| 00:00.04 | *** join/#asterisk evilrabbi (~evilrabbi@ip68-229-222-32.ok.ok.cox.net) |
| 00:00.08 | Qwell | but the fact that he isn't giving you a tracking number is suspicious |
| 00:00.13 | evilrabbi | ! |
| 00:00.15 | Qwell | I got one within like 2 days |
| 00:00.23 | netsurfer | exactly |
| 00:00.48 | netsurfer | no tracking number, no answer to the business phone number, no reply to 99% of my emails |
| 00:01.26 | netsurfer | oh.. and his domain names are registered to him at two different addresses |
| 00:01.31 | dsmouse | what did you buy? |
| 00:01.41 | netsurfer | X100P card from a dodgy ebayer |
| 00:01.50 | dsmouse | :( |
| 00:01.51 | evilrabbi | =\ |
| 00:01.53 | evilrabbi | that sucks man |
| 00:01.58 | evilrabbi | i've been burned on ebay before |
| 00:01.59 | evilrabbi | =( |
| 00:02.04 | dsmouse | I just got 2 of them I think |
| 00:02.09 | Qwell | it still strikes me as weird, because I went through that guy, and I think my boss did too |
| 00:02.40 | netsurfer | yeah Qwell me too, he has lots of positive feedback, but its all for small stuff.. very low value stuff |
| 00:02.50 | Luke-Jr | too bad there's no commonplace usage of PGP for auctions and such |
| 00:03.01 | Luke-Jr | a web of trust would really help with it, I think |
| 00:03.02 | Qwell | not sure why he would rip people off for < $20 |
| 00:03.07 | Qwell | Thats just silly |
| 00:03.15 | netsurfer | yep |
| 00:03.23 | netsurfer | lets see what paypal say |
| 00:03.38 | dsmouse | sometimes a seller just screws up too |
| 00:03.45 | netsurfer | his registered UK address is a very upmarket place in London.. Harley street no less |
| 00:03.57 | Qwell | UK address? |
| 00:04.07 | Luke-Jr | netsurfer: I wouldn't trust paypal to care... paypal sucks |
| 00:04.12 | netsurfer | yes, the auction said he was uk based |
| 00:04.15 | Qwell | he should be shipping from Walnut Creek, CA |
| 00:04.23 | netsurfer | Luke-Jr - yep of course they do, they are owned by ebay |
| 00:04.26 | Luke-Jr | any company demanding bank account #s is untrustworthy, IMO |
| 00:04.28 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 00:04.38 | netsurfer | Qwell - the auction said shipped from UK |
| 00:04.52 | Qwell | You said it was "neks" or whatever, right? |
| 00:04.58 | netsurfer | nkans |
| 00:05.12 | Qwell | yeah, mine shipped from right around the corner |
| 00:05.36 | netsurfer | something very dodgy.. I dont think his UK ebay account is legitimate |
| 00:06.16 | sudoer | does anyone know what I mean? |
| 00:06.16 | netsurfer | I did see he was selling stuff in the USA in his "other auctions" |
| 00:06.23 | sudoer | make * call you back |
| 00:07.04 | *** join/#asterisk Druken (~druken@CPE00119539b9cc-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
| 00:07.56 | netsurfer | Qwell - All units will be shipped via Royalmail Special Delivery within United Kingdom |
| 00:07.59 | *** join/#asterisk eye69 (magnus@ipv6.upcore.net) |
| 00:08.04 | Qwell | odd |
| 00:08.16 | *** join/#asterisk goatmilk (~goatmilk@130-127-45-50.chouse.resnet.clemson.edu) |
| 00:08.47 | Mother_ | this is really frustrating |
| 00:08.53 | Qwell | Item location: Walnut Creek, California |
| 00:09.00 | Qwell | netsurfer: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=5748126825 |
| 00:09.11 | netsurfer | Item location: London |
| 00:09.11 | netsurfer | United Kingdom |
| 00:10.00 | netsurfer | yep.. same seller different country |
| 00:10.28 | Druken | i've purchased from nkans before |
| 00:10.41 | Druken | twice actually |
| 00:10.45 | netsurfer | I certainly wont be purchasing from him again |
| 00:10.53 | Druken | oh? |
| 00:11.29 | netsurfer | had 1 email from him on tuesday, no delivery of any description. bought on 28th Jan |
| 00:11.49 | netsurfer | asked for tracking numbers, he wont send them |
| 00:12.16 | Druken | how did you pay ? |
| 00:12.29 | netsurfer | paypal |
| 00:12.53 | Druken | has it cleared? paypal is slower then fuck for clearing shit |
| 00:13.10 | netsurfer | yes he said the 1st one got lost in the post so he would send another |
| 00:13.16 | netsurfer | which is BS |
| 00:13.40 | Druken | well, i can say i have purchased twice from him, and gotten goods both times |
| 00:13.43 | Druken | no problems |
| 00:13.55 | Druken | i guess i'm just more patient |
| 00:14.12 | Qwell | I bought mine on Jan 31, had it by the 3rd ;/ |
| 00:14.29 | netsurfer | theres a difference in being patient and sitting back while someone rips you off.. he wont supply tracking numbers and wont answer the phone or reply to emails |
| 00:15.19 | *** join/#asterisk PTG123 (~PTG123@ip68-106-19-249.ph.ph.cox.net) |
| 00:15.50 | Druken | Qwell: shit, you must be in his country.. hehehe took like 2 weeks to get here :) |
| 00:16.02 | Qwell | Druken: same state even |
| 00:16.07 | Druken | that works |
| 00:16.13 | netsurfer | Druken - where are u located ? |
| 00:16.53 | Druken | ontario canada |
| 00:17.51 | Mother_ | what does the INVAL flag mean in the IAX2 debug? |
| 00:18.04 | Mother_ | I get that when the dial command executes |
| 00:18.10 | evilrabbi | Invalid ? |
| 00:18.22 | mrproper_ | anyone know why id be getting missing seperator errors when compiling pwlib for openh323 |
| 00:19.09 | Mother_ | nm found it |
| 00:19.31 | Mother_ | yeah, but invalid *what*? turns out apparently it's an invalid destination |
| 00:20.36 | evilrabbi | maybe it's tring to contact an IAX host that isn't there? |
| 00:20.58 | Mother_ | no, the IAX comms are fine, the hosts see each other, all is dandy |
| 00:21.03 | evilrabbi | ah |
| 00:21.04 | evilrabbi | ok |
| 00:21.09 | Mother_ | it's when the time comes to execute the Dial command |
| 00:21.55 | Mother_ | the setup is PSTN => * #1 <=> IAX2 <=> * #2 => SIP Phones |
| 00:32.06 | *** join/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 00:33.07 | Mother_ | Darwin35: I've tried removing the IAX from the end, I've tried with extension numbers, with s, I just cannot get it to accept a call |
| 00:35.08 | *** join/#asterisk rfc1918 (~Gentoo@pool-151-205-89-223.char.east.verizon.net) |
| 00:35.26 | Mother_ | haH |
| 00:35.55 | Mother_ | changing the context name seems to have cured it, but now the local * segfaults |
| 00:36.07 | Mother_ | and complains that it cannot trunk once a call is started... |
| 00:40.11 | dsmouse | ok. got the zap setup |
| 00:40.15 | dsmouse | now for sip :/ |
| 00:46.35 | *** join/#asterisk IsMe (~some@218.111.194.207) |
| 00:47.10 | *** join/#asterisk ManxPwr (~eric@dsl-209-205-172-111.i-55.com) |
| 00:47.38 | IsMe | ~seen ZX81 |
| 00:47.39 | jbot | zx81 <~ZX81@222-153-18-2.jetstream.xtra.co.nz> was last seen on IRC in channel #asterisk, 10d 12h 26s ago, saying: ':)'. |
| 00:49.13 | *** join/#asterisk zd_eyez (~zd_eyez@Ottawa-HSE-ppp257967.sympatico.ca) |
| 00:49.25 | Mother_ | it verks! |
| 00:50.05 | Mother_ | now....why on earth does zap channel get an event 17 (polarity reversal), but it doesn't interpret it as a hangup as it should? |
| 00:51.15 | *** part/#asterisk kietlak (~kietlak@11-mo3-6.acn.waw.pl) |
| 00:53.00 | Mother_ | any takers? |
| 00:54.34 | *** join/#asterisk cc (~cc@byte.fedora) |
| 00:55.06 | *** join/#asterisk Mother__ (~m@53.Red-217-126-93.pooles.rima-tde.net) |
| 00:55.21 | Mother__ | anyone know about this polarity reversal message then? |
| 00:55.35 | Mother__ | more in particular why doesn't it make simple switch end |
| 00:58.33 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 01:04.42 | Mother__ | does faxdetect work without spandsp? |
| 01:07.21 | tzanger | Mother__: yes |
| 01:07.50 | Mother__ | thanks, I'm going to try it... |
| 01:08.14 | Mother__ | this setup must be handled with tweezers |
| 01:09.54 | *** join/#asterisk yxa (~void@203.118.40.42) |
| 01:10.35 | *** join/#asterisk locksy (~locksy@mrtg.sisgroup.com.au) |
| 01:11.03 | *** join/#asterisk jterrero (~some@ool-43576e0d.dyn.optonline.net) |
| 01:11.05 | *** join/#asterisk lyroy (~lyroy@modemcable117.123-202-24.mc.videotron.ca) |
| 01:12.09 | lyroy | Is it possible to have asterisk messages in french or I have to record all the messages by myself? |
| 01:12.39 | Mavvie | iconv isn't really implemetned there. |
| 01:13.40 | Mother__ | w000000t! |
| 01:13.46 | Mother__ | it verks even better now |
| 01:13.52 | Mother__ | thanks tzanger :) |
| 01:26.13 | Darwin35 | pay to have them recorded in french |
| 01:29.01 | *** part/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 01:32.15 | lyroy | where Darwin? |
| 01:32.27 | Mother__ | grrrrr |
| 01:32.45 | *** join/#asterisk Tough_Nuts (~Tough_Nut@m210e36d0.tmodns.net) |
| 01:33.00 | *** join/#asterisk BBRodriguez (~alex@pD9EA7BDE.dip.t-dialin.net) |
| 01:33.00 | Mother__ | now I have a new problem: when the SIP phone picks up, it can talk etc. but when it's hangup, it just starts ringing again |
| 01:33.47 | Mother__ | it's as if the remote * is not seeing the hangup and starts the IAX2 call over again |
| 01:34.15 | *** join/#asterisk Othello (Othello@nusnet-154-210.dynip.nus.edu.sg) |
| 01:35.39 | *** join/#asterisk PMantis (~PrayingMa@65-37-0-159.nrp2.roc.ny.frontiernet.net) |
| 01:35.42 | loud | Mother, you might have a voicemail. |
| 01:36.12 | Mother__ | loud: nope, voicemail doesn't trigger at all |
| 01:36.43 | loud | do you get just one ring back ? |
| 01:36.55 | Mother__ | no, it carries on ringing |
| 01:37.02 | loud | ah |
| 01:37.08 | Mother__ | also, the remote * starts the call all over again |
| 01:37.37 | Mavvie | check the dialplan |
| 01:38.42 | *** join/#asterisk SeaForth (~SeaForth@c-24-1-126-202.client.comcast.net) |
| 01:39.05 | Mother__ | what would that have to do with the remote * starting the call over again? it's as if the SIP phone that picks up hangs the call, but nobody notices |
| 01:39.20 | Mother__ | I'll double-check anyway just in case |
| 01:39.54 | PMantis | Anyone know how high "priority" is allowed to be in extensions.conf? |
| 01:45.55 | CoaxD | you know, it is very annoying to be coming to work at 7:45pm after working all day at a different location |
| 01:46.18 | CoaxD | makes me think i'm just slave labor. But then, i think about it, and realize that i own both biz's, so...what is it, then, i'm hiring MYSELF for slave labor? .... |
| 01:46.57 | CoaxD | on a sunday no less.. |
| 01:47.06 | PMantis | LOL |
| 01:47.59 | *** join/#asterisk NormAst (HydraIRC@Ottawa-HSE-ppp4117547.sympatico.ca) |
| 01:48.23 | Mother__ | this is uberweird |
| 01:49.14 | Mother__ | when I hangup the SIP phone, which was coming from Zap/2 on the remote *, a new call on Zap/1 is spawn and sent, which causes the SIP phones to ring again |
| 01:49.18 | netsurfer | PMantis - dunno, but we have some at 201 here |
| 01:49.30 | Mother__ | and on the phone that was making the call I get the ringing tone again |
| 01:50.47 | Mother__ | if the phone that made the PSTN call hangs up first, then the process repeats, but when the SIP phone answers the call it gets the hangup tones |
| 01:50.52 | Mother__ | pffff |
| 01:51.28 | PMantis | I have a weird callerid issue. |
| 01:51.47 | PMantis | I was running CVSHEAD from December, then Jan... all was ok with callerid |
| 01:52.01 | CoaxD | pmantis: it might not be asterisk. telcos change things all the time. remember that. |
| 01:52.08 | PMantis | i went back to 1.0.5 for stability with SIP |
| 01:52.21 | *** join/#asterisk ZX81_laptop (~chatzilla@222-153-20-92.jetstream.xtra.co.nz) |
| 01:52.23 | PMantis | CoaxD, IAX2 and SIP terminations both |
| 01:52.52 | PMantis | Now, I get callerid with the number populated in the name and number field. |
| 01:53.17 | ZX81_laptop | so, anyone know if it is possible to do a callpickup (i.e. *8) on a ringing call via the manager interface? |
| 01:53.26 | ZX81_laptop | oh and hi |
| 01:53.28 | ZX81_laptop | :) |
| 01:54.02 | PMantis | I tried to set calleridname to blank if calleridname and calleridnum are the same... still is set. So, I tried SetCallerID, to set it only to a number. Still name and number both have the phone number in it. |
| 01:54.05 | mikegrb | ZX81_laptop: you could possible redirect it to another channel |
| 01:54.33 | ZX81_laptop | don't think you can redirect unless it is an existing call |
| 01:54.41 | ZX81_laptop | there were some posts about it |
| 01:54.47 | ZX81_laptop | guess I should try first though |
| 01:54.49 | ZX81_laptop | :) |
| 01:55.25 | PMantis | My questions boils down to: How do I blank the calleridname variable? |
| 01:55.45 | *** join/#asterisk JunK-Y (~junky@modemcable056.110-81-70.mc.videotron.ca) |
| 01:55.58 | PMantis | SetVar(CALLERIDNUM=) doesn't work |
| 01:56.01 | ZX81_laptop | setcid |
| 01:56.05 | ZX81_laptop | or something |
| 01:56.10 | JunK-Y | SetCidNum |
| 01:56.12 | ZX81_laptop | show application setc<TAB> |
| 01:56.17 | ZX81_laptop | name he wants |
| 01:56.25 | PMantis | SetCallerID(${CALLERIDNUM}) also doesn't work. :( |
| 01:56.26 | JunK-Y | SetCidName :) |
| 01:56.37 | PMantis | oh?? |
| 01:57.07 | ZX81_laptop | :) |
| 01:57.22 | bjohnson | Mother__: I had a ring back problem .. sound kind of, but not quite same as you .. was realted to call waiting on the SIP device .. turned it off and seems like problem went away |
| 01:58.23 | PMantis | arrrgh, still don't work :( |
| 01:59.04 | PMantis | my debug lines still print a full calleridname field: NoOp( - - - CallerID: ${CALLERIDNUM}/${CALLERIDNAME}) |
| 01:59.16 | PMantis | Yes, I ran reload. :) |
| 02:00.44 | JunK-Y | PMantis: whatcha trying to do exactly ? change the calleridnum or name? |
| 02:02.30 | PMantis | JunK-Y, My callerid info seems to be suddenly coming in with the number also in the name field. |
| 02:02.44 | PMantis | JunK-Y, i simply want to blank the CALLERIDNAME variable |
| 02:03.13 | niZon | PMantis: did you set it right in the client config? |
| 02:03.41 | JunK-Y | whats ur CLI's output exactly? |
| 02:03.48 | PMantis | JunK-Y, This callerid change seems to be related to the time when I wewnt from CVS to ver 1.0.5. |
| 02:04.00 | PMantis | JunK-Y, Mind if i pm you? |
| 02:04.43 | JunK-Y | no, but i dont have so much time. |
| 02:05.27 | *** part/#asterisk zd_eyez (~zd_eyez@Ottawa-HSE-ppp257967.sympatico.ca) |
| 02:05.50 | loud | mother |
| 02:06.43 | *** join/#asterisk MrEntropy (~entropy@ppp55-252.lns1.adl2.internode.on.net) |
| 02:06.44 | tzanger | a loud mother? |
| 02:06.45 | tzanger | hahaha |
| 02:06.55 | flewid | sup |
| 02:07.25 | flewid | has anyone ever seen an aastra phone just stop properly sending dtmf digits on the phone when it's plugged in |
| 02:07.32 | tzanger | no |
| 02:07.33 | tzanger | never |
| 02:07.42 | tzanger | I've had a PT390 for 5 or 6 years now |
| 02:07.52 | tzanger | I think it's been that long anyway |
| 02:07.57 | flewid | like, if i unplug the phone it works and i can dial, but if it's plugged in, it dials (i can hear the tones), but nothing gets accepted by the asterisk server |
| 02:08.03 | flewid | and asterisk says it doesnt' receive any digits |
| 02:08.10 | flewid | i tried resetting the phone to fact default, no avail |
| 02:08.46 | flewid | just started today |
| 02:14.07 | *** join/#asterisk file (~symlink@mctn1-1056.nb.aliant.net) |
| 02:14.30 | *** part/#asterisk sudoer (~sudoer@65.75.148.190) |
| 02:16.35 | bjohnson | any of those aastra phones actually voip? or are they all just fancy analog |
| 02:18.46 | *** join/#asterisk Guest^DJ (~some@218.111.194.207) |
| 02:19.02 | Guest^DJ | hi guys |
| 02:22.49 | *** join/#asterisk sudoer (~sudoer@65.75.148.190) |
| 02:23.31 | sudoer | can someone help me with runnign a system command? |
| 02:23.37 | sudoer | I am trying to run: ,System(sleep 3 && echo -e "Channel: IAX2/160@voipjet/{${CALLERIDNUM}\nApplication |
| 02:24.08 | sudoer | but the output I get is: Channel: IAX2/160@voipjet/{123123131nApplication |
| 02:24.51 | sudoer | the extra "{" and "n" are becing echoed |
| 02:25.55 | Mavvie | you want printf instead of echo |
| 02:25.58 | heath__ | i'd try taking them both out, and then just run another System(echo) |
| 02:26.31 | heath__ | or try NoOp witch is like Mavvie is saying |
| 02:26.31 | *** join/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
| 02:26.48 | sudoer | ok, ill try printf, thanks |
| 02:29.25 | sudoer | printf creates the same error |
| 02:29.50 | sudoer | I made a mistake about the }, but the \n is appearing as n instead of newlien though |
| 02:30.23 | Mavvie | what is the line now? |
| 02:31.18 | sudoer | ok, I thought it was my fault but the { is still appearing, weird: Channel: IAX2/160@voipjet/{16179099947nApplication: DisanData |
| 02:31.50 | heath__ | then just take \n out, then just add another System like this System(echo) |
| 02:32.22 | sudoer | but I am putting multiple lines into 1 file |
| 02:32.58 | Mavvie | sudoer: stop and think for a moment: |
| 02:32.59 | sudoer | in extensions.conf: exten => _X.,1,System(sleep 3 && printf "Channel: IAX2/160@voipjet/${EXTEN}\nApplication: Disa\nData: no-password,context" > /var/spool/asterisk/call &) |
| 02:33.12 | Mavvie | sudoer: n means n |
| 02:33.34 | Mavvie | sudoer: \n means: escaped n, i.e. take n litterally[sp] |
| 02:33.47 | sudoer | as in newline |
| 02:33.54 | Mavvie | sudoer: \\n means: escaped \, n, i.e. take the \ litterally[sp] |
| 02:34.16 | heath__ | oh yeah, Mavvie's right |
| 02:34.25 | Mavvie | now in printf... |
| 02:34.29 | Mavvie | n means n |
| 02:34.33 | Mavvie | \n means newline |
| 02:34.49 | Mavvie | just make sure you feed printf the \. |
| 02:34.51 | sudoer | ok, but I want a newline |
| 02:34.54 | Mavvie | just make sure you feed printf the \ and not the n |
| 02:35.02 | sudoer | I am, my example is right aboce |
| 02:35.13 | Mavvie | sudoer: think about what I just said. |
| 02:35.49 | sudoer | ok, I think I got ti |
| 02:36.51 | *** join/#asterisk jdims (~sleepbsd@24-119-121-6.cpe.cableone.net) |
| 02:36.55 | Mavvie | pbx_extension_helper: No application 'Exit' for extension (macro-call-nominum-do, s-VM, 2) |
| 02:36.55 | Mavvie | == Spawn extension (macro-call-do, s-VM, 2) exited non-zero on 'SIP/edwin-2166' in macro 'call-do' |
| 02:37.04 | Mavvie | heh... that's funny |
| 02:38.23 | Mavvie | now how do I exit a macro properly? |
| 02:38.52 | sudoer | Mavvie: I thought I need to pass \\n, but that is wrong, what am I doing wrong? |
| 02:39.05 | Mavvie | sudoer: I don't know, what does it say? |
| 02:39.33 | sudoer | same exact thing |
| 02:39.42 | sudoer | atextntext |
| 02:39.44 | Mavvie | can you paste it somewhere? |
| 02:39.57 | sudoer | ok |
| 02:40.57 | Darwin35 | use pastebin.ca |
| 02:41.04 | Darwin35 | dont paste to the channel |
| 02:41.14 | Darwin35 | ~pastebin |
| 02:41.15 | jbot | [pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 02:41.30 | sudoer | Mavvie: http://pastebin.ca/5782 |
| 02:43.16 | Mavvie | try four \'s |
| 02:43.21 | *** join/#asterisk bjohnson_ (~bjohnson@ip226-181.tor.istop.com) |
| 02:43.33 | srt | why don't you write a short agi for that? |
| 02:43.36 | Mavvie | first it has to be parsed by asterisk. |
| 02:43.41 | Mavvie | the it has to be parsed by the shell |
| 02:43.47 | Mavvie | then it has to be parsed by printf |
| 02:43.51 | heath__ | it should be three shouldn't it? |
| 02:44.00 | sudoer | oh |
| 02:44.22 | sudoer | how about the ${EXTEN} problem? |
| 02:44.40 | bjohnson_ | Can someone call my FWD number to test incoming? I can call out fine |
| 02:45.53 | *** join/#asterisk JustinT (toasted@adsl-065-007-243-200.sip.clt.bellsouth.net) |
| 02:46.01 | sudoer | 3 doesnt work either, it keeps doing the same thing |
| 02:46.09 | Mavvie | not 3 |
| 02:46.38 | srt | bjohnson_: you can originate a callback on the fwd site |
| 02:46.39 | Mavvie | sudoer: please explain to me what 3 \'s would be |
| 02:46.43 | Mavvie | s/be/do |
| 02:47.04 | sudoer | I was trying heath__ suggestion |
| 02:47.20 | *** part/#asterisk JustinT (toasted@adsl-065-007-243-200.sip.clt.bellsouth.net) |
| 02:47.29 | Mavvie | I don't care, you should only do things which are explained and which you understand. |
| 02:47.39 | Mavvie | Now tell me, what would "\\\n" do ? |
| 02:48.30 | sudoer | let * parse, then shell, the nprintf' |
| 02:49.22 | Mavvie | I don't see any reference to * in the question, nor to a shell or printf. |
| 02:49.32 | Mavvie | that leaves the question: what would "\\\n" do ? |
| 02:50.16 | bjohnson_ | srt: didn't see that |
| 02:50.24 | sudoer | I dont know then |
| 02:50.27 | srt | its kinda hidden ;) |
| 02:50.35 | srt | but if you like i can call ya |
| 02:50.44 | Mavvie | sudoer: look above. n means n, \n means an escaped n. Now you continue. |
| 02:51.21 | sudoer | ok, so I want to use escaped n |
| 02:51.36 | Mavvie | a double escaped to be exactly. |
| 02:51.43 | sudoer | \\n |
| 02:51.45 | Mavvie | a double escaped \n to be exactly |
| 02:52.09 | sudoer | does "\\n" mean double escaped n? |
| 02:52.10 | chipig | a |
| 02:52.27 | Mavvie | no, it's an escaped \ with an n |
| 02:53.02 | sudoer | \n\n ? |
| 02:53.10 | bjohnson_ | srt: jdims tried .. no luck. Must be NAT related. Gonna try tomorrow behind a different NAT router |
| 02:53.24 | Mavvie | Ha, you have shown that my question was wrong. Well done. |
| 02:53.37 | Mavvie | An escaped-escaped \n is what you need. |
| 02:54.00 | sudoer | \\\n? |
| 02:54.01 | srt | bjohnson_: k - if you wanna try it on your own: http://account.freeworlddialup.com/index_new.php?section_id=76 |
| 02:54.09 | Mavvie | sudoer: what is an "\\\n" ? |
| 02:54.19 | sudoer | a escaped escaped \n |
| 02:54.29 | jdims | what I don't understand bjohnson_ is why you can call out but not receive with NAT |
| 02:54.31 | srt | bjohnson_: or login at fwd website / support/faq / configuration guide / step 4 |
| 02:54.33 | Mavvie | no, it's an escaped \ - escaped n |
| 02:55.00 | sudoer | Mavvie: I am not sure what is correct to use then |
| 02:55.05 | Mavvie | sudoer: \\\\n |
| 02:55.13 | srt | bjohnson_: do you have a register line in sip.conf ? |
| 02:55.27 | Mavvie | and if that doesn't work, double it again. |
| 02:55.48 | Mavvie | sooner or later you will run out of interpreters and have your \n in printf. |
| 02:55.54 | heath__ | sudoer: when you used \\\n where you using echo or printf? |
| 02:55.58 | sudoer | printf |
| 02:55.59 | heath__ | were*** |
| 02:56.26 | sudoer | Mavvie: heath__ it works now, thanks |
| 02:57.42 | sudoer | I would have figured thisd out earlier, but it seems liek reload doesnt work |
| 02:57.51 | *** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-100.modem.logical.net) |
| 02:58.18 | sudoer | i was just doing reload earlier, and just until 2,3 times ago did i shutdown adn start asterisk, did I get different output |
| 03:00.45 | bjohnson_ | srt: this is just a SPA 2000 .. trying to connect in and out through it |
| 03:01.21 | bjohnson_ | I have * running here but am trying to setup 2000 for other remote users .. don't want additional config for them to do |
| 03:01.45 | bjohnson_ | I can dial to FWD and route back to my office * .. but not the other way |
| 03:01.47 | *** join/#asterisk Rick_Hunter (~rhunter@03-025.008.popsite.net) |
| 03:03.08 | *** part/#asterisk sudoer (~sudoer@65.75.148.190) |
| 03:03.47 | bjohnson_ | the SPA web interface even says it is ringing .. but no ring on the phones |
| 03:04.42 | bjohnson_ | phone was on mute |
| 03:06.37 | *** join/#asterisk Varanger (~varanger@201.240.144.132) |
| 03:08.27 | Mavvie | aaaaaaaargh. |
| 03:08.31 | Mavvie | the CLI should have a hostname in ir. |
| 03:08.33 | Mavvie | in it. |
| 03:08.42 | Mavvie | this is the third time that I stopped the wrong server. |
| 03:08.53 | Varanger | hi all! |
| 03:08.59 | Varanger | Asterisk rulez! |
| 03:09.09 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 03:12.35 | dsmouse | YAY! |
| 03:12.37 | *** join/#asterisk SeaForth (~SeaForth@c-24-1-126-202.client.comcast.net) |
| 03:12.39 | Varanger | hello all |
| 03:13.09 | Varanger | I want to ask something about Asterisk |
| 03:13.22 | dsmouse | that's what this channel is for :) |
| 03:13.47 | dsmouse | and I'll even answer if it's within my 6 hours of experiance |
| 03:13.57 | Varanger | I am home user, and I`d like to connect my telephone line to Asterisk |
| 03:14.03 | Mavvie | ~hardware |
| 03:14.06 | jbot | hardware is probably nice, however i can't get any of the code from the company |
| 03:14.06 | Varanger | what hardware do I need? |
| 03:14.17 | Mavvie | oh. jbot sucks sometimes :-) |
| 03:14.29 | dsmouse | Varanger: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=61841&item=5751502883&rd=1 |
| 03:14.34 | dsmouse | Varanger: that's the cheap way |
| 03:15.31 | dsmouse | also look at http://www.digium.com/index.php?menu=hardware_products |
| 03:15.39 | Varanger | dsmouse: I ll buy there! |
| 03:15.50 | dsmouse | and http://www.digium.com/index.php?menu=grayfaq |
| 03:15.51 | Mavvie | jbot: hardware is http://www.digium.com/index.php?menu=hardware_products. If you don't know what you need, start with an TDM400P and an FXS module. |
| 03:15.53 | jbot | ...but hardware is already something else... |
| 03:15.54 | JerJer | dsmouse: if you want crap, go for it |
| 03:15.58 | Mavvie | jbot: forget hardware |
| 03:15.59 | jbot | Mavvie: i forgot hardware |
| 03:16.00 | Mavvie | jbot: hardware is http://www.digium.com/index.php?menu=hardware_products. If you don't know what you need, start with an TDM400P and an FXS module. |
| 03:16.01 | jbot | okay, Mavvie |
| 03:16.14 | Varanger | dsmouse: I need a FXO card? |
| 03:16.28 | dsmouse | Varanger: FTR, that's just the first ebay hit I got, I'm not selling anything :) |
| 03:16.33 | Varanger | I have a normal telephone line, I dont have a ISDN one |
| 03:16.54 | dsmouse | http://www.digium.com/index.php?menu=fxsvfxo <- |
| 03:18.15 | dsmouse | jbot: fxo? |
| 03:18.16 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 03:20.29 | netsurfer | Varanger - just a warning, iv had bad experience with that ebayer.. still not received goods |
| 03:21.44 | bjohnson_ | while a digium one port fxp plus one port fxs is good .. I prefer a Sipura SPA 3000 for a one line analogue system |
| 03:21.54 | bjohnson_ | mainly due to the failover |
| 03:22.10 | PTG123 | Varanger: why connect your telephone line, and not use a voip provider? |
| 03:22.54 | Sedorox | I got a card from that person (knans, or whatever) recently.. shipped to canada... works nice |
| 03:24.46 | bjohnson_ | one fxo is only part of a system |
| 03:28.12 | *** join/#asterisk |nix (~inix@202.148.164.9) |
| 03:32.53 | MrEntropy | ~seen implicit |
| 03:33.04 | jbot | implicit <~bayan@209.80.0.43> was last seen on IRC in channel #asterisk, 2d 4h 28m 19s ago, saying: 'whats up'. |
| 03:35.16 | bjohnson_ | is there a way for * to tell where a north american toll free number will work (ie US or CAN)? |
| 03:36.03 | dsmouse | how much bandwidth does voip take, ballpark? |
| 03:36.32 | Carp1 | ~seen carp |
| 03:36.33 | jbot | carp <Carp@ip-204-97-151-148.modem.logical.net> was last seen on IRC in channel #asterisk, 103d 1h 46m 20s ago, saying: 'Sorry, Forgot about the location of everyone.'. |
| 03:36.37 | Sedorox | 24K/s per phone total in and out |
| 03:36.38 | Mavvie | how big is a road? |
| 03:36.40 | Sedorox | average |
| 03:37.08 | stepcut | hrm, I can place calls through sipphone, but with fwd I don't hear anything after I get connected :( |
| 03:41.14 | bjohnson_ | Sedorox: what codec? |
| 03:41.31 | bjohnson_ | dsmouse: depends on codec .. ulaw ismost at 80 kb I think |
| 03:41.44 | Sedorox | uhhhhh... think most of our stuff is ulaw... |
| 03:42.41 | *** join/#asterisk menger (~menger@static-88.243.240.220.dsl.comindico.com.au) |
| 03:42.45 | yashax | just testing :) kto nibyt govorit po russki? |
| 03:42.51 | bjohnson_ | might be 80 kbits each way = 24 kBytes |
| 03:43.05 | JunK-Y | ulaw is 64kb |
| 03:43.20 | Mavvie | JunK-Y: but you have IP overheard. |
| 03:43.32 | JunK-Y | 64kb is w/out overhead. |
| 03:43.46 | JunK-Y | w/ overhead i dont know, it goes to 80kb 6 |
| 03:43.48 | JunK-Y | ? |
| 03:44.00 | Mavvie | yes |
| 03:44.06 | bjohnson_ | dsmouse: other codecs use less .. some have ok qual;ity |
| 03:44.06 | JunK-Y | im on a gigabit network, ive no trouble with that atm. :) |
| 03:44.38 | tzanger | why does everyone use ulaw for VOIP? makes no sense whatsoever to me |
| 03:44.57 | tzanger | that's like saying "well I've got a fast internet connection, I may as well throw around uncompressed .wavs instead of .mp3s |
| 03:45.00 | Mavvie | tzanger: best quality. no loss. |
| 03:45.02 | bjohnson_ | tzanger: what do you use? |
| 03:45.11 | JunK-Y | tzafrir: cause it has no quality loss. |
| 03:45.12 | bjohnson_ | different internal than external? |
| 03:45.20 | tzanger | Mavvie: it's VOIP... who cares if it sounds like a good digital cell? |
| 03:45.21 | tzanger | gsm |
| 03:45.33 | Sedorox | tzanger: that seemed to be the only one we could get working well... |
| 03:45.35 | niZon | tzanger: CPU limitations could be a factor.. |
| 03:45.37 | Mavvie | tzanger: also, no CPU necessary for conversions to the PSTN. |
| 03:45.38 | *** join/#asterisk TheEmperor (~mattn@203.121.47.100) |
| 03:45.52 | tzanger | niZon: if you can't transcode a few channels of GSM you've got serious issues... My P90 (no MMX) could do it |
| 03:46.00 | tzanger | Mavvie: untrue |
| 03:46.04 | tzanger | zap uses slinear |
| 03:46.13 | niZon | hm ok :P |
| 03:46.18 | Mavvie | tzanger: You are currently labeled as -1, Troll :-) |
| 03:46.23 | tzanger | Mavvie: ha |
| 03:46.26 | Sedorox | anyone know a good ISDN Pri card for asterisk? |
| 03:46.33 | tzanger | Sedorox: T100P? |
| 03:48.14 | Mavvie | ${ARG1:-14} gives me only "enum.barne" instead of "enum.barnet.com" |
| 03:48.21 | Sedorox | hmmm |
| 03:49.16 | tzanger | tomorrow we go in to full-scale new jitter buffer testing |
| 03:49.32 | Sedorox | t100p is for isdn too? |
| 03:49.34 | tzanger | thurs/fri had some but we found incompatibilities with iax2 bridge optimization |
| 03:49.37 | dsmouse | in extentions, what's the syntax for say 81... |
| 03:49.45 | dsmouse | where ... is anyhing, anylength |
| 03:49.46 | tzanger | Sedorox: T100P is for T1. ISDN PRI rides on top of T1/E1 so yes |
| 03:49.56 | Sedorox | interesting |
| 03:49.58 | Sedorox | didn't know that |
| 03:50.02 | tzanger | Sedorox: yup |
| 03:50.08 | Sedorox | learn something new everday |
| 03:50.09 | Sedorox | ;-) |
| 03:50.13 | tzanger | physically T1 PRI and CAS T1 are no different |
| 03:50.20 | tzanger | it's all in what the bits mean |
| 03:50.34 | Sedorox | ah |
| 03:50.36 | Sedorox | thanks |
| 03:50.44 | tzanger | and even then, CAS T1 and ISDN PRI bits are pretty much 1-for-1 until you hit signaling |
| 03:51.03 | Sedorox | I have to read up on T1/ISDN stuff.. we're gonna go with one or the other... and you can use a T1 coming in for half data half voice.. or something like that.. right? |
| 03:51.12 | tzanger | yes |
| 03:51.23 | Sedorox | niceeee |
| 03:51.28 | tzanger | and T100P supports that too, although it's very unstable because of all the goddamned kernel changes in 2.6 |
| 03:51.44 | tzanger | ISDN PRI has so many nice features without fucking around |
| 03:52.04 | Sedorox | so the TE110P would be good for both data and voice? |
| 03:52.33 | tzanger | no timing oddities (just signaling oddities, <G>), a wealth of info about incoming and outgoing calls, good good stuff |
| 03:52.42 | tzanger | yes TE110P is fine, although I think it is 3.3V PCI only? |
| 03:52.54 | tzanger | TE410P is 3.3V only, TE405P is same thing but 5V only |
| 03:53.15 | tzanger | T100P is both but uses that fucking shitty TJ320 PCI chip |
| 03:53.32 | tzanger | it works but fuck I hate that chip |
| 03:53.58 | menger | tzanger, how come? |
| 03:54.17 | tzanger | menger: it's just the lowest bid, everything about it screams "I'm a ghetto PCI interface" |
| 03:54.27 | menger | ahh |
| 03:54.28 | Sedorox | lol.. so what is good for a start with good quality... for asterisk as a PBX.. and data for a office... |
| 03:54.36 | tzanger | the fact that Digium glues a heatsink to it also concerns me |
| 03:54.45 | Sedorox | the T100P or the TE110P? |
| 03:54.47 | tzanger | Sedorox: T100P will do your stuff |
| 03:54.57 | tzanger | I don't know much about TE110P |
| 03:55.17 | Sedorox | kk |
| 03:55.26 | Sedorox | will have to look int all this |
| 03:55.40 | tzanger | I have a pair of T100Ps and a pair of TE110Ps |
| 03:55.43 | tzanger | er TE405Ps sorry |
| 03:56.00 | Sedorox | hehe |
| 03:56.05 | Sedorox | don't need 4 T1's...... yet..... |
| 03:56.05 | Sedorox | lol |
| 03:56.40 | tzanger | :-) |
| 03:57.10 | Sedorox | actually.. just got * up 2 weeks ago.. leanred a lot in that time about PBX's.. voip... pots... all that... |
| 03:57.12 | tzanger | the TE110P uses the TJ320 too |
| 03:57.13 | Sedorox | no crying... |
| 03:57.15 | Sedorox | lol |
| 03:57.24 | tzanger | and from the spec sheet pic, it's a universal card too |
| 03:57.36 | Sedorox | hmmm |
| 03:57.39 | tzanger | looks like a T100P with a T1/E1 framer instead of T1 framer |
| 03:58.01 | Sedorox | yea |
| 03:58.12 | Sedorox | US/CAN doesn't have E1's do they? |
| 03:58.27 | tzanger | no |
| 03:58.31 | Sedorox | thought so |
| 03:58.44 | bjohnson_ | tzanger: you have any luck finding Stratford DIDs? |
| 03:58.51 | tzanger | no |
| 03:58.54 | Sedorox | so it isn't work the extra $100 for the TE110P |
| 03:58.58 | tzanger | haven't looked too hard either |
| 03:59.35 | *** join/#asterisk godsmoke (~godsmoke@66-108-159-216.nyc.rr.com) |
| 04:00.40 | bjohnson_ | were you just looking for a DID for your own use? |
| 04:00.48 | tzanger | bjohnson_: no |
| 04:00.59 | tzanger | at 1.25c/min or so I don't care about LD to stratford |
| 04:01.12 | tzanger | I want stratford DIDs because that means I can port my Listowel numbers over there |
| 04:01.25 | tzanger | since Listowel's CO terminates in the stratford switch |
| 04:01.43 | tzanger | and if someone can get PRIs in statford it means I can port listowel #s to them |
| 04:02.04 | *** part/#asterisk Varanger (~varanger@201.240.144.132) |
| 04:02.51 | bjohnson_ | so you don't actually need a PRI in Stratford .. just to know who will supply one? |
| 04:03.00 | tzanger | yes |
| 04:03.10 | *** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.we.client2.attbi.com) |
| 04:03.13 | tzanger | well techncially I'll need a stratford PRI |
| 04:03.21 | tzanger | to get Listowel numbers to appear on it with LNP |
| 04:03.29 | Wi_Fi | anyone know how to set time zone on CentOS system |
| 04:03.35 | tzanger | I can't get listowel #s on a Kitchener PRI without paying a lot of money |
| 04:03.42 | Wi_Fi | with NTP |
| 04:04.01 | sivana | :) |
| 04:04.24 | bjohnson_ | tzanger: so you need a host in Stratford to get the PRI and you use it? |
| 04:04.40 | tzanger | well this is all information fishing at this point |
| 04:04.51 | tzanger | I want to have 3 listowel #s appear on whatever PRI I get |
| 04:05.15 | tzanger | and it's important for 1 of those 3 to have full "roll down" on the entire PRI |
| 04:05.42 | bjohnson_ | I remembered that Wightman telecom is a new name around here but there are telecomunications and offer residential dsl .. you could talk to them since they likely have access to PRIs |
| 04:05.45 | tzanger | CFB apparently quits after 3 simultaneous calls unless you use transfer on Centrex |
| 04:06.01 | tzanger | bjohnson_: yeah |
| 04:06.07 | tzanger | Wightman's a little pissed at me |
| 04:06.10 | tzanger | I have VOIP working and they don't |
| 04:06.18 | tzanger | they rolled it out in Elmira and had to cancel it |
| 04:06.26 | Sedorox | tzanger: those cards.. they act like a csu/dsu I'm assuming, so you just plug from the wall into the card w/o any other HW? |
| 04:06.39 | tzanger | Sedorox: you will plug the telco smartjack into them |
| 04:07.07 | tzanger | technically I SUPPOSE they could drive thousands of feet of cable but I doubt it... |
| 04:07.08 | sivana | bjohnson_: where are you located? |
| 04:07.13 | tzanger | so the telco terminates to their smartjack |
| 04:07.17 | tzanger | and you connect to that |
| 04:07.23 | Sedorox | lol |
| 04:07.25 | tzanger | and yes, they're the DSU/CSU |
| 04:07.26 | Sedorox | well yea |
| 04:07.29 | Sedorox | kk |
| 04:07.33 | bjohnson_ | sivana: Stratford |
| 04:07.44 | Sedorox | I know a little bit of this stuff to be dangerous.. it sucks... |
| 04:07.45 | Sedorox | lol |
| 04:08.25 | tzanger | ok |
| 04:08.27 | Mavvie | hmm... |
| 04:08.32 | tzanger | how the blue fuck did I end up watching Cricket on CBC? |
| 04:08.36 | Sedorox | lol |
| 04:08.46 | tzanger | pakistan vs someone else I have no idea |
| 04:08.48 | tzanger | wtf |
| 04:08.48 | bjohnson_ | tzanger: sounds like you need a colo here or gamble on a startup |
| 04:09.05 | tzanger | bjohnson_: well I have to talk to Group and Telus and Sprint and Allstream to see just how much it will cost |
| 04:09.12 | Mavvie | <PROTECTED> |
| 04:09.12 | Mavvie | <PROTECTED> |
| 04:09.22 | Mavvie | that is not true. it should be "ringing", not answered. |
| 04:09.28 | tzanger | and past all that, call Bell and Allstream and ask them what it will cost to get rid of the 3 CFB limit |
| 04:09.38 | tzanger | and just call forward to a DID in Kitchener |
| 04:09.43 | bjohnson_ | ok |
| 04:09.46 | tzanger | it's odd though |
| 04:09.50 | tzanger | stratford is not local to Listowel |
| 04:09.56 | tzanger | yet our calls terminate at the switch there |
| 04:10.37 | bjohnson_ | part or Bell's initiative to give local calling to a major city to everyone .. but they don;t want to give away too much |
| 04:10.40 | tzanger | hmm |
| 04:10.45 | Mavvie | If I dial via an FXS zaptel device, does the state immediately go to answered? |
| 04:10.50 | tzanger | dandy.net says that there's a switch right in Listowel |
| 04:10.52 | tzanger | I don't buy that |
| 04:10.58 | Mavvie | (with an PRI zaptel device it goes first to ringing) |
| 04:11.01 | tzanger | when I had the PRI provisioned I had to call Stratford |
| 04:11.21 | tzanger | LTWLON01RS0 is the switch |
| 04:11.22 | tzanger | wtf |
| 04:11.36 | bjohnson_ | well .. I'm off for the night |
| 04:11.46 | tzanger | no way |
| 04:11.48 | jetscreamer | i need to lurk more here |
| 04:11.50 | tzanger | atwood does not have their own swtich |
| 04:11.57 | tzanger | ok now I know this is all virtual bullshit |
| 04:12.00 | *** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
| 04:12.58 | bjohnson_ | hehe |
| 04:13.27 | *** join/#asterisk LawrenceNa (~abc@218.208.119.12) |
| 04:13.45 | tzanger | anyway |
| 04:13.49 | tzanger | I gotta get dishes done and play some guitar |
| 04:13.54 | tzanger | probably in the opposite order |
| 04:13.54 | sivana | hehe |
| 04:15.16 | Mavvie | must be a difference between pri and fxs zaptel devices. |
| 04:16.44 | file[laptop] | Mavvie: no accurate way to do call progression detection on regular analog |
| 04:16.53 | file[laptop] | so calls are immediately flagged as answered |
| 04:17.36 | Mavvie | file[laptop]: that's what I thought, but that would totally ruin the dialstatus afterwards. |
| 04:17.43 | file[laptop] | yup. |
| 04:19.48 | Mavvie | file[laptop] so there is no way to figure out if the call wasn't answered? (or timed out) |
| 04:20.12 | file[laptop] | Mavvie: you can try call progression... go Google... but no telling if it won't disconnect you in the middle of a call |
| 04:22.44 | Druken | anyone here good with flash operator panel ? |
| 04:29.22 | herag | I can dial out just fine through my sipura to my * server, but I can't navigate extension menus through waitexten with it, why not? if I dial into my * server from my cell phone, and access the same menu, everything works fine, everything works fine, what am I doing wrong? |
| 04:29.53 | herag | I'm also ok navigating through 3rd party dtmf menus when I call 800 numbers for example from my sipura |
| 04:29.57 | herag | so I'm sorta at a loss |
| 04:31.09 | herag | what I'm trying to do is dial an extension on my sipura, and get to a menu, and then push a menu option and get to my voicemail |
| 04:31.28 | herag | I can access the menu also through my cell phone, and it works just fine |
| 04:31.48 | herag | I can _reach_ the menu from my sipura, but I can't choose a menu option, I just get a context timeout |
| 04:32.54 | Mavvie | file[laptop]: got it working, thanks! |
| 04:33.13 | JerJer | herag: how is your sipura sending DTMF indication? |
| 04:33.20 | JerJer | make it be the same in sip.conf |
| 04:33.29 | *** join/#asterisk datareactor (datareacto@203.81.192.33) |
| 04:35.18 | Mavvie | file[laptop]: well, as long as I call inside the US :-) |
| 04:48.06 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 04:54.34 | yashax | What's the difference between Cisco 7960 and 7960G? |
| 04:58.41 | stepcut | yeah! I can now receive incoming calls :) |
| 05:02.24 | Nugget | the 7960 has words on the buttons, the G has pictures. |
| 05:03.19 | yashax | that's it? really? |
| 05:03.23 | Nugget | yes |
| 05:04.07 | denon | G has global stuff |
| 05:04.11 | denon | it also can do standard PoE |
| 05:04.14 | denon | (so i hear) |
| 05:04.25 | Nugget | "global stuff" == "pictures instead of words" |
| 05:04.26 | *** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net) |
| 05:05.30 | Nugget | the 7960 can do PoE too, but the pinout changed between the 7960 and the 7960g. |
| 05:05.43 | Nugget | (the standard was finalized and differed from what they used for the earlier phones) |
| 05:06.10 | Nugget | just needs a different cable |
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| 05:17.03 | *** join/#asterisk sudhir492 (~sudhir@4.7.59.167) |
| 05:17.07 | sudhir492 | hi all |
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| 05:19.45 | *** mode/#asterisk [+o twisted[work]] by irc.freenode.net |
| 05:20.32 | Mavvie | MailboxExists(mailbox[@context]) |
| 05:20.34 | Mavvie | Conditionally branches to priority n+101 if the specified voice mailbox exists. |
| 05:20.44 | Mavvie | all this +101 is giving me a headache. |
| 05:22.26 | Mavvie | and it clashes in my dialplan. |
| 05:25.37 | *** join/#asterisk twisted[work] (~twisted@twisted-active-pdpc.developer.and.friend.of.asterisk) [NETSPLIT VICTIM] |
| 05:25.37 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) [NETSPLIT VICTIM] |
| 05:25.37 | *** join/#asterisk Qwell (~north@70-32-102-18.ontrca.adelphia.net) [NETSPLIT VICTIM] |
| 05:25.37 | *** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) [NETSPLIT VICTIM] |
| 05:25.37 | *** join/#asterisk {zombie} (zombie@soulasylum.penguincare.com.au) [NETSPLIT VICTIM] |
| 05:25.37 | *** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc) [NETSPLIT VICTIM] |
| 05:25.37 | *** join/#asterisk angler_ (~angler@suid.digium.com) [NETSPLIT VICTIM] |
| 05:25.37 | *** mode/#asterisk [+o twisted[work]] by irc.freenode.net |
| 05:25.46 | *** join/#asterisk SeaForth (~SeaForth@c-24-1-126-202.client.comcast.net) |
| 05:26.32 | *** join/#asterisk unixgeek (~unixgeek@216-220-234-197.exploremaine.com) |
| 05:36.33 | *** part/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 05:43.56 | *** join/#asterisk Othello (Othello@nusnet-154-210.dynip.nus.edu.sg) |
| 05:46.57 | *** join/#asterisk modulus_ (modulus@rm-f.net) |
| 06:03.39 | *** join/#asterisk kks (~kks@203.115.208.140) |
| 06:04.16 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 06:06.04 | *** join/#asterisk abenamer (~asdf@p67-117.acedsl.com) |
| 06:06.11 | abenamer | hello |
| 06:06.16 | abenamer | anyone on? |
| 06:07.55 | datareactor | yes |
| 06:09.38 | abenamer | oh |
| 06:09.40 | abenamer | ok |
| 06:09.46 | abenamer | am evaluating asterisk |
| 06:10.05 | abenamer | will there be a vxml module for asterisk someday? |
| 06:10.59 | Moc | abenamer, maybe |
| 06:11.08 | Moc | if someone decide to do it |
| 06:11.22 | datareactor | sorry i dont know much about it |
| 06:11.43 | abenamer | so nothing right now? |
| 06:11.45 | abenamer | hmm... |
| 06:11.59 | abenamer | the IVR functionality on asterisk is a bit crude |
| 06:12.07 | abenamer | still requiring C++ skills |
| 06:12.43 | abenamer | i'm looking also to see how many developers in NYC area |
| 06:12.51 | abenamer | i don't see much support for it even in NYC |
| 06:13.28 | *** join/#asterisk IsMe (~some@219.95.222.101) |
| 06:14.53 | abenamer | if I could find some local vendors I'd be more into it |
| 06:19.12 | abenamer | i'm looking to get an asterisk server |
| 06:19.19 | abenamer | for disaster recovery site |
| 06:19.28 | abenamer | it will be backing up an Avaya S8100 |
| 06:19.36 | abenamer | that sound feasible? |
| 06:20.13 | abenamer | ? |
| 06:21.48 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
| 06:24.48 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 06:25.06 | puzzled | morning |
| 06:29.10 | kks | i get a error msg when receiving incomming h323 call -> ClearCall: Request to clear call with token ip$192.168.0.150:33933/4096, cause 7. Can someone give me suggestion? |
| 06:33.19 | *** join/#asterisk JohnWayne (jeff@doublewide.hitechredneck.net) |
| 06:34.01 | JohnWayne | how come I have to reboot asterisk every 30 minutes to keep IAX outgoing calls from not going through? |
| 06:34.37 | JohnWayne | they always hang at -- Called user:pass@gwiaxt01.voicepulse.com/1234567890 |
| 06:34.37 | `Sauron | You could just fix your iax2 config... |
| 06:34.44 | JohnWayne | what is wrong with it? |
| 06:34.50 | jetscreamer | :) |
| 06:35.03 | `Sauron | WRong user, wrong pass, no path to VP |
| 06:35.08 | `Sauron | A number of possibillities. |
| 06:35.15 | *** join/#asterisk clive- (~pirch@myw-stp-66-18-83-222.sentechsa.net) |
| 06:35.27 | JohnWayne | it works fine when it works |
| 06:35.40 | JohnWayne | then it will just stop, and as soon as i restart asterisk it works again |
| 06:35.49 | JohnWayne | I meant to say restart asterisk, not reboot the server |
| 06:37.09 | JohnWayne | any ideas? |
| 06:37.36 | JohnWayne | it's not just voicepulse that does it either, it's the same with nufone, simpletelecom, iax.cc, etc |
| 06:39.41 | *** join/#asterisk drumkilla (~russell@12.21.241.80) |
| 06:39.41 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 06:42.37 | *** join/#asterisk harryjj (~comming@S010600055d210201.vs.shawcable.net) |
| 06:59.04 | *** join/#asterisk pranav (~dawda_pra@202.63.174.250) |
| 06:59.08 | pranav | hello |
| 06:59.24 | pranav | i need help fwd |
| 06:59.33 | pranav | hello |
| 07:03.23 | djin | mrempire, are you online? |
| 07:04.22 | *** join/#asterisk clive- (~pirch@myw-stp-66-18-86-79.sentechsa.net) |
| 07:05.26 | pranav | i need help for fwd |
| 07:06.19 | *** join/#asterisk Luke-Jr (~luke-jr@207.192.221.172) |
| 07:09.16 | Strom_C | pranav: quit whining and just ask the question, and someone will answer it |
| 07:12.27 | Essobi | Anyone know the link to the program that flowcharts your extensions.conf? I been googling all damn weekend and can't find it. |
| 07:17.34 | hmodes | http://megaglobal.net/docs/asterisk/html/dialviz.html |
| 07:17.44 | Essobi | TY! |
| 07:20.48 | *** join/#asterisk letherglov (~letherglo@8036aa5e.resnet.ucsd.edu) |
| 07:24.41 | brc_ | graphviz? |
| 07:24.51 | Essobi | that's it |
| 07:28.13 | PatrickDK | hmm, doesn't do macro's though I bet |
| 07:28.58 | Essobi | AFAIK |
| 07:35.19 | *** join/#asterisk pranav (~dawda_pra@202-63-174-250.broadband.isp.exatt.net) |
| 07:35.39 | pranav | hello any one is there |
| 07:35.42 | pranav | how to create billing using asterisk |
| 07:39.46 | eipi | ppl, any comment on nkans ebay user selling X100P??? |
| 07:39.49 | eipi | oems |
| 07:40.27 | *** join/#asterisk kks (~kks@203.115.210.253) |
| 07:47.29 | Darwin35 | i thought the x100p where phased out |
| 07:47.40 | *** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net) |
| 07:49.51 | PatrickDK | it i |
| 07:49.53 | PatrickDK | it is |
| 07:54.46 | pranav | hello |
| 07:54.58 | brc_ | http://www.switchvox.com/sv?cmd=screenshots&pic_id=1 |
| 07:55.23 | pranav | how to record the call using asterisk |
| 07:56.15 | *** join/#asterisk pulu (~chatzilla@65.77.78.3) |
| 07:56.42 | Darwin35 | we need a gui like tis for * |
| 07:57.54 | pulu | has anyone ever written an agi that does the same thing as authenticate but sets the accountcode to something else besides the pin? |
| 07:59.11 | Darwin35 | prav its call quality control and there is a macro |
| 08:00.31 | Darwin35 | nite all |
| 08:04.45 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
| 08:07.21 | *** join/#asterisk pranav (~dawda_pra@202.63.174.250) |
| 08:07.37 | pranav | hello everyone |
| 08:08.20 | pranav | how to create call records made by asterisk |
| 08:09.01 | *** join/#asterisk CoaxD (~coax@foshay.mpls.mn.pixius.com) |
| 08:09.13 | pranav | i want to listen a call which i made to the asterisk |
| 08:09.19 | CoaxD | Good lord, my whole network is screwed. Stupid extended power outage |
| 08:09.41 | brc_ | Darien, that IS a gui for asterisk |
| 08:09.43 | CoaxD | the whole town is dead on one side. the other is perfectly fine. STupid bastards |
| 08:09.57 | brc_ | CoaxD, no generator? |
| 08:10.12 | CoaxD | brc: Nahhh. We're small. If it is REALLY extended, I do have one and can run a cable.. |
| 08:10.41 | CoaxD | (Hell, the telco across the street'd probably run me a cable if i offered to take 'em out for lunch some afternoon. *lol*) |
| 08:10.58 | CoaxD | THEIR generator is happily buzzing away! |
| 08:11.28 | brc_ | heh |
| 08:11.30 | brc_ | chow |
| 08:11.39 | CoaxD | yepepr |
| 08:12.37 | CoaxD | So i headed in, grabbed a pistol, and started foraging through the town. the 15 minute drive took me 35 minutes due to all the fscking SNOW. I got to the building, and in I went. Fuck, I can't even see my keys. So i thought..fuck that. and headed to my |
| 08:12.49 | CoaxD | 'rents house. |
| 08:13.08 | CoaxD | soons' the powers back on, i'll go see what kind of damage a power outage did to all the ext2 everywhere |
| 08:13.29 | chipig | anyone seen Switchvox in person yet? |
| 08:14.14 | moonwick | grabbed a pistol? is this town full of zombies, or something? |
| 08:14.34 | CoaxD | moonwick: No, but whenever 100% of the power is down, things get unsafe fast. anywhere. |
| 08:15.01 | chipig | thats why i have a big UPS. |
| 08:15.04 | chipig | ill be safe. |
| 08:15.05 | chipig | :P |
| 08:15.09 | CoaxD | chip1g: I have 3 big UPS's |
| 08:15.18 | chipig | and a generator! |
| 08:15.20 | CoaxD | chipig: It'll hold power for an hour and a half. |
| 08:15.37 | CoaxD | chipig: Generator would be the useful part |
| 08:15.43 | CoaxD | chipig: *g* |
| 08:15.47 | moonwick | oic |
| 08:15.55 | CoaxD | chipig: Note: Live switchover generator :) |
| 08:16.24 | brc_ | PTS |
| 08:16.37 | brc_ | CoaxD, ReiserFS. Use it. Love it. |
| 08:16.42 | chipig | yes.. and don't have a big red button (see intermap) |
| 08:16.44 | CoaxD | brc: Um, no |
| 08:16.57 | implicit | only reiser4 |
| 08:17.05 | chipig | reiser4? are you crazy? |
| 08:17.07 | brc_ | chipig, actually, BRB's are required per fire safety code |
| 08:17.11 | brc_ | no |
| 08:17.13 | brc_ | not 4 |
| 08:17.14 | brc_ | not yet |
| 08:17.18 | chipig | brc_: yes.. but.. don't let bad people touch it. |
| 08:17.19 | chipig | :-/ |
| 08:17.24 | brc_ | chipig, yes |
| 08:17.29 | CoaxD | have had extremely bad luck with reiser in the past |
| 08:17.31 | brc_ | it needs to be behind glass |
| 08:17.37 | implicit | 3 is not very good |
| 08:17.39 | implicit | 4 is excellent |
| 08:17.44 | implicit | CoaxD: with reiser3? |
| 08:17.45 | CoaxD | seems i and other people really dont like it when our filesystems just randomly disappear under our computer systems |
| 08:17.50 | implicit | CoaxD: reiser3 is crap |
| 08:17.53 | chipig | xfs :) |
| 08:17.54 | brc_ | erm |
| 08:17.57 | brc_ | yeah I meant 5 not 4 |
| 08:18.08 | implicit | 5 isn't even being developed yet is it? |
| 08:18.14 | CoaxD | i dunno man. hans reiser spent a whole lotta money for nothin', i think |
| 08:18.18 | brc_ | I heard it was |
| 08:18.21 | chipig | does 4 still break apache? |
| 08:18.22 | CoaxD | I worked for SourceForge back when they had it "DONE" |
| 08:18.27 | brc_ | chipig, no |
| 08:18.37 | chipig | good. I didn't want to change our code anyways :) |
| 08:18.39 | CoaxD | WHY THE FUCK WOULD APACHE CARE ABOUT A FILESYSTEM?!#$ |
| 08:18.41 | implicit | checking namesys.com |
| 08:18.44 | CoaxD | You see what i mean? |
| 08:18.46 | chipig | CoaxD: .htaccess files |
| 08:18.47 | CoaxD | IMPROPER CODING. |
| 08:18.59 | chipig | CoaxD: files = directories = different errors returned. |
| 08:19.06 | implicit | yeah no reiser5 yet brc |
| 08:19.12 | CoaxD | abstraction makes it impossible for that to happen |
| 08:19.13 | brc_ | *shrug* |
| 08:19.17 | CoaxD | unless of course, it is not properly abstracted |
| 08:19.23 | CoaxD | which was 100% absolutely the case with reiser |
| 08:19.24 | implicit | CoaxD: or the abstraction layer changes |
| 08:19.28 | CoaxD | (and IS, apparently!) |
| 08:19.33 | CoaxD | implicit: Correct. |
| 08:19.42 | chipig | http://issues.apache.org/bugzilla/show_bug.cgi?id=31126 |
| 08:19.43 | implicit | CoaxD: which is not always a bad thing if you are adding functionality |
| 08:19.52 | implicit | but is bad |
| 08:20.23 | chipig | it was broken, at least for awhile :) |
| 08:20.23 | implicit | if you are getting stupid ass shit |
| 08:20.23 | CoaxD | implicit: Yah. and then you go in and mod every single driver the exact same way and it all works |
| 08:20.23 | CoaxD | implicit: And chances are, the existign filesystem code will not compile cleanly when said abstraction layer changes |
| 08:20.23 | CoaxD | (without being patched) |
| 08:20.38 | CoaxD | the only time is when the symbols dont change - but the behavior of the abstraction layer DOES |
| 08:20.45 | CoaxD | (And that is a *BAD* idea) |
| 08:20.48 | implicit | dont stress though |
| 08:20.52 | implicit | reiser4 is good |
| 08:21.04 | implicit | it has less bugs than asterisk :-p |
| 08:21.11 | *** join/#asterisk Sawwe (~junkmail1@62-20-5-66.customer.telia.com) |
| 08:21.14 | chipig | heh |
| 08:21.15 | CoaxD | there really isnt any reason to run reiser |
| 08:21.18 | chipig | asterisk-- |
| 08:21.18 | CoaxD | not when ext3 exists |
| 08:21.25 | implicit | CoaxD: not true |
| 08:21.26 | chipig | ext3 is slow. |
| 08:21.32 | implicit | chipig: yeah |
| 08:21.34 | CoaxD | ext3 has been clocked, for most intensive purposes, as being damn near as fast as realtime raw disk access |
| 08:21.38 | tzafrir | I use a nice backup software, faubackup |
| 08:21.44 | implicit | CoaxD: for what type of use? |
| 08:21.54 | CoaxD | implicit: See, thast the thing i havent tested much of |
| 08:21.56 | chipig | it depends on your needs. |
| 08:21.57 | implicit | small files? large files? lots of large files in one directory |
| 08:22.00 | chipig | benchmark to your needs. |
| 08:22.03 | implicit | lots of small files in one directory |
| 08:22.04 | implicit | ... |
| 08:22.05 | chipig | use what fits your eneds. |
| 08:22.06 | tzafrir | backs up everything to a filesystem. If a file already evests, it only makes a symlink |
| 08:22.07 | implicit | yeah |
| 08:22.08 | CoaxD | implicit: *ANY* sequential disk access is going to be near raw speeds |
| 08:22.16 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 08:22.22 | implicit | CoaxD: 'sequential' |
| 08:22.27 | tzafrir | but with ext3 it will soon fail due to lack of inodes |
| 08:22.35 | CoaxD | implicit: jump around from beginning to end and then middle yadda yadda yadda 15 million times a second is a different story |
| 08:22.39 | tzafrir | reiserfs is greatr |
| 08:22.47 | tzafrir | great, I mean |
| 08:22.55 | implicit | tzafrir: especially reiser4 |
| 08:22.58 | CoaxD | tzafrir: Dude, thats the fault of the user that generated the filesystem; not the filesystem |
| 08:23.10 | CoaxD | tzafrir: it is easy to create a filesystem with a gazillion inodes if you know yer gonna need 'em |
| 08:23.35 | implicit | damn this is getting insane |
| 08:23.46 | tzafrir | CoaxD, but not after you have files on it, without wiping out those files |
| 08:23.54 | CoaxD | reiser IS probably faster than ext3 at a LOT of stuff; but when bugs like APACHE DOESNT WORK!#$!#$!#$ comes up, that juist spells incorrect coding |
| 08:24.01 | CoaxD | *WAY* incorrect coding |
| 08:24.04 | chipig | heh |
| 08:25.03 | chipig | its actualy technically sort of a bug in behavoir... happens with all old filesystems.. but, is in no way required. |
| 08:25.06 | CoaxD | user processes should not have to be aware of what system processes are doing. Period. |
| 08:25.13 | chipig | its just a preoptimization in apache to avoid extra stat() calls |
| 08:25.29 | CoaxD | i say open a file, it should open it. i say write something to it, it should write. Period. I should not have to know which FILESYSTEM I AM OPERATING UNDER in order to make myself go! |
| 08:25.33 | chipig | so, I don't blame reiser completely :) |
| 08:25.35 | CoaxD | thats the whole point of unix! |
| 08:26.22 | CoaxD | okay, so yer saying that apache took advantage of a shortcut; and with a new filesystem, that shortcut cant work anymore, so.. |
| 08:26.30 | CoaxD | this still sounds to me like a shortcoming of the filesystem |
| 08:26.32 | chipig | yes.. |
| 08:26.46 | CoaxD | if 1 doesn't behave like the many, correct or not, i'd call that borkenness |
| 08:27.10 | chipig | anyways, im happy with XFS. |
| 08:27.12 | chipig | im less happy with chan_sip.c, but thats another story :P |
| 08:27.16 | CoaxD | they should have made reiser work correctly for the masses and then give the new behavior to those who want it but know it'll break shit |
| 08:27.34 | CoaxD | course i spose the reiser folks didnt know it was gonna happen that way and had probably overlooked it |
| 08:36.07 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 08:39.24 | pranav | hello everyone |
| 08:39.42 | pranav | how to create call records made by asterisk |
| 08:39.42 | pranav | i want to listen a call which i made to the asterisk |
| 08:40.37 | zoa | use apple quicktime |
| 08:42.03 | *** join/#asterisk emitrax (~emitrax@host193-51.pool80181.interbusiness.it) |
| 08:42.19 | *** part/#asterisk emitrax (~emitrax@host193-51.pool80181.interbusiness.it) |
| 08:49.21 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
| 08:50.24 | datareactor | praav http://www.voip-info.org/wiki-Asterisk+billing |
| 08:54.11 | *** join/#asterisk schurig (~schurig@p5080A03F.dip0.t-ipconnect.de) |
| 08:55.31 | *** join/#asterisk pranav (~dawda_pra@202-63-174-250.broadband.isp.exatt.net) |
| 08:56.16 | pranav | how to record call in asterisk |
| 08:56.35 | pranav | any one os there |
| 08:57.31 | pulu | monitor app |
| 08:57.57 | pranav | how to record call in asterisk |
| 08:58.23 | pranav | hello mr.pulu do you know |
| 08:58.32 | pulu | monitor app |
| 08:59.29 | pranav | ok from where should i go through it |
| 09:00.25 | pranav | mr.pulu ok from where should i go through it |
| 09:01.05 | pulu | http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Monitor |
| 09:01.22 | pulu | i've never done it before but it seems relatively straightforward |
| 09:03.06 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 09:03.38 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 09:04.47 | Delvar | morning |
| 09:08.06 | letherglov | hallo |
| 09:09.07 | CoaxD | okay, this is getting OLD |
| 09:09.37 | *** join/#asterisk Dibbler_ (~Dibbler@zidane.pi-net.net) |
| 09:09.39 | CoaxD | 2 hours later, the power is STILL out |
| 09:09.46 | moonwick | where is this? |
| 09:09.56 | CoaxD | moonwick: Hicksville, Minnesota |
| 09:09.59 | CoaxD | moonwick: (USA) |
| 09:10.03 | moonwick | heh |
| 09:10.08 | moonwick | what caused it? |
| 09:10.10 | CoaxD | for the record, we haven't had a sizeable outage in 2 years, give or take a few months |
| 09:10.22 | CoaxD | moonwick: God only knwos. I have power here about 3 blocks away, but everything north of main street is dead |
| 09:10.42 | CoaxD | i could string a fucking cable, for chrissakes |
| 09:10.49 | CoaxD | were it not for the major highway in the middle |
| 09:10.52 | moonwick | I was on the lucky side of town |
| 09:10.54 | CoaxD | (main street) |
| 09:11.08 | CoaxD | heh. We are NEVER on the lucky side of town |
| 09:11.24 | CoaxD | i suspect a big outage at a switching station |
| 09:11.42 | moonwick | maybe they ran out of electrons :( |
| 09:11.44 | CoaxD | i.e. a big grouping of cables got burned up due to a short caused by the wet snow |
| 09:11.54 | CoaxD | ya electrons are in short supply these days |
| 09:12.11 | CoaxD | and on top of it, i ate a shitload of meatloaf this evening, and it is now just bustin' my gut. |
| 09:12.30 | moonwick | lemme guess, well water? |
| 09:12.36 | CoaxD | moonwick: Nahhh. |
| 09:12.55 | CoaxD | not THAT rural :L) |
| 09:12.56 | CoaxD | er :) |
| 09:13.09 | CoaxD | (Rural enough, however, to not have KewlStart pots lines) |
| 09:13.14 | CoaxD | ours need busydetect! *lol* |
| 09:13.28 | moonwick | kewlstart? |
| 09:13.57 | CoaxD | (Yay for having to use an AbsoluteTimeout on voicemail() so as to catch busydetect breakage so the damn thing cant sit there and beep at voicemail for 3 days straight, whilst holding the line busy!) |
| 09:14.16 | moonwick | oic |
| 09:14.25 | CoaxD | moonwick: Kewlstart is a way to detect remote hangup |
| 09:14.59 | CoaxD | moonwick: on a phone solely powered by the POTS interface, call it, pick up the phone, and hang up the remote end. Does the power get cut to the phone when you hang up? If so, you have kewlstart |
| 09:15.28 | moonwick | oh |
| 09:15.38 | CoaxD | moonwick: There's also a technique called 'polarity reversal', which is pretty new, i guess, and is done in other countries than USA i think |
| 09:15.46 | moonwick | I don't think I've ever seen that. |
| 09:15.51 | *** join/#asterisk SeaForth (~SeaForth@c-24-1-126-202.client.comcast.net) |
| 09:15.51 | CoaxD | We're stuck on a Nortel DMS-100 over here. Quite old design |
| 09:16.12 | CoaxD | moonwick: (btw, we're about 200 feet from the central office. I heard THEIR generator running! Fuckers.) |
| 09:16.58 | CoaxD | moonwick: In any large urban area, kewlstart is the norm. Just that out here in the woods, the switch doesn't support it, so we're stuck with loop start |
| 09:17.22 | *** join/#asterisk Fabe_ (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 09:17.29 | CoaxD | (can't detect remote hangup with anything other than busydetect, which is listening for the line to make that long, annoying beeping sound after the remote end hangs up.) |
| 09:18.47 | moonwick | I've never noticed anything besideds the busy tone |
| 09:18.54 | moonwick | but it's been a long time since I've tried that |
| 09:19.48 | CoaxD | moonwick: Unless your area has had a switch hardware (or software) upgrade, it wouldn't ahve changed |
| 09:20.29 | CoaxD | i have a phone i use to test this sort of thing. Its a $7.99 walmart special. You know the one. Bell ringer, cheap handset |
| 09:20.49 | CoaxD | go 15 miles in any direction, they have kewlstart. But us? We don't have kewlstart. Bastards. |
| 09:21.02 | CoaxD | THings are SO much easier with kewlstart |
| 09:22.34 | *** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au) |
| 09:22.54 | moonwick | heh |
| 09:23.11 | soundguy | what is the echo test line for extensions.conf? I accidently deleted it :( |
| 09:24.25 | {zombie} | you just need to call the "Echo" application |
| 09:24.36 | {zombie} | eg exten => 600,1,Echo |
| 09:24.42 | *** part/#asterisk djin (~djin@gridfox.xs4all.nl) |
| 09:24.59 | soundguy | ta |
| 09:25.52 | soundguy | that didnt work |
| 09:27.17 | moonwick | maybe your echo test failed. :) |
| 09:27.41 | datareactor | exten => 600,1,Playback(demo-echotest) |
| 09:27.41 | datareactor | exten => 600,2,Echo |
| 09:28.07 | datareactor | the lines moonwick mentioned are o |
| 09:28.13 | datareactor | k |
| 09:28.42 | soundguy | ta |
| 09:28.44 | CoaxD | awww FUCK, they estimate 5:30am! Its 3:30 now! |
| 09:28.53 | *** join/#asterisk pranav (~dawda_pra@202-63-174-250.broadband.isp.exatt.net) |
| 09:29.02 | CoaxD | it is apparently a 'circuit breaker problem |
| 09:29.03 | CoaxD | '' |
| 09:29.03 | moonwick | sounds like it's time to go buy a generator |
| 09:29.19 | CoaxD | moonwick: I've got one. Just need to wait for work to be done to allow us to hook it up) |
| 09:29.35 | moonwick | heh |
| 09:29.41 | moonwick | ironic timing. |
| 09:29.47 | pranav | how to configure call record chennal on asterisk |
| 09:29.48 | CoaxD | moonwick: We're a small business. We haven't had a prob like this in years |
| 09:30.00 | CoaxD | pranav: Pay lots of money. I'll show you how |
| 09:30.10 | CoaxD | at 5:30am :P |
| 09:30.31 | CoaxD | pranov: Seriously, have a look at 'show application monitor' in an asterisk console |
| 09:30.42 | pranav | how much |
| 09:30.48 | CoaxD | pranav: I was kidding |
| 09:31.02 | pranav | yes |
| 09:31.18 | CoaxD | pranav: http://www.voip-info.org and search on 'monitor'. it'll most likely pull up a filename generation macro, and a syntax for using Monitor() |
| 09:31.32 | CoaxD | i'd cut and paste by config, but, well, power's out at the office. cant get there |
| 09:32.17 | CoaxD | s/by/my |
| 09:32.50 | *** join/#asterisk SeaForth (~SeaForth@c-24-1-126-202.client.comcast.net) |
| 09:34.16 | pranav | hello |
| 09:38.24 | *** join/#asterisk maruz (~maumar@adsl-123-3.38-151.net24.it) |
| 09:41.22 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 09:43.41 | *** join/#asterisk Savage-S (~savage@c514701e0.cable.wanadoo.nl) |
| 09:45.42 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.161.115.68.195.rev.coltfrance.com) |
| 09:47.22 | *** join/#asterisk fishboy1669 (proxyuser@62.69.81.129) |
| 09:47.42 | fishboy1669 | hi guys |
| 09:48.50 | *** join/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com) |
| 09:49.46 | fishboy1669 | god its quiet in here |
| 09:50.41 | moonwick | want me to bang on some pots and pans |
| 09:50.42 | moonwick | ? |
| 09:51.51 | soundguy | is there a wiki page for getting two asterisk servers to talk to eachother? |
| 09:51.59 | soundguy | like what to add to sip.conf and extensions.conf |
| 09:52.42 | moonwick | well, you'll want to start out by looking at iax2.conf |
| 09:52.53 | moonwick | since you'll be using that, not sip |
| 09:53.10 | soundguy | I dont have a iax2.conf |
| 09:53.10 | soundguy | only iax.conf |
| 09:53.18 | moonwick | ah, that's the one |
| 09:54.38 | soundguy | is there a wiki page? |
| 09:55.07 | moonwick | I'm not aware of one |
| 09:55.17 | moonwick | though the page for iax.conf might be useful |
| 09:55.51 | soundguy | http://www.voip-info.org/wiki-Asterisk+-+dual+servers << that maybe? |
| 09:57.13 | *** join/#asterisk ta[i]nted (~ta_i_nted@65-60-70-252-cust.telepacific.net) |
| 09:58.47 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 09:59.28 | Ogun | My configs for it: http://pastebin.ca/5777 |
| 10:04.32 | ta[i]nted | mr roboto |
| 10:05.31 | ta[i]nted | why do u have two general sections? |
| 10:06.09 | ta[i]nted | nm |
| 10:07.35 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 10:07.59 | Ogun | two hosts :) |
| 10:19.03 | soundguy | dont I need a register section for iax? |
| 10:19.37 | Ogun | Not necessarily. |
| 10:19.55 | Ogun | It still works, the downside with my way is that you get the IAX username&pwd in the CDR logs. |
| 10:23.57 | *** join/#asterisk bowman (~bowman@snert3.tal.de) |
| 10:25.07 | bowman | heya. I have a problem with Asterisk behind a router. I want to send out SIP packets containing my Internet IP address instead of 10.0.0.2 - I set externip=a.b.c.d (my IP address) in sip.conf but that didn't help - any hints? |
| 10:27.30 | moonwick | that oughta work |
| 10:27.31 | *** join/#asterisk cc (~cc@byte.fedora) |
| 10:29.24 | eye69 | bowman: Have you set nat=yes ? |
| 10:31.44 | *** join/#asterisk zyke (~zakforeve@84.45.132.117) |
| 10:32.01 | *** join/#asterisk pranav (~dawda_pra@202.63.174.250) |
| 10:32.20 | pranav | hello everyone |
| 10:32.33 | pranav | i am having a doubt |
| 10:33.33 | pranav | on the cli prompt when i give zap show channels |
| 10:33.45 | zyke | where is the best place to get US DIDs? |
| 10:33.56 | pranav | it says unable to find the channel |
| 10:34.23 | bowman | eye69: yep |
| 10:34.44 | pranav | but on the root prompt when i give ztcfg -vvvvv |
| 10:34.55 | pranav | it says 1 channel configured |
| 10:35.01 | pranav | where is the error |
| 10:38.06 | bowman | ah it works |
| 10:38.09 | bowman | :) |
| 10:39.33 | *** join/#asterisk k0ga (~roberto@62-15-230-33.inversas.jazztel.es) |
| 10:41.15 | pranav | hello |
| 10:41.21 | pranav | anyone there |
| 10:42.01 | pranav | y is no one answering |
| 10:43.02 | pranav | hello k0ga |
| 10:43.10 | pranav | are you there |
| 10:44.05 | pranav | i am asking a question RoyK |
| 10:45.15 | *** join/#asterisk zoa (~zoa@pirus.securax.be) |
| 10:45.27 | *** join/#asterisk r1 (~erwan@www.thiscow.com) |
| 10:45.55 | k0ga | hello pranav |
| 10:46.03 | pranav | ya hello |
| 10:46.43 | pranav | in the cli prompt wheni give zap show channels it says unable to find the channel |
| 10:47.47 | pranav | where as in the root prompt when i give ztfcg -vvvvv it says 1 channel configured |
| 10:50.24 | zyke | anyone has US DIDs? |
| 10:51.50 | *** join/#asterisk bowman (~bowman@snert3.tal.de) |
| 10:52.07 | bowman | heya. my ZAP channels all say "default law: alaw", but I'm in germany - I need ulaw :) where do I set this? |
| 10:53.24 | pranav | mr k0ga do you know where the error is? |
| 10:53.49 | nitram | bowman: huh? alaw is for europe |
| 10:54.00 | nitram | bowman: germany has alaw |
| 10:55.46 | bowman | nitram: oh. ehm. :D |
| 11:02.51 | nitram | hrhr |
| 11:03.17 | pranav | hello some one plz answer my question |
| 11:03.43 | k0ga | pranav, I don't know where the problem could be, if you give some adittional information maybe someone can help you |
| 11:06.48 | JunK-Y | pranav: its cause ya didnt setup the channels properly |
| 11:10.43 | *** join/#asterisk nazgool (~nazgoool@port-83-236-180-106.static.qsc.de) |
| 11:10.47 | nazgool | hi |
| 11:10.53 | *** join/#asterisk pranav (~dawda_pra@202-63-174-250.broadband.isp.exatt.net) |
| 11:11.19 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 11:21.09 | pranav | hello |
| 11:21.42 | nazgool | i have the impression that my asterisk matches too quickly on incoming calls (people don't have the time to finish dialing). what's the option to set the time for dialing? |
| 11:22.12 | Delvar | digitTimout? |
| 11:22.28 | *** join/#asterisk QRPartner (~andy@ns1.accu-com.com) |
| 11:22.33 | nazgool | ok thx Delvar |
| 11:22.42 | QRPartner | Hello and Good Morning |
| 11:24.44 | Delvar | morning |
| 11:25.35 | *** join/#asterisk Mother__ (~m@53.Red-217-126-93.pooles.rima-tde.net) |
| 11:25.37 | Mother__ | hi all |
| 11:25.46 | Mother__ | I have some doubts about auto-starting * on FC3 |
| 11:31.18 | *** join/#asterisk djin (~djin@217-19-18-130.dsl.cambrium.nl) |
| 11:33.16 | datareactor | Mother__ whats problem with it |
| 11:33.21 | *** join/#asterisk jerlique (jerlique@lnk254.adl0.adsl.esc.net.au) |
| 11:36.16 | jerlique | does anyone use agent logout and not queue member logout? |
| 11:39.32 | *** join/#asterisk negativecreep (~yama@202.147.174.97) |
| 11:39.34 | negativecreep | hi all |
| 11:40.03 | negativecreep | i have an isdn line and an X100P |
| 11:40.15 | negativecreep | can i connect the isdn line to the X100P..the jack size is different.. |
| 11:41.35 | {zombie} | X100P is for POTS (plain old telephone service) not ISDN |
| 11:42.10 | {zombie} | POTS is analogue, ISDN is digital |
| 11:44.00 | JunK-Y | for isdn, ya need a T100P. |
| 11:45.58 | negativecreep | JunK-Y: so no chance? |
| 11:46.53 | negativecreep | {zombie}: any chance i can connect the X100P to isdn line?? |
| 11:47.13 | {zombie} | dude.. read what I said |
| 11:47.18 | {zombie} | it's not gonna happen |
| 11:47.40 | {zombie} | you need an ISDN card |
| 11:48.28 | ta[i]nted | i have a coaxial cable |
| 11:48.54 | ta[i]nted | and commercial grade microwave |
| 11:49.15 | ta[i]nted | can i connect them? |
| 11:49.28 | negativecreep | thnx {zombie} |
| 11:49.30 | {zombie} | sure, why not |
| 11:49.45 | {zombie} | make sure you set up a webcam first |
| 11:50.42 | QRPartner | Anyone familiar with setting the alarm debounce for the T1 card (T100P) |
| 11:58.01 | *** join/#asterisk [ro]nic3try (~iancu@81.181.199.33) |
| 12:02.43 | *** join/#asterisk hellop (~hellop@cpe-69-75-47-251.hawaii.rr.com) |
| 12:04.27 | *** join/#asterisk Starfall (~starfall@212.135.91.19) |
| 12:04.40 | hellop | Anyone know what voice recognition software companies like Sprint and Compaq use on their phone systems? |
| 12:04.59 | *** join/#asterisk ddfire (~david@OL206-180.fibertel.com.ar) |
| 12:04.59 | hellop | It's soo accurate compared to perlbox.. |
| 12:05.05 | ddfire | hello |
| 12:05.35 | ddfire | i am looking for some info about asterisk and speech recognition |
| 12:05.43 | RoyK | ddfire: why???? |
| 12:06.21 | ddfire | RoyK, because i need to implent it |
| 12:07.08 | ddfire | RoyK, i need when someone call to my house say whit who he want to talk |
| 12:07.21 | ddfire | instead to press 1 or 2 ... |
| 12:08.31 | RoyK | ddfire: do you really want to treat people that badly? |
| 12:10.50 | ddfire | RoyK, .... please help me if you can |
| 12:11.24 | Mavvie | ddfire: you'd better do DTMF "name" talking instead of voice recognition. |
| 12:11.25 | JunK-Y | Feb 14 06:57:58 WARNING[17762]: app_checkblacklist.c:188 checkblacklist_exec: CheckBlacklist: Unable to connect to database server 10.0.0.10. |
| 12:11.25 | JunK-Y | Feb 14 06:57:58 WARNING[17762]: app_checkblacklist.c:189 checkblacklist_exec: CheckBlacklist: Reason: FATAL: connection limit exceeded for non-superusers |
| 12:11.26 | JunK-Y | wtf? |
| 12:12.14 | RoyK | ddfire: was that you on the mailing list? |
| 12:12.21 | RoyK | there was just something about sr on the ml@ |
| 12:13.14 | ddfire | i singup today in the list |
| 12:16.26 | ddfire | Mavvie, sorry i dont undesrtand you can repeat? |
| 12:16.47 | Mavvie | ddfire: you'd better do DTMF "name" talking instead of voice recognition. |
| 12:17.28 | ddfire | Mavvie, ... waht do you say? use the traditional mode to press buttons? |
| 12:18.12 | Mavvie | yes |
| 12:18.36 | ddfire | Mavvie, but i need to do whit speech recognition |
| 12:18.56 | Mavvie | speech recognition is not easy. |
| 12:19.18 | Mavvie | actually, it is difficult. |
| 12:19.26 | ddfire | i know, and i am looking for info because i know it isnt easy |
| 12:20.07 | Mavvie | I mean, really difficult. |
| 12:20.26 | ddfire | Mavvie, ok but i dont have any other option |
| 12:20.43 | Mavvie | if you just let them enter the 'name' with the DTMF keys, it is easy to figure out who is needed, specially if you only have a short list of people. |
| 12:20.48 | *** join/#asterisk pranav (~dawda_pra@202.63.174.250) |
| 12:20.59 | pranav | hello everyone |
| 12:21.05 | Mavvie | mavvie = 628843 |
| 12:21.55 | ddfire | Mavvie LOL no there area lot of large names and dificults like rusia names or germany names |
| 12:22.18 | hellop | Also, the VR should be easy because of a limited word list. |
| 12:22.27 | [ro]nic3try | I have 2 sip devices, and i'm tring to call from one to another.. what should i do to make it work ? |
| 12:22.27 | pranav | i am facing a problem in zaptel channel configuration |
| 12:22.41 | [ro]nic3try | wat do i need to configure ? |
| 12:22.52 | Mavvie | oh, cyrillic is a different song indeed. |
| 12:23.10 | pranav | ro:sip.conf and extensions.conf |
| 12:24.53 | [ro]nic3try | ok.. and the line context=from-sip |
| 12:25.01 | [ro]nic3try | what reprezent ? |
| 12:25.21 | pranav | ya thats the context |
| 12:25.32 | pranav | u need to kep it same |
| 12:26.09 | hellop | ddfire, so maybe perlbox could help with the VR |
| 12:26.14 | pranav | in fact i am also new but just a few days back i configured my 2 sip phones |
| 12:26.17 | [ro]nic3try | but where do i configure the rules on this context ? |
| 12:26.23 | hellop | But, it is not nearly real-time. |
| 12:26.26 | ddfire | perlbox??? |
| 12:26.32 | ddfire | hellop, tell me more please |
| 12:26.59 | pranav | i din't get ur question |
| 12:27.37 | hellop | ddfire just google for it. You have to install festival first. |
| 12:28.12 | hellop | ddfire, like I said before at best I got a 50% success rate.. But, it's open source, so maybe |
| 12:28.15 | [ro]nic3try | where an how do i configure context ? |
| 12:28.35 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 12:28.41 | pranav | see u'll have to read the documents |
| 12:28.57 | hellop | we could edit it to always choose an option. So, if given the choice between yes/no it would choose the closest match, and we could achieve a 100% success rate. |
| 12:29.06 | pranav | u must be having an example of sip.conf and extensions.conf |
| 12:31.52 | *** join/#asterisk mak_ (~mak@privat.ua-online.net) |
| 12:31.53 | mak_ | hi |
| 12:31.53 | pranav | hi mak_ |
| 12:31.53 | ddfire | hellop may i send you a priv msg? |
| 12:31.55 | mak_ | can I use SayDigits with Dial ? :) I need to dial and Say Digits :-) |
| 12:32.01 | hellop | ddfire, again, the secret to success with VR is to have a small word list. With regular cheapy microphones, and noise, VR is nearly impossible otherwise. |
| 12:32.04 | mak_ | pranav: |
| 12:32.51 | pranav | hi , i am facing a problem with zap channels |
| 12:33.58 | pranav | i have my zap channel configured but when i give zap show channels it says unable to find |
| 12:35.13 | pranav | whats the error |
| 12:36.23 | *** join/#asterisk CoaxD (coax@shell1.cornernet.com) |
| 12:36.39 | *** join/#asterisk santiago (~santiago@200.24.109.194) |
| 12:36.42 | CoaxD | whois brc_ brc_ |
| 12:36.44 | CoaxD | balh |
| 12:36.47 | CoaxD | er blah. BLAH! |
| 12:39.21 | pranav | any answers |
| 12:39.24 | *** join/#asterisk gernika (~athorsen@cmu-24-35-7-234.mivlmd.cablespeed.com) |
| 12:39.36 | pranav | ?? |
| 12:43.32 | *** join/#asterisk dtwilson (~dave@host217-36-121-129.in-addr.btopenworld.com) |
| 12:43.58 | dtwilson | hello, has digium stopped selling X101p cards? |
| 12:44.11 | wasim | dtwilson: affirmative, thats EoL |
| 12:44.35 | dtwilson | what's it been replaced by? |
| 12:45.15 | dtwilson | I know there are x101p clones out there - have some here in fact but am finding there's a fair bit of echo and wondered if having an original digium item might counter some of the echo |
| 12:46.22 | hellop | Where would I find professional Asterisk Consultants? |
| 12:46.31 | wasim | hellop: hello! |
| 12:46.43 | hellop | hi |
| 12:46.52 | wasim | dtwilson: its been replaced by the TDM01 |
| 12:47.12 | dtwilson | thanks wasim |
| 12:47.23 | pranav | hello |
| 12:47.37 | *** join/#asterisk JunK-Y (~junky@modemcable056.110-81-70.mc.videotron.ca) |
| 12:47.55 | JunK-Y | if 9690 root 25 0 3804 2608 3668 R 99.9 0.1 14:35.44 mpg123 |
| 12:48.16 | JunK-Y | a mpg123 process is taking 99.9% of my cpu, do i need to restart asterisk or kill that process? |
| 12:51.26 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 12:51.36 | ariel_ | good morning all. |
| 12:51.38 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.161.115.68.195.rev.coltfrance.com) |
| 12:52.29 | pranav | is there any method to record and listen the calls which are made |
| 12:53.09 | pranav | morning ariel |
| 12:53.12 | wasim | pranav: show application monitor |
| 12:54.23 | pranav | ok but i want to listen to the caslls which i have made |
| 12:54.30 | pranav | sorry calls |
| 12:54.40 | wasim | pranav: retrospectively? |
| 12:54.55 | wasim | pranav: you need an app_timemachine then |
| 12:55.15 | zoa | hey wasim |
| 12:55.21 | wasim | zoa! |
| 12:55.22 | *** join/#asterisk riksta (~rick@81-178-207-173.dsl.pipex.com) |
| 12:55.58 | pranav | ok |
| 12:57.33 | pranav | ok and to only record the calls which i have madei have to give the command "show application monitor" |
| 12:57.49 | pranav | write wasim? |
| 12:59.52 | *** join/#asterisk breakpoint (~rvramos@202.163.202.212) |
| 13:00.33 | breakpoint | hello everybody |
| 13:03.03 | *** part/#asterisk djin (~djin@217-19-18-130.dsl.cambrium.nl) |
| 13:04.08 | *** join/#asterisk CletusColeman (~CletusCol@c-24-0-179-254.client.comcast.net) |
| 13:07.25 | *** join/#asterisk breakpoint (~rvramos@202.163.202.212) |
| 13:07.39 | breakpoint | hi everyone |
| 13:08.16 | breakpoint | kind of newbie here |
| 13:08.27 | breakpoint | can anybody i can ask to? |
| 13:10.05 | pranav | wasim ?? |
| 13:10.08 | wasim | pranav: no, type "show application monitor" on *CLI to learn about the application called monitor which does what you want |
| 13:10.19 | wasim | breakpoint: ask away, someone or the other will answer |
| 13:10.36 | breakpoint | wasim: thanks |
| 13:10.56 | pranav | ok thanks wasim |
| 13:11.27 | breakpoint | i would like to accept something like an account number followed by the # sign |
| 13:11.41 | breakpoint | how can i do this? |
| 13:12.10 | breakpoint | exten => s,1,Background(enter-account-num) |
| 13:12.18 | breakpoint | exten => s,2,???? |
| 13:12.20 | wasim | breakpoint: ideally you'd do this in an AGI using GET_DATA |
| 13:12.51 | wasim | breakpoint: although you can use DBget also |
| 13:14.11 | breakpoint | am afraid i dont know how to use AGI for now |
| 13:14.43 | wasim | breakpoint: AGI rocks, do you know a little perl? |
| 13:14.44 | srt | breakpoint: what would you like to do with the entered account number anyway? |
| 13:15.00 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 13:15.38 | breakpoint | srt: after getting the account number i would query it in the database to get the password |
| 13:16.24 | breakpoint | wasim: not quite pearl, i use c,c++,pascal |
| 13:16.35 | pranav | wasim : do you know a perl |
| 13:16.50 | pranav | i mean perl language |
| 13:16.56 | wasim | breakpoint: thats fine as well, also see cmd Read and Authenticate |
| 13:17.01 | breakpoint | and after getting the password and valid then the call will pass through |
| 13:17.43 | srt | if its not a "legacy" db you could just use authenticate... |
| 13:17.45 | breakpoint | wasim: yeah, i already used Authenticate but that is for password only. will check what Read can do |
| 13:18.07 | breakpoint | srt: i use mysql in my database for postpaid and prepaid accounts |
| 13:18.12 | wasim | READ does in dialplan what GET_DATA does in AGI |
| 13:18.33 | pranav | i am using astguiclient and when i give the command /home/cron/ADMIN_area_code_populate.pl it gives an error "cant loacte net/mysql.pm in @INC" |
| 13:18.58 | srt | if you already have some code to access the db, agi will be a good way to go |
| 13:19.00 | wasim | pranav: you probably don't have mysql cpan stuff |
| 13:19.17 | srt | no need to do it in perl if you dont know hat, c, c++, etc works equally well |
| 13:19.23 | *** join/#asterisk _Brian (brian@unix01.voicenet.com) |
| 13:19.39 | srt | only java is a bit tricky ;) |
| 13:19.41 | breakpoint | srt: i already have an access to my db using MYSQL command |
| 13:19.53 | pranav | no from the cpan prompt when i give install Net::MySQL it says that net/mysql is up to date |
| 13:20.27 | tzafrir | net::mysql , actually (unless this is windows?) |
| 13:20.56 | tzafrir | but normal mysql access is through DBD, isn't it? |
| 13:21.14 | *** join/#asterisk breakpoint_bgo (~rvramos@202.163.202.212) |
| 13:21.25 | breakpoint_bgo | was disconnected |
| 13:21.32 | pranav | no i have an rpm of mysql with DBD also |
| 13:22.24 | srt | breakpoint_bgo: only access via the mysql cli or do you already have an application to maintain the accounts? |
| 13:22.32 | *** join/#asterisk ToyMan (~konversat@204-8-82-238.webjogger.net) |
| 13:23.11 | breakpoint_bgo | srt: can only access via MYSQL in the exten |
| 13:23.47 | breakpoint_bgo | srt: after that i would like to prompt user to enter their account number terminated by the # key |
| 13:24.01 | breakpoint_bgo | srt: then compare what the user entered and what i queried from DB |
| 13:25.12 | srt | breakpoint_bgo: as wasim said the best way to do that is via an agi - written in any language you feel comfortable with and that runs on your asterisk box |
| 13:25.43 | breakpoint_bgo | srt: thanks, will google that |
| 13:25.51 | wasim | breakpoint_bgo: wiki it |
| 13:26.03 | breakpoint_bgo | wasim: that's better :) |
| 13:26.15 | srt | breakpoint_bgo: http://www.voip-info.org/wiki-Asterisk+AGI |
| 13:26.26 | breakpoint_bgo | srt: thanks |
| 13:26.58 | breakpoint_bgo | BTW, is the connection between voicepulse and asterisk secured? |
| 13:27.49 | srt | probably not - if you dont use a vpn connection to voicepulse ;) |
| 13:28.08 | breakpoint_bgo | i mean, when a call is currently under way, can someone able to 'tap' and 'hear' the communication? |
| 13:28.47 | srt | yes same as with unencrypted email |
| 13:29.05 | breakpoint_bgo | srt: :( that's bad |
| 13:29.34 | wasim | breakpoint_bgo: encrypt it |
| 13:29.56 | nitram | damn hipath |
| 13:30.17 | breakpoint_bgo | wasim: do IAX support encryption? |
| 13:30.26 | wasim | breakpoint_bgo: its in the works |
| 13:30.52 | wasim | farfon should support it soon |
| 13:31.41 | breakpoint_bgo | wasim: is there any other IAX provider that supports encryption? |
| 13:32.05 | wasim | breakpoint_bgo: not that i'm aware of, its still very alpha, but i bet you could cajole jerjer to do so |
| 13:32.34 | wasim | jerjer == nufone |
| 13:33.32 | breakpoint_bgo | wasim: thanks |
| 13:33.34 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 13:36.01 | *** join/#asterisk eivindtr (~Eivind@193.91.146.34) |
| 13:37.02 | breakpoint_bgo | i was able to get the number entered after the # key using the Read command |
| 13:37.57 | breakpoint_bgo | exten => 9,1,Background(enter-account-number) |
| 13:38.16 | breakpoint_bgo | exten => 9,2,Read(accountnumber) |
| 13:38.34 | wasim | breakpoint_bgo: if they enter the number during the 9,1, it won't go to 9.2 |
| 13:38.35 | breakpoint_bgo | exten => 9,3,SayNumber(${accountnumber}) |
| 13:39.08 | wasim | breakpoint_bgo: you should use Read(accountnumber|enter-account-number) |
| 13:41.01 | breakpoint_bgo | wasim: that means that Read will play the sound enter-account-number while waiting for accountnumber to be entered? |
| 13:41.27 | wasim | breakpoint_bgo: show application read on *CLI |
| 13:41.40 | wasim | breakpoint_bgo: Read(variable[|filename][|maxdigits][|option]) |
| 13:42.44 | [ro]nic3try | need help : if i dial from my sip phone 1 it send an invite ro 3333@192.168.1.89 |
| 13:43.02 | [ro]nic3try | an my asterisk response 404 |
| 13:49.13 | breakpoint_bgo | wasim: got an error doing this |
| 13:49.41 | breakpoint_bgo | wasim: exten => 9,1,Read(accountnumber | agent-user) |
| 13:49.49 | wasim | breakpoint_bgo: no spaces |
| 13:50.04 | breakpoint_bgo | wasim: exten 9,2,SayNumber(${accountnumber}) |
| 13:50.10 | bjohnson | can anyone explain what you actually get access to with the DISA command? |
| 13:50.29 | breakpoint_bgo | wasim: error is ast_streatmfile: unable to open agent-user (format unknown) |
| 13:54.08 | ariel_ | bjohnson, if you are calling to your pbx from the outside and want to make an outside call you would use DISA it gives you a dialtone for an outside call. |
| 13:54.12 | breakpoint_bgo | wasim: thanks, i got it working! |
| 13:55.20 | *** part/#asterisk [ro]nic3try (~iancu@81.181.199.33) |
| 13:55.23 | *** join/#asterisk NTJOCK (~brian@txshirts.com) |
| 13:55.48 | NTJOCK | good morning |
| 13:55.49 | NTJOCK | ;) |
| 13:55.55 | breakpoint_bgo | wasim: now i have to face the problem of looping such that user will have only have 3 tries to enter their accountnumber |
| 13:56.17 | breakpoint_bgo | wasim: like the Authenticate can do |
| 13:57.12 | wasim | breakpoint_bgo: thats simple, use variables |
| 13:58.19 | jalsot | I have a quick question: is the zaptel driver stable enough with linux-2.6? [e.g. 2.6.11rc4] any experience? |
| 13:58.40 | NTJOCK | zaptel is working for me on deb 2.6 |
| 13:58.40 | wasim | <PROTECTED> |
| 13:58.44 | wasim | ugh ... |
| 13:58.48 | wasim | exten => s,1,SetVar(v1=0) |
| 13:59.02 | *** join/#asterisk inspired (mikael@host-81-191-127-166.bluecom.no) |
| 13:59.03 | wasim | exten => s,5,SetVar(v1=$[${v1} + 1]) |
| 13:59.03 | wasim | exten => s,6,GotoIf($[${v1} > 2]?s,7:s,3) |
| 13:59.08 | wasim | etc, etc |
| 13:59.13 | wasim | sorry for the paste, folks |
| 13:59.17 | jalsot | NTJOCK: which version of * do you use? stable-1_0 or HEAD? |
| 13:59.18 | NTJOCK | np |
| 13:59.21 | NTJOCK | head |
| 13:59.24 | NTJOCK | I had hell with stable |
| 13:59.34 | NTJOCK | I just grabbed it last nite |
| 13:59.40 | jalsot | what was the problem with HEAD? |
| 13:59.43 | NTJOCK | I kept having seg faults with head. |
| 13:59.58 | NTJOCK | I accidentally (following the )(*^(*&%$*%$&(* instructions) grabbed the "add-ons" |
| 14:00.10 | NTJOCK | and a couple of the add-ons kept causing seg-faults |
| 14:00.22 | breakpoint_bgo | wasim: thanks a lot |
| 14:00.26 | NTJOCK | I gave up and copied my configs, purged, and restarted. |
| 14:00.31 | tzanger | morn |
| 14:00.44 | wasim | breakpoint_bgo: my swiss banker will say you're welcome |
| 14:00.45 | NTJOCK | at least now when I hang up it doesn't crash! |
| 14:00.54 | jalsot | :) |
| 14:01.11 | jalsot | I'm using stable on 2.4 and don't have any problem |
| 14:01.14 | NTJOCK | I had started with the deb package, and then layered CVS. |
| 14:01.17 | NTJOCK | I don't recommend that |
| 14:01.18 | NTJOCK | :) |
| 14:01.21 | breakpoint_bgo | wasim: :) |
| 14:01.23 | NTJOCK | just go get the CVS |
| 14:01.27 | jalsot | just chan_misdn drops segfault on 2.4 kernels |
| 14:01.33 | jalsot | so I need to go for 2.6 |
| 14:01.57 | jalsot | I use 1_0 CVS branch ;) |
| 14:02.02 | NTJOCK | well, I'm a linux newbie... so I'll just say I run what my linux mentor tells me to. :) |
| 14:02.04 | jalsot | no deb packages |
| 14:02.12 | jalsot | :)) |
| 14:02.12 | loud | (08:09:56 pm) <tzanger> and it's important for 1 of those 3 to have full "roll down" on the entire PRI |
| 14:02.12 | loud | (08:09:59 pm) *** Quit: heath__ has quit irc ("Leaving") |
| 14:02.12 | loud | (08:10:22 pm) <bjohnson_> I remembered that Wightman telecom is a new name around here but there are telecomunications and |
| 14:02.16 | loud | offer residential dsl .. you could talk to them since they likely have access to PRIs |
| 14:02.17 | loud | (08:10:26 pm) <tzanger> CFB apparently quits after 3 simultaneous calls unless you use transfer on Centrex |
| 14:02.19 | loud | (08:10:42 pm) <tzanger> bjohnson_: yeah |
| 14:02.21 | loud | (08:10:48 pm) <tzanger> Wightman's a little pissed at me |
| 14:02.22 | NTJOCK | I had a lot of problems with the 1_0 branch. |
| 14:02.24 | loud | (08:10:51 pm) <tzanger> I have VOIP working and they don't |
| 14:02.26 | loud | (08:10:59 pm) <tzanger> they rolled it out in Elmira and had to cancel it |
| 14:02.28 | loud | (08:11:06 pm) <Sedorox> tzanger: those cards.. they act like a csu/dsu I'm assuming, so you just plug from the wall into the |
| 14:02.30 | loud | card w/o any other HW? |
| 14:02.32 | loud | (08:11:20 pm) <tzanger> Sedorox: you will plug the telco smartjack into them |
| 14:02.34 | loud | <PROTECTED> |
| 14:02.41 | jalsot | NTJOCK: what kind of problems? |
| 14:02.46 | NTJOCK | crashing * |
| 14:02.51 | jalsot | ah |
| 14:02.59 | NTJOCK | when you hung up with a SIP call it just did a seg-fault exit stage left. |
| 14:03.07 | NTJOCK | very annoying |
| 14:03.09 | NTJOCK | no traces in logs |
| 14:03.13 | *** part/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
| 14:03.25 | loud | ahora son todos consultores.ol\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\O |
| 14:03.31 | NTJOCK | I think it had to do with some mysql logging crap in the add-ons package |
| 14:04.02 | NTJOCK | at least now it works |
| 14:04.12 | NTJOCK | I finally got my sipura spa-841's in |
| 14:04.18 | bjohnson | ariel_: how does it select which outside line to use? Doesn't it just forward you to another context that "MAY" allow the caller to dial out if the exten's are there? |
| 14:04.19 | jalsot | ok, I will try 1_0 on 2.6 1st and later will look for HEAD - just don't know when is it in stable state :) |
| 14:04.26 | NTJOCK | so much easier to configure then the polycom 500's |
| 14:05.04 | breakpoint_bgo | bye for now |
| 14:05.07 | breakpoint_bgo | thanks to all |
| 14:05.16 | bjohnson | loud: ? |
| 14:05.23 | Moc | NTJOCK, it aint the same type of phone too hehe |
| 14:05.25 | NTJOCK | jalsot: if I have a simple extensions.conf, how (where) do I put the musiconhold in for a extension? |
| 14:05.31 | *** join/#asterisk djin (~marius@62.58.40.196) |
| 14:05.36 | *** part/#asterisk breakpoint_bgo (~rvramos@202.163.202.212) |
| 14:05.40 | *** join/#asterisk [ro]nic3try (~iancu@81.181.199.33) |
| 14:06.11 | NTJOCK | moc: true... but it should take hours to configure a phone... and the 500's are evil when it comes to FTP |
| 14:06.20 | jalsot | NTJOCK: hmm, I have never used moh |
| 14:06.25 | Moc | ?? I never had that much problem.. |
| 14:06.33 | jalsot | NTJOCK: I'm not an expert, yet :( |
| 14:06.40 | jalsot | NTJOCK: thanks for comments |
| 14:06.44 | NTJOCK | neither am i |
| 14:06.50 | NTJOCK | I'm just a babe with a box of legos here.... |
| 14:06.59 | *** join/#asterisk mogorman (~mogorman@dhcp-162.digium.com) |
| 14:07.00 | NTJOCK | LOL |
| 14:07.29 | NTJOCK | although, I have about $2K into my "thank you for calling digium" box.... :) |
| 14:07.34 | mak_ | can I use SayDigits with Dial ? :) I need to dial and then Say Digits :-) |
| 14:07.36 | NTJOCK | including system and phones |
| 14:07.45 | NTJOCK | LOL |
| 14:08.37 | *** join/#asterisk yashax (daemon@69.15.218.218) |
| 14:08.40 | mak_ | can you help me with this ? :) |
| 14:08.57 | wasim | mak_: umm .. you want to say digits to the called party? |
| 14:09.20 | yashax | What is the latest SIP firmware that works with Cisco 7960 on *? |
| 14:09.58 | srt | yashax: 07-3-00 |
| 14:10.50 | yashax | is this v6? |
| 14:11.01 | srt | v7 |
| 14:11.06 | srt | 7.3 |
| 14:11.15 | bjohnson | NTJOCK: add a 'm' arg to your dial command |
| 14:11.23 | yashax | got it.... works well with *? |
| 14:11.35 | srt | yes its fine |
| 14:12.19 | yashax | thank you VERY much for the info..... is it available for download from anywhere? I purchased the phones from ebay and need to flash them..... |
| 14:13.15 | [ro]nic3try | re all |
| 14:13.42 | [ro]nic3try | does anyone like to help a beginger in asterisk ? |
| 14:13.55 | [ro]nic3try | :) |
| 14:14.17 | srt | just ask [ro]nic3try |
| 14:14.19 | Moc | ask your question and we will see ;) |
| 14:14.47 | NTJOCK | bjohnson: so like exten=> 527,2,dial(sip/527|20|trm) ? |
| 14:15.04 | [ro]nic3try | i have just setup 2 dip devices on a local net |
| 14:15.15 | [ro]nic3try | and i canot make a call from one to another |
| 14:15.37 | [ro]nic3try | it just keep giving me a 404 not found eror |
| 14:15.39 | *** join/#asterisk shadebob (~shadebob@rnis-162-206-192-81.marocconnect.com) |
| 14:16.25 | [ro]nic3try | i have configures sip.conf and extension.conf |
| 14:17.06 | bjohnson | mak_: maybe 'M(x) -- Executes the macro (x) upon connect of the call' (arg to dial command)? or make a conference and join both people and the thing that says digits? Check the wiki to see if someone has played a file after the call is answered (I think someone has but may not have documented it) |
| 14:17.08 | bjohnson | ~docs |
| 14:17.09 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 14:18.04 | bjohnson | NTJOCK: yes, except remove the r .. assuming that is the dial command to call an extension (where you want the moh to play until answered) |
| 14:18.50 | NTJOCK | ok, I understand what you are saying, but what I want is when the person hits the hold button they here music on hold |
| 14:18.58 | NTJOCK | when it dials... I want it to be ringing |
| 14:19.53 | *** join/#asterisk carbon60 (~adam@gw.techsupport.ca) |
| 14:20.05 | carbon60 | Morning all. |
| 14:20.17 | carbon60 | Has anyone had issues with Polycom IP500s locking up hard? |
| 14:20.21 | bjohnson | NTJOCK: r fakes a ring .. even if it is busy |
| 14:20.49 | NTJOCK | oh. |
| 14:20.56 | bjohnson | NTJOCK: I think m will do what you want |
| 14:21.00 | NTJOCK | hmm, let me go look at the docs.... |
| 14:21.04 | NTJOCK | i'll remove the r then |
| 14:21.08 | NTJOCK | :) |
| 14:21.08 | NTJOCK | thanks |
| 14:21.37 | NTJOCK | I feel silly asking these questions, but * is so complicated... and there isn't really a nice way to get going other then trial and error. |
| 14:22.07 | bjohnson | ~docs |
| 14:22.08 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 14:22.19 | NTJOCK | been to all of the above |
| 14:22.21 | bjohnson | check out the musiconhold info on the wiki |
| 14:22.31 | NTJOCK | asteriskdocs was a disaster |
| 14:22.39 | NTJOCK | I did read the wiki. |
| 14:22.45 | bjohnson | I think you have to make sure you set a moh context that matches one in musiconhold.conf |
| 14:23.13 | bjohnson | in the general section of your sip.conf for example |
| 14:23.13 | NTJOCK | but the snippet was extension specific and I wasn't sure if the MOH was something that "lived" at an extension... or if that was an example of it for an extension |
| 14:23.35 | NTJOCK | ah ok |
| 14:23.40 | NTJOCK | I have the musiconhold.conf set |
| 14:23.43 | NTJOCK | but not sip |
| 14:23.53 | bjohnson | I haven't played with moh much since I'm just using * in front of a pbx .. not really as THE pbx |
| 14:24.02 | NTJOCK | I was going to do that |
| 14:24.08 | NTJOCK | but decided it was cheaper to just replace the pbx |
| 14:24.09 | NTJOCK | :) |
| 14:24.13 | NTJOCK | and we could get more features |
| 14:24.24 | bjohnson | make sure you edit the wiki when you figure it out to make it easier for the next guy |
| 14:24.29 | NTJOCK | * costs about what Nortel wanted for a Voicemail $y$tem |
| 14:24.38 | NTJOCK | hehe |
| 14:24.43 | NTJOCK | I guess that would be nice (editing the wiki).... |
| 14:24.52 | NTJOCK | * does a nice job of forwarding it to email |
| 14:24.57 | NTJOCK | a really nice job actually |
| 14:25.01 | NTJOCK | then you can have VSpam |
| 14:25.02 | *** join/#asterisk sabre (~urfos@69.149.209.83) |
| 14:25.02 | NTJOCK | ;) |
| 14:25.04 | *** join/#asterisk coppice (~chatzilla@245.195.17.210.dyn.pacific.net.hk) |
| 14:26.12 | NTJOCK | be back in a few... I'm off to burger thing for a ham crosancritter and some tater spots. |
| 14:26.13 | NTJOCK | ;) |
| 14:26.21 | [ro]nic3try | how do i use the demo from extension.conf ? |
| 14:26.26 | NTJOCK | 500\ |
| 14:26.30 | NTJOCK | er 5-- |
| 14:26.32 | NTJOCK | er 500 |
| 14:26.32 | NTJOCK | :) |
| 14:27.51 | bjohnson | or 600 |
| 14:27.54 | bjohnson | or a few others |
| 14:28.05 | bjohnson | look in the [demo] section of extensions.conf |
| 14:28.26 | bjohnson | exten => 600,s,1,jsdjsdkjsd |
| 14:28.32 | bjohnson | means extension 600 |
| 14:28.33 | mak_ | wasim: yes :) first Dial to some number and when answered say digits :) |
| 14:30.02 | djin | Question, I have a 7940(SIP) behind NAT and after some time I cannot reach the phone from *, but after I called from the phone itself the link is back. Is this a typical SIP-behind-NAT problem? |
| 14:34.04 | carbon60 | Has anyone had issues with Polycom IP500s locking up hard and needing to be power cycled? |
| 14:34.17 | Moc | nope |
| 14:34.25 | darkskiez | djin, put qualify=yes into the sip.conf |
| 14:34.37 | darkskiez | djin, under the appt entry. see if that helps |
| 14:34.59 | djin | darkskiez, thanks. I'll try that. |
| 14:35.32 | *** join/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu) |
| 14:38.32 | djin | darkskiez, just looked up some info on qualify on Wiki and it looks like it could resolve this issue, thanks again. |
| 14:39.38 | *** join/#asterisk eivindtr (~Eivind@193.91.146.34) |
| 14:39.50 | darkskiez | all i really want in the world in a setCalledParty equivalent for sip. |
| 14:42.06 | [ro]nic3try | thx. now the demo run ok |
| 14:42.20 | [ro]nic3try | but i still canot conect to my other phone |
| 14:43.24 | nazgool | if i want something to be executed inside a context, before any actual matching is done, how do i do that? is there a better way than to add a line with priority 1 matching on _. and increment the priorities of all the other stuff to start at 2? |
| 14:43.56 | *** join/#asterisk montag___ (~montag@host187-252.pool8175.interbusiness.it) |
| 14:44.07 | *** join/#asterisk eivindtr (~Eivind@193.91.146.34) |
| 14:44.54 | montag___ | hi, mi dial string is exten => _0XXXX.,1,Dial(Modem/g1:${EXTEN},120,r), all works correctly, i can place a call but i cannot ear a ringing tone when call is in progress, the client is kphone....any tips ???? |
| 14:45.02 | *** part/#asterisk QRPartner (~andy@ns1.accu-com.com) |
| 14:45.07 | *** join/#asterisk adjacent (~scott@office.bftwave.com) |
| 14:47.40 | ariel_ | bjohnson, sorry I was out feeding the baby. I use DISA like this and it puts me in my dialing context. |
| 14:47.58 | ariel_ | exten => 300,1,Authenticate(XXXXX) |
| 14:47.58 | ariel_ | exten => 300,2,DISA,no-password|toll-access |
| 14:47.59 | ariel_ | exten => 300,3,Hangup |
| 14:48.17 | bjohnson | why do you authenticate when a DISA password can be specified? |
| 14:48.58 | ariel_ | I use authenticate due it asked you for the password and correctly puts you into the next line. |
| 14:49.04 | ariel_ | It works better that way. |
| 14:50.14 | ariel_ | bjohnson, it actually ask you for the password. Test it without and you see all you get is a dial tone. really not very informative. |
| 14:51.15 | *** part/#asterisk datareactor (datareacto@203.81.192.33) |
| 14:51.50 | ariel_ | montag___, some times softphones and other device don't have an internal ringing. |
| 14:52.24 | montag___ | ariel: it's a client issue then.... |
| 14:52.36 | montag___ | asterisk don't provide a self ringing tone ? |
| 14:52.58 | ariel_ | you can add a line like exten => _0XXXX.,1,Ringing then exten => _0XXXX.,2,Dial |
| 14:53.20 | ariel_ | montag___, no it does not. unless your dialing zap ports. |
| 14:53.20 | bjohnson | ariel_: what happens when using authenticate if the wrong password is entered? |
| 14:53.28 | ariel_ | asks again |
| 14:53.34 | montag___ | thanks... |
| 14:53.59 | ariel_ | bjohnson, wait I have not tested with a wrong one just a sec let me try it. |
| 14:54.46 | ariel_ | bjohnson, yepper it ask 3 times if it's wrong it will give you the goodbye message. |
| 14:54.53 | bjohnson | ok |
| 14:55.12 | bjohnson | working on this: http://www.voip-info.org/tiki-index.php?page=Asterisk+user+authentication |
| 14:56.47 | *** join/#asterisk quigleymd (~quigleymd@24-53-142-3.chvlva.adelphia.net) |
| 14:57.43 | wasim | mak_: use the /var/spool/asterisk/qcall |
| 14:58.11 | [ro]nic3try | extern => 7757,1,dial(sip/7757,20,t) |
| 14:58.16 | [ro]nic3try | is this corect ? |
| 14:58.27 | *** join/#asterisk dg1nsw (~dg1nsw@saturn2.franken.de) |
| 14:58.32 | [ro]nic3try | where 7757 is my sip phonr |
| 14:59.04 | ariel_ | bjohnson, I also think you can use it via a text file. like exten => 300,1,Authenticate(/etc/asterisk/password.txt) |
| 14:59.57 | ariel_ | [ro]nic3try, it's spelled exten => |
| 15:05.25 | *** join/#asterisk yashax (~yasha_x@69.15.218.218) |
| 15:05.51 | *** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net) |
| 15:06.32 | [ro]nic3try | :D thx |
| 15:06.40 | *** join/#asterisk {-award-} (admin@pD956A8BC.dip.t-dialin.net) |
| 15:08.25 | [ro]nic3try | now works... :) |
| 15:08.31 | *** join/#asterisk mikayle (~talamar@akro-pool2-cs1.pool.dslohio.net) |
| 15:08.38 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfmom.dialup.mindspring.com) |
| 15:09.57 | *** join/#asterisk carlos-d-man (~carlos@201.128.216.236) |
| 15:10.22 | mak_ | wasim: /var/spool/asterisk$ ls -a |
| 15:10.23 | mak_ | . .. outgoing tmp vm voicemail |
| 15:10.27 | mikayle | any able to compile chan_sccp with the callerid error? |
| 15:11.57 | *** join/#asterisk TekLexus (~Mnemonic@167.206.75.24) |
| 15:12.59 | bjohnson | how is DISA without a password any different than waitexten? |
| 15:13.44 | ariel_ | bjohnson, disa actuall takes you to the context you want for outside dialing. I have not used the waitexten don't know it. |
| 15:14.03 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
| 15:14.49 | ariel_ | I guess actually DISA will take you and give you a dial tone to any other context like if you have a phone that only limites you to one group of number you could take them to another group of extens. |
| 15:15.18 | *** join/#asterisk dsfr (~dsfr@216.207.244.183) |
| 15:15.59 | *** join/#asterisk outsidefactor (barf@203-206-254-16.dyn.iinet.net.au) |
| 15:15.59 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
| 15:15.59 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:16.20 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 15:16.20 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 15:18.53 | yashax | Guys... Debating between Cisco 7600 and Polycom IP500... Huge price difference.... What do you think? |
| 15:20.00 | yashax | 7960.. sorry |
| 15:20.44 | bjohnson | Polycom IP500 are favs here in price/performance for business models |
| 15:21.06 | bjohnson | fav in the budget phones seems to be switching to Sipura SPA 841 |
| 15:21.39 | *** join/#asterisk HitTop (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
| 15:21.50 | *** join/#asterisk heka (~fasada@82.114.68.126) |
| 15:22.07 | bjohnson | availability of iax phones may change that budget phone fav in near future (haven't seen any IAX business models yet) |
| 15:22.09 | _Brian | yashax: I have tested both and I prefer the polycom. Provisioning is nicer and i think overall they look cleaner for a desktop phone |
| 15:22.12 | heka | Hello, where can I enable MYSQL_FRIENDS. I cant find it on asterisk/channels/Makefile |
| 15:22.26 | *** join/#asterisk jero (~boo@199.243.85.90) |
| 15:22.28 | _Brian | heka: do a search for realtime on the wiki |
| 15:22.40 | yashax | thank you for heads up... |
| 15:22.51 | *** join/#asterisk TheEmperor (TheEmperor@218.111.51.151) |
| 15:22.52 | heka | _Brian: I found this http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers |
| 15:23.06 | heka | but I cant find the MYSQL_FRIENDS line on asterisk/channels/Makefile |
| 15:23.44 | heka | _Brian: I found something :) |
| 15:23.45 | _Brian | heka: i assume you want to put your sip users into mysql? |
| 15:23.55 | _Brian | heka: realtime is what you want.. :) |
| 15:24.04 | heka | _Brian: yes thanks |
| 15:24.07 | HitTop | can anyone give me some suggestion on how to setup so that I can do a 3-way-calling? |
| 15:24.33 | _Brian | HitTop: find two parties that want to talk to you :) |
| 15:24.41 | _Brian | HitTop: sorry..had to say it.. :) |
| 15:25.47 | HitTop | _Brian: ..... ???? |
| 15:26.41 | *** join/#asterisk Hmmhesays (~Hmmmhesay@66.173.103.108) |
| 15:27.25 | HitTop | for meetme.. |
| 15:28.20 | HitTop | its for dialing in right? i want 3 way for dialing out |
| 15:28.50 | Nugget | hehe |
| 15:29.13 | tzafrir | I set up an IAX account on FWD. I can connect to their echo test but I get "busy" to any other number I tried |
| 15:29.26 | tzafrir | (echo test: 613) |
| 15:31.14 | *** join/#asterisk Nix (~Nix@81.213.125.220) |
| 15:33.05 | tzafrir | on asterisk logs I see just "busy" |
| 15:33.07 | ariel_ | HitTop, meetme is for both you dial a party you want transfer them to your meetme room then call the other party and do the same. |
| 15:33.09 | tzafrir | asterisk 1.0.5 |
| 15:34.03 | tzafrir | HitPop, a call file to dial "from" a meetme channel? |
| 15:34.57 | ariel_ | tzafrir, it might be iaxtel I just tried them and could not connect to them. |
| 15:35.29 | tzafrir | iax.fwdnet.net == iaxtel ? |
| 15:35.32 | *** join/#asterisk eKo1 (~bernd@63.245.57.70) |
| 15:35.36 | heka | isn`t cvs server working? |
| 15:35.37 | heka | cvs [checkout aborted]: connect to cvs.digium.com(66.225.202.81):2401 failed: Connection timed out |
| 15:37.22 | tzafrir | anybody here connected to fwd to test connection to? |
| 15:37.25 | HitTop | tzafrir: does that mean i need to go into a conference room first, then dial out from there? |
| 15:37.50 | Nugget | heka: working fine for me. |
| 15:37.50 | jaiger | tzafrir: I think I am, what's your #? |
| 15:38.06 | tzafrir | my #: 611334 |
| 15:38.13 | heka | Nugget: its working now, thanks :) |
| 15:38.46 | tzafrir | I haven't yet set it up for receiving , though, I believe |
| 15:40.18 | *** join/#asterisk roamer323 (~sing@Toronto-HSE-ppp3702363.sympatico.ca) |
| 15:40.18 | *** part/#asterisk roamer323 (~sing@Toronto-HSE-ppp3702363.sympatico.ca) |
| 15:40.29 | TekLexus | anyone know how to set up an extension to auto-answer the sound port of the sound card for paging system? |
| 15:40.47 | *** join/#asterisk roamer323 (~sing@Toronto-HSE-ppp3702363.sympatico.ca) |
| 15:40.56 | TekLexus | anyone know how to set up an extension to auto-answer the sound port of the sound card for paging system? |
| 15:41.04 | TekLexus | whoops, sorry |
| 15:41.23 | Nugget | the red zone is for the immediate loading and unloading of passengers. there is to be no parking in the white zone. |
| 15:45.25 | jaiger | tzafrir: "call rejected" |
| 15:45.41 | tzafrir | editing setup... |
| 15:46.36 | roamer323 | anyone have any luck ringing phones on asterisk, located behind a symmetric NAT, from a SER (sipphone.com) ; ports forwarded, no outcalling problem through the same asterisk |
| 15:48.25 | Hmmhesays | i see there is a new phpagi version |
| 15:48.27 | Hmmhesays | cool |
| 15:50.02 | greg_work | there is? |
| 15:50.32 | *** join/#asterisk stickynomore (~jeff@nsc66.147.11-46.newsouth.net) |
| 15:51.33 | *** join/#asterisk cmslaght (~cmslaght@mail.techgroupintl.com) |
| 15:52.34 | cmslaght | Does anyone know how to turn on DTMF tones on a SNOM190 so that when you dial you can hear the numbers being dialed? |
| 15:54.07 | [ro]nic3try | is there a limitation for mp3 in asterisk ? |
| 15:54.52 | cmslaght | yes |
| 15:55.01 | [ro]nic3try | what |
| 15:56.13 | cmslaght | i am trying to look it up on the wiki. I know that it must be Consistant Bit Rate vs VBR |
| 15:56.23 | [ro]nic3try | i want to listen to bad_boys.mp3 and asterisk return pool time out |
| 15:57.15 | *** join/#asterisk outsidefactor (~blah@203-206-254-16.dyn.iinet.net.au) |
| 15:57.52 | ariel_ | [ro]nic3try, the problem is with the mp3 file. Asterisk uses mpg123 if it can play it correctly then asterisk can too. |
| 15:58.20 | [ro]nic3try | i have instaled the mpg123 from asterisk |
| 15:58.44 | [ro]nic3try | .. was this a worng thing to do ? |
| 15:59.26 | ariel_ | [ro]nic3try, when your at the linux prompt what version do you get when you put mpg123 in. |
| 15:59.59 | [ro]nic3try | 0.59q |
| 16:00.17 | ariel_ | [ro]nic3try, wrong version you need .59r |
| 16:00.36 | [ro]nic3try | :) ok thx |
| 16:00.39 | ariel_ | from the asterisk /usr/src/asterisk make mpg123 |
| 16:00.58 | ariel_ | then go to the mpg123 directory and make linux |
| 16:01.16 | tzafrir | what distro is it, BTW? |
| 16:02.22 | *** join/#asterisk jets (jetsn@omega.pmt.org) |
| 16:03.56 | roamer323 | is the iaxlib available separately somewhere or only as part of gnophone? TIA |
| 16:05.07 | coppice | if you want an up to date iaxlib look at iaxclient.sourceforge.net |
| 16:07.00 | roamer323 | thanks, coppice. I thought that is mainly for win32 developers. Will take a second look. |
| 16:08.05 | Damin | ~seen JerJer |
| 16:08.07 | jbot | jerjer is currently on #asterisk. Has said a total of 63 messages. Is idling for 11h 34m 47s |
| 16:08.17 | *** join/#asterisk SeaForth (~SeaForth@c-24-1-126-202.client.comcast.net) |
| 16:08.20 | Hmmhesays | heh |
| 16:11.38 | [ro]nic3try | i have the 0.59r version |
| 16:11.59 | jets | where oh where is athnm |
| 16:12.56 | nazgool | is it possible to define some kind of sip alias? e.g. that when SIP/myalias is called, it actually rings on SIP/joe&SIP/jim ? |
| 16:13.26 | tzafrir | nazgool, why not use dialplan for that? |
| 16:13.40 | stepcut | ok, I think my music on hold is not work because .... I don't have mpg123 installed ;) |
| 16:14.15 | stepcut | Is it ok to use mpg321, or do I need mpg123 ? |
| 16:14.32 | tzafrir | you need mpg123 |
| 16:14.32 | ariel_ | nazgool, you can make variables in the general section like sip2=sip/1&sip/2 etc and in the dial string just do exten => 100,1,Dial(${SIP2} |
| 16:14.37 | jaiger | stepcut: it's very picky, you need mpg123 |
| 16:14.45 | tzafrir | is this debian? you can get mpg123 from non-free |
| 16:14.51 | eKo1 | Speaking of music on hold, I moved all the music in mohmp3 to mohmp3/elevator-crap and made the corresponding changes to musiconhold.conf but I don't here anything anymore. |
| 16:15.25 | nazgool | ok thx |
| 16:15.36 | tzafrir | eKo1, do you get any errors on the command-line? |
| 16:15.42 | eKo1 | Nope. |
| 16:15.46 | cmslaght | Does anyone know how to turn on DTMF tones on a SNOM190 so that when you dial you can hear the numbers being dialed? |
| 16:15.48 | stepcut | jaiger: thanks! |
| 16:15.58 | stepcut | tzafrir: unfortunately, it's freebsd :) |
| 16:16.00 | mikayle | anyone know a trick to get chan_sccp to compile? I'm getting the callerid error |
| 16:16.18 | *** join/#asterisk guest_0034 (~jp@pD9559403.dip.t-dialin.net) |
| 16:17.02 | stepcut | sweet! hold music works now :) |
| 16:18.02 | eKo1 | stepcut: If you move your music to another directory, will it still work? |
| 16:19.04 | eKo1 | I have a feeling that var/lib/asterisk/mohmp3 is hardcoded somewhere. |
| 16:19.07 | stepcut | eKo1: yes -- but I also did a full restart of asterisk after updating the musiconhold.conf |
| 16:19.49 | stepcut | eKo1: on freebsd, the music is install in /usr/local/share/asterisk/mohmp3, so I had to update the config file... |
| 16:20.02 | anthm | shhh "moh reload" |
| 16:20.22 | guest_0034 | hi, Does somebody know where I find information about hardware requirements of asterisk? |
| 16:20.35 | *** part/#asterisk djin (~marius@62.58.40.196) |
| 16:20.41 | eKo1 | I've done all that and it still doesn't work. So odd. |
| 16:20.44 | roamer323 | anyone know what the big diff is between the misery demo server @ digium and the IAXnet server? I use same settings and the IAXnet incoming audio is always garbled. thx |
| 16:20.57 | eKo1 | anthm: eh, there is no moh reload |
| 16:21.06 | anthm | there is no spoon |
| 16:21.13 | eKo1 | At least not in stable. |
| 16:21.14 | ariel_ | matrix |
| 16:21.28 | anthm | there is no stable |
| 16:22.02 | ariel_ | morning anthm hope everything is going well. |
| 16:22.13 | anthm | so far |
| 16:22.14 | eKo1 | Hmm...maybe it's a bug. |
| 16:22.37 | [ro]nic3try | can i use any mp3 for musiconhold ? |
| 16:23.05 | *** join/#asterisk Godsey (lanny@207-229-102-125.cortland.com) |
| 16:23.07 | Essobi | How does one reorder the default codec attempting? |
| 16:23.19 | Godsey | voip-info is down? I'm a fish out of water :) |
| 16:23.22 | Essobi | I just got some G729 licenses. |
| 16:24.17 | Essobi | Godsey It's not down |
| 16:24.25 | Essobi | besides, use google and the google cache |
| 16:24.42 | eKo1 | Hmm...I see now, I have to have a default => ... in musiconhold.conf |
| 16:24.53 | eKo1 | I can't just use random => ... |
| 16:25.11 | Godsey | back up |
| 16:25.35 | *** join/#asterisk urkle (~urkle@12-202-97-178.client.insightBB.com) |
| 16:26.51 | *** part/#asterisk [ro]nic3try (~iancu@81.181.199.33) |
| 16:27.19 | eKo1 | Well, at least the CLI says "Started music on hold, class 'default',". |
| 16:27.28 | eKo1 | I don't hear any sound though. |
| 16:29.18 | *** join/#asterisk yogurt2ungue (~charlie@203-132-126-200.fibertel.com.ar) |
| 16:29.59 | _Brian | anyone here use Polycom ip500's? |
| 16:30.33 | Connor- | anyone know how to get 7905's to upgrade to sip from sccp? I'v done it 3 times now, but, I have 2 phones that I can't get it going. |
| 16:30.55 | ariel_ | eKo1, on the command prompt for linux what do you get when you do mpg123 |
| 16:31.56 | eKo1 | OK, I think I see what the problem is. Why didn't I read this before?! |
| 16:32.07 | _Brian | by default on most systems mpg123 is installed in /bin/ (/bin/mpg123), whereas the distro is installed in /usr/local/bin/mpg123. You should check which one you are using. (you would do a 'which mpg123' and check your path) |
| 16:32.16 | mikayle | Connor: most of the problems I had with cisco upgrades were fixed by using REALLY old firmwares and moving up one at a time |
| 16:32.51 | mikayle | also, on a lot of distrocs, mpg123 is linked to mpg321 which doesn't work with * |
| 16:32.55 | anthm | get CVS head and say bub bye to mpg123 |
| 16:33.02 | eKo1 | Eh, that didn't work. I've been cheated! |
| 16:33.34 | mtqh | what does cvshead use? |
| 16:33.47 | Essobi | Hey.. why does my cisco 7960 default to G711 when G729 is available? |
| 16:34.02 | *** join/#asterisk cjk (~cjk@80.92.75.91) |
| 16:34.18 | ariel_ | anthm, I don't use CVS Head except for testing. Stable is what I use mainly. |
| 16:34.22 | Essobi | connor: start with sip 5.2 |
| 16:34.26 | Essobi | then 6.2 |
| 16:34.30 | Essobi | then the latest 7 |
| 16:34.36 | Essobi | I just did one last night from sccp |
| 16:34.49 | _Brian | eKo1: do a which mpg123 and then a mpg123 and give us the version number you are using |
| 16:35.11 | Essobi | so it there no way to weight the codecs to decide to use G729 over G711? |
| 16:35.28 | *** join/#asterisk jesse_132 (~chatzilla@207.246.72.150) |
| 16:35.30 | anthm | again CVS head has codec preferences |
| 16:35.31 | eKo1 | Ah, I must do a SetMusicOnHold before the Dial command. |
| 16:35.33 | roamer323 | in a dial plan, when I call Dial() with "**" (two asterisks) - the sip proxy complaints that it needs numerical form for the double *, how do I do that? |
| 16:35.33 | Essobi | 1-2-3! You're outh! |
| 16:35.40 | mikayle | Ess: it is 'weighted' by the allow sequence in the config |
| 16:35.43 | eKo1 | What a pain. |
| 16:35.45 | jesse_132 | Is it possible to get a 1/2 PRI 1/2 data T1? |
| 16:35.57 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 16:36.00 | Essobi | mik In Sip.conf? It's not working like that. |
| 16:36.06 | mikayle | jesse: yes |
| 16:36.26 | jesse_132 | mikayle: cool... does it usually cost the same? |
| 16:36.34 | Essobi | jesse_132 And the cards all support it |
| 16:36.40 | Essobi | oh hell no they don't. |
| 16:36.43 | Essobi | Data T1 > Voice T1 |
| 16:36.51 | mikayle | jesse: the line costs the same, it's the services that cost more |
| 16:36.55 | cjk | hi, * does not seem to recognize authentication from my grandstream, but it works from the pstn network. had this anything to do with dtmf settings? |
| 16:36.58 | jaiger | any polycom users around that are willing to test something for me? |
| 16:37.01 | Essobi | got to pay for an ISP terimination |
| 16:37.14 | mikayle | you need an inteligent CSU/DSU that can split out the voice channels and data channels |
| 16:37.34 | jesse_132 | Essobi: so they asterisk does or doesn't support it... ("oh hell no they don't") |
| 16:38.04 | Essobi | that was inresponse to "does it usually cost the same" |
| 16:38.06 | mikayle | jesse: I think Ess meant that they don't cose the same |
| 16:38.12 | mikayle | what he said :) |
| 16:38.16 | jesse_132 | :) |
| 16:38.18 | Essobi | YAWN |
| 16:38.24 | Essobi | Codecs are aggravating me. |
| 16:38.43 | jesse_132 | actually, anyone setup in Louisville, KY? I just a few DIDs for origination |
| 16:38.45 | anthm | get CVS head and codecs will work like you want. |
| 16:39.04 | anthm | along with music, and most other things |
| 16:39.19 | mikayle | yes, but chan_sccp doesn't compile with new CVS |
| 16:39.33 | anthm | then your mission is clear ! |
| 16:39.40 | eKo1 | Hmm...I still don't here shit. This sucks. |
| 16:39.42 | mikayle | rewrite chan_scp? |
| 16:39.49 | mikayle | ;) |
| 16:40.01 | anthm | its probably just a few api calls' prototypes changed and the flags stuff |
| 16:40.12 | meebey | the socket handling of chan_sip is not clean btw |
| 16:40.17 | Essobi | anthm I can't run -head |
| 16:40.45 | *** join/#asterisk LUTOR_ASI (~newbie@icable-docsis2-108.ibw.com.ni) |
| 16:41.01 | Essobi | That's retarded that codec order get ignored in sip.conf. |
| 16:41.04 | LUTOR_ASI | HI guys, |
| 16:41.22 | LUTOR_ASI | I have a little problem, my asterisk box can not detect remote hangup ... |
| 16:41.51 | anthm | hense why I recoded the entire codec handling system and added it to , you guessed it, CVS HEAD ! |
| 16:41.58 | *** join/#asterisk dsfr (~dsfr@216.207.244.183) |
| 16:42.11 | Essobi | anthm Ahh FFS. |
| 16:42.22 | Essobi | I won't have a decent spandsp. :| |
| 16:42.25 | coppice | anthm: does it handle T.38 now? :-) |
| 16:42.34 | Essobi | Tehe. |
| 16:42.47 | darkskiez | Yay, I've added a reject incoming call feature to the cisco 79[46]0 :] |
| 16:43.09 | Essobi | I surprised you arn't gunning for that T38 bounty |
| 16:43.12 | Essobi | Coppice. |
| 16:43.18 | Essobi | darkskiez How's that? |
| 16:43.37 | darkskiez | Essobi, When its all finished i'll put it on the wiki |
| 16:43.45 | Essobi | Roger dodger. |
| 16:43.48 | coppice | the bounty is no motivation, but I am implementing T.38 |
| 16:43.51 | Essobi | Dirty dirty hack? |
| 16:43.52 | LUTOR_ASI | anyone that could help me with remote hang up problem...? |
| 16:44.06 | Essobi | coppice $8,000 is no motivation? |
| 16:44.08 | Essobi | Jees. |
| 16:44.08 | *** join/#asterisk mountainm2k (~freenodei@cbit-98.bullseye9.com) |
| 16:44.10 | darkskiez | Essobi, cisco services and ast manager integration. |
| 16:44.20 | coppice | 8000? last time I looked it was 3000 |
| 16:44.31 | darkskiez | holy crapsticls |
| 16:44.34 | Essobi | darkskiez Ahh, dirty dirty hack. I thought maybe you had a new softkey or something. |
| 16:44.45 | Godsey | I think I'll go crazy soon :) |
| 16:44.46 | junky[work] | wow 8000$! |
| 16:44.51 | darkskiez | Essobi, Pressing [Services] 7, is pretty nice though. |
| 16:44.55 | Godsey | I have polycom ip300s working great @ work |
| 16:45.07 | Godsey | trying to setup same system @ home and nada :) |
| 16:45.14 | Essobi | Ehh, drop it the patches somewhere and I'll look at setting up a softkey for it. |
| 16:45.21 | Essobi | Softkeys are hotshit. :) |
| 16:45.28 | jaiger | Godsey: do you use the directory with your polycoms? |
| 16:45.30 | darkskiez | Essobi, yeh, not on sip firmware though :( |
| 16:45.48 | Essobi | No softkey remapping on SIP? Egads. |
| 16:45.53 | jesse_132 | so, is voiceconduits back up yet??? haven't seen any news since 12/30 "scam" thread... (they aren't a scam for those who luckly missed the thread) |
| 16:46.30 | coppice | Essobi: wiki still says 3000 |
| 16:47.00 | Essobi | Ah, someone same 8K Friday.. BKW I think. |
| 16:47.13 | Godsey | jaiger: yes |
| 16:47.32 | Godsey | but my problem is my damn phone isn't registering |
| 16:47.45 | Godsey | infact, sip debug on asterisk shows the phone isn't talking to it |
| 16:47.52 | Godsey | but, if I dial 1000 and let it sit there |
| 16:47.57 | Godsey | it'll send a timeout to server |
| 16:47.59 | jaiger | Godsey: do you use the call forwarding feature of the phone? |
| 16:48.08 | Godsey | sometimes |
| 16:48.36 | Essobi | Haha, google cache still said 5000 but the page said 3000, someone retracted their offers for T38. |
| 16:48.37 | ManxPwr | ~docs |
| 16:48.38 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 16:48.42 | jaiger | Godsey: I have a problem related to directory and call forward with my IP500/IP600 phones |
| 16:49.00 | Essobi | coppice How good/bad is that new spandsp that build against head? |
| 16:49.01 | Godsey | when I forward, I enter the # I want to forward to :) |
| 16:49.05 | _Brian | anyone know if itis possible to disable the call forwarding feature on the polycom ip500 (from the ipmid.cfg)? |
| 16:49.45 | jaiger | Godsey: right, but callers using the directory to call a forwarded extension get a different voicemail from direct dialing the same extension |
| 16:49.57 | Godsey | oh |
| 16:50.38 | jaiger | I'm wondering if other's have noticed that, if it's a bug or if I can work around it |
| 16:51.38 | jaiger | Godsey: I guess you haven't noticed that behavior? |
| 16:51.57 | Godsey | nope |
| 16:52.02 | *** join/#asterisk blitzrage (~blitzrage@blitzrage.documenter.extraordinaire.of.asterisk) |
| 16:52.13 | Godsey | since they never go to voicemail on asterisk if I forward :) |
| 16:52.32 | jaiger | Godsey: you never step away from the desk? |
| 16:52.41 | *** part/#asterisk blitzrage (~blitzrage@blitzrage.documenter.extraordinaire.of.asterisk) |
| 16:52.46 | Godsey | if it's forwarded it goes out of system |
| 16:52.52 | jaiger | ahh |
| 16:53.27 | coppice | I wonder how many bounties actually get paid out |
| 16:53.34 | Essobi | Heh. |
| 16:53.37 | Essobi | Who knows. |
| 16:55.12 | *** join/#asterisk felix (~felix@200.115.235.79) |
| 16:55.25 | *** join/#asterisk MEPHiST0 (mephisto@ccitt5.de) |
| 16:55.51 | *** join/#asterisk blitzrage (~blitzrage@blitzrage.documenter.extraordinaire.of.asterisk) |
| 16:55.55 | MEPHiST0 | 'lo |
| 16:56.22 | *** part/#asterisk blitzrage (~blitzrage@blitzrage.documenter.extraordinaire.of.asterisk) |
| 16:57.34 | mikayle | anyone using SCCP with new head, if so, which SCCP are you using? |
| 16:58.22 | hardwire | SCCP! |
| 16:58.27 | jesse_132 | SCCP? |
| 16:58.31 | jesse_132 | it works? |
| 16:58.33 | hardwire | !!!!! |
| 16:58.39 | mikayle | hey...be nice |
| 16:58.42 | *** join/#asterisk Darwin35 (~Darin@c-24-3-241-22.client.comcast.net) |
| 16:58.50 | mikayle | I got a cisco IP conf phone and want to make it work |
| 16:59.04 | mikayle | and chan_sccp doesn't compile anymore |
| 16:59.04 | hardwire | flash it to sip :) |
| 16:59.07 | mikayle | can't |
| 16:59.10 | hardwire | doh |
| 16:59.12 | mikayle | no sip for that model |
| 16:59.17 | hardwire | just send it to me then. |
| 16:59.20 | hardwire | :) |
| 16:59.44 | hardwire | not that I have any experience there |
| 16:59.55 | mikayle | I probably should of gotten an ATA unit and an analog polycom |
| 17:00.11 | MEPHiST0 | mmh one thing maybe s/o here can help me |
| 17:00.22 | MEPHiST0 | i'm planning to fully switch to SIP |
| 17:00.44 | MEPHiST0 | is asterisk capable of simulating an NTBA, so i can use my ISDN-TK thing in the future? |
| 17:01.04 | MEPHiST0 | if, what hardware is required? |
| 17:01.09 | hardwire | NTBA? |
| 17:01.14 | MEPHiST0 | just an ordinary ISDN Card? |
| 17:01.28 | TekLexus | what do u guys think about the sipura 841? I was going to go with a cisco, but this phone has all the features i want, and much less expensive... except it uses a AC adapter... |
| 17:01.29 | MEPHiST0 | hardwire: yep, ntba, it's commonly called that way in germany |
| 17:01.49 | MEPHiST0 | the isdn S0 Terminator |
| 17:02.14 | MEPHiST0 | (which comes with isdn lines on pstn's) |
| 17:02.42 | hardwire | funky |
| 17:03.28 | MEPHiST0 | current wring is: <pstn>-<ntba>-<isdn TK>-<analog phones> |
| 17:03.54 | MEPHiST0 | i want: <inet-router>-<asterisk>-<isdn TK>-<analog phones> |
| 17:04.21 | mikayle | why put the isdn-tk in the loop? |
| 17:04.36 | MEPHiST0 | to keep the equipment + wiring |
| 17:04.48 | mikayle | <rtr>-<asterisk>-<analog phones> |
| 17:04.49 | jets | anthmmmmmm |
| 17:05.14 | MEPHiST0 | mmh. how to connect analog phones to a linux machine running asterisk? |
| 17:05.22 | mikayle | www.digium.com |
| 17:05.34 | mikayle | pci card |
| 17:05.38 | mikayle | how many lines? |
| 17:05.48 | MEPHiST0 | 3-4 |
| 17:06.16 | hardwire | digiuiuium |
| 17:06.16 | mikayle | TDM400P |
| 17:06.25 | MEPHiST0 | so POTS is analog? |
| 17:06.37 | MEPHiST0 | or what does POTS mean? |
| 17:06.50 | Delvar | Plain Old Telephone System |
| 17:06.54 | MEPHiST0 | ah ic |
| 17:06.59 | mikayle | TDM40B |
| 17:07.06 | MEPHiST0 | mmh, that would suite my needs |
| 17:07.15 | MEPHiST0 | i guess analog faxes won't work via sip? |
| 17:07.24 | mikayle | there are wiki on faxing |
| 17:07.32 | MEPHiST0 | ah ty |
| 17:07.33 | mikayle | inbound or outbound? |
| 17:07.59 | MEPHiST0 | both |
| 17:08.08 | MEPHiST0 | but inbound is most important |
| 17:08.19 | mikayle | neither should be an issue... |
| 17:08.30 | MEPHiST0 | i need a solution for my homeoffice and i'm sick & tired of pstn mafia in germany |
| 17:08.33 | mikayle | outbound, just plug fax machine into TDM card |
| 17:09.29 | MEPHiST0 | so the quality over my sip gateway will be enuff? |
| 17:09.43 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 17:09.47 | mikayle | should be, if you have teh bandwidth |
| 17:09.54 | MEPHiST0 | 2.3mbit/s sdsl |
| 17:10.04 | *** join/#asterisk tclark (~TC@S0106000c413a1c61.gv.shawcable.net) |
| 17:10.14 | mikayle | :) |
| 17:10.14 | *** join/#asterisk pcm (~pcm@user-69-73-0-22.knology.net) |
| 17:10.31 | mikayle | I need to call my dsl company.... |
| 17:10.37 | mikayle | I ordered 5Mb/1Mb |
| 17:10.44 | mikayle | I get 400Kb/300Kb |
| 17:10.49 | mikayle | not too happy about that |
| 17:10.54 | MEPHiST0 | aehm |
| 17:11.09 | MEPHiST0 | you ordered 5mbit/s down, 1mbit/s up |
| 17:11.16 | mikayle | yep, ADSL |
| 17:11.22 | MEPHiST0 | and you get 400kb transferrate incoming and 300kb outgoiing? |
| 17:11.35 | mikayle | I think the line is too noisey... |
| 17:11.46 | MEPHiST0 | sounds like overconcentration |
| 17:11.53 | mikayle | took then 2 weeks to figure out they forgot to terminate my wires into the dsl head unit in the colo |
| 17:12.04 | MEPHiST0 | omg |
| 17:12.07 | MEPHiST0 | what provider? |
| 17:12.13 | mikayle | Verizon |
| 17:12.19 | MEPHiST0 | btw, what is FXO/FXS? |
| 17:12.31 | mikayle | http://www.digium.com/index.php?menu=fxsvfxo |
| 17:12.38 | *** join/#asterisk ionix (~ioni@66.38.219.151) |
| 17:12.40 | ionix | Heya |
| 17:12.47 | ionix | anyone has a way to make ANI reverse lookup & |
| 17:12.54 | ionix | like automatically with asterisk |
| 17:13.01 | mikayle | bbl, need food.... |
| 17:13.02 | ionix | i.e fast queries or download database |
| 17:13.15 | MEPHiST0 | mika: have fun and ty for info |
| 17:13.52 | mikayle | np |
| 17:14.32 | *** join/#asterisk Dabba (dabba@matrix.lgw.ip6net.net) |
| 17:18.15 | PTG123 | ionix you would need access to a database, you have that/ |
| 17:18.21 | *** join/#asterisk algorithmn (~algorithm@ool-18bce89c.dyn.optonline.net) |
| 17:18.35 | *** join/#asterisk bsd3 (~bsd@203.145.128.5) |
| 17:21.52 | nitram | w0000t |
| 17:21.59 | Dabba | Hi all, could anyone spare a moment to look at this problem with me, im trying to get sms & voice to work on the same msn, but asterisk seems to ignore the order i do things even thoughi did include => |
| 17:22.01 | Dabba | http://pastebin.ca/5808 |
| 17:22.05 | nitram | nt-mode works with eicon cards :) |
| 17:22.40 | jets | anthmmmmmm |
| 17:24.54 | nitram | so now i have an octobri and a diva server 4bri in nt-mode :) |
| 17:24.57 | nitram | sweet |
| 17:25.39 | thieumS | hello |
| 17:26.20 | thieumS | did anyone manage to build oh323 module on debian ? |
| 17:30.52 | sudhir492 | Anyone using READ? |
| 17:31.07 | sudhir492 | whne I try exten => s,1,Read(PHNUM,vm-enter-num-to-call,5) |
| 17:31.15 | sudhir492 | exten => s,2,NoOp(PHNUM is ${CALLERIDNUM}) |
| 17:31.32 | sudhir492 | sorry |
| 17:31.43 | Essobi | Mmm. |
| 17:31.56 | sudhir492 | exten => s,2,NoOp(PHNUM is ${PHNUM}) |
| 17:32.08 | Essobi | My caller ID isn't passing to my 5400's.. Anyone got any ideas before I go googling? |
| 17:32.10 | sudhir492 | I see that PHNUM is always blank |
| 17:34.21 | Essobi | thieumS What're you using oh323 with? |
| 17:34.25 | jets | Hey bitches |
| 17:34.47 | Essobi | Sup hooker |
| 17:35.06 | jets | You. |
| 17:35.08 | jets | Or me actually. |
| 17:35.47 | *** join/#asterisk debianUSR (~debiandud@203.115.185.216) |
| 17:35.55 | hardwire | doobie doobie doobie doo! |
| 17:39.16 | *** join/#asterisk dano_ (~bismantux@www.crosscountrycourier.com) |
| 17:39.30 | thieumS | Essobi: lib provided by openh323.org |
| 17:39.48 | thieumS | and oh323 provided by * /h323 directory |
| 17:40.12 | dano_ | What would experienced users of Asterisk recommend for a small (10 person) call center as far as hard phones go? |
| 17:40.23 | thieumS | i've got an error during the wrapper compile |
| 17:41.23 | Essobi | thieumS I meant enpoints.. what're you hooking * to via H323 |
| 17:44.19 | thieumS | Essobi: i'm not here yet, i just want to compile oh323 :p |
| 17:45.00 | *** join/#asterisk jjg (tink@216.253.86.223) |
| 17:45.30 | jjg | any got any recommendations for DID providers for continental US? |
| 17:46.23 | jjg | is anyone aware of a web page that shows DID providers? |
| 17:46.42 | Essobi | Well |
| 17:46.45 | Essobi | Let me put it this way |
| 17:46.56 | Essobi | the two h323 drivers are a sin. |
| 17:47.02 | Essobi | One has no scalability |
| 17:47.25 | Nix | both have threading bugs |
| 17:47.31 | Essobi | the other.. well crashed my 5400's with a bufer over flow. Granted the cisco's shouldn't have allowed any h323 to overflow but.. |
| 17:47.43 | Essobi | It's a sin either way you go |
| 17:47.58 | Essobi | get the SIP images on your ciscos and be happy. |
| 17:48.59 | *** join/#asterisk dsfr (~dsfr@216.207.244.183) |
| 17:50.28 | Essobi | Damnit |
| 17:50.36 | Essobi | My callerID isn't getting presented |
| 17:50.37 | Essobi | :\ |
| 18:01.25 | *** join/#asterisk r1 (~erwan@www.thiscow.com) |
| 18:01.33 | thieumS | ok Essobi :p |
| 18:01.58 | thieumS | i don't like h323 but some people are using it again |
| 18:02.11 | thieumS | still use it |
| 18:02.58 | jesse_132 | is the IAXy still the best remote extension device (that handles firewalls, NAT, ...) |
| 18:03.19 | Nix | erm.. |
| 18:03.21 | algorithmn | iax is the most nat friendly voip protocol |
| 18:03.22 | Nix | define best |
| 18:03.34 | Nix | its very good if you have firewalls and high bandwidth |
| 18:03.40 | algorithmn | i'm waiting for a 2 rj-11 port iaxy |
| 18:03.44 | jesse_132 | Nix: well, it is really the only one I know... you can define it in anyway you want... |
| 18:03.51 | Nix | not so good if you have low bandwidth links (ie. will not work) |
| 18:03.55 | jesse_132 | Nix: I thought iax was better than sip wrt bandwidth |
| 18:04.02 | Nix | not much |
| 18:04.03 | algorithmn | both use rtp |
| 18:04.04 | bsd3 | hi, friends! howdy.323 show codecs, shows none of the codecs; i even tried 'enable-all' |
| 18:04.05 | jesse_132 | Nix: can you define low bandwidth |
| 18:04.06 | bsd3 | sorry, 'enable=all' |
| 18:04.24 | Nix | and the way libiax and asterisk does framing it often uses more bandwidth than sip or h323 |
| 18:04.26 | jesse_132 | Nix: like cable/dsl? |
| 18:05.00 | Nix | minimum bandwidth for iaxy is about 45k each direction... |
| 18:05.22 | jesse_132 | Nix: shite... ok... those are not an option then... |
| 18:05.29 | Nix | with sipura or any other device using g723 you can get down to about 8-9k |
| 18:05.37 | Nix | or 10-12k with g729 |
| 18:05.44 | *** join/#asterisk dsmouse (~mouse@rrcs-24-199-146-243.midsouth.biz.rr.com) |
| 18:05.48 | dsmouse | asterisk++ |
| 18:06.02 | Nix | asterisk is not designed for low bandwidth situations.. |
| 18:06.15 | Nix | it puts one frame per packet |
| 18:06.20 | *** join/#asterisk dsfr (~dsfr@216.207.244.183) |
| 18:06.23 | *** join/#asterisk lung (~lung@24-148-96-186.ip.mhcable.com) |
| 18:06.35 | algorithmn | any sbc t3 card users?? |
| 18:06.38 | Nix | which makes it use 40-50% more bandwidth than most sip or h323 implimentations with the same codec |
| 18:06.47 | jesse_132 | Nix: thanks... I was about to order 20 IAXy's ... ;) |
| 18:07.06 | Nix | IAXys are nice for a LAN though.. depends on your usage situation |
| 18:07.44 | jesse_132 | Nix: it will be a mix... I'm gonna pick up a few IAXys and a few SPA2000s |
| 18:07.55 | *** part/#asterisk mikayle (~talamar@akro-pool2-cs1.pool.dslohio.net) |
| 18:08.00 | Nix | ok. |
| 18:08.28 | Nix | I wouldnt want to leave them unprotected on the internet either.. anyone can just overwrite their config.. no security at all |
| 18:08.48 | jesse_132 | Nix: whole shite |
| 18:08.54 | jesse_132 | Nix: you mean the spa2000s? |
| 18:08.55 | Nix | ;-) |
| 18:08.58 | Nix | no |
| 18:08.59 | Nix | IAXy |
| 18:09.01 | jesse_132 | ahh |
| 18:09.08 | jesse_132 | are the SPAs ok? |
| 18:09.10 | Nix | sipura is a nice machine |
| 18:09.11 | Nix | yes |
| 18:09.21 | JerJer | Nix: you have no clue |
| 18:09.29 | Nix | you can lock the SPAs as well if you want... stop your customers fooling with them |
| 18:09.36 | Nix | JefJer nice to see you again ;-) |
| 18:09.41 | algorithmn | Nix: i wasn't happy with the 2100 firmware |
| 18:09.58 | algorithmn | was gonna order a 2000 to confirm my suspision |
| 18:10.26 | Nix | algorithmn: yeah.. there have been issues with various SPAs.. like any device |
| 18:10.35 | JerJer | Nix show me how you plan on configuring an IAXy from the internet |
| 18:10.46 | JerJer | without the iax username and secret |
| 18:10.58 | Nix | I have over 20 different SIP and h323 devices though.. the SPAs are the nicest ones I have seen so far |
| 18:11.20 | algorithmn | have you toyed with the 2k n 2100? |
| 18:11.59 | algorithmn | 2100!=https |
| 18:12.07 | Nix | JerJer: you can provision the IAXy without a password |
| 18:12.20 | Nix | algorithmn: nope. sorry |
| 18:12.43 | Nivex | Nix: interesting. I've had nothing but issues with my spa-2100. |
| 18:12.55 | Nix | really? |
| 18:12.57 | algorithmn | don't worry, i'm just situating this dilemma in front of me |
| 18:13.02 | JerJer | 3k is decent |
| 18:13.05 | JerJer | only if it rapped iax |
| 18:13.15 | algorithmn | word |
| 18:13.55 | Nivex | erm excuse me spa-2000 |
| 18:14.01 | JerJer | but i have noticed that if there is no caller*id the spa3k's won't setup a sip call |
| 18:14.04 | Nivex | soo many model numbers to keep straight :) |
| 18:14.06 | Nix | VoIP protocol is mostly irrevalent as far as I am concerned.. there can be bugs in any device |
| 18:14.23 | JerJer | NIx: don't change the topic |
| 18:14.34 | algorithmn | have fun gang. on with more programming, situating, simulating, optimizing and propogating.. |
| 18:14.46 | JerJer | explain your statement |
| 18:15.07 | Nix | JerJer: which statement? |
| 18:15.33 | JerJer | (13:08:41) Nix: I wouldnt want to leave them unprotected on the internet either.. anyone can just overwrite their config.. no security at all |
| 18:15.39 | *** join/#asterisk Mneumonic (Mnemonic@ool-18ba58b4.dyn.optonline.net) |
| 18:15.48 | Nix | yep |
| 18:15.57 | JerJer | bullshit |
| 18:16.02 | JerJer | pure bullshit |
| 18:16.07 | Mneumonic | Anyone know why my SoftPhone would say "Call Not Approved"? It suddenly started happening yesterday night |
| 18:16.15 | Mneumonic | cant even call voicemail |
| 18:16.29 | Nix | JerJer: tell me the provisioning procedure then? |
| 18:16.45 | *** join/#asterisk jdims (~sleepbsd@24-119-121-6.cpe.cableone.net) |
| 18:16.58 | Nix | left time I provisioned my IAXy it didn't need a password.. you just run a program with the IP address of the IAXy |
| 18:17.03 | Nix | lef=last |
| 18:17.36 | Nix | that may have changed, but my IAXy blew up and I have't bothered to buy another due to its high bandwidth requirements and complete lack of fax support... |
| 18:17.55 | JerJer | you really have no clue |
| 18:18.02 | JerJer | i fax thru mine quite often |
| 18:18.11 | Nix | oh yeah? |
| 18:18.14 | JerJer | and the very first time you provision them you use a broadcast packet |
| 18:18.19 | Nix | with what codec? |
| 18:18.27 | Nix | not T.38 I assume :-) |
| 18:18.29 | JerJer | explain to me how you are going to get a broadcast packet in via the WAN interface |
| 18:18.38 | *** join/#asterisk r1 (~erwan@www.thiscow.com) |
| 18:18.51 | Nix | JerJer: you have never heard of directed broadcast I assume? |
| 18:19.17 | JerJer | funny |
| 18:19.36 | JerJer | if you are so smart why don't you disclose it to one of the many agencies out there |
| 18:19.44 | JerJer | or even digium then |
| 18:20.08 | algorithmn | proper security vulnerability disclosure is a crucial step in finding a security vulnerability |
| 18:20.53 | algorithmn | some can slant it as a matter of national security and depending on the vendor, will press civil charges |
| 18:21.32 | algorithmn | there recently has been a consortium of voip vendors to research and protect voip protocols |
| 18:21.36 | algorithmn | report it to them |
| 18:21.52 | Nix | JerJer: I could.. dont have the time or inclination |
| 18:22.10 | JerJer | bullshit |
| 18:22.15 | algorithmn | tell me.. i'll do it |
| 18:22.28 | algorithmn | it will take no added time to you |
| 18:22.30 | JerJer | algorithmn: he's here spreading FUD |
| 18:22.35 | JerJer | again |
| 18:22.43 | algorithmn | i've been through the disclosure process before |
| 18:22.57 | algorithmn | it's like riding a tricicle to me right now |
| 18:23.26 | *** join/#asterisk jarrod (jarrod@dipole.informationwave.net) |
| 18:23.36 | algorithmn | JerJer: technically, it is possable that he is being truthful |
| 18:23.44 | jarrod | hey is there a way to accept a call and have it turn around and call another number and connect them? |
| 18:23.52 | JerJer | algorithmn: no |
| 18:24.29 | `Sauron | Damn, I can't remember the question I had for JerJer now. |
| 18:24.30 | algorithmn | sequence numbers->tcp/udp session might not check for proper ip address |
| 18:24.30 | `Sauron | Sigh. |
| 18:24.33 | jarrod | like exten => 1117771111,1,Dial(${TRUNK},111111111) |
| 18:24.53 | algorithmn | or just spoof udp or blind tcp connection |
| 18:25.16 | JerJer | using a broadcast packet |
| 18:25.29 | Nix | algorithmn: exactly |
| 18:25.32 | algorithmn | unicast.. im not sure about broadcast |
| 18:25.36 | JerJer | bingo |
| 18:25.42 | ManxPwr | jarrod, That has got to be the worst Dial(.. line I've seen in a very long time! |
| 18:25.49 | jjg | does anyone know if you can get something like a DID, but totally IP based? |
| 18:25.54 | jarrod | im just doing it as a quick example |
| 18:26.03 | `Sauron | jjg: FWD |
| 18:26.10 | Nugget | jjg: set up a SRV record for your domain. |
| 18:26.17 | `Sauron | or that |
| 18:26.20 | jarrod | im wondering how to accomplish this goal |
| 18:26.23 | Nugget | then take calls at sip:extension@domain |
| 18:26.24 | Nix | JerJer: what reason would I have to spread FUD anyway? |
| 18:26.26 | ManxPwr | jarrod, Well I've never seen "1111111111" as a dial option. |
| 18:26.29 | Essobi | Hey ManxPwr .. My 5400 isn't dropping a CID to the PSTN.. any ideas where to start debugging it at? |
| 18:26.33 | jjg | I need to read on SRV |
| 18:26.35 | ManxPwr | Usually 1111111 would be a number or a timeout. |
| 18:26.39 | Nugget | or, alternately, register your phone numbers with e164.org |
| 18:26.47 | Nix | I notice you didn't disagree with my statement about bandwidth.. or fax support.. |
| 18:26.50 | ManxPwr | Essobi, You mean like the mailing list archive? |
| 18:26.54 | Nugget | and encourage people to add e164.org enum lookups to their dial plan |
| 18:26.54 | Nix | ;-) |
| 18:27.03 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 18:27.03 | jarrod | manxpwr: its not an option, its the number to be called |
| 18:27.04 | algorithmn | JerJer: maybe you should ask him what sequence number prediction can yield.. |
| 18:27.04 | jets | You. |
| 18:27.20 | JerJer | nix: um what about the part i fax thru my iaxy all the time |
| 18:27.31 | jarrod | and its not the actualy number, its just something i did real quick |
| 18:27.34 | ManxPwr | jarrod, the comma makes it an option. Perhaps you meant exten => 1117771111,1,Dial(${TRUNK}/111111111) |
| 18:27.34 | jarrod | man you are thick |
| 18:27.42 | Nix | JerJer: yep.. you didnt answer me with the codec though did you :-) |
| 18:27.54 | meebey | can I restart a chah? |
| 18:27.55 | jarrod | ah thats what i did in the actual conf file |
| 18:27.56 | meebey | chan even |
| 18:28.15 | meebey | I have a dynamic IP and the externip value is only resolved on startup |
| 18:28.15 | Essobi | ManxPwr lol |
| 18:28.20 | Essobi | :P |
| 18:28.21 | Nix | Alaw or ulaw I assume.. which is pretty useless internationally.. |
| 18:28.45 | JerJer | you can ONLY fax using u or a law |
| 18:28.48 | ManxPwr | Essobi, I wasn't being funny. CLID and DID issues with Ciscos are have been discussed on the mailing list. |
| 18:28.51 | JerJer | no questions asked |
| 18:28.52 | jets | Nix: You originally stated you can't fax. |
| 18:29.10 | Essobi | Roger Roger. |
| 18:29.36 | *** join/#asterisk eivindtr (~Eivind@062016241059.customer.alfanett.no) |
| 18:29.37 | nestAr | :sigh: |
| 18:29.40 | Nix | I said "complete lack of fax support" which is true |
| 18:29.49 | JerJer | and I have just proven you wrong |
| 18:29.54 | jets | It's not completely lacking, as JerJer can fax. |
| 18:29.57 | ManxPwr | jets, Faxing over VoIP is a lot like going bungee jumping with an old frayed bungee rope. It will work most of the time, but even eventually it will break and then you go splat |
| 18:29.57 | nestAr | owner of the company wants to sell pbx's now that we have an * box. :sigh: |
| 18:30.12 | nestAr | maintenance, service, etc.. |
| 18:30.16 | Nix | not that "you MAY be able to fax through it using ulaw if you are lucky and have enough bandwidth and a low enough latency and your providor supports it" |
| 18:30.18 | Nix | :-) |
| 18:30.37 | jets | Let's bitch in #asterisk-whine and support people in #asterisk!! |
| 18:30.45 | JerJer | jets: right on |
| 18:30.54 | JerJer | don't like the IAXy, don't buy one |
| 18:31.46 | Nix | ManxPwr: exactly |
| 18:32.25 | JerJer | notice I didn't say I was faxing over the internet either |
| 18:32.50 | JerJer | I fax to the local asterisk box using rxfax then txfax it out a real PRI |
| 18:32.51 | JerJer | problem solved |
| 18:32.52 | ManxPwr | JerJer, We all know you are not *quite* crazy enough to fax over IP. |
| 18:32.53 | JerJer | now shutup |
| 18:33.03 | jets | *laugh* |
| 18:33.21 | jets | So who wants to poke anthm till he helps me with chanspy? |
| 18:33.28 | *** join/#asterisk jayden (~ircatjerr@65.170.43.34) |
| 18:33.34 | jayden | hello all... |
| 18:33.44 | jets | sshhh don't tell harley |
| 18:33.47 | jayden | I need card recomendations, T100p vs TE100p |
| 18:34.09 | *** join/#asterisk pelotas (~alek@host-ip130-216.crowley.pl) |
| 18:34.11 | JerJer | TE410P or TE405P |
| 18:34.22 | JerJer | accept nothing less |
| 18:34.28 | ManxPwr | jayden, If you look on the Digium Web site you will see that the T100P is no longer for sale. |
| 18:34.43 | jayden | y, they are all over the place still... are there issues w/ those? |
| 18:34.46 | jets | Do we stil lhave development meetme's? |
| 18:35.06 | jayden | and is there non scalability advatages to the 4 port models, or just hte ports? |
| 18:35.12 | ManxPwr | jayden, They were replaced with the TE110P I believe. |
| 18:35.42 | jayden | y, it looks to be so, there are still a bunch of them at resellers for about $100 less than the 110's |
| 18:36.12 | JerJer | why be cheap? |
| 18:36.14 | Logan | How can I have a user input something like a password with DTMF (possibly using something like # to terminate the string)? |
| 18:36.23 | *** part/#asterisk bsd3 (~bsd@203.145.128.5) |
| 18:36.28 | ionix | read() |
| 18:36.53 | jayden | I have a need for a single PRI installation, integrating to a Nortel Meridian swithch, unlikely to grow... the question is, is there any good reason not to go w/ the 100 if I can get one for sub $500 |
| 18:37.13 | Logan | A pattern like _.# doesn't seem to work. |
| 18:37.16 | JerJer | old chipset |
| 18:37.23 | jayden | jets is getting kinky.... |
| 18:37.49 | jets | *laugh* ya i'm trying to get his attention. i'm needy. |
| 18:37.50 | dsmouse | Logan: wouldn't that match just a 1 digit password? |
| 18:38.10 | Logan | dsmouse: The . means one or more digit. |
| 18:38.16 | Logan | What I had before was _#. |
| 18:38.17 | dsmouse | oh. |
| 18:38.21 | *** join/#asterisk tzanger (~tzanger@165.154.13.35) |
| 18:38.30 | Logan | Which worked, but you had to wait several seconds after entering the password before it could tell that you were done. |
| 18:38.46 | Logan | I want a way for the user to immediately indicate that he's done, without forcing the password length to be anything in particular. |
| 18:39.14 | jayden | ok.. so 100p it is.. .anybody have a good source for these in the US? |
| 18:39.17 | jayden | oops |
| 18:39.21 | jayden | 110 that is |
| 18:39.21 | Logan | Maybe there's a better way to collect digits. |
| 18:39.51 | *** join/#asterisk cnj (~matt@host-24-225-150-239.patmedia.net) |
| 18:40.15 | slePP | Logan: Read()? |
| 18:40.32 | Logan | slePP: Looks perfect, thanks. |
| 18:42.04 | ManxPwr | Logan, No _.# will never work. |
| 18:42.21 | ManxPwr | Logan, "show application read" |
| 18:42.22 | tzanger | whee |
| 18:42.28 | tzanger | UPSes only lhast about 1/2hr |
| 18:42.46 | *** join/#asterisk Ayano (~erik_leee@209.143.187.254) |
| 18:43.16 | jayden | UPS's with lots of batteries last longer.... |
| 18:43.28 | tzanger | jayden: well a 1/2h outage is already more or less unheard of |
| 18:43.38 | jayden | or UPS, that have nothing plugged into them |
| 18:43.40 | jayden | :) |
| 18:44.23 | jayden | tell that to the better part of the east coast and midwest about a year or so ago when the whole grid dropped for a couple days |
| 18:44.59 | dsmouse | jayden: or .ca.us 2 years ago. |
| 18:45.00 | tzanger | does zapata's echo cancel work with PRI? I was thinking about that on the way back into the office and I don't think it does |
| 18:45.03 | algorithmn | any1 know y when system '/usr/sbin/asterisk -rx 'reload' in perl agi threads a process, it hangs? |
| 18:45.04 | dsmouse | "roll" |
| 18:45.10 | ManxPwr | tzanger, Yes. |
| 18:45.17 | tzanger | ManxPwr: it does? |
| 18:45.18 | jayden | generators with big tanks last a long time too :) |
| 18:45.28 | ManxPwr | tzanger, Yes, it does. |
| 18:45.29 | tzanger | jayden: well yes, but as I said, 0.5hr is already rare |
| 18:45.40 | *** join/#asterisk Meznev (~Elshar@ip205-68.oregonfast.net) |
| 18:45.55 | ManxPwr | We use echocancel=yes and exhotraining=600 (or is it 800, I don't recall) with our PRI. |
| 18:45.57 | tzanger | ManxPwr: hmm okay... echotraining too? i.e. it won't send the pulse until after the B chan call has been establisthed? |
| 18:46.06 | tzanger | ManxPwr: I get no echo cancel happiness on my PRI :-( |
| 18:46.06 | jayden | true... |
| 18:46.31 | ManxPwr | tzanger, Of course, any changes to zapata.conf requires a STOP and a START of Asterisk |
| 18:47.00 | ManxPwr | tzanger, turn off echo cancel and see if the echo is worse. |
| 18:47.06 | tzanger | ManxPwr: yes I know |
| 18:47.10 | tzanger | and no it's not really any worse |
| 18:47.14 | tzanger | dmesg isn't showing it cancelled |
| 18:47.18 | tzanger | by fax tone |
| 18:47.28 | ManxPwr | tzanger, fax tone disables echocan. |
| 18:47.34 | Logan | slePP: When I use Read, after it completes, it attempts to resolev what the user entered as an extension. |
| 18:47.41 | tzanger | ManxPwr: yes I know |
| 18:47.46 | tzanger | I said I am *not* seeing that in dmesg |
| 18:47.51 | ManxPwr | Ah. |
| 18:48.04 | ManxPwr | Maybe you have a typoe or something else in the config file. |
| 18:48.15 | jayden | a typoe :) |
| 18:48.22 | tzanger | :-) |
| 18:48.28 | tzanger | I don't think so but I'll double/triple check |
| 18:50.20 | tzanger | ManxPwr: |
| 18:50.20 | tzanger | Echo Cancellation: 32 taps, currently ON |
| 18:50.23 | tzanger | it's seeing my config |
| 18:51.03 | ManxPwr | tzanger, dunno what's going on then |
| 18:51.25 | tzanger | Bell Canada just sucks :-) |
| 18:51.30 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 18:52.16 | tzanger | ManxPwr: what do you use for taps? |
| 18:52.20 | tzanger | I've tried 256, 128, 64 and 32 |
| 18:52.31 | tzanger | and echotraining to yes and 400, although all that does is speed up the echo cancel |
| 18:52.46 | tzanger | the echo cancel never seems to settle anyway so speeding it up shouldn't affect anything :-) |
| 18:53.30 | _Brian | anyone know the maximum number of arguements that can be passed to a AGI from * ? |
| 18:54.29 | jayden | hey, BTW, is that SwitchVox that was on asterisk daily news open source? |
| 18:56.47 | _Brian | bueller? bueller? |
| 18:56.51 | slePP | Logan: Read shouldn't. |
| 18:57.19 | slePP | Logan: Read(VARNAME,,8) |
| 18:57.20 | Logan | slePP: I figured out the problem. Now I have a new one. |
| 18:57.33 | Logan | slePP: Is it possible to have a special digit (like *) break out of the Read? |
| 18:57.38 | *** part/#asterisk jjg (tink@216.253.86.223) |
| 18:57.45 | slePP | i think its hardcoded to # |
| 18:57.45 | nitram | anyone? anyone? ;) |
| 18:58.25 | Logan | slePP: In our callflow, we have a prompt for a password, or to press * to enter a different extension to login with. Maybe we'll just change it to use #, and treat the empty "password" as special. |
| 18:58.31 | _Brian | :) (i thought it was a easy question :) |
| 18:59.02 | slePP | Logan: yeh, i think it would take a littl source change to accept * instead of # |
| 18:59.54 | *** join/#asterisk stepcut (~user@207.67.194.2) |
| 19:00.33 | *** join/#asterisk Blackthorn (blackthorn@ws-10.smyth.net) |
| 19:01.03 | Blackthorn | Hello, offtopic question. Do you know where i could find irc help for ms sql issues? |
| 19:01.09 | jets | Call me stupid but i thought ztcfg used to say yellow or red alarm if the line was down |
| 19:01.40 | _Brian | Hey Stupid! |
| 19:01.42 | jets | :P |
| 19:01.45 | slePP | zttool does, jets |
| 19:01.46 | _Brian | couldnt resist |
| 19:01.51 | _Brian | never saw that from ztcfg |
| 19:01.55 | tzanger | jets: no that's /proc/zaptel/# |
| 19:01.59 | tzanger | jets: or zttool |
| 19:02.15 | tzanger | hey, hey hey hey.... hey stoopid... whatcha tryin' to do? |
| 19:02.55 | jets | zttool never compiled and a make zttool errors. #%*&#%* |
| 19:03.00 | Mneumonic | Anyone know why my SoftPhone would say "Call Not Approved"? It suddenly started happening yesterday night |
| 19:05.33 | ionix | anyone has a way to make ANI reverse lookup & |
| 19:05.33 | ionix | like automatically with asterisk |
| 19:05.33 | ionix | i.e fast queries or download database |
| 19:09.20 | jesse_132 | anyone work with broadvoice for bulk origination (1 phone number, lots of incoming calls at the same time)? |
| 19:11.14 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
| 19:11.44 | ManxPwr | Mneumonic, Is it a USA Republican SoftPhone? I hear approval for those is droping like a rock. |
| 19:12.19 | dano_ | What would experienced users of Asterisk recommend for a small (10 person) call center as far as hard phones go? |
| 19:13.47 | *** join/#asterisk Ayano (~erik_leee@209.143.187.254) |
| 19:14.28 | Mneumonic | ManxPwr - No, its the X-10 X-Lite Softphone |
| 19:14.57 | *** join/#asterisk harryvv (~comming@S010600055d210201.vs.shawcable.net) |
| 19:15.24 | jets | dano: cisco 7960's are good if you want cheaper the polycoms are good too |
| 19:15.35 | Moc | polycom are better than the 7960 |
| 19:15.44 | tzanger | the asterisk messaging is fucked |
| 19:15.57 | tzanger | I saw "echo can disabled because of tone (rx) on channel 1" in demsg |
| 19:15.57 | Moc | only reason to go with a 7960 is, boot faster, and design look better |
| 19:16.02 | tzanger | which is good (there's a fax there) |
| 19:16.11 | tzanger | but zap show channel 1 shows 64 taps, currently ON |
| 19:16.11 | Nugget | yeah, the cisco phones look a lot sexier. |
| 19:16.11 | Moc | but the display is better on the polycom so.. |
| 19:16.38 | Nugget | I want something with a backlit display. |
| 19:16.46 | jets | I've never purchased from this company but they do show a lot of phones you could research from them and buy there or somehwere else you trust |
| 19:16.46 | jets | http://www.voipsupply.com/index.php?cPath=95_106&desc=IP%20Phones%20between%20$100%20and%20$200%20VoIP%20SIP%20H.323%20MGCP%20IAX |
| 19:16.51 | Moc | polycom have a new model with backlit i think |
| 19:16.57 | Nugget | ooh, cool. |
| 19:17.01 | Nugget | that'd swing me away from the ciscos. |
| 19:17.07 | Sedorox | I bought gradnsteams from them.... fast shipping... |
| 19:19.22 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 19:19.55 | jets | Anyone know of any phone's that actually support publish/subscribe sip extensions? |
| 19:20.26 | Delvar | nn all |
| 19:20.48 | *** join/#asterisk dsfr (~dsfr@216.207.244.183) |
| 19:20.50 | Ayano | what is publish/subscribe? |
| 19:20.58 | *** part/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 19:20.59 | Moc | polycom and snom i think |
| 19:21.01 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 19:21.25 | *** part/#asterisk jesse_132 (~chatzilla@207.246.72.150) |
| 19:22.08 | jets | Ayano: a feature where you could theoretically subscribe to see if someone elses extension is in use... like for a receptionist |
| 19:22.29 | Moc | jets check polycom and snom |
| 19:22.35 | Ayano | Why not just use a gui? |
| 19:23.25 | jets | Ayano: because maybe they don't have a computer nearby but need to see if someone is on the line -- most key pbx systems support it is all |
| 19:23.26 | Moc | because... |
| 19:23.30 | jets | Moc: Ya i'm looking up the specs now. |
| 19:23.46 | jets | I don't think the asterisk feature request for the publish/subscribe has been written anyway |
| 19:24.01 | Moc | there is a lamp feature for * |
| 19:24.04 | Ayano | True, Yea, most do, I just didn't know that's what it was called. |
| 19:24.11 | tzanger | maybe the echo canceller on PRI is broken in -HEAD or something |
| 19:24.13 | tzanger | it can't be working |
| 19:24.18 | tzanger | not on PRI |
| 19:24.42 | dsmouse | hey, who sales cheap 1-800 voip #'s |
| 19:24.51 | Moc | nufone |
| 19:25.50 | PCadach | tzanger: Hello! Why you mean EC can't be working on PRI? |
| 19:26.18 | bkw_ | Moc don't forget asterlink |
| 19:26.51 | bkw_ | dsmouse, you can signup online for asterlink.com and be active in about 15 min |
| 19:26.53 | bkw_ | or less |
| 19:26.56 | Moc | bkw_, asterlink keep sounding in my head as a solution provider only.. will try to mention it next time ;) |
| 19:27.06 | PCadach | bkw_: Any success with recorded conference? |
| 19:27.19 | Moc | allthrought I never experienced your services ;) |
| 19:27.22 | PCadach | bkw_: Is it published somewhere? |
| 19:27.25 | bkw_ | PCadach, ya know after all that I ended up with a nice silence recording |
| 19:27.27 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@d01m-21-177.d4.club-internet.fr) |
| 19:27.34 | *** join/#asterisk JimVanM (~jimvanm@HSE-Toronto-ppp3490056.sympatico.ca) |
| 19:27.49 | Moc | lol |
| 19:28.09 | bkw_ | 139 meg one at that |
| 19:28.16 | PCadach | I have about 1,5 hours at beginning of conference in WAV49 (about 8,8 mbytes). |
| 19:29.29 | bkw_ | can I get it from you? |
| 19:29.36 | bkw_ | i'll make sure we don't make that mistake this week |
| 19:29.48 | KalD|Work | kram, you alive? |
| 19:29.57 | harryvv | pretty much ata and ipphones will disable a outgoing greeting message if the call is answered? |
| 19:30.39 | tzanger | PCadach: I don't think it is |
| 19:30.43 | harryvv | Or does anyone know. |
| 19:31.14 | Moc | another meeting this week ? |
| 19:31.37 | *** join/#asterisk quigleymd (~quigleymd@24-53-142-5.chvlva.adelphia.net) |
| 19:31.47 | PCadach | tzanger: EC is enabled on _any_ audio channel when call isn't bridged within zaptel stack. |
| 19:33.01 | *** join/#asterisk felix (~felix@200.115.235.79) |
| 19:33.08 | felix | hello |
| 19:33.40 | netsurfer | guys.. got a slight problem here.. sipgate.co.uk do landline to voip termination.. recently when I dial the number its ringing as engaged, however * connects it and acts all stupid, but the caller never actually IS connected, they just get the engaged tone. does this sound like an asterisk problem or a problem with the carrier ? |
| 19:33.47 | tzanger | PCadach: I swear it ain't working |
| 19:34.13 | tzanger | PCadach: echocan of 32,64,128,256, echotraining on or off or 400 or 800... nothing seems to work and I'm doing a full stop/start of asterisk after changes |
| 19:34.18 | felix | i have a TDM400P [with 4 FXO modules], which signalling i must use? |
| 19:34.27 | felix | fxo_ls? |
| 19:34.38 | jets | No, oddly enough |
| 19:34.52 | jets | you are fxo, the pstn is fxs, so you put it in fxs signalling |
| 19:35.01 | *** join/#asterisk SeaForth (~SeaForth@c-24-1-126-202.client.comcast.net) |
| 19:35.06 | felix | jets: ok, thanks. |
| 19:35.07 | jets | it should explain that in zapata.conf i believe |
| 19:35.09 | PCadach | tzanger: EC is working ok on my box (it is slightly outdated but built from CVS-HEAD tree). |
| 19:35.30 | felix | jets: yes. I'm doing that right now. |
| 19:35.50 | bjohnson | dsmouse: the first question about DIDs is "For what country?" |
| 19:35.52 | PCadach | bkw_: How I can send it for you? |
| 19:35.57 | tzanger | PCadach: I dunno... I have a Bell PRI -- TE405P -- IAX2 -- TE405P -- PRI -- KSU |
| 19:36.04 | tzanger | neither one seems to echo cancel |
| 19:36.13 | netsurfer | this is very odd, after the caller gets the engaged tone on the pstn, hangs up.. the SIP phone rings and shows the callers number, but the line is quiet when answered |
| 19:36.26 | dsmouse | bjohnson: hehe. I guess it is. I'm a USeian. |
| 19:36.46 | tzanger | I only get echo on the Bell PRI (incoming or outgoing calls, and even then on MOST calls but not all) |
| 19:37.01 | tzanger | anything going to nufone or any other VOIP provider... no echo |
| 19:37.59 | jets | hrm who has an iax2 meetme somewhere i cna connect to |
| 19:38.21 | jets | or any test exten |
| 19:38.59 | bjohnson | jets: fwd |
| 19:39.15 | jaiger | jets: what bjohnson said |
| 19:39.27 | bjohnson | jets: what jaiger said |
| 19:39.54 | Sedorox | <PROTECTED> |
| 19:39.54 | Sedorox | ch |
| 19:39.59 | Sedorox | er |
| 19:40.03 | Damin | So.. anyone going to Von need a Hotel room to sleep in? :) |
| 19:40.11 | jets | :1 |
| 19:40.15 | jets | echo shutup |
| 19:40.15 | jets | goto 1 |
| 19:41.26 | harryvv | BTW, is there a switch or command that will limit packet streaming when either party is talking? |
| 19:41.41 | djin | Something totally OT, but hopefully someone from US can help me on this. |
| 19:41.44 | djin | I need to check if am American company still exists. In NL we have a Chamber-of-Commerce database to check, but I understand in US it's different. How would I be able to check if a Minneapolis based company is still alive? |
| 19:41.49 | PCadach | tzanger: I had PSTN PRI -- TE410P -- TDMoE PRI -- Skinny RTP, and echo canceller had work more-or-less but get overloaded on high remote echo signal (attennuation less than about 7-8 dB). You can patch ztmonitor to record both voice directions into stereo file then analyze/play with it. |
| 19:44.06 | jaiger | bjohnson: Stack overflow |
| 19:44.09 | ManxPwr | ~docs |
| 19:44.10 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 19:44.37 | jero | can I check if a file exists in extensions.conf and Playback() it if so ? |
| 19:44.44 | jero | or must I use agi |
| 19:45.09 | *** join/#asterisk jgaviria (~jgaviria@63.245.86.116) |
| 19:45.30 | tzanger | PCadach: hmm okay |
| 19:46.01 | bjohnson | jero: just playback it? |
| 19:46.03 | kpfleming | anybody out there got a Cisco Call Manager running SIP? |
| 19:46.41 | jero | bjohnson, I was thinking so :) |
| 19:46.56 | jero | Can I pass Playback() an absolute filename ? |
| 19:47.02 | bjohnson | I don't think it will kill the call .. just dumps errors to logs |
| 19:47.19 | jero | I want to plat the VM's recorded user name after extension has been dialed |
| 19:47.22 | kpfleming | jero: yes |
| 19:47.32 | jero | s/plat/play |
| 19:47.33 | *** join/#asterisk netsurfer (~netsurfer@dreambox.myvnc.com) |
| 19:47.37 | jero | great |
| 19:47.41 | bjohnson | speaking of logs .. how do you rotate the logs so some actually get deleted |
| 19:47.50 | tzanger | bjohnson: logger rotate |
| 19:47.57 | *** join/#asterisk netsurfer (~netsurfer@dreambox.myvnc.com) |
| 19:48.18 | tzanger | personally I disagree with that entirely... logs should be opened, written and closed so a normal logrotate system can take them rather than have asterisk have some "special" rotate need |
| 19:48.24 | bjohnson | http://www.voip-info.org/tiki-index.php?page=logrotate |
| 19:48.24 | jero | so I can Playback(/var/spool/asterisk/voicemail/office/${EXTEN}/greet) |
| 19:49.15 | bjohnson | tzanger: logger rotate doesn't remove any .. and it makes the just-past-current messages file the biggest in the sequence |
| 19:49.27 | tzanger | bjohnson: I know |
| 19:49.28 | bjohnson | eg messages become messages.9 if there already .8 there |
| 19:49.31 | tzanger | what I'm saying is that isn't right :-) |
| 19:49.37 | bjohnson | it sucks |
| 19:49.40 | tzanger | I should not have to tell asterisk to rotate |
| 19:49.57 | tzanger | if you're writing THAT much to logs that open/write/close is too excessive then you have special needs |
| 19:50.06 | eKo1 | Anybody have a file called parking.h in the * source code? |
| 19:50.10 | tzanger | again, just all my opinion |
| 19:51.03 | bjohnson | right. especially with good utils like logrotate available |
| 19:51.38 | bjohnson | and seemingly little understanding of how the log system works here |
| 19:52.01 | bjohnson | I've been asking that question off and on for weeks now |
| 19:52.40 | tzanger | yeah |
| 19:52.53 | tzanger | basically you need to renumber them manually |
| 19:52.57 | bjohnson | here's a new question I have. My Sipura SPAs seem to re-ring my Nortel extensions after a hangup. Any idea how I can stop this? |
| 19:53.30 | eKo1 | I have logrotate rotating my asterisk logs just fine. |
| 19:53.41 | bjohnson | eKo1: mind adding details to that wiki page? |
| 19:54.04 | bjohnson | or give details to me, I'll test and update wiki |
| 19:54.04 | eKo1 | bjohnson: How about I post the logrotate script I'm using in the wiki. |
| 19:54.11 | bjohnson | sounds good |
| 19:54.21 | eKo1 | OK then. |
| 19:54.31 | bjohnson | http://www.voip-info.org/tiki-index.php?page=logrotate |
| 19:55.22 | *** join/#asterisk RoyK (~roy@44.80-203-227.nextgentel.com) |
| 19:57.02 | tzanger | hmm |
| 19:57.17 | tzanger | I only get color in my -rc sessions if my main asterisk session has a console (i.e. -gc) |
| 19:57.32 | netsurfer | me too |
| 19:57.52 | tzanger | netsurfer: is that for me? |
| 19:57.58 | netsurfer | yep tzanger |
| 19:58.01 | tzanger | heh |
| 19:58.08 | tzanger | I was fighting with -dev on this for a while |
| 19:58.13 | tzanger | I say -c stands for COLOR not console |
| 19:58.22 | tzanger | perhaps BOTH |
| 19:58.28 | machinehd | I have an * box with a tdm04b. Incoming pstn calls are perfect. However, the person making an outgoing call can hear random high pitched beeps. We tried this on 3 separate GS phones. Any ideas? |
| 19:58.42 | netsurfer | -r for remote -c for color - at least thats what I thought |
| 19:58.43 | tzanger | machinehd: cat /proc/interrupts |
| 19:58.59 | tzanger | netsurfer: -c is for console, apparently but it is hte only thing that makes the remote sessions come up in color |
| 19:59.15 | netsurfer | hehe kinda funny aint it |
| 19:59.24 | machinehd | tzanger, 10: 264268061 XT-PIC ehci_hcd, wctdm |
| 19:59.51 | netsurfer | ever since I added * to init.d I get lots of ghost connections, and -- Remote UNIX connection disconnected - is this normal? |
| 20:00.12 | tzanger | machinehd: are you using yoru USB |
| 20:00.41 | KalD|Work | netsurfer, it says that when you try to run multiple copies at the same time |
| 20:01.07 | machinehd | tzanger, nope. It's disabled in the bios. Do I still need to change something to disable it in the OS? |
| 20:01.35 | netsurfer | eek... ps x says asterisk has ~ 25 PIDs hehe |
| 20:01.39 | harryvv | kal, multiple copies of hwat? |
| 20:01.44 | harryvv | ohh man |
| 20:01.46 | harryvv | hahah |
| 20:01.55 | KalD|Work | harryvv, asterisk |
| 20:02.09 | harryvv | yea |
| 20:02.32 | KalD|Work | I've got my system setup w/ crontab and it tries to start * every 30 seconds - so I get a message like that every 30 seconds on the * console |
| 20:02.36 | netsurfer | http://pastebin.ca/5821 <--- LMAO |
| 20:02.56 | tzanger | machinehd: no |
| 20:04.05 | *** join/#asterisk hans (fugalh@falcon.fugal.net) |
| 20:04.46 | stevekstevek | tzanger: heya. |
| 20:06.06 | bjohnson | anyone have echo issues with SPA 3ks? I've read some posts about setting line impedence possibly helping for fxo-fxs calls .. but all posts were UK and don't use same system as NA |
| 20:06.20 | tzanger | stevekstevek: hey |
| 20:06.46 | stevekstevek | I put new patches up today.. Nothing to solve your particular issue, but still stuff you'd like. |
| 20:07.02 | stevekstevek | gotta leave early today, though, so that's probably it. |
| 20:07.21 | tzanger | stevekstevek: oh? |
| 20:07.37 | stevekstevek | it for today, that is. |
| 20:07.57 | stevekstevek | you'll like iax2 show netstats now. |
| 20:08.02 | tzanger | yes I am using that |
| 20:08.08 | tzanger | patch5 adds more to it? |
| 20:08.26 | stevekstevek | yeah -- the old one didn't support loss and stuff. |
| 20:08.41 | stevekstevek | *CLI> iax2 show netstats |
| 20:08.41 | stevekstevek | <PROTECTED> |
| 20:08.41 | stevekstevek | Channel RTT Jit Del Lost % Drop OOO Kpkts Jit Del Lost % Drop OOO Kpkts |
| 20:08.41 | stevekstevek | IAX2/guest@10.23.1.31:596 16 109 178 49 1 5 0 1 148 225 28 0 1 0 0 |
| 20:08.46 | tzanger | nice |
| 20:08.52 | tzanger | does this patch nicer than patch4 did ? ;-) |
| 20:09.03 | bjohnson | who here has DIDs from iax.cc user? I have some questions about their service. |
| 20:09.04 | stevekstevek | it should patch clean against CVS-head. |
| 20:09.11 | tzanger | bjohnson: I do |
| 20:09.15 | tzanger | don't use it much |
| 20:09.18 | tzanger | but it seems to work |
| 20:09.41 | bjohnson | tzanger: do they provide callerid on incoming calls? Name AND number? |
| 20:09.48 | tzanger | bjohnson: good question |
| 20:09.51 | tzanger | call me |
| 20:09.53 | bjohnson | ha |
| 20:10.28 | bjohnson | ... I don't have your number |
| 20:10.53 | bjohnson | tzanger <- has so many numbers he has to look his own number up |
| 20:10.58 | tzanger | bjohnson: 5194892955 |
| 20:11.49 | bjohnson | busy |
| 20:12.15 | tzanger | busy |
| 20:12.15 | tzanger | wtf |
| 20:12.15 | bjohnson | or is that congested? |
| 20:12.15 | tzanger | Feb 14 15:23:20 NOTICE[2433]: chan_iax2.c:7403 iax2_poke_noanswer: Peer 'sixTel' is now UNREACHABLE! Time: 42 |
| 20:12.17 | tzanger | fast busy = congested |
| 20:12.30 | bjohnson | I called through a pstn line .. no voip at this end for that call .. tried twice |
| 20:12.31 | tzanger | try now |
| 20:13.16 | bjohnson | ringing |
| 20:13.28 | tzanger | ringing?? |
| 20:13.32 | tzanger | I don't see the call coming in |
| 20:13.32 | bjohnson | yes |
| 20:13.46 | bjohnson | btw .. local call to me |
| 20:13.57 | bjohnson | (according to Bell) |
| 20:14.01 | tzanger | 5194892955 |
| 20:14.11 | tzanger | are you sure you're dialing it, I just called it myself and it worked |
| 20:14.29 | bjohnson | now can't be completed as dialed |
| 20:14.41 | tzanger | <PROTECTED> |
| 20:14.45 | tzanger | <PROTECTED> |
| 20:14.47 | tzanger | that is what I get |
| 20:14.50 | tzanger | no name and I *do* set outgoing name |
| 20:15.00 | bjohnson | now some message |
| 20:15.07 | tzanger | yeah |
| 20:15.08 | tzanger | that was me |
| 20:17.44 | bjohnson | you sound like a woman |
| 20:17.45 | *** join/#asterisk three55ml (~who@cs662589-157.satx.rr.com) |
| 20:18.54 | tzanger | that's not me |
| 20:19.16 | bjohnson | hehe |
| 20:20.23 | bjohnson | mind if I fire some questions at you that I sent to them and haven't had a reply yet? You might know some of the answers |
| 20:21.07 | bjohnson | btw .. does iax.cc ONLY support iax? |
| 20:21.14 | bjohnson | or do they also do SIP? |
| 20:21.37 | eKo1 | OK. Done. |
| 20:22.15 | eKo1 | Look here: http://www.voip-info.org/tiki-index.php?page=logrotate |
| 20:23.18 | jets | hrm |
| 20:23.25 | jets | does it really get this quiet aroung lunch? |
| 20:23.33 | bjohnson | today is dead |
| 20:23.46 | bjohnson | usually a bunch of newbies in here asking all sorts of questions |
| 20:23.55 | bjohnson | <- newbie |
| 20:24.00 | bjohnson | <- asking questions |
| 20:24.02 | tzanger | eKo1: nice, although that's not any different than bjohnson's first post |
| 20:24.14 | tzanger | you're rotating everything though which is nice :-) |
| 20:25.24 | bjohnson | I'll try it .. filename at the top is different |
| 20:25.47 | PCadach | tzanger: Do you want to get caller name over PRI? |
| 20:25.57 | tzanger | PCadach: I do alrady |
| 20:25.59 | tzanger | er already |
| 20:26.04 | tzanger | I can set callerid and name on my PRI too |
| 20:26.49 | PCadach | Hm-m-m... Is it passed over long-distance calls? |
| 20:27.00 | terrapen | anybody here using a WRAP board? |
| 20:27.04 | terrapen | or ever used one? |
| 20:27.06 | tzanger | PCadach: so long as it stays within the bell network, yes |
| 20:27.12 | tzanger | terrapen: yes I have, but not with asterisk |
| 20:27.21 | kpfleming | anybody here done a firmware upgrade on a Snom 200? |
| 20:27.21 | terrapen | tz: i wonder how it would work |
| 20:27.27 | tzanger | terrapen: it's got no balls |
| 20:27.32 | terrapen | `really? |
| 20:27.34 | tzanger | so don't expect tons of codec |
| 20:27.38 | tzanger | Geode 266Mhz has nothing |
| 20:27.52 | terrapen | hmmm |
| 20:28.07 | *** part/#asterisk felix (~felix@200.115.235.79) |
| 20:28.10 | terrapen | what codec does my Polycom ip500 use by default? |
| 20:28.18 | *** join/#asterisk evilrabbi (evilrabbi@64.123.157.107) |
| 20:28.22 | terrapen | i wonder if it could handle a few ip500s w/o transcoding |
| 20:29.02 | bjohnson | eKo1: first run looks more like what I expected from mine. I set to daily and will run for a week and delete mine from wiki is all is good |
| 20:29.41 | *** join/#asterisk pjm_uk (~pjm_uk@cpc1-pool3-3-0-cust116.sot3.cable.ntl.com) |
| 20:29.50 | sneak | hi guys |
| 20:29.56 | *** join/#asterisk Tough_Nuts (~Tough_Nut@12.171.163.38) |
| 20:30.00 | tzanger | terrapen: only one way to try it |
| 20:30.01 | PCadach | tzanger: Could you make test long-distance call, I'll not answer (but after 20 sec there is voicemail)? |
| 20:30.10 | sneak | in your opinions, what is the most stable and feature-rich hardphone that's not cisco? |
| 20:30.24 | PCadach | tzanger: er, international call I mean.. |
| 20:30.41 | tzanger | PCadach: I'm pretty sure that won't work, since it'll hop off of Bell Canada's network :-) |
| 20:30.44 | tzanger | PCadach: but yeah I can try |
| 20:31.04 | *** join/#asterisk VivekKhera (~vivek@onceler.kcilink.com) |
| 20:31.10 | *** join/#asterisk PakiPenguin (~uppal@202.176.254.1) |
| 20:31.59 | *** join/#asterisk clint_ (~clint@snap.helixsystems.com) |
| 20:32.38 | clint_ | anyone: AgentCallbackLogin and ackcall=no - anyone have this working? I always have to # ack the call... |
| 20:32.47 | tzanger | PCadach: ok let me know |
| 20:34.35 | PCadach | tzanger: I have two PRI links to PSTN through different exchanges (Alcatel S-12 and Huawei C&C08). Could you call me twice at +7-3232-200770 to go through C&C08 and at +7-3232-220201 to go through S-12? |
| 20:34.38 | PakiPenguin | hello there , i am trying to solve a problem ( i am sending calls out to my second * server from the first one ) , the call passes on fine , and it dials out from the second server fine , but after 1 / 2 secs , it gives me an error "unable to transfer" and the iax link between two * machines is dropped |
| 20:34.50 | PakiPenguin | http://pastebin.ca/5823 < -- here's the pastebin |
| 20:34.56 | terrapen | does anyone make a PCI board that can offload transcoding? |
| 20:35.18 | Silik0n | hah thats funny |
| 20:35.38 | PCadach | terrapen: As discussed some time ago such type of card will overload PCI bus. |
| 20:36.00 | terrapen | really?> |
| 20:36.04 | *** part/#asterisk djin (~djin@gridfox.xs4all.nl) |
| 20:36.06 | terrapen | its that much I/O? |
| 20:36.08 | terrapen | for just voice? |
| 20:36.20 | terrapen | voice =~ 64kbps? |
| 20:36.21 | *** join/#asterisk sivana (~sivana@165.154.13.35) |
| 20:36.35 | terrapen | i don't believe that |
| 20:36.45 | terrapen | you can put a GBE card on PCI |
| 20:36.53 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 20:36.55 | terrapen | how the hell could a transcoding card overload pci |
| 20:37.23 | sivana | anyone use AMD with *, ie: AMD SEMPRON 2300+ S462 |
| 20:37.34 | shido6 | A im Dee |
| 20:38.07 | PCadach | terrapen: It will good enough for limited number of channels processed in parallel, but this value is hardlimited by PCI speed and OS overhead for interrupt handling. |
| 20:38.56 | *** join/#asterisk kietlak (~kietlak@11-mo3-6.acn.waw.pl) |
| 20:39.01 | tzanger | PCadach: I'm trying, my damn phone system's limiting my digits |
| 20:39.19 | sneak | yeah - remember it's not so much pci bandwidth as it is latency, and the x86 architecture sucks wrt interrupts |
| 20:39.23 | *** part/#asterisk Nix (~Nix@81.213.125.220) |
| 20:39.36 | shido6 | sneak? |
| 20:39.37 | terrapen | this is voice we are talking about |
| 20:39.49 | terrapen | how bad could the latency possibly be? |
| 20:40.09 | harryvv | tzanger, seen those 15 pci slow expansion boards? Wonder if thay would work well with the zap cards. |
| 20:40.22 | eKo1 | Anybody have app_callingcard working? |
| 20:40.28 | harryvv | 15 pci slot boards. |
| 20:40.39 | PakiPenguin | why do i get "unable to transfer" when the other end picks up the call |
| 20:42.20 | *** join/#asterisk QRPartner (~andy@ns1.accu-com.com) |
| 20:42.28 | QRPartner | Hello |
| 20:42.38 | *** join/#asterisk dsfr (~dsfr@216.207.244.183) |
| 20:42.52 | PCadach | terrapen: To be sure encoded voice packet. To process each packet you must: a) place packet somewhere in memory accessible from transcoding card; b) notify transcoding card about packet available for processing; c) card will lock PCI bus for busmaster operation for downloading packet data; d) card will lock PCI bus for busmaster operation for uploading decoded voice packet into main memory; e) card will notify CPU about finished transcoding process |
| 20:43.07 | QRPartner | Can someone direct me to where timing is setup? My carrier is telling me it needs to be set from them (external) |
| 20:43.20 | tzanger | harryvv: so long as they can all have their own IRQs, no problem |
| 20:43.24 | harryvv | http://www.cyberresearch.com/store/product/307.2.htm 15 pci slow backplane. Possible to run zaps in these? looks to have many irq controler chips on them. |
| 20:43.27 | tclark | pacach: I guess atacomm dont believ that http://ipvolution.com/ |
| 20:43.45 | terrapen | hrmmm |
| 20:44.07 | terrapen | PCadach, how are these SSL-offloader cards doing upwards of 100Mb/s |
| 20:44.21 | shido6 | zapata.conf QRPartner |
| 20:44.24 | shido6 | err |
| 20:44.25 | shido6 | sorry |
| 20:44.27 | shido6 | zaptel.conf |
| 20:44.31 | shido6 | QRPartner |
| 20:44.43 | shido6 | check QRPartner and if ur stumped check /usr/src/asterisk/configs/zapata.conf.sample |
| 20:44.47 | PCadach | tzanger: and have its own PCI bus controller built in chipset. |
| 20:44.55 | shido6 | its all there in a comment for ya QRPartner |
| 20:45.08 | shido6 | read it 4 times and call me in the morning |
| 20:45.16 | PCadach | tzanger: Ok, forget about calls. |
| 20:45.17 | Silik0n | the ammount of data to offload for transcoding would be a monster... sure there might be bandwidth on the PCI-X bus for it, but I dunno about a PCI bus... best bet would be encoidng TDM calls with a DSP on a card. that would work... besides I dont see a real reason to use anything but Ulaw or Alaw on a LAN and then use a more compressing codec like G729 on WAN connects. I use ULaw on the net all the time and it sounds just find but thats only a small |
| 20:45.21 | tzanger | PCadach: sorry |
| 20:45.32 | tzanger | I can't get my KSU to let me release more than 13 digits |
| 20:45.46 | tzanger | and I need 14 |
| 20:45.50 | tzanger | to hit the 011 |
| 20:46.28 | PakiPenguin | damn , unable to transfer? why why why!!! |
| 20:48.01 | PCadach | tzanger: No problem. I just wanted to have international call with Calling name to see is it passed over international SS7 links and how it's represented by local exchanges on PRI links (by ROSE or by regular DISPLAY IE). |
| 20:48.35 | shido6 | err |
| 20:49.12 | shido6 | what is that, oh yeah |
| 20:49.13 | shido6 | pri intense debug span 1 |
| 20:49.20 | shido6 | there's your IE, ROSE info ... |
| 20:50.28 | sivana | is AMD any good with *? |
| 20:52.05 | *** join/#asterisk SeaForth (~SeaForth@c-24-1-126-202.client.comcast.net) |
| 20:52.22 | Silik0n | Silik0n yes works just fine |
| 20:52.28 | Silik0n | sivana rather |
| 20:52.32 | Silik0n | damned crappy autocomplete |
| 20:52.56 | *** join/#asterisk wolfson (~hehe@cpe-68-187-190-217.man.nc.charter.com) |
| 20:53.03 | sivana | Silik0n: are you familiar with AMD Sempron 2300+ S462 ? |
| 20:53.15 | Silik0n | sivana you can even tweak the Makefile a little for Athlon support if your uname output doesnt show athlon |
| 20:53.17 | PakiPenguin | can anyone PLEASE help me out? |
| 20:53.41 | Silik0n | not first hand... I have a stack of AthlonXP (32bit) boxes |
| 20:54.34 | *** join/#asterisk buddah (~hnic@208.179.86.5) |
| 20:54.37 | sivana | wondering if it could handle 19 simultaneuos calls |
| 20:54.41 | Silik0n | actually I think thats basically the same thing I have in my 2200 and 2400+ boxes |
| 20:54.52 | Silik0n | so long as you arent transcoding |
| 20:54.58 | buddah | does anyone know what would prevent calls from getting progress tones when dialing one of our dids? |
| 20:55.03 | *** join/#asterisk Ayano (~erik_leee@209.143.187.254) |
| 20:55.09 | buddah | when i call any of our polycoms, i get no tones |
| 20:55.22 | shido6 | cant u turn off callprogress |
| 20:55.28 | shido6 | read the zapata.conf.sample |
| 20:55.32 | buddah | ok |
| 20:55.40 | sivana | transcoding is defined as? |
| 20:55.52 | shido6 | Im not screaming at you :) its in /usr/src/asterisk/configs/zapata.conf.sample |
| 20:56.00 | shido6 | its a start |
| 20:56.01 | Silik0n | having Asterisk convert from one codec to another |
| 20:56.27 | Silik0n | ie: phone A on GSM -> asteirsk ->g729a -> phoneB |
| 20:56.44 | Silik0n | ALL HAIL KRAM! |
| 20:56.47 | buddah | lol |
| 20:56.48 | sivana | going from a channel bank to my switch |
| 20:57.34 | greg_work | okay, just finished my sipura config tool, for configuring a set of spa-841's, if anyone is interested: http://opensource.mwater.ca/projects/sipuraconfigtools |
| 20:57.36 | Silik0n | sivana channel bank -> T1 -> T100P -> asterisk -> G711U -> SIP Phone is not realyl transcoding |
| 20:58.20 | *** join/#asterisk Uajal (~icechat5@ool-182e86f3.dyn.optonline.net) |
| 20:59.14 | sivana | nortel -> channel bank -> asterisk on AMD |
| 20:59.57 | Silik0n | sivana analog ports off the nortel? |
| 21:00.02 | buddah | i found the callprogress in zapata |
| 21:00.08 | buddah | and uncommented it, so its =yes |
| 21:00.15 | buddah | and nothing |
| 21:00.30 | *** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc) |
| 21:00.43 | sivana | Silik0n: yes |
| 21:01.04 | Silik0n | that should work find just use G711 on the IP phones |
| 21:01.08 | Uajal | I used before * such ivr stuff wait_for_silence(n), flag is_answering_machine, event voice-barge-in. I didn't find anything similatr in * docs. Somebody knows about call progress detection? |
| 21:01.17 | sivana | it's a nortel MICS |
| 21:01.31 | Silik0n | you just said a bad word |
| 21:01.50 | sivana | well.. just changing out the dial tone basically |
| 21:01.55 | Silik0n | yeah |
| 21:02.03 | sivana | I'd love to put in new handsets.. but they aren't ready :) |
| 21:02.13 | Silik0n | so you are going to use * to play VoIP -> ATA for the MICS? |
| 21:02.49 | shido6 | ok |
| 21:02.53 | shido6 | since u dont listen to pm's sivana |
| 21:02.54 | shido6 | http://www.voip-info.org/files/nortel-asterisk-0.2.pdf |
| 21:02.58 | shido6 | there ya go |
| 21:02.59 | sivana | uhh... MICS -> channel bank -> their * -> ethernet bridge -> our * box -> PRIs |
| 21:03.15 | tzanger | that nortel-asterisk doc doesn't tell you shit |
| 21:03.17 | tzanger | trust me |
| 21:03.21 | tzanger | it was a very specific application |
| 21:03.22 | shido6 | its a start... |
| 21:03.38 | tzanger | there's no magic to using Norstar with asterisk on PRI or POTS |
| 21:03.40 | shido6 | sounds like ur pissed at that doc tz |
| 21:03.41 | shido6 | :) |
| 21:03.43 | tzanger | heh no |
| 21:03.48 | Uajal | Call Progress Detection in * - pls any suggestions! |
| 21:03.53 | harryvv | Is there a easier way to read Master.csv? |
| 21:04.32 | buddah | easier than what? |
| 21:04.40 | buddah | could plug it into excel and read it that way |
| 21:05.10 | buddah | or any other spreadsheet program will work |
| 21:05.27 | harryvv | okay |
| 21:05.47 | harryvv | budda, so excel will sort all the componets out then. |
| 21:06.13 | eKo1 | Beware though that large Master.csv files will take a while to load on Excel. |
| 21:06.24 | Uajal | Could anybody answer anything about Call Progress detection. Does it really work? |
| 21:06.44 | eKo1 | Uajal: on zap? |
| 21:06.56 | harryvv | yea, I guess i need to filter the cdr if its to look right. |
| 21:07.03 | Uajal | VoIP |
| 21:07.26 | buddah | yes |
| 21:07.39 | buddah | harryvv: should, .csv files work well with excell |
| 21:07.48 | buddah | harryvv: every one i had it sorts out well |
| 21:07.50 | Uajal | what is the AGI command to detect answering machine? |
| 21:07.53 | harryvv | okay |
| 21:09.19 | Uajal | is it debug message or else? |
| 21:09.51 | Uajal | Should it appear as "Answering machine answered"? |
| 21:11.25 | Uajal | eko1: any link to write IVR script with * ? |
| 21:11.55 | Silik0n | <PROTECTED> |
| 21:12.25 | Silik0n | damnit |
| 21:12.52 | cjk | hi i tried to restart *. but there is still someone calling someone. now * doesnt accept new channels. any way to cancel the restart |
| 21:13.06 | shido6 | cancel the restart? |
| 21:13.34 | shido6 | keep a backup of your working .conf files next time :) |
| 21:13.46 | bjohnson | I do * to Nortel CICs via 4 fxs ports into the Nortel analog trunks |
| 21:13.56 | bjohnson | sivana: ^^ |
| 21:14.53 | JimVanM | bjohnson: do you have a services card so you can receive DTMF? |
| 21:15.52 | tzanger | JimVanM: you do exist on irc |
| 21:16.04 | Uajal | eko1: I mean smth like "If person barged in then record his message then if silence longer than 4 sec then play message" can it be done with *? |
| 21:16.16 | JimVanM | tzanger: ya, but not much |
| 21:16.17 | tzanger | Uajal: anything is possible with asterisk |
| 21:16.25 | tzanger | Uajal: it's all a matter of how much time and energy you want to spend doing it |
| 21:16.34 | tzanger | Uajal: something like that is not immediately available on asterisk |
| 21:16.42 | tzanger | Uajal: so you need to either code it yourself or put up a bounty |
| 21:17.10 | JimVanM | tzanger: Mr Benjamen Shaw, I presume? |
| 21:17.15 | tzanger | haha |
| 21:17.15 | tzanger | yes |
| 21:17.49 | Uajal | yes I didn't find any AGI command concerning Progress detection. But Mark wrote that he made something concerning this? |
| 21:17.53 | JimVanM | tzafrir: dispensing Norstar hacks, are we? |
| 21:17.57 | tzanger | JimVanM: ha |
| 21:17.59 | tzanger | nah |
| 21:18.01 | VivekKhera | anyone out there that uses switchvox willing to share experience with it? it is on my short list of vendors and I'd like to hear what other users have to say. |
| 21:18.09 | tzanger | just saying the pdf online isn't really all that spectacular |
| 21:18.42 | tzanger | Uajal: progress detection is in * but I don't think anyone uses it in production due to high chance of false positives |
| 21:18.51 | JimVanM | tzanger: if you know nor* you don't need it - if you don't, it's almost as useless as nortel docs |
| 21:18.52 | bjohnson | JimVanM: just a Nortel CICs with a standard card (4 line in, 16 extensions) |
| 21:19.03 | tzanger | JimVanM: exactly |
| 21:19.12 | Uajal | So what is the way to determine wether it is answering machine or person? |
| 21:19.14 | tzanger | bjohnson: plug it into a TDM440P |
| 21:19.22 | tzanger | Uajal: are yout he same guy that was in -dev last week |
| 21:19.54 | *** join/#asterisk rvhi (~rv@mail.o-matrix.org) |
| 21:20.01 | JimVanM | bjohnson: if you have 16 extns, you either got them via a services card or a key code. If you've got the services card, you can use target lines |
| 21:20.12 | bjohnson | tzanger: I have it plugged into 2 SPA 3000s and a SPA 2000 to create 3 line appearances on the Nortel .. line 1 still connects directly to pstn |
| 21:20.19 | bjohnson | JimVanM: no |
| 21:20.22 | Uajal | I was in -dev but not last week. May be 3 weeks ago or more |
| 21:20.26 | tzanger | bjohnson: ok |
| 21:20.35 | tzanger | Uajal: well basically answering machine detection is done a number of ways |
| 21:20.45 | rvhi | does one know how to setup * to use mysql for sip.conf, extension.conf and voicemail.conf? |
| 21:20.48 | *** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net) |
| 21:20.57 | bjohnson | tzanger: other than some echo problems and a strange ring back after hanging up .. it is working |
| 21:20.58 | tzanger | there was a guy in last week who said his existing system listened for energy in a passband and that worked best, although i don't imageine how |
| 21:21.49 | JimVanM | bjohnson: bummer. still, the CICS has that autoattendant built in, so with the right delays you could have Asterisk dialplan entries connect directly to stations . . . |
| 21:21.55 | Uajal | My current IVR (not *) determine OK smth like 80% |
| 21:22.03 | Uajal | even more |
| 21:22.19 | tzanger | bjohnson: are you sure the ringback isn't the SPA saying you've got voicemail? |
| 21:22.50 | Uajal | Determine answering machine. My algorithm is to leave the message to answering machine but hang up on person |
| 21:23.29 | tzanger | Uajal: my general understanding of answering machine detection goes like this: Listen for voice energy > a few seconds upon answer. Most people say "Hello?" or "Welcome to MyCompany." and then wait for you to say something. Answering machines typically have longer amounts of talk |
| 21:24.07 | Uajal | Year but how can it be done in *? |
| 21:26.40 | bjohnson | JimVanM: I have a X100P fxo connected into the built in ATA port. Haven't tested it but theoretically I can dial extensions through that |
| 21:26.41 | tzanger | Uajal: My guess is to work in dsp.c and create something similar to the existing fax detection where you jump to an extension upon detecting a fax machine |
| 21:27.02 | bjohnson | JimVanM: but yes .. autoattendant and log into DISA of the Nortel is another option |
| 21:27.07 | JimVanM | bjohnson: yep, but no CLID then |
| 21:27.48 | Uajal | I tried to find in Internet any samples of answering machine detection with * and no result. |
| 21:28.09 | bjohnson | tzanger: I don't think so .. but I'll check. I only have one global voicemail box and I think the SPA are not set to use it .. but I should check what they ARE set to |
| 21:29.18 | bjohnson | JimVanM: we intend to keep the Nortel .. 4/16 should do use quite well for a number of years .. * is just there to take advantage of lower LD and other voip features |
| 21:29.24 | tzanger | Uajal: that is becuase it doesn't currently exist |
| 21:29.30 | tzanger | you'll have to do it yourself or put up a bounty |
| 21:30.44 | JimVanM | bjohnson: those things do what they do really well, so that's a good plan. Asterisk is great because you can migrate users over as needs dictate. then one day, you just unplug the nor* and put it up on ebay :-) |
| 21:30.51 | bjohnson | Uajal: hanging up on a person is rather irritating for that person (from personal experience) |
| 21:31.22 | bjohnson | JimVanM: and get extra $$ because it will be an antique by then |
| 21:31.38 | bjohnson | it's already 5+ years old |
| 21:31.46 | JimVanM | bjohnson: LOL! Actually, those things keep their value very well, they never die. |
| 21:32.31 | JimVanM | bjohnson: also, Citel makes a SIP gateway for Nor* sets - means you don't have to chuck your phones |
| 21:34.34 | Uajal | I mean different scenario for answering machine and person |
| 21:35.17 | denon | anyone remember off the top of their head where you enable UniqueID? |
| 21:35.30 | denon | thought it was defined by default now, guess not |
| 21:36.28 | Uajal | Ok, so the only way to check whether it is answering machine or person is to change asterisk code and recompile? |
| 21:36.42 | *** part/#asterisk VivekKhera (~vivek@onceler.kcilink.com) |
| 21:37.22 | *** join/#asterisk fwittekind (rom@pcp0010183025pcs.columbus.in.indy.comcast.net) |
| 21:37.58 | rvhi | i just tried my first setup of voicemailmain, how come it didn't prompt me to change settings on the first login? |
| 21:38.00 | fwittekind | I am trying to troubleshoot PRI CNAM, telco says it is enabled. what exactly should I look for the the pri debug output? |
| 21:38.54 | Essobi | Anyone know why when I issue a dial from perl AGI my callerID to my 7960 say "From asterisk asterisk"? |
| 21:39.20 | Nugget | because your callerid is unset or set to an invalid string. |
| 21:39.37 | Essobi | what's a "invalid" string? |
| 21:39.52 | Nugget | an invalid string. I don't know how else to say that. |
| 21:40.01 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 21:40.03 | Nugget | a string which doesn't resemble callerid. |
| 21:40.07 | ariel_ | good afternoon all |
| 21:40.10 | Essobi | I'm just trying $AGI->set_callerid('545'); |
| 21:40.26 | *** join/#asterisk VivekKhera (~vivek@onceler.kcilink.com) |
| 21:40.36 | Nugget | set it to 'asterisk <545>' |
| 21:40.42 | Essobi | Mmm. |
| 21:40.50 | Essobi | the perl AGI docs, suck then. ;) |
| 21:40.50 | Essobi | Example: $AGI->set_callerid('9995551212'); |
| 21:40.54 | *** join/#asterisk ManxPower (~eric@dsl-209-205-172-111.i-55.com) |
| 21:41.08 | bjohnson | JimVanM: someone was asking about that last week .. any idea what the Citel units cost and how many ports? |
| 21:41.10 | Nugget | I'm not saying that will work, just that I'd expect it to want that form. |
| 21:41.42 | Nugget | asterisk is the value you get if the channel code can't parse the callerid. |
| 21:41.45 | Essobi | Mmm. almost. |
| 21:41.54 | Essobi | <PROTECTED> |
| 21:42.06 | Essobi | yields From Unknown asterisk |
| 21:42.11 | Nugget | closer, then. |
| 21:42.15 | fwittekind | any1 know what to look for in the pri debug output for troubleshooting caller id name? switch type is ni2 from a dms100 |
| 21:42.18 | Essobi | Ayup. |
| 21:42.28 | Nugget | perhaps set_callerid wants two parameters, then. |
| 21:42.48 | JimVanM | bjohnson: i spoke to them at some length about that. They only come in a 24 port model, and are roughly USD$100 per port. I encouraged them to consider doing a 4, 8, and 16 port model. They're thinking about it. |
| 21:42.54 | *** join/#asterisk file (~symlink@mctn1-142166197096.nb.aliant.net) |
| 21:42.59 | Essobi | Mmm. |
| 21:43.10 | Essobi | It doesn't according to the docs. |
| 21:43.13 | ariel_ | fwittekind, I can tell you if all the settings are correct and you dchannel is set you should get CallerID if the provider is sending it to you. |
| 21:43.26 | JimVanM | bjohnson: if someone were to come up with an IAX to legacy PBX adaptor, I'd want shares! |
| 21:43.30 | Silik0n | ok the telecom world is going to hell... verizon is buying MCI |
| 21:43.46 | three55ml | I used to work for Verizon, they think they're a money tree |
| 21:43.50 | Silik0n | JimVanM: gotta EE? |
| 21:43.53 | Essobi | two params ain't it |
| 21:44.15 | ariel_ | Fact is that there buying it for less then what qwest was offering |
| 21:44.22 | JimVanM | Silik0n: have you heard of the open-source PBX called Asterisk? ;-P It's gonna chance ALL the rules! |
| 21:44.26 | Essobi | JimVanM It's called an asterisk box. ;) |
| 21:44.27 | Silik0n | EE with high density Multilayer board design wanted... contact me via PM |
| 21:44.32 | JimVanM | s/chance/change |
| 21:44.38 | Silik0n | hah yeah I know |
| 21:44.41 | fwittekind | ariel, that's what I have read. I get cid num, but not cid name. what does a valid cid name message look like in the debug? |
| 21:45.06 | Silik0n | actually an integrated asterisk box would just be the stuff |
| 21:45.11 | ariel_ | fwittekind, if the provider dose not supply the name your not getting it. It cost money to look up names. |
| 21:45.18 | Essobi | fwittekind Umm. depending on where you are, and what kind of line you payed for.. you're not getting the names. |
| 21:45.35 | JimVanM | Essobi: LOL! OK, I said that wrong. I meant an adaptor that'd let you plug in proprietary PBX sets |
| 21:45.38 | fwittekind | ariel_: telco says they looked in the switch, and confirmed it is set to send name. |
| 21:45.59 | *** join/#asterisk kimosabe (~kimo@216.60.60.103) |
| 21:45.59 | Essobi | JimVanM Haha, okay. A PRI card on the PBX makes it all gravy. |
| 21:46.13 | Essobi | A punch down to channel bank is the same. |
| 21:46.36 | JimVanM | Essobi: it's a start, but only if all you want is trunking |
| 21:46.36 | Essobi | I doubt anyone will make anything to plug the phone into directly |
| 21:46.42 | ariel_ | fwittekind, if your setting is right and your getting the number do a line that is exten => s,1,NoOp(${CALLERID}) this will give you both name and number if it's sent. |
| 21:46.54 | Essobi | true |
| 21:46.54 | Essobi | trus |
| 21:46.56 | Essobi | true |
| 21:47.07 | sivana | what's a good cisco 48 port switch? |
| 21:47.17 | Silik0n | 3548 |
| 21:47.22 | ariel_ | sivana, an non cisco brand |
| 21:47.28 | sivana | :) |
| 21:47.33 | fwittekind | ariel_: already did that. just showing number. (btw, this is 1.0.5) |
| 21:47.53 | JimVanM | Essobi: that's what citel did |
| 21:47.54 | *** join/#asterisk TekLexus (~Mnemonic@167.206.75.24) |
| 21:48.09 | ariel_ | fwittekind, I have 1.0.5 and even if your phone gets it incorrect the cli will show it if it's there |
| 21:48.19 | Essobi | PFFFFFFT. |
| 21:48.24 | Essobi | <PROTECTED> |
| 21:48.30 | Essobi | That still won't parse to my 7960. |
| 21:48.43 | Essobi | Anyone got callerID passing to their 7960's right? |
| 21:48.53 | Nugget | sure. |
| 21:48.55 | Nugget | all day long. |
| 21:49.12 | JimVanM | Essobi: you said "gotta EE" - what's EE? |
| 21:49.14 | Nugget | if you see "asterisk asterisk" the problem is not your 7960. |
| 21:49.20 | *** join/#asterisk mitsos (~mitsos@athe730i-1763.otenet.gr) |
| 21:49.22 | Nugget | I guarantee that you cisco is not coming up with that on its own. |
| 21:49.32 | Essobi | Umm |
| 21:49.32 | Essobi | dud |
| 21:49.35 | Essobi | dude |
| 21:49.36 | *** join/#asterisk harryvv (~comming@S010600055d210201.vs.shawcable.net) |
| 21:49.42 | fwittekind | ariel_: yeah, this much I know. are there multiple types of CID name service the telco might use? some supported and some not? |
| 21:49.42 | Essobi | I'm looking at the callerid on the AGI debugs. |
| 21:49.47 | Nugget | the problem is with that perl agi library. |
| 21:49.57 | Essobi | GRRRRR. |
| 21:50.11 | Essobi | 0.8 is a POS? |
| 21:50.11 | ariel_ | Essobi, when setting callerID in asterisk it wants it as "Name "<555-333-2222> |
| 21:50.14 | Essobi | :| |
| 21:50.20 | `Sauron | Sigh. I need to leave my SPA1k at home. |
| 21:50.26 | Essobi | Roger roger! |
| 21:50.27 | Nugget | ariel: the dashes are not required. |
| 21:50.35 | Nugget | nor are the quotes. |
| 21:50.43 | ariel_ | Nugget, right but " " are |
| 21:50.46 | Nugget | no. |
| 21:50.50 | Uajal | Guys, which product would you recommend as autodialer which can detect answering machine (if * cannot yet)? |
| 21:50.53 | `Sauron | I took it in to work this morning, but now I need/want it at home. Blah. |
| 21:51.09 | Nugget | if you look at the code, it grabs whatever's in < > as the number then strips off the quotes if they exist. |
| 21:51.13 | tzanger | Uajal: I'm sure there are telemarketer forums with that information... |
| 21:52.26 | mitsos | does anybody have compiled zaplib with 2.6.* kernel ? |
| 21:52.37 | ariel_ | Nugget, when ever I have had to set Name for callerID I have had to put it as " Name " and the number between <xxxxx> |
| 21:52.51 | kimosabe | tzager u still sellin voip service |
| 21:52.53 | Uajal | I worked on speech recognition and also I can write in C. So what is the best way to make this answ. mach. recogn.? |
| 21:53.07 | *** join/#asterisk TedC (~ted@gray.impulse.net) |
| 21:53.08 | `Sauron | Uajal: Using linux? |
| 21:53.25 | `Sauron | There is none, since IBM discontinued their speech recognition stuff. |
| 21:53.36 | Uajal | I worked C on windows and perl on Free BSD |
| 21:53.39 | ariel_ | fwittekind, I don't have any other responce right now. |
| 21:54.03 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfilp.dialup.mindspring.com) |
| 21:54.58 | Uajal | which files contain the result of fourier (or dct) transform? |
| 21:55.30 | *** part/#asterisk fwittekind (rom@pcp0010183025pcs.columbus.in.indy.comcast.net) |
| 21:57.38 | Essobi | Yea |
| 21:57.49 | Essobi | I copied the example exactly from 0.8's docs |
| 21:57.59 | Nugget | Rock 'n Roll McDonald's? |
| 21:58.01 | Essobi | and it's not setting the number on the chan_sip driver |
| 21:58.04 | Essobi | from the AGI |
| 21:58.08 | Essobi | :| |
| 21:58.15 | kimosabe | if i want to copy somthing with wget for music on hold how do i use wget |
| 21:58.55 | Essobi | ?! |
| 21:58.55 | terrapen | nugget, no, "Jesus is the Answer" |
| 21:58.55 | Essobi | wget url |
| 21:58.56 | terrapen | and it fucking rocks! |
| 21:58.58 | Essobi | haha |
| 21:59.04 | Nugget | rockin'! :) |
| 21:59.06 | kimosabe | thanks |
| 21:59.15 | terrapen | its with his Fiasco band |
| 21:59.16 | bkw_ | you mean to tell me you dont know how to use wget? |
| 21:59.22 | *** join/#asterisk ToyMan (~konversat@user-12lcqq2.cable.mindspring.com) |
| 21:59.32 | terrapen | talk about PBX torture |
| 21:59.38 | bkw_ | thats about like saying "How do you breath?" |
| 21:59.38 | terrapen | you could put Wesley Willis as your MoH |
| 21:59.42 | ManxPower | I don't suppose anyone has experience with Linux Kernel 2.6 and GRE tunnels? |
| 21:59.53 | ManxPower | The lartc.org docs don't work for this. |
| 22:00.00 | Uajal | I need to write the check of the spectrum to determine silence. And if silence is long (to make counter) then it is silence detection. And if long silence appear after 3 seconds - it is person. And if more it is answering machine or smth like this |
| 22:00.01 | terrapen | i could add it to my Telemarketer Torture extension |
| 22:00.07 | MEPHiST0 | manx: poptop? |
| 22:00.28 | terrapen | Play(rockandrollmcdonalds) |
| 22:00.36 | kimosabe | how can i check the music on hold |
| 22:00.39 | harryvv | anyone seen this? Unable to open pid file '/var/run/asterisk.pid': Permission denied |
| 22:00.46 | bkw_ | OMG |
| 22:00.52 | bkw_ | check permissions |
| 22:00.52 | bkw_ | duh |
| 22:00.57 | terrapen | heh harry, come on... |
| 22:00.58 | bkw_ | show application MusicOnHold |
| 22:01.00 | ManxPower | MEPHiST0, What is poptop? |
| 22:01.03 | terrapen | you can figure that one out |
| 22:01.15 | MEPHiST0 | a pptp server for linux (for serving GRE tunnels) |
| 22:01.27 | ManxPower | MEPHiST0, Well I don't want PPTP, I want GRE. |
| 22:01.36 | MEPHiST0 | PPTP utilzed GRE :> |
| 22:01.52 | MEPHiST0 | utilzes, even |
| 22:02.13 | ManxPower | Also I don't see the point in installing a whole new application when you can use the "ip" command included with most distros. See: http://lartc.org/howto/lartc.tunnel.gre.html |
| 22:02.17 | terrapen | Rock over London. Rock on Chicago. Mienekie: You've got a friend in the muffler business. |
| 22:02.35 | terrapen | nugget, did you know that Wesley Willis died? |
| 22:02.38 | ManxPower | GRE tunnels are not rocket science. |
| 22:02.44 | Nugget | no, I hadn't heard. :) |
| 22:02.47 | MEPHiST0 | manx: i guess you're right |
| 22:03.00 | MEPHiST0 | i just wanted to display, in what particular manner i'm in common with GRE |
| 22:03.42 | ManxPower | ONE of the many reasons I want GRE is because you can set it up with three commands on the linux BOX and 1 command on the Cisco box. |
| 22:03.57 | kimosabe | im using asterisk and i just put the music that will play when on hold how can i review that music |
| 22:04.15 | MEPHiST0 | manx: on a pix you can easyly setup gre/pptp with 2 commandlines |
| 22:04.17 | terrapen | put yourself on hold. |
| 22:04.29 | *** join/#asterisk jets (jetsn@omega.pmt.org) |
| 22:04.36 | *** part/#asterisk VivekKhera (~vivek@onceler.kcilink.com) |
| 22:05.12 | bjohnson | kimosabe: ; test music on hold |
| 22:05.12 | bjohnson | exten => 10001,1,Answer |
| 22:05.12 | bjohnson | exten => 10001,2,MusicOnHold(slimp3b) |
| 22:05.12 | bjohnson | exten => 10001,3,Hangup |
| 22:05.39 | bjohnson | I guess you don't really need 3 .. maybe not even |
| 22:05.40 | bjohnson | 1 |
| 22:06.19 | bjohnson | I think I found that in the examples on the wiki |
| 22:06.36 | terrapen | oh word |
| 22:07.02 | kimosabe | bjonson slimp3 b whats the b for back ground |
| 22:07.04 | terrapen | Wesley_Willis_-_Im_Sorry_That_I_Got_Fat_I_Will_Slim_Down.mp3 |
| 22:07.28 | bjohnson | kimosabe: it's the name I defined in musiconhold.conf .. use the one that you want to test |
| 22:07.56 | kimosabe | thnkas |
| 22:08.48 | *** join/#asterisk file[laptop] (~file_lapt@mctn1-142166197096.nb.aliant.net) |
| 22:09.05 | *** join/#asterisk Grooby (~Grooby@66.160.105.186) |
| 22:09.16 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 22:09.41 | kimosabe | bjohnson can u tell me thi e link for that please |
| 22:10.10 | bjohnson | for what? the wiki? |
| 22:10.12 | bjohnson | ~docs |
| 22:10.15 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 22:10.31 | Essobi | FFS. |
| 22:10.33 | Essobi | :\ |
| 22:11.01 | Essobi | This is driving me nuts. |
| 22:11.12 | Essobi | ManxPower You around? |
| 22:12.32 | Essobi | ~agi |
| 22:12.33 | jbot | i guess agi is the Asterisk Gateway Interface... similar to CGI for web applications AGI lets you script call control and access databases using your favorite language. AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages |
| 22:12.37 | Essobi | $newcallerid = "\"$name <($npa) $nxx-$station>\""; |
| 22:12.45 | Damascene | wow that's awesome! |
| 22:12.50 | Essobi | err.. |
| 22:12.51 | Essobi | <PROTECTED> |
| 22:12.58 | Essobi | I swear that looks right. |
| 22:13.19 | Essobi | BOB shows up the the chan_sip but the number doesn't parse to my 7960. |
| 22:13.53 | Essobi | I'm running head and the latest 7960 image. |
| 22:14.10 | epoch | what is the latest 7960 sip version, anyways? |
| 22:14.18 | mtqh | 7.3 |
| 22:14.27 | Essobi | I swear I'm going to drop kick it. |
| 22:14.46 | epoch | ah |
| 22:14.50 | epoch | guess I'm behind ;/ |
| 22:14.50 | epoch | heh |
| 22:15.09 | brc_ | move the quote |
| 22:16.00 | anthm | <> should only contain digits |
| 22:16.34 | Essobi | I'm looking at tcpdumps of the sip to my phone |
| 22:16.41 | Essobi | and the phone number isn't in the packets. |
| 22:16.42 | Essobi | :\ |
| 22:16.48 | Grooby | anyone know why i can't get firefly to run on iax2 w/ ilbc codec? |
| 22:16.54 | brc_ | Essobi, uh...see above? |
| 22:17.05 | Essobi | Oh. |
| 22:17.07 | `Sauron | Anyone here one of the devs who works on realtime? |
| 22:17.16 | Essobi | The one I'm using is from the 0.8 perl AGI examples |
| 22:17.40 | *** join/#asterisk carlos-d-man (mdkuser@201.128.216.236) |
| 22:17.42 | Essobi | <PROTECTED> |
| 22:17.49 | sivana | exit |
| 22:17.50 | Essobi | Same thing.. FROM Bob Asterisk |
| 22:18.05 | *** join/#asterisk sezuan (sezuan@port-212-202-57-119.dynamic.qsc.de) |
| 22:18.09 | anthm | <> should only contain digits ? |
| 22:18.20 | Essobi | <PROTECTED> |
| 22:18.23 | Essobi | Same thing |
| 22:18.29 | Essobi | FROM Bob Asterisk |
| 22:18.30 | *** join/#asterisk sivana (~sivana@165.154.13.35) |
| 22:18.49 | Essobi | Anyway I can throw AGI into debug to I can see what's getting set on the AGI? |
| 22:18.50 | Godsey | I just noticed macro-stdexten |
| 22:18.54 | *** join/#asterisk roamer323 (~sing@HSE-MTL-ppp72652.qc.sympatico.ca) |
| 22:18.56 | Essobi | maybe it's a bug.. |
| 22:18.58 | Godsey | do I call it in dial plan like: |
| 22:19.07 | Godsey | Macro(macro-stdexten,SIP/1001)? |
| 22:19.16 | brc_ | depends |
| 22:19.21 | cbachman | Essobi: the code seems to hint that the AGI just ends up calling ast_set_callerid which doesn't do any processing itself. What does the cdr show? or the manager event interface? |
| 22:19.22 | brc_ | on how you have the macro setup |
| 22:19.28 | Godsey | sample configs |
| 22:19.41 | Godsey | exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20$ |
| 22:19.42 | *** join/#asterisk firstsword (~root@host6614613596.biz.tor.fcibroadband.com) |
| 22:19.42 | brc_ | use the sample macro line then... |
| 22:19.48 | Godsey | I don't see one :) |
| 22:19.53 | brc_ | look harder |
| 22:20.02 | brc_ | I know it's in extensions.conf.sample |
| 22:20.07 | Godsey | doh! |
| 22:20.10 | *** join/#asterisk kietlak (~kietlak@11-mo3-6.acn.waw.pl) |
| 22:20.15 | Godsey | you don't have macro- in all places :) |
| 22:20.16 | Godsey | ok |
| 22:20.22 | firstsword | Hi... have anyone tried to fresh linksys wrt54g with openwrt? |
| 22:20.31 | Godsey | exten => 1234,2,Macro(stdexten,1234,${CONSOLE}) |
| 22:20.31 | *** join/#asterisk Tall-guy (tall-guy@hssxrg207-195-103-110.sasknet.sk.ca) |
| 22:20.34 | firstsword | i mean.. anyone here tried? |
| 22:20.36 | Godsey | so then 1001,SIP? |
| 22:20.39 | Grooby | hmm..i feel like a retard now |
| 22:20.57 | Tall-guy | Hey lads, quick analog question here, is CallerID polarity sensitive in N.America (Tip/Ring?) |
| 22:21.09 | CoaxD | Tall-guy: Absolutely not |
| 22:21.16 | Tall-guy | coax: I didn't think so.... |
| 22:21.26 | Godsey | the part I'm not following is extension and device to ring |
| 22:21.31 | CoaxD | Tall-guy: Remember, callerid is nothing more than a 1200baud carrier |
| 22:21.34 | CoaxD | (or is 300? I forget) |
| 22:21.55 | Tall-guy | now I have the unenviable task of proving to my telco that they aren't sending callerid down one of my trunks :) |
| 22:22.05 | CoaxD | Tall0-guy: Are you absolutely sure? |
| 22:22.18 | Tall-guy | coax: swapped the lines on my cards......problem followed the line, not the Zap channel |
| 22:22.31 | CoaxD | Tall-guy: That doesnt prove a thing |
| 22:22.32 | Tall-guy | u don't think so? |
| 22:22.33 | Tall-guy | why not? |
| 22:22.51 | brc_ | Godsey, it works like this |
| 22:22.55 | CoaxD | Tall-guy: What happens if 1 line needs an rxgain of -7.0 in order to properly hear fsk, but the other line only requires 0.0? |
| 22:22.58 | Godsey | ok I see |
| 22:23.01 | CoaxD | Tall-guy: (i.e. new wiring) |
| 22:23.03 | Godsey | I just need to use 1001,SIP/1001 |
| 22:23.08 | shido6 | then set it for that channel, CoaxD |
| 22:23.15 | CoaxD | shido: ABsolutely |
| 22:23.17 | brc_ | Macro(macroname,whateversheregoesinto ARG1,hereis ARG2, here is ARG3) |
| 22:23.34 | CoaxD | shido: But the point is, if he hasn't hooked up a soundcard and LISTENED FOR AN FSK, he wouldnt know if one was being sent or not |
| 22:23.37 | Godsey | thanks brc: my mind was stuck on extension == what to ring |
| 22:23.44 | CoaxD | shido: ie.. with ztmonitor |
| 22:23.45 | shido6 | soundcard? |
| 22:23.47 | brc_ | and when you write the macro context the name must start with macro- |
| 22:23.49 | Godsey | so when it split it into 2 parts I was mind phucked |
| 22:23.50 | shido6 | oh |
| 22:23.55 | Tall-guy | coaxd: I was going from on the assumption that my 4 trunks from my telco would be pumping out approximately same needs for rx gain |
| 22:23.57 | CoaxD | shido: I had to do this on friday |
| 22:23.58 | shido6 | or just make a call through the zaptel |
| 22:23.58 | CoaxD | er saturday |
| 22:23.59 | shido6 | :) |
| 22:24.05 | CoaxD | Tall-guy: Please dont ever assume that |
| 22:24.15 | CoaxD | Tall-guy: Chances are, you're right. But don't assume it |
| 22:24.27 | Tall-guy | coaxd: hmm, ok...let me play devils advocate here...... |
| 22:24.29 | CoaxD | Tall-guy: They could have a tansmit gain setting set wrong at the telco, too, and you'd never know it |
| 22:24.38 | CoaxD | tallguy: err, transmit |
| 22:24.42 | ionix | hey |
| 22:24.51 | CoaxD | Tall-guy: Thus, problem still follows line, not card |
| 22:24.51 | ionix | anyone has access to the RBOC reverse ANI db ? |
| 22:24.57 | Tall-guy | coaxd: the cheesy little callerid devices one buys from radioshack....they don't "auto adjust" rx/tx gain to lines right? |
| 22:25.09 | CoaxD | Tall-guy: Yes, they absolutely do |
| 22:25.13 | Essobi | cbachman It's showing up in the CLID of the CDRs. |
| 22:25.25 | Tall-guy | coaxd: izzat so? well color me informed :) |
| 22:25.31 | CoaxD | Tall-guy: Thats how they DONT have problems whereas asterisk, the "dumber" of the two, DOES |
| 22:25.39 | brc_ | yup |
| 22:25.44 | Tall-guy | coaxd: makes sense.... |
| 22:26.01 | Tall-guy | coaxd: so lets pretend for a second my telco is good, and my gains are bad...... |
| 22:26.05 | CoaxD | Tall-guy: Would you believe that this weekend, I found that I had to set my rxgain at -7.5 to make it work? |
| 22:26.10 | cbachman | Essobi, if it shows up in the cdrs correctly as to what the AGI is setting, then it sounds like the AGI->asterisk path is working at least |
| 22:26.25 | CoaxD | Tall-guy: I set up a sound card, and ran ztmonitor. I heard nothing but junk. I turned down the rxgain until it sounded reasonable |
| 22:26.32 | CoaxD | Tall-guy: Funny how when i did that, all my local echo went away |
| 22:26.52 | CoaxD | Tall-guy: And the phone sounded GOOD for a change |
| 22:27.02 | Tall-guy | coaxd: interesting.....apart from trial and error......is there a smarter way to do it with ztmonitor and a certain # of "****" ???? |
| 22:27.11 | CoaxD | Tall-guy: But i wouldn't have known that until i hooked up a soundcard and got alsa working for ztmonitor |
| 22:27.14 | kimosabe | some one im trying to configure music on hold im in music on hold.conf i add the rouyte but the music will not play |
| 22:27.16 | jalsot | does anybody know how does ztmonitor exactly work? What should it show if the monitored channel is inactive (no call on it)? the last state? |
| 22:27.20 | CoaxD | Tall-guy: No. |
| 22:27.25 | CoaxD | Tall-guy: You must listen to the FSK |
| 22:27.37 | CoaxD | Tall-guy: It sounds a lot like the first part of a modem connection. |
| 22:27.48 | Essobi | cbachman So what? SIP -> 9760 is broxored? |
| 22:27.50 | Essobi | :| |
| 22:27.58 | CoaxD | Tall-guy: I tried trial and error and failed. Why? Because I didnt know WHAT the problem was. (was rxgain too LOW? too HIGH? I could've been at it for years testing that.) |
| 22:27.59 | Essobi | You'd think more people running -head would bitch then. |
| 22:28.16 | Tall-guy | coad: so, if yer listening, what are you listening for? (ie: whats good/bad?) |
| 22:28.21 | CoaxD | Tall-guy: ztmonitor (without -v) is the way to go. it was SO easy once i got a sound card working |
| 22:28.30 | CoaxD | Tall-guy: Listen for a "modem connection" sound between the first ring and the second |
| 22:28.49 | Tall-guy | coaxd: ok....no modem connection sound = no CID info being received.... |
| 22:28.59 | CoaxD | Tall-guy: CORRECT |
| 22:29.04 | Essobi | FFS, This is driving me nuts. |
| 22:29.14 | jalsot | CoaxD: when I hangup the channel, will ztmonitor show the last state of the channel? |
| 22:29.15 | Tall-guy | coaxd: but how much of my "listening" is going to be dependant on rx/tx gain levels, and how much is going to be dependant on the volume control settings in my alsamixer ????? |
| 22:29.28 | cbachman | Essobi, perhaps. The other thing to look at is if it works if you set it in the dialplan itself. |
| 22:29.31 | CoaxD | Tall-guy: You'll just have to get a feel for it :) |
| 22:29.40 | Tall-guy | coaxd: fair enuf....I'll give it a whirl |
| 22:29.45 | CoaxD | tall-guy: I had that same problem that you just described |
| 22:29.52 | Tall-guy | (and its nice to talk to someone who can type as fast as me! |
| 22:29.56 | CoaxD | tall-guy: I did however listen to an mp3 to verify "normal sound levels" |
| 22:30.01 | CoaxD | Tall-guy: Hehe, oh man, i hearya |
| 22:30.09 | Tall-guy | double grin |
| 22:30.09 | CoaxD | Tall-guy: Too many slow typists. *g* |
| 22:30.29 | CoaxD | Tall-guy: So, i listened to an mp3, got it to comfortable levels, then played the line sounds with ztmonitor |
| 22:30.32 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 22:30.34 | CoaxD | Holy *SHIT* they were loud compared to that mp3 |
| 22:30.37 | CoaxD | and overdrivin |
| 22:30.39 | CoaxD | er driven |
| 22:30.47 | *** part/#asterisk dg1nsw (~dg1nsw@saturn2.franken.de) |
| 22:30.49 | Tall-guy | coaxd: (i didn't know that ztmonitor could output audio on the console...I've only used it on a ssh shell) |
| 22:30.53 | CoaxD | i kept cranking the rxgain down until it sounded about right - like the mp3 i played |
| 22:31.00 | CoaxD | tall-guy: Ohyes, just dont use -v |
| 22:31.48 | Tall-guy | coaxd: cool......and there's no reason to compare "MY" settings with "your" settings as our lines are completely different telcos.... |
| 22:31.58 | CoaxD | Tall-guy: Correct. Absolutely |
| 22:32.10 | Tall-guy | (I'm at rx 9, tx -2) on my pstns |
| 22:32.12 | CoaxD | Tall-guy: The reaosn i had to set my receive gains down so low is because the telco is what, 200 feet from my building? |
| 22:32.21 | CoaxD | tall-guy: Hmmm. That sounds high in ANY regard |
| 22:32.41 | Tall-guy | coaxd: when I dropped them too low, the voicemails being LEFT by callers were inaudible.... |
| 22:32.53 | CoaxD | Tall-guy: Oh god, dude, you must be 100 miles from the telco |
| 22:33.08 | Tall-guy | coaxd: i'd guess about 3KM |
| 22:33.14 | CoaxD | Tall-guy: Hmm. |
| 22:33.21 | Tall-guy | coaxd: but I'll use your methods.......see what I get.... |
| 22:33.31 | CoaxD | tall-guy: Yeah, i bet you got 1 line producing way better than the other |
| 22:33.49 | Tall-guy | coaxd: that'd be my guess.... |
| 22:33.49 | greg_work | CoaxD: my CO is probably 60 ft from my building. but the lines actually travel WAY up past the highway and down, i think its probably 2-3km ;P |
| 22:33.49 | CoaxD | tall-guy: And you'll need to split your rxgain/txgain settings per line |
| 22:33.59 | CoaxD | greg: hehe *g* |
| 22:34.00 | greg_work | (we share a parking lot with the CO actually) |
| 22:34.14 | CoaxD | greg: I can see my 50pair cable trapsing over the poles to the telco. :) |
| 22:34.23 | greg_work | :) |
| 22:34.44 | Tall-guy | coaxd: yup, more work :) |
| 22:35.12 | greg_work | apparently our building was the community phone center in the 70's, so it has a big trunk line coming down from their main switching location or something (they have new lines now that run from there to the CO) |
| 22:35.16 | wankel | coax! |
| 22:35.31 | CoaxD | wankel! |
| 22:35.56 | greg_work | we also have 4 2" conduits in our electrial room that go back out to the bottom of the pole .. they've pulled out whatever was in there though and now there just a couple 3 pair lines |
| 22:36.00 | dsmouse | wankel ?? |
| 22:36.13 | wankel | you too? |
| 22:36.30 | CoaxD | wankel: What you been up to, sir? |
| 22:36.42 | dsmouse | wankel: me too? |
| 22:36.53 | dsmouse | heheh |
| 22:37.12 | wankel | jontow was in here the other day |
| 22:37.44 | *** join/#asterisk Derkommissar (~Loving@66.64.215.7.nw.nuvox.net) |
| 22:37.45 | Tall-guy | coaxd: so just a random ztmonitor on my pstn lines (no call active), sound like CRAP (static/noisy), and my internal fxs lines (no calls) are nice and quiet |
| 22:38.02 | Derkommissar | does anyone here know how to ger around GNUGK ? |
| 22:38.32 | Grooby | this is really weird.... |
| 22:38.50 | Grooby | firefly => iax2 => asterisk using ilbc gets static noise |
| 22:38.56 | CoaxD | Tall-guy: MMmmhmmm :) |
| 22:38.58 | Grooby | but ulaw codec works |
| 22:39.39 | CoaxD | Tall-guy: You should set your rxgain/txgain separately for your FXS ports. Obviously, there's a BIG difference between them and your POTS lines ;) |
| 22:40.00 | CoaxD | Tall-guy: BTW, the RING will ALWAYS sound staticky and crappy. Its 90v of pure unadulturated voltage love |
| 22:40.13 | CoaxD | Tall-guy: But the actual SOUND of the live call? Thats different |
| 22:40.42 | Tall-guy | coaxd: my fxs and fxo are separated...by the way....but yeah, I hear ya. |
| 22:40.48 | CoaxD | Tall-guy: For some reason, my asterisk likes to open /dev/dsp regardless if i want it to or not. So, in order to use ztmonitor, i have to stop asterisk, load ztmonitor FIRST, and then START asterisk - so it errs out on /dev/dsp (due to busy) |
| 22:41.06 | CoaxD | Tall-guy: The thign to do will be to set your receive gain way down on your pots lines, and your transmit gain way UP on your handsets |
| 22:41.22 | Tall-guy | coaxd: yup.....I'm pickin up what yer laying down.... |
| 22:41.35 | CoaxD | Tall-guy: I hate these problems. I mean truely *HATE* |
| 22:41.55 | CoaxD | Tall-guy: If asterisk hardware could be somewhat intelligent about gain detection, it would just be so much easier. (but i certainly understand why they cant do it) |
| 22:42.09 | Tall-guy | coaxd: I'll do some monkeying around late today and tomorrow, I'll keep u posted if you give a damn |
| 22:42.19 | CoaxD | Tall-guy: I do give a damn, actually |
| 22:42.29 | wankel | mmm, ring voltage. |
| 22:42.39 | CoaxD | Tall-guy: I fought with these problems for months until the weekend |
| 22:42.40 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 22:42.48 | jgaviria | somebody has succefully installed spandsp? |
| 22:42.54 | CoaxD | Tall-guy: i finally bit the bullet, got alsa going on the box, a set of speakers over there, and had a listen |
| 22:43.12 | Tall-guy | coaxd: yeah, I got the speakers goin now :) |
| 22:43.27 | CoaxD | Tall-guy: it took me a while that day as i had to burn off all our recorded calls too. *lol* |
| 22:43.38 | Tall-guy | :) |
| 22:43.46 | CoaxD | tall-guy: plus i had to futz with cmos to get all irqs right. Its one of those boards that just sucks for irq sharing. |
| 22:43.53 | CoaxD | finally gave up and let it share the farking irq. *lol* |
| 22:44.18 | CoaxD | tall-guy: We just have a small store hooked up to asterisk for the purposes of recording phone calls, and also, our own voicemail system |
| 22:45.04 | CoaxD | tall-guy: oh yeah. and another asterisk server for our voip stuff. (my sister runs an office in the same building that is solely done with voip to NuFone) |
| 22:46.58 | Tall-guy | coaxd: well, my first test gave me NO modem connection sound on the line in question...but I heard it clearly on all the other channs.....I'm leaning towards the telco, as the other parts of the call volume sounded similarly loud |
| 22:48.53 | Grooby | hmmmmmm |
| 22:49.02 | Grooby | can asterisk transcode via iax2? |
| 22:49.07 | tzanger | Grooby: of course |
| 22:49.31 | dsmouse | Grooby: do you know what iax stands for? |
| 22:49.34 | Tall-guy | coaxd: I'm outta here....wife / supper/valentines = more fun than ztmonitor.....chat tomorrow :) |
| 22:49.47 | *** join/#asterisk magiconair (~chatzilla@pD9E262F9.dip.t-dialin.net) |
| 22:49.56 | CoaxD | Tall-guy: it very easily could be that. |
| 22:50.02 | CoaxD | Tall-guy: (i.e. the telco aint sending cid) |
| 22:50.14 | Tall-guy | coaxd: yup, methinks. |
| 22:50.14 | Grooby | inter asterisk exchange |
| 22:50.20 | CoaxD | Tall-guy: Have a great night! yes, wife / supper / valentines is MUCH more fun :) |
| 22:50.20 | Grooby | i am asking because I am having a little problem |
| 22:50.29 | Grooby | I am not sure if it's firefly problem or what |
| 22:50.30 | Tall-guy | coaxd: later dude, thanx for all yer help. |
| 22:50.34 | Grooby | i force firefl to use ilbc |
| 22:50.38 | CoaxD | Tall-guy: I have none of the above, so.. i'll settle for telling a good friend of mine happy valentines etc :) |
| 22:50.46 | Grooby | and when I dial out, i get noises.... |
| 22:50.53 | Grooby | but when I dial from landline to my firefox |
| 22:50.55 | *** part/#asterisk Tall-guy (tall-guy@hssxrg207-195-103-110.sasknet.sk.ca) |
| 22:50.57 | Grooby | i get perfect sound |
| 22:51.07 | Grooby | and going from landline to firefox the codec stays ulaw |
| 22:51.24 | Grooby | and firefox to landline is ilbc (i think..let me 2x check) |
| 22:51.56 | *** join/#asterisk eborn (~eborn@linda.fambus.nl) |
| 22:54.44 | eborn | hi, i'm trying to get dtmf caller id working with my X100P. I used ztmonitor to check for a FSK signal, and it's there: http://www.fambus.nl/~ebus/test.wav (6 secs into the audio file) |
| 22:54.50 | eborn | but asterisk doesn't detect it |
| 22:55.00 | eborn | the clidtest utility i found somewhere doesn't detect it either |
| 22:56.49 | eborn | i have a few settings in my zapata.conf: usecallerid = yes, cidsignalling = dtmf, cidstart = polarity, hidecallerid = no, callerid = asreceived |
| 22:57.00 | *** join/#asterisk dsfr (~dsfr@216.207.244.183) |
| 22:57.22 | *** join/#asterisk Jlau515 (~jackie@global-sf.keen.com) |
| 22:57.43 | *** join/#asterisk TekLexus (~Mnemonic@206.231.230.230) |
| 22:59.41 | Jlau515 | hi can some one help me with configuring my zapata card |
| 23:01.18 | terrapen | try the wiki, Jlau515 |
| 23:01.27 | CoaxD | Jlau515: That depends. Do you have a credit card? :) |
| 23:02.17 | *** join/#asterisk ipso (~ipso@d207-81-249-35.bchsia.telus.net) |
| 23:02.18 | wasim | Jlau515: which card? |
| 23:02.41 | *** join/#asterisk thefallen (PolarBear@thefallen.user) |
| 23:02.58 | ipso | Is there a way with Asterisk to setup SIP groups like Zap groups for dialing out? |
| 23:03.03 | jets | Gents any reason asterisk manager api wouldn't be showing caller id information, as well as manager apps like FOP, or RingXP? |
| 23:03.10 | Jlau515 | te110p |
| 23:03.12 | jets | I am originating a SIP call phone to phone |
| 23:03.26 | CoaxD | wasim: Farfon in mass production yet? Or do i still have to send money to a pakistani address and hope and pray for a phone someday? |
| 23:03.28 | wasim | Jlau515: should be simple enough, have you checked the wiki? |
| 23:03.37 | Jlau515 | yes |
| 23:03.46 | Jlau515 | when i run ztcfg -vv |
| 23:03.47 | wasim | CoaxD: well, we've got 30 odd in EU and soon some in US |
| 23:03.54 | Jlau515 | i see that its configure for 24 channels |
| 23:03.58 | thefallen | jets I think you have to set the callerid field for each user. |
| 23:04.01 | Jlau515 | but when i run asterisk |
| 23:04.06 | wasim | CoaxD: of course, nothing stops you from sending money to me :) |
| 23:04.08 | CoaxD | wasim: Ah |
| 23:04.11 | CoaxD | wasim: hehe |
| 23:04.12 | Jlau515 | it says that channel 16 is used for D-channel |
| 23:04.19 | Jlau515 | then craps out |
| 23:04.20 | jets | i have callerid= set on each user in sip.conf |
| 23:04.21 | CoaxD | wasim: I'm just wondering if yer ever gonna mass-produce them and get them out to market |
| 23:04.22 | thefallen | jets at least that's how I got it working :) |
| 23:04.25 | *** join/#asterisk ManxPwr (~eric@dsl-209-205-172-111.i-55.com) |
| 23:04.30 | CoaxD | wasim: i.e. so i can go to digitnetworks and order one or something. hehe |
| 23:04.37 | Grooby | grrr |
| 23:04.43 | Grooby | i am stummped |
| 23:04.53 | wasim | CoaxD: we probably are not going to directly mass produce them, but do them on behalf of someone, we're a core r&d shoppe |
| 23:05.02 | CoaxD | wasim: Ahhh |
| 23:05.15 | CoaxD | wasim: So basically, sell the "selling rights" to someone |
| 23:05.19 | wasim | CoaxD: we're also licensing it out, so others can as well, and a lot of the stuff is open source |
| 23:05.22 | CoaxD | wasim: And gain royalties on the R&D |
| 23:05.40 | CoaxD | wasim: Sure. Your firmware has got to be open source. it took you wayyy too little time to develop that |
| 23:05.47 | CoaxD | wasim: (i.e. the base OS for your firmware) |
| 23:05.48 | Grooby | woah! |
| 23:05.54 | Jlau515 | this is the error: Channel 16 is reserved for D-channel. |
| 23:05.54 | Jlau515 | Unable to register channel '1-23' |
| 23:06.08 | Grooby | can someone help me test something? |
| 23:06.19 | Jlau515 | chan_zap.so: load_module failed, returning -1 |
| 23:06.24 | ManxPwr | Jlau515, You are in the USA? |
| 23:06.28 | Jlau515 | correct |
| 23:06.33 | Grooby | whoever is using iax2, please dial 1800-456-3355..it's Dell's 800 number |
| 23:06.36 | ManxPwr | Jlau515, your card is configured for E-1 |
| 23:06.40 | Grooby | i get pure static when dialing that number |
| 23:06.43 | wasim | CoaxD: actually, we'd had a lot of the core network stacks and other OS stuff from before, just had to do the gsm/iax bits |
| 23:06.44 | Grooby | using ilbc |
| 23:06.50 | Grooby | but i just called my work and it works fine |
| 23:06.59 | wasim | CoaxD: and optimizations |
| 23:07.06 | CoaxD | wasim: Ah, you designed all the firmware stuff in previous adventures |
| 23:07.15 | *** join/#asterisk w0w0 (~apardo@80.26.162.178) |
| 23:07.16 | wasim | CoaxD: not all, i'd say 40% |
| 23:07.30 | CoaxD | wasim: Yeah, but that got you past needing to write a whole OS from scratch |
| 23:08.07 | CoaxD | wasim: and of course, then you just have to integrate whatever protocol you use, as well as the iax2 stuff to sit on top of it. Not near as bad as doing the OS design work |
| 23:08.30 | CoaxD | wasim: Internally, is your OS a 'unixy' type thing? or is the whole thing done in 1 thread? |
| 23:08.33 | wasim | CoaxD: actually, we did do another OS from scratch, just to test some other ideas |
| 23:09.08 | CoaxD | wasim: Sure. The idea is, though, it wouldnt be as complex as, say, rewriting the linux kernel. you don't really need a truely multitasking OS for this purpose. You can pretty much loop through everything. |
| 23:09.16 | CoaxD | wasim: (since the design is limited-functionality) |
| 23:09.33 | wasim | CoaxD: yep, but using the dsp as a host processor is fun in itself |
| 23:09.46 | CoaxD | wasim: hehe |
| 23:09.58 | CoaxD | wasim: Well, really, all you ahve to do is tell it to act like a processor, and it will |
| 23:10.09 | CoaxD | wasim: (no small task, i'm sure) |
| 23:10.14 | CoaxD | wasim: I dont know anything about it. I've never had to do it |
| 23:10.27 | wasim | CoaxD: i was lucky to have some very very smart people |
| 23:10.50 | wasim | CoaxD: with decades of experience between them, so it was a treat |
| 23:10.51 | CoaxD | wasim; I do know that I worked with FEP/OS, the operating system that the intelligent rs232 Digiboards run |
| 23:11.07 | CoaxD | wasim: i.e. the actual firmware code.. THat was nifty |
| 23:11.26 | CoaxD | wasim: Yeah, its tough stuff. But if they've got the experience, they probably had the ideas to start with of how they were gonna design it, and then did so. |
| 23:11.28 | wasim | CoaxD: well, i'm trying to push for open sourcing a fair bit soon, as promised |
| 23:11.46 | CoaxD | wasim: Sweet. So, how do you intend to make back the money you put into the design? |
| 23:11.54 | *** join/#asterisk clh (~jjg@64.241.37.140) |
| 23:11.58 | CoaxD | wasim: support for the product? Or? |
| 23:12.06 | CoaxD | wasim: or was this your 'gift to the community'? |
| 23:12.07 | wasim | CoaxD: we don't, being bankrupt has its charm as well |
| 23:12.13 | *** join/#asterisk heath__ (~heath@12-215-32-191.client.mchsi.com) |
| 23:12.22 | clh | anybody got any tips on trying to connect from panera to a * at home that 5060 has been forwarded to? |
| 23:12.25 | CoaxD | wasim: hehe |
| 23:12.28 | clh | i'm having RTP trouble |
| 23:12.33 | CoaxD | wasim: (Its a good tax writeoff, i'm sure) |
| 23:12.40 | wasim | CoaxD: actually, a licensing option on the firmware |
| 23:12.58 | CoaxD | wasim: Ahh. So the hardware design is open sourced; the firmware is licensed |
| 23:13.05 | CoaxD | wasim: That in and of itself aint a bad idea |
| 23:13.18 | CoaxD | wasim: It allows people to build on your design whilst paying out the nose for your firmware |
| 23:13.35 | wasim | CoaxD: some bits we're gonna keep, no, no, not through the nose at all |
| 23:13.43 | CoaxD | wasim: Cool |
| 23:14.02 | CoaxD | wasim: I suppose also that you guys have a fair bit of business regardless of this iax2 phone thing |
| 23:14.06 | wasim | CoaxD: our next challenge is to fill the firmware with features, right now its pretty basic |
| 23:14.13 | *** part/#asterisk yogurt2ungue (~charlie@203-132-126-200.fibertel.com.ar) |
| 23:14.17 | CoaxD | wasim: Sure. Supporting sip, for instance, might not be a bad idea |
| 23:14.36 | wasim | CoaxD: no, that would be contrary to our beliefs |
| 23:14.44 | *** join/#asterisk jsandnes (cryzeck@sms.trangest.no) |
| 23:14.49 | wasim | CoaxD: which is that SIP sucks |
| 23:14.55 | CoaxD | wasim: Sip does suck. |
| 23:15.01 | wasim | CoaxD: indeed |
| 23:15.10 | CoaxD | wasim: However, the thing is, not everybody is gonna be wanting to use your stuff with asterisk |
| 23:15.28 | wasim | CoaxD: they can use it with Yate! |
| 23:15.30 | CoaxD | wasim: They'll say.. "OOOH, COOL IP PHONE! Can i hook it up to my <insert proprietary thingie here> |
| 23:15.32 | CoaxD | " |
| 23:15.42 | wasim | CoaxD: sure, if that talks IAX |
| 23:15.45 | CoaxD | wasim: What is Yate? a IAX2 to <insert protocol here> proxy? |
| 23:15.54 | terrapen | its amazing how bad the engrish is in these phishing scams |
| 23:15.58 | wasim | CoaxD: no, its l-fy's switch |
| 23:16.04 | CoaxD | wasim: What is that? |
| 23:16.15 | CoaxD | terrapen: Yeah, you better believe it |
| 23:16.29 | CoaxD | terrapen: I applied for an account through USBank the other day. they have an entire pamphlet entitled "PHISHING" |
| 23:16.34 | dsmouse | terrapen: ??? |
| 23:16.35 | jsandnes | Hey guys, I need some help here :-) - I'm working on a open source Web management interface for asterisk, but i would need some tips from a few people what I should include in that interface, what features people want, and I need a good name :-) |
| 23:16.46 | CoaxD | terrapen: Every single account holder gets this pamphlet when they sign up |
| 23:17.31 | terrapen | i shouldn't have deleted this one |
| 23:17.33 | terrapen | it was funny |
| 23:17.40 | *** part/#asterisk clh (~jjg@64.241.37.140) |
| 23:17.43 | CoaxD | i never delete anything. even spam |
| 23:17.47 | terrapen | "Thank you for your cooperations." |
| 23:17.47 | CoaxD | i have gigs and gigs of mail |
| 23:18.01 | *** join/#asterisk brycec (~brycec@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 23:18.05 | CoaxD | the only reason I delete anything is if i mail myself a large attachment, or a large attachment that i only need once - has been received |
| 23:18.10 | CoaxD | beyond that, everything is kept. Period. |
| 23:18.13 | Jlau515 | how do i configure my zapatel card from e1 to t1 |
| 23:18.29 | CoaxD | jlau: You read the documentation. Thats how. |
| 23:18.30 | wasim | Jlau515: its the signalling in zaptel.conf |
| 23:18.47 | Jlau515 | in /etc/zaptel.conf i believe its set to t1 |
| 23:18.54 | CoaxD | jlau: I haven't read the documentation. But within 10 minutes after doing so, i can guarantee you I would know |
| 23:21.51 | *** join/#asterisk clh (~jjg@64.241.37.140) |
| 23:22.28 | wasim | Jlau515: what part of span=<span num>,<timing>,<line build out (LBO)>,<framing>,<coding>[,yellow] have you set to t1 ? |
| 23:23.01 | Jlau515 | span=1,1,0,esf,b8zs |
| 23:23.54 | *** join/#asterisk BradWaite (~root@66.180.106.106) |
| 23:24.17 | Jlau515 | the framing and coding are correctly set for t1 |
| 23:24.27 | ManxPwr | Jlau515, chekc the jumpers on the board to make sure they are set to T-1. |
| 23:24.48 | Jlau515 | there's jumpers |
| 23:24.50 | BradWaite | brb |
| 23:24.55 | Jlau515 | didnt know what |
| 23:24.56 | Jlau515 | that |
| 23:25.19 | ManxPwr | Jlau515, sometimes cards go out from Digium that are set for E-1 rather than T01 |
| 23:25.20 | ManxPwr | T-1 |
| 23:25.28 | *** join/#asterisk voxlinx (~voxlinx@66.180.106.106) |
| 23:25.31 | voxlinx | better |
| 23:25.34 | ManxPwr | Jlau515, I think you can override the jumbers somewhere in software, but I don't know where. |
| 23:25.43 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@d01m-4-110.d4.club-internet.fr) |
| 23:25.49 | voxlinx | Wow... channel's grown since I was here last |
| 23:26.15 | Jlau515 | yeh card is set to E1, thxs manxpwr |
| 23:26.35 | Jlau515 | jumper was there |
| 23:27.36 | jets | hrm am i dumb or is callerid on the manager api horked |
| 23:27.41 | jets | any fop users with the most recent cvs? |
| 23:27.52 | voxlinx | hey guys.. can't seem to send audio via my x10-* 1.0-PRI. |
| 23:27.57 | ManxPwr | jets, yes. |
| 23:28.17 | ManxPwr | jets, Fixed in 1.0.x CVS stable. |
| 23:29.02 | jets | I'm using head I believe. |
| 23:29.08 | jets | still broken in head? |
| 23:29.25 | voxlinx | hey manx, anything you can think of in a * or Zap config that'd kill outbound audio on a PRI? |
| 23:29.29 | ManxPwr | jets, no idea. I run production systems and so don't use the developement version of Asterisk |
| 23:29.36 | ManxPwr | voxlinx, no. |
| 23:30.30 | Grooby | doo dee doo dee doo |
| 23:30.31 | *** join/#asterisk seva (seva@sevatech.com) |
| 23:30.49 | Grooby | anyone here uses firefly with asterisk? |
| 23:33.48 | seva | i have an incredibly simple IAX peer (VoIPJet) and IAX user (iaxcomm soft phone) setup: http://sevatech.com/tmp/asterisk |
| 23:34.24 | seva | whenever i try to dial 12223334444 i get "Rejected connect attempt from 1.2.3.4, request '12223334444@default' does not exist |
| 23:34.55 | seva | i am missing something incredibly simple (first * attempt), any suggestions? |
| 23:35.06 | greg_work | ManxPwr: i get jumped on whenever i say i don't want to run HEAD in production |
| 23:35.06 | *** join/#asterisk hajekd (~hajekd@21.208.65.212.contactel.net) |
| 23:35.08 | ariel_ | seva, what is your dialing command |
| 23:35.34 | seva | i use iaxcomm and enter "12223334444" as the phone number |
| 23:36.26 | *** join/#asterisk obiyoda (~Jared@70-58-164-219.bois.qwest.net) |
| 23:36.55 | three55ml | seva: You have to set up 12223334444 as an extension in the [default] context |
| 23:36.55 | seva | ariel_: or if i try "dial 1222333444" from *CLI> prompt i get "No such extension '12223334444' in context 'default'" |
| 23:37.05 | three55ml | That's what the message is telling you |
| 23:37.14 | ariel_ | seva, your dialing rule in asterisk something like exten => _NXXNXXXXXX.,Dial(IAX2/username:password@voipjet) something like that |
| 23:37.26 | seva | yeah http://sevatech.com/tmp/asterisk/extensions.conf |
| 23:37.33 | seva | er _1NX... |
| 23:38.06 | FuRR_ | anyone know anything about "SIP Trunking" |
| 23:38.49 | FuRR_ | aparently we have a nortel succession box that supports "SIP Trunking" |
| 23:38.50 | three55ml | seva: Do you have a [general] section or did you delete it? |
| 23:39.00 | seva | don't have one |
| 23:39.06 | three55ml | Try putting one in |
| 23:39.16 | three55ml | Look at the sample extensions.conf on the wiki |
| 23:39.42 | ManxPwr | greg_work, Anyone that does that is an idiot. |
| 23:39.48 | seva | the sample in * distribution [general] has just 2 statement, static= and writeprotect= |
| 23:39.48 | ManxPwr | greg_work, Just put them on /ignore. |
| 23:40.15 | ManxPwr | seva, Even if it's empty Astrisk wants a [general] section. |
| 23:40.45 | three55ml | seva: What ManxPwr said. It's probably never even parsing your default context |
| 23:41.03 | seva | adding an empty [general] to extensions didn't help |
| 23:41.49 | three55ml | try adding "static=yes" and "writeprotect=no" |
| 23:42.18 | three55ml | Connect to the terminal through "asterisk -r" and watch for parse errors in extensions.conf when you run "reload" |
| 23:42.20 | seva | doesn't help |
| 23:42.30 | three55ml | You're reloading each time? |
| 23:42.34 | seva | there is a warning about IAX provisioning being disabled |
| 23:42.35 | seva | yeah |
| 23:42.40 | seva | i quit and restart with -c |
| 23:42.44 | seva | (-vc) |
| 23:42.59 | seva | and a warning about not fidning management conf |
| 23:43.04 | three55ml | Hmmm |
| 23:43.09 | three55ml | What's the context set to for your IAX user? |
| 23:43.25 | seva | default: http://sevatech.com/tmp/asterisk/iax.conf |
| 23:44.14 | seva | http://sevatech.com/tmp/asterisk/startup.txt |
| 23:44.43 | three55ml | Try running with -vvvgc to get a little bit higher level of verbosity |
| 23:44.48 | three55ml | See if it tells you anything else |
| 23:45.54 | *** join/#asterisk evilrabbi (~evilrabbi@ip68-229-222-32.ok.ok.cox.net) |
| 23:46.07 | seva | not much there either http://sevatech.com/tmp/asterisk/startup-verbose.txt |
| 23:46.20 | seva | hrm |
| 23:46.27 | seva | do i need asterisk.conf and extconfig.conf? |
| 23:46.46 | three55ml | I believe so |
| 23:46.57 | three55ml | At least asterisk.conf |
| 23:47.02 | three55ml | What version are you running? |
| 23:47.05 | ManxPwr | "The trouble with being poor is that it takes up all your time." <-- this is SO true. |
| 23:47.13 | seva | that looks like just to override compile time default (running 1.0.5) |
| 23:48.03 | three55ml | Why don't you just try make install-config and make sur everything is there |
| 23:48.22 | three55ml | I dont see it ever parsing your extensions.conf |
| 23:48.35 | seva | good point |
| 23:48.53 | three55ml | You could check permissions as well |
| 23:49.04 | seva | permissions are fine, as i am testing as root (on the laptop) |
| 23:49.15 | seva | what normally loads extensions.conf |
| 23:49.29 | ManxPwr | seva, this stuff is not that complicated. |
| 23:49.35 | seva | i agree |
| 23:49.41 | three55ml | Just try make install-config |
| 23:49.45 | ManxPwr | "show dialplan" will show you what Asterisk thinks the dialplan is. |
| 23:49.52 | three55ml | That should put everything you need in there, backup your hold ones if you want to keep them. |
| 23:50.33 | seva | show dialplan only has context 'parkedcalls' via res_features |
| 23:50.48 | *** part/#asterisk Grooby (~Grooby@66.160.105.186) |
| 23:50.58 | *** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com) |
| 23:50.58 | seva | three55ml: i really hate starting with giant config and then forgetting to remove stuff, i' prefer to start from very little and add as i need it |
| 23:51.06 | seva | but you are right, something important is missing ;) |
| 23:51.16 | *** join/#asterisk peted20 (~pete@24-113-67-25.wavecable.com) |
| 23:51.27 | three55ml | It's actually "make samples" if you end up doing it |
| 23:51.45 | seva | here we go |
| 23:51.52 | seva | i did a grep "extensions.conf" in modules |
| 23:51.58 | seva | and added pbx_config to load |
| 23:52.03 | seva | show dialplan looks a lot better |
| 23:52.08 | ManxPwr | seva, Post your extensions.conf to pastebin.ca |
| 23:52.22 | voxlinx | hey manx... turns out I'm not sending audio on SIP-SIP. |
| 23:52.32 | seva | ManxPwr: i think i got it ^ |
| 23:52.32 | ManxPwr | seva, /etc/asterisk/modules.conf should be LEFT THE DEFAULT unless you have problems. |
| 23:52.41 | voxlinx | ManxPwr * is reporting maximum retries exceeded, but I'm on a local lan... |
| 23:52.41 | seva | i don't want it loading all the damn modules |
| 23:53.01 | *** join/#asterisk Ayano (~erik_leee@209.143.187.254) |
| 23:53.41 | *** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net) |
| 23:53.50 | voxlinx | anyone have any ideas? |
| 23:54.08 | *** join/#asterisk dsfr (~dsfr@216.207.244.183) |
| 23:54.25 | Ayano | voxlinx: sorry just joined what was the q? |
| 23:54.42 | seva | but looks like i'll just have to load all, otherwise there are too many things i have to list manually |
| 23:54.50 | voxlinx | local lan, SIP-SIP via x-lite. * reports "Maximum retries exceeded" |
| 23:55.24 | voxlinx | google search shows that's usually a NAT issue, but since I'm on a local lan, I can't figure it out... |
| 23:55.40 | voxlinx | The real problem is lack of outbound audio. |
| 23:55.45 | Mavvie | voxlinx: use tcpdump and you'll see what is going on. |
| 23:55.53 | voxlinx | either sip-sip or sip-pstn |
| 23:57.12 | voxlinx | sniffing may show clues, but I would think that's rather low-level... |
| 23:57.47 | Mavvie | well, of course you're free to ignore suggestions from people who have been at it. |
| 23:58.18 | voxlinx | heh. |
| 23:58.26 | voxlinx | no offense intended... |
| 23:58.33 | Mavvie | here neither :-) |
| 23:58.44 | Mavvie | but it's called voice-over-*IP*. |
| 23:59.02 | voxlinx | yes, but I've had it working without problems in the past. |
| 23:59.13 | Mavvie | where the IP layer (well, icmp, udp :-) is showing you where it goes wrong. |
| 23:59.15 | voxlinx | matter of fact, this is the first sip problem I've had since playing with * |
| 23:59.28 | voxlinx | Unless it's not a problem at that layer. |
| 23:59.39 | Mavvie | yes, the user part is the most horrible and the least debuggable. |
| 23:59.45 | voxlinx | If the issue is in the application error, a sniffer won't do much good. |