisk Akiraa (n=Akiraaaa@79.112.17.3) |
| 14:48.21 | Katty | or there's some other process hogging resources on that lil workstation |
| 14:48.44 | coppice | there are millions of little USB things down the backs of couches. many of them memory sticks with the homework that was not eaten by the dog |
| 14:49.04 | Katty | hehe |
| 14:49.15 | Katty | my couch was partially eating by the dog |
| 14:49.21 | Katty | s/eating/eaten/ |
| 14:49.51 | Katty | riddick had gotten upstairs at there was this awful noise so i went to check it out.... he was dragging the couch across the room. |
| 14:50.03 | Katty | which, i must say, is quite a display for a 3 month old puppy |
| 14:50.12 | coppice | Katty: what about your homework? |
| 14:50.18 | *** join/#asterisk slidesinger (n=slidesin@c-68-44-99-50.hsd1.nj.comcast.net) |
| 14:50.22 | Katty | i don't do homework |
| 14:50.24 | Katty | i did my time |
| 14:51.23 | eppigy | hi Katty |
| 14:51.25 | voipmonk | Katty - tell the truth - you're keeping baby dino's in the house |
| 14:51.30 | voipmonk | the couch? |
| 14:51.32 | voipmonk | puppy? |
| 14:51.36 | *** join/#asterisk Yuy (n=tarnok@dsl-67-204-56-88.acanac.net) |
| 14:51.38 | voipmonk | who you foolin' ? :) |
| 14:51.51 | Katty | yes, i totally am keeping a trex in the house. |
| 14:51.54 | *** join/#asterisk chilicuil (n=sistemas@unaffiliated/chilicuil) |
| 14:52.03 | voipmonk | what does this puppy eat? |
| 14:52.12 | coppice | trex? isn't that a cooking fat? |
| 14:52.16 | Katty | short of everything i give him? |
| 14:53.11 | Katty | voipmonk: it's called Diamond's Naturals |
| 14:54.11 | Katty | voipmonk: riddick has allergies to corn, wheat, soy, or a combination of those three. |
| 14:54.31 | Katty | voipmonk: so i found a dog food at buchietts which is chicken based with no corn wheat or soy (= |
| 14:54.37 | Katty | it's chicken and rice or something |
| 14:55.42 | Katty | i wonder how fast a trex could have ran |
| 14:55.58 | Katty | googles |
| 14:56.07 | Katty | bah, 20mph |
| 14:56.13 | *** part/#asterisk slidesinger (n=slidesin@c-68-44-99-50.hsd1.nj.comcast.net) |
| 14:56.23 | coppice | trex is a tub of lard. it doesn't run at all. cf couch potato |
| 14:56.48 | Katty | do you know what you can do with a pound of lard? |
| 14:56.53 | Katty | you can melt it, in a big pan. |
| 14:57.05 | Katty | and then mix bird seed, oatmeal, cornmeal, dried fruit, and other goodies into it |
| 14:57.06 | coppice | make dumplings, or elect it |
| 14:57.08 | Katty | and hang it up for the birds. |
| 14:57.31 | Katty | yeah...don't eat things fried in lard. |
| 14:57.40 | Katty | that's just bad for you. really bad. |
| 14:58.08 | coppice | electing tubs of lard is bad for us, but that's what many countries do |
| 14:59.04 | Katty | http://i.imgur.com/NTGTq.jpg <- |
| 14:59.39 | Katty | ^- lard side effect |
| 15:01.53 | dlynes_laptop | Katty, wtf? |
| 15:02.47 | Katty | dlynes_laptop: hmm? |
| 15:02.55 | dlynes_laptop | Katty, thought you were going to show us someone with lard and drool dripping down their chin, or something |
| 15:03.05 | Katty | -_^ |
| 15:03.41 | Katty | mmmmmmmmno. |
| 15:03.52 | Katty | so the orange slices were not a big hit. |
| 15:04.00 | Katty | the peach went over okay |
| 15:04.04 | Katty | and the banana was devoured. |
| 15:05.11 | Katty | http://bitsandpieces.us/wp-content/uploads/2010/01/imagescomputer_engineer_help_wanted_sign_small.jpg <- cute idea, but far too easy. |
| 15:06.03 | angryuser | Hello i have one Zap channel stuck in Rsrvd State, zap show channel 34 is here >http://pastebin.com/m3ab22cf7 it is stuck bridged, any ideas ? asterisk 1.2 zap 1.2 |
| 15:06.22 | Katty | wow, 1.2 |
| 15:06.28 | Katty | haven't see that used in awhile |
| 15:06.29 | *** join/#asterisk jmayorga5_ (n=jmayorga@adsl-99-91-103-186.dsl.ipltin.sbcglobal.net) |
| 15:06.39 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 15:07.06 | Katty | hi jaytee |
| 15:07.11 | angryuser | Katty, many people still use 1.2 as they dont need much new stuff |
| 15:07.18 | jaytee | hi Katty |
| 15:07.28 | Katty | jaytee: you have snow up north? |
| 15:08.49 | jaytee | many people use 1.2 still because they have third party addins and customization that won't port to 1.4 or 1.6 and either can't afford the cost/time to upgrade or in a few rare instances are just too damn lazy |
| 15:09.46 | Katty | it's just odd to see it mentioned. |
| 15:10.06 | Katty | http://i.imgur.com/RwSQk.jpg <- *hee* "wear particle mask" |
| 15:10.07 | *** part/#asterisk fiddur (n=fiddur@192.121.104.121) |
| 15:10.21 | *** join/#asterisk UserReg_CL (n=COB@pc-135-211-30-200.cm.vtr.net) |
| 15:10.44 | Katty | wow, siberia has 40ft snow drifts. |
| 15:12.24 | coppice | Katty: how do you know the last user didn't just die of ebola? |
| 15:12.48 | Poincare | russellb: regarding that operator telling me iax is insecure, I thing he's getting ddos attacks and just needs something to blame |
| 15:13.43 | Katty | coppice: i'm....not following you |
| 15:14.05 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 15:14.28 | coppice | Katty: some notebooks really are dangerously infected |
| 15:14.49 | voipmonk | well |
| 15:14.54 | voipmonk | iax is insecure |
| 15:14.57 | Katty | i'm so completely lost |
| 15:14.57 | *** join/#asterisk af_ (n=getsmart@88-149-240-203.dynamic.ngi.it) |
| 15:15.10 | voipmonk | any asterisk system running iax can be shut down |
| 15:15.10 | Katty | either i'm not awake, or it's just going over my head |
| 15:15.14 | voipmonk | digium knows this but refuses to fix it |
| 15:15.21 | eppigy | I am eating avocado, canned chicken, and yogurt |
| 15:15.29 | Naikrovek | voipmonk: explain |
| 15:15.30 | coppice | Katty http://i.imgur.com/RwSQk.jpg <- *hee* "wear particle mask" |
| 15:15.42 | Katty | ohhh |
| 15:15.44 | Katty | ha |
| 15:15.54 | Katty | brb, gotta move mah car |
| 15:16.25 | drfreeze | Morning |
| 15:17.09 | drfreeze | Anyone built an * system using a solid-state drive? I'm wondering how long one would last, especially if you were doing call monitoring |
| 15:19.39 | voipmonk | yeah... |
| 15:21.29 | Katty | MFin BRRRR |
| 15:21.58 | *** join/#asterisk blkry (n=chatzill@64.147.222.130) |
| 15:22.00 | Katty | drfreeze: better not take it through the airport ;P |
| 15:22.24 | drfreeze | Katty: :), why? |
| 15:22.33 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 15:22.35 | Katty | facepalms. |
| 15:22.38 | drfreeze | are solid state drives explosive? |
| 15:22.44 | Katty | apparently. |
| 15:22.49 | drfreeze | BTW, I am flying today |
| 15:23.02 | voipmonk | where to |
| 15:23.03 | Katty | or at least, one air port security thought it was. |
| 15:23.07 | drfreeze | They always have a problem with the maginifying glass I carry in my laptop case. |
| 15:23.12 | *** join/#asterisk phl4kx (n=phl4kx@mail001.centralcorp.upnorte.edu.pe) |
| 15:23.18 | *** join/#asterisk casix (n=casix@xenpbxedifici.adamvozip.es) |
| 15:23.19 | Katty | you might start a fire with it |
| 15:23.36 | drfreeze | It does funny things to the xray I think |
| 15:23.56 | Katty | good possiblity |
| 15:23.57 | drfreeze | it goes thru with no problems about 1 in 20 times |
| 15:24.05 | coppice | drfreeze: is it lead crystal? |
| 15:24.14 | drfreeze | I think |
| 15:24.21 | Katty | ah, lead. |
| 15:24.52 | coppice | lead crystal comes out totally black on those X-ray machines. denser than steel |
| 15:28.53 | Katty | eppigy: did you save me any avacado? :> |
| 15:31.40 | Akiraa | This makes me wonder: can you reasonably hope to haul important electronics through US airports these days? |
| 15:32.28 | Katty | nope |
| 15:34.22 | *** join/#asterisk benngard (n=benngard@90-230-92-67-no148.tbcn.telia.com) |
| 15:34.32 | eppigy | Katty: I have several at home |
| 15:35.19 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 15:35.24 | *** join/#asterisk The_Boy_Wonder (n=vossel@asterisk/batman-developer/dvossel) |
| 15:35.35 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 15:35.45 | Zeeek | Hellooooooo |
| 15:35.51 | eppigy | HI |
| 15:36.01 | Zeeek | It's beer o'clock over here in Euroland |
| 15:36.15 | ariel_ | Akiraa: I hope so, I just took some testing electronic equipment through the airport. A T1 test unit, a Wireless signal test system as well |
| 15:36.32 | Katty | herroes. |
| 15:37.37 | *** join/#asterisk RobH (n=robh@rob.tech.wikimedia.org) |
| 15:37.46 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:37.46 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:37.54 | coppice | taking terminal blocks through airports in your luggage is fun. they *always* make you open your bags. the terminals look like bullets :-) |
| 15:38.28 | Katty | 'terminal blocks'? |
| 15:38.39 | ariel_ | your correct, I had to turn on the testers for them to see that there worked |
| 15:38.44 | leifmadsen | you're* |
| 15:39.21 | ariel_ | English teacher |
| 15:39.26 | leifmadsen | it helps to make a difference if you say: you-er vs. yor |
| 15:39.35 | leifmadsen | your and you're shouldn't even sound the same |
| 15:39.40 | Katty | oh, that |
| 15:39.53 | Katty | i thought they were called 66 blocks |
| 15:40.01 | coppice | in days of yore the apostrophe was optional |
| 15:40.07 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 15:40.11 | leifmadsen | coppice: :) |
| 15:40.11 | Katty | hi riddlebox |
| 15:40.16 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 15:40.49 | riddlebox | hey |
| 15:41.18 | p3nguin | wonders what happened to the big bad storm we should have received. |
| 15:42.17 | ariel_ | I think you can say that weather is and has always been very un predictable, oh and since Al Gore we also can now always blame it on Global Warming. Even if it's cold. |
| 15:42.17 | Katty | p3nguin: well... |
| 15:42.19 | Katty | p3nguin: it slowed down |
| 15:42.25 | Katty | p3nguin: we're just now getting hit, down south |
| 15:42.38 | Katty | p3nguin: you can have a look on crittercam at the snow (= |
| 15:43.04 | p3nguin | katty: It'll probably crash my browser. |
| 15:43.17 | *** join/#asterisk timeshell (n=chatzill@gw.lusi.on.ca) |
| 15:43.21 | timeshell | Greetings |
| 15:43.35 | Katty | p3nguin: 42ndgeekstreet.blogspot.com |
| 15:43.39 | Katty | p3nguin: i embedded it for dlynes_laptop |
| 15:44.10 | timeshell | Here's a question: Is there a module for asterisk that would allow for one to use an incoming channel as a dialup ppp connection? |
| 15:44.24 | timeshell | specifically an incoming SIP channel |
| 15:44.25 | voipmonk | wow |
| 15:44.31 | voipmonk | ppp over sip |
| 15:44.36 | voipmonk | seriously? |
| 15:44.45 | voipmonk | that would suck |
| 15:44.49 | timeshell | lol |
| 15:44.51 | timeshell | Yah it would |
| 15:44.54 | voipmonk | look |
| 15:44.58 | voipmonk | im about to show my age here |
| 15:45.03 | Katty | uh oh |
| 15:45.04 | voipmonk | do some research on slirp |
| 15:45.05 | Katty | sells tickets |
| 15:45.06 | voipmonk | or tia |
| 15:45.26 | timeshell | slip is pre ppp |
| 15:45.30 | timeshell | I'm familiar with slip |
| 15:45.36 | voipmonk | slirp |
| 15:45.43 | voipmonk | its old skewl |
| 15:45.52 | timeshell | Ok, and why may I ask? |
| 15:45.54 | voipmonk | you'll have more luck |
| 15:46.08 | voipmonk | ppp over sip is going to bite worse than faxing over sip |
| 15:46.26 | ariel_ | what is ppp or is this what you are calling RAS |
| 15:46.28 | timeshell | Really? |
| 15:46.39 | timeshell | Why would it be worse that faxing? |
| 15:46.59 | timeshell | If the sampling is greater than the modem speed, it should be relatively stable shouldn't it? |
| 15:47.15 | coppice | continuous modem operation is far more quirky than the fairly short bursts used for FAXing |
| 15:47.31 | timeshell | But I guess the question still is, how would one integrate that into asterisk... direct an incoming call over SIP to a ppp/slip/slirp? |
| 15:48.04 | timeshell | Additionally, I don't think I'm looking for faster than 14400 for a modem connection. |
| 15:48.05 | *** join/#asterisk MAbbas (i=Jinbaba@115.186.24.157) |
| 15:48.13 | timeshell | Not 57600 |
| 15:48.30 | p3nguin | katty: Here's what the radar looked like at 1:30. We should have been snowed upon, according to it. :/ http://imagebin.org/82409 |
| 15:48.39 | MAbbas | Hi everyone, how can I get channel variables in AGI script? |
| 15:49.12 | timeshell | thinking... "I suppose iaxmodem would do it..." |
| 15:49.15 | MAbbas | I tried.. AGI Rx << get variable DIALSTATUS |
| 15:49.16 | MAbbas | AGI Tx >> 200 result=0 |
| 15:49.20 | coppice | timeshell: why do you need that? I am genuinely interested in the kinds of apps people still have for modems |
| 15:49.25 | voipmonk | good lord |
| 15:49.27 | *** join/#asterisk asteriskATmarmuD (n=mundt@193.158.65.23) |
| 15:49.36 | asteriskATmarmuD | hi there |
| 15:49.50 | timeshell | coppice Only for the rare occasion that I don't have a suitable internet connection and I'm desperate for one. |
| 15:49.56 | timeshell | My incoming trunks are all SIP |
| 15:50.12 | ariel_ | coppice: I can give you one, HOA have to communicate to the gates and most only do it via modem 9600 bps |
| 15:50.26 | asteriskATmarmuD | anyone experienced in using digium tdm410 |
| 15:50.36 | coppice | one interesting use is dialing into the VoIP server to fix it :-) |
| 15:50.41 | ariel_ | asteriskATmarmuD: just ask your question |
| 15:51.07 | asteriskATmarmuD | ok, I installed the card correctly, but I got no dial tone on the connected phone |
| 15:51.20 | coppice | ariel_: HOA? |
| 15:51.26 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 15:51.32 | Katty | p3nguin: hmm, yeah |
| 15:51.44 | ariel_ | Home Owers Assosiation, they control our gates |
| 15:52.00 | timeshell | coppice... Would sort of defeat the purpose since the only access to the server would be over the internet anyway |
| 15:52.02 | Katty | p3nguin: at least we're not in the ice regon |
| 15:52.16 | coppice | ariel_: ah. do you know what kind of modem they use? |
| 15:52.26 | fatnasty1 | MAbbas: be carefull using agi, it can cause terrible sound quality. use fastagi instead. |
| 15:52.33 | timeshell | coppice All the trunks are over the internet. The dialup would be over a SIP trunk over the internet. |
| 15:52.47 | ariel_ | most of them use standard modems but at 9600 baud |
| 15:53.22 | coppice | ariel_: most security stuff uses really slow modems. |
| 15:53.59 | ariel_ | yes for some reason or another they still use old stuff |
| 15:54.12 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 15:54.30 | fatnasty1 | MAbbas: Or better yet use AMI instead of AGI. |
| 15:54.58 | [TK]D-Fender | AGI has nothing to do with audio quality |
| 15:55.04 | ariel_ | they have the phone lines on there mb and use it for inbound calls to configure the system like alarms systems and also use the line for when someones gets to the gate and dials an owner to let them in |
| 15:55.12 | MAbbas | fantasy1: i am using FastAGI |
| 15:55.15 | coppice | ariel_: security is not an industry that moves very quickly |
| 15:55.20 | [TK]D-Fender | Where do people come up with this stuff? |
| 15:55.51 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:56.42 | p3nguin | katty: I loaded your embedded video page; then I pressed the play button on the image. It loaded a new browser window to show me the video in your original ustream page... but it didn't crash the browser! |
| 15:56.46 | Katty | ryan just called me--he thinks i should come home :< |
| 15:57.03 | Katty | p3nguin: that is just...crazy |
| 15:57.14 | *** join/#asterisk Daviey (n=Daviey@ubuntu/member/pdpc.gold.Daviey) |
| 15:57.36 | p3nguin | katty: If I press the play button on the lower left corner, it plays in your embedded page. |
| 15:58.21 | Katty | tricksy |
| 15:59.30 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 15:59.56 | MAbbas | here is my dialplan entry, exten => _80X,1,AGI(agi://10.110.32.44:4572) |
| 16:00.16 | MAbbas | and in agi script I do AGI Rx << EXEC Dial Agent/201 |
| 16:00.16 | MAbbas | AGI Tx >> 200 result=-1 |
| 16:00.25 | fatnasty1 | [TK]D-Fender: Thats where your wrong |
| 16:00.32 | MAbbas | I want to get DIALSTATUS variable? |
| 16:01.36 | [TK]D-Fender | fatnasty1: AGI itself doesn't stream audio, it is an interface to issue commands that * itself will execute |
| 16:02.02 | [TK]D-Fender | fatnasty1: And it's "you're" |
| 16:02.04 | fatnasty1 | [TK]D-Fender: we tried running AGI on every inbound call on an Asterisk VM platform, spwaning a seperate proccess many times a second can make the audio sound like shit. |
| 16:02.26 | [TK]D-Fender | fatnasty1: thats a server load issue, not the mere fact that its AGI |
| 16:02.43 | [TK]D-Fender | fatnasty1: Smaller load = jsut fine |
| 16:03.06 | MAbbas | fantasy1: the thing is its something to do with asterisk handling all the load .. [TK]D-Fender said is absolutly right |
| 16:03.14 | [TK]D-Fender | fatnasty1: that's like saying a station wagon is slower than a bicycle... just because there is a traffic jam. |
| 16:03.36 | fatnasty1 | [TK]D-Fender: well whatever dude, dont use AGI if you dont have to. |
| 16:03.45 | fatnasty1 | use AMI |
| 16:03.52 | [TK]D-Fender | fatnasty1: I don't |
| 16:03.53 | MAbbas | wen we use FastAGI we basically transfer the load to AGI server .. |
| 16:04.03 | [TK]D-Fender | fatnasty1: And AMI and AGI have nothing to do with each other. |
| 16:04.13 | [TK]D-Fender | fatnasty1: Stop comparing apples & oranges |
| 16:04.31 | fatnasty1 | [TK]D-Fender: you can accomplish many of the same tasks, |
| 16:04.44 | [TK]D-Fender | fatnasty1: AGI is for dialplan processing. AMI is for issuing other random functions not related to the current channel (sine it isn't frm a channel |
| 16:05.15 | *** join/#asterisk _abc_ (n=no@unaffiliated/ccbbaa) |
| 16:05.21 | fatnasty1 | [TK]D-Fender: I bet its hard to hang out with you. |
| 16:05.52 | fatnasty1 | [TK]D-Fender: and thats all i have to say about that.............. |
| 16:06.09 | [TK]D-Fender | fatnasty1: You are comparing two completely different things and trying to shift your way out of it. |
| 16:06.24 | MAbbas | [TK]D-Fender: here is my dialplan entry, exten => _80X,1,AGI(agi://10.110.32.44:4572) |
| 16:06.24 | MAbbas | and in agi script I do AGI Rx << EXEC Dial Agent/201 |
| 16:06.24 | MAbbas | AGI Tx >> 200 result=-1 |
| 16:06.24 | MAbbas | I want to get DIALSTATUS variable? |
| 16:06.26 | [TK]D-Fender | fatnasty1: And now a third. |
| 16:06.43 | [TK]D-Fender | MAbbas: I don't know... DO you? |
| 16:07.07 | fatnasty1 | [TK]D-Fender: You are an arrogant confrontational individual. |
| 16:07.14 | _abc_ | feels that he joined in the middle of an odd discussion |
| 16:07.17 | [TK]D-Fender | MAbbas: Do you expect me to knwo your wants? the dial command was accepted. if your AGI continues they chances are the call didn't go through |
| 16:07.20 | _abc_ | or several |
| 16:07.49 | [TK]D-Fender | fatnasty1: If by "confront" you mean "willing to point out that advice you are giving here to people is wrong", then yes |
| 16:08.02 | fatnasty1 | [TK]D-Fender: in your opinion |
| 16:08.18 | [TK]D-Fender | fatnasty1: You seem to take everything contradictory to what you are saying as a personal attack. This is problematic |
| 16:08.44 | Zeeek | [TK]D-Fender I resemble that remark!!! |
| 16:08.46 | MAbbas | I am asking how can I get DIALSTATUS of last Dial() in my AGI script? |
| 16:08.50 | [TK]D-Fender | Zeeek: :) |
| 16:08.55 | fatnasty1 | [TK]D-Fender: If you run an agi script on every call you will spawn a seperate process on every call, this can and will cause shitty audio. |
| 16:09.02 | [TK]D-Fender | MAbbas: its a variable like any other. Go get it |
| 16:09.06 | Zeeek | seperate is spelled separate |
| 16:09.18 | fatnasty1 | [TK]D-Fender: also VIOP is part of the PSTN. |
| 16:09.35 | [TK]D-Fender | fatnasty1: if I have 2 calls going on in my server at any one time on a quad-core system with 4 gigs I will get shitty audio? |
| 16:10.06 | MAbbas | this is what I get after DIAL() is finished AGI Rx << get variable DIALSTATUS |
| 16:10.07 | MAbbas | AGI Tx >> 200 result=0 |
| 16:10.15 | fatnasty1 | [TK]D-Fender: Nope, but that is an unrealistic situation. |
| 16:10.16 | MAbbas | i.e. variable is not set .. |
| 16:10.31 | [TK]D-Fender | fatnasty1: No, its perfectly realistic. I don't get that many calls. |
| 16:10.31 | voipmonk | ? |
| 16:10.37 | Kobaz | how would i force a registration entry in asterisk |
| 16:10.38 | voipmonk | wears a question mark on his face |
| 16:10.44 | [TK]D-Fender | fatnasty1: You are casting aspersions as to my requirements. |
| 16:10.53 | Kobaz | if i do a database put on SIP/registry, sip show peers doesn't have the ip |
| 16:10.57 | fatnasty1 | [TK]D-Fender: Then why would you have purchased such an over speced server? |
| 16:11.06 | Kobaz | SIP/Registry rather |
| 16:11.25 | [TK]D-Fender | fatnasty1: Quad core systems with 4 gig is a common DESKTOP these days |
| 16:11.31 | _abc_ | so what can cause large banging noises in sip connections? what is the normal way in which network load and drop-outs manifest themselves? |
| 16:11.35 | [TK]D-Fender | fatnasty1: And I could get away with far less. |
| 16:11.51 | voipmonk | does that matter fatnasty1 ? what matters is you're getting shitty audio - agi and shitty audio go together like... pump gas and peanut butter and jelly |
| 16:12.22 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 16:12.29 | [TK]D-Fender | fatnasty1: 10 AGI's in swing at a time... how big a box you do you really thing I need? You also didn't quanlify as to the actual load the AGI's would be placing due to their function. Are your AGI's doing more than mine? |
| 16:12.42 | [TK]D-Fender | fatnasty1: You have no substance for comparison. |
| 16:12.56 | [TK]D-Fender | fatnasty1: Instead have jumped to a wholesale conclusion. |
| 16:13.05 | voipmonk | set the scene for me fatnasty1 , you're getting audio quality issues - whats in the mix, what are you doing to make your calls? ? |
| 16:13.17 | fatnasty1 | Well use AGI all you want then. But when you run a script on every call, a separate process will spawn on every call, and this can and does become problematic. |
| 16:13.23 | *** join/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 16:13.33 | _abc_ | so what can cause large banging noises in sip connections? what is the normal way in which network load and drop-outs manifest themselves? |
| 16:13.41 | [TK]D-Fender | voipmonk: AGI is nasty apparently. Not the fact that is running a 5 million record SQl query for each on 200 simultaneaous calls to the same AGI per second. |
| 16:13.42 | fatnasty1 | Doesnt matter what you THINK about it. I have SEEN it happen |
| 16:14.25 | fatnasty1 | [TK]D-Fender: yout 5 million record query is 1 process, thats not the point. |
| 16:14.29 | MAbbas | <fatnasty1>: does same thing happen if you use FastAGI? |
| 16:14.38 | Skeeter- | anyone can tell me about Openvox product??? |
| 16:14.41 | voipmonk | what is the point? |
| 16:14.51 | fatnasty1 | MAbbas: I havent tried it yet, but I assume not as it sends the load to a different server. |
| 16:14.56 | [TK]D-Fender | fatnasty1: AGI isn't bad. The scale of how many, and what they are doing is your problem. No-one else's. |
| 16:15.19 | [TK]D-Fender | Skeeter-: Avoid. Cusctom support = bad, and are based on old Digium designs |
| 16:15.33 | fatnasty1 | voipmonk: If you run an AGI script on every call, and your system gets alot of calls, you will run into audio problems. |
| 16:15.42 | Skeeter- | [TK]D-Fender, thanks, i saw the look like product of digium |
| 16:15.59 | Skeeter- | [TK]D-Fender, i only used Sangoma so far, do you suggest anything else?? |
| 16:16.05 | MAbbas | <[TK]D-Fender>: lets say, If I open 100 FastAGI connections at a time .. does that posses any problem on * performance(sound etc)? |
| 16:16.13 | [TK]D-Fender | [11:14]<fatnasty1>[TK]D-Fender: yout 5 million record query is 1 process, thats not the point. <- Yes... PER CALL. That's the point. after 200 simultaneous calls your server load will skyrocket |
| 16:16.20 | fatnasty1 | MAbbas: hell yes it will. |
| 16:16.43 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:16.48 | [TK]D-Fender | 11:15]<fatnasty1>voipmonk: If you run an AGI script on every call, and your system gets alot of calls, you will run into audio problems. <- again unqualified because you don't specify a fixed # of calls, and don't have a server spec to compare it against |
| 16:16.50 | voipmonk | fatnasty1: I can see that working out if you arent aware of how to build your system to scale, like not using agi for 1m minutes/day on an under powered single system. |
| 16:17.27 | [TK]D-Fender | fatnasty1: "Your server can't handle 200 calls at a time! --- "You don't know what I'm running, so stop commenting". |
| 16:17.50 | voipmonk | how many calls is a lot? |
| 16:17.51 | fatnasty1 | Just kidding AGI is awesome I try to spawn as many processes as I can with every call, it can't cause a problem. |
| 16:18.09 | fatnasty1 | Everyone should use it alot, especially you MAbbs. |
| 16:18.10 | [TK]D-Fender | fatnasty1: What part of "the numbers matter" are you failing to understand? |
| 16:18.10 | Kobaz | [TK]D-Fender: any idea? my sip master? |
| 16:18.12 | *** join/#asterisk grEvenX (n=even@cC0FD00C3.dhcp.bluecom.no) |
| 16:18.30 | [TK]D-Fender | fatnasty1: Yes AGI places a load. How much VS what you can handle is the issue and the details matter. ALL of the details. |
| 16:18.44 | fatnasty1 | i agree with everything Tk says, he is right. |
| 16:18.48 | voipmonk | force registration entry in asterisk by ........ adding a register line in sip.conf |
| 16:19.04 | voipmonk | Kobaz |
| 16:19.14 | Kobaz | no, the opposite |
| 16:19.23 | voipmonk | ? |
| 16:19.35 | voipmonk | am I on Punk'd today? |
| 16:19.36 | Kobaz | make asterisk be aware of a phone at ip x |
| 16:19.38 | MAbbas | fatnasty1: Sarcasm? |
| 16:19.44 | Kobaz | without having the phone directly register with it |
| 16:19.53 | fatnasty1 | Nope, use AGI. I was just kidding. |
| 16:19.57 | [TK]D-Fender | MAbbas: How about you take a step back and explain what you actually need to DO. |
| 16:20.14 | voipmonk | host = ip in the peer stanza in sip.conf, Kobaz |
| 16:20.18 | [TK]D-Fender | Kobaz: Aware in what way? |
| 16:20.33 | *** join/#asterisk bmoraca_work (n=bmoraca@66-242-174-254.ceres.bvn.net) |
| 16:20.34 | Kobaz | [TK]D-Fender: as if the phone registered with it |
| 16:20.39 | voipmonk | Kobaz: whats your sip peer look like in sip.conf? |
| 16:20.47 | Kobaz | voipmonk: dynamic ips |
| 16:20.58 | Kobaz | although i'm seriously considering making all the phones static ips via dhcp |
| 16:20.59 | [TK]D-Fender | Kobaz: Please rephrase from the top.... |
| 16:20.59 | voipmonk | host = dyndns name then |
| 16:21.02 | MAbbas | <[TK]D-Fender> well, for each of my call I spawan a FastAGI connection which uses Dial() to connect call to agent |
| 16:21.14 | voipmonk | but it wont be 'registered' |
| 16:21.19 | [TK]D-Fender | MAbbas: What does FastAGI do for you exactly? |
| 16:21.33 | fatnasty1 | Fact is there is no doubt TK knows more about * than I do. But, I will tell you that I have had problems with AGI on large servers with heavy call volume, this is a fact. |
| 16:21.47 | Kobaz | [TK]D-Fender: i have two asterisk boxes, for failover... it's a simple failover setup, machines A and B are identical, physical switch on a t1 and network interface... both servers have an ip of 192.168.24.12 on the 'shared' network interface |
| 16:21.49 | MAbbas | <[TK]D-Fender>: connects incoming call to agent of my choice |
| 16:22.00 | voipmonk | what is heavy call volume? what processor were you using? what were you doing in the agi for every call? |
| 16:22.07 | Kobaz | [TK]D-Fender: so when i flip from one server to another, if a call comes in, asterisk doesn't know where the phone is until they reregister |
| 16:22.16 | [TK]D-Fender | MAbbas: if all AGI is doing is issuing the Dial command I fail to see what you're using an AGI. you could jsut issue that dial directly in the dialplan. |
| 16:22.23 | [TK]D-Fender | MAbbas: What ELSE is that AGI doing? |
| 16:22.42 | voipmonk | tying shoes |
| 16:22.48 | voipmonk | making sandwhiches.. |
| 16:22.55 | voipmonk | babysitting the kids... |
| 16:23.13 | voipmonk | playing traffic cop |
| 16:23.16 | MAbbas | in AGI server I basically use some hueristic to come up with best Agent match for incoming call .. |
| 16:23.39 | fatnasty1 | voipmonk: probably 40 calls a min or so. most very short durration, this was on an HP DL360 2 proc, lots of ram, 8 gigs or so. the agi script was running a mysql query and setting an * variable based on the result. |
| 16:24.10 | MAbbas | <[TK]D-Fender>: What it basically is doing .. overriding Queue policy(roundrobin, etc .. ) |
| 16:24.13 | [TK]D-Fender | [11:21]<fatnasty1>Fact is there is no doubt TK knows more about * than I do. But, I will tell you that I have had problems with AGI on large servers with heavy call volume, this is a fact. <- so far you haven't given details on the number of calls, the actual function of your AGI's, how long a channel sits in an AGI call as opposed to returning to the dialplan, or the specs of your server,... |
| 16:24.15 | [TK]D-Fender | ...or call-rate. These would be "facts" You have not given us facts. You have given us a summary conclusion. AGAIN |
| 16:24.40 | voipmonk | there's all kinds of issues with that,, most noobs open a connection for every call, fatnasty1 - im willing to bet money that thats what you were doing :) |
| 16:24.58 | [TK]D-Fender | MAbbas: What does it do after the DIAL Agent/XXX ends? |
| 16:25.27 | MAbbas | <[TK]D-Fender>: Just that nothing else .. |
| 16:25.40 | bmoraca_work | fatnasty1: i'd imagine that func_odbc and a stored procedure would be infinitely more efficient than launching AGI and running through 2-3 more layers to get there |
| 16:26.07 | MAbbas | <[TK]D-Fender>: But the problem is I have to know, what was the result of Dial() |
| 16:26.22 | fatnasty1 | voipmonk: probably |
| 16:26.23 | [TK]D-Fender | MAbbas: then you should not be issuing the DIAL inside of the AGI. you should exit the AGI the moment you have the information it needs to lookup and you should let the DIALPLAN issue the dial. this way you will have NO AGI in memory during an active call. |
| 16:26.48 | [TK]D-Fender | MAbbas: You could eliminate this load almost instantly. |
| 16:27.26 | MAbbas | <[TK]D-Fender>: how do I tell dialplan to Dial() agent of my choice? |
| 16:27.32 | voipmonk | http://www.merriam-webster.com/dictionary/heuristic |
| 16:27.53 | [TK]D-Fender | MAbbas: .........you're telling AGI to DIAL... how crap! What do you think it's doing? Dial(Agent/12345) ! |
| 16:28.35 | [TK]D-Fender | MAbbas: Its just as straightforward as nomal dialplan. AGI can't do that much more to * directly than normal dialplan can. Its jsut calling the app no better than you can on your own. |
| 16:28.55 | Zeeek | <friendly spam> Join us in a half hour for the VUC with Plantronics in 30 minutes, Counterpath in 90 minutes. IRC #vuc - http://vuc.me </friendly spam> |
| 16:29.13 | Zeeek | To opt out of our friendly spam program, please go to .... |
| 16:29.29 | MAbbas | <[TK]D-Fender>: my dialplan is - exten => _80X,1,AGI(agi://10.110.32.44:4572) |
| 16:30.10 | [TK]D-Fender | MAbbas: exten => _80X,2,Dial(Agent/${agenttodialwhichyoushouldsetinyouragirightbeforeleaving}) |
| 16:30.12 | MAbbas | <[TK]D-Fender>: besides, my heristics code is written in C# .. so I can not port it on * |
| 16:30.43 | [TK]D-Fender | MAbbas: the SELECtION of the agent warrants the AGI... Dialing the Agent after selection is NOT. |
| 16:31.06 | Zeeek | C# or you might B flat |
| 16:31.42 | [TK]D-Fender | Zeeek: .. Db ..... |
| 16:31.48 | [TK]D-Fender | (just sayin') |
| 16:31.55 | [TK]D-Fender | rocks out |
| 16:32.22 | bmoraca_work | C-G-E...the three chords that play nearly every rock song ever! |
| 16:32.40 | Zeeek | that was from grade school, look both ways before crossing |
| 16:32.41 | [TK]D-Fender | bmoraca_work: Um.... also close but not quite :) |
| 16:32.41 | MAbbas | <[TK]D-Fender>: so essentialy, you are saying AGI should set a channel variable and quit and dialplan use that variable to dial agent .. I should set a channel variable in AGI and close my connection .. |
| 16:32.52 | [TK]D-Fender | bmoraca_work: 1st, 4th, & 5th magor within |
| 16:32.56 | Zeeek | A G I are the three chords played most often here |
| 16:32.56 | [TK]D-Fender | "major* |
| 16:33.11 | [TK]D-Fender | "within any given major key" |
| 16:33.15 | bmoraca_work | I? |
| 16:33.20 | Zeeek | you just reminded me to cue up the music for the http://vuc.me weekly concert |
| 16:33.29 | Zeeek | I IV V |
| 16:33.36 | MAbbas | <[TK]D-Fender>: thanks ! |
| 16:33.44 | [TK]D-Fender | Zeeek: you should do one on streaming / MoH / royalties 7 licensing |
| 16:33.50 | [TK]D-Fender | & |
| 16:34.06 | bmoraca_work | lol |
| 16:34.17 | Zeeek | I'd love to TK, ! Wo to invite? |
| 16:34.24 | Zeeek | s/Wo/Who/ |
| 16:34.29 | bmoraca_work | indeed, that should be brought to attention...but no one would pay attention. |
| 16:34.58 | MAbbas | <[TK]D-Fender>: One thing how do I let know that my Dial() command was successful/un-successful? |
| 16:35.15 | [TK]D-Fender | MAbbas: ... same way as we do for EVERY call. |
| 16:35.15 | *** join/#asterisk chendy (n=chatzill@116.25.172.169) |
| 16:35.34 | [TK]D-Fender | MAbbas: ..... this is seriously 101 grade stuff...."core show application dial" |
| 16:36.08 | voipmonk | allow the applications to do what they do best and don't be afraid to hand off the information to their partners for best dialplan or asterisk optimization. it should feel natural when reading the dialplan. this does this and that does this but using this to do that and that to do this leads to fatnasty1's scaling issue. if u run into cpu problems - maybe there's another app that does what u need that can take the load off another app. its |
| 16:36.08 | voipmonk | use applications as team members to reach your goal while keeping the overall load to a minimum. |
| 16:36.46 | _abc_ | guys, how do sip dorpouts sound in your parts?! |
| 16:36.52 | _abc_ | *dropouts |
| 16:37.02 | fatnasty1 | If you have to use agi, use fastagi. |
| 16:37.08 | *** join/#asterisk garymc (n=chatzill@host86-158-86-203.range86-158.btcentralplus.com) |
| 16:37.17 | [TK]D-Fender | voipmonk: In his case he left the AGI around far longer than needed and the collection of them is the issue. Easily circumvented with sub-nominal effort |
| 16:38.01 | _abc_ | in my case i hear loud banging noises |
| 16:38.04 | _abc_ | is that normal? |
| 16:38.21 | bmoraca_work | fatnasty1: the problem isn't his using AGI for what he needs...his problem is that he's using AGI wrong. |
| 16:38.39 | [TK]D-Fender | _abc_: Can't hear you, speak up! |
| 16:39.03 | [TK]D-Fender | bmoraca_work: Well using it right.. for the wrong tasks :) |
| 16:39.07 | _abc_ | [TK]D-Fender: i hear noise on the channel ... |
| 16:39.23 | [TK]D-Fender | _abc_: This place is always noisy... |
| 16:40.31 | fatnasty1 | bmoraca_work: Im not really paying attention to what his problems are, Im just making a general statement about AGI, I'd say if your going to use it, use fastagi instaead, even if its not currently nessisary it can only help you down the road. |
| 16:40.55 | MAbbas | <[TK]D-Fender>: Pardon man, I send DIALSTATUS, using another agi connection and disconnect.. right? |
| 16:41.15 | bmoraca_work | fatnasty1: and I still maintain that func_odbc with stored procedures would be faster |
| 16:41.46 | fatnasty1 | bmoraca_work: AGI isnt just for SQL. |
| 16:42.01 | bmoraca_work | fatnasty1: i never said it was. stored procedures can do a lot of things. |
| 16:42.31 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 16:42.47 | fatnasty1 | bmoraca_work: Well I wouldnt use a stored proc to do a screen scrape on a website for football scores. |
| 16:43.13 | fatnasty1 | An AGI script may be used for this. |
| 16:43.18 | asteriskATmarmuD | digium TDM410 and no dial tone on the connecte analog phone??? any hints? |
| 16:43.32 | *** join/#asterisk yoshx (n=yoshx@84.7.169.144) |
| 16:43.49 | *** join/#asterisk keyp (n=keyp@66.184.128.98) |
| 16:44.17 | bmoraca_work | fatnasty1: no, probably not. then again, why would you need a phone service for that when it's far easier to just look at the website myself? untenable examples don't disprove the general theory. |
| 16:44.33 | p3nguin | When a call comes in, I start MixMonitor() before the call goes into the queue. There isn't much time before someone answers, so that part isn't the problem... The problem is that when the call is transferred to me, the MixMonitor() doesn't keep recording. Is there any way to make it continue recording through the transfer? |
| 16:45.36 | fatnasty1 | bmoraca_work: well if you run a fantasy football site, and want your users to be able to dial in and get scores is why. |
| 16:45.47 | Zeeek | join us now, we're getting started: sip:200901@login.zipdx.com or #vuc here on Freenode.net |
| 16:45.50 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 16:46.40 | bmoraca_work | once again, untenable example. aside from that, it would be far more efficient of you to pull those numbers from a database, rather than scrape them from a website every single time someone calls. look, i'm not going to argue with you anymore. keep living in your lala land, and i'll keep working in the real world. |
| 16:46.53 | fatnasty1 | lol |
| 16:47.27 | fatnasty1 | your basicly saying the only use for AGI is SQL and I disagree. |
| 16:47.41 | p3nguin | The phone's transfer key is being used to perform an attended transfer. MixMonitor() records while the caller is on hold (while the answerer is calling someone else to establish the attended transfer), but as soon as they complete the transfer by pressing the button a second time or hanging up, MixMonitor() stops. |
| 16:48.18 | bmoraca_work | fatnasty1: i never once, EVER, said that. i said that there is no reason to use agi when stored procedures can do 99% of the job far more efficiently. the last 1% are examples that would never occur in the real world. end of story. |
| 16:48.42 | _abc_ | omg AGI+SQL in real time is really bad speed-wise |
| 16:48.58 | _abc_ | although a lot of places use that, long distance calling cards too |
| 16:49.03 | fatnasty1 | Is your world the real world? |
| 16:49.11 | Qwell | Is your SQL server on the other side of the Atlantic? |
| 16:49.21 | _abc_ | this is due to the fail mantra 'database==SQL' |
| 16:49.34 | asteriskATmarmuD | ;) |
| 16:49.40 | fatnasty1 | Im outa here, take a shit guys. |
| 16:49.42 | fatnasty1 | exit |
| 16:49.50 | [TK]D-Fender | [11:40]<MAbbas><[TK]D-Fender>: Pardon man, I send DIALSTATUS, using another agi connection and disconnect.. right? <- Really? Why? |
| 16:49.53 | Qwell | fail |
| 16:49.58 | _abc_ | basically a gdb hash lookup is more than enough and you can have 10 million records and it will barely blip the system load |
| 16:50.03 | bmoraca_work | fails at failing...wow |
| 16:50.10 | [TK]D-Fender | MAbbas: You haven't said what you want to DO after the call doesn't go through. |
| 16:50.18 | asteriskATmarmuD | who wants to help me? |
| 16:50.25 | [TK]D-Fender | MAbbas: again this is asking my to make wild guesses about your needs. |
| 16:50.25 | asteriskATmarmuD | digium TDM410 and no dial tone on the connecte analog phone??? any hints? |
| 16:51.16 | voipmonk | what else is involved in this call asteriskATmarmuD ? |
| 16:51.23 | [TK]D-Fender | Qwell: yeah, he's an entire boatload of fail. A life less qualified indeed. |
| 16:51.32 | MAbbas | <[TK]D-Fender>: if Dial is un-successful , I add call to Queue for default processing |
| 16:52.08 | [TK]D-Fender | MAbbas: then just call Queue right after. Do you CARE why the call didn't go through? Also you should dial with a Timeout so it doesn't ring forever |
| 16:53.53 | MAbbas | <[TK]D-Fender>: right - am I stay corrected about letting know DIALSTATUS, using AGI connection? |
| 16:54.33 | [TK]D-Fender | MAbbas: there is no purpose to being in a AGI after the moment you haev retreived the agent # to call. |
| 16:54.43 | [TK]D-Fender | MAbbas: NONE. |
| 16:55.00 | [TK]D-Fender | MAbbas: Dial the Agent # you retrieved. Go into Queue. |
| 16:55.04 | [TK]D-Fender | MAbbas: The End. |
| 16:55.26 | Naikrovek | end credits |
| 16:55.31 | Naikrovek | up tempo music |
| 16:55.41 | [TK]D-Fender | rocks out ... AGAIN |
| 16:55.49 | MAbbas | <[TK]D-Fender>: I have to let my application know, that its decision wan implemented or not .. ! |
| 16:55.56 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/bronze/beek) |
| 16:55.58 | MAbbas | *was |
| 16:56.21 | voipmonk | erf? |
| 16:56.52 | *** part/#asterisk beek (n=klinebl@pdpc/supporter/bronze/beek) |
| 16:57.00 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/bronze/beek) |
| 16:57.28 | *** join/#asterisk ruied (n=ruied@89.180.121.75) |
| 16:57.38 | *** join/#asterisk rgsteele (n=rgsteele@207.106.239.81) |
| 16:58.05 | [TK]D-Fender | MAbbas: What application? |
| 16:58.22 | [TK]D-Fender | MAbbas: What decision? |
| 16:58.33 | MAbbas | <[TK]D-Fender> MAbbas: there is no purpose to being in a AGI after the moment you haev retreived the agent # to call. <- I disconnect agi and after Dial() is finished .. I connect to agi and tell my application that its decision was implement or not .. |
| 16:58.42 | MAbbas | decision = Agent # |
| 16:58.47 | [TK]D-Fender | mab...... |
| 16:58.54 | [TK]D-Fender | MAbbas: NO. |
| 16:59.16 | [TK]D-Fender | MAbbas: Enter AGI. get the Agent to call from it. Return to dialplan. DIAL the agent from dialplan. Then what? |
| 17:00.34 | MAbbas | <[TK]D-Fender>: Let my huristic application know that Agent # it provided was called or not? to maintain stats in my huristics applicaion |
| 17:00.38 | *** join/#asterisk pkecastillo (n=pirruar@190.113.141.122) |
| 17:01.09 | [TK]D-Fender | MAbbas: How do you pass it that information? |
| 17:01.21 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 17:02.03 | *** join/#asterisk jtodd (i=amy33aok@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 17:02.03 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 17:02.05 | MAbbas | <[TK]D-Fender>: I will pass call's uniqueid along with DIALSTATUS |
| 17:02.17 | MAbbas | <[TK]D-Fender>: as agi arguments |
| 17:02.34 | [TK]D-Fender | MAbbas: NO. that AGI DOES something with this information. WHAT does it DO with it? |
| 17:02.51 | [TK]D-Fender | MAbbas: AGI is where you are thing of atking this action. WHAT is the action itself? |
| 17:02.54 | *** join/#asterisk DarkFibre (n=DarkFibr@64.129.95.226) |
| 17:02.55 | [TK]D-Fender | taking* |
| 17:04.45 | MAbbas | <[TK]D-Fender>: Action is Agent # generation to serve the call .. But I have to maintain statistics about incoming call and whom it was Dial()ed |
| 17:05.06 | [TK]D-Fender | MAbbas: where do you STORE the damn statistics?!?!?! |
| 17:05.29 | MAbbas | in DB .. |
| 17:05.45 | [TK]D-Fender | \mabbHow complex a DB call? A simple INSErT? |
| 17:05.58 | MAbbas | <[TK]D-Fender>: yes |
| 17:06.02 | [TK]D-Fender | MAbbas: You can do even THAT directly in dialplan <---- |
| 17:06.31 | [TK]D-Fender | MAbbas: MySQL and ODBC is quite functional directly from dialplan apps. |
| 17:06.53 | [TK]D-Fender | Lunch, back in a few minutes. |
| 17:07.01 | MAbbas | <[TK]D-Fender> ok |
| 17:09.33 | *** join/#asterisk DarkFibre (n=DarkFibr@64.129.95.226) |
| 17:09.41 | _abc_ | [TK]D-Fender: you pushed him down the 'database==mySQL' fail mantra ... |
| 17:11.06 | bmoraca_work | _abc_: mysql is simply an example of an RDBMS. ODBC is an abstraction layer that allows you to connect to any database. the only "fail mantra" here is yours. |
| 17:11.30 | *** join/#asterisk DarkFibre (n=DarkFibr@64.129.95.226) |
| 17:12.50 | _abc_ | the only fail mantra here is to consider a remote database access on a loaded db server an option in a relatime system, and have people wait for the machines to do what they do (slowly) under load |
| 17:13.11 | _abc_ | often on the phone which costs a lot of money |
| 17:13.19 | _abc_ | (when calling in) |
| 17:13.26 | Katty | dlynes_laptop: the adaptor fell behind the couch...i moved it back to where it should be (= |
| 17:13.42 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 17:13.59 | _abc_ | i can quote several real life examples here, most of them related to calling card applications where i clocked 30 seconds and more until the lady in the box tells the remaining card value |
| 17:14.16 | Katty | dlynes_laptop: it's going to look a little washed out again, i pulled the blinds so i can watch out the window (= |
| 17:14.16 | _abc_ | although it is 'just' a simple card number lookup |
| 17:14.24 | bmoraca_work | _abc_: a properly configured database system can provide realtime data. views, for instance, were designed specifically for that. proper indexing in the database can drastically reduce the amount of time it takes to look something up. |
| 17:14.57 | bmoraca_work | _abc_: improperly implemented examples do not invalidate the entire practice. |
| 17:15.15 | _abc_ | bmoraca_work: true, but that requires a dedicated server whose load can be kept under control and provided for with hardware. it also requires the db to be local to the client application, and not shared with other unpredictable users. |
| 17:16.01 | _abc_ | bmoraca_work: so get back to the real world, where everyone is penny-pinching and running on obsolete hardware with non-genius admins, give them SQL and stored procedures, and wait about 5 minutes for them to hang themselves with it |
| 17:16.05 | bmoraca_work | _abc_: no it doesn't. they can coexist just fine. if your database is getting to the point where it is not performing well on a shared server, it's probably time for a dedicated server. |
| 17:16.43 | bmoraca_work | _abc_: that's not the real world. sorry, it just isn't. but, you're welcome to believe so if you'd like. |
| 17:17.18 | _abc_ | ahh, and who is going to convince the boss it's time to pay for it? it runs like it is anyway, right? and the right answer is gdbm which will really do the lookup and update in a blink because it is not SQL, and it will be as fast as an asteroid on the ancient hardware |
| 17:18.46 | _abc_ | i work a lot with embedded systems, assembly and also embedded asterisk with marginal cpu capacity, low ram, and often no disk at all. SQL is a mistake grandfathered from univeristy via web V2.0 |
| 17:19.42 | *** join/#asterisk Corydon76-dig (i=silver@c-69-137-80-31.hsd1.tn.comcast.net) |
| 17:19.42 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 17:20.01 | *** join/#asterisk MAbbas (i=Jinbaba@58-27-164-211.wateen.net) |
| 17:20.22 | bmoraca_work | _abc_: you're not going to run large database queries in an embedded system. yet another untenable example. in your own example of calling card balance lookup...you don't run that on an embedded system, and you'd be a moron to try. depending on your client base, you run that on very, very large servers. why would you even bring embedded systems into the equation? |
| 17:20.39 | _abc_ | incidentally gdbm is a component of mySQL, i.e. mySQL runs on gdbm |
| 17:21.09 | _abc_ | bmoraca_work: thanks for the compliments, the application runs great on the embedded diskless system. |
| 17:21.11 | MAbbas | <[TK]D-Fender> MAbbas: MySQL and ODBC is quite functional directly from dialplan apps. |
| 17:21.21 | MAbbas | <[TK]D-Fender>well, I do not have direct access DB, I call API of another application that stores DB entries |
| 17:21.29 | bmoraca_work | _abc_: you must have 15 customers. congrats. but you're still a moron. |
| 17:21.49 | *** join/#asterisk JraNil (n=JraNil@85.133.218.127) |
| 17:21.56 | _abc_ | bmoraca_work: i heard in argentina they throw capybarras with a catapult to communicate |
| 17:22.05 | Naikrovek | *thud* |
| 17:22.12 | Naikrovek | capybaras are freakin' huge |
| 17:22.17 | Naikrovek | (for rodents) |
| 17:22.38 | [TK]D-Fender | MAbbas: What kind of API calls? |
| 17:22.45 | *** join/#asterisk RobH (n=robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 17:23.16 | MAbbas | <[TK]D-Fender> non standard, DB application exposes its own API's |
| 17:23.37 | [TK]D-Fender | MAbbas: At worst you would call a second AGI for the sole purpose of doing this logging. This means that GI is only in use for the bare minimum of time. |
| 17:23.54 | [TK]D-Fender | AGI* |
| 17:24.27 | MAbbas | <[TK]D-Fender>: yes, I just pass DIALSTATUS as agi arg and read it and close the connection .. |
| 17:24.27 | *** join/#asterisk lil_Mac (n=johnsmit@cpe-71-74-167-121.indy.res.rr.com) |
| 17:25.13 | [TK]D-Fender | MAbbas: Yes, that way you don't have AGI's sucking up resources & connection for active calls in progress. Only for a moment at the very start, and at the very end |
| 17:25.14 | *** join/#asterisk TheDavidFactor (n=chatzill@fw1.safedataisp.net) |
| 17:25.58 | MAbbas | <[TK]D-Fender> Thanks man! You have been big help .. |
| 17:26.26 | *** part/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 17:27.58 | *** join/#asterisk _cgc (n=_cgc@94-193-99-128.zone7.bethere.co.uk) |
| 17:28.01 | _cgc | hi everyone |
| 17:28.41 | _cgc | does anyone know why I would be getting the following when a call comes in over a sip trunk: http://pastebin.ca/1770969 |
| 17:29.43 | [TK]D-Fender | _cgc: tahts the end of it, is it? |
| 17:29.46 | _cgc | if I change the sip peer in sip.conf to 'host=dynamic' it works, but then outgoing calls don't work |
| 17:29.54 | _cgc | [TK]D-Fender: yes |
| 17:30.32 | [TK]D-Fender | _cgc: * is challenging their invite and they just give up. Add "insecure=port,invite" to your [voiptalk] entry |
| 17:30.50 | *** join/#asterisk bl4 (n=bl4qkuba@host-166.arcadia-srv-216-83-132.fiber.net) |
| 17:31.15 | *** join/#asterisk titter (n=titter@c-76-101-240-142.hsd1.fl.comcast.net) |
| 17:31.17 | _cgc | [TK]D-Fender: ok will try, thank you very much :) |
| 17:32.22 | *** join/#asterisk Godfather_ (n=Godfathe@79.109.251.250.dyn.user.ono.com) |
| 17:32.46 | _cgc | [TK]D-Fender: brilliant! that worked perfectly :) |
| 17:32.49 | Godfather_ | o/ |
| 17:33.02 | [TK]D-Fender | _cgc: You're welcome. |
| 17:33.09 | titter | Anyone here splitting a PRI into BRI's, or splitting a PRI some other way? I have two PRI's at one site, and one connects to our main Asterisk box, the other goes to our Asterisk box dedicated to fax. I would like to split some channels from the 2nd PRI, and bring them into my main Asterisk box. |
| 17:33.12 | [TK]D-Fender | NEXT!@!@@! (c) BKW |
| 17:33.43 | *** join/#asterisk DarkFibre_XPS (n=DarkFibr@64.129.95.226) |
| 17:34.41 | [TK]D-Fender | titter: if you have 2 PRI's, 2 *'s, and each * is connected to its own PRI.... then why would you send calls between your *'s via BRI? |
| 17:35.07 | [TK]D-Fender | titter: this adds senseless hardware and complexity into the mix |
| 17:36.35 | jaytee | [TK]D-Fender, you're certainly racking up some good karma today |
| 17:37.00 | titter | [TK]D-Fender: Basically the first PRI to our main * is completely in use, all 23 chans. Just trying to find a solution to better load balance. |
| 17:37.28 | carrar | titter, have phone company roll calls over to the second PRI |
| 17:37.28 | titter | [TK]D-Fender: Hence the 2nd part of my question, splitting the PRI some other way. |
| 17:37.38 | titter | carrar: Thanks. |
| 17:37.41 | carrar | then via SIP point those to the primary * |
| 17:38.05 | titter | That is much easier. |
| 17:39.20 | *** join/#asterisk terracon (n=greisky@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 17:39.33 | [TK]D-Fender | jaytee: My karma ran over your dogma :D |
| 17:39.45 | *** join/#asterisk Alagar (n=Administ@122.164.41.40) |
| 17:41.25 | *** join/#asterisk lanning (n=lanning@208.87.235.224) |
| 17:42.03 | Katty | halp, being attacked by slobbery monster |
| 17:42.24 | jaytee | slobbery? is it's name Riddick? |
| 17:42.32 | Katty | how did you guess?! |
| 17:42.46 | [TK]D-Fender | GETT OFF HER RYAN! |
| 17:42.50 | [TK]D-Fender | :O |
| 17:42.56 | Katty | he is very determined and depositing slobbery squeaky toy on my lap |
| 17:43.00 | jaytee | just lucky, plus ferrets are very slobberous |
| 17:43.14 | Katty | ferrets don't slobber ^_- |
| 17:43.31 | jaytee | neither does Ryan probably but I bet he drools and snores in his sleep |
| 17:43.47 | Katty | he definately snores sometimes |
| 17:43.51 | Katty | j8923j |
| 17:43.53 | Katty | sorry |
| 17:44.02 | Katty | pup wants to type! |
| 17:44.09 | *** join/#asterisk pietro (n=pietro@88-149-224-77.dynamic.ngi.it) |
| 17:44.35 | jaytee | I've been trying to get my hands on a j8923j for over two months now but they're always on backorder |
| 17:45.04 | Katty | lol |
| 17:45.30 | Katty | zE4~S <- lol. squeaky orange tennis balls are very hard to come by ;P |
| 17:46.43 | titter | lol |
| 17:49.06 | _abc_ | is that a dog typing? |
| 17:49.27 | Katty | no, that's a dog trying to give me his squeaky toy while i'm typing |
| 17:49.28 | *** join/#asterisk simplydrew (n=simplydr@pool-74-97-177-245.prvdri.fios.verizon.net) |
| 17:50.01 | _abc_ | uhh, ok. anything is possible on irc. log in to a root shell and see if he is a not a cat! |
| 17:50.10 | _abc_ | by letting him type |
| 17:50.19 | *** join/#asterisk rfoxpct (n=SMB921@mail.pctrouble.com) |
| 17:50.22 | _abc_ | or her |
| 17:50.41 | Katty | riddick has no interest in the keyboard |
| 17:50.45 | Katty | he just wants me to throw the ball |
| 17:51.03 | Katty | if he walks up to me, and i don't take the ball, he will dump it in my lap |
| 17:52.24 | *** join/#asterisk comradeb14ck (n=comradeb@72.37.252.50) |
| 17:52.29 | Katty | hi |
| 17:52.38 | comradeb14ck | hi |
| 17:55.41 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 18:02.33 | anonymouz666 | BitchX still exist? |
| 18:02.34 | anonymouz666 | lol |
| 18:02.59 | black | barley =p |
| 18:05.17 | voipmonk | http://www.amazon.com/Dream-Me-Lightweight-Adjustable-Turquoise/dp/B002CVTLBE/ref=reg_hu-wl_item-added |
| 18:05.22 | voipmonk | err... |
| 18:09.13 | drmessano | Where's the USB port on that? |
| 18:11.18 | *** join/#asterisk albertoandrade (n=albertoa@187.59.74.212) |
| 18:12.01 | *** join/#asterisk dandre (n=daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 18:12.11 | _abc_ | anonymouz666: what's so funny about that? |
| 18:13.17 | _abc_ | anonymouz666: if you meant the irc client then try hydra |
| 18:29.46 | Naikrovek | anyone know if there are any polycom resellers in india |
| 18:29.55 | Naikrovek | i can't seem to find any via google or polycom.com |
| 18:34.41 | *** join/#asterisk ttl- (n=patrick@d5153A420.access.telenet.be) |
| 18:35.49 | *** join/#asterisk DarkFibre_XPS (n=DarkFibr@64.129.95.226) |
| 18:39.52 | *** join/#asterisk asteriskATmarmuD (n=mundt@193.158.65.23) |
| 18:40.33 | asteriskATmarmuD | last time for today, trying my luck |
| 18:41.09 | asteriskATmarmuD | my analog card and dahdi seem to be installed and configured correctly, but I can't get a dial tone on my phone |
| 18:41.26 | Naikrovek | do you have FXO modules or FXS |
| 18:41.29 | Naikrovek | or both |
| 18:41.31 | asteriskATmarmuD | FXS |
| 18:41.35 | asteriskATmarmuD | only |
| 18:41.39 | Naikrovek | then I have no idea why you don't get dialtone |
| 18:41.49 | [TK]D-Fender | asteriskATmarmuD: What colour are the modules? Did you connect the molex? Feel like actually showing us configs & card status this time? |
| 18:41.50 | asteriskATmarmuD | great |
| 18:41.51 | Naikrovek | check the logs, make sure DAHDI is booting |
| 18:42.01 | asteriskATmarmuD | dahdi works |
| 18:42.04 | Naikrovek | "booting" wtf am i talking about |
| 18:42.10 | [TK]D-Fender | asteriskATmarmuD: Who shot J.R.? What's the Caramilk secret? How much could a woodchuck chuck? |
| 18:42.19 | asteriskATmarmuD | ringing of dahdi channel 1 shown on CLI (LOG) |
| 18:42.29 | Naikrovek | if DAHDI works then you get dialtone. you don't get dialtone so either DAHDI isn't working/configured or your phone is busted |
| 18:42.39 | Naikrovek | but i'm no expert on it |
| 18:42.48 | asteriskATmarmuD | ok, this is what I thought |
| 18:42.51 | Naikrovek | if something isn't working there will be a log entry of it |
| 18:42.54 | Naikrovek | somewhere |
| 18:42.58 | [TK]D-Fender | Screw logs. |
| 18:42.58 | Naikrovek | or |
| 18:42.59 | Naikrovek | maybe |
| 18:43.02 | Naikrovek | the card is bad or something |
| 18:43.03 | asteriskATmarmuD | moment please |
| 18:43.05 | [TK]D-Fender | * CLI only. |
| 18:43.08 | voipmonk | well |
| 18:43.16 | voipmonk | no |
| 18:43.37 | Naikrovek | i don't know why i'm so angry about that question... he's been polite and not flooded |
| 18:43.41 | voipmonk | fxo? |
| 18:43.49 | asteriskATmarmuD | dmesg -> 33.833895] Found a Wildcard TDM: Wildcard TDM410P (4 modules) |
| 18:43.51 | Naikrovek | he said FXS a bit ago |
| 18:44.41 | [TK]D-Fender | Naikrovek: And I'm waiing for the more conslusive answer |
| 18:44.42 | [TK]D-Fender | conclusive* |
| 18:44.42 | Naikrovek | waiting* |
| 18:44.42 | Naikrovek | yeah i know |
| 18:44.43 | asteriskATmarmuD | <PROTECTED> |
| 18:44.43 | asteriskATmarmuD | <PROTECTED> |
| 18:44.43 | asteriskATmarmuD | <PROTECTED> |
| 18:44.43 | asteriskATmarmuD | <PROTECTED> |
| 18:44.43 | asteriskATmarmuD | <PROTECTED> |
| 18:44.44 | asteriskATmarmuD | <PROTECTED> |
| 18:44.46 | asteriskATmarmuD | <PROTECTED> |
| 18:44.47 | Naikrovek | eeee |
| 18:44.50 | asteriskATmarmuD | sorry |
| 18:44.53 | asteriskATmarmuD | for the mess |
| 18:44.56 | voipmonk | asteriskATmarmuD: use pastebin.ca |
| 18:44.57 | Naikrovek | use a pastebin |
| 18:45.03 | black | http://pastie.org/ |
| 18:45.03 | black | ! |
| 18:45.06 | Naikrovek | once is forgiveable |
| 18:45.07 | asteriskATmarmuD | thx |
| 18:45.16 | voipmonk | asteriskATmarmuD: are these fxs or fxo? |
| 18:45.17 | Naikrovek | but that's it :) |
| 18:45.24 | carrar | It's FRI |
| 18:45.33 | carrar | no rules friday! |
| 18:45.36 | Naikrovek | carrar: i like it |
| 18:45.40 | asteriskATmarmuD | ;) |
| 18:45.44 | asteriskATmarmuD | 4 FXS modules |
| 18:45.44 | black | FRIDAY |
| 18:45.45 | black | WOO |
| 18:45.54 | black | (>'.')> |
| 18:45.55 | Naikrovek | black friday? dealz? |
| 18:45.57 | black | <('.'<) |
| 18:45.58 | asteriskATmarmuD | yeah friday night, I am still at work |
| 18:46.07 | Naikrovek | european! |
| 18:46.08 | black | It's only 10:45am here |
| 18:46.17 | black | But i'm still partying, baby! |
| 18:46.18 | Naikrovek | noon:45pm here |
| 18:46.30 | asteriskATmarmuD | <PROTECTED> |
| 18:46.37 | Naikrovek | asteriskATmarmuD: you got the channel going again |
| 18:46.40 | Naikrovek | thank you! (i mean that) |
| 18:46.48 | asteriskATmarmuD | shows every channel is "in service" IS THIS GOOD, I THOUGHT SO |
| 18:46.57 | Naikrovek | yes in service is good |
| 18:47.12 | Naikrovek | probably not a DAHDI problem, i would guess. voipmonk or [TK]D-Fender can correct me if they like |
| 18:47.15 | asteriskATmarmuD | then why the helöl doesnt it work |
| 18:47.28 | Naikrovek | no idea :/ |
| 18:47.31 | asteriskATmarmuD | phone tested on normal line works, on digium tdw410 not.... |
| 18:47.39 | Naikrovek | huh |
| 18:47.41 | voipmonk | last dahdi problem i had I was using sip to dial out the fxo , I had to add ignoresdpversion=yes into sip.conf |
| 18:47.51 | [TK]D-Fender | asteriskATmarmuD: I don't see answers to ANY of my questions. |
| 18:48.06 | carrar | You can't handle the answers!! |
| 18:48.22 | asteriskATmarmuD | had the same problem 2 times today, about 2-3 hours ago |
| 18:48.31 | asteriskATmarmuD | guess I have got my lucky minutes |
| 18:49.02 | asteriskATmarmuD | ok, noone got any hint on: dahdi works but no dial tone |
| 18:49.07 | asteriskATmarmuD | ? |
| 18:49.34 | asteriskATmarmuD | is there some info you need to see dahdi is fully working, did I provide everytrhing already |
| 18:49.42 | [TK]D-Fender | asteriskATmarmuD: Where are you configs? |
| 18:49.50 | Naikrovek | asteriskATmarmuD: find out what [TK]D-Fender is looking for, put it in a pastebin somewhere and he'll be happy to help you |
| 18:49.55 | [TK]D-Fender | asteriskATmarmuD: where is the confirmation on the colour of the modules? |
| 18:49.55 | Naikrovek | this is what I would do if I were you |
| 18:49.56 | Naikrovek | or |
| 18:50.02 | Naikrovek | contact voipmonk and he'll help you. |
| 18:50.05 | [TK]D-Fender | asteriskATmarmuD: where is the confirmation that you plugged in the molex connector? |
| 18:50.06 | Naikrovek | both VERY capable |
| 18:50.07 | asteriskATmarmuD | ah, pastebin |
| 18:50.08 | Naikrovek | cery smart |
| 18:50.10 | asteriskATmarmuD | checkin it nox |
| 18:50.12 | asteriskATmarmuD | now |
| 18:50.12 | Naikrovek | very* |
| 18:51.53 | Akiraa | Is there a traffic sniffing tool that specializes on VoIP (SIP) traffic? |
| 18:52.15 | Naikrovek | sip debug |
| 18:52.18 | Naikrovek | :P |
| 18:52.28 | Naikrovek | wireshark recognizes SIP traffic |
| 18:52.33 | carrar | tcpdump |
| 18:52.44 | carrar | ngrep |
| 18:52.48 | Naikrovek | you could use that if you really wanted to see that, but sip debug in asterisk console will give you what you want i think |
| 18:54.06 | leifmadsen | tshark |
| 18:54.30 | leifmadsen | for GUI, wireshark understands SIP and IAX2 |
| 18:54.41 | Naikrovek | iax2 as well.. nice |
| 18:55.36 | asteriskATmarmuD | so I gathered some info |
| 18:55.38 | asteriskATmarmuD | http://pastie.org/800778 |
| 18:55.51 | asteriskATmarmuD | would be pleased I you could help |
| 18:55.55 | asteriskATmarmuD | stuck since days |
| 18:58.05 | Naikrovek | bmoraca_work: you there? |
| 18:58.19 | bmoraca_work | lurking, yes. what do you need? |
| 18:58.28 | Naikrovek | bmoraca_work: you're polycom reseller, yes? |
| 18:58.32 | asteriskATmarmuD | [TK]D-Fender: molex plugged |
| 18:58.35 | Naikrovek | i need to find a polycom reseller in india |
| 18:58.44 | bmoraca_work | Naikrovek: yes, but I can't sell internationally |
| 18:58.56 | Naikrovek | i know, but maybe you can find a list of resellers |
| 18:59.00 | [TK]D-Fender | asteriskATmarmuD: What colour are the modules? |
| 18:59.07 | asteriskATmarmuD | I also switched it, to check if it is the problem |
| 18:59.28 | Naikrovek | bmoraca_work: i'm in the US anyway; i just need to find a reseller in india for my india guys. soooo tired of shipping phones to them from here |
| 19:00.17 | asteriskATmarmuD | [TK]D-Fender: I am sitting at this "server" right know.... it is 4x FXS as ordered |
| 19:00.39 | asteriskATmarmuD | [TK]D-Fender: I wanted to say, can't really open it right now... but will try.... |
| 19:00.58 | bmoraca_work | Naikrovek: if they don't have a "Where to Buy" list on their website, I don't have any other way to get that information |
| 19:01.12 | Naikrovek | bmoraca_work: okay. i can't find one. i could be blind, though. |
| 19:01.21 | ManxPower-work | 3 days until the Polycom SDK is released. |
| 19:01.31 | Naikrovek | ooOOohh |
| 19:02.01 | bmoraca_work | Naikrovek: you could try calling their India main branch... http://www.polycom.co.in/ |
| 19:02.17 | ManxPower-work | If it's anything like the rest of stuff from Polycom it will be poorly documented, hard to use, and kick ass when you get it working |
| 19:02.20 | Naikrovek | bmoraca_work: yeah been cruising their site. 12:30am there now, saturday morning. probably not in the office |
| 19:02.28 | Naikrovek | ManxPower-work: true dat |
| 19:03.04 | bmoraca_work | ManxPower-work: what's the purpose of the SDK? open the firmware to development for applications similar to their own licensable apps? |
| 19:03.05 | Naikrovek | their documentation staff needs a kick in the tokus sometimes. though the admin guide for the soundpoint phones is pretty good |
| 19:03.40 | *** part/#asterisk ruied (n=ruied@89.180.121.75) |
| 19:03.51 | ManxPower-work | bmoraca_work: I don't know, but based on the references to it, I think it's mainly a dev kit for the microbrowser. In any case ANY more docs on the microbrowser is a good thing. |
| 19:04.05 | bmoraca_work | definitely |
| 19:04.22 | ManxPower-work | I would love to be able to specify only digits in a form field in the microbrowser, for example. |
| 19:04.35 | Naikrovek | ah i found a reseller in india |
| 19:04.45 | Naikrovek | via voip-info wiki of all places |
| 19:05.05 | bmoraca_work | ManxPower-work: does the microbrowser support push? or is that only available in those update messages you were wrangling with earlier? |
| 19:05.25 | *** join/#asterisk dkirker-openmobl (n=dkirker@openmobl/ceo/dkirker) |
| 19:05.31 | Qwell | ManxPower-work: it doesn't support that? it's a rather basic feature... |
| 19:05.35 | Qwell | even Ciscos support that |
| 19:05.41 | Katty | :>>>> |
| 19:05.46 | ManxPower-work | bmoraca_work: there's lots of very cool things in the Polycom Web Devel manual. XMLHTTPrequest is supported, lots of other stuff that I can't figure out. |
| 19:05.47 | Katty | chipping sparrow has been discovered! |
| 19:05.52 | Naikrovek | what do polycoms support that cisco doesn't? naive question perhaps |
| 19:05.53 | *** join/#asterisk moy (n=moy@74.12.123.169) |
| 19:05.57 | asteriskATmarmuD | [TK]D-Fender: 4 green FXS modules |
| 19:06.04 | ManxPower-work | http push and pull appear to be supported |
| 19:06.38 | asteriskATmarmuD | [TK]D-Fender: I don't have any clue what might be wrong... if you or someone else could look over my pastie ... would be great |
| 19:06.41 | bmoraca_work | Naikrovek: the ability to modify ring tones/cadences/behavior (distinctive ring/auto answer) at calltime, rather than in configs |
| 19:06.47 | Katty | Polycom's support a smaller budget. |
| 19:06.54 | asteriskATmarmuD | gotta go now... its late... back to work on monday |
| 19:06.58 | Naikrovek | yeah other than being cheaper |
| 19:07.02 | Naikrovek | asteriskATmarmuD: good luck |
| 19:07.11 | bmoraca_work | Naikrovek: that basically means that they properly support intercoms and Cisco's (at least using SIP firmware) don't. |
| 19:07.35 | asteriskATmarmuD | thx guys, bye |
| 19:07.38 | *** part/#asterisk asteriskATmarmuD (n=mundt@193.158.65.23) |
| 19:08.00 | Naikrovek | bmoraca_work: okay |
| 19:09.08 | bmoraca_work | Naikrovek: polycoms are nice to use because they're an open architecture and polycom actively supports use on other PBXes. Cisco doesn't. |
| 19:09.14 | bmoraca_work | (officially) |
| 19:09.17 | Naikrovek | bmoraca_work: yeah i know that |
| 19:09.30 | Naikrovek | i'm talking technically, what can polycom phones do that cisco phones can't |
| 19:09.37 | Naikrovek | wonder if there's a list somehwere |
| 19:09.43 | bmoraca_work | i doubt it |
| 19:09.52 | Naikrovek | probably right |
| 19:10.02 | bmoraca_work | BLF also worls a lot better on Polycoms than it does on Ciscos (again, at least under SIP) |
| 19:10.12 | Naikrovek | yeah blf is nice |
| 19:10.19 | Naikrovek | i set it up here a couple days back |
| 19:10.37 | Naikrovek | in prep for the ip650 w/sidecar i've ordered |
| 19:11.00 | bmoraca_work | the attendant console for the Cisco phones (7900 series) don't work with SIP firmware, either. so if you need an attendant console, you can't use Cisco 7900 series phones |
| 19:11.12 | bmoraca_work | at least that was the case last time i looked in to it |
| 19:12.21 | Katty | p3nguin: there is a /ton/ of snow coming down |
| 19:12.24 | bmoraca_work | i don't have any experience with the Cisco 500 series phones yet, so i don't know how they compare to the Polycoms, but being the next evolution of the SPA-900s, I imagine that they're just as open architecturally, but suck in quality |
| 19:12.45 | [TK]D-Fender | Polycom supports multiple calls per line-key, better SIP support, non-hostile to open standards, more stable firmware, more highly configurable, G.722, ad-hoc conferencing, and a growing featureset. |
| 19:13.15 | [TK]D-Fender | Cisco = several steps behind |
| 19:13.36 | bmoraca_work | [TK]D-Fender: Cisco can do multiple calls per line key and ad-hoc conferencing as well. they're also just as stable once you get them set up |
| 19:14.16 | carrar | Cisco just looks prettier |
| 19:14.22 | carrar | Polycom's work better |
| 19:14.24 | [TK]D-Fender | bmoraca_work: Can they now? that's news. what about provisioning from more than just tftp, legal issues, firmware, etc? |
| 19:15.16 | bmoraca_work | [TK]D-Fender: i never claimed Cisco phones were as good as Polycoms. i was simply pointing out that some of those features are available. |
| 19:16.23 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:16.27 | [TK]D-Fender | bmoraca_work: Sorry, I'm not shooting your idea down here, I'm asking you for an update on what Cisco's are like NOW as you would undoubtedly be more up to date on where they stand now. |
| 19:17.06 | Naikrovek | i have a phone set up with 8 line appearances on one like key |
| 19:17.08 | [TK]D-Fender | bmoraca_work: I'm open-minded... just so few products give me reason to change opinions :) |
| 19:17.32 | [TK]D-Fender | Naikrovek: 8 appearances on a single key? What phone? |
| 19:17.38 | Naikrovek | ip321 :) |
| 19:17.46 | Naikrovek | they support up to 8 per line key |
| 19:17.51 | carrar | Cisco with SIP Code is still very limited |
| 19:17.51 | Naikrovek | default is 2 |
| 19:17.59 | [TK]D-Fender | Naikrovek: Ummm... no, thats CALL's, not LINE appearances |
| 19:18.02 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 19:18.17 | bmoraca_work | [TK]D-Fender: well, to answer that, 7900 series phones still only provision via TFTP, are still encumbered by legal and licensing issues...they're easier to configure than Polycoms, though. like I said, I have no experience with the Cisco 500 series yet, so I can't comment on those (but they were designed to be open) |
| 19:18.27 | [TK]D-Fender | Naikrovek: And last I checked, 32X/33X only supported 2 calls per key |
| 19:18.46 | Naikrovek | well i scrolled through a lot more than 2 on my desk phone the other day |
| 19:18.46 | [TK]D-Fender | bmoraca_work: the 500's look like a minor refresh of the 900's |
| 19:18.50 | Naikrovek | which is a 320 |
| 19:18.52 | bmoraca_work | [TK]D-Fender: yeah, not enough DSP resources to do otherwise |
| 19:19.06 | bmoraca_work | Naikrovek: active calls? |
| 19:19.13 | Naikrovek | all were on hold |
| 19:19.33 | Naikrovek | maybe i unlocked the 9th gate or something to make it happen |
| 19:19.44 | [TK]D-Fender | Naikrovek: I'll check up on this..... |
| 19:19.47 | Naikrovek | i thought "neat, no one will ever use this" and went on to something else |
| 19:20.04 | [TK]D-Fender | Naikrovek: the IP 30X supported 5 until recent firmwares cut them back. |
| 19:20.09 | bmoraca_work | [TK]D-Fender: that's what i suspected, but i'd like to see how they compare from an aesthetic point of view and configurations (let's face it, 900's were ugly inside and out) |
| 19:20.48 | [TK]D-Fender | bmoraca_work: I'd seen worse... the 900's were a good suggestion to those where Polycom/Cisco were not cost-competitive (Asia / Europe, etc) |
| 19:21.55 | *** join/#asterisk Tech_Travis (n=Travis@mail.techglia.com) |
| 19:30.06 | *** part/#asterisk _abc_ (n=no@unaffiliated/ccbbaa) |
| 19:31.15 | *** join/#asterisk Buklov (n=buklov@213.138.71.254) |
| 19:32.03 | carrar | hahahah http://www.youtube.com/watch?v=SXmv8quf_xM |
| 19:32.08 | carrar | WS |
| 19:34.25 | bmoraca_work | rofl "tracer tee" |
| 19:35.13 | bmoraca_work | wow, that guy's a fucking moron |
| 19:36.19 | Naikrovek | yeah |
| 19:36.24 | Naikrovek | saw that yesterday |
| 19:36.34 | bmoraca_work | "all these guys, I can't view them because my connection's not as good as theirs" |
| 19:36.35 | bmoraca_work | rofl |
| 19:36.42 | Naikrovek | it's always funny when someone is dumb and think they're smart |
| 19:36.52 | Naikrovek | like me for example |
| 19:36.56 | Naikrovek | always good for a laugh |
| 19:39.08 | carrar | heh |
| 19:39.08 | carrar | yeah |
| 19:39.14 | carrar | thats awesome stuff |
| 19:39.27 | bmoraca_work | http://www.youtube.com/watch?v=0MDQtDRu46A - "after months and months and months of hacking limewire" rofl. this guy's pure comedy |
| 19:40.44 | [TK]D-Fender | Naikrovek: I see indeed they pushed it from 2 to 8. About time :) |
| 19:40.49 | [TK]D-Fender | Naikrovek: Thanks for the heads up |
| 19:41.07 | *** join/#asterisk jo8330 (i=d04149c9@gateway/web/freenode/x-slvxpqplyeonrael) |
| 19:41.11 | jo8330 | hi folks. |
| 19:41.58 | jo8330 | i have a quick question regarding originating calls via AGI. Is there a way to originate without having to specify a specific channel number? |
| 19:42.30 | [TK]D-Fender | jo8330: You always need a Channel: |
| 19:42.35 | jo8330 | eg. I do "DAHDI/1/5555555555" to originate... However, is there a way that I don't have to specify "1" |
| 19:42.46 | jo8330 | and have asterisk choose the first available channel? |
| 19:42.48 | [TK]D-Fender | jo8330: what do you WANT to call? |
| 19:42.51 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 19:42.58 | [TK]D-Fender | jo8330: this is no different from what you would pass DIAL() |
| 19:43.14 | [TK]D-Fender | jo8330: How about passing it a GROUP instead of a fixed channel? |
| 19:43.32 | *** join/#asterisk kalib (n=lkhlui@osiris.aspec.com.br) |
| 19:43.33 | [TK]D-Fender | jo8330: This a a basic and well documented DAHDI parameter |
| 19:43.38 | jo8330 | I think that's what i need to know, thanks. I'll look into that. |
| 19:43.44 | [TK]D-Fender | jo8330: GROUP your channels together to pick from the pool |
| 19:44.22 | kalib | Hi people. In my CLI I can see some channels with "dahdi show channels". How can I unable for example the channel 66? |
| 19:44.36 | kalib | And how to enable it again? Just wanna make some tests... |
| 19:44.41 | jo8330 | actually I have them all in a single group |
| 19:44.49 | Naikrovek | [TK]D-Fender: yah |
| 19:44.54 | jo8330 | is there a way I can get astierks to choose an avail chanenl for that group, without having me manage them myself? |
| 19:45.19 | [TK]D-Fender | jo8330: you put them into a group and don't know how to DIAL the group? |
| 19:45.51 | bmoraca_work | kalib: asterisk cannot administratively take down channels or interfaces |
| 19:46.00 | kalib | hum... |
| 19:46.09 | jo8330 | i'm only familiar with dialing with a single channel via "DAHDI/<chan#>/<phone#>" |
| 19:46.25 | bmoraca_work | jo8330: the book might help you... |
| 19:46.28 | bmoraca_work | ~book |
| 19:46.29 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 19:46.36 | [TK]D-Fender | jo8330: group = 1. Dial(DAHDI/g1/1234567890) |
| 19:46.39 | jo8330 | thanks |
| 19:47.04 | [TK]D-Fender | jo8330: G / g (one is ascending order of selection, the other is descending |
| 19:47.05 | bmoraca_work | g, G, r, R...all valid group prefixes that do different things. read about them in the BOOK |
| 19:47.14 | bmoraca_work | or [TK]D-Fender can tell you :P |
| 19:47.24 | jo8330 | thanks guys! |
| 19:47.34 | [TK]D-Fender | bmoraca_work: yeah, this one felt worth handing out... when its a solitary letter :) |
| 19:47.43 | bmoraca_work | lol |
| 19:47.53 | [TK]D-Fender | bmoraca_work: But i didn't say which was which ;) |
| 19:47.54 | jo8330 | lol |
| 19:48.00 | [TK]D-Fender | jo8330: GO READ! |
| 19:48.05 | jo8330 | book is free too, how awesome is that. |
| 19:48.15 | Naikrovek | very |
| 19:48.26 | bmoraca_work | jo8330: bout as awesome as a face-plant into a big tub of topless asians |
| 19:48.38 | jo8330 | i findi t hard to find documentation in general out side of voip-info.org |
| 19:48.38 | [TK]D-Fender | Protein powder is whey awesome. |
| 19:48.47 | Naikrovek | heh |
| 19:49.17 | Nugget | haha |
| 19:49.18 | [TK]D-Fender | bmoraca_work: .... what kind of "asians"? This is dangerously open... |
| 19:49.26 | jo8330 | lol |
| 19:49.55 | bmoraca_work | [TK]D-Fender: japanese schoolgirls? to me, "topless" implies "female" |
| 19:50.22 | [TK]D-Fender | bmoraca_work: I'd like the think so but... </ackbar> |
| 19:50.33 | bmoraca_work | IT'S A TARP! |
| 19:50.44 | Naikrovek | tarp? |
| 19:51.08 | bmoraca_work | (jab at Barakbar) |
| 19:51.16 | Naikrovek | oh |
| 19:51.22 | *** join/#asterisk kW_ (n=kW@pD9EAC1CE.dip.t-dialin.net) |
| 19:51.29 | bmoraca_work | Naikrovek: http://www.icanhasforce.com/wp-content/uploads/2008/01/star-wars-ackbar-tarp.jpg |
| 19:51.44 | Naikrovek | heh. hehe. hehehe. |
| 19:52.00 | kW_ | Hello! How can I force asterisk to do RTP proxying between SIP users which are mutually unroutable directly? |
| 19:52.01 | jo8330 | epic |
| 19:52.02 | [TK]D-Fender | bmoraca_work: \o/ WIN |
| 19:52.13 | Defraz | Anyone using an adtran Total Access 924e for a PRI sip gateway? |
| 19:52.16 | [TK]D-Fender | kW_: "canreinvite=no" |
| 19:52.32 | bmoraca_work | Defraz: not this again... Yes, I do, and yes, it works both ways :) |
| 19:52.46 | kW_ | [TK]D-Fender: in sip.conf for each such user? |
| 19:52.54 | [TK]D-Fender | kW_: Yes |
| 19:53.14 | Naikrovek | comcast beta testing ipv6 to consumers? oooh. wish my router supported it |
| 19:53.44 | *** join/#asterisk RichardLynch (n=RichardL@c-98-193-36-91.hsd1.il.comcast.net) |
| 19:53.53 | bmoraca_work | Naikrovek: I've got a /64 of IPv6 addresses...never really felt like setting up the tunnel to them, though. |
| 19:53.55 | Qwell | Naikrovek: Go buy a linksys and put one of the open source firmwares on it |
| 19:54.09 | Naikrovek | have a linksys, thinking of ddwrt right now... |
| 19:54.21 | jo8330 | are there any other recommended documentation sources other than voip-info.org and the asterisk book? |
| 19:54.30 | [TK]D-Fender | jo8330: the * source tarball |
| 19:54.32 | Qwell | jo8330: Asterisk itself |
| 19:54.34 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:54.39 | bmoraca_work | jo8330: what they said |
| 19:54.43 | jo8330 | ok |
| 19:54.59 | [TK]D-Fender | jo8330: that is THE primary source of documentation. Everything else is 3rd party, and potentially dated or outright wrong |
| 19:55.16 | Defraz | bmoraca_work: A while back I never saw a reply from you. |
| 19:55.19 | [TK]D-Fender | (Especially the WIKI) |
| 19:55.24 | Defraz | Do you have a sample config? |
| 19:55.34 | Defraz | Just got some time thought i would fool around with it again. |
| 19:55.42 | bmoraca_work | Defraz: what is it you're trying to do with it? |
| 19:56.11 | Defraz | Well, I have a pri from Qwest. and I want it to pass the calls sip to my asterisk box. |
| 19:56.41 | bmoraca_work | Defraz: the trick with that particular device is that your SIP username MUST be all-numeric. i have never been able to get it to register otherwise. maybe it's an Asterisk thing. |
| 19:56.50 | Defraz | Eventually I want to simulate a PRI the other way too. Like have it connect sip to my asterisk box and deliver a PRI to a older pbx. |
| 19:57.21 | Defraz | I don't even know where to start. I have my cisco 1760 doing it but not this Adtran. |
| 19:57.30 | bmoraca_work | Defraz: well, are you having trouble with a particular part of the config? |
| 19:57.32 | Defraz | So if you have a config I could just look at to get some ideas. |
| 19:58.16 | *** part/#asterisk kalib (n=lkhlui@osiris.aspec.com.br) |
| 19:58.24 | Defraz | of where to go. |
| 19:59.10 | bmoraca_work | Defraz: the Adtran KB would be a great place to start. there's a walkthrough that will get you about 75% of the way there. the rest should be pretty easy to deduce. these devices are very similar to Cisco (though way easier because you don't have to screw with dial-peers) |
| 19:59.16 | RichardLynch | I'm working on a custom module for freepbx. I have written a foo_get_config() function. It's not getting called. Is this strictly a freebpx issue, or is this actually part of asterisk?... |
| 19:59.47 | bmoraca_work | Defraz: i'll see if i can find you the link |
| 20:00.06 | bmoraca_work | RichardLynch: #freepbx |
| 20:00.36 | *** join/#asterisk dkirker-openmobl (n=dkirker@openmobl/ceo/dkirker) |
| 20:00.36 | bmoraca_work | Defraz: KB home for that device: http://kb.adtran.com/display/2/index.asp?c=12&cpc=BrduHgTSN7q8U34W3REU5Gb07vRy0ko8jy4dp5CbvV82dL&cid=2&cat=2037&catURL=kb%3D1%26L1cid%3D1007%26L2cid%3D1021%26L3cid%3D2037%26level%3D4%26&r=0.1092951 |
| 20:00.47 | Defraz | perfect thanks |
| 20:00.53 | Defraz | I hear good things about them. |
| 20:01.02 | bmoraca_work | Defraz: sample config: http://kb.adtran.com/display/2/kb/article.asp?aid=3371 |
| 20:01.08 | bmoraca_work | Defraz: i love em |
| 20:01.10 | RichardLynch | bmoraca_work: Thanks. Haven't had much luck there, so far, but will be more patient, I guess... :-) |
| 20:01.59 | Defraz | SO the other way around pri to sip |
| 20:02.02 | bmoraca_work | Defraz: like I said, that sample config will get you started, but won't get you 100% of the way there. also, the "debug sip cldu" debug mode WILL be your fried |
| 20:02.03 | Defraz | should be about the same. |
| 20:02.18 | Defraz | okay cool cool. |
| 20:02.19 | Defraz | thanks |
| 20:02.22 | Defraz | I will star tfrom there |
| 20:02.24 | bmoraca_work | Defraz: yep, only differences are the settings of the PRI virtual interface (have it behave like user instead of network, etc) |
| 20:03.00 | Defraz | cool beans. thanks |
| 20:03.04 | *** part/#asterisk RichardLynch (n=RichardL@c-98-193-36-91.hsd1.il.comcast.net) |
| 20:03.07 | bmoraca_work | Defraz: also, your grouped-trunk dial patterns need to be more precise going the other way |
| 20:03.25 | titter | How does the power go out on a perfect day in Florida at 3pm? Old people who hit power poles ... |
| 20:03.44 | bmoraca_work | Defraz: also, remember that only T1 0/3 and T1 0/4 can be used as PRIs and will need T1 crossovers to connect them to CPE equipment (and probably your provider as well) |
| 20:04.16 | bmoraca_work | lunchtime! |
| 20:04.17 | *** join/#asterisk ktwilight_ (n=keliew@110.49-240-81.adsl-dyn.isp.belgacom.be) |
| 20:04.46 | Chainsaw | titter: We use underground cables in the UK. That might be an idea... |
| 20:05.09 | titter | It's mixed here depending on the location |
| 20:05.11 | Naikrovek | Chainsaw: new neighborhoods in the US almost all have underground power and communication |
| 20:05.20 | *** part/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 20:05.23 | Defraz | Thanks |
| 20:05.31 | Naikrovek | but it's a real pain to retrofit existing infrastructure to be underground |
| 20:05.38 | Naikrovek | and not worth it financially |
| 20:05.40 | titter | My development has underground lines ... problem is outside of my development is still poles. |
| 20:06.12 | titter | Also Florida ... is about 6 feet above sea level where I am at ... can't dig very deep |
| 20:06.47 | titter | Oh well <3 battery backups. |
| 20:07.28 | Naikrovek | generac |
| 20:07.29 | *** join/#asterisk t_ (i=tom@freenode/staff/tomaw) |
| 20:07.51 | Naikrovek | probably not feasible at home, but for business generac generators are dirt cheap |
| 20:08.01 | Naikrovek | $5k for a 20kW generator |
| 20:08.02 | titter | yup, at home today lol |
| 20:08.08 | titter | We have solar at my office |
| 20:08.20 | Naikrovek | yeah but solar can't power the whole office, can it? |
| 20:08.27 | titter | We store enough energy to go almost an entire business day |
| 20:08.33 | Naikrovek | oh nice |
| 20:08.46 | titter | It's a lot of panels, but the comapny got a very nice tax refund |
| 20:08.53 | Naikrovek | how many batteries |
| 20:09.38 | titter | Off the top of my head I don't know ... I would have to ask the owner. From what I was told it was enough to store 7-8 hours of our usage |
| 20:09.49 | Naikrovek | how many people in the office |
| 20:09.54 | titter | 20 people |
| 20:10.01 | Naikrovek | ah that's not too bad then |
| 20:10.04 | titter | Nah |
| 20:10.18 | Naikrovek | custom inverters/switchers or some vendor solution? |
| 20:10.20 | titter | 3 servers, some switches, 30 computers at max, and about 25 Polycoms |
| 20:10.29 | titter | Local vendor |
| 20:10.48 | titter | I had no part in it, besides be curious to what it does |
| 20:10.58 | Naikrovek | tripplite has some UPS controller thing that you can hook auto/marine batteries up to |
| 20:11.12 | titter | nice |
| 20:11.16 | Naikrovek | charge controller / inverter thing |
| 20:11.29 | Naikrovek | but it only does 4200W I think |
| 20:11.36 | titter | I have a small battery backup at home, enough to handle my asterisk server, my desktop, and routers for 30 minutes |
| 20:11.36 | Naikrovek | if I did that here, I'd need many |
| 20:12.14 | titter | If i build a new home it will have a generator for backup |
| 20:12.43 | titter | propane system like genrac |
| 20:12.55 | Naikrovek | generac can run on natural gas as well |
| 20:13.01 | bmoraca_work | Defraz: if you run in to a specific issue, i can certainly help you with that |
| 20:13.01 | Naikrovek | natural gas = no refilling |
| 20:13.02 | titter | no natural in florida |
| 20:13.06 | Naikrovek | ah |
| 20:13.08 | Naikrovek | why not |
| 20:13.23 | *** join/#asterisk gushi (n=danm@prime.gushi.org) |
| 20:13.43 | titter | not in my area |
| 20:13.48 | titter | im sure its somewhere in florida |
| 20:13.50 | Naikrovek | huh |
| 20:13.52 | Naikrovek | okay |
| 20:14.06 | titter | ive never looked because it's not common here |
| 20:14.15 | gushi | Hey all...I tried googling the asterisk site, but didn't find a ready answer...what's rough specs for an asterisk server that may need to handle a t1's worth of calls simultaneous, and still be comfortable to transcode if need be. |
| 20:14.28 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 20:14.31 | Naikrovek | there's some law in illinois that natrual gas lines be at least 6ft underground, so maybe that's it |
| 20:14.36 | Naikrovek | given your height above sea level |
| 20:14.50 | titter | I would assume it has to do with the sea level |
| 20:14.56 | titter | I am sure north florida has it |
| 20:15.58 | titter | Most homes on natural gas have propane tanks outside |
| 20:16.04 | ManxPower-work | Naikrovek: I assume that is because the ground freezes and thaws to a significant depth there. |
| 20:16.46 | ManxPower-work | There are similar rules about water lines in the north. Where I am in the south, I think 18" is as far down as water lines must be buried |
| 20:16.52 | Naikrovek | ManxPower-work: well normally we never see more than a week at a time of below freezing; usually there are days of time between freezes that prevent the freeze from going down more than a foot or so |
| 20:16.57 | Naikrovek | ManxPower-work: but you may be right |
| 20:17.28 | Naikrovek | all of illinois certainly does not see the same length of freezing. chicago can be frozen for months at a time |
| 20:17.44 | bmoraca_work | i can't wait to move in to a house with gas appliances...i'm sick of cooking on electric |
| 20:17.59 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:18.01 | Naikrovek | gas stoves are nice |
| 20:18.07 | Naikrovek | turn off the heat and the heat goes away |
| 20:18.15 | Naikrovek | turn on the burner and everything gets hot fast |
| 20:18.22 | Naikrovek | induction stoves are even faster than gas tho |
| 20:18.30 | Naikrovek | induction stovetops i mean |
| 20:18.33 | Naikrovek | those are fascinating |
| 20:18.52 | Naikrovek | boil a gallon of water in 90s |
| 20:18.52 | bmoraca_work | i don't like them. had one in the last house i lived in. i want good old gas. |
| 20:19.06 | Naikrovek | yeah gas is nice |
| 20:19.35 | Naikrovek | after a decade of it though you start to notice that your kitchen walls are starting to turn yellow |
| 20:19.45 | Naikrovek | gotta clean that off and/or repaint |
| 20:20.09 | bmoraca_work | i'd imagine that depends much more on what you cook than how you cook it |
| 20:20.22 | bmoraca_work | i wouldn't mind a house smelling like garlic though :) |
| 20:22.13 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 20:26.41 | *** join/#asterisk Frost7217 (n=Frost721@pool-173-57-96-245.dllstx.fios.verizon.net) |
| 20:28.39 | *** join/#asterisk Frost7217 (n=Frost721@pool-173-57-96-245.dllstx.fios.verizon.net) |
| 20:30.39 | Frost7217 | My organization is considering implementing asterisk. We have agents in the field on municipal wifi and use their VOIP phone's as cell phones. If they use the e911 service, will they be able to connect to the local 911 service, or the 911 service associated with the companies VOIP system? |
| 20:31.11 | Naikrovek | probably the company's voip system |
| 20:35.56 | bmoraca_work | Frost7217: e911 can be done a few ways. usually, the LIDB lookup is set up in such a way it uses the ANI information of the caller to route to the correct PSAP |
| 20:36.56 | bmoraca_work | Frost7217: that said, there are much more complex and feature-rich ways to do this, and I don't believe that Asterisk supports those yet. you can, however, contract with a 3rd party e911 service provider rather than using your regular termination provider and gain access to some of these |
| 20:37.30 | *** join/#asterisk L2SHO (n=adam@mail.voicepulse.com) |
| 20:37.37 | bmoraca_work | Frost7217: that could include, even, a GPS lookup of the caller's truck and forwarding that information to the provider. |
| 20:37.49 | jo8330 | bye all, take care |
| 20:38.40 | Frost7217 | interesting, thank you |
| 20:39.09 | bmoraca_work | Frost7217: in short, if your remote users are stationary and your provider allows you to have multiple pilot numbers, there's no problem. in long, anything is possible provided you want to pay for it. |
| 20:40.00 | L2SHO | if I do $[${flag}&${REGEX("${somevar}" ${someregex})}] and ${flag} is FALSE, does the regex still get executed, or is the expression code smart enough to know that it should immediately return 0? |
| 20:40.52 | bmoraca_work | L2SHO: i imagine that the regex will still be executed because the expression isn't necessarily false until all elements of it are computed and evaluated |
| 20:41.23 | bmoraca_work | L2SHO: for instance, two false components results in the expression returning true. |
| 20:41.37 | L2SHO | FALSE & FALSE should still be FALSE |
| 20:42.29 | Naikrovek | false & false = false, doesn't it |
| 20:42.33 | bmoraca_work | L2SHO: yes, it should. but the operator is examined after the operands |
| 20:43.20 | Naikrovek | for an & it should just stop if one of them is false. in java you use %% for that |
| 20:43.21 | Naikrovek | um |
| 20:43.22 | Naikrovek | && |
| 20:44.06 | L2SHO | bmoraca_work: thats not what I wanted to hear :( |
| 20:44.29 | L2SHO | Naikrovek: you would think so, |
| 20:44.43 | bmoraca_work | L2SHO: you should confirm in the source code (the only place you'll find this answer) before making any decisions, but I suspect that will be the order |
| 20:45.08 | Naikrovek | i think bmoraca_work is right. it'll evaluate both sides, then compare them using the operator and return the result |
| 20:45.08 | L2SHO | bmoraca_work: ya, I was just about to ask if anyone knows where in the source this stuff would be |
| 20:45.16 | Naikrovek | probably not any wisdom in the evaluation other than that |
| 20:46.00 | Skeeter- | anyone got a good tutorial to make NUT UPS work |
| 20:46.18 | Skeeter- | or anything similar that would shutdown a server froma UPS APC battery |
| 20:46.31 | Naikrovek | powerd? |
| 20:47.21 | Chainsaw | apcupsd? |
| 20:47.35 | *** join/#asterisk smooth_penguin (n=smoove@59.95.50.25) |
| 20:47.48 | Naikrovek | there are daemons that monitor UPS' and you can make them do whatever you want |
| 20:48.12 | Skeeter- | Naikrovek, which one do you suggest |
| 20:48.15 | *** join/#asterisk girlny (n=girlny@CPE00195b4be142-CM001a668ec076.cpe.net.cable.rogers.com) |
| 20:48.19 | Skeeter- | powerd forum seems to be down |
| 20:48.29 | Naikrovek | all i've used is powerd |
| 20:48.50 | Naikrovek | check forums for whatever distribution you're using |
| 20:48.55 | Naikrovek | someone will have asked this before |
| 20:49.30 | girlny | hellow my asterisk server cpu just spikes to 100% every few seconds , i did some testing and it only hapens when i set up sip extensions ... any clue of how can i solve this ? |
| 20:50.14 | Naikrovek | does it do that repeatedly forever or ONLY when you add an extension |
| 20:50.35 | girlny | it does it for ever and when i delete the extensions it goes |
| 20:50.50 | Naikrovek | girlny: also, are you using FreePBX or Trixbox or Elastix or one of those |
| 20:51.38 | girlny | using freepbx only and asterisk 1.6.1.5 |
| 20:53.53 | bmoraca_work | girlny: it's likely httpd and mysqld and php doing everything they need to do to update the database and data files. use "top" to figure out which service is spiking and then look at the logs for that service to determine what is actually happening |
| 20:53.56 | Naikrovek | you may have better luck if you ask in #freepbx |
| 20:56.34 | girlny | 2228 asterisk 15 0 39096 11m 7748 S 99 0.7 8:08.12 asterisk |
| 20:56.48 | Naikrovek | wow 8 hours of cpu time |
| 20:56.57 | Naikrovek | or is that 8 days |
| 20:57.00 | Naikrovek | 8 something |
| 20:57.34 | bmoraca_work | that's not much...one of my boxes is at 275 hours |
| 20:57.44 | Naikrovek | of asterisk cpu time? |
| 20:57.48 | Naikrovek | that's not uptime |
| 20:57.48 | bmoraca_work | yep |
| 20:57.53 | Naikrovek | wow |
| 20:57.54 | bmoraca_work | CPU time, not uptime |
| 20:58.08 | girlny | so what can this be |
| 20:58.09 | girlny | ? |
| 20:58.22 | Naikrovek | well something is up |
| 20:58.29 | bmoraca_work | girlny: i suspect there's a loop in some freepbx module or script or something. |
| 20:58.30 | girlny | if i delete the extension it fully disapeers |
| 20:58.41 | Naikrovek | but we'd just be guessing if we tried to tell you what it was at this point |
| 20:58.45 | girlny | like goes to normal |
| 20:59.15 | bmoraca_work | girlny: sounds like there's a loop in freepbx, whether it be in the AMI login or something else. chances are, there's nothing we can really do to help you. |
| 20:59.54 | girlny | http://pastebin.com/m5f448ca1 |
| 21:00.22 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 21:00.42 | girlny | thats the sip_general_custom.conf |
| 21:01.01 | girlny | can it be something to do with that |
| 21:01.12 | Naikrovek | how often does the cpu spike |
| 21:01.38 | Naikrovek | i think canreinvite is only "yes" or "no" |
| 21:01.39 | bmoraca_work | girlny: probably not. but, then again, i don't know everything that freepbx is doing behind the scenes |
| 21:02.00 | bmoraca_work | nonat is appropriate...it means "yes if the device is not behind a NAT" |
| 21:02.55 | girlny | it spikes every 10 secs or 5 |
| 21:03.23 | bmoraca_work | girlny: seriously, we cannot help you here. you need to talk to someone who has a similar setup to you, and that's not here |
| 21:05.00 | Naikrovek | #freepbx |
| 21:05.38 | *** join/#asterisk fofware (n=chatzill@host216.190-30-166.telecom.net.ar) |
| 21:06.50 | girlny | ok but there is any way i can check the full asterisk log , becase when i check i dont see allot of activity |
| 21:07.06 | girlny | also when i log in to asterisk -r i get -- Remote UNIX connection |
| 21:07.24 | Naikrovek | /var/log/asterisk/full has the full log i believe |
| 21:07.43 | Naikrovek | it'll be pretty bare until you start doing things like making calls |
| 21:08.40 | girlny | what im tring to untherstand is that the loop wont show ? |
| 21:09.40 | vader- | so are there any suitable Asterisk GUI's yet? |
| 21:10.10 | vader- | that are open source |
| 21:10.55 | voipmonk | frepbx |
| 21:11.08 | girlny | freepbx |
| 21:11.10 | vader- | that are good to manage 100+ phones? |
| 21:11.12 | voipmonk | maybe the one you build |
| 21:11.12 | kaldemar | oh the irony |
| 21:11.46 | ChannelZ | OT - does anyone know the MIME type for .m4v vids for the iPhone that it likes? |
| 21:15.14 | *** join/#asterisk ruied (n=ruied@92.250.24.120) |
| 21:20.53 | Katty | oh |
| 21:20.53 | Katty | hi |
| 21:20.57 | Katty | i fell asleep |
| 21:21.39 | p3nguin | Why does SendFAX use /var/www/faxes/ to find the files, and is there any way to change it? |
| 21:22.23 | p3nguin | Specifying the full path to the file causes /var/www/faxes/ to be prepended onto the front of the full path specified. |
| 21:23.38 | p3nguin | Wait, stupid macro. |
| 21:23.41 | p3nguin | disregard! |
| 21:24.01 | Katty | mymymy a mourning dove, ground bird, is sitting up on the tray feeder |
| 21:24.24 | Katty | someone must be really hungry! |
| 21:24.55 | bmoraca_work | p3nguin: i think someone was kvetching about that a few weeks ago :P |
| 21:28.54 | *** join/#asterisk Cain` (n=Geek@unaffiliated/cain) |
| 21:32.18 | *** join/#asterisk TimeRider (n=steve@78.32.26.1) |
| 21:32.39 | Katty | holy pasta fajioli |
| 21:32.43 | Katty | we got SNOW! |
| 21:33.02 | *** join/#asterisk daidoji (n=daidoji7@99.48.50.198) |
| 21:33.52 | Katty | p3nguin: SNOWBALL FIGHT IN 10MIN BE THERE |
| 21:34.00 | p3nguin | haha |
| 21:34.11 | [TK]D-Fender | Katty: Want snow? TAKE MINE DAMMIT |
| 21:34.18 | Katty | i fell asleep on the couch earlier |
| 21:34.27 | Katty | i guess it's been snowing for all this time |
| 21:34.35 | bmoraca_work | it snowed in Central California earlier this winter...hasn't snowed here in 40 years |
| 21:35.08 | Katty | snow is very very very common here |
| 21:35.46 | bmoraca_work | 90 minute drive to snow here...90 minute drive to ocean, as well...300 minute drive to Disneyland, though :P |
| 21:37.20 | *** join/#asterisk neurosys (n=neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 21:38.30 | *** join/#asterisk evil_gordita (n=evilgord@ip70-188-50-186.rn.hr.cox.net) |
| 21:39.03 | Katty | pretty red bellied woodpecker |
| 21:39.11 | Katty | wolfing down peanuts |
| 21:39.19 | p3nguin | [Jan 29 15:38:32] ERROR[3054]: res_fax_digium.c:1817 dgm_fax_start: FAX handle 0: failed to queue document '/tmp/test.tif' |
| 21:39.23 | p3nguin | What could cause this? |
| 21:39.37 | Katty | format |
| 21:39.52 | p3nguin | /tmp/test.tif: TIFF image data, little-endian |
| 21:40.16 | Katty | i'm not sure what you're trying to tell me with that line. |
| 21:40.23 | Katty | but i will assume this isn't a windows TIFF image. |
| 21:40.40 | p3nguin | It's just a TIFF image. |
| 21:40.53 | p3nguin | I created it using bmp2tiff. |
| 21:41.11 | [TK]D-Fender | where do we see permissions, ertc? |
| 21:41.24 | Katty | well, Just a Tiff.... |
| 21:41.26 | p3nguin | -rw-r--r-- 1 root root 2694 Jan 29 15:37 /tmp/test.tif |
| 21:41.45 | Katty | you can print to a tiff printer and not get a proper document |
| 21:41.58 | Katty | i don't really understand why, but i've had it happen to me |
| 21:42.10 | Katty | i just always assumed it was because windows is retarded |
| 21:42.16 | [TK]D-Fender | p3nguin: root? And * is running as root? |
| 21:42.26 | *** join/#asterisk Caplain (i=shayne@caplain.loves.boys.fbi.gov.silverelitez.org) |
| 21:42.29 | p3nguin | It is readable by ALL. |
| 21:42.40 | p3nguin | Asterisk is running as "asterisk" |
| 21:42.49 | Katty | female woodpecker! |
| 21:42.58 | Katty | downy woodpecker at that :> |
| 21:43.04 | Katty | i named her Cher |
| 21:43.27 | [TK]D-Fender | p3nguin: changer the perms & owner and test |
| 21:43.37 | Katty | that squirrel is retarded. it's snowing heavily....and the wind is blowing the snow around like a blizzard |
| 21:43.48 | p3nguin | [tk]d-fender: What do you want the perms to be? It's already readable by everyone. |
| 21:43.52 | Katty | i guess he got snow in his ear, cause he ran to the other side of the tree trunk and was pawing at it |
| 21:44.01 | Katty | but he won't leave...he's just sitting there stuffing his face. |
| 21:44.01 | p3nguin | I'll happily do it, but I need to know what you have in mind. |
| 21:44.18 | Katty | Major General has already gone to his squirrely nest up in the neighbor's tree |
| 21:45.16 | *** join/#asterisk MedicineMan (n=medicine@75.87.82.200) |
| 21:45.27 | MedicineMan | hey quick question for you all |
| 21:45.35 | Katty | the answer is 3.14 |
| 21:45.38 | MedicineMan | rofl |
| 21:45.39 | MedicineMan | lol |
| 21:45.40 | Katty | or possibly 42 |
| 21:45.51 | MedicineMan | i'm trying to run the make menuselect |
| 21:45.55 | Katty | and yes, starlings DO like banana. |
| 21:45.55 | MedicineMan | on 1.4.28 |
| 21:46.09 | MedicineMan | and it tells me ncurses needs to be installed |
| 21:46.14 | MedicineMan | but its already isntalled |
| 21:46.17 | MedicineMan | what am i missing |
| 21:46.22 | p3nguin | Can fax be sent over IAX2 channels as long as I use G.711? |
| 21:46.24 | Chainsaw | MedicineMan: Is this a Debian system? |
| 21:46.30 | MedicineMan | Centos |
| 21:46.41 | Katty | Centos, the Fresh Maker |
| 21:46.46 | bmoraca_work | MedicineMan: you need ncurses-devel as well |
| 21:46.49 | Chainsaw | MedicineMan: Hm, okay. On a Debian system I'd say you're missing the -dev package for ncurses. |
| 21:47.01 | MedicineMan | i have that installed as well |
| 21:47.04 | p3nguin | MentOS, the fresh operating system |
| 21:47.06 | bmoraca_work | MedicineMan: install it again |
| 21:47.08 | Chainsaw | MedicineMan: Not sure what the RedHat equivalent is. I do dislike distributions that only install half of a software pack. |
| 21:47.22 | Katty | bmoraca_work: ehhh |
| 21:47.35 | Katty | bmoraca_work: hmm |
| 21:47.53 | MedicineMan | how would i reinstall it. yum remove ncurses? |
| 21:47.56 | Katty | bmoraca_work: i'm not sure how i feel about mintoy operating systems |
| 21:48.18 | bmoraca_work | MedicineMan: no, "yum install ncurses-devel" |
| 21:48.20 | Katty | aww. a starling ran Cher off :< |
| 21:48.35 | MedicineMan | i've already done that one as well |
| 21:48.43 | bmoraca_work | MedicineMan: humor me. |
| 21:48.45 | Katty | makes a note to hang another suet feeder |
| 21:48.45 | MedicineMan | both ncurses and ncurses-devel is in stalled |
| 21:48.59 | Qwell | MedicineMan: and what happened when you re-ran configure? |
| 21:49.08 | MedicineMan | ]# rpm -qa|grep ncurses |
| 21:49.08 | MedicineMan | ncurses-devel-5.5-24.20060715 |
| 21:49.08 | MedicineMan | ncurses-5.5-24.20060715 |
| 21:49.14 | MedicineMan | it runs fine |
| 21:49.33 | Katty | p3nguin: you think dominos is running? |
| 21:49.46 | Katty | p3nguin: and by running i mean delivering |
| 21:49.56 | p3nguin | katty: Absolutely. |
| 21:50.13 | eppigy | is your refridgerator running? |
| 21:50.13 | Katty | in this weather!? |
| 21:50.15 | MedicineMan | make[1]: Entering directory `/usr/src/asterisk-1.4.28/menuselect' |
| 21:50.16 | MedicineMan | make[2]: Entering directory `/usr/src/asterisk-1.4.28/menuselect' |
| 21:50.16 | MedicineMan | make[2]: `menuselect' is up to date. |
| 21:50.16 | MedicineMan | make[2]: Leaving directory `/usr/src/asterisk-1.4.28/menuselect' |
| 21:50.16 | MedicineMan | make[1]: Leaving directory `/usr/src/asterisk-1.4.28/menuselect' |
| 21:50.16 | MedicineMan | ************************************************** |
| 21:50.18 | MedicineMan | *** Install ncurses to use the menu interface! *** |
| 21:50.18 | eppigy | and by running I mean delivering |
| 21:50.19 | MedicineMan | ************************************************** |
| 21:50.21 | MedicineMan | menuselect changes NOT saved! |
| 21:50.23 | Chainsaw | MedicineMan: No flooding! |
| 21:50.26 | Qwell | MedicineMan: and what happened when you re-ran configure? |
| 21:50.31 | Qwell | ~pastebin |
| 21:50.31 | infobot | [~pastebin] 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 |
| 21:50.39 | p3nguin | medicineman: How about a little less flooding and a little more pastebin? |
| 21:50.39 | Katty | eppigy: my fridge is so NOT delivering |
| 21:50.47 | Katty | eppigy: it's on strike, i think |
| 21:50.50 | MedicineMan | wow |
| 21:50.52 | MedicineMan | newb here |
| 21:50.59 | Katty | eppigy: having recently joined a union |
| 21:51.00 | eppigy | Katty: :< |
| 21:51.04 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 21:51.09 | [TK]D-Fender | MedicineMan: Did you redo "./configure"? |
| 21:51.13 | Katty | eppigy: otherwise known as empty. |
| 21:51.15 | eppigy | unions are terrible |
| 21:51.16 | MedicineMan | yes |
| 21:51.25 | MedicineMan | it ran all fine.. no errors |
| 21:51.25 | Katty | eppigy: they can be. |
| 21:51.26 | praet | check the makefile for where its looking for ncurses |
| 21:51.31 | eppigy | many businesses would be profitable if not for union |
| 21:51.32 | eppigy | s |
| 21:51.49 | Katty | eppigy: and many people, like my father, would already have been retired if it wasn't for unions |
| 21:51.56 | Katty | eppigy: but unions can serve a purpose. |
| 21:52.01 | eppigy | word |
| 21:52.02 | p3nguin | Do I have to use SIP to send a fax with SendFAX? |
| 21:52.17 | Katty | p3nguin: mmmm, no that i'm aware of |
| 21:52.21 | Katty | p3nguin: you want my blog page? |
| 21:52.34 | [TK]D-Fender | p3nguin: Expect failure on anything by DAHDI and SIP + T.38 |
| 21:52.41 | p3nguin | katty: If it's the one with the video on it, I saw it already. |
| 21:52.43 | Katty | i seem to recall something sip not working well |
| 21:52.47 | Katty | p3nguin: err |
| 21:53.13 | p3nguin | I guess I can switch from IAX2 over to SIP just to test it out. |
| 21:53.25 | Katty | http://42ndgeekstreet.blogspot.com/2009/11/asterisk-faxing.html |
| 21:53.33 | praet | MedicineMan: do you have a /usr/lib/libncurses.so |
| 21:53.42 | MedicineMan | let me check |
| 21:53.53 | Katty | p3nguin: ^-- url |
| 21:53.59 | Katty | p3nguin: go to very very very bottom |
| 21:54.05 | Qwell | MedicineMan: or ignore me and waste your time. that's fine too. |
| 21:54.07 | Katty | p3nguin: adjust as needed. |
| 21:54.14 | MedicineMan | yes i do |
| 21:54.39 | MedicineMan | i'm new to this IRC stuff, i'm not trying to ignore nayone |
| 21:54.43 | p3nguin | katty: Yeah, I already read that, and it doesn't seem to adress SIP/IAX2. |
| 21:54.49 | Qwell | re-run configure, and show me the output |
| 21:54.51 | Qwell | ~pastebin |
| 21:54.52 | infobot | [~pastebin] 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 |
| 21:55.20 | p3nguin | katty: I don't have any Zap channels, so I can't duplicate your Dial() command. |
| 21:55.38 | Katty | p3nguin: i still use dial for sip channels. |
| 21:55.50 | Katty | p3nguin: but i've never tried to send a fax out on one |
| 21:56.10 | p3nguin | let me switch to SIP and see if it makes any difference. |
| 21:56.26 | Katty | holy crap |
| 21:56.28 | MedicineMan | http://pastebin.com/m7c532a0 |
| 21:56.30 | Katty | someone almost drove into my yard |
| 21:56.36 | Katty | err slid into my yard |
| 21:56.42 | MedicineMan | i wasn't able to capture it all but that is some |
| 21:56.52 | MedicineMan | er most |
| 21:57.09 | Chainsaw | Katty: And the joke's on them, because you've got it on video. |
| 21:57.16 | Katty | i totoally do!!! |
| 21:57.28 | Katty | it's not like they're going to hit anything but a hill |
| 21:57.56 | Katty | but still.... |
| 21:58.38 | Katty | starlings are eattin that banana up |
| 21:58.43 | Katty | it's amazing to watch |
| 21:59.14 | MedicineMan | did that help qwell |
| 21:59.32 | p3nguin | Using SIP, same failure. |
| 21:59.37 | Katty | bummer. |
| 22:02.11 | p3nguin | I don't know what else to try; * simply says it cannot queue the file. |
| 22:02.21 | p3nguin | Oh, wait... where would it queue the file? |
| 22:04.18 | MedicineMan | @Qwell did that help |
| 22:04.19 | p3nguin | Every directory in /var/spool/asterisk/ is owned by asterisk and is writeable, so that's probably not the cause. |
| 22:04.50 | Qwell | MedicineMan: make -C menuselect/ clean |
| 22:05.14 | *** join/#asterisk torrancew (n=torrance@ip70-186-186-21.br.br.cox.net) |
| 22:05.14 | [TK]D-Fender | MedicineMan: You also need libtermcap <- |
| 22:05.21 | [TK]D-Fender | MedicineMan: You also need libtermcap-devel <- |
| 22:05.23 | [TK]D-Fender | also |
| 22:05.30 | eppigy | HOLLER |
| 22:05.31 | Katty | p3nguin: did you recieve a fax yet? |
| 22:05.54 | *** join/#asterisk Primer (n=daniel@www.ceregatti.org) |
| 22:05.54 | torrancew | would anyone reccommend a panasonic cordless for use with a PAP2 ATA and Asterisk? Just need basic features, and need the handset to ring - our old handsets do not |
| 22:05.55 | MedicineMan | both of those are installed |
| 22:06.04 | p3nguin | katty: No, that's why I am trying to send outbound. I'm trying to use HP's "we'll fax you back" thingy. |
| 22:06.13 | pigpen | Hi all, I ran into an issue a couple of years ago that I want to revisit. Page Groups in 1.4.21.2 is there a max # of sip extensions?? I am needing one page of 100 and two groups of 50'ish. |
| 22:06.15 | Katty | p3nguin: see if you can recieve one |
| 22:06.33 | pigpen | In the past I have manually made several groups that are called at once. |
| 22:06.41 | p3nguin | katty: You're going to fax me? |
| 22:06.46 | pigpen | Successfully hitting 300+ |
| 22:07.03 | Primer | Anyone know if some features of Polycom 501's are simply not settable via the phone itself, or the web based admin of the phone? I'd like to enabled url dialing, but I can't find any example other than via cfg files |
| 22:07.12 | Katty | p3nguin: i am sitting at home on the couch |
| 22:07.15 | Katty | p3nguin: in my pjs. |
| 22:07.20 | Katty | p3nguin: no, i'm not going to fax you |
| 22:07.28 | MedicineMan | http://pastebin.com/m2526ae69 |
| 22:07.34 | MedicineMan | check that out Qwell |
| 22:07.43 | p3nguin | katty: All the free online test-yer-fax apps either don't work (broken) or I have reached my quota and they won't send any more. |
| 22:07.46 | pigpen | Primer, you will want to do the automated deployment as that is where the real features are at. |
| 22:07.48 | Katty | p3nguin: but if you can't recieve faxes it might shed some light on something |
| 22:07.48 | MedicineMan | thats from doing your make -c menuselect/ clean |
| 22:07.54 | MedicineMan | then doing ./configure |
| 22:07.59 | Qwell | -C, not -c |
| 22:08.09 | MedicineMan | right |
| 22:08.10 | p3nguin | katty: If I could receive, I wouldn't be trying to send this fax right now. |
| 22:08.12 | MedicineMan | thats what i did |
| 22:08.23 | Qwell | show me everything. not just parts |
| 22:08.23 | Primer | pigpen: except these are phones that are provided to us by a third party, and I don't have access to said files |
| 22:08.58 | pigpen | if they reside under the control of your dhcp server it is possible. |
| 22:09.06 | Primer | pigpen: in the past I was able to sniff the network. It was using ftp with clear text everything. I was then able to make this phone use my ftp server, and gain access |
| 22:09.15 | Primer | but they've since switched to https |
| 22:09.17 | Primer | well, I do |
| 22:09.53 | Primer | but they've also flashed to a newer firmware, and I'm not sure the files I have are compatible |
| 22:09.53 | pigpen | you have the physical device, the power is in your hands.... |
| 22:10.10 | pigpen | you can get the files you need from pretty much any polycom distributor. |
| 22:10.31 | pigpen | just download the app set you have loaded...or better yet...upgrade it. |
| 22:11.21 | Primer | pigpen: why is it that not all features are exposed via the phone's config UI or web interface? |
| 22:11.28 | Primer | laziness? |
| 22:11.56 | pigpen | they were designed for mass deployment, no since having all this "overhead" at the device. Also users can jack with them. |
| 22:12.39 | Primer | I mean, I have the damn |
| 22:12.41 | Primer | err |
| 22:12.45 | Primer | I mean, I have the damn password |
| 22:12.46 | pigpen | it is much easier to modify a single file and have that setting push out to hundreds of phones. |
| 22:13.02 | Primer | yes, I understand that |
| 22:13.15 | Primer | but why limit the phone's built in ability to set these things? |
| 22:13.26 | pigpen | you will not be able to "exercise: the ability of that phone until you do a network deployment. |
| 22:13.31 | pigpen | Then by Snom. |
| 22:13.43 | pigpen | s/by/buy |
| 22:14.22 | Primer | not sure what you mean |
| 22:14.23 | *** part/#asterisk MedicineMan (n=medicine@75.87.82.200) |
| 22:14.36 | pigpen | snom has it all available at each phone. |
| 22:14.43 | pigpen | and can have mass deployment. |
| 22:15.23 | *** join/#asterisk e4 (n=e4@rrcs-76-79-59-194.west.biz.rr.com) |
| 22:17.35 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:20.16 | pigpen | not to be blunt, but that is the way it is. |
| 22:21.19 | *** join/#asterisk albertoandrade (n=albertoa@189.101.104.94) |
| 22:24.24 | Primer | I changed DHCP option 66 to point to a local web server, and the phone is asking for 000000000000.cfg |
| 22:24.33 | Primer | is that the "new" sip.cfg? |
| 22:24.41 | Primer | it never asks for sip.cfg |
| 22:25.14 | pigpen | have you ever setup a polycom phone deployment? |
| 22:25.50 | pigpen | I have always used tftp or ftp. |
| 22:25.56 | pigpen | never http. |
| 22:26.07 | carrar | Primer, use FTP |
| 22:26.07 | pigpen | google it, there are many good articles. |
| 22:26.12 | bmoraca_work | Primer: it may be that some of the features are not conviniently configurable via either interface. |
| 22:26.52 | Primer | meh, this is what they configured |
| 22:26.57 | Primer | their option 66 is https |
| 22:28.00 | bmoraca_work | Primer: option 66 simply lets you specify the NAME of a boot server. what your devices do with that NAME is up to the devices. |
| 22:28.14 | bmoraca_work | Primer: TFTP server IP address is Option 150 |
| 22:28.16 | Primer | pigpen: my goal is not a "polycom deployment". My goal is to 1) set the second, and perhaps third lines of this 501 to my asterisk without killing the first line, which goes to the SIP provider |
| 22:28.24 | Primer | 2) enable url dialong |
| 22:28.43 | Primer | the SIP provider that owns the phone, which is at my work |
| 22:28.52 | Primer | dialing, even |
| 22:29.45 | pigpen | you can setup some of these other lines in the web interface. but much of the options, which many find necessary, are only in the .cfg file for the phone/sip.cfg |
| 22:30.17 | pigpen | you can half ass this all you want, eventually, you will find yourself plowing into this. Granted, it is not easy. |
| 22:30.28 | Primer | I've actually had this all working before, but I achieved it by sniffing the network, watching the phone get its files in clear text over ftp, then using the sniffed credentials to get every file it got and have them locally, setting up an ftp server, then making it get those files from my local ftp |
| 22:30.30 | pigpen | I have done many, but each time, I always forget something stupid. |
| 22:30.34 | Primer | then I changed the files to suit me |
| 22:31.01 | Primer | pigpen: right, for example, you can't set the outbound proxy of the second and third lines via the web interface |
| 22:31.03 | pigpen | I have never had to "sniff" the network to find out what the phone is doing....just look at the configs. |
| 22:31.08 | pigpen | and google. |
| 22:31.28 | Primer | I had no config |
| 22:31.30 | pigpen | Primer, hence, forget the web interface.....do a mass phone deployment. |
| 22:31.41 | Primer | I was only able to get them once I sniffed them |
| 22:31.41 | pigpen | go download the polycom config files. |
| 22:31.57 | Primer | you keep saying mass deployment...there's only one phone! |
| 22:31.59 | pigpen | I think you are trying to make this more difficult than it really is. |
| 22:32.12 | bmoraca_work | Primer: would it not be easier to just call your provider and have them modify the configs on their end? |
| 22:32.30 | pigpen | Or hire someone who can set this up for you. |
| 22:32.36 | riddlebox | does anyone know if there is a site where people have made a collection of agi scripts? that you could browse and download? |
| 22:32.38 | Primer | bmoraca_work: if they were willing to do that, sure! "Hi, uhhh can you add a second line that going to my asterisk? Ok, thanks!" |
| 22:33.15 | bmoraca_work | Primer: if you're leasing the phone from them, they should have no problem modifying the phone's configuration files for you. I'm sure they'd prefer that to you screwing with the phone. |
| 22:33.35 | Primer | bmoraca_work: they were the ones that gave me the admin password :) |
| 22:33.39 | p3nguin | katty: I received a fax and it seems there is no problem with ReceiveFAX(). |
| 22:34.15 | bmoraca_work | Primer: and if they were willing to do that, why wouldn't they be willing to add a couple line appearance configs to your config file? |
| 22:35.10 | *** join/#asterisk corretico (n=laguilar@201.201.46.106) |
| 22:36.35 | Primer | bmoraca_work: not sure if they'd be willing to do that, but I never asked |
| 22:39.52 | p3nguin | katty: So I turned around and refaxed the one I received, since Asterisk created that tif, I figure it must be a valid format. It didn't show the failure. |
| 22:42.10 | *** part/#asterisk Tech_Travis (n=Travis@mail.techglia.com) |
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| 23:05.02 | Docteh | riddlebox: i dont think anybody has made such a site yet |
| 23:05.47 | riddlebox | Docteh, it would be nice if it did exist, just to be able to browse it and download ones you like |
| 23:14.47 | *** part/#asterisk torrancew (n=torrance@ip70-186-186-21.br.br.cox.net) |
| 23:16.18 | Katty | peeks in |
| 23:16.40 | Katty | herroes |
| 23:18.45 | Primer | ok, I copied the mac_address.cfg and sip.cfg that I was using before over to my web server, the phone found those and loaded them, and everything is back to normal :) |
| 23:18.59 | *** join/#asterisk c4t3l (n=c4t3l@c-76-31-57-251.hsd1.tx.comcast.net) |
| 23:19.06 | Katty | mmm, normality |
| 23:19.10 | c4t3l | hello world |
| 23:19.57 | Katty | c4t3l: ohai |
| 23:20.03 | Katty | c4t3l: do you have snow too? |
| 23:21.22 | c4t3l | not quite |
| 23:21.46 | c4t3l | Katty: it did snow here once or twice tho (houston, tx) |
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| 23:25.34 | Katty | ahhh |
| 23:25.47 | Katty | it's been snowin here since about 8am |
| 23:26.23 | Katty | 9.5hrs |
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| 23:32.53 | Katty | hi mister tee |
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| 23:43.06 | *** part/#asterisk Primer (n=daniel@www.ceregatti.org) |
| 23:43.48 | jaytee | hi Katty |
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