| 00:03.26 | ruben23 | hi |
| 00:05.42 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 00:10.20 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 00:17.38 | leifmadsen | Yurik: nice! You should provide that as a patch! :) |
| 00:17.43 | leifmadsen | http://issues.asterisk.org |
| 00:17.49 | Yurik | leifmadsen: I just signed up :) |
| 00:18.04 | Yurik | I am not sure patch is ideal, it is a oneliner; but asterisk runs :) |
| 00:18.18 | leifmadsen | [TK]D-Fender: if you didn't notice earlier, I just made a new tag: ~asteriskversioning <-- tells you about how 1.6 release processes work. Might be useful. |
| 00:18.26 | leifmadsen | Yurik: every bit helps :) |
| 00:27.27 | Yurik | leifmadsen: https://issues.asterisk.org/view.php?id=15392 |
| 00:35.28 | *** join/#asterisk rue_mohr (n=rue@h24-207-90-17.cst.dccnet.com) |
| 00:35.48 | rue_mohr | russellb, howdy, so what do we call the official fix? |
| 00:37.35 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
| 00:37.53 | *** join/#asterisk Ryan09 (n=Ryan09@mail.intersoft.net.au) |
| 00:39.18 | Ryan09 | Hi, would any one know if there's a way to get asterisk to start the music on hold from a random place in a single mp3? |
| 00:39.19 | *** part/#asterisk ruben23 (n=RPL@124.107.3.178) |
| 00:40.13 | Ryan09 | (well, it doesn't have to be an mp3... but a random place within the file, instead of starting from the start every time) |
| 00:42.33 | rue_mohr | do your hold music from a stream |
| 00:42.39 | rue_mohr | it'll pick up whereever the stream is |
| 00:43.17 | *** join/#asterisk exothermc (n=miles@74.85.64.30) |
| 00:46.33 | *** join/#asterisk zamba (i=marius@flage.org) |
| 00:47.26 | Ryan09 | ...ok, thanks. Is there an easy way to set up the stream on the machine running asterisk, or would I have to use squeezebox or something like that? |
| 00:48.17 | *** join/#asterisk serph (n=serph@CPE001f5b006e5e-CM001371144daa.cpe.net.cable.rogers.com) |
| 00:53.48 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 00:58.11 | Docteh | Ryan09: use shoutcast or icecast |
| 00:58.14 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 00:59.51 | Ryan09 | Thanks a lot, I'll look in to it :) I guess another option would be to split the file in to several parts... |
| 01:01.40 | Docteh | how long is the file? a looping stream might be more interesting |
| 01:02.14 | *** join/#asterisk Deeewayne (n=dwayne@76.29.245.9) |
| 01:02.14 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 01:03.09 | Docteh | cute |
| 01:03.28 | Docteh | e911 story on ctv news, isn't the GPS stuff so last year? |
| 01:03.42 | *** join/#asterisk jbjapan (n=jerome@mail.biosjp.com) |
| 01:03.52 | jbjapan | Hi |
| 01:04.04 | engien | i thought that was last decade |
| 01:11.03 | *** join/#asterisk aces1up (n=Now@ip70-173-52-152.lv.lv.cox.net) |
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| 01:47.51 | *** join/#asterisk AlmightyOatmeal (n=jamie@66-190-59-171.dhcp.mdsn.wi.charter.com) |
| 01:49.08 | AlmightyOatmeal | i need a little help on an outbound dialing rule.. if i wanted to dial only a 7 digit number and have asterisk append the first 4 digits of the 10 digit number, i would do exten => _1608NXXXXXX right? |
| 01:49.40 | AlmightyOatmeal | if that is true, how would i add it to a context that already has exten => _1NXXNXXXXXX |
| 01:49.55 | AlmightyOatmeal | like have what i have as 1, 2, 3, and my new one as 4? |
| 01:50.23 | *** join/#asterisk s0lid (n=s0lid@122.53.110.183) |
| 01:50.42 | AlmightyOatmeal | that doesn't seem quite right |
| 01:51.06 | AlmightyOatmeal | how would i make an outbound rule that if i only enter 7 digits it prefixes 1608 to the front, OR i can dial the full 10 digit number? |
| 01:54.54 | carrar | no |
| 01:55.18 | AlmightyOatmeal | how would i go about that? |
| 01:55.22 | Docteh | exten => XXXXXXX,1,Dial(1608$EXTENSION) <-- something like that |
| 01:55.25 | carrar | exten => _NXXXXXX,1,Dial(SIP/1608${EXTEN}) |
| 01:55.42 | carrar | mine is correct |
| 01:55.44 | Docteh | haha i knew i had that wrong |
| 01:55.54 | carrar | we |
| 01:55.54 | AlmightyOatmeal | but what if i want to dial the full 10 digit number instead? |
| 01:55.59 | carrar | ll mine is wrong too :) |
| 01:56.55 | Docteh | AlmightyOatmeal: add a line right after it matching a 10 digit number |
| 01:56.56 | carrar | exten => _NXXXXXX,1,Dial(SIP/1608${EXTEN}@sippeer) |
| 01:57.03 | *** join/#asterisk jplank (n=GBove@cpe-075-181-097-208.carolina.res.rr.com) |
| 01:57.37 | AlmightyOatmeal | exten => _1NXXNXXXXXX, n, dial(SIP/${EXTEN}@sip.broadvoice.com,30) |
| 01:57.37 | AlmightyOatmeal | exten => _NXXXXXX, n, dial(SIP/1608${EXTEN}@sip.broadvoice.com,30) |
| 01:57.53 | AlmightyOatmeal | or instead of 'n' i should do 1 then 2? |
| 01:58.09 | carrar | exten => _NXXXXXX,1,Dial(SIP/1608${EXTEN}@sip.broadvoice.com,30) |
| 01:58.12 | Docteh | well for 10 digit dialing you dont dial the 1 btw |
| 01:58.34 | carrar | if 1 is not 1, it is a 'n' |
| 01:58.46 | carrar | (not 2 or 3 etc...) |
| 01:58.53 | AlmightyOatmeal | ah |
| 01:58.58 | carrar | 1 |
| 01:58.58 | carrar | n |
| 01:58.59 | carrar | n |
| 01:59.05 | AlmightyOatmeal | yeah |
| 01:59.06 | carrar | the actual letter 'n' |
| 01:59.22 | AlmightyOatmeal | i keep getting extension not found when i dial a 7 digit number :( |
| 01:59.37 | carrar | exten => _NXXXXXX,1,Dial(SIP/1608${EXTEN}@sip.broadvoice.com,30) |
| 01:59.50 | jplank | if I'm installing HPEC and have a HW echo can module on my 2400, do I need to remove my module first? |
| 02:00.17 | AlmightyOatmeal | carrar: i've done: exten => _1NXXNXXXXXX, 2, dial(SIP/${EXTEN}@sip.broadvoice.com,30) |
| 02:00.20 | AlmightyOatmeal | exten => _NXXXXXX, 3, dial(SIP/1608${EXTEN}@sip.broadvoice.com,30) |
| 02:00.25 | AlmightyOatmeal | 1 is setting my callerid |
| 02:00.44 | carrar | change 2 & 3 to the letter n |
| 02:00.53 | carrar | oh |
| 02:00.55 | carrar | no |
| 02:00.59 | AlmightyOatmeal | ? |
| 02:01.01 | carrar | you have different extension |
| 02:01.04 | carrar | they all should be 1 |
| 02:01.21 | carrar | binpaste me the whole section |
| 02:01.23 | carrar | ~bp |
| 02:01.23 | infobot | from memory, bp is pb backwards |
| 02:01.28 | carrar | ~pb |
| 02:01.28 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 02:02.00 | AlmightyOatmeal | http://pastebin.ca/1473616 |
| 02:02.16 | carrar | wrong |
| 02:02.17 | AlmightyOatmeal | oops, i had the first n as 2, and the second as 3 |
| 02:02.17 | carrar | hang on |
| 02:03.02 | *** join/#asterisk xa0z (n=interex@75-129-229-76.dhcp.mtvr.il.charter.com) |
| 02:03.53 | xa0z | Is it possible to run Asterisk on the same box as a firewall, such as pfSense or m0n0wall? |
| 02:04.12 | AlmightyOatmeal | xa0z: anything is possible |
| 02:05.03 | xa0z | Hmm, is it likely to work properly? |
| 02:05.09 | *** join/#asterisk salzh (n=Administ@202.106.126.231) |
| 02:05.41 | AlmightyOatmeal | xa0z: depends how you configure it but i'm sure it would, i have my freebsd asterisk box behind pfsense |
| 02:05.52 | carrar | http://pastebin.ca/1473617 |
| 02:06.25 | xa0z | AlmightyOatmeal, no I don't want it behind anything... I want them on the same box. |
| 02:07.17 | AlmightyOatmeal | carrar: i get service unavailable when i dial a 7 digit number |
| 02:07.31 | carrar | did you change 'THISEXTENSION' |
| 02:07.37 | AlmightyOatmeal | lol |
| 02:07.47 | AlmightyOatmeal | i should stop drinking captain morgan :P |
| 02:07.49 | jplank | xa0z: depends on what you want to use your box for |
| 02:07.55 | jplank | as a firewall or a pbx? |
| 02:08.08 | rob0 | Captain Morgan should stop eating AlmightyOatmeal. |
| 02:08.08 | carrar | err 'THISEXTENSION' should be 'THIS CONTEXT' |
| 02:08.13 | carrar | sorry |
| 02:08.27 | carrar | http://pastebin.ca/1473618 |
| 02:08.33 | carrar | Whatever the context is this is ibn |
| 02:08.34 | carrar | in |
| 02:08.51 | *** join/#asterisk chendy (n=chatzill@58.251.102.197) |
| 02:08.54 | carrar | Whats the context this is? |
| 02:09.02 | AlmightyOatmeal | aah yes, that worked |
| 02:09.08 | AlmightyOatmeal | bows to carrar |
| 02:09.09 | AlmightyOatmeal | :) |
| 02:09.23 | carrar | it's messy but you can improve it |
| 02:09.31 | carrar | you NEVER need to use priorities numbers |
| 02:09.33 | carrar | cept for 1 |
| 02:09.38 | AlmightyOatmeal | that will give me something to start with, many thanks |
| 02:09.40 | carrar | use 'tags' |
| 02:09.57 | carrar | AVAILCHAN can be a lot of things |
| 02:09.59 | carrar | like BUSY |
| 02:10.01 | carrar | etc.. |
| 02:10.04 | carrar | you can key on that |
| 02:10.10 | AlmightyOatmeal | thanks :) |
| 02:10.24 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 02:10.40 | AlmightyOatmeal | if you're ever in south central wisconsin ill have my rents buy you a beer ;) |
| 02:10.50 | xa0z | So, the likely hood of running asterisk or trixbox on the same system as pfSense or the like, is not good? |
| 02:11.08 | jplank | trixbox |
| 02:11.10 | AlmightyOatmeal | xa0z: pfsense is not built for running ports but its entire plausible though i wouldn't |
| 02:11.12 | jplank | probably not |
| 02:11.19 | carrar | show application dial |
| 02:11.24 | carrar | <PROTECTED> |
| 02:11.24 | carrar | <PROTECTED> |
| 02:11.25 | carrar | <PROTECTED> |
| 02:11.37 | jplank | xa0z: the problem is, why would your run anything on your phonesystem? |
| 02:11.49 | jplank | its a phonesystem, not a swiss army knife |
| 02:12.07 | carrar | woops |
| 02:12.19 | jplank | you could build a firewall out of a $20 PC, or even a free one a friend is throwing away, why waste CPU cycles? |
| 02:12.21 | xa0z | The problem is, I already have a system with 4 NICs in it running pfsense, I'd like not to have another system to use my 2 phones with CallCentric. |
| 02:12.31 | jplank | oh |
| 02:12.41 | jplank | in that case, I guess it would work |
| 02:13.03 | jplank | but if that asterisk is going to be "behind" the firewall, that might prove to be annoying |
| 02:13.45 | xa0z | Right now I have my old DTA310 connected to CallCentric with my regular phone but would rather be using sip phones for more options. |
| 02:13.59 | carrar | AlmightyOatmeal, change AVAIL to DIALSTATUS |
| 02:14.21 | jplank | if we are talking two phones at home, it would be fine |
| 02:14.49 | xa0z | Yea, with 2 channels I'd rather have 2 phones ;) |
| 02:14.59 | jplank | I just don't know how fun to setup it will be when pfsense and * is on the same box, but then again, I can't say I ever setup pfsense, so who am I to talk |
| 02:15.11 | xa0z | :/ |
| 02:15.44 | carrar | AlmightyOatmeal, http://pastebin.com/m658773fe |
| 02:15.51 | jplank | how would * send traffic through pfsense, out one interface and back into another? |
| 02:15.56 | jplank | sounds kind of annoying |
| 02:16.00 | jplank | but I guess that would work |
| 02:17.03 | jplank | is pfsense going to be *'s gateway? |
| 02:18.16 | jplank | but then again, I don't know how pfsense works, I guess if it works like IP Tables, it will be transparent to * and its all a moot point |
| 02:19.15 | carrar | prob need to capitalize it too |
| 02:22.54 | carrar | I love it when I post the incorrect dialplans |
| 02:37.07 | s14ck | lol |
| 02:37.09 | s14ck | http://xkcd.com/537/ |
| 02:40.34 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 02:47.35 | *** join/#asterisk metfan2007 (n=metfan@189.146.148.103) |
| 03:20.54 | mmlj4 | pfsense is a high-falutin' edge-device distro |
| 03:21.07 | *** part/#asterisk xa0z (n=interex@75-129-229-76.dhcp.mtvr.il.charter.com) |
| 03:21.26 | mmlj4 | if you've used one (smoothwall, IPcop, etc.) you've used them all |
| 03:21.33 | jplank | oh |
| 03:21.48 | jplank | well then, I wouldn't want to run * on a smoothie |
| 03:22.34 | jplank | riddle me this, why does digium's site say tech support is 7am-8pm CST, but they are actually are open 24/5 |
| 03:22.52 | engien | so they can play halo at night |
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| 04:28.31 | mmattice | what's the "best" digium PRI card for low port count? (likely only 1 needed) |
| 04:29.28 | mmattice | also, is starpy the only twisted based AGI interface left? |
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| 05:07.32 | AlmightyOatmeal | i wish my sip provider subscribed to cname databases... |
| 05:07.41 | AlmightyOatmeal | i wish i could add myself for cheap to a cname database |
| 05:07.42 | AlmightyOatmeal | :( |
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| 05:41.50 | JT | can anyone refresh me as to what each bit means in relation to a pri?: Status: Provisioned, In Alarm, Down, Active |
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| 05:54.04 | AlmightyOatmeal | boy why doesn't anyone want to buy a cisco 7911G sip phone? |
| 05:54.10 | AlmightyOatmeal | kicks ebay |
| 05:54.52 | JT | $5? |
| 05:56.41 | AlmightyOatmeal | ... |
| 05:58.41 | AlmightyOatmeal | i just dont get it, i have the sip firmware and configs and started the bidding at $0.99 and i have no bids and no watchers and like a day and a half to go |
| 06:02.28 | JT | it's just an old phone |
| 06:03.12 | AlmightyOatmeal | and annoying to configure |
| 06:03.13 | AlmightyOatmeal | but it works |
| 06:03.20 | MT`AwAy | AlmightyOatmeal> accepting paypal and delivering worldwide ? |
| 06:03.23 | AlmightyOatmeal | and i'm surprised it hasn't found its way into sombody's heart |
| 06:03.29 | AlmightyOatmeal | MT`AwAy: both are affirmative |
| 06:03.35 | MT`AwAy | url? |
| 06:03.38 | MT`AwAy | :D |
| 06:04.48 | AlmightyOatmeal | item 180372705923 |
| 06:05.14 | AlmightyOatmeal | i'm using my gf's ebay account because paypal has some stupid new policy on sellers with less than 100 feedback |
| 06:08.41 | hardwire | sigh |
| 06:10.11 | *** join/#asterisk juny4ng (n=junyangn@119.234.22.123) |
| 06:10.16 | MT`AwAy | haha |
| 06:10.20 | MT`AwAy | 46 usd for delivery |
| 06:10.37 | AlmightyOatmeal | really? |
| 06:10.48 | AlmightyOatmeal | where are you, a tiny islant in the middle of the ocean? |
| 06:10.54 | MT`AwAy | somewhat |
| 06:10.58 | MT`AwAy | an island called Japan |
| 06:11.03 | AlmightyOatmeal | oh heh |
| 06:11.05 | AlmightyOatmeal | strange |
| 06:11.54 | MT`AwAy | well, deliveries are expensive to Japan |
| 06:11.59 | MT`AwAy | from japan it's not |
| 06:12.04 | MT`AwAy | things are weird~ |
| 06:12.26 | AlmightyOatmeal | cheapest i could do is riority MailŽ International Small Flat-Rate Box if it will fit in that, thats like $13 USD |
| 06:13.07 | AlmightyOatmeal | otherwise its first class mail international at like $25 like you said |
| 06:13.23 | MT`AwAy | well anyway don't worry, most likely someone will buy it within the last 5 minutes or so |
| 06:13.34 | AlmightyOatmeal | pouts |
| 06:14.04 | JT | what's the policy for sellers with less than 100 feedback? |
| 06:14.24 | AlmightyOatmeal | JT: paypal holds the funds for 21 days, even if said funds are needed to ship said item |
| 06:14.42 | AlmightyOatmeal | i have new accounts because my old ebay/paypal accounts were hijacked |
| 06:16.53 | JT | how did they get hijacked? |
| 06:17.04 | AlmightyOatmeal | i dont know |
| 06:18.44 | MT`AwAy | argh |
| 06:18.54 | MT`AwAy | I'm going to need a new paypal account too |
| 06:19.07 | MT`AwAy | my current paypal account can't have its country changed |
| 06:19.20 | AlmightyOatmeal | hmm |
| 06:19.26 | AlmightyOatmeal | why would you do that? |
| 06:19.36 | MT`AwAy | I moved to Japan 2 weeks ago |
| 06:19.45 | MT`AwAy | and do not plan on returning |
| 06:20.18 | JT | must be poor email security or a week password to get your account hijacked |
| 06:20.25 | JT | s/week/weak/ |
| 06:20.39 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 06:20.50 | AlmightyOatmeal | JT: neither were the case afaik, my passwords are 12 chars long and a bountiful mix |
| 06:20.55 | MT`AwAy | or fall victim of phishing |
| 06:21.00 | MT`AwAy | or got a virus/backdoor |
| 06:21.08 | JT | well i can't imagine it happened by magic |
| 06:21.10 | MT`AwAy | or logged in from a public place or someone else's computer who had a virus |
| 06:21.18 | AlmightyOatmeal | potentially a virus thanks to my gf or someone sniffing the wifi at the local college |
| 06:21.21 | AlmightyOatmeal | shrugs |
| 06:21.27 | MT`AwAy | ah |
| 06:21.28 | AlmightyOatmeal | in either case, there was severe suckage involved |
| 06:21.30 | MT`AwAy | college |
| 06:21.33 | MT`AwAy | it explain things |
| 06:21.34 | JT | signon is over ssl |
| 06:21.35 | rue_mohr | all my attacks have been attemts to log in as extenstions from 1000 to 9999 with passwords that match the extentions |
| 06:21.41 | JT | unless they can crack ssl |
| 06:22.02 | JT | packet sniffing won't do much |
| 06:22.22 | MT`AwAy | says: gf's fault |
| 06:22.43 | AlmightyOatmeal | she opened up IE tonight and i found a gazillion more spyware loaded heh |
| 06:22.48 | AlmightyOatmeal | i can't keep up with her :P |
| 06:22.56 | rue_mohr | you shoudl delete ie from the system |
| 06:23.07 | rue_mohr | rename is to iexplode.exe |
| 06:23.09 | AlmightyOatmeal | she uses firefox, but was using IE to west a blog post |
| 06:23.12 | AlmightyOatmeal | heh |
| 06:23.23 | AlmightyOatmeal | s/west/test |
| 06:23.28 | rue_mohr | they have found ways occasionsally to get firefox to start ie |
| 06:23.30 | Docteh | haha i deleted the shortcuts to ie so my friend now starts msn just to get ie |
| 06:23.45 | Docteh | msn messenger i mean |
| 06:24.05 | rue_mohr | hah, I just noticed the channel, you are SO offtopic |
| 06:24.15 | MT`AwAy | somewhat |
| 06:24.29 | AlmightyOatmeal | rue_mohr: if you followed the chain of events you'd know we are right on topic |
| 06:24.36 | MT`AwAy | ip phone > ebay sale > account stolen > account security > browser security |
| 06:24.41 | rue_mohr | that would take work and commitment |
| 06:24.52 | AlmightyOatmeal | or just read what MT`AwAy said |
| 06:25.07 | rue_mohr | I think thats work? |
| 06:25.17 | rue_mohr | I'm busy coding... |
| 06:25.41 | MT`AwAy | I'm not busy because I'm at work (what kind of logic is this?) |
| 06:25.49 | AlmightyOatmeal | MT`AwAy: same here |
| 06:26.01 | AlmightyOatmeal | i have a summer job as a hotel night auditor.. i'm bored as hell :P |
| 06:26.12 | rue_mohr | go audit someone |
| 06:26.26 | rue_mohr | random audits can be healthy |
| 06:26.30 | AlmightyOatmeal | :P |
| 06:26.48 | MT`AwAy | sounds like a fun job |
| 06:26.49 | rue_mohr | "NOBODY EVER expects the apnish inquisition!" |
| 06:26.51 | AlmightyOatmeal | for $7/hr i will stick to the bare minimum thanks |
| 06:27.01 | rue_mohr | apnish/spanish/ |
| 06:27.10 | MT`AwAy | I'm project manager |
| 06:27.18 | MT`AwAy | for people coding PHP |
| 06:27.26 | rue_mohr | you project managment? |
| 06:27.33 | AlmightyOatmeal | i'm grammar police |
| 06:27.37 | AlmightyOatmeal | and you're under arrest |
| 06:28.07 | rue_mohr | "I declare my room a seperate country in which the rules of speeling are diffrent" |
| 06:28.24 | AlmightyOatmeal | ha |
| 06:29.00 | MT`AwAy | rue_mohr> basically I determine in advance what is going to get wrong or difficult in project and how to overcome that thanks to my past 8 years as a php dev |
| 06:29.22 | rue_mohr | good good |
| 06:29.34 | rue_mohr | what part of asterisk are you doing php for? |
| 06:30.04 | rue_mohr | I did a speed dial editor for the system dial list |
| 06:30.17 | rue_mohr | I need to do another one for the call display rewriter |
| 06:30.39 | MT`AwAy | got a system using AGI that will select best route for outgoing calls based on cost, line quality and amount of calls currently routed |
| 06:30.46 | rue_mohr | so as more company cell phones are added we can put them in the list, our provider always says the same thing for incomming cell calls |
| 06:31.44 | rue_mohr | everyone enjoyed using my php page to edit their speed dials :) |
| 06:31.46 | MT`AwAy | when one of our subscribers calls externally, a route is calculated by an external php system (and the max number of seconds for the call based on the customer's balance is computed), and once the call is completed, the total cost is computed and deducted |
| 06:32.14 | *** join/#asterisk xrmx__ (n=rm@host176-254-dynamic.14-87-r.retail.telecomitalia.it) |
| 06:32.22 | rue_mohr | your employees must be really scattered |
| 06:32.46 | MT`AwAy | not using asterisk only internally |
| 06:32.51 | MT`AwAy | we also sell voip telephonly |
| 06:32.55 | MT`AwAy | telephony* |
| 06:33.05 | rue_mohr | the cat says its WAY past my bedtime and if I dont make with getting to bed he's gonna sleep on me instead |
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| 06:33.23 | Docteh | dundundunnn |
| 06:33.25 | rue_mohr | MT`AwAy, what area of the world? |
| 06:33.41 | rue_mohr | west canada perhapps? |
| 06:33.51 | MT`AwAy | rue_mohr> mainly europe, but also japan |
| 06:34.09 | rue_mohr | bah, absolutly no use to me |
| 06:34.11 | rue_mohr | gnight |
| 06:34.21 | MT`AwAy | :D |
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| 07:14.14 | McL0VIN | howdy folks |
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| 07:16.58 | cousin_luigi | hello |
| 07:17.05 | cousin_luigi | is there any IP phone or ATA with an OSS firmware? |
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| 07:24.17 | aces1up | where can i get the latest zapata drivers? |
| 07:26.36 | kaldemar | zaptel was renamed to dahdi, so dahdi is the latest. http://downloads.asterisk.org/pub/telephony/ |
| 07:29.00 | McL0VIN | i have a idea and i am not sure if it is possible or no. can i assign 'dialplan reload' to a keyboard key e.g the '*' in the keypad? |
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| 07:34.39 | kaldemar | McL0VIN: there will be an alias function in 1.6.2 |
| 07:34.55 | kaldemar | http://svn.digium.com/svn/asterisk/branches/1.6.2/configs/cli_aliases.conf.sample |
| 07:35.19 | McL0VIN | kaldemar: nice! |
| 07:37.40 | aces1up | kaldmar i am running centos and i installed my kernel sources via yum install kernel-devel but when doing a make all on the dahdi it says cannot find kernel sources are there any other packages i should install? |
| 07:39.03 | kaldemar | you're probably missing kernel headers |
| 07:40.15 | kaldemar | kernel-devel should include those, though. do you have a symlink to the source dir? |
| 07:41.06 | kaldemar | meaning a linux-2.6 named symlink that points to linux-headers-2.6.xx.... or something like that |
| 07:41.27 | liri | what should I be looking for if extension A and B are on a call and extension C is calling one of them but doesn't hear the busy sound it gets back a normal ringing ringback sound. |
| 07:41.47 | aces1up | kaldemar i dunno how can i check ? |
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| 07:42.39 | kaldemar | aces1up: see what is in /usr/src/ |
| 07:43.57 | aces1up | i see the source in /usr/src/kernels |
| 07:45.03 | kaldemar | i don't use centos myself so you better ask someone who does. |
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| 07:57.08 | McL0VIN | \amethyst: are you coming to * |
| 08:00.54 | liri | any ideas about asterisk's function for call waiting/busy? |
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| 08:19.02 | mort_gib | liri: Concept is different from analogue lines, on the phone (depending on brand/model) the recipient will hear a slight beep when the second call comes in |
| 08:19.48 | mort_gib | liri: I favour playing "The callee is in another call, please hold the line or press one for VM" |
| 08:20.10 | mort_gib | liri: Or that's what my clients prefer |
| 08:27.47 | liri | mort_gib: bare with me on this... |
| 08:28.59 | liri | mort_gib: I'm not talking about the called extension, I'm talking about the caller - instead of hearing a normal ringback sound (for an extension which is not on a call) I want it to hear the call waiting sound. |
| 08:29.52 | liri | mort_gib: let's stick to what you said - how do I configure it so the caller hears that message "the calle is in another call"... ? |
| 08:30.17 | mort_gib | SIPPEER |
| 08:30.31 | mort_gib | if you are using sip 'phons |
| 08:31.22 | liri | mort_gib: what about it? |
| 08:33.46 | mort_gib | liri: Private |
| 08:34.53 | liri | k |
| 08:36.13 | cousin_luigi | any idea about a completely OSS phone? |
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| 08:47.29 | xrmx__ | cousin_luigi, openmoko? gigaset gives you the source and toolchain for cordless ip phone, snom gives that too |
| 08:48.12 | cousin_luigi | xrmx__: thanks, I'll look into those |
| 08:49.45 | jkroon | Does anybody here have any experience with the Diva PRI cards? |
| 08:49.59 | jkroon | specifically their Dialogic Corporation PRI |
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| 09:03.35 | eliteSchaf | has anyone got a debian init-script for asterisk? |
| 09:03.55 | mort_gib | make config |
| 09:05.15 | eliteSchaf | and the script will be located under contrib/scripts? |
| 09:05.35 | eliteSchaf | mort_gib: oh thx :) |
| 09:05.49 | mort_gib | np |
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| 09:21.09 | tzafrir_laptop | electus, or the debian package itself... |
| 09:26.01 | zeeesh | i hv installed speex-1.2rc1.tar.gz, when compiling asterisk, but when trying to translating ulaw to speex a single peer to peer call, then getting error "chan_sip.c:4339 sip_call: No audio format found to offer. Cancelling call to 322XXXXXX .. -- Couldn't call 322XXXXXX"? |
| 09:26.58 | zeeesh | ulaw to alaw or gsm translating working but when trying with speex ,,, getting error ? |
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| 09:51.12 | czindy | Hello, Good morning! |
| 09:52.28 | czindy | I have a problem eith dahdi2.2.0 and wanpipe3.4.1 install. Here are the details: http://pastebin.com/d3b2d089a please help to resolve this. It is a copmile problem. Thank you. |
| 09:56.24 | zeeesh | how to uninstall dahdi-linux-complete? |
| 09:57.28 | *** join/#asterisk Chris-NB (n=chris@85-126-61-10.work.xdsl-line.inode.at) |
| 09:57.29 | Chris-NB | hi |
| 09:57.44 | Chris-NB | is it possible to play a busy during dialing to a sip phone? |
| 09:58.01 | Chris-NB | i know, stupid question, but I need this for a customer |
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| 09:59.19 | tzafrir_laptop | Playtones(busy) ? |
| 10:02.26 | liri | what about playing a call-waiting sound if the callee is on a call with another extension? |
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| 10:07.00 | Chris-NB | tzafrir_laptop, I've seen this and tried it: http://www.voip-info.org/wiki/view/Asterisk+cmd+Playtones but it's not working (1.4.20) |
| 10:08.03 | Chris-NB | liri, If a phone is busy and 2nd comes in, I call the phone for 1 sec (to have an unanswered call on the phone) and then play a busy. But that sounds like the call was rejected and not busy. |
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| 10:42.56 | McL0VIN | why when i use zoiper i am not able to listen to moh or main-menu sounds? |
| 10:43.05 | McL0VIN | works just fine with x-lite |
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| 11:05.02 | jkroon | [Jun 25 19:00:01] WARNING[15809]: chan_capi.c:5933 cc_init_capi: CAPI not installed, chan_capi disabled! <-- last line of output from asterisk failing to start up. I'm not sure whether it's chan_capi causing the problem or not - any suggestions? |
| 11:06.04 | jkroon | ok, has to be chan_capi, adding noload => chan_capi.so allows asterisk to start up. |
| 11:07.08 | kaldemar | the warning says that you don't have CAPI installed, and therefore chan_capi cannot be loaded. quite obvious. |
| 11:10.01 | *** join/#asterisk iksik (i=xk@livedata.pl) |
| 11:11.31 | jkroon | kaldemar, ls -l /dev/capi20 says different :) |
| 11:16.55 | jkroon | open("/dev/capi20", O_RDWR) = 12 |
| 11:16.56 | jkroon | ioctl(12, CAPI_INSTALLED, 0) = -1 ENXIO (No such device or address) |
| 11:20.29 | *** part/#asterisk AdvoWork (n=AdvoWork@unaffiliated/advowork) |
| 11:28.57 | kaldemar | does ldd /usr/lib/asterisk/modules/chan_capi.so show anything interesting? |
| 11:31.58 | jkroon | kaldemar, no, the interesting part is that ioctl. |
| 11:34.55 | jkroon | i'm guessing it's something to do with firmware not loading properly or something equally silly. |
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| 11:47.02 | WeazelON | hi guys, i have an issue with the asterisk attended transfer in Asterisk 1.2.17 via Gentoo 2.6.19-gentoo-r5 ---- If one wants to speak first to the object of the transfer, then stay until they answer, otherwise hang up > > and the transfer is completed. Problem is - that if they hang up before the object of the transfer answers, the call drops, the only thing i can see in the logs is something to do with "Unable to play beep" |
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| 12:06.33 | mahiti-irc | hi |
| 12:06.57 | mahiti-irc | anyone knows how do i change the default sip port in AsteriskNOW ? |
| 12:08.54 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 12:09.05 | kaldemar | someone in #asterisknow might |
| 12:11.02 | mahiti-irc | kaldemar, noone does a reply there ! |
| 12:12.09 | kaldemar | give them time. most people here don't use GUI's at all. |
| 12:18.49 | mahiti-irc | oh |
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| 12:22.44 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 12:44.11 | beek | Is it possible to use the licenses for Allison-8kHz and two ports that I purchased direct from Cepstral last year with res_cepstral? |
| 12:44.45 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:44.51 | beek | morning jaytee |
| 12:44.56 | jaytee | morning beek |
| 12:45.17 | beek | You're using Ceptral aren't you? |
| 12:45.20 | beek | Cepstral |
| 12:45.26 | jaytee | it's already 77F here and supposed to get up in the 90's again, ugh |
| 12:45.48 | jaytee | beek, yeah but just for converting text files in Spanish to spanish prompts |
| 12:46.40 | *** join/#asterisk dpreacher (n=dpreache@117.195.35.72) |
| 12:46.44 | beek | I have licenses for two-ports and Allison 8Khz that I got from Cepstral and want to use them with res_cepstral but of course the register program won't deal with the keys. I need to track down how to migrate the licenses. |
| 12:46.47 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 12:46.51 | liri | wb mort_gib |
| 12:47.03 | mort_gib | Hi |
| 12:47.12 | liri | I've debugged the problem to figure out the problem |
| 12:47.23 | liri | the gateway is sending 183 with sdp and then 180 with sdp right after it |
| 12:47.34 | Aiatek | good morning everybody |
| 12:47.42 | beek | morning |
| 12:48.20 | mort_gib | liri: Gateway?? :-) What gateway?? |
| 12:49.04 | dpreacher | hello, is there some way asterisk notifies if a iax2 reload has been successful? i am writing a script to check on that, but as of now the only way i find it possible is to check if iax2 show peers show the trunk as OK. is there a better way. a status variable of sorts? |
| 12:49.28 | *** join/#asterisk deadpigeon (n=deadpige@office.xpressamerica.net) |
| 12:54.36 | jaytee | dpreacher, chanisavail()? |
| 12:55.04 | dpreacher | i see. let me look it up briefly |
| 12:55.10 | dpreacher | thanks meanwhile jaytee |
| 12:55.23 | jaytee | it's an app, not a variable |
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| 12:58.00 | dpreacher | ohk.. |
| 12:59.53 | czindy | Hello, If I connect two asterisk server with iax2 channel, is it possible to compress the sounds (traffic) between them? |
| 13:01.39 | leifmadsen | czindy: that is the job of the codec, regardless of the signalling used |
| 13:01.51 | leifmadsen | use a compressed codec |
| 13:02.30 | Aiatek | you can save bandwith with an iax trunk not compress the sound |
| 13:03.12 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 13:04.12 | leifmadsen | Aiatek: well, ya, but that is only effective when you have many simultaneous calls -- you're just saving the data bit rate on the signalling portion of IAX2. Most of the DBR is going to be with the codec (audio) stream |
| 13:05.02 | Aiatek | yes |
| 13:05.22 | leifmadsen | so it's two fold :) |
| 13:05.30 | leifmadsen | czindy: see what Aiatek and I said |
| 13:05.55 | leifmadsen | this blood computer keeps sucking me back in! I'm going to the gym... NOW. |
| 13:06.13 | czindy | Yes, thank you. |
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| 13:12.24 | Katty | distributes hugs and breakfast burritos |
| 13:12.56 | Aiatek | burritos y enchiladas |
| 13:13.04 | Pan3D | :p |
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| 13:16.34 | Katty | enchiladas are not for breakfast. |
| 13:19.41 | Katty | dear windows. wheni tell you Restart Later, that means I will restart later. you do not need to continiously nag me about it every 15 minutes. kthx, Katty. |
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| 13:27.26 | mort_gib | Katty: What do you use Windows for?? |
| 13:31.44 | dpreacher | for updating it...? |
| 13:33.12 | mort_gib | I started running secunia on a few Windows PC's |
| 13:34.06 | mort_gib | So if the normal windows update takes time, then try maintaining some 160+ boxes with ALL updates |
| 13:34.09 | mort_gib | %£!"$^&*)"$£"_ |
| 13:34.51 | Katty | i don't feel like debating what you think about windows, mort_gib |
| 13:35.36 | mort_gib | Just surprised you are using it... |
| 13:36.01 | Katty | well you got your surprise in for the day. isn't that exciting. |
| 13:36.26 | *** join/#asterisk pawpro (n=Miranda@213.166.12.34) |
| 13:36.42 | mort_gib | Yeah, my knees are a' shakin' :-) |
| 13:36.48 | Katty | super duper |
| 13:44.53 | leifmadsen | mort_gib: I use Windows, Linux, and OSX |
| 13:45.09 | leifmadsen | Windows for Quickbooks, OSX for fun stuff, and Linux for every day use |
| 13:45.18 | leifmadsen | every OS has its uses |
| 13:45.25 | leifmadsen | (except for maybe FreeBSD) |
| 13:46.03 | mort_gib | leifmadsen: I use OpenBSD fro Firewalls, Linux for laptop/desktop Debian for Asterisk/Squid etc etc and I support all the rest of the OS's |
| 13:46.22 | mort_gib | I never got around to try FreeBSD, it's on my list though |
| 13:46.38 | mort_gib | They ALL have their use, true :-) |
| 13:50.17 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 13:54.31 | Katty | feels grumpy |
| 13:54.47 | dpreacher | jaytee i read about using the command online but on my asterisknow 1.5.0 i am not able to run chanisavail. i tried running it as asterisk -rx chanisavail to get some message on usage but even that did not come...saying no such command. where should I be running chanisavail from? |
| 13:56.01 | Katty | jaytee is busy feeling the elephants right now. |
| 13:56.09 | Katty | ... |
| 13:56.10 | Katty | feeding. |
| 13:57.24 | *** join/#asterisk Yurik (n=yrashk@S01060023dfdf07dd.vc.shawcable.net) |
| 13:58.29 | dpreacher | so do you have an answer for my question Katty? |
| 13:58.46 | Katty | nope |
| 13:58.52 | dpreacher | anyone else |
| 13:59.16 | Katty | you might try waiting awhile and asking later |
| 13:59.20 | Katty | there's not many people around yet |
| 14:00.12 | [TK]D-Fender | dpreacher: chanisavail is a DIALPLAN application, not a CLI COMMAND |
| 14:01.00 | dpreacher | so where do I run this command? like asterisk -rx "iax2 chanisavail(channelname)" ? |
| 14:02.02 | Katty | [TK]D-Fender: i think that first bit of your comment didn't sink in properly. |
| 14:02.30 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 14:02.53 | [TK]D-Fender | dpreacher: DIALPLAN. this is stuff you run in EXTENSIONS.CONF and if for processing while in a call <- |
| 14:03.08 | [TK]D-Fender | drmessano: This has nothing to do with * CLI. |
| 14:03.16 | [TK]D-Fender | dpreacher: This has nothing to do with * CLI. |
| 14:03.18 | [TK]D-Fender | gah |
| 14:03.22 | dpreacher | :) |
| 14:03.53 | dpreacher | fender i'll look up more on this...but thanks for clarifying...my intention was to get the status out to a shell script that i was writing |
| 14:04.15 | [TK]D-Fender | dpreacher: When would this script be run? |
| 14:04.39 | Katty | woah, did i just see a [TK]D-Fender grammatical error? |
| 14:04.52 | dpreacher | reloads...i just need to check if reloads really got the lines back online...trying to format code for nagios spec |
| 14:05.03 | [TK]D-Fender | Katty: What you is talking about!??! |
| 14:05.10 | dpreacher | just keeping a tab on successful and unsuccessful reloads |
| 14:05.19 | mmattice | which digium digital trunk cards do hardware offload? |
| 14:05.22 | jaytee | [TK]D-Fender, he's running AsteriskNOW so he won't be able to figure it out |
| 14:05.39 | [TK]D-Fender | dpreacher: that is not a valid way to verify that |
| 14:05.39 | Katty | hi jaytee |
| 14:05.56 | dpreacher | what else do you suggest fender? |
| 14:06.07 | mahiti-irc | kaldemar, still no replies dude |
| 14:06.43 | jaytee | mornin Katty *hugs* |
| 14:06.45 | [TK]D-Fender | have another system place a call to it. See if you can connect to the running * Process at all. |
| 14:06.58 | Katty | hugs on jaytee |
| 14:07.45 | dpreacher | you really are helping me or having a good time [TK]D-Fender ? |
| 14:07.48 | dpreacher | :( |
| 14:08.08 | [TK]D-Fender | dpreacher: 2 valid suggestions. |
| 14:08.14 | dpreacher | tell me |
| 14:09.01 | [TK]D-Fender | dpreacher: Trying to. I question jsut how much is sinking in. |
| 14:09.06 | dpreacher | asterisk has no bloody way to tell a reload has taken place, no exit code, nothing? |
| 14:09.38 | [TK]D-Fender | dpreacher: Absolutely none |
| 14:09.41 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 14:09.48 | Katty | dpreacher: here's an idea |
| 14:09.54 | dpreacher | tell me |
| 14:09.56 | Katty | dpreacher: why don't you write one, and contribute it to the project. |
| 14:10.05 | kaldemar | mahiti-irc: yes, gui people may not be too active. in plain asterisk, you modify parameter port, bindport or bindaddr in sip.conf depending on the version. |
| 14:10.17 | [TK]D-Fender | dpreacher: "reload" as far as CLI is concerned is pretty transparent |
| 14:10.44 | kaldemar | mahiti-irc: the gui may or may not override the change or cause some other screwup if done by hand. |
| 14:10.47 | [TK]D-Fender | kaldemar: Always been bindport & bindaddr since 1.0 AFAIK |
| 14:11.16 | tzafrir_laptop | dpreacher, I think that there are some pretty bloody ways. I figure you can e.g. monitor the logs for the right message |
| 14:11.18 | jaytee | and "reload" using AsteriskNOW is as opaque as wood |
| 14:11.30 | dpreacher | fender thats true that we can observe reloads happening with a asterisk -rvvv open...but to tell it to script "look lines are scrolling...say its done" is something else |
| 14:11.42 | tzafrir_laptop | so with enough sweat, blood and tears, it's doable |
| 14:11.47 | mahiti-irc | kaldemar, if not by hand then how do i change the port? |
| 14:11.51 | kaldemar | [TK]D-Fender: no, it was "port" in 1.0. |
| 14:12.05 | Katty | oh boy. |
| 14:12.14 | jaytee | oh girl. |
| 14:12.17 | dpreacher | i was goin thru logs tzafrir. but only log with a text called reload [CONFIGRELOAD] had some irrelevant data |
| 14:12.26 | *** join/#asterisk oej (n=olle@240.Red-83-33-40.dynamicIP.rima-tde.net) |
| 14:12.43 | *** join/#asterisk aiksa[LV] (n=aiksa[LV@mx.fiveplus.lv) |
| 14:12.56 | Katty | jaytee: i feel bitchy today |
| 14:13.02 | dpreacher | if someone could point out the write log or write text in log to grep for and process that'd help too |
| 14:13.03 | kaldemar | [TK]D-Fender: you made me check a bunch of tags, you. |
| 14:13.10 | jaytee | Katty, I feel bastardly today |
| 14:13.24 | kaldemar | mahiti-irc: i don't know if the GUI has an option for the port. i don't use it. |
| 14:13.28 | Katty | jaytee: :< |
| 14:13.36 | dpreacher | I don't understand VoIP so well as of now, so it'll be some time before I'm comfortable to write something to contribute |
| 14:13.47 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:14.27 | [TK]D-Fender | kaldemar: Well it was always "port" for peer entries |
| 14:15.05 | [TK]D-Fender | dpreacher: You don't seem to be successfully processing the concept "what you are looking for does not exist" |
| 14:15.26 | kaldemar | yes. bindport came with 1.2. |
| 14:15.40 | dpreacher | i agree fender...will work out something else |
| 14:16.50 | Aiatek | [TK]D-Fender the channel #asterisk-es is supported by this channel or not? |
| 14:17.19 | [TK]D-Fender | dpreacher: What you have are "live tests" like I have already give you. You could also use AMI to see if it responds (could get usable info like Zaptel/DAHDI status this way as an extra verification of being "up") |
| 14:17.26 | [TK]D-Fender | Aiatek: Not |
| 14:17.37 | [TK]D-Fender | Aiatek: This channel supports this channel |
| 14:17.39 | Aiatek | [TK]D-Fender ok |
| 14:18.31 | eppigy | hi |
| 14:18.55 | jaytee | buenos dias, muchacho! |
| 14:19.33 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:19.51 | eppigy | HOLA ME AMIGO |
| 14:20.20 | Katty | hugs on eppigy |
| 14:20.53 | jaytee | Estoy trabajando muy difÃcilmente para continuar el aspecto del trabajo difÃcilmente. |
| 14:22.03 | Aiatek | jaytee where are you from? |
| 14:22.32 | jaytee | Beta Omicron IV in the Pleiades sector. |
| 14:23.10 | [TK]D-Fender | Aiatek: Milky Way, Sol 3, northern hemisphere, and you? |
| 14:23.19 | aiksa[LV] | :) |
| 14:23.27 | Aiatek | the nest planet near the sun |
| 14:23.31 | jaytee | everyone always leaves out the third spiral arm |
| 14:23.33 | Aiatek | next* |
| 14:23.45 | aiksa[LV] | [TK]D-Fender: you should put it on you cv |
| 14:24.06 | Aiatek | hehehehhhehe |
| 14:24.11 | *** join/#asterisk propellerhead (n=yogurt2u@host107.190-136-119.telecom.net.ar) |
| 14:24.32 | *** join/#asterisk jtodd (i=mm8p03q3@ns.fox-den.com) |
| 14:24.32 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 14:27.23 | [TK]D-Fender | aiksa[LV]: "Willing to relocate with a range of 50 parsecs |
| 14:27.29 | leifmadsen | anyone know of a softphone that I could provision from a central server, a la polycom? |
| 14:27.55 | [TK]D-Fender | leifmadsen: IIRC Zoiper has that |
| 14:28.03 | leifmadsen | [TK]D-Fender: oh ya? I should check that! |
| 14:28.57 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:28.57 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:30.55 | eppigy | takes the opportunity to smell Katty's neck |
| 14:31.14 | *** join/#asterisk odenkos (n=feeds@85-135-133-212.adsl.slovanet.sk) |
| 14:32.50 | Aiatek | [TK]D-Fender do you speak spanish? |
| 14:33.06 | [TK]D-Fender | Aiatek: Only when I'm ordering from Taco Bell |
| 14:33.19 | Aiatek | hehehehhe |
| 14:34.02 | Aiatek | im asking because we have a opensource congress in my university in september |
| 14:34.03 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 14:34.21 | [TK]D-Fender | Aiatek: My suggestion : filibuster :p |
| 14:34.49 | *** join/#asterisk cheetr (n=tom@oh.you.can.do.such.a.lot.with.a.wompom.com) |
| 14:35.35 | Aiatek | we was looking for an international asterisk speaker |
| 14:36.24 | cheetr | i have a quick question, I've just been setting up asterisk 1.6, and all of the playback prerecorded sounds are being replayed at doublequick speed so they're totally unusable |
| 14:36.49 | Aiatek | three years ago we bring richard stallman |
| 14:36.54 | cheetr | has anyone heard of this before? |
| 14:37.41 | *** join/#asterisk morpheus08 (n=agoradf@88-149-140-45.dynamic.ngi.it) |
| 14:37.45 | morpheus08 | help |
| 14:38.08 | morpheus08 | can help my for a new version asterisk |
| 14:38.20 | morpheus08 | i have a problem with cdr |
| 14:39.04 | morpheus08 | my inglesh not is good but help my |
| 14:40.25 | odenkos | morpheus08: how can we help you, if we don't know what the problem is? |
| 14:41.47 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:42.40 | morpheus08 | cdr report ed databse don't is empity |
| 14:47.06 | *** join/#asterisk kaptengu (n=kaptengu@unaffiliated/kaptengu) |
| 14:47.39 | [TK]D-Fender | bbiab |
| 14:48.04 | Aiatek | ~bbiab |
| 14:48.04 | infobot | i guess bbiab is be back in a bit |
| 14:49.02 | *** join/#asterisk mltlnx (n=mltlnx@cpe-24-193-11-14.nyc.res.rr.com) |
| 14:52.32 | *** join/#asterisk layne (n=layne@ool-44c0048f.dyn.optonline.net) |
| 14:52.44 | *** join/#asterisk Whitor (n=Whitor@64.128.237.124) |
| 14:53.05 | tomodachi | what about encrypted sip traffic between clients and asterisk? |
| 14:53.24 | tomodachi | for example so one can "relatively" safely use laptops allover the world to connect to our pbx |
| 14:53.38 | cheetr | like a vpn? |
| 14:53.39 | *** part/#asterisk gego (n=rick@b238085.customer.hansenet.de) |
| 14:53.46 | Aiatek | tomodachi use vpn |
| 14:54.10 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 14:54.28 | *** join/#asterisk oej_ (n=olle@240.Red-83-33-40.dynamicIP.rima-tde.net) |
| 14:54.28 | *** join/#asterisk jjshoe (n=jjshoe@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 14:54.32 | Kobaz | tomodachi: i like openvpn with a udp server and no compression |
| 14:55.00 | Kobaz | tomodachi: if you use compression, the results of packet loss will be grealy amplified |
| 14:55.29 | *** join/#asterisk rgsteele||work (n=rgsteele@207.106.239.76) |
| 14:56.40 | tomodachi | yeah we have vpn but its another auth layer and password that requires ppl to connect etc.. |
| 14:56.42 | cheetr | is that really a massive hardship for encrypted data? i mean, you'd have to enter some kind of key at some point |
| 14:56.53 | SuPrSluG | how can i set this to yes agi show = No stream file Sends audio file on channel |
| 14:56.58 | *** join/#asterisk oej (n=olle@240.Red-83-33-40.dynamicIP.rima-tde.net) |
| 14:57.23 | rgsteele||work | Hey folks. For some reason, I'm getting the "WARNING[<pid>] file.c: File <somefile> does not exist in any format", even though the file does exist, has the right perms/ownerships, and is in a format approved by my Asterisk installation (.gsm, .wav, and .sln) |
| 14:57.36 | rgsteele||work | Is there any documentation on debugging that problem? |
| 14:57.52 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 14:58.33 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:58.42 | Aiatek | rgsteele||work paste the output error in pastebin and show us |
| 14:59.21 | cheetr | Aiatek: any thoughts on why slin files might play back at the wrong speed? this is a brand new asterisk 1.6 install |
| 15:00.06 | bsilberman | hello all |
| 15:00.19 | bsilberman | question regarding dialing and asterisk 1.4.24 |
| 15:00.32 | rgsteele||work | http://pastie.org/524280 |
| 15:00.33 | NovceGuru | anybody know if you can set a handytone to auto answer? not seeing the options anywhere |
| 15:01.15 | bsilberman | we use asterisk here to dial out, which works fine, but now I am getting reports from a few people doing teh calling that once the remote site answers, they are unable to generate DTMF codes such as 0 for operator... |
| 15:01.15 | NovceGuru | ok maybe I should of googled handytone and not ht-286 |
| 15:01.26 | *** join/#asterisk oej (n=olle@170.Red-88-2-166.staticIP.rima-tde.net) |
| 15:01.51 | bsilberman | the current dtmfmode=rfc2883 is set, i have read that I could set it to =auto |
| 15:01.53 | bsilberman | thoughts? |
| 15:02.07 | rgsteele||work | Aiatek: The perms on the parent dirs are all correct as well (0755), so it really shouldn't have any problems seeing those files on the filesystem. |
| 15:02.08 | *** join/#asterisk oej_ (n=olle@240.Red-83-33-40.dynamicIP.rima-tde.net) |
| 15:03.51 | Aiatek | bsilberman paste what asterisk display in a console when trying to play the files |
| 15:04.13 | bsilberman | okay... i'll get back to you guys |
| 15:04.14 | bsilberman | thanks |
| 15:05.26 | Aiatek | rgsteele||work paste what asterisk display in a console when trying to play the files |
| 15:05.35 | *** join/#asterisk bsilberman (n=bsilberm@65.213.221.252) |
| 15:09.42 | *** join/#asterisk wierdo (n=jimmy@wifi-traf5.networx-bg.com) |
| 15:09.52 | rgsteele||work | Aiatek: http://pastie.org/524293 |
| 15:10.27 | rgsteele||work | Aiatek: That's with SIP debugging ramped up for the inbound host |
| 15:10.47 | rgsteele||work | But, I don't see any useful info in there - nothing beyond what was in the debug logs |
| 15:11.28 | *** part/#asterisk cheetr (n=tom@oh.you.can.do.such.a.lot.with.a.wompom.com) |
| 15:11.31 | Aiatek | what version of asterisk you are using |
| 15:11.45 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 15:11.57 | rgsteele||work | 1.4.17 |
| 15:13.32 | Aiatek | did you try to put your audio files in ./var/lib/asterisk/sounds |
| 15:13.34 | rgsteele||work | Yep, didn't make any difference. |
| 15:13.41 | *** join/#asterisk jdnWEST (n=jdn@mx1.westparkcom.net) |
| 15:13.53 | rgsteele||work | This used to work in 1.2 |
| 15:14.02 | rgsteele||work | I just recently upgraded, and am testing out the new installation. |
| 15:14.15 | kn0x | anyone know of a good way to monitor asterisk for call drops? |
| 15:14.24 | Aiatek | rgsteele||work are you using an extensions of files in dialplan |
| 15:14.27 | Aiatek | ?>? |
| 15:14.34 | kn0x | we've got over 5000 calls per day and they all abort 16 |
| 15:14.40 | kn0x | most are fine |
| 15:14.48 | kn0x | but some poeple atre complaining of sudden call drops |
| 15:14.54 | kn0x | and the hangup cause are allways 16 |
| 15:15.04 | kn0x | every now and then we'll get a cause 0 |
| 15:15.11 | kn0x | but anyone have any ideas? |
| 15:15.48 | rgsteele||work | Aiatek: Nope, but previously I never needed to - it just checked against a list of acceptable extensions and searched for the filename ending in any of those. |
| 15:16.13 | Aiatek | you dont have to use extension |
| 15:16.32 | rgsteele||work | Yeah, AFAIK, you can't use the extension, but that's a discussion for another day. |
| 15:16.32 | [TK]D-Fender | rgsteele||work: "core show modules like format" <- |
| 15:16.40 | [TK]D-Fender | rgsteele||work: make sure your modules are loaded |
| 15:16.51 | [TK]D-Fender | rgsteele||work: And you must not specify the extension |
| 15:17.52 | jplank | whats the dahdi tool the removes unneeded modules from dahdi_genconf? |
| 15:17.53 | rgsteele||work | [TK]D-Fender: Yeah, the extension is not specified. Also, that command doesn't seem to work. |
| 15:18.12 | rgsteele||work | [TK]D-Fender: From the * console, right? |
| 15:18.15 | *** join/#asterisk blargman_ (n=blarg@12.234.16.130) |
| 15:18.21 | [TK]D-Fender | rgsteele||work: Yes |
| 15:19.14 | kaldemar | module show like format then |
| 15:19.21 | rgsteele||work | [TK]D-Fender: Well, "core show codecs audio" works |
| 15:19.44 | rgsteele||work | [TK]D-Fender: http://pastie.org/524305 |
| 15:20.21 | [TK]D-Fender | rgsteele||work: No. Verify that the MODULES loaded |
| 15:20.32 | *** join/#asterisk qdk_ (n=qdk@81.7.168.130) |
| 15:20.32 | *** join/#asterisk joako_ (n=joako@opensuse/member/joak0) |
| 15:21.24 | *** join/#asterisk ricko73 (n=dhartman@wilug/newlug/ricko73) [NETSPLIT VICTIM] |
| 15:21.28 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-f7cd96c2564a302b) [NETSPLIT VICTIM] |
| 15:21.33 | rgsteele||work | [TK]D-Fender: That command you pasted doesn't seem to exist: http://pastie.org/524308 |
| 15:22.04 | *** join/#asterisk oej (n=olle@170.Red-88-2-166.staticIP.rima-tde.net) |
| 15:22.09 | *** join/#asterisk Whitor (n=Whitor@64.128.237.124) [NETSPLIT VICTIM] |
| 15:22.28 | [TK]D-Fender | rgsteele||work: modules show like format |
| 15:23.03 | rgsteele||work | [TK]D-Fender: http://pastie.org/524311 |
| 15:23.19 | rgsteele||work | Looks like gsm is valid and loaded |
| 15:23.29 | [TK]D-Fender | rgsteele||work: Ok, check the perms on the local folder itself |
| 15:23.33 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) [NETSPLIT VICTIM] |
| 15:23.45 | rgsteele||work | So, not sure why it claims the file doesn't exist. I did; 0755 and asterisk:asterisk |
| 15:23.55 | *** join/#asterisk Maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 15:24.03 | *** join/#asterisk oej (n=olle@240.Red-83-33-40.dynamicIP.rima-tde.net) |
| 15:24.20 | jplank | I thought dahdi automagically removed zaptel? |
| 15:24.36 | *** join/#asterisk hfb (n=hfb@pool-98-112-240-188.lsanca.dsl-w.verizon.net) |
| 15:24.36 | jplank | why did I have to do a service zaptel stop before startting dahdi correctly? |
| 15:24.53 | jplank | is it because I forgot to stop zaptel before compiling dahdi? |
| 15:25.21 | [TK]D-Fender | jplank: "Automagically" - I swear every time I hear that word I want to f-ing kill someone. |
| 15:25.26 | jplank | sorry |
| 15:25.43 | jplank | that was the word that was used to describe it to me |
| 15:25.47 | [TK]D-Fender | jplank: And Zaptel was seizing your hardware of course you want it out of the way. |
| 15:25.50 | rgsteele||work | [TK]D-Fender: http://pastie.org/524314 |
| 15:26.19 | [TK]D-Fender | rgsteele||work: Ok, out of ideas ATM |
| 15:26.27 | jplank | I was under the impression dahdi did all the for me? since I stopped zaptel after compiling dahdi, should I recompile for good measure, or should I be good? |
| 15:26.45 | rgsteele||work | [TK]D-Fender: Yeah, me too - guess I'll keep digging. Thanks for the attempts. |
| 15:28.52 | jplank | dahdi_genconf is replying back with Empty configuration -- no spans, did I screw something up? |
| 15:29.38 | jplank | there we go |
| 15:32.55 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:35.26 | tzafrir_laptop | jplank, it is indeed recommended to "stop zaptel". Or rather" unload the zaptel modules |
| 15:36.38 | tzafrir_laptop | jplank, though the module zaptel itself can't be unloaded as long as asterisk is still running |
| 15:37.11 | tzafrir_laptop | Which reduces the ability to do that automagically |
| 15:37.32 | tzafrir_laptop | hopes [TK]D-Fender's IRC client is not connected to a TTS software |
| 15:40.20 | jplank | I always stop asterisk when working with zaptel/dahdi, I've seized remote boxes twice forgetting to do that |
| 15:40.30 | jplank | done with the upgrade anyway |
| 15:40.41 | jplank | a dahdi_cfg cleared all my errors once I stopped zaptel |
| 15:40.50 | vAd0r | ,mysqladmin: Can't turn off logging; error: 'Access denied; you need the SUPER privilege for this operation' |
| 15:40.57 | vAd0r | why do i get that when i try to change my password |
| 15:41.05 | Nugget | nothing about mysql is SUPER :) |
| 15:41.31 | [TK]D-Fender | vAd0r: How is * causing that? |
| 15:41.40 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 15:41.55 | vAd0r | im typing mysqladmin -u asteriskuser -p password MYNEWPASSWORD |
| 15:42.09 | vAd0r | then it asks me to type the password to allow it and i put amp109 |
| 15:42.30 | [TK]D-Fender | vAd0r: Maybe you should ask in #freepbx or #mysql |
| 15:42.45 | vAd0r | they dont know |
| 15:42.53 | odenkos | LoL |
| 15:42.54 | odenkos | :D |
| 15:43.03 | vAd0r | http://freepbx.org/support/documentation/faq/changing-the-mysql-password |
| 15:43.04 | jaytee | vAd0r, just Google "resetting mysql password" |
| 15:43.05 | *** join/#asterisk l2trace99 (n=jr@138.210.113.253) |
| 15:43.20 | vAd0r | i know how to do it it is like i dont have permission |
| 15:43.32 | rgsteele||work | tzafrir_laptop: Have you done any work on file.c? |
| 15:45.10 | odenkos | vAd0r: well then why are you trying to change it if you don't have permission? :D |
| 15:46.01 | vAd0r | im logged in as root |
| 15:46.03 | vAd0r | i should |
| 15:46.12 | rgsteele||work | I'm trying to debug a problem where file.c won't open files that are there, have the right permissions and ownerships, and an extension for which a module that's been successfully loaded. But, strace doesn't reveal where it's actually looking for the file. |
| 15:46.48 | Deeewayne | Anyone here using Fax For Asterisk ? |
| 15:46.49 | Joel | vAd0r #mysql |
| 15:47.19 | vAd0r | thats a greate idea i didnt try that first |
| 15:47.31 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:47.35 | Joel | vAd0r being logged into a machine as root has nothing to do with mysql's 'root' user. #myqlfordummies? |
| 15:49.39 | *** join/#asterisk theron (n=theron@216.51.246.211) |
| 15:49.40 | *** join/#asterisk seanmh (n=johndoe@198.59.129.24) |
| 15:50.46 | vAd0r | what is the default mysql password |
| 15:51.45 | jplank | another successful upgrade from zaptel to dahdi, I'm getting good at this ;P |
| 15:51.51 | Nugget | yay dahdi |
| 15:51.53 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:52.13 | eppigy | vAd0r: is it blank by default |
| 15:52.23 | jplank | I hate upgrading anything, but the half duplex issue with digium cards on zaptel was killing me |
| 15:52.31 | eppigy | so you can usually log in with just mysql -u root |
| 15:56.46 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 15:58.40 | *** join/#asterisk tobias (n=tobias@216.27.28.176) |
| 15:58.54 | tzafrir_laptop | rgsteele||work, what do you mean? What is the exact line you see in strace? |
| 16:00.39 | *** part/#asterisk dpreacher (n=dpreache@117.195.35.72) |
| 16:01.18 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 16:04.00 | *** join/#asterisk mltlnx (n=mltlnx@asams.mserve.com) |
| 16:06.36 | jdnWEST | Any recomendations on a particular model for a cheap SIP---> PRI gatway that supports 2 ports maybe? |
| 16:07.19 | rgsteele||work | tzafrir_laptop: The same thing I get in the console: http://pastie.org/524293 |
| 16:07.55 | rgsteele||work | Asterisk tells me that it's loaded the .gsm extension, but appears to only look for .ulaw files (if you look in that posting) |
| 16:09.16 | rgsteele||work | tzafrir_laptop: 'module show like format' says that .gsm and .wav modules have been loaded, but the asterisk error message show it's just looking for ulaw: "File local/hours-open does not exist in any format; file.c:871 ast_streamfile: Unable to open local/hours-open (format 0x4 (ulaw)): No such file or directory" |
| 16:09.28 | rgsteele||work | [TK]D-Fender: That might shed some light on the situation. |
| 16:09.52 | rgsteele||work | It looks like the message says "I can't find any format of the file (I looked for ulaw)" |
| 16:09.54 | *** join/#asterisk jksM (i=jks@193.189.93.254) |
| 16:10.30 | [TK]D-Fender | rgsteele||work: No, that says the preferred format based on the channel |
| 16:10.53 | rgsteele||work | Oh. |
| 16:10.55 | [TK]D-Fender | jdnWEST: clarify "gateway" please... |
| 16:10.55 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:10.56 | rgsteele||work | sighs |
| 16:12.08 | rgsteele||work | I have *zero* idea why this doesn't work - I think the error it's spouting is not actually the problem, which makes it pretty difficult to debug. |
| 16:13.38 | *** join/#asterisk zchaos (n=none@CPE001d7ef0ba9d-CM001ceab63f9a.cpe.net.cable.rogers.com) |
| 16:16.26 | jdnWEST | D-Fender - Something that has a PRI interface that goes to my legacy switch, and eithernet on the other that I point at vitelity or someone |
| 16:17.26 | [TK]D-Fender | jdnWEST: So no * in the middle, straigh PRI > SIP. NOT cheap... |
| 16:17.47 | jdnWEST | Fender: quite aware, need "enteprise grade" and i'm going to have 2 of em |
| 16:17.47 | [TK]D-Fender | jdnWEST: You'd be looking at an AudioCodes Mediant gateway |
| 16:17.49 | *** join/#asterisk ingenius (n=alektro@host147.190-229-174.telecom.net.ar) |
| 16:17.58 | jdnWEST | Fender: That better than the adtrans? |
| 16:18.19 | [TK]D-Fender | jdnWEST: I'd trust Adtran on name alone |
| 16:18.55 | jdnWEST | Fender: I've noticed that all the serious voip shops arn't selling adtran anymore, and seem to push Audio codes. |
| 16:19.14 | [TK]D-Fender | jdnWEST: http://www.voipsupply.com/audiocodes-mediant-1000-2e1 |
| 16:19.56 | jdnWEST | Fender: I've got 2 asterisk boxes also, but i'd rather this be done by something dedicated :) |
| 16:20.01 | jdnWEST | fender: thanks! |
| 16:20.32 | Katty | mew. |
| 16:24.12 | rgsteele||work | Where can I change the preferred audio format? |
| 16:24.13 | Katty | hum. |
| 16:24.18 | Katty | rgsteele||work: for? |
| 16:25.03 | rgsteele||work | Asterisk. It's looking for ulaw files by preference before all other audio formats. |
| 16:25.08 | Katty | one of our sales people was arguing with me about what was better. a sprint blackberry or an iphone. |
| 16:25.11 | Katty | is peeved. |
| 16:25.55 | rgsteele||work | Katty: I've managed to stump the channel as far as my audio problem goes, so my latest idea/yet-to-be-executed-hack is to change the audio preference to .gsm or .wav and see if it changes anything |
| 16:28.11 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 16:29.30 | *** join/#asterisk neurosys (n=neurosys@173-9-159-182-miami.txt.hfc.comcastbusiness.net) |
| 16:36.16 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 16:36.39 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 16:36.55 | *** join/#asterisk micols (n=mio@rlogin.dk) |
| 16:37.02 | *** join/#asterisk Victor_Yure_ (n=victor@unaffiliated/victoryure/x-837844) |
| 16:38.41 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
| 16:44.29 | *** join/#asterisk bbryant1 (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 16:48.27 | rgsteele||work | [TK]D-Fender: Take a look at this... are you sure it's not saying it's only going to accept ulaw? http://pastie.org/524398 |
| 16:48.42 | rgsteele||work | I can't find any other rational explanation for this behavior and these log messages. |
| 16:51.16 | *** join/#asterisk clintc (n=clintc@n128-227-127-75.xlate.ufl.edu) |
| 16:51.28 | jplank | if I have 2 digit extensions starting with a 1, and the customer wanted to be able to dial without a 9, something like 1XT|1XXXXXXXXXX should work with polycom, right? |
| 16:54.22 | eppigy | TRABAJO |
| 16:55.49 | *** join/#asterisk kuku1 (n=kuku1@66.103.31.15) |
| 16:56.20 | kuku1 | When running 1.6, I can no longer add a cell phone number to be an agent in a queue, any suggestsions ? |
| 16:56.53 | Nugget | why would it matter that it's a cell phone number? how would asterisk even know. |
| 16:57.03 | Nugget | maybe you could clarify what "can no longer add" really means |
| 16:57.08 | Nugget | and re-think the nature of the problem |
| 16:58.02 | kuku1 | ok |
| 16:58.52 | kuku1 | I had the code working in 1.2 |
| 17:00.38 | kuku1 | I'll get some pastebin urls shortly |
| 17:03.43 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:04.18 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 17:06.57 | kuku1 | AgentCallbackLogin no longer exists in 1.6 |
| 17:09.26 | leifmadsen | kuku1: that is true -- it was deprecated in 1.4 |
| 17:10.59 | chrisdus | hello all... i have managed to fix most of the problems we had with our Asterisk server thanks to all your help, much appreciated! I still have one problem left, and am not sure if I understand the concept ... voicemail.. Now, i can leave voicemail messages no problem when I dial an extension and he does not pick up, AND the message is sent via email to the user too... the problem is that when we dial the voicemail number from the phone (MITEL) which |
| 17:11.34 | leifmadsen | chrisdus: your client cut off at "...phone (MITEL) whic" |
| 17:11.40 | leifmadsen | chrisdus: I suggest X-Chat :) |
| 17:11.53 | chrisdus | Ha, cool thanks leifmadsen! |
| 17:12.16 | chrisdus | Now, i can leave voicemail messages no problem when I dial an extension and he does not pick up, AND the message is sent via email to the user too... |
| 17:12.20 | chrisdus | the problem is that when we dial the voicemail number from the phone (MITEL) which is blinking "Hey! You have a voicemail!", get to the voicemail menu, but it will then state there are no messages! |
| 17:12.24 | chrisdus | this better? :D |
| 17:13.09 | kuku1 | ok, so here are my errors: http://pastebin.ca/1474437 |
| 17:13.27 | kuku1 | if someone needs more lines let me know |
| 17:13.45 | chrisdus | The voice mail number (8500) is coded as: |
| 17:13.48 | chrisdus | exten => 8500,1,VoicemailMain |
| 17:13.48 | chrisdus | exten => 8500,n,Goto(s,6) |
| 17:13.51 | kuku1 | leifmadsen: so if its gone, now i have problems adding extensions that are cell phone numbers |
| 17:14.11 | leifmadsen | kuku1: it needs to be done via dialplan now -- there is an example script in the doc/ directory |
| 17:14.27 | *** join/#asterisk tobias (n=tobias@216.27.28.176) |
| 17:14.34 | kuku1 | do i have to do it using the NEW dialplan ? |
| 17:14.44 | leifmadsen | kuku1: no, you can use regular dialplan |
| 17:15.02 | leifmadsen | I will be writing an article on converting from using AgentCallbackLogin() at some piont |
| 17:15.04 | kuku1 | I read it, and I only saw a reference to the new dialplan |
| 17:15.20 | leifmadsen | kuku1: yes, it is written in AEL, but you don't have to use AEL |
| 17:15.31 | leifmadsen | plus, you can use both AEL and regular dialplan at the same time |
| 17:15.37 | kuku1 | i havent seen an example in non-ael then... |
| 17:15.56 | kuku1 | im stuck, my tech support line is not dialing the tech's cell phones.... |
| 17:16.45 | leifmadsen | you didn't test this on a development system first? |
| 17:16.54 | leifmadsen | 1.4 and 1.6.x are quite different |
| 17:16.57 | eppigy | who has time for that |
| 17:17.04 | leifmadsen | eppigy: I have no idea! |
| 17:17.17 | leifmadsen | not me -- I still run 1.4 at home, and 1.6.x only in new developments |
| 17:17.37 | eppigy | I am still running 1.4.11 enterprise wide |
| 17:24.26 | *** join/#asterisk Abel408 (i=42a2b66c@gateway/web/freenode/x-0b1e691a41be49e0) |
| 17:25.34 | Abel408 | Hello everyone! I believe I'm having a firewall issue, but I wanted to make certain. I can call others from asterisk and they can her me and I can hear them, but when they call me niether party can hear eachother |
| 17:26.48 | kuku1 | leifmadsen: yeh, i didnt. Any urls i can look at ? |
| 17:26.55 | leifmadsen | none that I'm aware of |
| 17:27.08 | [TK]D-Fender | Abel408: ... |
| 17:27.10 | [TK]D-Fender | ~sipnat |
| 17:27.10 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:27.11 | leifmadsen | conversion of AgentCallbackLogin() to dialplan has not been well documented |
| 17:27.13 | [TK]D-Fender | ^^^^ |
| 17:27.21 | leifmadsen | I will change that, but not today. |
| 17:28.03 | *** join/#asterisk moy (n=moy@74.12.123.90) |
| 17:28.04 | eppigy | never change |
| 17:28.04 | jplank | grrr I guess I was wrong about this dahdi upgrade, when I run dahdi show channels its blank, and when I do dahdi show status, every channel (2400p) shows red |
| 17:29.56 | jplank | actually dahdi show status shows no alarms |
| 17:30.17 | jplank | neither does dahdi scan |
| 17:31.31 | Abel408 | [TK]D-Fender: huh? |
| 17:32.45 | jplank | jheh, no chan_dahdi.conf |
| 17:32.46 | jplank | :) |
| 17:32.47 | [TK]D-Fender | Abel408: Read the GUIDE I just had infobot link you |
| 17:32.59 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:33.20 | Abel408 | ok thanks |
| 17:37.38 | Katty | fullllllllllllllllllllllllll |
| 17:37.40 | Katty | sprawls. |
| 17:45.45 | *** join/#asterisk ruben23 (n=RPL@124.107.3.178) |
| 17:47.26 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 17:49.57 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
| 17:50.26 | *** join/#asterisk scarecrowed (n=fabiano@200.101.115.243) |
| 17:51.36 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 18:00.52 | eppigy | Katty: :D |
| 18:07.00 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 18:08.58 | *** join/#asterisk Micc (n=dotirc@c-76-121-255-52.hsd1.wa.comcast.net) |
| 18:09.15 | Micc | I recently upgraded to 1.6.1.1 and now when I reload it crashes. |
| 18:09.35 | Micc | It looks like it crashes after res_odbc reconnects. |
| 18:09.54 | Kobaz | micc: recompile with debugging flags and get a backtrace |
| 18:10.30 | Micc | kobaz, do I set that in the menuselect? |
| 18:10.39 | Kobaz | it's in one of the menues |
| 18:10.43 | Juggie | also make sure there are no old modules loaded |
| 18:10.45 | Kobaz | additional options or something |
| 18:10.56 | Juggie | you can blow away your modules dir and do a make install again to ensure you only have modules from 1.6.1.1 |
| 18:10.59 | Kobaz | yeah, reinstall cleanly |
| 18:11.03 | Kobaz | delete /var/lib/asterisk/modules |
| 18:11.05 | Kobaz | and reinstall |
| 18:11.20 | *** join/#asterisk gsiener (n=gsiener@216.255.164.190) |
| 18:11.20 | Kobaz | and then try again.. and then if it crashes.. get the backtrace |
| 18:11.30 | *** join/#asterisk steve2 (n=steve@87-194-204-58.bethere.co.uk) |
| 18:12.06 | Micc | DEBUG_THREADS? |
| 18:12.10 | Kobaz | nope |
| 18:12.13 | Kobaz | debugging symbols |
| 18:12.15 | Kobaz | or something like that |
| 18:12.18 | Kobaz | i dont have it in front of me |
| 18:12.39 | Micc | is it under Compiler Flags? |
| 18:12.42 | Micc | I don't see it. |
| 18:12.46 | Kobaz | probably |
| 18:13.02 | Kobaz | i'll find it in a sec... bidding on some stuff on ebay |
| 18:15.30 | steve2 | hi guys |
| 18:15.30 | steve2 | does askerisk support voice modems? |
| 18:15.32 | Qwell | steve2: no |
| 18:15.32 | Kobaz | you wouldn't want it to anyway |
| 18:15.32 | Qwell | exactly |
| 18:15.32 | steve2 | doh |
| 18:15.32 | kaldemar | Micc: DONT_OPTIMIZE under compiler flags |
| 18:15.33 | steve2 | well i have a device that accepts AT commands on one com port, and send/receive raw PCM 8Khz voice on another |
| 18:15.34 | steve2 | is there a channel driver that's close to that functionality I can edit? |
| 18:15.34 | gsiener | Hi all. Not sure where I should asking so I will try here. I've got an Asterisk box (1.4.21-2) that has a number of incoming IPKall accounts (via IAX2). This has been working w/o issue, and the asterisk box has a dedicated public ip. The firewall/router we had (Sonicwall) just died and we have a Linksys dd-wrt router in its place as we await a replacement. I've forwarded all ports and the public IP via iptables and our outgoing call service is wor |
| 18:15.34 | steve2 | i.e. this thing is full duplex, not like AT+VRX on voice modems etc |
| 18:15.35 | *** join/#asterisk telnettech (n=telnette@12.71.238.130) |
| 18:15.43 | kaldemar | gsiener: your message got cut at "call service is wor..." |
| 18:16.13 | gsiener | ...call service is working and registered via the proper ip. Incoming calls on the IPKall lines are not going through, and I don't see anything on tcpdump or in the asterisk logs. Any thoughts on what to troubleshoot? Is there a port besides 4569 that IAX2 expects to use? |
| 18:18.34 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 18:20.12 | Katty | paints nails. |
| 18:20.31 | Kobaz | watches intently |
| 18:20.55 | kaldemar | gsiener: if you can't see any packets on the asterisk box, it sounds like a network problem. IAX2 only uses UDP 4569. |
| 18:21.27 | gsiener | kaldemar: thanks, just wanted to confirm that. |
| 18:24.15 | Micc | ok. so after I'm running the dont_optimize build, how do I get a back_trace? |
| 18:24.26 | *** part/#asterisk gsiener (n=gsiener@216.255.164.190) |
| 18:27.18 | Abel408 | [TK]D-Fender: I don't think it is a nat problem because the ip address never gets changed |
| 18:28.19 | *** join/#asterisk thansen (n=thansen@c-24-10-236-153.hsd1.ut.comcast.net) |
| 18:29.08 | Micc | it tells me format_mp3 isn't built with this version of asterisk, but I just rebuilt it. |
| 18:30.12 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 18:30.52 | ariel_ | afternoon folks |
| 18:31.06 | Micc | oh, I guess it doesn't know if its compatible or not, it just says it wasn't installed by this version of asterisk. |
| 18:31.50 | Micc | Its from asterisk-addons-1.6.1.0 |
| 18:32.04 | Micc | its one minor version off, that shouldn't matter should it? |
| 18:32.27 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:34.58 | Katty | Kobaz: linkin park after midnight. |
| 18:36.40 | Kobaz | heh |
| 18:37.47 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-4463911808cef870) |
| 18:38.18 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 18:38.47 | *** join/#asterisk LemensTS (n=customgt@207.160.41.191) |
| 18:40.14 | LemensTS | Hello All. Im trying to receive a digit pressed using exec background via phpagi...http://pastebin.com/m65f53f72 The result is say 0 instead of 5 (what im pressing) |
| 18:40.15 | ariel_ | Ice Cream in the Kitchen at 3:30 pm this Friday.... but I am want some now! |
| 18:40.42 | LemensTS | stream_file says it plays in the cli, but i dont hear it in the phone. |
| 18:40.51 | LemensTS | *noticed other people have the same problem |
| 18:41.16 | *** join/#asterisk ACK-NAK (n=Miranda@home.chicagoventure.com) |
| 18:41.17 | LemensTS | http://pastebin.com/m65f53f72 there is clickable now sorry |
| 18:41.17 | telnettech | TK: I am needing some help |
| 18:41.33 | rob0 | Psychiatric help, 5c. |
| 18:41.45 | telnettech | http://pastebin.com/d6fca3db3 I am getting this error when i load chan_dahdi.so module. I have included my config files |
| 18:42.16 | telnettech | TK: will you please look at this and tell what i may have done incorrectly |
| 18:43.19 | Abel408 | Is this a firewall issue? We are not getting audio on either end for incoming calls. Everything else works fine. We are NOT behind nat. |
| 18:43.53 | ariel_ | is rtp being blocked? |
| 18:44.16 | Abel408 | possibly. I will check |
| 18:46.16 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 18:47.34 | *** join/#asterisk Primer (n=vi@216.239.132.98) |
| 18:48.09 | *** join/#asterisk boch (n=fran@200.61.191.9) |
| 18:48.33 | Primer | Anyone running on Centos 5.3, run /etc/init.d.dahdi (with all modules disabled in /etc/dahdi/modules.conf commented out), and have their box instantly lock up? |
| 18:48.38 | Primer | Or, have its network die? |
| 18:49.04 | boch | anyone used MESSAGE method with SPA922? |
| 18:49.18 | Primer | I just ran /etc/init.d/dahdi start, which reported: No hardware timing source found in /proc/dahdi, loading dahdi_dummy |
| 18:49.27 | Primer | And my ssh connection immediately died |
| 18:51.23 | *** join/#asterisk Joel (n=jjshoe@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 18:51.32 | tzafrir_laptop | Primer, your server decided to panic? |
| 18:51.58 | rgsteele||work | Weird - asterisk seems to crash every time it registers the VPB driver |
| 18:52.06 | tzafrir_laptop | can you (or anybody) see what's on the console? |
| 18:52.10 | Primer | tzafrir_laptop: it seems so |
| 18:52.28 | tzafrir_laptop | rgsteele||work, hmm... I have an open bug for that |
| 18:52.37 | rgsteele||work | Ah, yeah https://issues.asterisk.org/view.php?id=12096 |
| 18:52.55 | rgsteele||work | I even tried a noload => chan_vpb.so but it doesn't seem to help |
| 18:52.55 | tzafrir_laptop | rgsteele||work, what version of asterisk? on older version is was a genuine bug in libvpb |
| 18:53.08 | rgsteele||work | 1.4.17 |
| 18:53.41 | tzafrir_laptop | rgsteele||work, hang on: the crash is probably in the module that loads right after vpb |
| 18:54.15 | rgsteele||work | tzafrir_laptop: Ah, okay, this time noload worked |
| 18:54.23 | rgsteele||work | tzafrir_laptop: Maybe I fat-fingered it last time. |
| 18:54.23 | tzafrir_laptop | the issue I had in mind is: https://issues.asterisk.org/view.php?id=14970 |
| 18:54.29 | tzafrir_laptop | patch there is pure voodoo |
| 18:55.04 | rgsteele||work | tzafrir_laptop: I don't have any VoiceTronix cards on this box - just a pair of Digium TDM400P's |
| 18:55.28 | tzafrir_laptop | 1.4.17 - sounds like it is before the fix . The code you see there is probably not yet in chan_vpb |
| 18:55.31 | rgsteele||work | Once I issued a 'noload => chan_vpb.so', it seemed to not crash :) |
| 18:56.02 | tzafrir_laptop | rgsteele||work, yes, check your vpb config (/etc/vpb-something) |
| 18:57.22 | leifmadsen | Asterisk 1.4.26-rc4 and Asterisk 1.6.0.11-rc1 are now available! Please see http://www.asterisk.org/node/48604 and http://www.asterisk.org/node/48605 for more information. Thank you for your continued support of Asterisk! |
| 18:59.14 | telnettech | need some help. getting following error on CLI http://pastebin.com/d6fca3db3 I am getting this error when i load chan_dahdi.so module. I have included my config files |
| 19:01.31 | *** join/#asterisk gpled (n=gpled@firewall.fccfurn.com) |
| 19:02.01 | gpled | anyone know of a good source for open source, on hold music? |
| 19:02.37 | leifmadsen | gpled: http://www.opensourcemusic.com ? |
| 19:04.04 | leifmadsen | plus it has links to other sites |
| 19:04.14 | gpled | leifmadsen: thanks |
| 19:04.16 | *** part/#asterisk LemensTS (n=customgt@207.160.41.191) |
| 19:04.21 | leifmadsen | gpled: google helped me out... |
| 19:04.44 | Abel408 | Ok here is my issue. Asterisk is connected to a media gateway which is connected to a Fujitsu phone system. There is no NAT. Everything works great except niether end can hear eachother only on incoming calls. I can make an external call fine. I can also leave a message on asterisk from an outside line and I can hear that message from an internal line. What could this be? |
| 19:05.17 | Primer | This is what I get for installing a package from a third part provider...sigh...now, since the rpm for dahdi set it to load on boot, my box crashes as soon as the dahdi init.d script runs |
| 19:05.34 | leifmadsen | Abel408: canreinvite=no ? |
| 19:05.52 | Abel408 | where is that? |
| 19:05.55 | vAd0r | anyone know any free inbound sip providers |
| 19:06.02 | leifmadsen | Primer: do an interactive startup and skip it |
| 19:06.06 | leifmadsen | Abel408: sip.conf |
| 19:06.16 | Primer | leifmadsen: it's a VM 3000 miles away |
| 19:06.42 | Primer | and the serial console info I have for it is in an email...on the box... |
| 19:06.43 | leifmadsen | Primer: oops then :) |
| 19:06.58 | rgsteele||work | tzafrir_laptop: Hang on a sec |
| 19:07.40 | Abel408 | @leifmadsen: It is set to default... yes |
| 19:07.55 | leifmadsen | Abel408: and I gave you the option that may make it work |
| 19:09.56 | rgsteele||work | tzafrir_laptop: Were you looking for something specific in that config file? |
| 19:10.14 | tzafrir_laptop | is there any card listed there? |
| 19:10.37 | Abel408 | @leifmadsen: That setting is for users behind a nat correct? |
| 19:10.41 | ruben23 | hi anyone havig issue with voip provider, |
| 19:10.52 | leifmadsen | Abel408: yes |
| 19:11.03 | leifmadsen | Abel408: or whenever you don't want audio to be reinvited |
| 19:11.25 | Abel408 | So shouldn't I leave this as yes? |
| 19:13.52 | leifmadsen | Abel408: it depends on your configuration, but if the reinvite isn't happening correctly, or your network routing and configuration doesn't allow for the audio to pass directly between end points, then you may need to disable it |
| 19:13.52 | leifmadsen | Abel408: I'm just throwing out a guess since you didn't give a whole lot of info to go on |
| 19:14.20 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 19:15.40 | *** join/#asterisk s0lid (n=s0lid@122.53.110.183) |
| 19:17.21 | *** join/#asterisk metfan2007 (n=metfan20@201.103.89.156) |
| 19:18.16 | metfan2007 | hi all! I need to do a trunk between Asterisk and a Nortel Meridian Option 11, do you know if is it possible with SIP? |
| 19:20.12 | *** join/#asterisk rgsteele||work (n=rgsteele@207.106.239.76) |
| 19:20.22 | *** join/#asterisk yerwish (n=geekangl@86.51.52.131) |
| 19:20.36 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:20.56 | rgsteele||work | tzafrir-laptop: Did you need me to relay the contents of my vpb config file(s)? |
| 19:26.19 | Abel408 | @leifmadsen: Can I give you more information that you might need to help me? Thanks for being helpful |
| 19:26.33 | leifmadsen | Abel408: you can give it to the room and if someone can help, they will |
| 19:27.10 | *** join/#asterisk j_kroon (n=jkroon@dsl-240-167-45.telkomadsl.co.za) |
| 19:28.52 | rgsteele||work | [TK]D-Fender: I figured out why the audio wasn't playing. |
| 19:29.06 | rgsteele||work | Asterisk was looking at /usr/share/asterisk/sounds, not /var/lib/asterisk/sounds |
| 19:29.25 | rgsteele||work | It would have been *awesome* if strace or debug/verbose settings or ANYTHING had let me know what the fully qualified path was |
| 19:29.32 | rgsteele||work | Would have saved several hours of hair pulling. |
| 19:31.31 | Abel408 | Here is my problem: The Asterisk server is NOT behind NAT. On an incomming call neither side has audio. Internal and outgoing calls work fine. An outside line can also leave a message on the asterisk server and the inside line can listen to the message. One would think that the problem lies between the asterisk box and the inside phone, but internal calls work fine. It is only when an outside line dials to an inside line. |
| 19:33.03 | *** join/#asterisk mltlnx (n=mltlnx@cpe-74-73-146-155.nyc.res.rr.com) |
| 19:33.04 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:34.11 | SuPrSluG | anyone know of a presence client like HUD lite that will work with opensips/kamailio ? |
| 19:34.50 | *** join/#asterisk mltlnx (n=mltlnx@67.102.249.50) |
| 19:35.48 | *** join/#asterisk mltlnx (n=mltlnx@cpe-74-73-146-155.nyc.res.rr.com) |
| 19:36.04 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.216) |
| 19:38.28 | *** part/#asterisk gpled (n=gpled@firewall.fccfurn.com) |
| 19:40.39 | Abel408 | @leifmadsen: I love you |
| 19:41.03 | leifmadsen | Abel408: that's what my g/f says too |
| 19:41.14 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 19:41.17 | leifmadsen | (except she adds, "long time") |
| 19:41.24 | seanbright | leifmadsen: right, except Abel408 is telling the truth |
| 19:41.33 | seanbright | zing |
| 19:41.40 | ariel_ | love you long time,, hummm far east |
| 19:41.42 | *** join/#asterisk korihor (n=korihor@190.205.253.235) |
| 19:41.52 | rob0 | Doesn't your GF call you Blitzrage? |
| 19:42.15 | Abel408 | @leifmadsen: canreinvite=no worked. Now I gotta figure out why |
| 19:42.49 | leifmadsen | Abel408: because the audio was being reinvited to a location that isn't routable in your topology |
| 19:43.13 | leifmadsen | the two end points either can't talk to each other directly, one of the end points doesn't support what you're doing, or some other random reason |
| 19:43.18 | Abel408 | @leifmadsen: yea I think an acl on one of the cisco routers is blocking it |
| 19:44.19 | Katty | sways |
| 19:44.43 | ariel_ | I am hungry |
| 19:45.32 | Qwell | ariel_: what're you making us? |
| 19:46.06 | ariel_ | I have to hit the Vending machine. |
| 19:47.01 | *** join/#asterisk mltlnx (n=mltlnx@cpe-74-73-146-155.nyc.res.rr.com) |
| 19:51.45 | *** join/#asterisk timeshell_atwork (n=chatzill@gw.lusi.on.ca) |
| 19:54.17 | ariel_ | Qwell: Granola bar, or Famos Ages cookies.??? |
| 19:54.45 | e0n` | FreePBX or ney |
| 19:55.00 | e0n` | need something easily manageable by someone other than myself technically that can setup extensions easily |
| 19:56.02 | ariel_ | e0n`: AsteriskNow, PBX in a Flash or Elastix |
| 19:56.06 | ariel_ | all good |
| 19:56.57 | e0n` | got it |
| 19:57.07 | *** join/#asterisk steve2 (n=steve@87-194-204-58.bethere.co.uk) |
| 19:57.11 | Micc | anyone using 1.6.1.1 setup multiple parking lots yet? |
| 19:58.51 | ariel_ | Micc: not I |
| 19:58.59 | ariel_ | still on 1.4.25.1 |
| 19:59.03 | ariel_ | or less |
| 19:59.19 | Micc | I haven't been able to get it to work yet. |
| 19:59.59 | *** join/#asterisk exc_ess (n=exc_ess@69-165-165-12.dsl.teksavvy.com) |
| 20:02.54 | kuku1 | Qwell: any new regarding AgentCallbackLogin in 1.6, documentation to do that non-ael |
| 20:08.02 | *** join/#asterisk mltlnx (n=mltlnx@cpe-74-73-151-103.nyc.res.rr.com) |
| 20:13.20 | saxa | hi, is there someway I can configure a voicemail if my remote asterisk box is offline ? |
| 20:13.39 | saxa | i use IAX2 to connect two asterisk boxes |
| 20:14.24 | saxa | so when the box which receives a call on Zap/1 then call the remote box via IAX2 and if this remote box doesnt answer |
| 20:14.44 | saxa | I would like that the voicemail activates on the first box |
| 20:15.33 | *** join/#asterisk Greek-Boy (n=greek@41.222.89.77) |
| 20:15.41 | [TK]D-Fender | saxa: Its your dialplan, do whatever you want |
| 20:15.52 | saxa | :) |
| 20:16.06 | saxa | so this is possible i suppose |
| 20:16.20 | kuku1 | [TK]D-Fender: any suggestions on how to do agentcallbacklogin in 1.6 ? (its not in 1.6 ) |
| 20:16.41 | saxa | but how can I activate the voicemail just on that s extension answer ? |
| 20:17.03 | [TK]D-Fender | kuku1: "core show application addqueuemember" <- |
| 20:17.44 | [TK]D-Fender | saxa: "Activate"? Its your dialplan, Dial the other box, and if the call isn't answered you dialplan CONTINUES. Just shove your call to Voicemail there |
| 20:18.18 | saxa | oh ok, i dint tought that, sorry |
| 20:18.21 | kuku1 | [TK]D-Fender: I have addqueuemember |
| 20:18.37 | kuku1 | [TK]D-Fender: here are my errors: http://pastebin.ca/1474437 |
| 20:18.49 | saxa | hides, as know i remember that |
| 20:19.01 | *** join/#asterisk eric_hill (i=eric_hil@204.94.175.11) |
| 20:19.15 | *** join/#asterisk thansen (n=thansen@7.247.sfcn.org) |
| 20:19.38 | *** join/#asterisk degrade (n=degrade@unaffiliated/degrade) |
| 20:20.01 | [TK]D-Fender | kuku1: You have no "1" priority for your "1" exten, and previous failed logins probably left illegal devices with WHITE-SPACE in them |
| 20:20.05 | eric_hill | Anyone else see asterisk 1.6.1.1 stop servicing SIP requests after 60 minutes? UDP in-Q starts filling (netstat -an) and phones go unregistered. |
| 20:20.15 | [TK]D-Fender | kuku1: Go dump your queue tos ee what bad entries you added and remove them |
| 20:20.32 | [TK]D-Fender | kuku1: "queue show service_queue" |
| 20:20.41 | kuku1 | [TK]D-Fender: I do< I just didnt' include it. |
| 20:21.01 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 20:21.01 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 20:21.46 | kuku1 | [TK]D-Fender: http://pastebin.ca/1474641 |
| 20:21.56 | *** join/#asterisk BadHAL (n=nn@70.99.106.38) |
| 20:22.18 | [TK]D-Fender | kuku1: A whole buch oc clrealy bad ones there |
| 20:22.22 | Katty | okay now i'm just plain pissy. |
| 20:22.23 | kuku1 | ok |
| 20:22.28 | [TK]D-Fender | kuku1: A whole bunch of clearly bad ones there |
| 20:22.31 | kuku1 | I need to delete them somehow |
| 20:22.41 | kuku1 | do I need to make a new queue then.. |
| 20:22.46 | *** join/#asterisk dmz (n=dmz@64.203.233.195.dyn-cm-pool35.hargray.net) |
| 20:22.47 | [TK]D-Fender | kuku1: "core show application removequeuemember" <- |
| 20:26.29 | *** join/#asterisk propellerhead (n=yogurt2u@host106.190-30-247.telecom.net.ar) |
| 20:26.36 | timeshell_atwork | TK |
| 20:26.43 | timeshell_atwork | Sangoma vs Asterisk |
| 20:26.51 | timeshell_atwork | I mean Digium |
| 20:26.53 | timeshell_atwork | :p |
| 20:27.06 | timeshell_atwork | Sangoma vs Digium TDM analog card |
| 20:27.15 | kuku1 | [TK]D-Fender: I dont know how to remove the bad ones, since its obvious I have some spacing problems |
| 20:27.51 | [TK]D-Fender | kuku1: "core show application removequeuemember" <- |
| 20:28.38 | [TK]D-Fender | timeshell_atwork: grammar is very English good your! |
| 20:28.58 | [TK]D-Fender | timeshell_atwork: Talks does funny Yoda, hhhmmmmmmmmmmmmMMMMMMMMM?!?!?!?!??! |
| 20:29.00 | ISO9001 | lol |
| 20:29.11 | kuku1 | [TK]D-Fender: Yes |
| 20:29.22 | rob0 | Miss Piggy |
| 20:29.31 | [TK]D-Fender | hhhhhhhhiiiiyaaaaaaaaa!! |
| 20:34.41 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 20:36.08 | kuku1 | [TK]D-Fender: I Still get WARNING[11373]: chan_sip.c:4526 create_addr: No such host: 111 |
| 20:36.09 | kuku1 | when I have exten => 1,n,AddQueueMember(service_queue,Agent/111) or SIP/111 |
| 20:36.43 | [TK]D-Fender | kuku1: And I don't see you showing me that its legal |
| 20:37.47 | kuku1 | hm, i dont know what you mena but I'll check the docs |
| 20:38.07 | [TK]D-Fender | kuku1: How Do I know thats a valid SIP peer? |
| 20:39.09 | kuku1 | Example: AddQueueMember(techsupport,SIP/3000) |
| 20:39.21 | kuku1 | sip/111 is not valid |
| 20:39.29 | kuku1 | but in 1.2 it worked |
| 20:39.35 | [TK]D-Fender | kuku1: How Do I know thats a valid SIP peer? <---- |
| 20:39.37 | kuku1 | 111 is an agent |
| 20:39.58 | [TK]D-Fender | kuku1: Well you are adding a SIP PEER. Not an AGENT |
| 20:40.22 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 20:40.46 | kuku1 | exten => 1,n,AddQueueMember(service_queue,Agent/111) ? |
| 20:40.55 | kuku1 | isnt this an agent ? |
| 20:41.44 | [TK]D-Fender | kuku1: exten => 1,n,AddQueueMember(service_queue,SIP/121) <- Not an AGENT. Please wake up and look at what you're doing |
| 20:42.18 | kuku1 | correct, that worked in 1.2, it doesnt work in 1.6 ' |
| 20:42.21 | [TK]D-Fender | kuku1: And you are pasting only 1 little line, and not the entire output of your failed attempts |
| 20:42.32 | kuku1 | I also tried Agent/111 and Agent /121, and still didnt work |
| 20:42.45 | kuku1 | ok, one sec. |
| 20:42.45 | [TK]D-Fender | kuku1: And you aren't showing me the COMPLETE attempt |
| 20:42.55 | kuku1 | ill paste all |
| 20:44.11 | kuku1 | [TK]D-Fender: http://pastebin.ca/1474668 |
| 20:45.15 | [TK]D-Fender | kuku1: .... ALREADY THERE. Stop adding members to the queue every &#^$ing time you place a queue call! |
| 20:45.49 | [TK]D-Fender | [Jun 25 15:43:35] WARNING[12746]: app_queue.c:4636 aqm_exec: Unable to add interface 'SIP/110' to queue 'service_queue': Already there <-- Already F-ing logged in, stop loggin them in dammit! |
| 20:45.50 | kuku1 | ok, but it still doenst rignt 121 and 111 |
| 20:46.07 | kuku1 | 110 and 120 work fine im ok with those warnings |
| 20:46.10 | [TK]D-Fender | kuku1: And how do you THINK that "Agents" get calls? |
| 20:46.13 | kuku1 | 111 and 121 dont work |
| 20:46.22 | kuku1 | I dont know, it worked in 1.2 |
| 20:46.38 | kuku1 | I had exten => 111, dial(sip/xxx) |
| 20:46.45 | kuku1 | and it was working. |
| 20:46.49 | [TK]D-Fender | kuku1: the "agent would have to be LOGGED IN. The only way that happens here is AgentLogin() <- |
| 20:47.37 | [TK]D-Fender | kuku1: And yes that means that you agents would have to SIT in that call indefinitely |
| 20:48.16 | [TK]D-Fender | kuku1: 1.6 AddQueueMember does not point to DIALPLAN unless that is the DEVICE you specified |
| 20:49.23 | kuku1 | ok,im trying to understand where the problem is |
| 20:49.39 | [TK]D-Fender | kuku1: "Agent" = TOTALLY DEAD. Stop using them |
| 20:50.04 | kuku1 | ok |
| 20:50.26 | kuku1 | but i still dont understand how to do it correctly, I need it to call 2 cell phone numbers.... |
| 20:51.13 | [TK]D-Fender | kuku1: So do add a DEVICE that will ring them <- |
| 20:52.18 | [TK]D-Fender | kuku1: Right now you are failing to grasp how to use even just boring local devices |
| 20:52.44 | kuku1 | :) |
| 20:52.53 | kuku1 | add a dveice... im thinking... |
| 20:53.05 | kuku1 | but only devices i know are sip and zap |
| 20:53.17 | [TK]D-Fender | kuku1: Go read *'s list of channel-type a few dozen times. |
| 20:53.25 | kuku1 | ok |
| 20:57.46 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 21:04.59 | *** join/#asterisk mocker (n=user@shell.mocker.org) |
| 21:13.05 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 21:14.01 | j_kroon | is there any isdn guru's here? i'm pretty much at the point of giving up. |
| 21:15.50 | Nugget | hits j_kroon with a SPID |
| 21:18.29 | j_kroon | i've got a diva card, the second i start the capi stuff up the files under /dev/capi/ goes away ... stop it again and they're back. |
| 21:18.29 | j_kroon | and no matter what i do asterisk won't start when i enable chan_capi (presumably due to a misconfiguration somewhere). |
| 21:18.29 | j_kroon | would you know what /etc/capi.conf needs to look like for a Diva PRI card? |
| 21:23.03 | Katty | is anyone familiar with raids? and ntfs. |
| 21:23.25 | Katty | specifically, if i take an ntfs based drive, out of a raid, and connect it to an xp workstation... |
| 21:23.28 | Katty | can i see anything? |
| 21:23.41 | j_kroon | Katty, what raid levels? |
| 21:23.46 | Katty | i don't know. |
| 21:23.47 | ACK-NAK | Just to confirm: If you have HWEC it does not require a parameter in system.conf, ONLY requrement is 'echocancel = yes' in chan_dahdi.conf, is that correct? |
| 21:23.51 | Katty | not a mirror. |
| 21:23.57 | j_kroon | and no, the filesystem would be ntfs, which runs on top of a raid. |
| 21:24.00 | Katty | well, i can't confirm that either. |
| 21:24.06 | j_kroon | not a mirror? then no. probably not. |
| 21:24.28 | j_kroon | ACK-NAK, yes. but I highly recommend ripping that hwec out of there and using oslec. |
| 21:25.17 | ACK-NAK | j_kroon: Really? I have always been told that HWEC is the best solution, cost aside. |
| 21:25.21 | Katty | j_kroon: have you confirmed that? |
| 21:25.57 | j_kroon | Katty, raid1 == mirror. raid0 == stripe. raid4/5/6 == parity based redundancy. |
| 21:26.17 | j_kroon | the only option where you can take a single drive elsewhere and have it work is raid1. |
| 21:26.19 | Katty | that's not what i asked. |
| 21:26.24 | Katty | what i asked, is have you confirmed that answer. |
| 21:26.28 | Katty | or is it just what you think |
| 21:26.35 | errr | heh |
| 21:26.50 | j_kroon | i know raid about as well as anybody. read the second line i just posted. |
| 21:26.58 | j_kroon | and I _know_ that for a fact :) |
| 21:27.03 | ACK-NAK | Raid? http://www.baarf.com - Battle Against Any Raid Five |
| 21:27.05 | *** join/#asterisk mltlnx (n=mltlnx@cpe-24-193-11-14.nyc.res.rr.com) |
| 21:27.10 | Katty | k |
| 21:27.20 | ACK-NAK | Baarf.com is pretty funny. |
| 21:27.39 | j_kroon | ACK-NAK, hehe. it has it's place. but you need a VERY GOOD controller or you're looking for trouble. |
| 21:28.38 | ACK-NAK | j_kroon: the thing that got my attention was the potential for undetected corruption. Given the cost of storage these days, I find fewer and fewer cases for raid 5 over a straight mirror. |
| 21:29.06 | j_kroon | ACK-NAK, in HUGE arrays where the physical space is actually a problem :) |
| 21:30.12 | j_kroon | but yes, the undetected corruption is an issue ... but those same corruptions on a raid1 results in the same problem - which bit got corrupted? |
| 21:30.42 | ACK-NAK | j_kroon: Indeed. The darned definition of 'large' keeps changing. A decade ago we rand a 10gb Oracle database. They used to send us to the VLDB department Very large database. Now 10GB oracle instance is considered a toy. |
| 21:30.43 | j_kroon | and it can (And EMC CLARiiONs do) be detected with a periodic consistency scan. |
| 21:32.37 | ACK-NAK | I'm not an expert on the low level mechanics of the XOR parity calculation or spanning, so I can't render an personal opinion, but the point of the site is that there is a markedly higher risk of undetected corruption using raid5, all other considerations aside. |
| 21:33.42 | j_kroon | the risk is higher because your span is wider, not because of the parity. |
| 21:34.17 | j_kroon | anyway, i need help with this diva card. I'm out of ideas. |
| 21:34.38 | j_kroon | and google is being rather unhelpful tonight, or I'm simply not googling for the right stuff. |
| 21:35.15 | ACK-NAK | j_kroon: You're right. There's a good fit for raid5 it in lots of cases. |
| 21:35.37 | ACK-NAK | j_kroon: do you like AMCC or Adaptec or 'other' |
| 21:35.56 | j_kroon | adaptec. stay the &)(&*)&^(^%&^%*^(*#@^%$(&*@%$)*@^ away from anything with intel. |
| 21:36.11 | j_kroon | if you can get your hands on something dac960 you won't regret it. |
| 21:36.29 | ACK-NAK | does AMCC/3ware use an intel chip? |
| 21:36.55 | j_kroon | just make very, very, very sure that you've got a battery backup on your raid controller. most of them DON'T have write intent bitmaps so a single power failure can lose your array if you don't have battery backing. |
| 21:37.13 | j_kroon | not sure about the amcc thing. |
| 21:37.22 | ACK-NAK | Good advice. Not so with Adaptec? |
| 21:37.24 | *** join/#asterisk mayo (n=mayo@a221-smpafs01.blockb-142.stargate.ca) |
| 21:37.27 | mayo | hello |
| 21:37.40 | mayo | could anyone shed some slight what could be causing "got hangup request, cause 102" ? |
| 21:37.50 | j_kroon | adaptec have their own, and the dell fakes is in fact LSI (megaraid), not bad either. |
| 21:38.20 | ACK-NAK | j_kroon: thanks! |
| 21:38.44 | j_kroon | mayo, http://networking.ringofsaturn.com/RemoteAccess/isdncausecodes.php |
| 21:38.57 | j_kroon | Cause No. 102 - recovery on timer expiry |
| 21:39.04 | mayo | j_kroon: oh awesome, thanks! |
| 21:39.36 | j_kroon | i wish i knew about that stuff a year ago. |
| 21:40.28 | mayo | j_kroon: i get a crapload of incoming calls that die with cause 102 ... trying to figure out what the hell is going on |
| 21:40.49 | j_kroon | that description doesn't help much. never seen 102 myself, so good luck. |
| 21:41.24 | mayo | j_kroon: hehe yeah |
| 21:41.34 | mayo | j_kroon: thanks though .. |
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| 21:54.28 | exc_ess | Hey all, I'm running Asterisk 1.6.1 on Fedora 11. I'm trying to set up an extension which will call the Directory application. The extension works fine the first time you call it, but if you call a second time * crashes with a segfault in app_directory_plain.so |
| 21:54.39 | e0n` | what flavor of nix is asterisknow based off of? |
| 21:54.48 | exc_ess | I can't find any reference to this particular error anywhere on the net. |
| 21:56.09 | exc_ess | Anyone have any idea what I could do about this? |
| 21:56.23 | drmessano | Michael Jackson - dead |
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| 21:58.22 | e0n` | drmessano: where did you read that at? |
| 21:59.11 | drmessano | Where didn't I read it? |
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| 22:00.04 | Nugget | everywhere just says "rushed to hospital." TMZ is the only site I've seen that's bold enough to pronounce him dead. |
| 22:00.12 | e0n` | yeah |
| 22:00.14 | e0n` | my point exactly |
| 22:00.23 | e0n` | everywhere says "Hospital" no where says "dead" |
| 22:00.26 | e0n` | except tmz |
| 22:01.40 | *** join/#asterisk vegbox (n=kevinle@adsl-64-173-83-182.dsl.lsan03.pacbell.net) |
| 22:02.01 | vegbox | Hi, I got an old DOS voice mails system. Its giving out and I would like to replace it with a asterisk box. I have been doing some reading and it seems like i need a new computer with an FXO card and asterisk. Of course it needs to be configured but is that all I need? |
| 22:02.47 | Primer | yup |
| 22:03.06 | Primer | all it is is voicemail though? |
| 22:03.12 | Primer | not an actual PBX? |
| 22:04.47 | vegbox | just voicemail |
| 22:04.54 | vegbox | we still use NEC analog phone system |
| 22:05.12 | vegbox | right now we just dial ext. 250 and get to our voicemail system (from there enter pin etc) |
| 22:05.38 | vegbox | DOS = Replay Voice Mail |
| 22:05.39 | vegbox | Lol |
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| 22:10.22 | vegbox | Primer: ? |
| 22:10.32 | vegbox | No more sage advice lol? |
| 22:11.32 | Primer | Well, most people run asterisk as their PBX + voicemail |
| 22:12.11 | Primer | but, presuming your old voicemail system worked with an fxo card, I see no reason why asterisk can't perform the same task |
| 22:15.54 | beek | vegbox: I have my Iwatsu PBX using Asterisk as its voicemail system using 8 analog lines. |
| 22:16.13 | vegbox | Oh okay |
| 22:16.22 | vegbox | So you dial an ext to get voicemails? |
| 22:16.28 | beek | You will have to write the dialplan to make this work. |
| 22:17.22 | beek | Yes. The PBX has eight analog channels which dials eight channels on the * box. They're configured to terminate into a context that emulates the VM system that was previously attached. |
| 22:18.19 | vegbox | Man that sounds like a journey |
| 22:18.22 | vegbox | I hope I can figure it out |
| 22:18.30 | beek | It was only an evening's work. |
| 22:18.36 | vegbox | I mean ive done the Ip Phones and VoiceMail |
| 22:18.41 | vegbox | Never analog phones and voicemail |
| 22:18.42 | beek | Do you know anything about Asterisk and dialplans? |
| 22:18.53 | vegbox | Nope, any good reads? |
| 22:18.57 | beek | ~book |
| 22:18.57 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 22:19.35 | vegbox | Can you give me a summary of what dialplans are? |
| 22:20.21 | beek | Summary: the programming that makes Asterisk do something useful and not just sit there buring up power. |
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| 22:21.32 | beek | vegbox: Which Phone system/model are you running? |
| 22:21.53 | vegbox | NEC Electra |
| 22:22.44 | beek | vegbox: mind a PM? |
| 22:22.48 | vegbox | Sure |
| 22:22.50 | vegbox | I dont mind |
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| 22:32.15 | j_kroon | does anybody know what the D-Channel protocol in China (Shanghai) is? |
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| 22:33.04 | superbock | hi |
| 22:33.20 | superbock | Channel Location State Application(Data) |
| 22:33.20 | superbock | Zap/1-1 s@internal:1 Rsrvd (None) |
| 22:33.31 | superbock | does anyone know why this happens? |
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| 22:36.48 | j_kroon | superbock, i _suspect_ it's something to do with malfunctioning hangup detection. |
| 22:36.52 | j_kroon | or a missing Hangup() |
| 22:37.08 | j_kroon | i've seen it too, and then i fixed some other stuff and it went away ... so I'm not sure. |
| 22:37.17 | j_kroon | Please select NT-2 (Direct Dial In, Direct Inward Dialing) mode <-- WTF is NT-2 ? |
| 22:39.21 | KavanS | shit |
| 22:39.24 | KavanS | michael jackson died |
| 22:39.29 | superbock | j_kroon: this is very occasional, so I doubt it's a missing hangup. I've researched about malfunctioning hangup detection but I'm not sure how that applies to an FXS port (I only see this happen with FXS, not FXO's). Problem is that sometimes this requires a reboot, other times dahdi and asterisk restart is enough |
| 22:39.30 | KavanS | heh now I feel old... |
| 22:40.15 | j_kroon | superbock, i only had it in a specific case, call comes in off DAHDI/ ... goes to a SIP phone which calls back out again over another DAHDI/ channel. The SIP side never hung up, and neither did the DAHDI channels. |
| 22:40.56 | j_kroon | As long as the call was actually up though the outbound DAHDI had Dial() as the app, and as soon as the call actually got hung up this changed to (None). |
| 22:41.11 | j_kroon | but BOTH the in and out DAHDI channels got "locked" up. |
| 22:41.59 | superbock | so a show channels would show both of them in rsrvd and none? |
| 22:43.09 | superbock | I've also seen a report about missed interrupts, increasing latency up to 30ms, and then it blocks the port. But I don't see latency increasing to that level |
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| 22:46.41 | superbock | this post in the forum pretty much describes my problem: http://forums.digium.com/viewtopic.php?t=67724&sid=f3e327022b41d7aff319661732e8b313 |
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| 22:58.00 | j_kroon | superbock, i've seen weird things whenever i get that lost interrupts issue in dmesg (one-way voice on DAHDI/ ... sudden jitter on SIP/ channels etc ...), setting latency=7 solves this for me (it normally increases up to 4) |
| 23:03.44 | dacs | anyone having issues with zoiper working with * |
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| 23:17.40 | dacs | can someone help me please with an issue i have in my dialplan |
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| 23:19.21 | dacs | http://pastebin.ca/1474824 |
| 23:19.52 | beek | So, what is the problem? |
| 23:21.16 | Aiatek | dacs whats your problem? |
| 23:22.14 | dacs | i think i found it out , but let me ask your expertise |
| 23:28.15 | dacs | Aiatek: when i dial extension 102 , i get invalid |
| 23:28.32 | dacs | is it because i have my i exten above it ?! |
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| 23:49.17 | lanning | there is no order to the extensions. what context is the incoming channel dropping into? |
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