| 00:03.05 | ChannelZ | uh oh |
| 00:04.05 | p3nguin | hmm? |
| 00:05.28 | ChannelZ | nm |
| 00:06.17 | ChannelZ | Anyone have any crazy-buzy mailing lists to suggest? Trying to debug a totally bizarre email server problem and need a regular flow of crap coming in |
| 00:15.17 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 00:18.36 | artyx | anyone know n3glv's conference line # |
| 00:20.48 | *** join/#asterisk wonderworld (~w@dsdf-4d0a1153.pool.mediaWays.net) |
| 00:22.02 | *** join/#asterisk bytemaster (~qweq@host81-150-217-167.in-addr.btopenworld.com) |
| 00:27.50 | slav3_kitten | remember ageis you can use include => context to allow your local sets to outbound dial etc |
| 00:38.42 | wonderworld | hi ppl |
| 00:45.46 | navaismo | artyx, ask to n3glv |
| 00:47.24 | slav3_kitten | good engineering practices should be to put a hangup() for every extension right? |
| 00:48.30 | p3nguin | I have an explicit Hangup() on probably every extension that doesn't end in a Goto() or some variation of Goto(). |
| 00:48.52 | p3nguin | If the call stops within that extension, I end it with a Hangup(), even if it is not ever going to be used. |
| 00:49.33 | slav3_kitten | *nods* i just want to make sure i'm doing this right. i figured a hangup would terminate the call if something went wonky |
| 00:49.40 | p3nguin | Keep in mind, without a hangup and without any additional steps, the call is going to inevitably die anyway. |
| 00:50.06 | dijib | you guys still going? |
| 00:50.17 | slav3_kitten | what dijib ? |
| 00:50.28 | p3nguin | The Hangup() would only get used if the call is wonky AND the previous steps exited. |
| 00:51.07 | slav3_kitten | right |
| 00:51.43 | *** part/#asterisk ghost75 (~trechber@dslb-178-010-042-022.pools.arcor-ip.net) |
| 00:52.30 | slav3_kitten | i'm trying to get my dial plan to be decent looking and maybe try to play with the asterisk db |
| 01:05.36 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 01:09.32 | *** join/#asterisk debuggerboy (~debuggerb@117.196.164.203) |
| 01:09.43 | *** part/#asterisk debuggerboy (~debuggerb@117.196.164.203) |
| 01:14.06 | SeRi | my comcast downstream channels.... Freq 615.00 MHz |
| 01:14.11 | SeRi | WTF!!!!! |
| 01:14.53 | slav3_kitten | seems legit |
| 01:16.03 | *** join/#asterisk Sedated (~gokeefe@unaffiliated/sedated) |
| 01:16.12 | p3nguin | What did you expect it to be? |
| 01:16.16 | *** part/#asterisk Sedated (~gokeefe@unaffiliated/sedated) |
| 01:16.35 | kjs | 1001Mhz |
| 01:32.15 | *** join/#asterisk deo_ (~deo@58.71.19.178) |
| 01:35.11 | slav3_kitten | kjs, loss is lower @ 615 |
| 01:36.10 | *** part/#asterisk deo_ (~deo@58.71.19.178) |
| 01:38.41 | dijib | SeRi: fix my shit shaper |
| 01:38.43 | dijib | :D |
| 01:38.53 | dijib | i mean call monitor emails |
| 01:39.38 | SeRi | lol |
| 01:39.50 | SeRi | I have to fix my shit first |
| 01:39.57 | dijib | seriously i dont know whats up.. |
| 01:40.14 | SeRi | I cant right now but when I get a chance I'll help you |
| 01:40.14 | *** join/#asterisk barfoo (~kvirc@bzq-109-67-191-133.red.bezeqint.net) |
| 01:40.29 | dijib | k call 2663 |
| 01:40.51 | SeRi | ok |
| 01:42.00 | dijib | what are you woring on now? |
| 01:42.44 | barfoo | I'm trying to use regex in dialplan to get a match and store in a var. The problem: it returns an empty var. here is the syntax: |
| 01:43.06 | barfoo | same => n,Set(regx="(ig[a-zA-Z0-9]Ex)") same => n,Set(cid="DigiExt)") same => n,Set(cid2=$[${cid} : ${regx}]) same => n,Verbose(substring is: ${cid2}) |
| 01:43.27 | barfoo | Can you tell the problem? |
| 01:44.27 | dijib | man that sounded like crap |
| 01:44.52 | dijib | p3nguin: what s SeRi's problem? he is choppy like hell. Syn flood/ |
| 01:44.54 | dijib | ? |
| 01:47.49 | p3nguin | How would I know? I'm not his admin. |
| 01:49.43 | dijib | :D |
| 01:49.51 | dijib | then what is chan_motif?? |
| 01:51.30 | SeRi | damn straight he is not my admin |
| 01:51.34 | SeRi | I am my own admin! |
| 01:51.39 | SeRi | recognize! |
| 01:51.40 | SeRi | LMAO |
| 01:51.43 | SeRi | lol |
| 01:52.04 | dijib | recognize that call was crap mr HD |
| 01:52.07 | dijib | g722 |
| 01:52.24 | dijib | how do i get chan_motif working with google_voice? |
| 01:52.33 | dijib | my i key is effed |
| 01:54.18 | file | there's a guide on wiki.asterisk.org |
| 01:57.32 | slav3_kitten | ok this is stupid question i seem to have missed in the book. how do i get an identifier for the set making the call to an extension |
| 01:59.08 | WIMPy | Like the caller ID? |
| 01:59.10 | dijib | huh? |
| 02:00.35 | slav3_kitten | yea much like caller ID |
| 02:00.50 | slav3_kitten | dijib, if i have a phone in the work room call an extension |
| 02:00.54 | WIMPy | ... but ...? |
| 02:01.20 | slav3_kitten | for database stuff |
| 02:01.34 | slav3_kitten | i think i missed something i should probably re read the section |
| 02:02.47 | SeRi | lolthings are not going good so far :( |
| 02:03.32 | slav3_kitten | SeRi, shit on fire? |
| 02:04.14 | SeRi | slav3_kitten: yes. I migrated my phone system and thats all fucked up.... I cant imagine what is going to happen when I move the firewall portion |
| 02:04.38 | p3nguin | seri: I started using g722 when I'm on my cell phone... HUGE difference. |
| 02:04.57 | p3nguin | Now I want some desk phones that support it. |
| 02:04.59 | SeRi | p3nguin: bad ass! |
| 02:05.05 | SeRi | It is. |
| 02:05.16 | SeRi | I did a test call with WIMPy and it was awesome |
| 02:05.18 | p3nguin | Lots of sound that was cut off without it. |
| 02:05.40 | WIMPy | Everyone should use it. |
| 02:05.48 | p3nguin | Just listening to Allison I can tell a world of difference over the 8k codecs. |
| 02:11.59 | dijib | she doesnt sound that good |
| 02:12.03 | SeRi | man my fucking ast is fucked up |
| 02:12.14 | SeRi | Damn piece of shit. It seems I cant do anything with it |
| 02:12.23 | SeRi | going to the fucking trash |
| 02:12.25 | dijib | why did you migrate your system and did you uplug it or rebuild your entire topology? |
| 02:12.38 | SeRi | dont ask stupid questions |
| 02:12.41 | dijib | what was the cpu again? |
| 02:12.54 | SeRi | sorry man I am pist |
| 02:13.04 | dijib | no doubt... return to sender. |
| 02:13.14 | dijib | or tell the credit card company it never arrived |
| 02:13.33 | dijib | or vendor |
| 02:13.50 | SeRi | lol |
| 02:14.06 | dijib | seriously... |
| 02:14.14 | dijib | tracing on the shipping? |
| 02:18.21 | barfoo | Can you take a look at my code please? |
| 02:18.25 | barfoo | same => n,Set(regx="(Dig[a-z]Ex)") |
| 02:18.34 | barfoo | same => n,Set(cid="DigiExt)") |
| 02:18.42 | barfoo | same => n,Set(cid2=$[${cid} : ${regx}]) |
| 02:19.01 | SeRi | ~pastebin |
| 02:19.01 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 02:19.05 | barfoo | the problem: the var cid2 is empty |
| 02:19.55 | barfoo | I know pastebin but also that people don't like to click links |
| 02:22.53 | barfoo | Ok here is a link: http://pastebin.com/0WezPmj4 |
| 02:23.00 | *** join/#asterisk Gokee2 (~gokee2@24-113-140-9.wavecable.com) |
| 02:23.10 | *** join/#asterisk Kako_ (~Kako@ip-95-223-21-26.unitymediagroup.de) |
| 02:23.14 | dijib | noojee fax free license? |
| 02:24.39 | Gokee2 | Hello everyone, I am wanting to dial a digit on an active call. The dial command only seems to dial on a new connection. Is there another command to use to simply dial a digit on the existing connection? |
| 02:25.49 | dijib | cant you do soething like that with features? |
| 02:30.06 | p3nguin | Yep. |
| 02:30.39 | dijib | and for email to fax? possible? |
| 02:33.59 | *** join/#asterisk moy (~moy@unvlon55-1176057127.sdsl.bell.ca) |
| 02:36.00 | Gokee2 | Hmm, features to send a digit tone? How would that work? Or is the features in reference to something else? |
| 02:37.43 | p3nguin | Did you not indicate that you wanted to be able to dial a digit during a call and make something happen? |
| 02:38.27 | dijib | http://latino.foxnews.com/latino/news/2012/11/13/john-mcafee-antivirus-software-founder-wanted-in-murder-case-in-central-america/ |
| 02:38.31 | dijib | The creator of McAfee Antivirus is wanted for murder. The trial is expected to last 30 days but the judge won't give a verdict until you pay $29.99. |
| 02:38.34 | p3nguin | Perhaps your query has been misinterpreted. |
| 02:38.45 | Gokee2 | I want asterisk to dial the number. So, I initiate a call to asterisk from my entry gate, asterisk picks up I put in my access code, asterisk dials 9 and lets me in. |
| 02:39.50 | p3nguin | Okay, so you want asterisk to SendDTMF(9) in response to your entry of your PIN? |
| 02:40.37 | p3nguin | Is this merely a call between your phone and asterisk, or is there another end point in the call that I don't know about? |
| 02:40.43 | Gokee2 | p3nguin, Exactly! Thanks :) I knew I must be missing something simple. |
| 02:41.38 | Gokee2 | No other endpoints just did not know there was a SendDTMF command... I was searching for things like send digits |
| 02:43.37 | *** join/#asterisk SeRi (~wtf@pdpc/supporter/professional/seri) |
| 02:43.43 | SeRi | p3nguin: I have a question |
| 02:43.56 | p3nguin | And *I* have an answer. |
| 02:44.02 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 02:44.21 | SeRi | chan_sip.c:22871 handle_request_invite: Call from '' (209.62.1.2:5060) to extension 'my number' |
| 02:44.31 | SeRi | Why is it comming '' from voip.ms? |
| 02:45.04 | SeRi | I am registered but calls are not comming in |
| 02:45.13 | SeRi | they are been send to my misc context |
| 02:46.02 | dijib | i had this issue and it was a setting in the voip.ms console |
| 02:46.12 | SeRi | dijib: ? |
| 02:46.13 | dijib | remember that p3nguin when i first signed up |
| 02:46.14 | dijib | ya |
| 02:46.18 | dijib | let me check |
| 02:46.23 | p3nguin | I remember. |
| 02:46.34 | p3nguin | It was the type of device setting. |
| 02:46.43 | dijib | Current Balance |
| 02:46.44 | dijib | $51.71 |
| 02:47.32 | SeRi | hu? |
| 02:47.37 | SeRi | It never happen to me before |
| 02:47.41 | SeRi | it has worked |
| 02:47.47 | SeRi | and I have it as asterisk |
| 02:47.54 | p3nguin | I'm pondering it. I'll let you know my answer in a moment. |
| 02:48.24 | p3nguin | Have you changed something between the last time it was working and now when it isn't working? |
| 02:48.44 | dijib | SeRi: Account Settings for SIP account, Inbound Settings -> device type= IP PBX Server, Asterisk or softphone. apply, apply all |
| 02:49.44 | SeRi | I have it that way |
| 02:49.52 | SeRi | It has for 2 years or so |
| 02:50.02 | dijib | ok then, then its in your sip.conf |
| 02:50.05 | p3nguin | (2048.24) <p3nguin> Have you changed something between the last time it was working and now when it isn't working? |
| 02:50.16 | SeRi | p3nguin: no. |
| 02:50.18 | dijib | he changed computers |
| 02:50.21 | SeRi | Just new hardware |
| 02:50.25 | p3nguin | So it just magically stopped working? |
| 02:50.49 | SeRi | well saying is not working os not the correct term I guess. |
| 02:50.57 | dijib | i think it must be those elves you guys were talking about earlier |
| 02:51.00 | SeRi | it works but when it comes in from voip.ms is coming in as '' |
| 02:51.15 | SeRi | I never seen that |
| 02:51.19 | dijib | what happends when you change that setting to ATA? |
| 02:51.26 | dijib | have never seen |
| 02:53.09 | p3nguin | Changing the setting to ATA causes all calls to be sent to the 's' extension. |
| 02:53.33 | dijib | right so whats his issue then |
| 02:54.19 | SeRi | a reboot took care of it |
| 02:54.26 | SeRi | now calls come in ok. |
| 02:54.30 | SeRi | I guess it was my system |
| 02:54.48 | p3nguin | You probably could have run sip reload and fixed it. |
| 02:54.53 | SeRi | p3nguin: do you know where I can get the digium fax scrip to build the module? |
| 02:54.54 | dijib | od\ |
| 02:55.03 | SeRi | p3nguin: I did and it did not |
| 03:03.05 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 03:05.10 | dijib | core restart now |
| 03:08.31 | carrar | ERROR |
| 03:08.37 | dijib | lol |
| 03:08.47 | carrar | Please type in root password for access: |
| 03:09.27 | slav3_kitten | how long does it take for a sip connection to time out in the event of a loss of internet? |
| 03:09.48 | dijib | 300? |
| 03:09.50 | dijib | i dont know |
| 03:10.50 | dijib | 60seconds? |
| 03:11.24 | slav3_kitten | never encountered it? |
| 03:11.41 | dijib | never cared to notice |
| 03:11.53 | slav3_kitten | hmm |
| 03:12.11 | dijib | your saying i lose my asterisk servers internet connection, howlong till the itsp's failover knows about the loss ? |
| 03:12.32 | slav3_kitten | yea |
| 03:12.40 | dijib | yeh never counted |
| 03:13.45 | slav3_kitten | like if i'm on the phone with my girl in the UK, and suddenly isp switch gear drops out due to 1/2 second power loss which my ups covers for my gear. how long max should i plan on the call running before it drops an i quit getting billed |
| 03:14.55 | unicron | is it just me, or are the examples in extensions.conf in asterisk 1.8 completely wrong? |
| 03:15.15 | dijib | they changed the sytaxes some over the years |
| 03:15.42 | unicron | ya like with every major revision. the least they could do is search and replace on their docs :( |
| 03:16.04 | dijib | they do replace them |
| 03:16.31 | unicron | i can't figure out why my dialplan is broken |
| 03:16.32 | dijib | they keep the old docs for legacy versions and write or update new ones for bleeding edge |
| 03:16.38 | dijib | lets see it |
| 03:16.59 | dijib | pastebay. |
| 03:17.01 | dijib | net |
| 03:17.26 | p3nguin | unicron: It's just you. |
| 03:17.42 | dijib | slav3_kitten: sip_reg_timeout |
| 03:17.47 | unicron | :( |
| 03:17.52 | dijib | ;maxexpirey=3600 ; Max length of incoming registration we allow |
| 03:17.52 | dijib | ;defaultexpirey=120 ; Default length of incoming/outgoing registration |
| 03:18.08 | dijib | 2min |
| 03:18.09 | dijib | ? |
| 03:19.16 | slav3_kitten | ohhhhhh i thought it had to register once a call for some reason. makes much more sense now |
| 03:19.42 | dijib | slav3_kitten: check your PM |
| 03:19.54 | slav3_kitten | i saw |
| 03:20.01 | slav3_kitten | i actually check mine dijib |
| 03:26.28 | SeRi | p3nguin: you up for a g722 test? I am stress testing the new system. 100% at one core out of 4. |
| 03:26.47 | unicron | dijib: http://pastebay.net/1155873 |
| 03:27.34 | p3nguin | Maybe in a little while. |
| 03:27.51 | SeRi | p3nguin: cool. |
| 03:28.48 | unicron | dijib: everything works except the stupid Gosubs for stdexten |
| 03:29.00 | unicron | can't figure out what it wants, wiki is no use |
| 03:29.23 | unicron | when i dial a user extension, i get an infinite loop |
| 03:30.02 | ChannelZ | SeRi: I think my softphone does g722 if you want me to call something |
| 03:30.26 | SeRi | ChannelZ: awesome. Will do in one sec. Testing fax now |
| 03:30.56 | p3nguin | unicron: You know what Gosub() does and is for? |
| 03:31.28 | unicron | p3nguin, i thought it is like Goto but it returns to the place it is called from? |
| 03:31.45 | p3nguin | That's right. |
| 03:32.34 | dijib | unicron: line 71? |
| 03:32.59 | unicron | dijib: that is an example of the many Gosubs for stdexten that do not work |
| 03:33.24 | unicron | or maybe one of them is breaking it somehow and the rest would work |
| 03:34.37 | dijib | just letting you guys know SeRi and I are in 2663 agan |
| 03:36.14 | *** join/#asterisk deo_ (~deo@222.127.13.226) |
| 03:36.24 | *** part/#asterisk deo_ (~deo@222.127.13.226) |
| 03:43.09 | *** join/#asterisk hebber (~hebber@node-14pg.pool-125-25.dynamic.totbb.net) |
| 03:49.31 | dijib | SeRi: what are you doing? |
| 03:59.56 | *** join/#asterisk timahvo1 (~rogue@197.176.169.131) |
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| 04:07.44 | barfoo | my asterisk is configured on port 5060 |
| 04:07.47 | barfoo | how to register a SIP provider which operates on port 5061 ? |
| 04:11.04 | SeRi | append :5061 to your register string |
| 04:11.40 | SeRi | also use port 5061 pn the peer entry if need it |
| 04:11.50 | SeRi | s/pn/on/ |
| 04:12.29 | SeRi | ok I am now avail for g722 testing if any body is up to it |
| 04:12.46 | barfoo | ok, and is that all? |
| 04:12.54 | SeRi | barfoo: yes. |
| 04:14.43 | barfoo | so i do this on [general] only |
| 04:16.41 | barfoo | and not for the provider specific configurations |
| 04:16.51 | SeRi | barfoo: register => user:pass@itsp.com:5010 |
| 04:16.52 | barfoo | if that's correct |
| 04:16.58 | barfoo | thanks very much |
| 04:42.10 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 04:42.11 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 06:13.27 | *** join/#asterisk artyx (U2FsdGVkX1@junction.googleplex.net) |
| 06:13.50 | artyx | So i just dropped asterisk 1.8.18 onto a system, compield it. and then noticed amportal is nowhere to be found |
| 06:13.55 | artyx | Is that hiding in a subpackage somewherE? |
| 06:16.56 | [TK]D-Fender | Amportal is not Asterisk |
| 06:18.00 | artyx | Oh. that'd be why. i havent dropped fpbx in yet. ignore me then ;) |
| 06:29.31 | *** join/#asterisk mokmeister (~mokmeiste@109.78.34.118) |
| 06:38.46 | [TK]D-Fender | artyx: What card BTW? |
| 06:40.33 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 06:54.38 | artyx | B600d |
| 06:54.51 | artyx | pci base 4 fxo 1 fxs sangoma |
| 06:55.57 | [TK]D-Fender | I know it... Good card |
| 06:56.39 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-215-110.red.bezeqint.net) |
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| 07:06.37 | mirela666 | Hi, anyone got idea how to destroy originated call on my hangup (I don;t want it to be lunched if I hangup my call) |
| 07:11.47 | mirela666 | since it is a sync Originate (waiting for other end to answer to continue dialplan) I have no info what channel to kill - right, all i found is hat i can kill all channels with SoftHangup(Tech/Resource,a) but that I can't do |
| 07:12.38 | mirela666 | does Originate maybe create call-file which i can try to delete? |
| 07:14.05 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
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| 07:40.46 | LiuYan | Does the yum repository of asterisk stop maintained? http://packages.asterisk.org |
| 07:41.07 | carrar | People still use yum? |
| 07:41.25 | LiuYan | i'm one of them |
| 07:43.34 | LiuYan | it seems there're no update anymore since September |
| 07:48.39 | carrar | Why not download from src |
| 07:48.43 | carrar | thats the best way |
| 07:49.50 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 07:50.49 | LiuYan | i don't think it's the best way for end users |
| 07:51.32 | kaldemar | end users should not be installing servers anyway. :P |
| 07:51.45 | artyx | What version of redhat/cent ? |
| 07:52.19 | carrar | LiuYan, it is actually |
| 07:52.39 | carrar | using something you have no idea how it was compile is more risky |
| 07:53.10 | artyx | carrar: http://www.markinthedark.nl/news/ubuntu-linux-unix/69-installing-freepbx-28-with-asterisk-18-on-centos-55.html, if its built on 6.x google for a centos 6 .. theres a nice howto for that as well |
| 07:53.13 | carrar | or use svn |
| 07:53.21 | artyx | oops. LiuYan even |
| 07:53.30 | carrar | THanks artyx!! :) |
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| 08:08.22 | LiuYan | That can't convince me it's the best way. If it is, I think we should build linux in the same way, LinuxFromScratch instead of a familiar distro. But thanks anyway. artyx & carrar. |
| 08:08.43 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 08:13.39 | artyx | LiuYan: If you dont want to build it from scratch, I suggest using some sort of distro that has it incorporated into it at install time (freepbx, or pbiaf come to mind) |
| 08:16.09 | artyx | At least until you have a better idea of what you want out of it |
| 08:17.01 | carrar | haha |
| 08:17.02 | carrar | ok |
| 08:17.06 | carrar | yeah |
| 08:17.14 | carrar | LFS is great BTW |
| 08:17.47 | carrar | I'm pretty sure you don't use FLS |
| 08:17.48 | carrar | LFS |
| 08:18.10 | artyx | I got in on LFS when lfsbook was brand new... ;) |
| 08:18.31 | artyx | Back when you had to statically link everything, then recompile after dropping it on, i dunno if you still have to or if they have some vanilla deployment |
| 08:18.36 | artyx | because it was so tedious to do anything |
| 08:19.24 | artyx | when i worked for RH. i did all sortsa crazy stuff.. but before RH i used slack, i liked it. |
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| 08:29.07 | kmdm | Hi guys, asterisk 1.6.x with dynamic queue members being Invalid after restart, I have in modules.conf: preloads for pbx_config/pbx_ael and then a load order of: chan_sip,iax2,local then app_queue... but it's still Invalid after a restart, also, "module reload app_queue.so" doesn't fix the problem either - but adding/removing the entries does. |
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| 08:38.34 | Yxa | for asterisk 10, should hyperthreading be on? |
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| 09:08.14 | ChannelZ | I've never had hyperthreading do anything good, anywhere. |
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| 09:14.35 | artyx | but its fun to see 16 cpu's available on top |
| 09:15.21 | jacekowski | ChannelZ: say that in ##hardware |
| 09:15.25 | jacekowski | ChannelZ: you will get eaten alive |
| 09:15.36 | jacekowski | ChannelZ: people there get 30x the performance with HT enabled |
| 09:15.56 | artyx | Whats a free software sipphone to use with testing asterisk (not *nix unfortunately) windows 7 |
| 09:16.07 | jacekowski | 3cx |
| 09:16.08 | artyx | linphone isnt' registering |
| 09:16.20 | ChannelZ | hurray for them |
| 09:17.13 | ChannelZ | artyx: zoiper, blink |
| 09:17.16 | artyx | thanksIs that a client? or a server |
| 09:18.20 | artyx | Okay. zoiper seems to fit the bill |
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| 09:36.27 | surferboy | anyone know what file the voice "the person at extension is not available" is in? |
| 09:37.10 | surferboy | oh it's vm-theperson.gsm |
| 09:40.22 | kaldemar | surferboy: that only has the "the person at extension" part. there's a file in the sounds package that tells the file contents. |
| 09:40.42 | surferboy | kaldemar, which package? |
| 09:40.55 | kaldemar | the sounds one. how did you install? |
| 09:40.56 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 09:41.02 | surferboy | what I'm trying to do is replace that voice with my own |
| 09:41.16 | surferboy | how do you recommend that I do that? |
| 09:41.33 | kaldemar | start by answering the question i asked. |
| 09:41.42 | surferboy | I never installed anything |
| 09:41.53 | kaldemar | you must have installed asterisk somehow... |
| 09:41.54 | surferboy | someone else did |
| 09:42.02 | surferboy | I'm just trying to figure this out |
| 09:42.09 | surferboy | I have a basic idea |
| 09:42.16 | ChannelZ | it's strung together from different sounds |
| 09:42.22 | kaldemar | what language are your sound files in and what version of asterisk are you using? |
| 09:42.25 | surferboy | yeah |
| 09:42.37 | surferboy | so can I someone replace all of them? |
| 09:42.43 | surferboy | en |
| 09:42.58 | ChannelZ | do 'locate vm-theperson' in a shell and then you'll know where they all are. |
| 09:43.07 | kaldemar | look into /var/lib/asterisk/sounds/en/core-sounds-en.txt |
| 09:43.12 | surferboy | 1.8.7.0 |
| 09:44.30 | surferboy | kaldemar, how does that file help? |
| 09:44.34 | kaldemar | the whole message is a combination of vm-theperson, some "say" for the extension and vm-isunavail. |
| 09:44.41 | kaldemar | surferboy: did you take a look at it? |
| 09:44.48 | surferboy | yip |
| 09:44.54 | surferboy | just tells me what the files are |
| 09:45.06 | ChannelZ | imagine that |
| 09:45.08 | surferboy | kaldemar, how can I make it only play one of those files? |
| 09:45.10 | kaldemar | surferboy: it tells you which file says which. i.e. it tells you what yo replace and with what. |
| 09:45.57 | kaldemar | surferboy: modify the source code. and loose the ability to tell which extension is in question. |
| 09:46.12 | surferboy | lol |
| 09:46.16 | surferboy | not an option |
| 09:46.21 | surferboy | is there a simple way |
| 09:46.34 | kaldemar | replacing the files with your own is the simplest possible way. |
| 09:46.53 | surferboy | yeah |
| 09:46.55 | surferboy | I have one file |
| 09:47.08 | kaldemar | well divide it into two files then. |
| 09:47.13 | surferboy | if I replace vm-theperson with my gsm file |
| 09:47.19 | surferboy | will the others still play? |
| 09:47.45 | kaldemar | yes they will play. asterisk does not recognize the file contents, it just plays them. |
| 09:48.02 | ChannelZ | huh. Mine calls the cops if I try to change them. |
| 09:48.04 | kaldemar | do you find it too hard to split your file into two? |
| 09:49.46 | surferboy | kaldemar, umm... how can I stop asterisk from speaking the ext after vm-theperson has played? |
| 09:49.49 | surferboy | it's kinda working |
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| 09:52.26 | surferboy | guys I would love some assistance |
| 09:52.30 | ChannelZ | <kaldemar> surferboy: modify the source code. |
| 09:52.34 | surferboy | you are both way smarter than me |
| 09:52.49 | *** join/#asterisk carrar (tim@osburn.jp) |
| 09:52.50 | ChannelZ | Unfortunately pretty much all of Voicemail's behavior is baked into the code. |
| 09:54.30 | kaldemar | surferboy: if you want flexibility, then MiniVM would probably suit your needs better. it requires more effort to implement though. |
| 10:00.39 | kaldemar | that particular change to remove the extension playback would be as simple as removing 3 lines of code in app_voicemail.c but you'd shoot yourself in the leg by taking the path of such minor source modifications. just do it the way it is supposed to, and replace two files. |
| 10:01.01 | RZero | Does any body know if its possible to send a fax during call over a DAHDI channel ? |
| 10:01.59 | surferboy | kaldemar, it's more than two files though |
| 10:02.42 | artyx | I am stumped... I am using freepbx and asterisk 1.8, i have configured my dahdi channels, and dahdi show channels shows me 4 fxo ports, and 1 fxs port. I can call the fxs port and the phone rings, but when i pick up the phone i just get a fast busy, checking the debug gives me a hangup immediately when it goes offhook |
| 10:02.52 | artyx | Ive reformatted, reinstalled. and have the same error |
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| 10:03.22 | artyx | should the context be from-zaptel from-dahdi or from-internal for this line, ..... |
| 10:03.25 | surferboy | kaldemar, and would I need to re record the message to take into account that the extension number gets played back |
| 10:03.48 | artyx | in chan_dahdi_additional i have a context listed of from-internal, but i just get the fast busy |
| 10:04.19 | kaldemar | surferboy: i assure you, reconding two files again will cause you less work overall. |
| 10:04.41 | kaldemar | artyx: try asking in #freepbx |
| 10:04.46 | surferboy | so is that the only way????? |
| 10:05.46 | artyx | kaldemar .. freepbx is the frontend, it generates asterisk config files,though the dialplan might bea little large, what generated the from-internal context was dahdi (asterisk module i think) so i also asked here |
| 10:06.08 | artyx | im googling with no success, checking mailing lists now |
| 10:06.18 | kaldemar | artyx: i know what freepbx is. but note the following: |
| 10:06.22 | kaldemar | ~freepbx |
| 10:06.23 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 10:09.14 | surferboy | kaldemar, where can I get hold of vm-theperson.gsm |
| 10:09.18 | surferboy | I deleted it by accident |
| 10:09.41 | kaldemar | http://downloads.asterisk.org/pub/telephony/sounds/ |
| 10:11.46 | Yxa | my first step to installing asterisk is to compile libpri. it the errors: pridump.c:45:24: error: dahdi/user.h: No such file or directory |
| 10:13.06 | Yxa | must i compile dahdi first instead? |
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| 10:18.58 | RZero | When trying to send a fax during a call over DAHDI channel , the sendfax starts but then hangs up, I pri span debug on but I dont see any thing in the debug to say why it hangs up, here is whats on the console http://pastebin.com/kfEHk6mD, the idea is the person calls from a fax machine listens to tt-weasels wait 7 seconds then presses the ready button on their fax machine. |
| 10:20.07 | RZero | I can send the fax using a call file no problem |
| 10:20.19 | RZero | but this needs to be done during a call |
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| 10:28.58 | surferboy | how can I get a voice to sound like that asterisk robot? |
| 10:29.07 | surferboy | att text to speech |
| 10:29.11 | surferboy | is not the same |
| 10:30.41 | carrar | You mean Allison? |
| 10:31.29 | carrar | http://www.digium.com/en/products/ivr/allisonsmith/ |
| 10:33.37 | carrar | The AT&T one used to block numbers |
| 10:33.40 | carrar | or mixed them |
| 10:33.54 | carrar | so you couldn't use it for stuff without paying them for it |
| 10:34.12 | carrar | Though I think you could spell out the numbers and it worked |
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| 10:59.05 | Kako | hi, when I set a Channel-var with AMI: "Action: Setvar, Variable: CDR(AccountCode)" How can I get the value in AEL? |
| 10:59.07 | Kako | I was trying by ${CHANNEL(AccountCode)}, but I receive "func_channel_read: Unknown or unavailable item requested: 'AccountCode'" |
| 11:00.19 | *** join/#asterisk ghost75 (~trechber@dslb-188-105-020-031.pools.arcor-ip.net) |
| 11:00.26 | *** join/#asterisk surferboy (~surferboy@109.83.business-adsl.cybersmart.co.za) |
| 11:00.43 | ghost75 | if i goto different context will $EXTEN change ? |
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| 11:02.08 | surferboy | carrar, how can I make her speak something? |
| 11:02.43 | carrar | You first start by reading that page |
| 11:07.21 | kaldemar | ghost75: EXTEN is the extension, not the context. whether it changes depends on the current extension. |
| 11:07.27 | *** join/#asterisk surferboy (~surferboy@109.83.business-adsl.cybersmart.co.za) |
| 11:07.29 | surferboy | what is a voice prompt file? |
| 11:07.35 | surferboy | carrar, |
| 11:07.47 | carrar | forme, it's a wav file |
| 11:08.06 | ghost75 | so it will not change if there is like exten => s in other context |
| 11:08.16 | carrar | or use SOX to convert it to something else |
| 11:08.39 | ghost75 | because in h it will change |
| 11:09.26 | kaldemar | ghost75: under the hood it changes every time the current extension changes, but the extension might be the same as the previous one. |
| 11:09.43 | surferboy | umm.. |
| 11:09.50 | surferboy | so it's just a audio wav file? |
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| 11:15.11 | ghost75 | surferboy: what you want to do? |
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| 12:01.20 | Kako | hi, I've set a Channel-var with AMI: "Action: Setvar, Variable: CDR(AccountCode), Value: test, Channel: ...". When i run CoreShowChannels I see the set values, so I know the values are set. How can I use this Value in a AEL? I've tried ${CHANNEL(AccountCode)}, ${CDR(AccountCode)} and ${AccountCode}. They are all empty... :-( |
| 12:03.15 | *** join/#asterisk deo (~deo@112.198.90.243) |
| 12:03.39 | kaldemar | Kako: show what you do. |
| 12:08.17 | kaldemar | Kako: ${CDR(AccountCode)} is how you access the variable, you must be doing some invalid assumptions. |
| 12:08.59 | Greenlight | Possibly your setting it one one channel, and trying to get it of a channels thats dialled from there |
| 12:09.11 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.101) |
| 12:10.03 | Kako | I've put the AccountCode in both channels and wehn running CoreShowChannels I see it in both Channels. What I want to do is somthing like this: |
| 12:10.05 | Kako | Verbose(1,### AccountCode was set: ${AccountCode} ###); |
| 12:10.41 | Kako | or Set(CDR(AccountCode)=${AccountCode}); |
| 12:11.18 | kaldemar | and what exactly are you doing now? |
| 12:11.42 | Kako | My problem is, thet the accountcode set by AMI is not been written to CDR, so I'm trying to force it... |
| 12:11.55 | Kako | Now I' doing: |
| 12:11.56 | Kako | Verbose(1,### AccountCode was set: ${AccountCode} ###); |
| 12:12.03 | Kako | The Result is empty |
| 12:12.29 | kaldemar | why would it not be empty if the variable you set is CDR(AccountCode)? |
| 12:12.51 | Kako | I've tried ${CHANNEL(AccountCode)}, ${CDR(AccountCode)} and ${AccountCode}. They are all empty... |
| 12:13.00 | kaldemar | tried how? |
| 12:13.12 | kaldemar | show what you do. precisely. |
| 12:13.29 | Kako | Verbose(1,### Channel AccountCode was set: ${CHANNEL(AccountCode)} ###); |
| 12:13.31 | Kako | Verbose(1,### CDR AccountCode was set: ${CDR(AccountCode)} ###); |
| 12:13.32 | Kako | Verbose(1,### AccountCode was set: ${AccountCode} ###); |
| 12:14.14 | kaldemar | ${CDR(AccountCode)} will have the value when set on the right channel and the appropriate channel is the one executing that Verbose. |
| 12:14.40 | kaldemar | show output of the manager communication and a CLI output. |
| 12:15.00 | Kako | getting it, will put it as pastebin... |
| 12:20.21 | Kako | Here the pastebin: http://pastebin.com/RUKcDeRm |
| 12:21.13 | Kako | 1. Step: Connect to SIPs |
| 12:21.14 | Kako | 2. Set Vars |
| 12:21.16 | Kako | 3. Check if set by CoreShowChannels |
| 12:21.17 | Kako | 4. Hangup, so that h rule runs |
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| 12:23.19 | gavimobile | folks, my google voice stoped working, I didn't make any changes. is it possible that its not from my side? |
| 12:23.32 | gavimobile | im using asterisk 1.8 and I know google voice is best used with asterisk 11 |
| 12:25.11 | kaldemar | Kako: copy and paste what you have in setCDRdetails to a new pastebin. |
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| 12:26.03 | kaldemar | "dialplan show setCDRdetails" would be even better. |
| 12:27.18 | Kako | kaldemar: her the macro: http://pastebin.com/n7g5bUA5 |
| 12:28.17 | salz212 | what is the release date of asterisk 11.1 ? any link to find out .. tried couldn't find one.. |
| 12:28.34 | kaldemar | Kako: ezi_setCDRdetails is not setCDRdetails |
| 12:29.28 | Kako | kaldemar: yes and no, type error after cut and paste :-) |
| 12:31.10 | Kako | kaldemar: there is only this on macro, no other. the trueth is, I forgot to cut ezi_ before pasting. |
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| 12:31.40 | salz212 | what is the release date of asterisk 11.1 ? any link to find out .. tried couldn't find one.. |
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| 12:32.23 | Greenlight | Its in the topic |
| 12:32.33 | Greenlight | Just look up ^^ |
| 12:32.47 | kaldemar | Kako: i asked you to copy and paste to avoid any manual changes. "dialplan show setCDRdetails" |
| 12:32.48 | Kako | kaldemar: this is the full macro as it is read in asterisk rigth now. I was trying to make i readable |
| 12:32.56 | Kako | http://pastebin.com/RPR1hjse |
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| 12:35.11 | kaldemar | Kako: i can't help you if you don't show anything real. |
| 12:36.50 | Kako | here the dump of "dialplan show": http://pastebin.com/2yBiyjJM |
| 12:37.36 | kaldemar | your CLI output was setCDRdetails, not ezi_setCDRdetails. |
| 12:38.36 | Kako | ok, new run, now with the german words without translations and changes... creating now a new one... |
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| 12:49.44 | Kako | kaldemar: hoppefully now long and full enought: http://pastebin.com/S0sMPCP4 |
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| 12:53.58 | kaldemar | Kako: nothing wrong there. something before Gosub("SIP/4137-00000035", "ezi_setCDRdetails,s,1") must be undoing your change to the account code. |
| 12:55.13 | kaldemar | Kako: i tested that very thing myself with a gosub in the h exten. works as expected. either something unsets the account code or you have a encountered a bug. |
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| 13:08.41 | FlashDeluxe | Hi! I got an asterisk server running and i got this extension: http://paste.debian.net/209512/ If the user is busy and somebody calls in, asterisk leads him to the Voicemail. But i want asterisk to playback a busy tone. How do i do it? |
| 13:10.08 | kaldemar | FlashDeluxe: check DIALSTATUS variable contents after Dial and do your decisions based on that. |
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| 13:13.21 | FlashDeluxe | kaldemar: ok thanks! |
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| 13:23.28 | madduck | So we have 10 numbers and they shall be assigned to 3 phones (3 each) and one menu service. What means do I have to distinguish on the phones which number has been rung? Does Asterisk or SIP provide for that? |
| 13:24.23 | p3nguin | madduck: You can manipulate the caller id. |
| 13:24.26 | madduck | i.e. can a SIP phone ring differently? |
| 13:24.37 | madduck | p3nguin: but the caller ID should of course remain unchanged |
| 13:24.40 | madduck | or wait… hah! |
| 13:24.48 | p3nguin | Some phones have distinctive ringing. |
| 13:25.04 | madduck | I want to make it such that there is a default dial out and to use any of the other two numbers, you dial *1 and *2 before the number |
| 13:25.09 | madduck | so I simply add that to the callerid ;) |
| 13:25.20 | madduck | that'll do and play nicely with calling back |
| 13:26.11 | p3nguin | Set(CALLERID(num)=*1${CALLERID(num)}) |
| 13:26.15 | madduck | yeah! |
| 13:26.31 | p3nguin | or Set(CALLERID(num)=*1 ${CALLERID(num)}) |
| 13:27.14 | p3nguin | I usually change the callerid name, but you can change the number if it will help solve your problem. |
| 13:27.36 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 13:28.01 | [TK]D-Fender | I'd change the name instead of the number unless I needed to mod the number for how my dialplan would best dial it back the way I'd like. |
| 13:28.39 | p3nguin | That's what he said he does anyway. |
| 13:28.46 | madduck | [TK]D-Fender: exactly that's why I would change the number… |
| 13:31.55 | [TK]D-Fender | madduck, I might guess so that you mod your OUTGOING CallerID so they see it as the division they called to in the first place (if you miss the call) |
| 13:33.05 | WIMPy | I'd use 3 accounts and let the phone take care of it. |
| 13:33.28 | [TK]D-Fender | WIMPy, Certainly an idea. |
| 13:33.57 | madduck | unfortunately, our phones can only do 6 accounts altogether |
| 13:34.08 | madduck | we are using an El Cheapo Gigaset home thingie for now |
| 13:34.16 | madduck | while it works… |
| 13:35.20 | WIMPy | Didn't you say 3 per phone? |
| 13:35.29 | madduck | incoming trunk numbers |
| 13:35.38 | madduck | we bought 10 numbers from the provider |
| 13:35.53 | madduck | 3 go to each of the phones each and each of the phone should be able to dial out with any of the three |
| 13:35.59 | madduck | this is because we work in three contexts |
| 13:36.32 | WIMPy | Seems to fit. |
| 13:37.28 | *** join/#asterisk ruied (~AndChat66@246.105.103.87.rev.vodafone.pt) |
| 13:40.47 | SeRi | good morning all. |
| 13:41.19 | WIMPy | Hi SeRi |
| 13:41.31 | SeRi | whats going on WIMPy |
| 13:41.54 | SeRi | is freaking cold here |
| 13:42.06 | Kako | kaldemar: Sorry, was on the phone... No, bevor the mentionated line was nothing. Here the CLI output befor it (http://pastebin.com/TYDdgdP1) Maybe I should increase my debug/verbose level... BTW I'm using Asterisk 1.6.2.9-2+squeeze5. Thanks for your help. |
| 13:42.19 | WIMPy | Breakfast :-) |
| 13:42.55 | SeRi | That sounds like an excellent idea. |
| 13:43.12 | Nivex | is 1.6 even still supported? |
| 13:43.31 | [TK]D-Fender | no |
| 13:43.40 | WIMPy | Nivex: No, abd it wasn't the best anyway. |
| 13:43.49 | [TK]D-Fender | Kako, and I see 2 warnings in there. which one are you asking for help on? |
| 13:44.02 | WIMPy | Kako: Befor you try anything else, you should upgrade. |
| 13:44.25 | Nivex | I'm glad they got a relatively recent 1.8 available in squeeze-backports now that digium doesn't maintain packages |
| 13:50.26 | *** join/#asterisk frey (~frey@130.208.247.19) |
| 13:51.10 | kaldemar | [TK]D-Fender: neither. he was setting CDR(AccountCode) on a channel from AMI but it didn't show up in dialplan with ${CDR(AccountCode)}. |
| 13:51.48 | [TK]D-Fender | kaldemar, And we don't see any kind of AMI debug in there... lovely... |
| 13:51.59 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 13:52.16 | kaldemar | [TK]D-Fender: nope. it was pasted before, i just asked for earlier CLI output and that was it. |
| 13:52.35 | [TK]D-Fender | relink.... |
| 13:52.48 | *** join/#asterisk kriger (~norge@68-190-90-80.dhcp.mdsn.wi.charter.com) |
| 13:53.13 | kaldemar | http://pastebin.com/S0sMPCP4 |
| 13:53.22 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 13:54.29 | [TK]D-Fender | And raw dialplan? |
| 13:55.03 | kaldemar | does not exist. |
| 13:55.53 | [TK]D-Fender | Kako, Please pastebin your dialplan for what we see in those calls. |
| 13:57.59 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:57.59 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 13:59.46 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 13:59.47 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:00.15 | *** join/#asterisk serafie (~erin@nat/digium/x-vvjqnupvgqvtifch) |
| 14:01.05 | Kako | [TK]D-Fender: this will take a while... all or only those show in "-- Executing ["? |
| 14:01.44 | [TK]D-Fender | Kako, It should have less time than it did just for that response... |
| 14:01.58 | [TK]D-Fender | Kako, and yes, I asked for all the bits we SEE |
| 14:06.51 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:07.41 | Kako | [TK]D-Fender: ok, here a copy of extensions.ael (http://pastebin.com/p2MrvVKW) |
| 14:07.47 | Maliuta | [TK]D-Fender: I like your attention to detail :) When you're not around I miss it :) |
| 14:08.46 | Maliuta | Kako: what's with the excessive commenting? |
| 14:10.11 | Kako | Maliuta: 99% of th file is the original from Gemeinschaft3.1. And the last time I cut parts for better readability I was asked to leave it as it is, so.... sorry :-) |
| 14:10.25 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 14:10.45 | Kako | Maliuta: thats because I asked "all?" |
| 14:14.34 | Kako | My part of the code you see from line 101 - 134 and the lines with "&ezi_setCDRdetails();", the rest is not from me |
| 14:20.14 | [TK]D-Fender | Kako, I'm wondering if those functions survive the end of call properly at all. Do a hard SET before you place the call just to see if that value sticks |
| 14:20.43 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 14:20.48 | [TK]D-Fender | Kako, and I presume that your accountcode ISN'T being written to the actual CDR, correct? |
| 14:21.21 | [TK]D-Fender | Maliuta, Yeah, all the subtlety of a Rusty-Nail Enhanced ClueBat (tm) :) |
| 14:22.23 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:22.23 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:25.24 | kaldemar | [TK]D-Fender: i even tested that same thing, with an h exten that uses gosub with a noop. shows the AMI-set account code as expected. |
| 14:25.55 | [TK]D-Fender | I remember Goto wrecking some things... |
| 14:26.05 | [TK]D-Fender | kaldemar, So you figure it's version-bug related? |
| 14:26.15 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-yzumopwuitaefykc) |
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| 14:26.23 | kaldemar | the version i tested on was 11. |
| 14:26.49 | kaldemar | i don't see anything wrong in his outputs, i'd suspect a bug. |
| 14:27.30 | frey | Hello. I have an old OmniPBX that I would like to replace with Asterisk, but I'm new to the field. :) |
| 14:28.03 | frey | I have old 4028 phones that are being used all over my university. Do you know if I can use them with Alterisk? Or do I need to replace them? |
| 14:28.40 | slav3_kitten | anyone have a sound file saying "caller id" |
| 14:29.02 | WIMPy | slav3_kitten: Say it. |
| 14:29.36 | slav3_kitten | WIMPy, my voice is crap an i got no mic |
| 14:29.40 | p3nguin | There's one that comes with asterisk, I believe. |
| 14:29.46 | Maliuta | [TK]D-Fender: I prefer to be a subtle as a .50 calibre round to the head :) |
| 14:30.04 | Kako | [TK]D-Fender: Yes it is not written to CDR if I try to set the AccountCode with AMI "Setvar" for a Channel the ist already "open". But when creating a call with AMI "Originate" the AccountCode ist written correctly to CDR. |
| 14:30.09 | WIMPy | slav3_kitten: You don't have a phone with a mic? |
| 14:30.15 | slav3_kitten | Maliuta, .50AE .500S&W or .50BMG |
| 14:30.35 | slav3_kitten | WIMPy, i can use that to record sounds? |
| 14:30.57 | WIMPy | slav3_kitten: Have you ever used something like voicemail? |
| 14:31.09 | slav3_kitten | not set it up yet, didn't dawn on me either |
| 14:31.16 | [TK]D-Fender | frey, How many? |
| 14:31.16 | Maliuta | slav3_kitten: .50 Desert Eagle ... say what you want about the Israelis, they know how to make a serious weapon/round. :) |
| 14:31.25 | slav3_kitten | Maliuta, so .50AE |
| 14:31.33 | WIMPy | slav3_kitten: Otherwise there's the simple Record(). |
| 14:31.34 | Kako | The uncomment Set(CDR... lines were already in use and they would be written into CDR. So a Hard SET will work. |
| 14:31.40 | [TK]D-Fender | frey, I'm looking for docs on it. It seems to say they're IP phone, but I'm looking for "SIP" and "open to configuration". |
| 14:31.52 | Maliuta | kaldemar: I did away with GOTO by transitioning to AEL for my dialplans |
| 14:32.03 | slav3_kitten | nods at WIMPy and starts looking in his book |
| 14:32.30 | Maliuta | slav3_kitten: you've been using * for ages and haven |
| 14:32.32 | frey | [TK]D-Fender: I don't know. Over 100 maybe. :) |
| 14:32.36 | Maliuta | 't setup VM? |
| 14:32.38 | p3nguin | I didn't know people intentionally switched *TO* AEL. |
| 14:32.39 | [TK]D-Fender | Maliuta, DE stopping power isn't THAT great to validate the excess barrel length & handling difficulty let alone the rater sad magazine capacity. |
| 14:33.02 | slav3_kitten | Maliuta, it's been like 2 weeks |
| 14:33.12 | [TK]D-Fender | Maliuta, Far better to carry a hot-loaded dual stack 1911 varient, and far cheaper. |
| 14:33.31 | *** join/#asterisk AkkerKid (~AkkerKid@50-200-18-202-static.hfc.comcastbusiness.net) |
| 14:33.31 | slav3_kitten | [TK]D-Fender, don't forget it's also a gas operated handgun which is prone to FTE if you limp wrist it which is really easy to do with it |
| 14:33.32 | Maliuta | p3nguin: I found it more flexible than the stock standard extensions.conf |
| 14:33.47 | Maliuta | slav3_kitten: ahh, I've gotten you mixed up with someone else :) |
| 14:33.47 | [TK]D-Fender | slav3_kitten, Kicks like a bitch... |
| 14:34.07 | slav3_kitten | yea it has excessive recoil, making it easier to limp wrist |
| 14:34.13 | Maliuta | slav3_kitten: get a USB headset, the sound quality is not too bad. |
| 14:34.14 | slav3_kitten | Maliuta, i've been in the channel for months |
| 14:35.11 | Maliuta | slav3_kitten: and there are a heap of *nix sound tools to produce professional quality sound files that can be transcoded to all the formats you need for * |
| 14:35.12 | slav3_kitten | we are still back to my voice sounding like ass. i kind of plan on having the uk girl i'm setting this all up to call record an mp3 for me so i can chop it up into the segments i want |
| 14:36.12 | Maliuta | slav3_kitten: voice people are easy to find. I use some Australian voice files ... but tt-monkeys sounds more like ducks, so I use the US version of that |
| 14:36.50 | slav3_kitten | she's british, i love her voice |
| 14:36.51 | [TK]D-Fender | slav3_kitten, http://www.the-m-factor.com/html/specs_6.html <- 13rd compared to the DE's 8. |
| 14:37.03 | Maliuta | slav3_kitten: there are a heap of different soundfiles for * out there, I packaged the .au ones for debian and I need to do up a bunch of the extra UK ones that I have kicking about |
| 14:37.23 | *** join/#asterisk mdg (6c5df7d9@gateway/web/freenode/ip.108.93.247.217) |
| 14:37.36 | slav3_kitten | [TK]D-Fender, do you know the diff between an XD and an XDM |
| 14:37.36 | Maliuta | slav3_kitten: what is it with yanks and Commonwealth accents? |
| 14:38.12 | slav3_kitten | Maliuta, i think it's the things she lets me do to her when she visits :P |
| 14:38.17 | [TK]D-Fender | slav3_kitten, Not specifically. I glossed over the series once as they are at least mechanically interesting... |
| 14:38.27 | slav3_kitten | m is match iirc |
| 14:38.42 | Maliuta | not that most yanks can tell the difference between London english accents, Australian accents, Kiwi accents and South African accents :) |
| 14:38.51 | mdg | Hi, I installed asterisk 1.8.18.0 and did then executed `make samples`. This placed extensions.{conf,ael,lua} in /etc/asterisk. I am interested in configuring the dialplan through lua, how can I tell asterisk to use extensions.lua and no others? |
| 14:39.02 | Maliuta | slav3_kitten: pics pls ;P |
| 14:39.03 | [TK]D-Fender | slav3_kitten, I don't own any actual firearms, but used to work for an international distribution company. |
| 14:39.23 | slav3_kitten | [TK]D-Fender, i you're american why not?? |
| 14:39.23 | Maliuta | mdg: empty the other files :) |
| 14:39.26 | mdg | I read through the relevant parts of TFoT3 and didnt see any mention on how/where to specify extensions.lua |
| 14:39.29 | [TK]D-Fender | slav3_kitten, And my father used to collect so I'd seen more than most people's fair share. |
| 14:39.34 | slav3_kitten | Maliuta, are you a male or female? |
| 14:39.36 | WIMPy | mdg: Remove the others. |
| 14:39.46 | [TK]D-Fender | slav3_kitten, I'm not. Besides, gun's aren't my style.... |
| 14:40.01 | slav3_kitten | [TK]D-Fender, airsoft? :P |
| 14:40.08 | Maliuta | slav3_kitten: I have 3 DNA strands from a bone marrow transplant, so that's a loaded question. |
| 14:40.36 | slav3_kitten | has that affected the sex organs you were manufactured with? |
| 14:40.44 | jmetro | I could tell an australian and London, but havent heard a kiwi or south african accent to be able to know. |
| 14:40.59 | Maliuta | slav3_kitten: the DNA I was born with is male. The donor DNA is female (native american too). The mutant strand (a combination of mine and hers) is of undetermined gender :) |
| 14:41.19 | slav3_kitten | Maliuta, then no pics for you :P |
| 14:41.28 | slav3_kitten | jmetro, same here |
| 14:41.36 | Maliuta | I plan on getting my shooters licence early next year |
| 14:41.52 | Maliuta | slav3_kitten: you're putting up a pay-per-view website then? ;) |
| 14:42.00 | mdg | Maliuta: WIMPy: Thanks, I tried renaming the files before I came in here (I assume if it isnt 'extensions.*' asterisk wont consider it) and reloading the dialplan. When I execute 'dialplan show' now, there are 3 contexts, 2 noop()s and 1 park(), none of which are in extensions.lua. Does this mean I have an error in extensions.lua and asterisk fell back to a safe default ? |
| 14:42.06 | slav3_kitten | fist bumps Maliuta on gun ownership |
| 14:42.21 | slav3_kitten | Maliuta, my friend has them on his fetlife account |
| 14:42.25 | Maliuta | slav3_kitten: and yes it has effect my sex organs ... but that's a whole other issue |
| 14:42.40 | slav3_kitten | Maliuta, really? that's surprising to me |
| 14:42.47 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 14:42.48 | Maliuta | slav3_kitten: a/c name? I have had a fetlife account for like a decade |
| 14:43.00 | WIMPy | mdg: Extensions can be generated by other modules. Take a look at features.conf. |
| 14:43.01 | [TK]D-Fender | Maliuta, So when people tell you to go fuck yourself you can tell them "Still waiting for the recombinant DNA to finish mutating, I'll get back to you on that!"? ;) |
| 14:43.10 | slav3_kitten | does it affect like hormone production or something? |
| 14:43.20 | mdg | WIMPy: ok thanks for your help and time |
| 14:43.39 | Maliuta | slav3_kitten: shouldn't be, reproductive issues are common in the cancer community. Chemo, radiation, and the general meds you have to use all have a big effect on reproductive health |
| 14:44.00 | slav3_kitten | well yea chemo and rads are pretty terrible on the lil swimmers |
| 14:44.06 | [TK]D-Fender | frey, Well first confirm that. If so then Sure, you probably can use them. |
| 14:44.14 | slav3_kitten | sorry you got cancer, not really sure what to say though |
| 14:44.17 | Maliuta | slav3_kitten: my hormone levels are "normal", have you heard of "hypergonadism" ??? |
| 14:44.36 | slav3_kitten | nope, but google has |
| 14:44.39 | Maliuta | slav3_kitten: that and a whole heap of other issues ... it's cost me more than one relationship in the last 9 years. |
| 14:44.43 | [TK]D-Fender | spins up "Big Balls" - AC/DC on the playlist... |
| 14:45.18 | *** join/#asterisk ruied (~AndChat66@246.105.103.87.rev.vodafone.pt) |
| 14:45.35 | slav3_kitten | [TK]D-Fender, fitting |
| 14:45.43 | Maliuta | slav3_kitten: how about just dealing with it, I don't need pity, or sorrow, or anything else. Just accept it, that's better than anything someone in my situation can hope for. |
| 14:46.04 | Maliuta | [TK]D-Fender: how about "She's got the jack" :) |
| 14:46.22 | slav3_kitten | there is no pitty or sorrow, but it just sucks you had to get cancer man. just like it just sucks that i had to get disabled at work for the rest of my life |
| 14:46.23 | Maliuta | [TK]D-Fender: at least Acca-Dacca is an Aussie band |
| 14:47.02 | Maliuta | slav3_kitten: it happened 9 years ago, I came to terms with it fairly quickly. I doesn't really need a response. |
| 14:47.27 | slav3_kitten | well i'm glad you're not dead. you seem like a cool guy |
| 14:47.29 | Maliuta | slav3_kitten: what happened? and have you taken the employer to the cleaners yet? |
| 14:47.56 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 14:47.58 | Maliuta | slav3_kitten: I've been getting that alot since I started meeting new people post BMT :) |
| 14:48.05 | slav3_kitten | Maliuta, fell an broke my spine, got some damaged nerves that they say will never heal. i have to get around on crutches and got terrible pain in one limb |
| 14:48.11 | slav3_kitten | aka shit happens. |
| 14:48.37 | bulkorok | hey... is displaysystemname in manager.conf deprecated in 1.8 ?! |
| 14:48.47 | slav3_kitten | got the car modified for left foot driving, got a meager settlement which paid off some debts incurred after the accident |
| 14:49.19 | Maliuta | slav3_kitten: I know what that is like ... I am on a waiting list to have my left hip replacement replaced |
| 14:50.06 | slav3_kitten | well i got some scalpels and access to a mill/lathe as well as some titanium |
| 14:51.27 | jmetro | i've got a door hinge , if you angle it the right way... |
| 14:51.43 | slav3_kitten | i think we got this |
| 14:52.43 | *** join/#asterisk ruied (~AndChat66@246.105.103.87.rev.vodafone.pt) |
| 14:54.29 | mdg | The reason why my extensions.lua wasnt loading was because pbx_lua module was never compiled. pbx_lua was never compiled because I did not have lua-devel installed |
| 14:54.53 | mdg | +2xp |
| 14:58.44 | frey | [TK]D-Fender: You mean how many 4028 IP Touch VOIP phone's I have? |
| 14:59.44 | *** join/#asterisk coppice (~chatzilla@m121-202-52-221.smartone.com) |
| 15:01.14 | *** join/#asterisk Yxa (~Yxa@bb119-74-74-189.singnet.com.sg) |
| 15:01.55 | Maliuta | slav3_kitten: if the injury is that debilitating you should have been up for a bigger payment to cover loss of income for period X |
| 15:02.00 | Yxa | libpri-1.4.13 doesn't compile. errors dahdi/user.h not found |
| 15:02.08 | Yxa | libpri-1.4.12 compiles fine |
| 15:02.36 | WIMPy | Yxa: Install dahdi first. |
| 15:02.40 | Maliuta | slav3_kitten: sounds like you had a bad PI lawyer ... I know the relevant laws here in .au and what you can/can't claim for and get. I also know alot of really good lawyers :) |
| 15:03.01 | [TK]D-Fender | frey, Yes |
| 15:03.11 | slav3_kitten | Maliuta, i got the max my state allowed which since i accepted workers compensation was limited to a meager amount. it also took me years to get the lawyers to work it out |
| 15:03.23 | Yxa | WIMPy i thought we always install libpri first |
| 15:03.38 | frey | [TK]D-Fender: Does the number mater that much? I think there are more than 100, could be 200. |
| 15:03.41 | WIMPy | Yxa: Not any longer. |
| 15:04.18 | frey | [TK]D-Fender: I also have a few analog phones connected to it, but I would you ATA boxes instead. |
| 15:05.23 | [TK]D-Fender | frey, Ok, go verify if they support standard SIP. If so, hopefully standard PoE as well so you can just swap out the entire core. |
| 15:05.39 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:05.39 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:06.10 | slav3_kitten | ok anyone good with international dialing to the uk i fail to understand it fully |
| 15:06.20 | Maliuta | slav3_kitten: which state? I'm aware that alot of stuff varies from state to state over there. I was thinking of getting a class action off the ground against states which don't allow GCSF to be given to healthy people, that means that if a bone marrow donor is found in that state an adult stem cell harvest isn't possible |
| 15:06.36 | slav3_kitten | +44-xxxx-xxx-xxx |
| 15:06.36 | slav3_kitten | uk = england |
| 15:06.38 | WIMPy | slav3_kitten: International is a question of from, not to, usually. |
| 15:07.17 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:07.17 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:07.18 | slav3_kitten | when i dial out via my outbound provider do i need to send the number as "044-xxxx-xxx-xxx" or "44-xxxx-xxx-xxx" |
| 15:07.19 | WIMPy | slav3_kitten: Not sure the rest of the uk like to be called England. |
| 15:07.31 | WIMPy | But what's the question? |
| 15:07.34 | slav3_kitten | WIMPy, i corrected what i eant by uk |
| 15:07.38 | Maliuta | slav3_kitten: and the difference between the after effects of a BMT with actual marrow vs. adult stem cells is huge. I probably wouldn't have the degree of issuse I've had post transplant if my donor could have donated stem cells |
| 15:07.54 | Kako | [TK]D-Fender, kaldemar, WIMPy: updated now to asterisk amd64 1:1.6.2.9-2+squeeze8 (not a real jump :-() ... no changes, will now download and compile 1.6.2.24... Gemeinschaft will not run with 1.8 an newer, so... well, I hope this solves my problem. |
| 15:08.10 | WIMPy | slav3_kitten: Whatever you use for international calls in the country you're calling FROM. |
| 15:08.11 | slav3_kitten | Maliuta, that kinda sucks that stem cells are not in common usage |
| 15:08.15 | Maliuta | WIMPy: talk to the Scots, Welsh and Irish about it ... it's painful |
| 15:08.47 | *** join/#asterisk ruied (~AndChat66@246.105.103.87.rev.vodafone.pt) |
| 15:08.50 | slav3_kitten | WIMPy, ive never called outside the country and my outbound service inserts no country codes by default |
| 15:08.50 | WIMPy | Kako: That's bad. I wouldn't use anything older than 1.8. |
| 15:09.24 | wonderworld | hey guys, i have been working on an asterisk-based project for quiet some time and was wondering if you could have a look and tell me what you think. it's not launched yet. but it will be launched soon. http://vocieforall.com/index2.php |
| 15:09.35 | Maliuta | slav3_kitten: talk to the state legislatures, it's their fault. The rest of the world is fine with it. I think that people get adult stem cells mixed up with embryonic stem cells and just kybosh the whole thing |
| 15:10.11 | Maliuta | Kako: running debian? there are 1.8 packages out there. |
| 15:10.15 | slav3_kitten | Maliuta, the general population is easily confused often times |
| 15:10.19 | WIMPy | wonderworld: unable to determine IP address for host name 'voiceforall.com'. |
| 15:10.49 | wonderworld | damn type. its... http:/voiceforall.com/index2.php |
| 15:11.05 | WIMPy | typo again |
| 15:11.06 | Maliuta | slav3_kitten: when a large group of people get together the IQ of the groups drops by about 50 point :) |
| 15:11.28 | *** join/#asterisk Slivka (~Slivka@31.40.53.196) |
| 15:11.32 | slav3_kitten | true that |
| 15:11.38 | *** join/#asterisk leedm777 (~leedm777@nat/digium/x-ihakgcvaardepeiz) |
| 15:11.59 | *** join/#asterisk bchia (~Adium@nat/digium/x-dogkmlulnfvqpghb) |
| 15:12.10 | WIMPy | wonderworld: Not very informative that page. |
| 15:13.16 | wonderworld | i tried to keep things as simple as possible... but you are right, maybe a bit more info would be nice. |
| 15:13.21 | WIMPy | invites wonderworld to #asterisk-de |
| 15:13.32 | jmetro | just watched a TED talk about stem cells and how they are starting to lab-grow them |
| 15:13.57 | WIMPy | wonderworld: You know what it's about. I don't. And that doesn;t change by looking at that page. |
| 15:14.04 | coppice | I blame sesame street for not giving george bush an adequate explanation of what stem cells are |
| 15:14.56 | *** join/#asterisk jamicque (~jamicque@80.50.125.74) |
| 15:16.05 | wonderworld | it's a browser based confbridge |
| 15:16.19 | jamicque | hi @ll. Is there any way to configure asterisk to accept any regetrations? I want to configure a serwer in my local net which will accept every reg and call. Making asterisk to accept every call is quite easy (allowguest to yes) is it possible to make somrthing similar to regestrations? |
| 15:16.47 | [TK]D-Fender | jamicque, autocreatepeer=yes |
| 15:17.17 | p3nguin | You don't need a device to be registered to accept calls from it. Just know its IP address. |
| 15:17.34 | *** join/#asterisk ruied (~AndChat66@246.105.103.87.rev.vodafone.pt) |
| 15:20.47 | sp00kz | <PROTECTED> |
| 15:21.01 | [TK]D-Fender | sp00kz, Nope |
| 15:21.08 | sp00kz | damn :( |
| 15:21.19 | WIMPy | sp00kz: That would probably be too good. |
| 15:21.30 | AkkerKid | anyone know how to reset the admin password of a polycom SIP650 that's tied to a hosted provider? |
| 15:21.31 | sp00kz | isn't it slated for including? |
| 15:21.47 | Chainsaw | AkkerKid: Factory reset, then you're back to 456. |
| 15:22.09 | AkkerKid | Chainsaw: how can I factory reset when i don't know the admin password? |
| 15:22.21 | WIMPy | sp00kz: Take a look at the FAQ on their website and take your own conclusions. |
| 15:22.54 | [TK]D-Fender | sp00kz, I'd bet on "no" |
| 15:23.08 | Chainsaw | AkkerKid: Hold down 4 6 8 * on bootup. |
| 15:23.17 | Chainsaw | AkkerKid: When it *then* asks for a password, enter the phone MAC. |
| 15:23.18 | sp00kz | damn, need to get on that! |
| 15:23.33 | Chainsaw | AkkerKid: No colons, all letters in lower case. |
| 15:24.04 | AkkerKid | oh ,lowercase... |
| 15:24.05 | AkkerKid | hmmm |
| 15:24.07 | jmetro | Akker - polycoms should be 55555 i believe. |
| 15:24.10 | *** join/#asterisk jblack (~jblack@pool-108-57-61-167.sctnpa.east.verizon.net) |
| 15:24.20 | Chainsaw | AkkerKid: Yes, upper case won't work. |
| 15:24.51 | AkkerKid | didn't work on try one. |
| 15:24.59 | *** join/#asterisk navaismo (~navaismo@189.191.12.116) |
| 15:25.33 | Chainsaw | AkkerKid: Your window of opportunity is only about 15 seconds long. Time is of the essence. |
| 15:25.34 | Greenlight | I think I've found a bug in the AMI. When I execute a "Bridge" action, if the second channel doesn't exist, in addition to giving an error that it doesn't exist, it continues to try and bridge me anyways, resulting in Channel1 hanging up |
| 15:25.53 | Greenlight | Am i correct in thinking that's not right? |
| 15:25.56 | coppice | sp00kz: they probably just use the toolkit from the chip vendor to provide their software stack, and they really don't have much incentive to add major new functions like IAX to it |
| 15:26.12 | Chainsaw | Greenlight: The failure mode could just be undocumented at this time. |
| 15:27.22 | Greenlight | So, it's supposed to hangup the Channel1 if Channel2 doesn't exist? This is happening in niche cases, where Channel2 has just hangup before I execute the bridge |
| 15:27.36 | *** join/#asterisk cklimos (~Claude@209.5.121.227) |
| 15:27.49 | Chainsaw | Greenlight: So even if you test beforehand, it might still disappear on you. Hmm. |
| 15:28.59 | Greenlight | It does seem incorrect behavior to hangup the call in such cases |
| 15:29.16 | *** join/#asterisk k610 (~Instantbi@cred.epid.ucl.ac.be) |
| 15:29.36 | Chainsaw | Greenlight: Agreed. But I have found that what is convenient and what actually happens can offer differ substantially in the world of Asterisk. |
| 15:29.37 | Greenlight | Usually with AMI commands, if they give an error response, they don't then try and do it anyway |
| 15:29.45 | Chainsaw | offer? often. |
| 15:29.59 | Greenlight | heh :) |
| 15:30.28 | Greenlight | It almost sounds like a "return;" is missing from the function somewhere |
| 15:31.05 | AkkerKid | Chainsaw: i did everything in under 15 seconds the full mac in lowercase and when i hit enter, it just drops back to a normal screen |
| 15:31.09 | Greenlight | But I didn't want to delve in if it was a known issue or if it was there for a reason |
| 15:31.14 | AkkerKid | then '456' still doesn't work |
| 15:31.39 | Chainsaw | AkkerKid: And you have isolated the phone from the network to ensure it is not retrieving a pre-set profile using the boot loader? |
| 15:32.04 | Chainsaw | AkkerKid: Use a PSU temporarily, or deliberately set a VLAN ID that will not get connectivity. |
| 15:32.12 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 15:32.13 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 15:32.22 | Chainsaw | Greenlight: It might be worth having a quick trawl through the JIRA bug tracker for Asterisk. |
| 15:32.46 | jmetro | Chainsaw: If i remember right, his phones are manually configured with their TFTP IP address, so a factory would erase that and then give them nowhere to boot off. |
| 15:32.51 | AkkerKid | Chainsaw: it's not connected to any network cable at all. |
| 15:33.01 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 15:33.12 | AkkerKid | jmetro: this phone is unrelated to that issue. |
| 15:33.30 | *** join/#asterisk Pitel_IPEX (~pitel@gatekeeper.bm.ipex.cz) |
| 15:33.52 | Greenlight | Yea will do - was being lazy coming here first, in case someone knew off hand :) |
| 15:34.25 | *** join/#asterisk keycruncher (~Adium@pa-67-234-31-102.dhcp.embarqhsd.net) |
| 15:36.04 | Pitel_IPEX | I have a little trouble with agi scripts... how can I pass sentence with comma as parameter? like, AGI(greet.php, Hello, my name is Derp), the comma after Hello should not mean end of parameter |
| 15:36.47 | slav3_kitten | jmetro, why is he not setting the dhcp option to tell them what the TFTP ip is? |
| 15:36.55 | slav3_kitten | on cisco phones it's option 150 |
| 15:37.15 | slav3_kitten | but i'm sure there is an official bootp or tftp option |
| 15:40.50 | navaismo | Pitel_IPEX, you can work around changing the coma with other symbol and then replace it in the script. did you tried escaping the comma? |
| 15:41.33 | Pitel_IPEX | quotes seems to work |
| 15:42.25 | *** join/#asterisk ruied (~AndChat66@114.32.166.178.rev.vodafone.pt) |
| 15:43.15 | [TK]D-Fender | Pitel_IPEX, Another suggestion is to save it in a var and pick up the var in AGI |
| 15:53.28 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 15:53.42 | *** join/#asterisk freckle (~jon@firebox.enta.net) |
| 15:55.19 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 15:57.20 | pabelanger | ~itsp-us |
| 15:57.27 | pabelanger | ~itsp |
| 15:57.27 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 15:57.33 | pabelanger | ~itsplist-us |
| 15:57.33 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 15:57.57 | Maliuta | ~itsp-au |
| 15:58.23 | freckle | anyone know if it is possible for another channel to PauseMonitor a different monitored channel? |
| 16:00.37 | jamicque | [TK]D-Fender -> thx autocreatepeer worked as a charm |
| 16:01.36 | *** join/#asterisk CodaDevil (~dberetta@c50-26-189-5.abilcmtk01.tx.dh.suddenlink.net) |
| 16:02.48 | [TK]D-Fender | jamicque, You're welcome. |
| 16:02.53 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 16:19.48 | *** join/#asterisk blee (~blee@72.188.117.219) |
| 16:23.26 | *** join/#asterisk fritz09 (~Adium@pop1-1677.catv.wtnet.de) |
| 16:24.07 | SeRi | fucking termites |
| 16:24.53 | WIMPy | Have you become too fat for your chair and are now blaming termites? ;-) |
| 16:25.20 | Maliuta | Woooo! I know have automatic vim syntax highlighting for ael :) |
| 16:25.46 | Maliuta | Looks as though some stuff could be added to the syntax file though :) |
| 16:25.52 | fritz09 | hi |
| 16:30.46 | *** join/#asterisk drcallcenter (~drcallcen@207.50.14.53) |
| 16:30.56 | drcallcenter | Hello |
| 16:34.27 | Maliuta | meh bed time |
| 16:36.10 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 16:36.11 | *** mode/#asterisk [+o file] by ChanServ |
| 16:37.36 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.122) |
| 16:39.09 | *** join/#asterisk eclark (~eclark@chii.monkeyvoodoo.net) |
| 16:42.32 | eclark | i'm having issues with Asterisk 10.4.0 and WaitForSilence. canadian telcos seem to have recently made a change where their end won't transmit any packets when the remote end is silent |
| 16:43.26 | eclark | waitforsilence appears to get stuck waiting for silence (since it's not getting any packets to check for silence), resulting in the dialplan getting hung up on that step |
| 16:44.36 | Yxa | is asterisk 11 stable? for a new deployment, would you guys recommend I go with 11 or stick with 10 |
| 16:45.05 | eclark | err, on outbound calls, the remote end sends no packets when its end is silent |
| 16:52.55 | [TK]D-Fender | Yxa, *11 has had it's first bug-fix release out... might be good to go... |
| 16:53.04 | [TK]D-Fender | Yxa, Check the current tracker on it first |
| 16:53.09 | ageis | anyone have any cool AGI scripts they want to share, in PHP or Perl |
| 16:53.18 | [TK]D-Fender | Yxa, But *10 is on a death-clock |
| 16:53.54 | [TK]D-Fender | plenty you could google up, but that is a psychotically wide request. AGI's could be for anything at all... |
| 16:54.03 | [TK]D-Fender | ageis, ^ |
| 16:54.16 | ageis | alright |
| 16:55.50 | *** join/#asterisk jacobw (~jacobw@fedora/jacobw) |
| 16:55.57 | [TK]D-Fender | ageis, Samples in the book as well already |
| 16:56.21 | ageis | alreadY? |
| 16:56.38 | [TK]D-Fender | yes. |
| 16:56.40 | [TK]D-Fender | ~book |
| 16:56.40 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:56.43 | jacobw | i want to know how use an If like statement in extensions.conf, can someone point me in the right direction? |
| 16:57.06 | [TK]D-Fender | jacobw, Book above... |
| 16:57.12 | jacobw | yeah :) |
| 16:57.17 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 16:57.27 | jacobw | what is the language actually called? |
| 16:57.31 | [TK]D-Fender | jacobw, "core show application GotoIf", and read up on expressions |
| 16:57.47 | ageis | [TK]D-Fender is there a copy of this book for asterisk 1.6 |
| 16:57.51 | AkkerKid | Question: I set option 66 on my router to direct the phones to the TFTP configuration server. If I factory default the phones, they'll assume option 66 goes FTP and not TFTP. Is there something I'm doing wrong or do I still have to set every phone to TFTP manually? |
| 16:57.58 | jacobw | i don't think it's AGI, AEL seems to be something different too, and 'Asterisk extensions' isn't particulary helpful as a keyword |
| 16:58.11 | AkkerKid | these are polycom 3XX BTW. |
| 16:59.49 | jacobw | [TK]D-Fender: my problem with GotoIf is that i can't prevent execution of B if A is true, i have execute B or A and B |
| 17:00.00 | [TK]D-Fender | jacekowski, Umm... NO. |
| 17:00.29 | *** join/#asterisk rgsteele (~rgsteele@173-15-180-105-BusName-Philadelphia.hfc.comcastbusiness.net) |
| 17:00.30 | [TK]D-Fender | jacobw, go make an actual sample and we'll show you how to fix it. |
| 17:00.47 | freckle | anyone know if it is possible for another channel to PauseMonitor a different monitored channel? |
| 17:02.29 | ageis | [TK]D-Fender: do you think asterisk has enough native scripting capability to read/parse/interpret a file with hours of operation for three locations, or should I use AGI |
| 17:02.56 | ageis | Also has anyone used this https://github.com/drivefast/asterisk-res_json |
| 17:03.03 | Qwell | ageis: Asterisk can do plenty via dialplan. |
| 17:03.05 | [TK]D-Fender | who are you making that file? how often is it updated? |
| 17:03.08 | [TK]D-Fender | how* |
| 17:03.30 | [TK]D-Fender | ageis, There are lots of things you could probably do at the same time that would make doing it in dialplan dead easy instead of depending on that file. |
| 17:03.31 | jacobw | [TK]D-Fender: http://pastebin.com/ZKpQUfCN |
| 17:03.42 | ageis | [TK]D-Fender we're going to pull it from another server periodically, let's say hourly. it can be any format we want it to. Basically 3 keys with 7 values each. (hours for 6 days and an 0/1 emergency override) |
| 17:03.46 | ChannelZ | CURL! |
| 17:04.04 | ageis | yeah I know I can CURL |
| 17:04.08 | *** join/#asterisk drcallcenter (~drcallcen@207.50.14.53) |
| 17:04.26 | Qwell | jacobw: If you're just doing one thing, why not ExecIf()? |
| 17:04.30 | ageis | [TK]D-Fender: unfortunately my boss has decreed that I must design it this way. I wanted to use a PHP script to write a piece of the dialplan config which would then get included. |
| 17:04.40 | [TK]D-Fender | jacobw, http://pastebin.com/33LS0hx3 |
| 17:04.55 | drcallcenter | Would anyone be able to help me with dtmf signals being ignored by a third party ivr, when an agi script is run when the ivr call is placed |
| 17:04.56 | drcallcenter | ? |
| 17:05.01 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 17:05.15 | [TK]D-Fender | jacobw, Or you could just use "core show function IF" in a single set and do it in one line. |
| 17:06.28 | *** join/#asterisk chris_n (~Chris@184.7.21.42) |
| 17:07.24 | [TK]D-Fender | ageis, i fail to see his point... |
| 17:08.16 | ageis | [TK]D-Fender: we want a "read only" operation so that if the remote site we're pulling this file from is down then it will stay unmodified, so no hard interdependency or brute forcing scripting capability |
| 17:08.17 | [TK]D-Fender | ageis, there are a lot of "interpretations" on how you could deal with this... |
| 17:08.49 | ageis | I argued for a cron PHP script that writes out the dialplan based on data from this retrieved file |
| 17:08.53 | [TK]D-Fender | ageis, I see no difference in pulling an actual * config. |
| 17:08.59 | ageis | He insists I need to do it with Asterisk |
| 17:09.02 | ageis | Ah |
| 17:09.06 | ageis | That's a good idea |
| 17:09.11 | [TK]D-Fender | <ageis> I argued for a cron PHP script that writes out the dialplan based on data from this retrieved file <- this works too |
| 17:09.31 | [TK]D-Fender | ageis, I'd sooner do that than live-parse that POS every damn call... |
| 17:09.44 | ageis | yeah |
| 17:09.53 | ageis | I like your design, have the remote server write the config |
| 17:09.57 | [TK]D-Fender | he wants to put it out flat? OK... parse it LOCAL on pull though... |
| 17:10.28 | [TK]D-Fender | ageis, It's a philosophical difference if you generate the actual dialplan AFTER the pull, or AS the format of what you pull. |
| 17:11.08 | [TK]D-Fender | ageis, technically it should be more secure to parse a more "raw" file provided on pull though. that prevents hijacking and code injection. |
| 17:11.32 | ageis | Ah |
| 17:12.24 | [TK]D-Fender | ageis, might make you boss "happy". So pull raw format, parse on successful pull. |
| 17:12.53 | ageis | Yah. but I'm wondering what my raw format should be like. PHP arrays? What kind of data structures can I use that Asterisk can handle. JSON/XML ? |
| 17:13.02 | ghost75 | [TK]D-Fender: did u use geoip on apache ? |
| 17:13.15 | ageis | that was my initial question |
| 17:13.32 | [TK]D-Fender | ghost75, ? |
| 17:13.45 | ghost75 | you mentioned geoip yesterday |
| 17:14.04 | [TK]D-Fender | ageis, Well your boss is already micro-managing your job .. you mean he hasn't already dictated the format too? ;) |
| 17:14.22 | ageis | [TK]D-Fender: lol nope |
| 17:14.26 | [TK]D-Fender | ghost75, I did? please link the exact reference and to whom I was referring. |
| 17:14.45 | ghost75 | not sure lol |
| 17:14.46 | [TK]D-Fender | ageis, I hate twits like that.... You don't let morons design systems. |
| 17:15.08 | [TK]D-Fender | ageis, Describe the goal and the factors, but not the precies means. |
| 17:15.47 | ageis | [TK]D-Fender: Yeah I mean... the way he wants to design things is super cautious and as failsafe as possible, even though I explained that we can do my design with enough error checking and validation that it's completely solid |
| 17:16.16 | ageis | [TK]D-Fender: We're developing a dynamic dialplan where we can set Open/Closed and hours of operation for 3 locations over 6 days of the week. |
| 17:16.32 | ageis | And it will behave differently depending on those variables |
| 17:16.50 | ageis | That stuff will be managed/set in a CMS module on another server, which our phone server needs to communicate with to grab |
| 17:17.27 | *** join/#asterisk ghost75 (~trechber@dslb-188-105-020-031.pools.arcor-ip.net) |
| 17:17.40 | ageis | So If the module doesn't write the dialplan in asterisk format, I'm wondering what kind of data structure I should use to export this file, how should I script the intrepreting of this file in Asterisk, whether I should use AGI, etc. |
| 17:18.15 | ageis | Ok he's fine with the module writing the config---so I don't need to worry about that anymore :) |
| 17:18.45 | [TK]D-Fender | ageis, pull & LOCAL parse make sense. |
| 17:19.04 | slav3_kitten | cackles |
| 17:19.09 | [TK]D-Fender | satisfies both ends and allows you to make site-specific adaptations. |
| 17:19.19 | slav3_kitten | now to setup my voip.ms crap *looks nervous* |
| 17:19.31 | *** join/#asterisk oej_ (~olle@2001:16d8:cc57:1000::42:1003) |
| 17:20.39 | jacobw | Qwell, [TK]D-Fender, thanks |
| 17:21.46 | [TK]D-Fender | jacobw, You're welcome. remember... you aren't stuck running through that other code... just jump to the end :) |
| 17:22.29 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 17:26.15 | gg608f | hi all, I donwloaded Asterisk 11.1.0-rc1 anyone knows if this contains vp8 support ? |
| 17:26.53 | WIMPy | What is vp8? |
| 17:27.03 | ghost75 | can u please click and tell me if you see your country: http://ghostadmin.dyndns.org:8081/geoiptest.php |
| 17:27.27 | WIMPy | ghost75: Empty |
| 17:27.37 | ghost75 | both fields? |
| 17:27.46 | WIMPy | yes |
| 17:28.09 | freckle | ghost75: Your Country is correct, country code is blank |
| 17:28.26 | ghost75 | are you using proxy WIMPy |
| 17:28.46 | WIMPy | ghost75: Yes. The one next door. |
| 17:29.30 | ghost75 | wonder why this doesnt give country code |
| 17:29.56 | WIMPy | You shouldn't put too much trust in these services anyway. |
| 17:30.16 | ghost75 | i use it to block spammers from china and so on |
| 17:30.17 | WIMPy | But so far it has worked with this IP. |
| 17:30.44 | elico | ghost75: whats the php code? |
| 17:31.02 | ghost75 | $country_name = apache_note("GEOIP_COUNTRY_NAME"); |
| 17:31.02 | ghost75 | $country_code = apache_note("GEOIP_COUNTRY_CODE_BY_NAME"); |
| 17:32.42 | elico | ghost75: well did you took the time to see the example at php.net ? |
| 17:33.01 | ghost75 | i tried like 5 examples |
| 17:33.22 | elico | it seems like it works ..code .. based on hostname and not based on plain IP |
| 17:33.39 | elico | it should work with IP.. |
| 17:33.57 | ghost75 | its just a local database full of ip ranges where it looks |
| 17:34.24 | elico | by the way... the whole code... |
| 17:34.33 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 17:35.53 | ghost75 | what do you see now on the page? |
| 17:36.55 | *** join/#asterisk andy09usa (~andy09usa@audotov.com) |
| 17:37.05 | slav3_kitten | crap ok i have a question here. apparently nowakeHHMM is pattern matching to my outbound sip. is there any way to prevent the "nowake" from pattern matching the 10 digit outbound patten? |
| 17:37.27 | elico | works |
| 17:37.44 | ghost75 | all 3 ? |
| 17:37.47 | elico | yes |
| 17:37.50 | ghost75 | damn |
| 17:38.05 | [TK]D-Fender | slav3_kitten, ... huh? |
| 17:38.12 | ghost75 | its GEOIP_COUNTRY_CODE without _BY_NAME |
| 17:38.21 | elico | ghost75: what caused it? and how is related to asterisk? |
| 17:38.32 | ghost75 | not related at all lol |
| 17:38.41 | ghost75 | but thx for testing |
| 17:38.46 | elico | funny you |
| 17:38.58 | elico | I am using a lot with ruby for these kind of tasks |
| 17:38.59 | ghost75 | i cannot test myself as it detects local |
| 17:39.03 | slav3_kitten | [TK]D-Fender, i'm running a wakeup call script, it's matching my 10 digit outbound dial |
| 17:39.11 | slav3_kitten | maybe i'll just make it *wake an *nowake |
| 17:40.16 | slav3_kitten | or if i shift it position wise on the includes... |
| 17:40.30 | *** join/#asterisk cusco (~tralala@2001:41d0:1:6caf::1) |
| 17:41.56 | cusco | hi |
| 17:42.03 | elico | ghost75: you can always use a proxy.. |
| 17:42.06 | cusco | asterisk manager event AgentCalled |
| 17:42.15 | cusco | can I rely on the block line position? |
| 17:42.18 | cusco | like line 5 |
| 17:42.33 | cusco | take this for example: http://paste.debian.net/209569/ |
| 17:42.35 | *** join/#asterisk oej (~olle@2001:16d8:cc57:1000::42:1003) |
| 17:42.51 | cusco | before lines 4 and 5 would not show |
| 17:42.54 | cusco | and now they're showing |
| 17:42.56 | cusco | thus mi asking |
| 17:43.00 | cusco | can I rely on line number? |
| 17:43.01 | *** join/#asterisk imox (~imox@91-66-32-57-dynip.superkabel.de) |
| 17:47.38 | elico | cusco: like if maybe it's the ID and not the real number? |
| 17:47.49 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.111) |
| 17:51.21 | *** join/#asterisk mokmeister (~mokmeiste@109.78.34.118) |
| 17:52.06 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 17:54.22 | *** join/#asterisk Greenlight (~email@cpc1-dund9-0-0-cust1000.sgyl.cable.virginmedia.com) |
| 17:59.20 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 18:01.25 | jmetro | trying to configure the front screen on a snom 870...or just add my park buttons on there somehow. suggestions? |
| 18:07.48 | cusco | elico: ? |
| 18:09.58 | *** part/#asterisk eclark (~eclark@chii.monkeyvoodoo.net) |
| 18:10.24 | *** join/#asterisk imox (~imox@91-66-32-57-dynip.superkabel.de) |
| 18:20.10 | slav3_kitten | looks a bit confused |
| 18:20.42 | slav3_kitten | ok so if someone calls in an you have no ivr or anything setup. your extensions.conf has exten => phone,1,Asnwer() |
| 18:20.54 | slav3_kitten | will it just give them a dial tone where they can enter an extension? |
| 18:21.39 | jmetro | hm |
| 18:21.40 | [TK]D-Fender | slav3_kitten, No. |
| 18:21.42 | jaytee | When setting up a Polycom 331/321 phone to mount on a wall, how do you get the little handset hook to come out. Do you have to pry the plug in the back off with a knife or something? I can't get it to budge with just my fingers. |
| 18:21.59 | *** join/#asterisk kresp0 (~kresp0@81.61.24.25.dyn.user.ono.com) |
| 18:22.00 | [TK]D-Fender | slav3_kitten, want tone? background an audio file of it. Or use Disa() (horrible idea). |
| 18:22.08 | [TK]D-Fender | slav3_kitten, You SHOULD be making a basic IVR for this |
| 18:22.19 | [TK]D-Fender | jaytee, Yes |
| 18:22.19 | slav3_kitten | well i was just curious what would happen with no IVR |
| 18:22.33 | [TK]D-Fender | slav3_kitten, * doesn't jsut wait for input out of nowhere |
| 18:22.49 | jmetro | you set up an ivr, and just dont play a sound file. |
| 18:22.57 | jmetro | dead air. |
| 18:23.15 | slav3_kitten | so it'd answer then see nothing else to do an hang up, in that example?? |
| 18:24.19 | [TK]D-Fender | correct |
| 18:24.21 | elico | cusco: what did you asked? to rely on the line as what? |
| 18:24.30 | jmetro | though you can set the timeout. |
| 18:24.35 | slav3_kitten | nifty |
| 18:24.37 | [TK]D-Fender | Run out of steps typically leads to autofallthrough |
| 18:24.53 | [TK]D-Fender | Which is the default state |
| 18:24.58 | [TK]D-Fender | So timeouts = irrelevant |
| 18:25.01 | jaytee | [TK]D-Fender, thanks. It's been awhile since I had to wall mount one of these. |
| 18:25.04 | jmetro | anyone know how to setup the front screen of a snom 870 while youre all alive =p |
| 18:25.51 | [TK]D-Fender | jaytee, i've got a 320 in my cafeteria, and a 430 in my lobby here |
| 18:30.52 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:30.52 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 18:32.31 | *** join/#asterisk n8ideas (~joshua@65.112.207.3) |
| 18:32.56 | n8ideas | can anyone possibly give me a little insight on the reason for the msg_id addition to voicemail messages in ODBC voicemail in Asterisk 11? |
| 18:33.07 | *** join/#asterisk Takapa (vegard@svanberg.no) |
| 18:37.27 | Qwell | n8ideas: So that messages have a unique ID that can be referenced. |
| 18:38.01 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 18:39.16 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 18:42.28 | *** join/#asterisk _Corey_ (~chatzilla@173-161-229-46-Philadelphia.hfc.comcastbusiness.net) |
| 18:48.10 | slav3_kitten | tcpdump clusterfuck, run it via ssh :D |
| 18:48.43 | slav3_kitten | it's like kaboom flood of ssh traffic |
| 18:56.49 | SeRi | waz up slav3_kitten |
| 18:57.19 | SeRi | cool my port will finish tomorrow :) |
| 18:57.33 | slav3_kitten | SeRi, trying to debug an ssh thing, while connected to ssh because i totally forgot i was |
| 19:00.14 | *** join/#asterisk fritz09 (~Adium@2002:542e:5694:e472:90cc:91a8:add5:dfb6) |
| 19:04.13 | *** join/#asterisk jiku (~jikz@106.195.222.137) |
| 19:04.19 | jiku | Hi all.. |
| 19:04.34 | jiku | have a small issue with asterisk with x100p.. |
| 19:04.58 | jiku | trying to pass manager Originate sequence.. |
| 19:05.20 | jiku | Channel: dahdi/g0/012321231 |
| 19:05.29 | jiku | asterisk does originate the call to this channel |
| 19:05.35 | jiku | but it automatically answers.. |
| 19:06.27 | jiku | http://pastebin.com/egAMKWNt |
| 19:06.58 | jiku | instead of asterisk waiting for the user in this phone, answering, it answers by itself. |
| 19:07.08 | jiku | any idea why this is happening ? |
| 19:08.21 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 19:08.28 | navaismo | jiku, analog channels doesnt have answer supervision as far i know, but you can ask to your telco if they support it and check the settings in your chan_dadhi.conf |
| 19:08.49 | navaismo | if you want answer suppervivion use an itsp provider |
| 19:09.09 | jiku | navaismo, no wonder.. this config worked in other location and that had a pri line.. |
| 19:09.16 | jiku | navaismo, you would be right. |
| 19:09.36 | jiku | so i can originate to a sip channel correct? |
| 19:09.41 | jiku | and that wouldn't be a problem right ? |
| 19:10.37 | navaismo | Pri line is digital line too, and yes I suppose with "sip trunk " will work fine |
| 19:11.24 | jiku | thansk navaismo.. i will check.. |
| 19:14.27 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 19:19.46 | *** join/#asterisk eject_ck (~Evgeniy@213.159.242.65) |
| 19:21.32 | eject_ck | Hi all , how can I in dialplan pass variable and get result from external programm ? |
| 19:22.24 | eject_ck | basically I want to send phone number and get email from my csv file :) |
| 19:24.30 | navaismo | eject_ck, you can use agi with phpagi |
| 19:29.29 | slav3_kitten | ok SeRi |
| 19:29.54 | eject_ck | navaismo: thank you |
| 19:29.59 | slav3_kitten | i should be able with ports 10000-20000 forwarded to my asterisk box to run tcpdump port 15000 |
| 19:30.33 | slav3_kitten | then have a friend hping3 the 15000 port an see it right? |
| 19:31.42 | eject_ck | navaismo: where I can get examples ? |
| 19:31.49 | eject_ck | for phpagi ? |
| 19:32.22 | [TK]D-Fender | eject_ck, "core show function SHELL" |
| 19:33.06 | navaismo | eject_ck, http://phpagi.sourceforge.net/ |
| 19:33.16 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 19:33.21 | SuperNull | Hey gents. |
| 19:34.21 | navaismo | eject_ck, and here http://phpagi.sourceforge.net/phpagi22/api-docs/__examplesource/exsource__home_phpagi_devel_phpagi_examples_ping.php_a8afcfe0902072cc636965d2b445e3fc.html |
| 19:34.28 | navaismo | and here http://phpagi.sourceforge.net/phpagi22/api-docs/__examplesource/exsource__home_phpagi_devel_phpagi_examples_dtmf.php_eb30ab5fd2d98b4b41f251416fc7ed37.html |
| 19:34.39 | SuperNull | Anyone use openser/sips as a SBC with pbx user validation for outbound ? |
| 19:36.10 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:36.10 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 19:38.22 | *** join/#asterisk afink (4172ce22@gateway/web/freenode/ip.65.114.206.34) |
| 19:47.46 | [sr] | is it possible |
| 19:47.50 | [sr] | in any way or form |
| 19:48.17 | [sr] | to "proxy" isdn interfaces so that i could eventually record everything that passed thro there? |
| 19:48.38 | [sr] | like if this "proxy" (hardware) was a squid |
| 19:48.50 | Chainsaw | [sr]: You want a "passive tap". |
| 19:48.56 | Chainsaw | [sr]: Sangoma seems particularly adept at that. |
| 19:49.03 | [TK]D-Fender | [sr], there is no "proxy". * is a B@BUA |
| 19:49.09 | [TK]D-Fender | ~b2bua |
| 19:49.09 | infobot | b2bua is probably a Back 2 Back User Agent. Additional information is available on wikipedia: http://en.wikipedia.org/wiki/Back-to-back_user_agent |
| 19:49.20 | [TK]D-Fender | [sr], Dial in, dial out. The End. |
| 19:49.41 | Chainsaw | All I see is Dido, and a song starts in my head. |
| 19:50.31 | [sr] | [TK]D-Fender: i was not thinking as a part os asterisk, but Chainsaw seems to answered my question, it was something that come to my mind it it would even be possible |
| 19:50.48 | [sr] | it it = if it |
| 19:51.10 | Chainsaw | [sr]: The government has a need for it, so you can bet that there are hardware manufacturers that manufacture the kit to do it. |
| 19:52.00 | Chainsaw | [sr]: I can't say I particularly like the Sangoma driver layout, with that strange extra layer between DAHDI and the adapter... but I've seen passive tap references in it. |
| 19:54.11 | [sr] | Chainsaw: im searching google, well i see, had no idea that was possible to "intercept" the call's before the PBX, no matter if it's asterisk or other |
| 19:54.31 | Chainsaw | [sr]: If you cable it right, neither side notices. |
| 19:57.08 | [sr] | i wonder if this would be possible to do with as asterisk, inserting the line and then the line go out for the PBX, use asterisk just as an intercept (of course asterisk has recording caps on it's own, that's just an idea) |
| 19:57.25 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.110) |
| 19:57.49 | Chainsaw | [sr]: You can likely use Asterisk to make sense of the ISDN data you've passively tapped. |
| 19:58.15 | Chainsaw | [sr]: But it wouldn't be quite like a proxy with two sets of ISDN adapters. It would be a single, specifically-modified ISDN adapter. Or purpose-made tap board. |
| 19:58.45 | [sr] | i see |
| 19:59.07 | [sr] | hey i have a question |
| 19:59.41 | [TK]D-Fender | Radio Shack : You have questions ... we have batteries. |
| 19:59.48 | Chainsaw | *mechanical noise* Please speak your question now. |
| 20:00.21 | [sr] | not related, i tried to make an instalation of asterisk, with a BRI card, and the connection to the NTBA's are ok, everything was tested but cannot make call's, on the network side, there's different ways to configure the BRI channels? |
| 20:01.02 | [sr] | maybe i didn't explained quite well |
| 20:01.40 | Chainsaw | [sr]: It's important to get the ISDN adapter configured correctly. For the correct country & protocol, particularly. |
| 20:02.17 | Chainsaw | [sr]: As is often the case, American signalling differs from rest-of-world signalling. |
| 20:02.50 | [sr] | Chainsaw: country is the same, protocol was the same also |
| 20:03.05 | [sr] | euroisdn, bri_cpe_ptmp |
| 20:03.48 | [sr] | it works on other places, less on the place i want to install it, on about 20 instalattions i made, its the 1st time this happened, already tried with 2 new NTBA's |
| 20:04.28 | [sr] | from the operator side, they have different configurations for BRI? i have no idea, maybe an old configuration since the PBX that it's there has > 15 years? |
| 20:08.47 | [sr] | Chainsaw: i always test it on my company BRI's, and then they work without problems on the final destination BRI's, and now i got stuck with this, i'm not sure is it's possible or if exists different configuration in the operator side also |
| 20:09.25 | Chainsaw | [sr]: Short answer: Yes. |
| 20:10.24 | [sr] | Chainsaw: ah, for which words/info should i search for the operator side confs? |
| 20:11.04 | Chainsaw | [sr]: Particularly whether they want PTMP on your end. |
| 20:12.23 | [sr] | i tried with ptmp on and of on the NTBA's |
| 20:14.12 | *** join/#asterisk kpettit (~kpettit@99.116.144.119) |
| 20:15.48 | [sr] | Chainsaw: for switchtype cant be nothing different than euroISDN on my case since the contry is the same, where can i see a list of possible signalling options? i tried with the usual bri_cpe_ptmp and bri_net (this one complains) and none worked |
| 20:17.05 | Chainsaw | [sr]: You are a cpe, that is fairly certain. Customer Premises Equipment. You are not originating the network, that is generally left to your telco. |
| 20:19.03 | [sr] | Chainsaw: i see, it must he something on the operator side, can i ask you what it would it possible be? remember that this workes in other places, and on this place doesn't with some configuration |
| 20:20.18 | Chainsaw | [sr]: Well I can tell you're in Germany, based on your terminology. We call that an NT1. |
| 20:21.10 | [sr] | Chainsaw: portugal |
| 20:21.22 | *** join/#asterisk jiku (~jikz@106.79.10.113) |
| 20:21.36 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.110) |
| 20:22.37 | Chainsaw | [sr]: Some ISDN NT1s have two outputs, but only work properly if you use the first. |
| 20:24.03 | slav3_kitten | growls |
| 20:24.10 | Chainsaw | [sr]: Other then that, Italy uses American-style 21BQ signalling where everyone else uses 4B3T. |
| 20:24.33 | Chainsaw | [sr]: So if the problematic circuit is near the Italian border, try the other signalling and see if that works better. |
| 20:24.51 | [sr] | Chainsaw: well they all have two, but tried the 1st port and nothing, |
| 20:24.58 | [sr] | Chainsaw: way far from itally :) |
| 20:25.07 | Chainsaw | [sr]: Then they really should all be 4B3T. |
| 20:26.01 | *** join/#asterisk vlad_sta_ (~vlad_star@83.149.8.102) |
| 20:27.34 | madduck | At the moment, for a SIP registrar, I have both a register=> line and a friend stanza in sip.conf. Do I need the friend stanza? What for? I would like to clean up 'sip show peers' to only show the clients… |
| 20:28.03 | Qwell | madduck: You need both. |
| 20:28.10 | Qwell | Why is it a friend though? No idea. |
| 20:29.08 | slav3_kitten | anyone use voip.ms? |
| 20:29.15 | [TK]D-Fender | Qwell, Could be they send calls authed by name from various IP's |
| 20:29.21 | madduck | Qwell: why do I need both? Why has noone eliminated this horrible redundancy yet? ;) |
| 20:29.28 | [TK]D-Fender | Qwell, slav3_kitten Lots of people do. |
| 20:29.51 | Qwell | madduck: Patches welcome. |
| 20:30.09 | slav3_kitten | ok, ever had your asterisk box say it's registered but the voip.ms page not show any registrations? |
| 20:30.11 | [TK]D-Fender | madduck, Because telling * to register has nothing to do with the AUTH requirement for actual calls they'll send you. And doesn't define codecs for them. Or anything for that matter. |
| 20:30.27 | [TK]D-Fender | madduck, Registering has nothing to do with processing an actual call. |
| 20:30.41 | [TK]D-Fender | slav3_kitten, Show us |
| 20:31.38 | slav3_kitten | <PROTECTED> |
| 20:32.05 | slav3_kitten | and the voipms portal home, says no registrations found |
| 20:32.30 | slav3_kitten | calls to my DID do connect though |
| 20:32.30 | *** join/#asterisk bchia (~Adium@nat/digium/x-zarqfzcdjmvcwdrg) |
| 20:32.31 | madduck | [TK]D-Fender: right. I understand. Is it possible to prevent a friend from appearing in "sip show peers"? |
| 20:33.42 | *** join/#asterisk eclark (~eclark@chii.monkeyvoodoo.net) |
| 20:33.51 | eclark | how do i add a tech prefix to a call in a .call file? |
| 20:34.37 | [TK]D-Fender | madd no |
| 20:34.56 | eclark | the normal "Extension: SIP/[trunk]/[prefix][extension]" isn't working. immediate hangup |
| 20:35.45 | eclark | err, "Channel" not "Extension" |
| 20:36.26 | [TK]D-Fender | eclark, just like you would with Dial() |
| 20:36.59 | eclark | i get an immediate hangup when i do it like that, though: NOTICE[22929]: pbx_spool.c:369 attempt_thread: Call failed to go through, reason (1) Hangup |
| 20:37.20 | *** join/#asterisk jblack_ (~jblack@65.242.101.42) |
| 20:37.36 | [TK]D-Fender | eclark, You should probably be showing us the EXACT sample in full along with all output. |
| 20:37.58 | kpettit | I'm having a problem getting "automon=*1" toggle call recording to work. Other features in the map work, and always record calls work. I just can't get it to toggle call recording on a live call. |
| 20:38.00 | kpettit | Any ideas? |
| 20:40.00 | [TK]D-Fender | kpettit, idea : show us your config and complete attempt |
| 20:40.06 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 20:40.18 | eclark | here's the full .call file and error: http://pastebin.com/uw7aGjkv |
| 20:40.53 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 20:40.55 | kpettit | [TK]D-Fender, which part of the config? The feature map just has a couple lines. |
| 20:41.46 | [TK]D-Fender | eclark, SIP DEBUG isn't enabled through that, is it? |
| 20:42.08 | eclark | debug isn't on, no |
| 20:42.16 | eclark | it doesn't even make it into the dialplan |
| 20:43.29 | *** join/#asterisk vfabi (~fabi@host-static-89-41-121-42.moldtelecom.md) |
| 20:44.03 | eclark | i also just noticed i copied the wrong error. the only difference is the pound sign on the one i should have copied |
| 20:44.21 | eclark | (i'd tried two .call files, one with the #, one without. same error both times) |
| 20:47.27 | [TK]D-Fender | eclark, Indeed the # looked blatantly wrong and I wanted you to have actually seen the attempt |
| 20:47.48 | [TK]D-Fender | eclark, And it doesn't have to make it to the dialplan... the DIAL fails |
| 20:47.58 | eclark | it errors out the same way with or without the # |
| 20:48.05 | [TK]D-Fender | elguero, Show the complete call. |
| 20:48.13 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 20:48.19 | [TK]D-Fender | eclark, Along with your even creating the call file |
| 20:49.09 | eclark | i'm not sure what you mean with that last line, [TK]D-Fender |
| 20:49.26 | [TK]D-Fender | eclark, I want to see precisely where you make and drop it in |
| 20:50.13 | eclark | i have a file whose contents are in that pastebin. i move it into /var/spool/asterisk/outgoing, and it generates the error in the pastebin |
| 20:50.36 | slav3_kitten | ok i have a stupid question here. if i include a context that has includes. are all those included as well? |
| 20:50.52 | jmetro | yep |
| 20:51.12 | [TK]D-Fender | eclark, New call debug...... |
| 20:51.14 | jmetro | wish the guys at trixbox knew that. |
| 20:51.26 | jmetro | cause god their extensions.conf is nasty looking. |
| 20:51.27 | [TK]D-Fender | jmetro, Trixbox has been dead for YEARS |
| 20:51.31 | slav3_kitten | so best not to include localsets in an inbound context else everything will be allowed |
| 20:51.38 | [TK]D-Fender | People still using it should be dragged out and shot... |
| 20:51.55 | [TK]D-Fender | Along with people who configure Polycom phones via the web interface. |
| 20:52.18 | slav3_kitten | [TK]D-Fender, can polycom pull their config from a tftp like cisco phones can? |
| 20:52.33 | [TK]D-Fender | jmetro, And you're not supposed to be messing with GUI generated dialplan anyway |
| 20:52.41 | [TK]D-Fender | slav3_kitten, Always. |
| 20:52.51 | slav3_kitten | < never used polycom |
| 20:53.08 | [TK]D-Fender | slav3_kitten, FTP, TFTP, FTPS, HTTP, HTTPS..... |
| 20:53.09 | slav3_kitten | imho tftp hosted configs are the way to do it |
| 20:53.49 | jmetro | well yeah |
| 20:54.02 | slav3_kitten | nice |
| 20:54.41 | jmetro | just trying to figure out how Snom does it...apparently requires multiple config files |
| 20:54.54 | eclark | [TK]D-Fender: new call: http://pastebin.com/yhVQBY6B |
| 20:55.13 | slav3_kitten | sighs as he tries to figure out how to do an auto attendant properly |
| 20:55.30 | *** join/#asterisk nix8n82 (~nate@64.118.54.50) |
| 20:55.53 | [TK]D-Fender | <[TK]D-Fender> eclark, SIP DEBUG isn't enabled through that, is it? <------ |
| 20:56.05 | jmetro | 0 is a valid option, and make commonly used options first. And dont talk too much! |
| 20:56.52 | eclark | no, should i have enabled SIP debug? |
| 20:57.37 | [TK]D-Fender | wonders if he's going to have to say it a THIRD time. |
| 21:00.15 | eclark | you asked if it was, i told you no |
| 21:00.17 | eclark | here's with debug http://pastebin.com/8HMnGdNf |
| 21:00.42 | eclark | i notice the # is being converted to %23 |
| 21:01.11 | [TK]D-Fender | SIP/2.0 404 Invalid dialing format - support@flowroute.com |
| 21:01.35 | [TK]D-Fender | Where do you see a description from them that what you are dialing should be considered acceptable? |
| 21:02.00 | eclark | i'm doing what they said to do when i talked with their support |
| 21:02.40 | [TK]D-Fender | eclark, Is it in writing somewhere? |
| 21:03.05 | [TK]D-Fender | eclark, It certainly doesn't look right... |
| 21:03.33 | eclark | https://support.flowroute.com/entries/20593986-ip-based-authentication-outbound |
| 21:05.18 | *** part/#asterisk CodaDevil (~dberetta@c50-26-189-5.abilcmtk01.tx.dh.suddenlink.net) |
| 21:06.20 | [TK]D-Fender | eclark, And are you registering to them? |
| 21:06.41 | eclark | yes |
| 21:06.50 | [TK]D-Fender | Read the text in BLUE on that page |
| 21:07.11 | [TK]D-Fender | Seems to say this is not required for you |
| 21:08.25 | [TK]D-Fender | "Please note: This setup is only for systems on a static external IP" |
| 21:08.28 | eclark | except that on some calls, they end up asking for credentials after the call has started, and asterisk doesn't send them |
| 21:08.29 | [TK]D-Fender | Also right up top |
| 21:08.40 | eclark | this is a system on a static external IP |
| 21:09.13 | [TK]D-Fender | eclark, just because, set "pedantic=yes" under [general] and retest |
| 21:09.36 | [TK]D-Fender | eclark, Last I checked that was only for inbound, but lets see if there is a difference |
| 21:11.11 | eclark | same invalid dialing format error |
| 21:11.49 | [TK]D-Fender | I am expecting to actually SEE this attempt/// |
| 21:15.30 | SeRi | ok I am starting to love the jetway |
| 21:17.26 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 21:23.28 | trumee | anybody using csipsimple? |
| 21:23.35 | trumee | on android phone? |
| 21:25.12 | *** join/#asterisk brian98 (~brian98@79.97.2.26) |
| 21:25.36 | [TK]D-Fender | And ... time's up. Heading home. BBIAB. |
| 21:27.36 | pbxbrian | Hello Hello |
| 21:28.56 | pbxbrian | Quick question for the gurus in the house |
| 21:29.46 | pbxbrian | from time to time some customers manage to use an attendant console pickup feature in some way that leaves the channel stuck. |
| 21:29.57 | pbxbrian | core show channels shows SIP/101-b6aa36 *8@from-inside-context Ring (None) |
| 21:30.20 | pbxbrian | and soft hangup that channel just gives Requested Hangup on channel 'SIP/101 |
| 21:30.37 | pbxbrian | the only way to get rid of the channel is to core restart asterisk |
| 21:35.40 | slav3_kitten | so for dialing say 3 extensions it should be Dial(SIP/something[&SIP/somethingelse[&SIP/onemoresomething]]) ? |
| 21:39.23 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 21:42.14 | slav3_kitten | nope that didn't work, gave everyone busy/congested |
| 21:44.53 | slav3_kitten | hmm |
| 21:49.20 | kaldemar | slav3_kitten: drop the square brackets |
| 21:50.14 | kaldemar | they indicate something optional in the syntax |
| 21:51.37 | slav3_kitten | ah |
| 21:51.41 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:51.42 | slav3_kitten | sorry i misread it :| |
| 22:04.38 | SeRi | slav3_kitten: you using y22g? |
| 22:04.56 | SeRi | s/y22g/722g/ |
| 22:05.35 | slav3_kitten | ulaw so 711 |
| 22:07.45 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.102) |
| 22:12.10 | ghost75 | i enabled g.729 but everything comes in at g.711 :< |
| 22:12.55 | slav3_kitten | ghost75, does your carriers support 729? |
| 22:13.17 | slav3_kitten | iirc 729 is lower quality then ulaw so disallow=ulaw |
| 22:13.42 | slav3_kitten | so basically it'll go with the highest quality codec it can, someone correct me if i'm wrong. |
| 22:13.55 | ghost75 | i the caller wants g.711 then he get it |
| 22:14.06 | ghost75 | i want lower bandwidth |
| 22:14.07 | SeRi | most will be 711 |
| 22:14.19 | SeRi | g722 mostly for point to point or ip calls |
| 22:14.24 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:14.24 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 22:14.33 | SeRi | unles supported internally |
| 22:14.36 | slav3_kitten | ghost75, then if they allow 729 disallow ulaw |
| 22:14.57 | SeRi | most itsp allow g722 inside their network |
| 22:14.57 | Qwell | How, precisely, did you enable G.729? |
| 22:15.02 | ghost75 | if i block g.711 i guess it wont work |
| 22:16.02 | [TK]D-Fender | Why are we "guessing"? |
| 22:16.55 | ghost75 | guessing before trying |
| 22:17.51 | slav3_kitten | ghost75, this is pretty simple. what does your sip carrier allow for their codecs. if they allow ulaw and 729, then disable ulaw via disallow=all, then only allow=g729 in your sip.conf section for the provider |
| 22:20.42 | SeRi | slav3_kitten: there is no such thing as a sip carrier :) There is ITSP :P |
| 22:21.05 | slav3_kitten | SeRi, you knew what i meant and i couldn't remember the proper term |
| 22:21.11 | SeRi | lol |
| 22:21.23 | SeRi | who is your tcp/ip carrier? |
| 22:21.35 | SeRi | j/k :P |
| 22:21.59 | slav3_kitten | SeRi, some giant backbone |
| 22:22.23 | slav3_kitten | oh speaking of those cunts... i can say cunts in here right? |
| 22:23.19 | ghost75 | i disabled alaw/ulaw and called my nr from mobile and got g.729 (but here i always had g.729) |
| 22:24.40 | slav3_kitten | so i set some port forward shit up on my router an tell a friend to portscan me and generate some tcp/udp traffic on the ports i opened up. nothing shows up in tcpdump. then hour later he does a udp on the iax port and i see some traffic so it shows my shit is working right. call the isp, "no we are not natting you or doing anything that'd prevent a nmap" |
| 22:24.52 | slav3_kitten | except the nmap showed literally nothing at all |
| 22:25.15 | slav3_kitten | after the call, it shows the proper things open aside from the udp 10k-20k |
| 22:25.55 | ghost75 | reminds me from our telekom here |
| 22:26.15 | *** join/#asterisk Galen (~Galen@rrcs-24-43-17-235.west.biz.rr.com) |
| 22:26.21 | ghost75 | never ever say you use other devices than they sell |
| 22:27.43 | slav3_kitten | ghost75, i pay them 120 dollars a year to get a public IP |
| 22:28.01 | ghost75 | kind of public :> |
| 22:28.08 | slav3_kitten | sort of public |
| 22:28.18 | slav3_kitten | not really public just one less layer of nat |
| 22:28.45 | ghost75 | do use it for mailserver too? |
| 22:29.46 | slav3_kitten | nope |
| 22:30.02 | slav3_kitten | asterisk is the only thing i host |
| 22:32.00 | ghost75 | once i told them my line is working unstable and they told me i have unsupported equipment lol |
| 22:32.36 | slav3_kitten | core set verbose 3 |
| 22:32.43 | slav3_kitten | wrong window :\ |
| 22:33.16 | ghost75 | there is also telekom in us ? |
| 22:34.15 | slav3_kitten | huh? |
| 22:34.36 | ghost75 | or is it called t-mobile? |
| 22:38.06 | slav3_kitten | we have t-mobile |
| 22:38.51 | ghost75 | avoid at all cost |
| 22:40.32 | slav3_kitten | oh? |
| 22:45.40 | *** join/#asterisk Kako (~Kako@ip-95-223-21-26.unitymediagroup.de) |
| 22:46.49 | Kako | hi, after removing my debian package, compiling from source and restarting I'm getting lots of "xmldoc.c: Couldn't find application ??? in XML documentation". how can I fix it? |
| 22:48.24 | mjordan | you most likely have applications in modules that are registering themselves to Asterisk that are not part of the documentation that was pulled from the source you compiled from |
| 22:48.38 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 22:49.02 | mjordan | this can be for a number of reasons, one of which is that the modules installed by your debian package are still present in your modules directory but were not in the one you installed from source |
| 22:49.28 | mjordan | you should make sure you have completely uninstalled the debian package and all of the things that it installed before you install the one from source. |
| 22:50.43 | Kako | well, im very shure that apllications like Dial, Version an Echo should be compiled and installed with ther source compilation. I guess, that a path on y configuration file in /etc/asterisk points to an old destination.... |
| 22:51.07 | Kako | because Im not using a new configuratino created from samples, but the old one. |
| 22:51.52 | Kako | all other parts should be removed. |
| 23:00.29 | Kako | Option: I dont need the HTML/XML Documentation on the server. Is there a way to "not create" it, so that asterisk doesn't try to find it? |
| 23:01.12 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:03.51 | Kako | for what are thex necessary? |
| 23:11.04 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 23:25.45 | *** join/#asterisk jsjc (~Adium@54.Red-83-35-54.dynamicIP.rima-tde.net) |
| 23:32.54 | paulc | join #a2billing |
| 23:33.08 | paulc | like that, but with a slash, duh |
| 23:34.30 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-yzumopwuitaefykc) |
| 23:42.22 | *** join/#asterisk jsjc (~Adium@54.Red-83-35-54.dynamicIP.rima-tde.net) |
| 23:42.28 | ghost75 | sends bill to paulc |
| 23:46.52 | *** join/#asterisk jsjc (~Adium@54.Red-83-35-54.dynamicIP.rima-tde.net) |
| 23:50.03 | *** join/#asterisk jsjc (~Adium@54.Red-83-35-54.dynamicIP.rima-tde.net) |