| 00:00.23 | mvanbaak | JT: I really dont get this: free to <some places> and paid to <other places> |
| 00:00.29 | mvanbaak | we dont have that in .eu |
| 00:00.50 | mvanbaak | we get a line, and that's to $world |
| 00:01.14 | mvanbaak | only exception on that is internal ISP traffic |
| 00:01.16 | JT | mvanbaak: try living on an island 13000+km away from other big Internet centres :) |
| 00:01.30 | mvanbaak | like, my apt-get upgrade stuff goes at 10gb |
| 00:01.45 | mvanbaak | because bit hosts a debian mirror |
| 00:01.50 | JT | you have a 10Gbit/s ethernet link? |
| 00:01.55 | mvanbaak | yup |
| 00:02.01 | mvanbaak | well, fiber |
| 00:02.04 | JT | nice, termination hardware costs too much though |
| 00:02.08 | JT | err |
| 00:02.12 | mvanbaak | nah |
| 00:02.14 | mvanbaak | we rent it |
| 00:02.17 | JT | is it ethernet or not? |
| 00:02.24 | JT | ethernet can run on fibre or copper |
| 00:02.28 | mvanbaak | it's fiber to our cisco |
| 00:02.36 | JT | is it ethernet? |
| 00:02.42 | mvanbaak | yup |
| 00:02.46 | JT | or SDH |
| 00:02.48 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 00:03.04 | mvanbaak | ethernet |
| 00:03.25 | JT | ok |
| 00:03.35 | mvanbaak | our cisco gear has vlans to their juniper |
| 00:03.40 | mvanbaak | from there it's SDH |
| 00:03.50 | mvanbaak | SDH to SARA and AMS-IX |
| 00:04.03 | *** join/#asterisk egecko (n=sam@cpe-76-176-238-225.san.res.rr.com) |
| 00:04.20 | JT | from a consumer or co-lo customer standpoint, it's the same cost for domestic and international, unless it's domestic peered |
| 00:04.23 | pkunkra | JT, alright. i'm going to buy a clamp meter to solve this power problem of mine. i'm googling them now. do you have any brand/model recommendations? |
| 00:04.24 | JT | here |
| 00:04.40 | JT | but for providers with overseas links, it's the overseas connections that cost more |
| 00:04.41 | mvanbaak | and they have 16colors 10ge SDH stuff from their juniper to ams-ix and de-ix |
| 00:04.55 | JT | wavelengths? |
| 00:05.14 | mvanbaak | JT: no idea, that's their business |
| 00:05.23 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:05.37 | mvanbaak | we just get a 10ge connection to their core network and 100mbit to AMS-IX |
| 00:05.58 | JT | i think that's what you mean |
| 00:06.36 | mvanbaak | wb seanbright |
| 00:06.47 | mvanbaak | hows the cdr-redux going ? |
| 00:06.53 | seanbright | great to be back mvanbaak |
| 00:07.06 | seanbright | eh |
| 00:07.09 | seanbright | slowly :) |
| 00:07.17 | mvanbaak | I hear ya |
| 00:07.21 | seanbright | paying work has been demanding most of my attention |
| 00:07.28 | mvanbaak | I've been working on the new multiparking stuff |
| 00:07.33 | *** join/#asterisk equanimity (n=alex@cpc2-oxfd7-0-0-cust208.oxfd.cable.ntl.com) |
| 00:07.35 | riddlebox | has anyone configured sla.conf? |
| 00:07.41 | seanbright | yeah. i've been following your commits |
| 00:07.47 | mvanbaak | cool |
| 00:08.06 | mvanbaak | I got a great lecture about AST_LIST stuff tonight |
| 00:08.23 | seanbright | yeah? |
| 00:08.29 | mvanbaak | putnopvut made me understand that stuff |
| 00:08.37 | seanbright | awesome |
| 00:08.40 | mvanbaak | uhhuh |
| 00:08.58 | seanbright | i'm still relatively an asterisk code virgin |
| 00:09.06 | seanbright | but i hope to get lucky soon |
| 00:09.07 | mvanbaak | before tonight I was dependant on oej to tell me stuff |
| 00:09.07 | seanbright | ;) |
| 00:09.15 | mvanbaak | but now I'm able to do it on my own |
| 00:09.52 | *** join/#asterisk rfernandez (n=rfernand@201.137.54.18) |
| 00:10.10 | seanbright | well that's awesome |
| 00:11.43 | mvanbaak | uhhuh |
| 00:12.12 | *** join/#asterisk Ironhand (i=x@xyx.nl) |
| 00:12.15 | rfernandez | hi! i have a customer that have a openvox a1200p and want 12 fxo extra trunks, if i connect another pci card i will have troubles of irq? |
| 00:13.00 | putnopvut | mvanbaak: you're quite welcome. Always glad to teach people to use the coding tools Asterisk provides |
| 00:13.41 | JT | rfernandez: why so many fxo? |
| 00:13.41 | mvanbaak | rfernandez: depends |
| 00:13.58 | mvanbaak | if the slots does not share IRQ/DMA you're fine |
| 00:15.05 | rfernandez | mvanbaak, how can i check this? |
| 00:15.32 | rfernandez | JT, cause the customer doesnt want a E1 =S wants analog lines cause its a pbx shared with 4 enterprises (6 trunks per enterprise) |
| 00:15.36 | mvanbaak | dont recall |
| 00:15.54 | rfernandez | oh sorry |
| 00:15.58 | mvanbaak | lspci or something |
| 00:16.17 | mvanbaak | I gave up on linux like a year ago |
| 00:16.39 | rfernandez | ahh ok |
| 00:17.08 | mvanbaak | it's all bsd here now |
| 00:17.16 | rfernandez | jejeje i see |
| 00:17.57 | *** part/#asterisk josephp (n=none@shell4.sea5.speakeasy.net) |
| 00:19.39 | JT | rfernandez: lol |
| 00:19.47 | JT | the customer is silly |
| 00:19.49 | JT | but oh well |
| 00:19.57 | rfernandez | yes |
| 00:20.06 | rfernandez | but well a customer its a customer isnt it? |
| 00:20.10 | rfernandez | ;) |
| 00:21.21 | rfernandez | so in the experience of you it dont represent any trouble isnt it? |
| 00:23.03 | riddlebox | has anyone tried this, I cannot get it to work http://www.asterisk.org/node/48342 |
| 00:25.54 | JT | rfernandez: but by the same customer, not every customer is a customer worth having :) |
| 00:28.51 | rfernandez | jejeje youre right |
| 00:29.13 | *** join/#asterisk cesar_CR (n=cesar@celord.ice.co.cr) |
| 00:29.54 | [hC] | argh |
| 00:30.02 | ruied | I would like to display the billing information with cdr table plus a column with the price per call, I'm recording cdr in postgresql, is there any tool for this? a2billing, is it a good solution? is there any other methods to do this? |
| 00:30.03 | [hC] | why am I getting double dtmf on this aastra |
| 00:30.51 | cesar_CR | [hC] |
| 00:31.01 | [hC] | hey man.. do you still need an iax example? |
| 00:32.02 | cesar_CR | well I need an explanation of the errors that I'am getting |
| 00:32.08 | [hC] | I can try! |
| 00:32.14 | rfernandez | jc but! if the customers can sign checks we can paint the asterisk in pink if he wants it! |
| 00:33.27 | cesar_CR | in the * that is making the call a get this : WARNING[2994]: chan_iax2.c:7542 socket_process: Call rejected by 200.91.75.45: No authority found |
| 00:33.49 | cesar_CR | and in the destinations * this : NOTICE[4599]: chan_iax2.c:7761 socket_process: Host 200.91.94.122 failed to authenticate as celord |
| 00:35.42 | [hC] | so there is a problem in your iax.conf or your dial string |
| 00:35.47 | [hC] | do you know about the pastebin? www.pastebin.ca |
| 00:35.57 | cesar_CR | yes.. |
| 00:36.00 | [hC] | paste your iax.conf from both machines in there, and your dial command |
| 00:36.02 | [hC] | then show it to me |
| 00:36.06 | [hC] | ill tell you where the problem is |
| 00:36.12 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 00:36.12 | cesar_CR | ok |
| 00:36.46 | *** join/#asterisk ZenBSDi (n=zenbsdi@unaffiliated/ZenBSDi) |
| 00:37.54 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-fe6fd91cab979807) |
| 00:39.31 | *** join/#asterisk SteveTotaro (n=root@c-69-243-124-5.hsd1.md.comcast.net) |
| 00:47.03 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) [NETSPLIT VICTIM] |
| 00:50.27 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 00:51.52 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 00:57.20 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 00:57.56 | grandpapadot | Hi all. What's the best set of sound files for asterisk 1.2? Will the 1.4 sound files work? And is there an archive with sln formatted files? I didn't see any on digium's website. |
| 01:03.50 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 01:08.09 | ruied | I'm compiling asterisk-addons and it reports the following error: http://paste.uni.cc/18411 what could be wrong? |
| 01:08.49 | ruied | I'm compiling it to have moh reading .mp3 files |
| 01:09.09 | *** join/#asterisk stansmith (n=stansmit@d60-65-93-136.col.wideopenwest.com) |
| 01:09.43 | stansmith | what forum should i post in if i found something that would be useful to other people |
| 01:09.48 | stansmith | on the digium boards |
| 01:10.21 | denon | you could tell us right here |
| 01:10.28 | stansmith | uhm |
| 01:11.08 | stansmith | well, i figured out a work around to do something with Swift(), the cepstral plug in |
| 01:11.25 | denon | dumping to wav first to save channels? |
| 01:11.34 | stansmith | no? but thats crazy |
| 01:11.37 | stansmith | i didnt know that |
| 01:11.50 | SteveTotaro | www.voip-info.org |
| 01:11.58 | SteveTotaro | or the digium forums |
| 01:12.04 | stansmith | SteveTotaro, yea but what forum |
| 01:12.07 | stansmith | not support, right? |
| 01:12.19 | stansmith | i mean, i dont want to look like a complete noob |
| 01:12.20 | SteveTotaro | no, i have never been there, just saw it |
| 01:12.29 | SteveTotaro | not sure what forums exist |
| 01:12.42 | SteveTotaro | you should start a blog on your site |
| 01:12.45 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:13.02 | stansmith | do you have a blog? |
| 01:13.04 | SteveTotaro | you could be the next asterisk daily news or nerdvittles |
| 01:13.19 | SteveTotaro | nah, i don't have time for blogs |
| 01:13.25 | SteveTotaro | just irc and email |
| 01:13.54 | SteveTotaro | plus, i charge for what i do |
| 01:14.08 | SteveTotaro | so putting up howtos doesn't help biz |
| 01:14.19 | denon | actually, it does |
| 01:14.28 | denon | putting up howtos gives you some credibility |
| 01:14.31 | SteveTotaro | limited howtos |
| 01:14.33 | denon | gets your name out there |
| 01:14.39 | SteveTotaro | i have credibility |
| 01:14.45 | denon | nah, odds are people aren't going to hire you for those specific things |
| 01:14.57 | denon | it's not like a dozen howtos are going to cover it |
| 01:15.04 | SteveTotaro | spoke at astricon for large call center deployments |
| 01:15.13 | denon | and if your customers were the type to read those things, there's already way more info on the web than you'd ever cover |
| 01:15.24 | SteveTotaro | that gets you more cred, trade shows |
| 01:16.03 | denon | look at open source .. do you give digium any less credit just because they published all their secrets? |
| 01:16.41 | stansmith | http://www.pastebin.ca/922809 |
| 01:16.58 | SteveTotaro | they have plenty of secrets |
| 01:17.02 | stansmith | its a work around to get dynamic TTS with asterisk, while performing a Read() at the same time |
| 01:17.07 | SteveTotaro | let's be real here |
| 01:17.15 | stansmith | is that clever or am i just a foolish rookie? |
| 01:17.37 | SteveTotaro | i think that may already be on the wiki |
| 01:17.44 | stansmith | damn! |
| 01:17.50 | SteveTotaro | check, maybe not |
| 01:17.55 | stansmith | i thought ive seen everything there is online with app_swift |
| 01:18.14 | *** part/#asterisk stubert (n=stu@techtools.actusa.net) |
| 01:18.15 | SteveTotaro | yeah, i wrote an app_swift article there years ago |
| 01:19.04 | stansmith | :-/ |
| 01:19.21 | stansmith | that was the first thing i felt like i figured out myself without having to see a HOWTO |
| 01:19.54 | *** join/#asterisk MaliutaWrk (i=nikolai@119.11.101.90) |
| 01:20.01 | drmessano | I started my own wiki since most of the other wikis suck |
| 01:20.16 | stansmith | drmessano, address? |
| 01:20.23 | SteveTotaro | ZT_CHANCONFIG failed on channel 1: Inappropriate ioctl for device (25) |
| 01:20.27 | SteveTotaro | WTF!!!! |
| 01:20.32 | SteveTotaro | Damn BRI crap |
| 01:20.49 | stansmith | !chapter 14 |
| 01:20.53 | stansmith | ~chapter 14 |
| 01:21.09 | *** join/#asterisk BeeBuu (n=beebuu@219.132.190.199) |
| 01:21.50 | SteveTotaro | are you talking to me about chapter 14? |
| 01:22.21 | stansmith | no i read chapter 14 today, it is inspiring to say the least |
| 01:26.26 | stansmith | ~asterisk daily news |
| 01:26.27 | jbot | it has been said that daily asterisk news is http://www.sineapps.com/news.php for html and http://www.sineapps.com/rssfeed.php for rss |
| 01:26.32 | stansmith | kthx |
| 01:27.05 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 01:29.03 | stansmith | is there a common way to get "#" as the variable "my_var" in "Read(my_var)"? |
| 01:30.27 | drmessano | BRI = Barely Readable Interface |
| 01:31.32 | SteveTotaro | bri fixed!!!! |
| 01:31.37 | SteveTotaro | asterisk sux!!!! |
| 01:31.41 | SteveTotaro | i rule!!!! |
| 01:31.55 | SteveTotaro | channel 1,2 = no good |
| 01:32.04 | SteveTotaro | channel 1-2 = joy |
| 01:32.30 | drmessano | channel 50-50 <-- Chance of BRI working |
| 01:32.32 | drmessano | lol |
| 01:32.36 | SteveTotaro | < [ 00 a1 01 05 ] |
| 01:32.36 | SteveTotaro | 1 |
| 01:32.36 | SteveTotaro | < Supervisory frame: |
| 01:32.36 | SteveTotaro | 1 < SAPI: 00 C/R: 0 EA: 0 |
| 01:32.36 | SteveTotaro | < TEI: 080 EA: 1 |
| 01:32.37 | SteveTotaro | 1 < Zero: 0 S: 0 01: 1 [ RR (receive ready) ] |
| 01:32.39 | SteveTotaro | < N(R): 002 P/F: 1 |
| 01:32.41 | SteveTotaro | < 0 bytes of data |
| 01:32.43 | SteveTotaro | 1 -- ACKing all packets from 1 to (but not including) 2 |
| 01:32.45 | SteveTotaro | 1 -- Since there was nothing left, stopping T200 counter |
| 01:32.47 | SteveTotaro | 1 -- Stopping T203 counter since we got an ACK |
| 01:32.49 | SteveTotaro | 1 -- Nothing left, starting T203 counter |
| 01:32.51 | SteveTotaro | 1 -- Got RR response to our frame |
| 01:32.53 | SteveTotaro | 1 -- Restarting T203 counter |
| 01:33.00 | SteveTotaro | sorry for the flood, to excited .... |
| 01:33.04 | drmessano | heh |
| 01:33.16 | drmessano | We'll overlook it THIS TIME... |
| 01:33.24 | drmessano | But we're watching you |
| 01:33.39 | SteveTotaro | damn man, i spent ALL day at the customer site, system down |
| 01:33.46 | drmessano | damn |
| 01:34.09 | SteveTotaro | i still don't know if I can get the new Sangoma 4 port FXO to work with a bri card |
| 01:34.18 | SteveTotaro | so much patched zatpel crap |
| 01:34.59 | JT | which bri card |
| 01:36.04 | SteveTotaro | Junghanns quad port |
| 01:36.10 | SteveTotaro | but Sangoma sells their own |
| 01:36.28 | JT | yes, they're quite different |
| 01:36.30 | SteveTotaro | and they have one file that does all their devices |
| 01:36.33 | JT | and blogs do help business imho |
| 01:36.53 | SteveTotaro | blogs do, but possibly not detailed howtos |
| 01:37.20 | SteveTotaro | a blog on ROI and enough implementation details to prove you have the skillz |
| 01:37.52 | JT | howtos is giving back to the community, and also help with business |
| 01:37.55 | JT | win win really |
| 01:37.59 | SteveTotaro | who makes a sip based four port fxo |
| 01:38.19 | stansmith | yea i mean, i would buy a product if there company had mad howtos |
| 01:38.22 | SteveTotaro | i am going to be releasing some GPL stuff in a week or so |
| 01:38.23 | JT | audiocodes, mediatrix, vegastream and patton at least i think |
| 01:38.26 | stansmith | thats why i buy intel-based laptops |
| 01:39.03 | SteveTotaro | because you are an intel fanboy |
| 01:39.07 | SteveTotaro | ;) |
| 01:39.12 | SteveTotaro | AMD baby!!! |
| 01:39.28 | stansmith | why shouldnt i be? intel fosters native linux wifi drivers |
| 01:39.51 | drmessano | AMD pwns |
| 01:39.54 | SteveTotaro | really, my ip2200 was a bitch |
| 01:40.06 | stansmith | ipw3945 "just works" with ubuntu |
| 01:40.15 | stansmith | it wasnt hard at all in suse10 |
| 01:40.27 | SteveTotaro | probably because some guy in his underwear reverse engineered the drivers |
| 01:40.30 | stansmith | i thought ip2200 was in the kernel |
| 01:40.42 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 01:40.43 | SteveTotaro | it is now but took long enough |
| 01:40.49 | stansmith | ha |
| 01:40.52 | stansmith | ~bsd |
| 01:40.53 | jbot | BSD is a UNIX operating system. An asterisk port is currently availible if you feel you must, or a way to set your pc back 30 years, progress is overrated |
| 01:40.54 | SteveTotaro | i have had this laptop for three years |
| 01:41.09 | drmessano | ~epoch |
| 01:41.10 | jbot | 1 Jan 1970 - The expiration date on side of the carton of Unix |
| 01:41.16 | stansmith | ~2038 |
| 01:41.45 | SteveTotaro | for a company selling services, it makes no sense to blog a complete howto |
| 01:41.49 | *** join/#asterisk JunK-Y (n=junky@modemcable153.55-201-24.mc.videotron.ca) |
| 01:41.54 | SteveTotaro | it does for a company selling hardware though |
| 01:42.03 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:42.05 | stansmith | or a company selling software + services |
| 01:42.24 | SteveTotaro | what software? opensource? |
| 01:42.36 | SteveTotaro | i will follow your blog and never call you, thanks |
| 01:42.44 | stansmith | sugarcrm |
| 01:42.57 | SteveTotaro | ~pos |
| 01:42.58 | jbot | See ibmpos |
| 01:42.58 | stansmith | ~sugarcrm |
| 01:43.03 | stansmith | ~ibmpos |
| 01:43.04 | jbot | [ibmpos] IBM Piece Of S#$t or a stinking piece of antique technology made to drive programmers insane |
| 01:43.04 | JT | SteveTotaro: only if you want to think about the world so one-tracked |
| 01:43.12 | JT | blogs can help service businesses too |
| 01:43.13 | stansmith | oic |
| 01:43.41 | JT | just because you tell someone how to do one specific thing, doesn't mean they want to do it themselves, or that they might not need you anyway |
| 01:43.42 | SteveTotaro | that is what having a business and accounting degree do to a guy |
| 01:44.04 | SteveTotaro | no, they will hire a guy in pakistan to do it via ssh |
| 01:44.14 | SteveTotaro | for $5/hr |
| 01:44.25 | stansmith | pwn |
| 01:44.47 | drmessano | I still laugh over the guy that wanted me to ship a PBX to india and have me support it |
| 01:44.49 | JT | not everyone is like that |
| 01:44.55 | drmessano | Total role reversal |
| 01:44.59 | stansmith | haha |
| 01:45.17 | drmessano | Im dead serious.. and he completely didn't get it |
| 01:45.22 | SteveTotaro | with the freefall of the dollar this role reversal is going to become reality |
| 01:45.33 | stansmith | :-/ |
| 01:45.40 | SteveTotaro | it's good for us |
| 01:45.43 | stansmith | o |
| 01:45.44 | stansmith | :-D |
| 01:46.04 | SteveTotaro | as long as prices for staples don't go up |
| 01:46.06 | BeeBuu | hello,all |
| 01:46.09 | stansmith | ~shark |
| 01:46.10 | jbot | There are NO sharks in #Asterisk. We want to be perfectly clear on that. |
| 01:46.24 | drmessano | ~sike |
| 01:46.25 | jbot | [Sike!] The preceeding statement was intented to be humourous, and may have involved false statements used in a humorous manner, or an exaggeration of a real life situation with humourous intent, but in no way should be taken as pure fact or not considered questionable at best. |
| 01:47.04 | BeeBuu | is there any USB GSM adapter for asterisk? |
| 01:47.19 | SteveTotaro | bluetooth |
| 01:47.24 | SteveTotaro | and chan_mobile |
| 01:47.28 | drmessano | Yep |
| 01:47.43 | BeeBuu | bluetooth? |
| 01:47.51 | SteveTotaro | well i put my first customer on 1.4 |
| 01:47.56 | SteveTotaro | we will see how it goes |
| 01:48.05 | drmessano | GSM adapters are old and busted, Bluetooth + chan_mobile + a cell phone is the new hotness |
| 01:48.11 | SteveTotaro | small pbx, nothing fancy |
| 01:48.17 | BeeBuu | i need to make over 100 people to meet ,any suggestion? |
| 01:48.31 | SteveTotaro | app_meetme |
| 01:48.40 | drmessano | BeeBuu: You've ignored all the suggestions given so far |
| 01:48.49 | BeeBuu | SteveTotaro: i mead hardware~~~ |
| 01:48.52 | SteveTotaro | or advertise free booze (open bar) |
| 01:49.06 | BeeBuu | :-) |
| 01:49.41 | SteveTotaro | you could buy a proprietary system with a conference bridge and pay ~$50-$75k |
| 01:49.46 | drmessano | Dual Quad Core 3.4GHZ Xeons and 8GB Ram should do it |
| 01:50.08 | *** join/#asterisk Wayhigh (i=noid@www.kevinlynn.com) |
| 01:50.10 | BeeBuu | drmessano: only one system? |
| 01:50.12 | SteveTotaro | opteron would definitely work better |
| 01:50.17 | Wayhigh | sup steve? |
| 01:50.22 | SteveTotaro | hey man |
| 01:50.41 | drmessano | If you got 100 lines, sure |
| 01:50.41 | SteveTotaro | hell of a day but finally fixed my issue, customer was down all day with me onsite |
| 01:50.48 | BeeBuu | 8GB RAM～～ |
| 01:51.16 | BeeBuu | how come this 8 GB RAM? |
| 01:51.27 | SteveTotaro | he is exaggerating |
| 01:51.50 | SteveTotaro | but the more, the better |
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| 01:51.50 | *** join/#asterisk sbingner (n=john@pdpc/supporter/sustaining/sbingner) [NETSPLIT VICTIM] |
| 01:51.50 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) [NETSPLIT VICTIM] |
| 01:51.50 | *** join/#asterisk BugBlue (n=mendel@clamans.mobach.nl) [NETSPLIT VICTIM] |
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| 01:51.51 | *** join/#asterisk JT (n=j@unaffiliated/jt) [NETSPLIT VICTIM] |
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| 01:51.51 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) [NETSPLIT VICTIM] |
| 01:51.51 | *** join/#asterisk Uni (n=mpartin@smtp.academicsuperstore.com) [NETSPLIT VICTIM] |
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| 01:51.51 | *** join/#asterisk mace (n=mace@debian/developer/mace) [NETSPLIT VICTIM] |
| 01:51.51 | *** join/#asterisk nixguy (n=matmoj@fw.packetfront.com) [NETSPLIT VICTIM] |
| 01:51.51 | *** join/#asterisk iCEBrkr (n=icebrkr@cyberdyne.org) [NETSPLIT VICTIM] |
| 01:52.21 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 01:52.22 | drmessano | Ok, 4GB Ram |
| 01:52.23 | drmessano | Ram is cheap, who cares |
| 01:52.31 | SteveTotaro | if you are ever configuring multiple zap channels do not do :channels=1,2, |
| 01:52.38 | SteveTotaro | do channels 1-2 |
| 01:52.48 | BeeBuu | SteveTotaro: why not group? |
| 01:53.02 | SteveTotaro | i am a lone wolf |
| 01:53.10 | SteveTotaro | not a pack animal |
| 01:53.37 | BeeBuu | :-P |
| 01:53.55 | SteveTotaro | channels=1,2 should work |
| 01:54.07 | SteveTotaro | i have seen it in many samples |
| 01:54.10 | BeeBuu | i got astribank when i searched, is it good? |
| 01:54.21 | SteveTotaro | ask tzafrir |
| 01:54.50 | SteveTotaro | xorcom is a good company from what i gather, never used their stuff except their version of bristuff |
| 01:55.18 | BeeBuu | is there any product with GSM? |
| 01:55.39 | SteveTotaro | do you want 100 gsm connections to a conference? |
| 01:55.49 | SteveTotaro | get a BTS and a cell tower |
| 01:56.07 | BeeBuu | SteveTotaro: yes,that's what i want~~~! |
| 01:56.19 | SteveTotaro | www.altobridge.com |
| 01:56.32 | BeeBuu | checking... |
| 01:56.42 | SteveTotaro | http://www.altobridge.com/emergency1.html |
| 01:56.49 | SteveTotaro | i carry one in my trunk |
| 01:57.38 | BeeBuu | fireman! |
| 01:58.16 | BeeBuu | my mission just like that.... |
| 01:58.24 | SteveTotaro | no, blackhat cell phone operator |
| 01:58.39 | SteveTotaro | please enter your bank account info |
| 01:58.46 | SteveTotaro | please enter your pin |
| 01:58.54 | SteveTotaro | app_monitor |
| 02:00.27 | *** join/#asterisk g0mb0 (n=root@external.micom.mng.net) |
| 02:01.00 | SteveTotaro | seriously though, what is your mission? |
| 02:01.22 | SteveTotaro | i have a good relationship with altobridge |
| 02:01.44 | SteveTotaro | do you have a budget or is this just an idea |
| 02:02.09 | *** join/#asterisk jbigbee (n=jbigbee@24.96.154.69) |
| 02:03.55 | BeeBuu | i still don't understand what's the altobridge offer.. |
| 02:04.14 | TJNII | Anyone played with a SNOM 220? Any good? |
| 02:05.01 | *** join/#asterisk egecko (n=sam@cpe-76-176-238-225.san.res.rr.com) |
| 02:05.09 | *** join/#asterisk PepOSX (n=angeldav@190.72.149.163) |
| 02:05.53 | TJNII | It runs a PPC chip with a 2.4 linux kernel, so I'm pretty confident even if it proves unusable out of the box I can hack it to do .... something. |
| 02:07.39 | grandpapadot | Is there a way to force calls from a queue to ring it's members the entire time there is someone in the queue without the queue pause/restart? |
| 02:08.12 | *** join/#asterisk nebulON (n=nebulon@200-161-125-209.dsl.telesp.net.br) |
| 02:08.30 | *** join/#asterisk marc7 (n=marc@64.46.14.64.novuscom.net) |
| 02:08.52 | TJNII | grandpapadot: Queue pause/restart? |
| 02:09.33 | grandpapadot | ilbc rocks |
| 02:10.58 | SteveTotaro | the offer is a cellphone BTS tower and VSAT |
| 02:11.07 | SteveTotaro | in a suitcase |
| 02:11.14 | grandpapadot | TJNII: It seems that it will ring a member for 10-15 seconds, then pause a while, then restart. |
| 02:11.17 | SteveTotaro | it's all right there in english |
| 02:11.24 | *** join/#asterisk TrXuk (n=trx@cpc5-flee1-0-0-cust532.glfd.cable.ntl.com) |
| 02:12.04 | TrXuk | Yo... I need to give people from my DMZ sip access to my internal PBX... I can see sip_contrack_nat for Iptables, but it looks like that was last updated in 2004 |
| 02:12.17 | TJNII | grandpapadot: Check the retry setting in queues.conf |
| 02:12.26 | grandpapadot | TJNII: Thanks. |
| 02:12.37 | TrXuk | Is this built into iptables by default now? can someone please point me to some docs of how it's done nowadays?? |
| 02:12.38 | TrXuk | cheers |
| 02:15.07 | SteveTotaro | 2 !! Unknown IE 42 (cs5, Unknown Information Element) |
| 02:15.26 | SteveTotaro | not sure how euroisdn works on a US BRI but it does |
| 02:15.42 | SteveTotaro | besides funky errors about unknown IEs |
| 02:16.47 | SteveTotaro | what do you need to do with iptables? |
| 02:17.58 | TrXuk | I want to allow sip traffic in / out of my network |
| 02:18.09 | TrXuk | without opening up a f*cktonne of UDP ports |
| 02:18.17 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:18.18 | TrXuk | like SIP connection tracking o |
| 02:19.01 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 02:20.35 | *** join/#asterisk L|NUX (i=L_NUX@unaffiliated/lnux/x-10290) |
| 02:20.36 | drmessano | I still don't understand why it matters how many ports you open |
| 02:20.49 | *** join/#asterisk MaliutaWrk (i=nikolai@119.11.100.39) |
| 02:21.29 | JT | it doesn't |
| 02:21.31 | drmessano | If the SAME app has a listener on 2 ports or 2000 ports, it's the same app... If it a matter of vulnerability, 1 port is as bad as 10000 |
| 02:21.34 | JT | some people are just insane |
| 02:21.39 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 02:22.24 | drmessano | Then theres the argument of "Well, I dont want them to know what I have listening, and less ports makes it less likely |
| 02:22.41 | SteveTotaro | just move 5060 |
| 02:22.45 | drmessano | Ok, instead of scanning 10000-20000, I will just scan 10000-10001.. ZOMG PWNED ME |
| 02:23.47 | JT | TrXuk: just open the required port range, and forget about sip aware stuff |
| 02:23.51 | JT | that usually breaks sip |
| 02:24.00 | drmessano | Just ask Cisco |
| 02:24.10 | obnauticus | lol |
| 02:24.27 | drmessano | PIX's work sooo well with SIP |
| 02:24.35 | TJNII | "A firewall is like a virgin: once it ain't, it ain't." |
| 02:24.39 | drmessano | lol |
| 02:24.48 | *** join/#asterisk Robba (n=rob@mail.ampwest.com.au) |
| 02:25.01 | obnauticus | drmessano i use pf and i get no problems |
| 02:25.04 | obnauticus | freebsd with pf |
| 02:25.14 | drmessano | Probably not |
| 02:25.15 | Robba | Has anyone here setup an asterisk box to use a Cisco 2851 as a sip gateway? |
| 02:25.21 | drmessano | That's still not a PIX |
| 02:25.31 | obnauticus | PIX firewalls are meh |
| 02:25.41 | drmessano | OpenWRT passes sip better |
| 02:25.47 | drmessano | Hell |
| 02:25.48 | obnauticus | than freebsd with pf? |
| 02:25.50 | obnauticus | uhh |
| 02:25.52 | obnauticus | ok lol. |
| 02:25.54 | drmessano | WRT54G STOCK passes it better |
| 02:25.59 | nebulON | burn it |
| 02:26.54 | drmessano | IMHO, a PIX is only good as a branch office client that VPNs into a enterprise Cisco solution |
| 02:27.24 | drmessano | Buying one as a stand alone box is effing teh dumb |
| 02:30.46 | grandpapadot | PIX sucks, dated. The ASA is much better. |
| 02:31.23 | TJNII | Does anyone know of an affordable ATA that supports rotaries? |
| 02:31.49 | TJNII | I have 2 rotaries, and while the idea of making an adapter does pique my interest, I don't have the time. |
| 02:32.37 | *** join/#asterisk linux_galore (n=Richard@dsl-203-33-161-39.NSW.netspace.net.au) |
| 02:32.38 | grandpapadot | lol, wtf are you still doing with rotary phones? |
| 02:32.51 | JT | TJNII: zap cards work with decadic dialling |
| 02:32.53 | TJNII | Retro, man |
| 02:32.58 | grandpapadot | lol |
| 02:33.13 | TJNII | House was madin in the 50s, kitchen is canary yellow. The rotary wall phone just fits. |
| 02:33.21 | *** part/#asterisk linux_galore (n=Richard@dsl-203-33-161-39.NSW.netspace.net.au) |
| 02:33.23 | grandpapadot | Someone needs to make a SIP bat phone, old style. Bright red, no buttons, just pickup and it dials a preset number... |
| 02:33.31 | drmessano | They do |
| 02:33.42 | drmessano | I found one somewhere |
| 02:33.47 | grandpapadot | drmessano: eh? Really? I'll bu one where? |
| 02:33.50 | TJNII | telephonedepot iirc |
| 02:35.01 | TJNII | No, not telephonydepot. W(here)TH did I find that? |
| 02:35.39 | grandpapadot | http://www.redhotphones.com/inbybareddep.html |
| 02:37.04 | TJNII | My PAP2s support dialing an extension when picked up, that + a bat phone and you've got it. |
| 02:37.10 | obnauticus | damn assholes... |
| 02:37.18 | obnauticus | open source labs keeps outbidding me on an ebay auction |
| 02:37.18 | obnauticus | >: | |
| 02:38.25 | grandpapadot | What item? |
| 02:38.32 | obnauticus | >:| |
| 02:38.35 | obnauticus | a serial console switch |
| 02:38.44 | grandpapadot | lol, I'm buying one RIGHT NOW, no shit! |
| 02:38.48 | obnauticus | ... |
| 02:38.51 | obnauticus | what brand |
| 02:39.17 | grandpapadot | no, the batphone .... |
| 02:39.19 | obnauticus | ... |
| 02:39.25 | pigpen | anyone know when 1.4.19 will be released? (rumors, etc..) I need the page app update. |
| 02:39.25 | obnauticus | this guy has bought like 20 of these things |
| 02:40.17 | drmessano | The PAP2 solution is the way to go |
| 02:41.00 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 02:41.42 | grandpapadot | I've like 10 of those things in a box... They would be useful if they did g729a on both ports at the same time. |
| 02:41.47 | *** join/#asterisk chendy (n=chendy@121.35.144.22) |
| 02:41.52 | grandpapadot | Or ilbc |
| 02:41.56 | grandpapadot | my new favorite codec |
| 02:42.07 | obnauticus | speex is good i ehar |
| 02:42.16 | obnauticus | i still like ulaw |
| 02:42.38 | drmessano | IPKALL has iLBC |
| 02:42.49 | *** join/#asterisk kiscokid (n=ron@38.104.140.82) |
| 02:42.50 | drmessano | which is teh cool |
| 02:42.57 | grandpapadot | ipkall? |
| 02:43.20 | drmessano | www.ipkall.com |
| 02:44.12 | grandpapadot | drmessano: That your biz? |
| 02:44.19 | JT | obnauticus: serial console, like a terminal server? |
| 02:44.22 | drmessano | No |
| 02:44.41 | obnauticus | yes |
| 02:44.42 | kiscokid | Anyone know how to trace the signalling when a call gets picked up on the other end using Zap devices? |
| 02:44.47 | obnauticus | JT like a cyclades. |
| 02:44.55 | JT | yeah exactly |
| 02:44.59 | JT | that's what i meant |
| 02:45.07 | JT | serial terminal server |
| 02:45.08 | obnauticus | ya |
| 02:45.18 | drmessano | I think Gizmo does iLBC too |
| 02:45.19 | obnauticus | I still don't know why i want one |
| 02:45.21 | sbingner | JT, lightwave SCS |
| 02:45.24 | obnauticus | I just want one |
| 02:45.24 | obnauticus | lol |
| 02:45.25 | grandpapadot | How does IPKall make money? |
| 02:45.37 | JT | you'll hate me but, the other day i pickets up 3 * 1RU Perle 24 port units |
| 02:45.41 | JT | AUD$1 each |
| 02:45.44 | sbingner | they are good :p |
| 02:45.48 | JT | s/pickets/picked/ |
| 02:45.52 | obnauticus | JT can i have one? |
| 02:45.59 | obnauticus | what brand are they |
| 02:46.03 | JT | obnauticus: not sure :P |
| 02:46.04 | JT | Perle |
| 02:46.22 | JT | you can configure any serial port to have a modem attached |
| 02:46.26 | obnauticus | errrr... |
| 02:46.33 | JT | so you can dial in for out of band access too all of the rest of the ports |
| 02:46.35 | obnauticus | I think the one im looking at can do that too |
| 02:46.38 | drmessano | http://lists.digium.com/pipermail/asterisk-biz/2005-July/007081.html |
| 02:46.47 | JT | the ports can connect to anything with a serial management port |
| 02:46.50 | drmessano | Thats the answer to the IPKALL question |
| 02:46.53 | obnauticus | JT ya |
| 02:46.56 | obnauticus | most can |
| 02:47.12 | riddlebox | who do you guys suggest for a cheap sip provider, I want to port my cell number, just to keep the number |
| 02:47.29 | kiscokid | voicepulse |
| 02:47.36 | obnauticus | JT I still prefer cyclades ones though |
| 02:47.54 | JT | the perle ones seem pretty nice |
| 02:48.02 | obnauticus | in comparison to cyclades? |
| 02:48.25 | grandpapadot | Ahh.. interesting and creative... Kudos to ipkall |
| 02:48.37 | JT | in general, haven't looked specifically at cylades external units |
| 02:48.45 | JT | but the perles do ssh out of the box |
| 02:48.50 | JT | and radius and ldap |
| 02:48.52 | obnauticus | ya |
| 02:48.53 | obnauticus | this one too |
| 02:49.02 | obnauticus | it has more ... like ridiculous auth methods |
| 02:49.24 | drmessano | IPKALL works well too |
| 02:49.31 | obnauticus | SSH v2, RADIUS, ip filtering... |
| 02:49.52 | obnauticus | secure in band and out of band network managment |
| 02:49.55 | JT | yep |
| 02:50.07 | obnauticus | SH v1 and v2; Local, RADIUS, TACACS+ and LDAP Authentication; Token-based strong authentication (SecurID); Local backup user authentication support; PAP/CHAP authentication; IP Packet and Security Filtering ; User Access Lists per port; System Event Syslog |
| 02:50.10 | obnauticus | SSH* |
| 02:50.10 | JT | it seems slanted towards remote admin |
| 02:50.23 | JT | but i see no reason why it can't be used to terminate a bunch of dialup modems |
| 02:50.26 | JT | the perle |
| 02:50.34 | obnauticus | lol |
| 02:50.39 | obnauticus | hook my cell phone into it |
| 02:50.45 | JT | i guess they don't market things that way anymore? |
| 02:50.52 | obnauticus | i dunno |
| 02:51.06 | obnauticus | most people just go out and buy stuff that comes with LOM packages |
| 02:51.06 | JT | since people moved to access servers |
| 02:51.09 | obnauticus | like iLO and DRAC |
| 02:51.11 | JT | then broadband |
| 02:51.12 | obnauticus | with ipmi |
| 02:51.18 | obnauticus | that have SoE integrated... |
| 02:51.24 | obnauticus | broadband is through err |
| 02:51.24 | drmessano | DRAC.. ewww |
| 02:51.31 | obnauticus | DRAC is pretty good |
| 02:51.34 | obnauticus | i prefer iLO though |
| 02:51.41 | drmessano | iLO is much better |
| 02:51.41 | *** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 02:51.43 | drmessano | Well |
| 02:51.52 | *** part/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 02:51.55 | drmessano | Drac > II is better |
| 02:52.04 | drmessano | Drac II suxored |
| 02:52.13 | JT | i'm thinking of getting a phone line in the datacentre, and a gsm modem or a gsm phone with serial cable |
| 02:52.16 | obnauticus | A termy server should be better than both |
| 02:52.19 | JT | for redundant OOB management :) |
| 02:52.36 | obnauticus | JT im making it so that i can call my * boxen that can send commands to other servers |
| 02:52.41 | obnauticus | via serial |
| 02:52.44 | obnauticus | through the PSTN |
| 02:53.24 | x86 | obnauticus: dial-up? |
| 02:53.30 | obnauticus | no |
| 02:53.33 | obnauticus | menu systems |
| 02:53.48 | x86 | how are you going through the PSTN with serial? |
| 02:53.52 | obnauticus | AGI scripts |
| 02:54.06 | x86 | ok so you're not going through the PSTN with serial... |
| 02:54.09 | obnauticus | PSTN -> Asterisk -> AGI scripts that send commands over ethernet.... |
| 02:54.10 | x86 | gotcha |
| 02:54.10 | obnauticus | no |
| 02:54.11 | obnauticus | ya |
| 02:54.13 | obnauticus | you got it |
| 02:54.36 | x86 | your previous statements were misleading :) |
| 02:54.42 | obnauticus | stream SNMP stats to a log which can be spoken through Festival |
| 02:54.46 | obnauticus | on certian hosts. |
| 02:54.52 | obnauticus | or SNMP trap events. |
| 02:55.21 | obnauticus | the funny and ironic part is, if my asterisk box goes down, i have no redundancy on it |
| 02:55.23 | obnauticus | im fixning it |
| 02:55.24 | obnauticus | lol |
| 02:55.56 | obnauticus | Anyone here have an article on * redundancy? |
| 02:56.15 | x86 | google does |
| 02:56.19 | obnauticus | a good one* |
| 02:56.35 | x86 | ask lmadsen |
| 02:56.54 | x86 | he does that professionally ;) |
| 03:01.02 | *** part/#asterisk beek (n=klinebl@static-71-240-222-16.alt.east.verizon.net) |
| 03:04.08 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 03:04.25 | *** join/#asterisk sailense (n=sailense@202.70.125.71) |
| 03:04.51 | sailense | hey, does anyone know a good VoIP termination carrier for the US? |
| 03:05.02 | obnauticus | for IAX, voipjet i hear is good |
| 03:05.17 | sailense | how about net2phone or latinode? |
| 03:05.18 | obnauticus | but i got 20 bucks from them and they still haven't given me my MD5 string for auth after like 5 months and 20 emails. |
| 03:05.46 | *** join/#asterisk jameswf-home (n=james@199.72.100.49) |
| 03:06.51 | sbingner | obnauticus, uh just log in and click "PBX Setup" -- it's listed there |
| 03:08.25 | obnauticus | ya |
| 03:08.26 | obnauticus | it doesn't show up |
| 03:08.27 | obnauticus | lol |
| 03:08.33 | sbingner | sailense, I've been having good results with voip.ms |
| 03:08.40 | obnauticus | i emailed their `fast support' email 20 times |
| 03:08.43 | obnauticus | and never got a reply |
| 03:08.56 | sailense | @sbingner: thnx, i'll check them out |
| 03:08.57 | obnauticus | from 3 different emails |
| 03:09.23 | sbingner | sailense, connect.voicepulse.com has been reliable too ;) |
| 03:09.37 | sailense | awesome, thanks for the suggestions |
| 03:09.44 | obnauticus | http://img442.imageshack.us/img442/7730/seeon0.png <-- |
| 03:09.46 | obnauticus | that's what happens to me |
| 03:10.02 | obnauticus | it's epic homosexual. |
| 03:11.13 | eric2 | dumb question... but how do I get my own context? from within the dial plan |
| 03:11.24 | eric2 | or my own username? |
| 03:11.34 | obnauticus | you mean |
| 03:11.45 | obnauticus | if you have a user (peer) defined in sip.conf |
| 03:11.55 | obnauticus | how do i put him in a certian context in the dialplan (extensions.conf) ? |
| 03:12.45 | eric2 | n, when you're in a macro, you can put ${MACRO_CONTEXT} or ${MACRO_EXTEN) |
| 03:13.19 | eric2 | but if you're just in the extensions.conf file, I need to get the username or context that my phone belongs to |
| 03:13.38 | obnauticus | dunno lol |
| 03:13.40 | *** join/#asterisk Navion (n=billp@208-106-107-201.dsl-s.california.com) |
| 03:14.23 | Navion | Does anyone know if there is support for echo canceller disable for modems in the zaptel drivers? |
| 03:14.48 | stansmith | MatBoy? |
| 03:16.31 | Navion | Back in the bad old days when there were analog telephones, and I worked for a phone company, the echo cancellers would detect modem answer tone and disable so they wouldn't interfere with the modems. |
| 03:17.08 | stansmith | how long ago was that |
| 03:17.39 | Navion | Hmmm... 1980's |
| 03:18.40 | Navion | We had the sme thing on TDM implimented echo cancellers for bandwidth managers too. |
| 03:18.56 | *** join/#asterisk jpeeler (n=jpeeler@adsl-065-005-230-151.sip.lft.bellsouth.net) |
| 03:20.18 | Navion | I actually worked in a CO once that was dial pulse only (step by step) |
| 03:20.19 | sbingner | eric2, you can get the accountcode -- that's what it's for |
| 03:20.37 | sbingner | the context is just where it starts out... the phone doesn't 'belong' to that context... it's just IN that context |
| 03:21.07 | [TK]D-Fender | Navion, Yes, thats automatic for modems & FAXs |
| 03:22.02 | Navion | <sigh> OK, then it just doesn't work for my Tivo's modem |
| 03:22.23 | Navion | I noticed that Linksys has a "Modem passthrough mode" code [*99]. |
| 03:24.10 | Navion | It also has a FAX passthrough mode [#99] why it's a different mode I have no idea and Linksys won't tell mere mortals what their configuration parameters mean (or do). |
| 03:25.04 | TJNII | I should get my modem working through my ATA |
| 03:25.10 | TJNII | For the sheer pointlessness of it |
| 03:26.15 | Navion | Three is method to the madness of making the DirecTivo work over a SIP connection... |
| 03:28.40 | sbingner | there is also madness to the method |
| 03:28.54 | *** join/#asterisk Sweet-P (n=me@ip68-111-191-246.sd.sd.cox.net) |
| 03:29.12 | Sweet-P | ok, i've got asterisk and freepbx to help out...inbound calling works perfect, i just can't dial out |
| 03:29.18 | Navion | I'm always open to alternative methods |
| 03:29.31 | Sweet-P | i think I have my dial rules wrong |
| 03:29.46 | Navion | 9|. |
| 03:29.50 | sbingner | Navion, does it work if you disable all the echo cancellation manually? |
| 03:30.00 | Sweet-P | what is "9|." |
| 03:30.41 | Sweet-P | ahh...wait |
| 03:30.44 | *** join/#asterisk thrills3k3r (n=root@66-193-179-217.static.twtelecom.net) |
| 03:30.44 | Navion | Haven't tried that yet. The idea of leaving it disabled after I paid all that money to get it would hurt my sensibilities. |
| 03:30.55 | sbingner | Navion, it's a troubleshooting step... |
| 03:31.06 | Navion | True. |
| 03:31.06 | stansmith | lo? |
| 03:31.35 | Navion | Sweet-P> do you have outgoing trunks defined? |
| 03:31.55 | Sweet-P | i do, I think I'm getting confused between dial rules and dial plans..reading now |
| 03:32.52 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-28.dllstx.fios.verizon.net) |
| 03:32.53 | Navion | You need to have "Dial Patterns" in your "Outbound Route" |
| 03:33.04 | Sweet-P | http://freepbx.org/support/documentation/howtos/howto-route-dial-patterns-and-trunk-dial-rules |
| 03:33.05 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 03:33.11 | Sweet-P | RTFM I guess! |
| 03:33.20 | Navion | You can use the wizard to do a simple one. |
| 03:33.21 | [TK]D-Fender | ~freepbx |
| 03:33.21 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 03:33.23 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 03:33.33 | Sweet-P | thanks |
| 03:34.26 | Navion | Then assign a trunk in the "Trunk Sequence" and it should work. |
| 03:35.41 | stansmith | k |
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| 03:36.06 | thrills3k3r | is anyone responsible for asterisk.org on at the moment? |
| 03:39.18 | TJNII | I can now call any phone in the house from any other phone in the house. So cool. So pointless. |
| 03:39.33 | cmantito | I had an analogue PBX in my house for that |
| 03:39.36 | cmantito | everyone hated it |
| 03:39.37 | cmantito | I loved it :P |
| 03:41.18 | Navion | TJNII> Learning to embrace your pointlessness is a big step. |
| 03:41.41 | TJNII | It wouldn't be so pointless if I had more than one housemate..... |
| 03:41.55 | Navion | At least you can write a dial plan so no one has to dial 9 now. |
| 03:42.11 | TJNII | Heh. I make them dial 9. |
| 03:42.29 | cmantito | My old PBX made everyone dial *1 |
| 03:42.56 | TJNII | 10 digit dialing mandatory? |
| 03:43.09 | Navion | Have your housemate move around between rooms. Then you can write a really cool Follw Me. |
| 03:43.23 | cmantito | *1 switched you to an outside line at which point you could dial anything the local telco would let you |
| 03:43.32 | TJNII | Aah. |
| 03:43.39 | cmantito | in other words, no, not necessarily |
| 03:43.47 | cmantito | it was acceptable to do *1NXXXXXX |
| 03:44.08 | Navion | I had a Panasonic 308. It worked fine so long as you wanted to do everything the way Panasonic wanted you to. |
| 03:44.36 | Navion | It did work with the DirecTivo's too. :( |
| 03:44.58 | cmantito | http://tbn0.google.com/images?q=tbn:1TBfCVISLSf0DM:http://secure.businessphonesdirect.com/bmz_cache/d/dc0bc19c3a9af06dafd8fe6190c434a0.image.150x150.jpg <-- the phones for mine |
| 03:45.34 | cmantito | it demanded *thsoe* phones |
| 03:45.43 | cmantito | required 4-wires too |
| 03:46.25 | Navion | The 308 would let you use their 4 wire phones or regular loop start 2w phones. |
| 03:46.26 | cmantito | I still have it and about 30 phones here somewhere |
| 03:47.37 | Navion | After the first EMP I'll switch back to the Setp-by-Step switch. |
| 03:47.48 | cmantito | heh |
| 03:47.52 | Navion | Step-by-Step that is... |
| 03:48.06 | cmantito | figured |
| 03:48.21 | tengulre | hi,all |
| 03:48.25 | cmantito | lo |
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| 03:49.46 | thrills3k3r | anyone recommend a middle east VoIP DID provider? |
| 03:50.00 | stansmith | thats deep |
| 03:50.22 | WilliamK | wholesale or retail? |
| 03:50.41 | thrills3k3r | either I guess |
| 03:50.41 | plik | hah! unbelievable: http://cgi.ebay.co.uk/VOIP-Complete-stand-alone-PBX-Software-CD-Download_W0QQitemZ270215124152QQihZ017QQcategoryZ11908QQcmdZViewItemQQ_trksidZp1742.m153.l1262 |
| 03:51.08 | WilliamK | figures someone would do that |
| 03:51.17 | cmantito | thrills3k3r: what CC? |
| 03:51.36 | pigpen | too funny. |
| 03:51.48 | thrills3k3r | cmantito: a few actually, one sec |
| 03:51.51 | stansmith | some people dont know how to burn iso |
| 03:51.55 | cmantito | thrills3k3r: kk |
| 03:51.59 | drmessano | Old news |
| 03:52.41 | cmantito | thrills3k3r: /msg me the country codes |
| 03:53.21 | stansmith | o? |
| 03:53.23 | plik | stansmith: that dot download requires that you fo (and provide your own blank CD) |
| 03:53.32 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-148-108.satx.res.rr.com) |
| 03:53.55 | stansmith | huh?! |
| 03:54.05 | plik | bleh pesky laggy connection |
| 03:54.12 | thrills3k3r | cmantito: +90 (Turkey), +971 (UAE), +964 (Iraq) |
| 03:54.21 | stansmith | Quantity:50 available |
| 03:54.23 | pigpen | funny enough he also offers "Mozilla Firefox Ebay Edition" |
| 03:54.25 | cmantito | one mo please |
| 03:54.26 | stansmith | thats whats funny ^ |
| 03:54.30 | thrills3k3r | np |
| 03:54.39 | Sweet-P | wow |
| 03:55.01 | pigpen | Man, that has to violate something, somewhere. |
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| 03:55.09 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:55.21 | stansmith | lol@feedback |
| 03:55.23 | Sweet-P | k, got all of my stuff right...when I call inbound from my cellular, everything is good, when I call from my softphone, my outbound audio just echos back to me and doesn't reach the distant end |
| 03:55.34 | Sweet-P | i think it's my firewall |
| 03:55.52 | cmantito | thrills3k3r: sorry, I got none |
| 03:56.15 | thrills3k3r | cmantito: no problem, thanks for checking |
| 03:56.18 | cmantito | np |
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| 04:22.52 | Sweet-P | alright, I have a weird setup question |
| 04:23.20 | Sweet-P | when you use the localnet param...it's supposed to have your local networks, right? |
| 04:23.42 | Sweet-P | what happens if I want to register my sip phone at my house, behind the nat, from another country at a hotspot? |
| 04:23.53 | Sweet-P | is there any special configs that would prevent me from registering? |
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| 04:29.05 | LemensTS | Hello. How much am I looking at a server that is just going to handle 200 concurrent calls, G729 no transcoding? |
| 04:29.06 | *** join/#asterisk Corydon76-vcch (i=purple@pdpc/supporter/bronze/Corydon76-home) |
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| 04:33.26 | lmadsen | LemensTS: price out a quad-core Intel system with 4GB of RAM |
| 04:33.38 | lmadsen | that is basically $1500 or so |
| 04:33.46 | lmadsen | so not much |
| 04:34.06 | lmadsen | Corydon76-vcch: you! |
| 04:34.41 | LemensTS | lmadsen: Really? That sounds good! Would that be dell poweredge server? |
| 04:34.42 | lmadsen | ya... I built one, got a switch and some other things for $1500 with taxes, etc... |
| 04:34.58 | lmadsen | LemensTS: I don't know... I built an AMD quad-core system with 4GB of ram for about $1000 |
| 04:35.14 | WilliamK | you get the q6600 or the q6700? |
| 04:35.20 | lmadsen | but yes, I like the dell poweredge 2950s |
| 04:35.39 | lmadsen | WilliamK: I didn't get a dell, I just built it out of off-the-shelf parts |
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| 04:36.03 | lmadsen | my friend runs a consulting business and has access to all the warehouses |
| 04:36.08 | lmadsen | anyways, bedtime for me |
| 04:36.30 | LemensTS | How much GHz ya think? |
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| 04:39.48 | LemensTS | How do you guys build those servers that cheap? |
| 04:40.40 | LemensTS | A quad core xeon 3.16ghz with 4gb ram in a poweredge 2950 is $6000 |
| 04:42.37 | outtolunc | do a comparision at newegg |
| 04:42.48 | WilliamK | pricegrabber, pricewatch, etc... |
| 04:42.57 | WilliamK | use the shopping search engines |
| 04:43.09 | WilliamK | and for the select few that can buy wholesale, that's another venue :) |
| 04:43.19 | Corydon76-dig | People think that since Dell makes some cheap desktops that all their machines are the cheapest. Those cheap machines are loss leaders. |
| 04:43.31 | outtolunc | i stopped building systems years ago, because people could buy direct for as cheap or cheaper than i could build |
| 04:43.33 | JT | LemensTS: the easy way to make them cheap is to remove the Dell badge |
| 04:43.39 | Corydon76-dig | and they're far from the only ones doing that trick |
| 04:43.39 | JT | LemensTS: saves a few thousand dollars |
| 04:43.58 | Corydon76-dig | The only way to be sure is to comparison shop exactly what you want |
| 04:44.19 | WilliamK | often the best way to find cheap things is to buy mb/cpu/memory seperate from the rest of the order |
| 04:44.25 | Corydon76-dig | Don't accept someone's specs. Make your own and compare what they'll offer for what you need |
| 04:44.33 | WilliamK | some will lower prices on items to get your other biz at a higher margin |
| 04:45.44 | WilliamK | also buying the newest chip on the market isn't smart shopping |
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| 04:45.58 | WilliamK | that chip usually cost 700+ more than the one right below it |
| 04:46.03 | WilliamK | for a 150mhz less |
| 04:46.29 | WilliamK | most people won't notice a 150mhz diff nowdays |
| 04:46.33 | WilliamK | if it was 600mhz, definitely |
| 04:47.08 | WilliamK | if you're a pro gamer going for 25k at tournaments or something, then spend away |
| 04:47.30 | WilliamK | but most of those people are putting their $ into badass video |
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| 04:48.17 | outtolunc | and 1ms monitors <G> |
| 04:48.51 | outtolunc | i got my ... handed to me the other day by a guy that i used to kick his butt... he got a new system <G> |
| 04:49.28 | WilliamK | reason? because that 1ms-8ms allows him to see you before you walk around a corner in counter-strike |
| 04:49.37 | WilliamK | before you see him |
| 04:49.39 | outtolunc | nods |
| 04:49.52 | outtolunc | he's got 3 shots in your head before you can blink |
| 04:50.01 | kiscokid | My Polycom phones start a timer when the other end of the call gets connected but when I call some 800 numbers that doesn't happen |
| 04:50.06 | WilliamK | we used to do that crap to lil punks when I worked at another ISP and we were on the local lan |
| 04:50.09 | WilliamK | :) |
| 04:50.38 | kiscokid | there must be some signalling where the other end tells * that they are off hook |
| 04:51.23 | kiscokid | when I call a certain 800 number it never starts the timer and we hang up after 1 minute |
| 04:51.35 | WilliamK | outtolunc, that's also why I like having 20/20 FiOS here at home |
| 04:51.41 | WilliamK | though I'm not much of a gamer anymore |
| 04:51.47 | WilliamK | need to put the servers back up one day |
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| 04:52.03 | *** mode/#asterisk [+o russellb] by ChanServ |
| 04:52.07 | WilliamK | think VZ go bananas if I do thou |
| 04:52.21 | Sweet-P | how do I configure asterisk if I want to register my sip phone to my asterisk behind a nat, from another country at a hotspot? |
| 04:52.25 | WilliamK | although I could drop it on a few net connections in the datacenter |
| 04:53.11 | jameswf-home | Sweet-P: vpn |
| 04:53.19 | Sweet-P | hmmm...damn |
| 04:53.24 | Sweet-P | thanks though |
| 04:53.36 | WilliamK | or open a tunnel through the NAT |
| 04:53.47 | WilliamK | and config asterisk with the externip= in sip.conf |
| 04:54.09 | kiscokid | isn't there some stuff at voip-info that talks about NAT? |
| 04:54.13 | Sweet-P | i have my ports forwarded on my NAT, |
| 04:54.19 | WilliamK | yep and it's in the sample docs |
| 04:54.22 | Sweet-P | i have externhost set to my host |
| 04:54.50 | Sweet-P | i've done all of that, so if that's all i have to do, then I've missed something somewhere else |
| 04:55.12 | drmessano | Ports open? |
| 04:55.22 | WilliamK | which ports did you open and forward? |
| 04:55.33 | Sweet-P | yup, 5060-5160, 2070, 3478, 10000-20000 |
| 04:55.38 | Sweet-P | just inc ase |
| 04:55.50 | WilliamK | are you behind NAT at the hotspot? |
| 04:55.56 | drmessano | UDP? |
| 04:56.01 | Sweet-P | all ports udp |
| 04:56.18 | Sweet-P | WK: I won't actually know until I get there |
| 04:56.29 | WilliamK | it might work, and might not if you are |
| 04:56.32 | Sweet-P | but I'm guessing I'll see both setups |
| 04:56.36 | WilliamK | you headed to china? |
| 04:56.47 | Sweet-P | i live in usa, moving to italy |
| 04:56.52 | Sweet-P | will be traveling europe |
| 04:56.52 | WilliamK | ah |
| 04:57.10 | Sweet-P | i bought a linksys wip330 so I can use voip on the road |
| 04:57.31 | Sweet-P | but I also want to use it at my house...without changing the config everytime I leave the house |
| 04:57.41 | WilliamK | worst case, get a sprint evdo card w/ 3G router and prepaid intl roaming |
| 04:57.46 | WilliamK | think it's like 100/mo |
| 04:58.16 | Sweet-P | hmmm, it's an option, but I'd really like to just carry my wip330 and that's it |
| 04:58.45 | WilliamK | I'm getting a vehicle adapter for the 3G Router |
| 04:58.50 | WilliamK | which will make it nice |
| 04:59.01 | WilliamK | james, which edge service? |
| 04:59.03 | WilliamK | tmobile? |
| 04:59.10 | jameswf-home | AT&T |
| 04:59.24 | WilliamK | got a new HSPDA card? |
| 04:59.31 | jameswf-home | Iyeah.. |
| 04:59.38 | jameswf-home | maybe |
| 04:59.39 | jameswf-home | what |
| 04:59.53 | WilliamK | what model of a aircard do you have? |
| 05:00.03 | jameswf-home | we are in downtown charleston 15 hotels no good wireless bahh |
| 05:00.17 | jameswf-home | sierra 875u |
| 05:00.29 | russellb | jameswf-home: charleston <3 |
| 05:00.50 | jameswf-home | I actually ate dinner with jared tonight |
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| 05:01.00 | WilliamK | ya it is |
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| 05:01.11 | WilliamK | you might wanna check and see what mode it's operating in |
| 05:01.37 | russellb | jameswf-home: jared smith? |
| 05:01.42 | jameswf-home | yeah |
| 05:01.53 | russellb | oh really ... so what's going on in charleston? |
| 05:02.00 | WilliamK | Wireless data at speeds up to 3.6 Mbps on HSDPA capable networks |
| 05:02.07 | jameswf-home | a group of us went to an overpriced restraunt lol |
| 05:02.08 | russellb | and why am i not there? |
| 05:02.21 | jameswf-home | Freepbx training deal |
| 05:02.26 | WilliamK | russellb, slacking again? |
| 05:02.27 | WilliamK | =) |
| 05:02.29 | jameswf-home | you hate freepbx |
| 05:02.34 | jameswf-home | :) |
| 05:02.42 | russellb | jared went to that? heh, interesting. |
| 05:02.53 | russellb | i don't hate freepbx, i hate trixbox |
| 05:02.57 | jameswf-home | were here for pr not the training |
| 05:03.13 | [hC] | jameswf-home: did you see evans's easy button thing? |
| 05:03.17 | russellb | well damn, i wish i would have known ... that's where i'm from, where all my family is, etc. ... |
| 05:03.18 | jameswf-home | russellb: will you be at astricon in phoenix... |
| 05:03.18 | [hC] | er, not easy button.. magic button. |
| 05:03.31 | russellb | always looking for an excuse to go |
| 05:03.34 | WilliamK | james, by the way the newer model cards are 880/881 which do up to 7.2Mbps down |
| 05:03.35 | russellb | jameswf-home: yup |
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| 05:03.43 | russellb | jameswf-home: where'd you eat? |
| 05:04.13 | jameswf-home | Coast |
| 05:04.17 | [hC] | jameswf-home: i'm trying to get a copy of the response point magic button clone he did with lumenvox, but he has only announced he'd give it away at the conference so far, and i couldnt make it in time. |
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| 05:05.46 | jameswf-home | well russellb maybe we can figure ou lunch when your in phoenix... that way you can see I 'm not evil :) |
| 05:06.11 | russellb | heh, i never thought you were evil |
| 05:06.24 | jameswf-home | damn I have failed :) |
| 05:06.56 | russellb | maybe misguided ? :p |
| 05:07.51 | jameswf-home | I even had a half hour talk to day with mark from sangoma I think people were in awe... |
| 05:09.02 | russellb | what's your position btw ? |
| 05:09.20 | russellb | i know who you work for, just not what you do .. |
| 05:10.22 | jameswf-home | I would say I am the same as jared... no offcial titl but I am the community interface with a touch of development and incidental sales |
| 05:10.38 | jameswf-home | I am the official company spell checker :) |
| 05:10.50 | russellb | fair enough. |
| 05:17.18 | drmessano | I wish I could have gone to the training |
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| 06:52.13 | yang | Could someone advise me the issue about zaptel channels...In my case I have only defined channels 4&5 in zapata.conf, does it mean that if the call is coming to the first channel that I cannot hear it (this is what happens) - here is the config http://openpaste.org/en/5310/ |
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| 08:35.38 | nixguy | how does FreePBX differ from asterisk? |
| 08:37.22 | marc7 | nixguy: the power of google! - http://www.voipuser.org/forum_topic_12076.html |
| 08:37.45 | nixguy | you know how ppl on forums are! |
| 08:37.55 | nixguy | if you want pure untainted knowlede you come to #asterisk |
| 08:38.05 | nixguy | but ok ok ill drink your "muddy water" |
| 08:38.07 | nixguy | thnx :D |
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| 08:38.53 | marc7 | for every silly forum question, there's a good forum answer archived somewhere in the internets. sometimes google, sometimes gmane.org, sometimes marc.theaimsgroup.com |
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| 09:12.57 | ^shark_ | hi guys how do i force the 7912g ip phone to boot firmware, a good tutorial might be helpful |
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| 09:33.15 | marc7 | ^shark: have you found the config file compiler yet? |
| 09:33.50 | *** join/#asterisk Hyphenex (n=Hyphenex@60-241-73-120.tpgi.com.au) |
| 09:33.57 | Hyphenex | What's the best IAX2 client for windows? |
| 09:34.05 | Hyphenex | ... that *crosses fingers* works behind NATs |
| 09:34.07 | BeeBuu | how can i invite someone to my conference by dial his phone? |
| 09:34.38 | BeeBuu | meetme() command? |
| 09:36.00 | marc7 | Hyphenex: have you looked at http://www.voip-info.org/wiki-Asterisk+IAX+clients and http://www.voip-info.org/wiki/view/VOIP+Phones (Soft Phones, halfway down the page) |
| 09:36.45 | Hyphenex | marc7: yeah, they don't rate them between say 1 and 10 though :P |
| 09:38.05 | marc7 | i couldn't comment on "best" as I don't use windows... but I've heard "things" about ZoIPer |
| 09:39.02 | sempuki | anyone know a simple tutorial where I can find out how to set up a simple conference using App_Conference on Asterisk and at least two soft phones? |
| 09:39.14 | marc7 | not really wanting to stand behind it myself yet, but give it a shot and come back during the day when more people are alive ;-) |
| 09:40.01 | madduck | why would asterisk send 183 and 180 in response to an INVITE, but then not send a ring back tone? |
| 09:40.48 | Hyphenex | marc7: good "thinks" |
| 09:40.49 | Hyphenex | things** |
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| 09:42.06 | sempuki | there seems to be lots about meetme, but none for App_conference |
| 09:42.11 | marc7 | Hyphenex: the only two softphones I really have tons of experience with are X-Lite and ZoIPer, but that's just because I'm never too far from a Cisco or Polycom phone that's been flashed to support SIP |
| 09:42.25 | sempuki | ekiga is a very nice softphone |
| 09:42.38 | marc7 | ^shark_: if you're still around, you should /msg me regarding the 7912's. they're a royal pain in the ass to setup, and I can point you in the direction of files that would be of use, as I just setup ten of them a week ago |
| 09:43.26 | Hyphenex | marc7: SIP does not work with X-Lite on other peoples computers |
| 09:43.27 | marc7 | sempuki: still waiting on the OSX build for ekiga ;) |
| 09:43.35 | Hyphenex | it appears to be a NAT issues as far as I can tell |
| 09:43.40 | marc7 | Hyphenex: ah, yes. I meant to say the "special" X-Lite build |
| 09:44.07 | badcfe | tests in dialplan for equality is done with = , not with == right? |
| 09:44.15 | Hyphenex | the "special" X-Lite build? |
| 09:44.22 | Hyphenex | marc7: I want badly |
| 09:44.24 | marc7 | Hyphenex: yes, there's a build floating around with SIP support. |
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| 09:44.51 | Hyphenex | marc7: you mean IAX2? |
| 09:45.44 | Hyphenex | ... I can't find it |
| 09:46.28 | marc7 | Hyphenex: well, *I* meant SIP, you were originally talking about IAX2 though. |
| 09:46.39 | Hyphenex | IAX2 > SIP? |
| 09:47.22 | sempuki | so one one really uses app_conference? |
| 09:47.58 | marc7 | sempuki: I should be working on a project now that either uses app_conference or app_meetme, but haven't quite dug into it yet |
| 09:48.46 | sempuki | Im pretty sure Ive done something small and stupid to make my conference fail to work... |
| 09:48.56 | sempuki | both soft phones dial in, and connect |
| 09:49.05 | marc7 | Hyphenex: IAX2 can work well with NAT networks and has a jitter buffer, but SIP is more widely adopted |
| 09:49.18 | sempuki | and conference show stats says theres one conference |
| 09:49.27 | sempuki | but I dont actually hear anyone's voice |
| 09:49.53 | Hyphenex | marc7: well, I use SIP internally fine, it's just that when people try registering in from outside, we generally get registration times out or one way audio |
| 09:50.52 | marc7 | Hyphenex: yeah, your method of troubleshooting via introducing IAX softphones is a good idea. |
| 09:53.06 | sempuki | *CLI> conference list AConferenceBridge |
| 09:53.07 | sempuki | User #: 1 Channel: SIP/test4-08351f00 Flags:VMT |
| 09:53.07 | sempuki | User #: 0 Channel: SIP/test3-0831d4a8 Flags:VMT |
| 09:53.22 | sempuki | neither one can hear the other |
| 09:55.19 | marc7 | sempuki: i'm no expert. i could imagine using ethereal or tcpdump to see if it's a network or configuration problem... |
| 09:55.28 | marc7 | just not something I've played around with much |
| 09:55.57 | sempuki | marc7: Im not sure Im leet enough with my network haxing to do so... |
| 09:57.01 | marc7 | hmmm, I remember with app_meetme, there's a little bell that plays when someone joins the conference. are both people hearing that? |
| 09:57.55 | badcfe | hmm. what i said was wrong everybody. actually == is for equality test in extentions.conf, but it seems to be = in AEL |
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| 10:04.22 | Hyphenex | what does this mean? [Feb 29 21:05:07] NOTICE[3884]: chan_iax2.c:7332 socket_process: Rejected connect attempt from 10.1.1.2, who was trying to reach '1000@' |
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| 10:08.11 | Der-Tim | hi there |
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| 10:09.46 | Hyphenex | I don't know how to set up IAX2 :( |
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| 10:10.15 | Der-Tim | i got a strange issue with two asterisk boxes which should connect each other by using iax2... one of the boxes has a static ip, the other doesn't... the box with dialup is also behind a nat, but the router is configured to forward port 4569 udp to the asterisk box... but still there's an unreachable on the dynamic box... anyone got time to help me out? |
| 10:10.18 | SAL123 | Hello. When playing sound, the "ts" number is not in proper order. Sometimes there are numbers about 002168 sometimes about 3504991820. Output sound is dropping out. |
| 10:12.47 | *** join/#asterisk InsolentDreams (n=Insolent@62.157.211.194) |
| 10:13.25 | InsolentDreams | Anyone know how/why/if there is a workaround to the queue status showing a agent still in a call when they transferred that call to someone else? |
| 10:14.17 | InsolentDreams | This happened on 1.2, and now on 1.4.18 also. |
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| 10:16.54 | SAL123 | Heh, many questions, no answers. :-) |
| 10:17.03 | InsolentDreams | :P :( |
| 10:17.05 | marc7 | InsolentDreams: I haven't seen that problem, but I have seen queue members get "stuck" in weird states, switching from "Not in use" to "Unavailable" for no reason. a "reload" seems to sort it out, |
| 10:17.45 | InsolentDreams | Well if that does fix it that's nice, but then how often can I run a reload? :\ |
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| 10:18.01 | Hyphenex | hmm, in asterisk, I'm getting a [Feb 29 21:18:04] NOTICE[3880]: chan_iax2.c:7364 socket_process: Rejected connect attempt from 10.1.1.2, request '1000@default' does not exist |
| 10:18.33 | InsolentDreams | I have a queue overview and monitoring system in place, and I need valid statuses for all queue members. Right now I queue every 5 seconds for statuses, how often can I reload and not cause major chaos? :P |
| 10:18.42 | Hyphenex | when in my extensions.conf I have exten => _1XXX,1,Dial(SIP/${EXTEN}) |
| 10:19.46 | marc7 | oh I entirely agree with you, that isn't an acceptable solution. using a cron script to invoke a reload through the asterisk manager interface is not something I want to be doing ever. It doesn't look like any of the regulars are around, best to try your question during north american daylight hours. |
| 10:20.08 | Hyphenex | marc7: what am I doing wrong? |
| 10:20.57 | marc7 | Hyphenex: do you have a user or peer setup in your iax.conf? |
| 10:21.00 | InsolentDreams | Will do marc, thanks. Yeah sucks being in Germany, as far as IRC goes. Thanks |
| 10:21.10 | Hyphenex | marc7: friend |
| 10:21.11 | marc7 | InsolentDreams: you could try asterisk-users@ |
| 10:21.22 | InsolentDreams | willdo |
| 10:21.50 | marc7 | InsolentDreams: the mailing list is very high traffic, search first to avoid being hit with a cluebat, there could be an ongoing conversation about the problem you're describing. |
| 10:22.16 | Der-Tim | even if i try to use registration for my iax connections problem, i only see POKEs and REGREQs going from dynamic host to static * |
| 10:22.19 | Der-Tim | any idea? |
| 10:22.36 | Hyphenex | marc7: should it be friend? |
| 10:22.41 | Hyphenex | type=friend, or something else? |
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| 10:24.00 | Hyphenex | marc7: please help :P |
| 10:24.02 | marc7 | Hyphenex: I wouldn't expect that to be a problem for your testing, but if you're giving IAX2 softphones to people, maybe type=user (optionally, with a secret specified) is more appropriate. |
| 10:24.27 | marc7 | Hyphenex: doing my best, I'm no expert. |
| 10:24.47 | Hyphenex | marc7: Your my expert ;) |
| 10:24.51 | Strom_C | type=user is pointless if you want the phones to receive calls too |
| 10:25.04 | marc7 | see? already Strom knows more than I do. |
| 10:25.11 | Hyphenex | yeah, I kind of did want them to receive calls |
| 10:26.05 | Strom_C | Hyphenex: what you have is a dialplan problem, not an iax.conf problem |
| 10:26.41 | Hyphenex | Strom_C: Really? So I don't use exten => _1XXX,1,Dial(SIP/${EXTEN})? |
| 10:26.47 | Strom_C | you do |
| 10:26.54 | Strom_C | but you have to make sure it's in the right context |
| 10:27.38 | Hyphenex | it's in [Scott] |
| 10:27.40 | Der-Tim | by now i've moved forward a little bit... i can now see on the dynamic *, that it's connected to the static *, but on the other side, there's still "unreachable"... any ideas? |
| 10:27.41 | Strom_C | your phones are looking for that extension in the "default" context |
| 10:28.05 | Strom_C | you either need to set context=scott in iax.conf or you need to include scott in your default context |
| 10:28.25 | Hyphenex | Strom_C Ahhk, I think I see. Thanks :) |
| 10:29.33 | Hyphenex | Awesome. Thanks Strom_C :) |
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| 10:55.54 | marc7 | could anyone shed any light as to why some $CDR variables are showing up blank? for example, I can NoOp(${CDR(start)}) to see when the call's started, but a NoOp(${CDR(billsec)}) (or duration) just shows up at 0 at any interval of the call. the data being fed into cdr_mysql is accurate, so this is puzzling to say the least. |
| 10:56.04 | *** join/#asterisk s0ck (n=m@unaffiliated/s0ck) |
| 10:57.10 | s0ck | anyone still running 1.2? :) |
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| 11:06.35 | tzafrir_home | s0ck, occasionally, as a testing environment |
| 11:08.25 | *** join/#asterisk hi365_m (n=hi365@213.151.62.23) |
| 11:12.15 | s0ck | tzafrir_home: the box i setup a while ago here is still running 1.2. just turned it back on now, it aint been used for a while |
| 11:12.20 | s0ck | spose i should jump on the 1.4 bandwagon |
| 11:12.46 | marc7 | feeling adventurous? there's always 1.6 ;-) |
| 11:12.54 | s0ck | hehe |
| 11:12.55 | s0ck | any tangible improvements? |
| 11:15.01 | marc7 | you can't really "touch" or "feel" improvements to asterisk, so in that regard, no. i'm sure there's spiffy new features though! |
| 11:15.55 | nixguy | <PROTECTED> |
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| 11:32.59 | Der-Tim | anyone here who could help me with that issue? >> register_verify: No registration for peer |
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| 11:56.25 | ccesario | hello, I'm making tests with agi but I dont get write error messages do STDEER ... only to STDOUT ..... any idea ? |
| 12:00.58 | tzafrir_home | I have no idea what you're talking about. AGI? |
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| 12:07.41 | ccesario | tzafrir_home, yes :) http://www.voip-info.org/wiki/view/Asterisk+AGI+php |
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| 12:13.03 | Asterisk-nob | how can i get "reload" command output in a file? |
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| 12:42.07 | alexfranci | hello! i have some troubles with asterisk. i have a geographic phone number via external SIP proxy: I set trunk sip and it works with soft-phone like Twinkle. But i would like that when i call the geographic number via normal line, my asterisk would answer and play a song (moh), but i couldn't. can somebody help me? |
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| 12:56.36 | hi365_m | what does this mean?? |
| 12:56.37 | hi365_m | WARNING[8706] chan_sip.c: sip_xmit of 0x9543550 (len 776) to 212.150.88.20:5060 returned -2: Network is unreachable |
| 13:00.43 | RoyK | hi365_m: try to ping that address |
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| 13:02.30 | riddlebox | has anyone configured sla.conf? I have followed the docs for it, but when I do sla show trunks, or sla show stations it shows nothing configured |
| 13:04.09 | hi365_m | RoyK:pining fine. its an incoming did, it hits my server, goes thru some hoops (set language, goto, etc) but chokes on Answer() |
| 13:05.32 | *** join/#asterisk myiagy (n=myiagy@200.215.59.133) |
| 13:06.26 | alexfranci | hi365, can anybody help me? |
| 13:06.28 | alexfranci | please |
| 13:06.28 | RoyK | hi365_m: enable full logging in logger.conf, set verbose 9, set debug 9, logger rotate, reproduce, logger rotate, pastebin the file named full.x where x is highest |
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| 13:16.31 | hi365_m | RoyK: full => notice,warning,error,debug,verbose <--- like this? |
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| 13:21.54 | RoyK | hi365_m: bingo |
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| 13:23.05 | ZeNN | i've got really crappy audio when i listen to voicemail using a LAN connection to the PBX with coden u-law. Making a phonecall through the PBX to the outside world with an AVM Fritzcard has good quality. But when dialing into the PBX from the outside world to the IVR menu, the low quality audio is experienced again. Is there a way to set the quality of the audio (IVR / Voicemail) to a better quality then what i've got now |
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| 13:35.41 | alexfranci | well, i have problem with moh |
| 13:36.06 | alexfranci | i can hear from inside, but not from outside (calling my geographical number via FXO line) |
| 13:36.10 | alexfranci | can anybody help me? |
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| 13:46.39 | hi365_m | RoyK: logger rotate isnt doing anything |
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| 13:46.50 | hi365_m | current log file is 6mb |
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| 13:50.06 | Zefk | Hi, after a swith statement in ael the ${EXTEN} became the pattern entry in that context. anyone knows why and how to avoid this ? |
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| 13:51.24 | RoyK | hi365_m: then rotate the logs manually :P |
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| 13:54.46 | tzafrir_home | use logrotate... |
| 13:55.09 | RoyK | tzafrir_home: see above - it wasn't for periodic rotation - just for debug |
| 13:55.18 | *** join/#asterisk af_ (n=getsmart@88-149-241-244.dynamic.ngi.it) |
| 13:55.31 | RoyK | still, if the 'logger rotate' command is broken, it should be fixed or at least removed |
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| 14:11.56 | *** join/#asterisk docelmo (n=vircuser@h74.196.180.12.cable.rstb.cablerocket.net) |
| 14:12.05 | hi365_m | RoyK: this is the last couple of lines, should be enough |
| 14:12.11 | hi365_m | http://pastebin.ca/923357 |
| 14:13.28 | Havokmon | I'm trying to troubleshoot my aastra "No service" issue. I'm doing a basic tcpdump on *. It seems Aastra 'renews' almost every 2 mins. If * doesn't respond within 5 seconds, the Aastra says "No Service" |
| 14:13.51 | *** join/#asterisk nirz (n=nir@IGLD-83-130-229-206.inter.net.il) |
| 14:14.19 | e` | i have to asterisk servers that call each other over iax2. on random occasions serv1 calls dont reach serv2. if you try again, sometimes it works. it's never the other way around. it's always from serv1 going to serv2. any ideas? |
| 14:14.32 | Havokmon | There was only 1 active call when this last happened, and it's a new Dell T105. I haven't 'seen' load exceed 15%... What should I try? |
| 14:14.45 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:16.05 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 14:17.53 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 14:18.21 | Katty | hai |
| 14:19.04 | grandpapadot | Is there a minimum in asterisk 1.2 to the Queue timeout? I'm trying to stack some queues but giving a 10 second timeout doesn't seem to do anything. |
| 14:19.28 | grandpapadot | Using Queue(someq|trT|||10) |
| 14:20.29 | *** join/#asterisk ber___ (i=brad@neu.cow.org) |
| 14:20.58 | ber___ | are there irc scripts out there to re-login and identify infinite loop |
| 14:21.08 | ber___ | freenode keeps splitting and booting |
| 14:21.20 | *** join/#asterisk seanbright (i=seanbrig@65.207.74.18) |
| 14:21.43 | hi365_m | all ok here. try switching servers, cliens, or isp's |
| 14:22.02 | Katty | anyone use call barging? |
| 14:23.45 | grandpapadot | Ok, I have a 1.2 call queue that if it has members it ignores my 10-second timeout in Queue(someq|trT|||10) but if I take all the members out, it times out at 10 seconds. Any ideas? |
| 14:25.01 | *** join/#asterisk BipBip (n=BipBip@194.65.5.235) |
| 14:25.13 | _ShrikE | Katty: yes, with chanspy |
| 14:28.54 | RoyK | hi365_m: sorry - don't know then |
| 14:29.57 | *** join/#asterisk mocker (n=kyle@mocker.org) |
| 14:30.05 | mocker | Good morning. |
| 14:30.30 | RoyK | good afternoon |
| 14:31.19 | mocker | RoyK: :) |
| 14:31.31 | madduck | why would asterisk send 183 and 180 in response to an INVITE, but then not send a ring back tone? |
| 14:31.36 | *** join/#asterisk stansmith (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 14:31.54 | Katty | _ShrikE: was it hard to get working? |
| 14:33.19 | mocker | Is there any way to detect an Answer on the remote end w/ Asterisk? |
| 14:33.40 | stansmith | ${CHANNELSTATUS} ? |
| 14:33.50 | mocker | I'd like to wait until the phone line is actually answered before starting to play a message, but most of the examples I see just use Waits or Loops |
| 14:34.06 | *** join/#asterisk MaliutaWrk (i=nikolai@119.11.106.217) |
| 14:34.39 | ManxPower | mocker: Yes. |
| 14:34.49 | *** join/#asterisk pwd5 (n=feng@MTRLPQ02-1177745966.sdsl.bell.ca) |
| 14:35.01 | ManxPower | Use anything except for loopstart or groundstart FXO |
| 14:35.15 | mocker | ManxPower: On a PRI? |
| 14:35.20 | stansmith | so that leaves kewlstart? |
| 14:35.22 | _ShrikE | Katty: not that I remember |
| 14:35.29 | ManxPower | mocker: yes, it works fine on a PRI |
| 14:35.38 | ManxPower | stansmith: no, that leaves PRI, E&M, E&M/Wink |
| 14:35.44 | puzzled | hi |
| 14:35.52 | ManxPower | Heck, I don't even known if E&M does answer supervision |
| 14:35.55 | mocker | ManxPower: And ${CHANNELSTATUS} is the variable to check? |
| 14:36.07 | ManxPower | mocker: no. DIALSTATUS or HANGUPCAUSE |
| 14:36.14 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 14:36.22 | ManxPower | but you can only get that info after the call ends |
| 14:36.46 | mocker | Ahh. |
| 14:36.47 | mocker | Ok. |
| 14:37.07 | ManxPower | mocker: almost all VoIP providers use PRIs, so it will also work on most SIP/IAX connections as well |
| 14:37.22 | mocker | ManxPower: But only *after* the call is d one. |
| 14:37.41 | ManxPower | mocker: Just remember, if the call is going to a cell phone, it will be considered answered when the cellphone voicemail picks up. |
| 14:37.50 | mocker | ManxPower: Yup, fine with that. |
| 14:37.51 | ManxPower | mocker: exactly what are you trying to accomplish? |
| 14:38.02 | mocker | ManxPower: Detect when calls start failing. |
| 14:38.07 | mocker | Almost like a remote monitoring app. |
| 14:38.21 | mocker | I think DialStatus will work well. |
| 14:38.49 | mocker | Wish there was a way to do it during the call instead of after, but for this application I don't think it matters. |
| 14:39.09 | Katty | _ShrikE: i don't suppose you documented the process |
| 14:39.22 | [TK]D-Fender | mocker: You want to know once the call is answered? |
| 14:39.33 | [TK]D-Fender | Katty: Mew. |
| 14:39.39 | creativx | woof! |
| 14:39.39 | ManxPower | mocker: For all outgoing calls my scripts check the HANGUPCAUSE, if it's not a cause we know means "OK", then we log all the info with a script |
| 14:39.41 | [TK]D-Fender | Katty: "core show application chanspy" |
| 14:39.59 | mocker | [TK]D-Fender: Yeah. |
| 14:40.03 | Katty | [TK]D-Fender: mew. |
| 14:40.03 | stansmith | ${SWIFT_DTMF} is my friend |
| 14:40.07 | [TK]D-Fender | mocker: To do what? |
| 14:40.10 | ManxPower | actually we just retry the call using a different carrier, if that fails THEN we log it |
| 14:40.12 | stansmith | i know this cause he helps me out with challenging problems |
| 14:40.26 | ManxPower | You should know that most carriers randomly fail a small number of calls for no reason we can see |
| 14:40.30 | mocker | [TK]D-Fender: Detect when calls start failing. |
| 14:40.37 | mocker | Like a remote monitoring app. |
| 14:41.04 | mocker | ManxPower: Oh, that's the bored tech. :) |
| 14:41.09 | mocker | Let's fail... THIS call. |
| 14:41.20 | [TK]D-Fender | mocker: Dialstatus will do that.. there is no such thing as "start failing". Its either dialing, IN CALL, or finished with some sort of status. |
| 14:41.29 | Katty | [TK]D-Fender: interesting |
| 14:41.40 | Katty | [TK]D-Fender: i think i might like snooping |
| 14:41.43 | Katty | [TK]D-Fender: i mean. |
| 14:41.45 | stansmith | :-O |
| 14:41.48 | [TK]D-Fender | Katty: EVAL! |
| 14:41.50 | Katty | [TK]D-Fender: making our server better for the company. |
| 14:42.03 | *** join/#asterisk pa (n=paolo@unaffiliated/pa) |
| 14:42.08 | Katty | [TK]D-Fender: ;) |
| 14:42.12 | [TK]D-Fender | Katty: "Power corrupts.... absolute power is kinda neat" |
| 14:42.16 | Katty | [TK]D-Fender: i'm sure the sales managers would like it for the telemarketers |
| 14:42.24 | ManxPower | mocker: if the call fails, Dial will return immediatly anyway |
| 14:42.26 | [TK]D-Fender | Katty: NEVER give them that power!!!! |
| 14:42.30 | Katty | [TK]D-Fender: and that 'private whisper' mode is... |
| 14:42.41 | Katty | [TK]D-Fender: so they can talk to the telemarketer, but client can't hear? |
| 14:42.44 | _ShrikE | Katty: We use Chanspy(SIP,qw) |
| 14:42.56 | [TK]D-Fender | Katty: thats what *I* thought, but apparently whisper plays to both |
| 14:42.58 | ManxPower | mocker: don't expect to make this work for anything except PRI |
| 14:43.03 | Katty | [TK]D-Fender: fascinating. |
| 14:43.52 | Katty | how do you tell it which extension you want to listen in on? |
| 14:43.59 | ManxPower | syntax, isn't that when you have to pay a tax on booze and smokes |
| 14:44.02 | Katty | Chanspy(SIP/${EXTEN}? |
| 14:45.18 | Katty | i don't understand the different between w and W 'whisper' and 'private whisper' |
| 14:46.20 | stansmith | haha |
| 14:46.28 | budol | [TK]D-Fender: so they can talk to the telemarketer, but client can't hear???? HOW I CAN DO THIS???? |
| 14:46.51 | ManxPower | budol: lucky for you all the info you need is on your screen right now |
| 14:47.12 | budol | ManxPower : REALLY WAIT |
| 14:47.17 | budol | thanks dude |
| 14:47.41 | ManxPower | budol: I see you changed your nick from drmessano |
| 14:49.55 | mocker | ManxPower: We'll be using a PRI, so that should be fine. |
| 14:51.19 | Katty | [TK]D-Fender: exten => *110,1,Chanspy(110) would chanspy SIP/110? |
| 14:51.38 | [TK]D-Fender | Katty: Whats that cute instruction page say abaout that parameter? |
| 14:51.52 | Katty | [TK]D-Fender: oh, the one that's confusing the jebus out of me? |
| 14:51.59 | [TK]D-Fender | Katty: Yeah, that one! |
| 14:52.42 | Katty | [TK]D-Fender: i'm having a hard time reading it since it's not in Kat. |
| 14:52.55 | Katty | [TK]D-Fender: this 'agent' stuff is smidgen confusing |
| 14:53.02 | [TK]D-Fender | Katty: I might suggest a career choice re-alignment... |
| 14:53.26 | ManxPower | Katty: look on the wiki for examples or search the mailinglist archives |
| 14:53.28 | ManxPower | ~docs |
| 14:53.28 | jbot | from memory, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 14:53.31 | ManxPower | Cmailinglist |
| 14:53.35 | Katty | [TK]D-Fender: wow. |
| 14:53.37 | ManxPower | ~mailinglist |
| 14:53.38 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 14:53.59 | Katty | [TK]D-Fender: i'm not even sure how to respond to that statement. |
| 14:54.25 | [TK]D-Fender | Katty: would probably looks something like this ;) http://ratonland.org/img/articles/bunny-pancake.gif |
| 14:54.25 | InsolentDreams | How do you go about fixing the bot's text, that jbot's second link needs the comma removed, and the third link doesn't work |
| 14:55.20 | Katty | [TK]D-Fender: i'm sorry i don't know as much as everyone else in here. |
| 14:55.59 | Katty | [TK]D-Fender: you've ruined my entire morning with that comment. |
| 14:56.03 | Katty | [TK]D-Fender: thanks. |
| 14:56.06 | budol | is this a good reference to configuring asterisk for call barging with Chanspy? |
| 14:56.12 | budol | is this a good reference to configuring asterisk for call barging with Chanspy? http://www.asteriskguru.com/board/call-barge-in-vt1623.html |
| 14:57.00 | stansmith | budol: trust no one |
| 14:57.39 | budol | lol |
| 14:57.43 | budol | hehe |
| 14:57.46 | stansmith | teehee |
| 14:58.07 | [TK]D-Fender | Katty: Hey, that was a cute bunny! |
| 14:58.55 | [TK]D-Fender | Katty: And the nudge factor wasn't on your experience, just that you really need to lock down on being able to parse instructions of how to pass parameters to an app. that format of documentation is prolific. |
| 14:59.08 | [TK]D-Fender | InsolentDreams :.. |
| 14:59.10 | [TK]D-Fender | ~docs |
| 14:59.10 | jbot | [~docs] Asterisk documentation can be found at http://www.digium.com/index.php?menu=documentation , http://www.asteriskdocs.org , http://www.asteriskguru.com , the WIKI (~wiki), or the BOOK (~book) |
| 14:59.11 | Katty | [TK]D-Fender: i'm sorry, i don't feel like talkign to you right now. |
| 14:59.18 | [TK]D-Fender | Katty: :( |
| 14:59.44 | [TK]D-Fender | Katty: and the bunny was what I offered you as an example of how you might respond to ME <- |
| 15:00.22 | InsolentDreams | [TK]D-Fender: I mean how can I update the jbot's text it spits out, since it was wrong/obsolete |
| 15:00.39 | [TK]D-Fender | InsolentDreams: And its now fixed |
| 15:01.00 | InsolentDreams | ~mailinglist |
| 15:01.00 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 15:01.07 | InsolentDreams | Eh, nope, still there |
| 15:01.16 | [TK]D-Fender | InsolentDreams: oh, that one's bad too? |
| 15:01.27 | [TK]D-Fender | InsolentDreams: jsut an extra "," right? |
| 15:01.29 | InsolentDreams | Last link is bad, and second to last link has a comma in it accidently at the end |
| 15:01.35 | InsolentDreams | Last link is a link to a link-farm |
| 15:01.37 | [TK]D-Fender | (or bad spacing more like) |
| 15:01.53 | ManxPower | I see you removed the note telling people how to use google to search the mailing list |
| 15:02.04 | *** join/#asterisk jmacz (n=jmacz@190.90.35.212) |
| 15:02.17 | ManxPower | nevermind |
| 15:02.28 | grandpapadot | Is there a way to loop through a mysql result set and grab values for use in a Dial() statement? I'm trying to build database drive hunt groups ... |
| 15:02.40 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:02.40 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:02.45 | *** join/#asterisk ddunavant (n=David@pool-71-191-30-72.washdc.east.verizon.net) |
| 15:04.02 | [TK]D-Fender | mailinglist~ |
| 15:04.05 | [TK]D-Fender | ~mailinglist |
| 15:04.06 | jbot | [~mailinglist] Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives |
| 15:04.52 | [TK]D-Fender | grandpapadot: its your dialplan, do what you like with it. |
| 15:05.32 | [TK]D-Fender | grandpapadot: You can do it in pure dialplan or depending on complexity you might be better off in AGI |
| 15:06.07 | defswork | I've got aastra 55i and expansion module with 20 extension blfs - when a ring group rings the blfs light up but sometimes don't go off - is there a workaround for this ? |
| 15:06.47 | *** join/#asterisk tobias (n=tobias@cpe-066-026-085-055.nc.res.rr.com) |
| 15:07.19 | grandpapadot | [TK]D-Fender: Ahh man, how easy was that. No problem using just MySQL() |
| 15:08.06 | Katty | [TK]D-Fender: while i agree with you, i don't appreciate you telling me i need a career change. |
| 15:08.09 | grandpapadot | [TK]D-Fender: I try to stay away from AGI on congested servers, seems like shelling out then JIT compiling a script seems like a lot of wasted overhead... |
| 15:09.23 | mocker | asterisk-users is too high volume. :( |
| 15:09.53 | [TK]D-Fender | grandpapadot: The needs define the means. |
| 15:09.55 | *** join/#asterisk gr0mit (n=tim@dhcp4.zuk40.mot-tools.co.uk) |
| 15:09.57 | *** join/#asterisk wmaulik (n=wmaulik@158.59.192.218) |
| 15:10.24 | dandre | Hello, |
| 15:10.40 | gr0mit | hello |
| 15:10.53 | [TK]D-Fender | grandpapadot: And big CPUs invented code-bloat :) Necessity is the mother of invention, over-abundance is the bastard father of waste |
| 15:11.04 | gr0mit | is anyone here using astmanproxy? |
| 15:11.14 | dandre | if I call Dial command in my dialplan, How can I get the result? |
| 15:11.46 | stansmith | ${DIAL_STATUS} ? |
| 15:12.02 | e` | i have to asterisk servers that call each other over iax2. on random occasions serv1 calls dont reach serv2. if you try again, sometimes it works. it's never the other way around. it's always from serv1 going to serv2. any ideas? |
| 15:14.15 | dandre | Is it available at the next priority or at prioriti+101? |
| 15:14.38 | [TK]D-Fender | dandre: priority jumping was deprecated in 1.2..... |
| 15:14.55 | dandre | ok |
| 15:14.57 | *** part/#asterisk Navion (n=billp@208-106-107-201.dsl-s.california.com) |
| 15:15.03 | [TK]D-Fender | dandre: and its ${DIALSTATUS} . Go read "core show application dial" |
| 15:15.24 | cesar_CR | I am using a sip cliente ins the same pc where asterisk is, and when a call another extension the do not hear me, and when another extention calls me they do hearme. |
| 15:15.51 | cesar_CR | I am using bindport and bindaddrs options.. |
| 15:15.56 | cesar_CR | is that normal ? |
| 15:16.14 | [TK]D-Fender | Movie time! |
| 15:16.48 | coppice | HTPC == DVD player with massive fan noise :-) |
| 15:18.50 | dandre | ok My problem is that I have this: |
| 15:18.50 | dandre | exten => s,1,Dial(SIP/10&SIP/11) |
| 15:18.50 | dandre | I want to write data in cdr only for the peer that answer le call. How should I do? |
| 15:19.02 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:19.02 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 15:20.16 | Zefk | I have an ISDN trunk that I can dial 3 patterns through it. The dialplat for each pattern is the same. How can I write the dialplan only once ? |
| 15:20.19 | tzanger | coppice: haha |
| 15:20.24 | dandre | in fact it is exten => s,1,Dial(Local/10@default&Local/11@default) |
| 15:20.47 | tzanger | coppice: I just discovered that slin->ulaw is pretty straightforward, but before ulaw is transmitted all its bits are inverted. |
| 15:20.53 | tzanger | that's not unusual, I've seen that kind of thing before |
| 15:21.09 | tzanger | however slin->alaw is equally straightforward but before alaw is transmitted all the even bits are inverted... wtf??!! |
| 15:21.21 | [TK]D-Fender | coppice: HTPC = an PC I hook up to my projector (with its 120" screen), and my Dolby Digital receiver :) |
| 15:21.25 | [TK]D-Fender | any* |
| 15:21.55 | [TK]D-Fender | Thats right... I have an 8' wide caller-id box ;) |
| 15:22.09 | [TK]D-Fender | *whee* |
| 15:22.12 | coppice | the alternate bits is a whitening thing, because they defined something channel related in the channel instead of the codec for some stupid historical reason |
| 15:22.14 | iCEBrkr | 'morning! |
| 15:22.53 | coppice | i mean in the codec instead of the channel |
| 15:23.23 | tzanger | coppice: hmm I wonder if I can turn this even bit whitening into a dental product |
| 15:24.34 | tzanger | coppice: based on other tests here, it does seem like a 3rd harmonic and not some mirror image of the signal in the passband (I thinkt hat' sthe right term) |
| 15:25.05 | *** join/#asterisk FlatFoot (n=bigflatf@80.88.192.83) |
| 15:25.25 | coppice | we DSP folk hate anything that's not white. we really should have a KKK algorithm or something |
| 15:25.36 | tzanger | if I feed in 500Hz sine, I get out 500 at 0dBm, then 1500, 2500, 3500 at -9dBm I think, and 2000 and 3000 at -18 I think |
| 15:25.59 | Katty | [TK]D-Fender: i got it to work (chanspy) by the way. |
| 15:26.23 | coppice | sounds pretty harmonious |
| 15:26.32 | Katty | [TK]D-Fender: so take that, you pancake and bunny picture producer. |
| 15:27.28 | tzanger | coppice: http://www.mixdown.ca/~andrew/dump/audio.png <-- I believe that was the 500Hz... it's almost as if *one* of the bits is inverted, or two are swapped |
| 15:27.36 | tzanger | but as I sweep the frequency the harmonics move appropriately |
| 15:30.08 | coppice | kinky :-) |
| 15:30.44 | stansmith | unappropriate for IRC |
| 15:31.14 | coppice | pictures like that are sooo much easier to relate to than all those complex numbers :-) |
| 15:31.31 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 15:33.23 | *** join/#asterisk jomarie (n=asterisk@158.59.192.218) |
| 15:33.25 | tzanger | coppice: haha |
| 15:33.29 | coppice | I wonder what could be unappropriate for IRC? :-\ |
| 15:33.29 | wmaulik | I was wondering what the second line in this was for, i am going to link the first 3 lines of part of our sip.conf file [218] |
| 15:33.29 | wmaulik | type=friend |
| 15:33.29 | wmaulik | secret=welcome |
| 15:33.36 | lmadsen | unappropriate? :) |
| 15:33.47 | lmadsen | wmaulik: it's a password... |
| 15:34.13 | wmaulik | no thats secret=welcome, i was wondering about type=friend |
| 15:34.27 | lmadsen | it's the type you are defining for the peer |
| 15:34.39 | lmadsen | check the docs to determine what the types are, and what they are for |
| 15:34.45 | drmessano | So this secret=Wxz74g!h3 in my sip.conf is a password? |
| 15:35.00 | lmadsen | drmessano: sometimes :) |
| 15:35.01 | drmessano | What about secret=hg465dmfFd ? |
| 15:35.03 | coppice | looks more like leet speak |
| 15:35.13 | lmadsen | drmessano: what'd you just call my mother?! |
| 15:35.19 | drmessano | lol |
| 15:35.56 | ManxPower | drmessano: just plays at being stupid. He's good at it. |
| 15:36.15 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 15:36.17 | wmaulik | would asterisk guru have an explanation about the types |
| 15:36.34 | ManxPower | He likes to tell newbies totally wrong information just because he things it's fun. |
| 15:36.58 | drmessano | ManxPower: Plays at being an ass, then calls everyone else an ass. It's amusing and unhelpful |
| 15:37.06 | stansmith | ~types |
| 15:37.17 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:37.21 | ManxPower | drmessano: telling newbies totally wrong information is neither amusing nor helpful |
| 15:37.26 | drmessano | I've never told anyone any wrong information |
| 15:37.36 | drmessano | So go crawl back in your cave |
| 15:37.38 | coppice | oooh, Mr Perfect |
| 15:38.30 | *** join/#asterisk angryuser (n=aster@df01t2-213-44-151-250.d4.club-internet.fr) |
| 15:38.40 | drmessano | ManxPower: I may make a few jokes here and there, but I have never intentionally misled someone, so you can drop this crap and your attitude, now |
| 15:38.47 | drmessano | and that my last comment on that |
| 15:42.07 | stansmith | people cant make it to the ballots to chose leadership, but we can make it jacob's and the dealership |
| 15:42.11 | *** join/#asterisk Cyon (n=cyon@fxp1.dmz1.cro1.bestweb.net) |
| 15:43.03 | *** join/#asterisk jameswf-home (n=james@mobile-166-214-245-212.mycingular.net) |
| 15:45.04 | angryuser | drmessano hi |
| 15:45.16 | b11d | . |
| 15:45.22 | drmessano | Hi |
| 15:46.44 | angryuser | i have installed sipproxd with pf sence, never used sip proxy before, how should i setup asterisk ? |
| 15:47.05 | angryuser | do i need to create all user in sip proxy, or only onbound peers ? |
| 15:47.12 | angryuser | outbound |
| 15:47.29 | drmessano | Not sure.. never messed with any sip proxies before |
| 15:47.29 | angryuser | outbound, i think ;) |
| 15:48.00 | angryuser | yeah, i need something with traffic shafting |
| 15:48.27 | angryuser | so pf sense was not so bad idea |
| 15:48.44 | drmessano | actually, I take that back |
| 15:49.13 | drmessano | I did use sipproxd on a router once, but I was unaware that it had user accounts.. |
| 15:49.27 | madduck | can someone please tell me how to make asterisk send the ring back tone when it sends 183 and 180 in response to INVITEs? |
| 15:49.37 | drmessano | I thought it was a shim for NAT transversal more or less |
| 15:49.41 | drmessano | ok |
| 15:50.39 | ManxPower | madduck: it should do that by default |
| 15:50.48 | madduck | ManxPower: it doesn't though... :/ |
| 15:50.57 | ManxPower | madduck: you might want to make sure you have a /etc/asterisk/indications.conf |
| 15:50.58 | madduck | i see the 180 and 183 coming in, but there is no sound |
| 15:51.43 | ManxPower | Do you mean when Asterisk RECEIVES the 180 and 183? |
| 15:51.50 | madduck | no, it sends them |
| 15:51.50 | *** join/#asterisk jameswf (n=james@mobile-166-214-225-111.mycingular.net) |
| 15:51.57 | madduck | i make a call with the client |
| 15:52.00 | madduck | it sends INVITE to asterisk |
| 15:52.07 | madduck | asterisk sends 183 then 180 |
| 15:52.11 | madduck | but there is no tone |
| 15:52.18 | ManxPower | any tones played after an Answer (or implied answer) are handled by indications.conf |
| 15:53.08 | iCEBrkr | It's all chatty in here today |
| 15:53.29 | stansmith | LemensTS: keys open doors |
| 15:53.29 | madduck | ManxPower: i can hear sound as soon as the peer picks up |
| 15:53.32 | madduck | but no dialing whatsoever. |
| 15:53.53 | angryuser | pfsense got nat, but it is written on their page "you cannot use multiple phones with a single outside server without using a dedicated public IP per device" |
| 15:54.05 | ManxPower | madduck: Do you or do you not have a valid /etc/asterisk/inidcations.conf? |
| 15:54.08 | [TK]D-Fender | tzanger>but as I sweep the frequency the harmonics move appropriately <- I read this and only think about guitars :) |
| 15:54.18 | madduck | ManxPower: i do have the default... |
| 15:54.21 | angryuser | hey fender |
| 15:54.22 | madduck | it looks good. |
| 15:54.31 | ManxPower | madduck: good, then that's not the problem. |
| 15:54.48 | madduck | ManxPower: http://scratch.madduck.net/__tmp__indications.conf -- i only changed s/us/ch/ in the first line |
| 15:55.02 | grandpapadot | Will Dial() in 1.2 support entering an extension while ringing? |
| 15:55.07 | ManxPower | angryuser: that sounds like a limitation of pfsense |
| 15:55.16 | coppice | [TK]D-Fender: well, he seems to have a bent string on his guitar http://www.mixdown.ca/~andrew/dump/audio.png |
| 15:55.23 | ManxPower | as most routers let you have multiple phones behind their nat going to an external server all the time |
| 15:55.31 | angryuser | <ManxPower> but "The sipproxd package will provide a work around for this issue" ;) |
| 15:55.34 | ManxPower | grandpapadot: no |
| 15:55.40 | grandpapadot | Thanks. |
| 15:55.51 | drmessano | angryuser: Not necessary for what you described |
| 15:55.53 | ManxPower | angryuser: why not use a NAT system that does not have that limitation? |
| 15:56.03 | drmessano | I have lots of phones behind NAT to an external server |
| 15:56.08 | *** join/#asterisk _mm_ (n=mmclain@cpe-76-93-236-54.dc.res.rr.com) |
| 15:56.30 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 15:56.43 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 15:56.45 | tzanger | [TK]D-Fender: :-) I haven't played in a long time now :-( |
| 15:56.51 | ManxPower | madduck: for debugging change it back to us and stop and start asterisk |
| 15:57.00 | [TK]D-Fender | tzanger: Must to correct! |
| 15:57.35 | madduck | ManxPower: i did that. same problem. |
| 15:57.40 | JerJer | anyone happen to know how many Digium Transcoder cards one could have in a system? I got a guy wanting to use a 13 Slot PCI expansion doohicky |
| 15:58.01 | tzanger | agreed, I intend on correcting it this weekend |
| 15:58.16 | ManxPower | madduck: weird |
| 15:58.30 | angryuser | ManxPower, maybe i will try ipcop |
| 15:58.31 | ManxPower | madduck: how are you dialing out? SIP, IAX, PRI? |
| 15:58.37 | madduck | and i tried twinkle and ekiga, both... |
| 15:58.40 | madduck | ManxPower: SIP |
| 15:59.04 | ManxPower | madduck: can you pastebin the output of a single call that has this issue? |
| 15:59.14 | madduck | yessir, sec |
| 15:59.16 | nixguy | calling 1 number i want 2 phones to ring but only if non of the phones are "busy" is this possible |
| 15:59.24 | nixguy | ? |
| 15:59.26 | file | JerJer: I think the driver is hardcoded at 24... and they shouldn't generate interrupts unless being used... that exhausts my answer |
| 15:59.27 | madduck | ManxPower: client-side log okay? or do you want asterisk sip debug? |
| 15:59.29 | ManxPower | nixguy: yes |
| 15:59.37 | nixguy | ManxPower: master please tell me more |
| 15:59.44 | ManxPower | madduck: I don't read sip debug without lots of cold hard cash. |
| 15:59.54 | madduck | :) |
| 16:00.22 | ManxPower | madduck: I want the output of the call from the server that you think should be providing ringback |
| 16:01.00 | ManxPower | nixguy: The easiest way is to slightly change your requirements. It would be trivial to "ring two phones at the same time, but not ring a phone if it's busy" |
| 16:01.01 | madduck | ManxPower: http://rafb.net/p/iEFaK590.txt |
| 16:01.09 | madduck | ManxPower: this is twinkle's log |
| 16:01.24 | ManxPower | madduck: now repaste it without sip debug |
| 16:01.35 | madduck | nah, this isn't sip debug |
| 16:01.43 | nixguy | ManxPower: the reason is, if you have both a mobile and a staitonary phone, you dont want it ringing beside you @ your desk constantly when already speaking in your mobilephone |
| 16:01.47 | madduck | ManxPower: i am unsure what you mean... |
| 16:02.12 | madduck | i have a softphone connecting to asterisk connecting to an upstream provider on _0X numbers |
| 16:02.22 | ManxPower | madduck: Here is what I am assuming. SIP Phone -> Asterisk -> Internet -> SIP ITSP |
| 16:02.40 | ManxPower | so, as I said I want the asterisk cli output of a call with this issue. |
| 16:03.44 | grandpapadot | How does one "unset" a variable in asterisk 1.2? Set it to nothing or null? |
| 16:03.45 | madduck | ManxPower: http://rafb.net/p/1fdRux12.txt -- nothing else is written to the CLI |
| 16:04.23 | ManxPower | madduck: Then you have something MAJORLY wrong. I don't see any Dial lines in that output. |
| 16:04.38 | ManxPower | You can't do a call without a Dial line . |
| 16:04.44 | madduck | you do mean asterisk -r output? |
| 16:04.49 | madduck | or asterisk -vvvc? |
| 16:05.01 | Corydon76-dig | grandpapadot: in C or in the dialplan? |
| 16:05.09 | ManxPower | madduck: do you understand the difference between -r and -c ? |
| 16:05.10 | *** join/#asterisk nirz (n=nir@IGLD-83-130-229-206.inter.net.il) |
| 16:05.11 | *** join/#asterisk ruied (n=ruied@pal-213-228-184-21.netvisao.pt) |
| 16:05.22 | JerJer | ManxPower: an r ? |
| 16:05.36 | madduck | ManxPower: not with respect to log output. -r connects to a running instance, -c opens the CLI after starting asterisk |
| 16:05.57 | ManxPower | correct. |
| 16:06.14 | ManxPower | asterisk -rvvv or -cvvv is what you want |
| 16:06.15 | Corydon76-dig | What does -rc do, then? :-) |
| 16:06.19 | grandpapadot | @Corydon76-dig: In the dial-plan ... |
| 16:06.22 | madduck | ManxPower: http://rafb.net/p/NSX98596.txt |
| 16:06.35 | ManxPower | Corydon76-dig: hopefully generate an error |
| 16:06.35 | Corydon76-dig | grandpapadot: just set it to blank |
| 16:06.35 | grandpapadot | Maybe just set it to a non-existing variable ${NULL} or something? |
| 16:06.37 | Corydon76-dig | ManxPower: nope |
| 16:06.52 | madduck | the sipsock_read errors are after i hang up |
| 16:07.17 | ManxPower | madduck: how long did you let the call rung? |
| 16:07.22 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 16:07.58 | madduck | only a few seconds, but no dial tone appears even after a while. lemme rerun with a longer time |
| 16:08.12 | ManxPower | madduck: first you say ringing, now you say dialtone. which is it? |
| 16:08.25 | Qwell | ManxPower: it's really congestion |
| 16:08.31 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
| 16:08.31 | madduck | the ring back tone, the beep beep beep thing you hear when you call someone while their phone is ringing |
| 16:08.34 | gr0mit | anyone seeing it quietly stop talking? |
| 16:08.35 | ManxPower | madduck: I'm more interested in if hte CLI output changes if you leave it ring for 15 seconds |
| 16:09.10 | madduck | nothing after 15 seconds |
| 16:09.20 | madduck | (still ringing, now 30) |
| 16:10.39 | JerJer | file: thanks |
| 16:10.58 | JerJer | Anyone ever tried to run Asterisk on an IBM blade server situation ? |
| 16:11.27 | Qwell | JerJer: it's "just another server", right? |
| 16:11.48 | JerJer | kinda, i guess |
| 16:12.08 | JerJer | sorta worried about IRQ sharing and what not |
| 16:12.19 | nixguy | is therea a gui the enduser can use to configure forwarding etc? |
| 16:12.23 | Qwell | they're independent machines, aren't they? |
| 16:12.25 | stansmith | IRQ is as IRQ does |
| 16:12.26 | nixguy | freepbx seems like an admingui tool |
| 16:12.44 | Qwell | nixguy: Switchvox |
| 16:12.50 | JerJer | Qwell: the docs I have read talked about shared this and shared that |
| 16:13.01 | ruied | I'm having a problem, I use two networks (10.1.0.0/16 and 10.2.0.0/16) separated by a router, if I place the phones in the network outside the * network (10.1.0.0/16), the phones can register but they can't talk, asterisk reports the following error: http://paste.uni.cc/18421 |
| 16:13.03 | Qwell | JerJer: sure, on the backplane, or whatever |
| 16:13.11 | JerJer | yeah |
| 16:13.19 | Qwell | each machine things it has it's own hardware though - and on the PCI things, they do |
| 16:13.31 | Qwell | there are PCI modules you can fit on a blade |
| 16:13.43 | Qwell | at least on the one I've played with (albeit briefly) |
| 16:13.50 | ruied | I have iptables -P FORWARD ACCEPT at the router... (to debug) |
| 16:14.05 | JerJer | the 13 PCI expansion thingy goes along with the blade servers - same system |
| 16:14.17 | ruied | so, all packets can/should pass form one network to the other.... |
| 16:14.24 | Qwell | JerJer: right, but I think they fit onto the individual blades though |
| 16:14.34 | JerJer | hmm - ok |
| 16:14.41 | Qwell | at least - on the one I've seen |
| 16:14.55 | JerJer | gonna have to build in some 'uncharted waters' language into this contract :) |
| 16:15.20 | Qwell | heh |
| 16:15.39 | nixguy | Qwell: switchvox seems to do the jobb but its not a frontend to asterisk |
| 16:15.49 | Qwell | nixguy: no, it's a full distro |
| 16:16.31 | ruied | I seems somehow asterisk is adding the public adddress ( BYE to call '75eb28c320a57cc14c19c73c2dd4429a@xxx.xxx.yyy.22) |
| 16:18.07 | stansmith | ruied: is it possible to flush the tables completely? |
| 16:18.12 | *** join/#asterisk Datax (n=Datax@glou.nurvnet.org) |
| 16:18.22 | Datax | Hi all |
| 16:18.29 | Datax | I have a "strange" issue |
| 16:18.33 | stansmith | ~hi Datax |
| 16:18.35 | jbot | Many greetings, Datax, most strange traveller, to this IRCdom of plenty. |
| 16:18.54 | Datax | I can dial through my sip provider but whenever the person on the other end picks up the phone the line dies |
| 16:18.59 | Datax | and I get this error message : [Feb 29 17:15:24] NOTICE[23646]: chan_sip.c:12189 handle_response_invite: Failed to authenticate on INVITE to '"John Gillespie" <sip:160086@10.2.0.11>;tag=as24d9a70d' |
| 16:19.16 | ruied | stansmith, do you think it could be that problem? I can try... |
| 16:19.25 | Datax | If the other person tries calling me I don't get anything (no ringing, no lines in CLI) |
| 16:19.38 | *** join/#asterisk osas (n=nnnnnnnn@nslu2-linux/osas) |
| 16:20.29 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:20.29 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:21.55 | stansmith | ruied: i always flush tables when in doubt |
| 16:23.17 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:27.39 | ruied | stansmith, ok, goint to try... |
| 16:28.09 | Datax | anyone have an idea what my problem might be ? |
| 16:30.53 | *** join/#asterisk MmixX (i=mmixx@202.124.138.69) |
| 16:32.47 | *** join/#asterisk lisandropm (n=lisandro@190.1.22.76) |
| 16:39.35 | *** join/#asterisk MaliutaWrk (i=nikolai@119.11.106.217) |
| 16:40.20 | *** join/#asterisk amir_ (n=amir@gentoo/developer/amir) |
| 16:40.37 | *** part/#asterisk amir_ (n=amir@gentoo/developer/amir) |
| 16:45.33 | *** join/#asterisk qdk (n=qdk@195.242.194.42) |
| 16:45.56 | LemensTS | Im trying to use the Transfer button on a Polycom 550. It does not work, in features.conf, [featuremap] i have blindxfer as #* and atxfer as *1 |
| 16:46.09 | LemensTS | i have a t in my dial command also |
| 16:46.57 | *** join/#asterisk ccvp (n=ax@66.0.46.210) |
| 16:47.13 | ccvp | hello |
| 16:47.25 | _ShrikE | LemensTS: the transfer button on the polycom is a feature of the phone, it does not use features.conf |
| 16:47.34 | ccvp | having issues with asterisk running in an 802.1ae environment |
| 16:47.42 | ccvp | using line rate encryption on nexus 7000 |
| 16:47.57 | LemensTS | Shrike: yea i tried to find how to change it in the web management of the Polycom, not in thier. |
| 16:48.17 | *** join/#asterisk Datax (n=Datax@glou.nurvnet.org) |
| 16:48.30 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 16:48.43 | [TK]D-Fender | LemensTS: You shouldn't be using * for DTMF transfers witha Polycom phone |
| 16:49.06 | *** join/#asterisk Havokmon (n=chown@mx1.vfemail.net) |
| 16:49.08 | Zeeek | Boyz and Gz Asstra is our guest in a few minutes on the VoIP Users Conference live: http://x2z.eu |
| 16:49.34 | Zeeek | meet us in IRC #voip-users-conference for the play by play |
| 16:49.53 | LemensTS | TDK-Fender: What should I be using then? |
| 16:49.56 | ccvp | so am i the only person in alabama that uses asterisk heh wtf |
| 16:49.58 | _ShrikE | is asstra a typ or are you implying something zeeek? |
| 16:49.59 | Havokmon | Are there /join voip-users-conference |
| 16:50.02 | ccvp | most people in here seem to be in UK/europe |
| 16:50.03 | Havokmon | lmao |
| 16:50.08 | _ShrikE | s/typ/typo |
| 16:50.12 | Zeeek | Aastra |
| 16:50.14 | ccvp | shrike, rule 1, no sed on irc |
| 16:50.15 | ccvp | thx |
| 16:50.30 | *** join/#asterisk joshaidan (n=joshaida@S0106001c1023e838.tb.shawcable.net) |
| 16:50.39 | ccvp | :) |
| 16:50.45 | Zeeek | s/asstra/aastra/ |
| 16:51.01 | _ShrikE | ccvp: no sed.. but there is jbot :p |
| 16:51.17 | *** join/#asterisk emmix-devin (n=devinsai@adsl-65-67-91-245.dsl.ltrkar.swbell.net) |
| 16:51.48 | pkunkra | no sed on irc? |
| 16:51.54 | pkunkra | never heard of that rule before. |
| 16:52.16 | marc7 | could anyone shed any light as to why some $CDR variables are showing up blank? for example, I can NoOp(${CDR(start)}) to see when the call's started, but a NoOp(${CDR(billsec)}) (or duration) just shows up at 0 at any interval of the call. the data being fed into cdr_mysql is accurate, so this is puzzling to say the least. |
| 16:52.32 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:52.36 | emmix-devin | I am using a sangoma A101D card faxing out with hylafax and iaxmodem, is there a way to disable echo cancel on when sending faxes throgh the iaxmodems |
| 16:53.01 | InsolentDreams | marc7 I had that problem with asterisk 1.2, recently upgraded to 1.4 and fixed the problem |
| 16:53.11 | InsolentDreams | marc7, what version? |
| 16:53.13 | marc7 | InsolentDreams: 1.4.18, no dice! |
| 16:53.18 | InsolentDreams | Hrm |
| 16:54.35 | Katty | hi Zeeek! |
| 16:56.15 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 16:56.53 | emmix-devin | anyone had problem with fax transmitions with hylafax using iaxmodems |
| 16:56.55 | Havokmon | The VOIP conference is starting |
| 16:57.16 | cesar_CR | hey guys I hae this error when traying to call another asterisk via IAX: Call rejected by 200.91.XX.XX: No authority found |
| 16:58.49 | *** join/#asterisk egecko (n=sam@cpe-76-176-238-225.san.res.rr.com) |
| 16:59.17 | JerJer | ohhh - happy leap day!! |
| 17:00.12 | ruied | stansmith, I did a flush to all iptables and put the forward in -P ACCEPT... the phones register from one network to another, but it seems that the voice packets don't get there if they pass the router. The phone rings and it stablishes the call but no voice |
| 17:00.17 | [TK]D-Fender | ccvp: You're right, no SED on IRC |
| 17:00.47 | [TK]D-Fender | s/You're right, no/Actually JBOT might beg to differ in that there is/ |
| 17:01.29 | drmessano | I like apples |
| 17:01.36 | [TK]D-Fender | mmmm pie |
| 17:01.42 | drmessano | s/I like apples/Beer is my fav food/ |
| 17:01.46 | drmessano | yay |
| 17:02.10 | cpm | nice |
| 17:02.19 | drmessano | yep |
| 17:02.21 | scooby2 | deadlocks every day since moving to this te212p |
| 17:02.25 | drmessano | s/yep/it rox/ |
| 17:02.44 | stansmith | ~stansmith |
| 17:03.12 | pkunkra | drmessano, hey. i have an electrical recommendation to ask of you. :-) |
| 17:03.14 | InsolentDreams | ruled: watch some tcpdump action and find out why. I'm sure something's prolly not being forwarded. |
| 17:03.38 | pkunkra | i have to buy a clamp meter to measure power consumption on the servers. |
| 17:03.53 | pkunkra | was thinking of the fluke 332. do you have any recommendations? |
| 17:04.25 | Wayhigh | pkunkra: you'd rather use a clamp meter than something like a killawatt just so you don't have to turn off the servers? |
| 17:04.37 | Wayhigh | clampmeters aren't nearly as accurate are they? |
| 17:04.44 | ruied | is there any special setting to get asterisk phones comunicating from one network to another(10.1.0.0/16 to 10.2.0.0/16). The phones can register but they can't talk to each other... I can get one way conversation if on phone is in the same netork that * is(10.2.0.0/16), but can't get audio back from the (10.1.0.x) phone to the 10.2.0.x phone... |
| 17:04.50 | pkunkra | right. i can't turn off the servers. |
| 17:05.10 | [TK]D-Fender | LemensTS: the transfer button on the phone itself |
| 17:05.12 | pkunkra | i get mixed reviews about how accurate they are. to be honest. i don't really know. |
| 17:06.01 | ruied | I have a linux router in the middle with iptables -P FORWARD ACCEPT, and no redirection, so I think the router is not the problem |
| 17:06.01 | ruied | all phones has the right gateway (they can register) |
| 17:06.20 | ruied | could it be some asterisk missconfiguration in sip.conf? |
| 17:06.26 | [TK]D-Fender | ruied: I'm betting your phones have no idea that you HAVE a gateway to that other subnet. |
| 17:06.31 | stansmith | routing tables? |
| 17:06.33 | pkunkra | wayhigh, however, the next set of power strips i buy must have meters built in. |
| 17:06.43 | [TK]D-Fender | ruied: did you set up a static route on your primary gateway? |
| 17:06.44 | pkunkra | power consumption has been the bane of my existance. |
| 17:07.18 | Wayhigh | I think that company that makes those is near here |
| 17:07.36 | pkunkra | and preferable, power strips that have snmp capability. |
| 17:07.37 | LemensTS | TKD-Fender: Yea i would but it doesnt work, that is the problem. *1 works |
| 17:07.41 | Wayhigh | certainly the one that makes the power strips that connect via usb to remove phantom loads is.. |
| 17:08.24 | pkunkra | yeah, i've seen them in other datacenters. |
| 17:08.34 | Wayhigh | hey.. btw.. I lubborz my tankless water heater |
| 17:08.39 | ruied | [TK]D-Fender, the phones have the default gw that can reach asterisk server, they can register ring and pickup, but no audio, it shouldn't be a gateway problem, I think (they can register) |
| 17:09.11 | pkunkra | used to work for travelocity, every single one of their strips has a display built in. |
| 17:09.21 | marc7 | hey guys, any idea why I couldn't access ${CDR(billsec)} or 'duration)}, but I *could* get things like |
| 17:09.25 | pkunkra | its really nice. |
| 17:09.38 | marc7 | "start" and other things that don't have to do with call length? (sorry for the broken sentence) |
| 17:10.16 | pkunkra | hmm. looked up kill-a-watt. this doesn't look like production equiment. wouldn't fit very well on the power strips i have. |
| 17:10.40 | pkunkra | i need a deviced that can do measurements on every server at regular intervals. |
| 17:10.44 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 17:11.02 | Wayhigh | pkunkra: no it's not production equipment |
| 17:11.16 | Wayhigh | I don't know of anything that can do port by port measurements |
| 17:11.19 | marc7 | pkunkra: maybe APC's got an addressable power strip that would fit the bill. |
| 17:12.00 | ruied | [TK]D-Fender, I have canreinviting=no and nat=yes in [general] sip.conf is this ok? (I think I'll need nat for future, Internet phones) |
| 17:12.01 | *** join/#asterisk ManxPower (n=manxpowe@207.191.118.2) |
| 17:12.38 | pkunkra | marc7, yeah. i would peg them for having one anyone does. |
| 17:12.54 | pkunkra | if they're out there, there very very new and very $$$ |
| 17:13.19 | marc7 | we're running one of the APC switched rack PDU's... and while the model we're using (the AP7900's) don't have per-port statistics, they give us an overall phase load and the system will alert us via SNMP or E-Mail if it exceeds the limit we've set |
| 17:13.29 | drmessano | canreinvite |
| 17:13.34 | pkunkra | oh. nice! |
| 17:13.48 | ice_croft | hi all |
| 17:15.19 | stansmith | ~hi ice_croft |
| 17:15.20 | jbot | Many greetings, ice_croft, most strange traveller, to this IRCdom of plenty. |
| 17:16.20 | pkunkra | marc7, i think some servers have management cards that how power consumption readings as well |
| 17:16.38 | LemensTS | If i call into asterisk from vonage, DTMF does not work in the IVR. Is thier a way i can make it work with it? I see you cant control DTMF in Iax channels |
| 17:16.39 | pkunkra | mine do not though. |
| 17:16.45 | ice_croft | does anybody have audiocodes mp-114? |
| 17:18.13 | pkunkra | lemensts, try rfc2833 in your asterisk config. you're probably sending the tones in-band. |
| 17:19.41 | ManxPower | LemensTS: IAX2 only supports one kind of DTMF -- Out of Band, so it is pointless to have a config option for it. |
| 17:20.12 | ManxPower | As you never told us what protocol you are using, that's all the help we can give |
| 17:20.26 | *** join/#asterisk Entr4nced (n=IMG001@67-129-213-39.dia.static.qwest.net) |
| 17:21.10 | LemensTS | ManxPower: Yea thats what i read. If I call from a vonage line, to a voipstreet number that is routed to an asterisk via IAX, then is routed to the IVR asterisk VIA IAX. I imagine codec is ulaw not positve |
| 17:21.26 | ManxPower | LemensTS: I assume you are using Phone -> Vonage -> PSTN -> Asterisk |
| 17:21.32 | marc7 | last time I'll ask for an hour or so... but NoOp(${CDR(billsec)} on a five minute call shouldn't return 0. why would it? running 1.4.18, would be happy to try 'almost' anything. |
| 17:22.08 | ManxPower | marc7: chances are all that stuff is closed and zerod out by the time you try to Noop it |
| 17:22.23 | stansmith | i got my report card today, i saw that it had all F's, i took it back to my teacher and told her to throw some D's on it |
| 17:22.25 | LemensTS | ManxPower: Phone -> Vonage -> VoipStreet -> Asterisk -> Asterisk |
| 17:22.26 | ManxPower | after all the call IS hung up |
| 17:22.37 | ManxPower | LemensTS: that's too complicated for me to help you |
| 17:22.56 | marc7 | ManxPower: even if I were to Background() an audio file that was 60 seconds long, NoOp the CDR(billsec) variable, then Background() another audio file... still in the same context of the same channel.... |
| 17:23.04 | LemensTS | manxpower: Also tried Phone -> Vonage -> local panama line -> asterisk -> asterisk same thing. |
| 17:23.33 | [TK]D-Fender | ruied: if * knows about that other subnet, "canreinvite=no" is enough to begin with. |
| 17:23.43 | ManxPower | marc7: maybe it's not set until the call is over? |
| 17:23.57 | LemensTS | Manxpower: If i do phone -> pots -> Voipstreet -> asterisk -> asterisk it works, vonage is the problem. I wasnt sure if thier was some configuring i could change on the asterisk to allow DTMF to work for it |
| 17:24.24 | ManxPower | LemensTS: you are not including even close to enough information. |
| 17:24.42 | marc7 | ManxPower: I'm actually trying to make use of the variable in the "hangup" context, so exten => h,1,Set(CallTime=${CDR(billsec)}) |
| 17:24.44 | drmessano | LemensTS: Can you call out from Vonage to a Cell and hear DTMF? |
| 17:24.46 | ManxPower | but as I said, it's so complicated. |
| 17:25.11 | marc7 | ManxPower: but it doesn't work there either |
| 17:25.38 | LemensTS | drmessano: Yes I can hear DTMF on the cell |
| 17:25.43 | ManxPower | marc7: *nod* many variables are not available in "h". I don't bill for calls, so CDR stuff is not something I do often, but I CAN tell you how many other variable act,. |
| 17:25.55 | marc7 | ;-) |
| 17:26.14 | drmessano | So somewhere between Vonage and Voipstreet the DTMF is getting lost |
| 17:28.17 | *** join/#asterisk nvrpunk (n=root@81.90.21.227) |
| 17:28.36 | drmessano | You can try calling voipstreet, but Vonage will be of little help |
| 17:28.53 | LemensTS | drmessano: Yea if i call from vonage, to a landline in panama that is connected to the first asterisk, it doesnt work either. If i call from a usa POTS line to that landline in panama that is connected to the asterisk it works. |
| 17:29.12 | nvrpunk | what is the default file extension for g729 encoded audio? |
| 17:29.19 | nvrpunk | or is there not one |
| 17:29.30 | *** join/#asterisk Jason99 (n=jason@jason.unitz.ca) |
| 17:29.32 | marc7 | ManxPower: okay, so could you at least recommend a reference which should be the authority on when CDR(billsec) and CDR(duration) should be available? is the Asterisk:TFOT book the best source, or am I going to have to start looking through code? ;-) |
| 17:30.07 | ManxPower | marc7: #asterisk-dev did not answer my request for more info on where CDR(billsec) is availble |
| 17:30.11 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 17:30.14 | *** part/#asterisk FlatFoot (n=bigflatf@80.88.192.83) |
| 17:30.18 | marc7 | ManxPower: because I can see your argument, about asterisk not wanting to make the CDR(billsec) data available until the call has fully been completed/destroyed and the data written to disk. |
| 17:30.20 | drmessano | LemensTS: You won't be able to fix this in your hardware or software, it's a carrier issue |
| 17:30.21 | Jason99 | is there a way to read postgres DB from extensions.conf? |
| 17:30.37 | LemensTS | drmessano: Ok thanks for the help though |
| 17:30.40 | [TK]D-Fender | Jason99: Yes, via ODBC |
| 17:30.45 | drmessano | LemensTS: Np |
| 17:31.04 | Jason99 | [TK]D-Fender: I'm used to using the MYSQL() cmd.. what would i be using instead? |
| 17:31.22 | [TK]D-Fender | Jason99: the BOOK. |
| 17:31.24 | *** part/#asterisk osas (n=nnnnnnnn@nslu2-linux/osas) |
| 17:31.27 | [TK]D-Fender | ~book |
| 17:31.27 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 17:31.59 | Jason99 | thanks |
| 17:32.26 | drmessano | <PROTECTED> |
| 17:32.42 | drmessano | So much for a brand new $100 2800 series router |
| 17:33.14 | denon | of course, if that's $78M at Cisco MSRP .. |
| 17:33.18 | denon | it may have just been a router or two |
| 17:33.20 | drmessano | ROFL |
| 17:33.24 | drmessano | HAHAHHA |
| 17:33.27 | drmessano | Nice... very nice |
| 17:33.36 | Qwell | give them some credit |
| 17:33.38 | denon | :) |
| 17:33.39 | Qwell | it might have been 3 |
| 17:33.43 | drmessano | HA |
| 17:33.55 | drmessano | 2 routers and a WIC |
| 17:34.11 | denon | I can see the knockoff firmware now .. "Please kindly enter your IP address>" |
| 17:34.22 | drmessano | AH yes |
| 17:34.23 | denon | "How you say .. segfault?" |
| 17:35.28 | drmessano | LQQK C1CSO MOST EXCELLENT 2800 NETWORK ROUTER |
| 17:35.34 | drmessano | $99 |
| 17:35.36 | drmessano | ha |
| 17:36.01 | drmessano | Sad thing is |
| 17:36.02 | denon | now, when they start knocking off CRS-1s |
| 17:36.14 | drmessano | They probably knocked it off, and fixed the bugs in the process |
| 17:36.31 | denon | doubtful, the knockoffs have no real world load to test with |
| 17:36.39 | Entr4nced | its a 1u linux box painted green |
| 17:36.44 | denon | as much as we mock cisco, they do run a large percentage of the net |
| 17:37.01 | denon | knocoff junipers would be nice too |
| 17:37.13 | drmessano | denon: NOT true |
| 17:37.13 | denon | knockoff |
| 17:37.16 | Entr4nced | they're better than bay networks |
| 17:37.19 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 17:37.27 | denon | drmessano: they don't run a large percentage of the net? |
| 17:37.30 | drmessano | Everywhere but China.. in China, C1CSO routers run the net |
| 17:37.34 | denon | hah |
| 17:37.42 | drmessano | Govt funded bootlegging |
| 17:37.48 | stansmith | the great firewall of china? |
| 17:38.04 | denon | Nice C1SC0 you mean |
| 17:38.11 | Datax | Hi, I've just tried this : exten => s,1,Dial(SIP/100&SCCP/101) and it won't work but it will if I remove the second entry |
| 17:38.14 | denon | everything in china seems to be "nice" |
| 17:38.17 | drmessano | You have to have the typo too |
| 17:38.23 | Datax | is there a problem when mixing SIP an SCCP in the same dial line ? |
| 17:38.30 | drmessano | C1CSO #1 NICE ROUTER BOX |
| 17:39.01 | denon | one dolla |
| 17:39.21 | drmessano | It wouldn't surprise me at all if the chinese govt funded the knock off of expensive US hardware to help bolster their infrastructure |
| 17:39.42 | drmessano | Not only for Chinese use, but bolster their economy by selling the knockoffs to the outside too |
| 17:39.43 | Datax | works now |
| 17:39.47 | Datax | I messed up ;) |
| 17:39.56 | denon | who needs to fund the knockoff? we have them made there . . |
| 17:40.04 | denon | the plants just run two shifts, one for us, one for them |
| 17:40.23 | drmessano | heh |
| 17:40.36 | denon | there are some really good articles out there about how that happens |
| 17:40.38 | denon | and it actually does |
| 17:40.43 | drmessano | I believe it |
| 17:40.50 | denon | sometimes at the very same plant, other times they build knockoff plants specifically for it |
| 17:41.18 | drmessano | Brand new Linksys PAP TOO |
| 17:41.18 | denon | companies like cisco in particular have gotten really aggressive controlling the plants there, so probably not quite so easy for those products |
| 17:41.39 | drmessano | Thats where the shadow plant comes in ;) |
| 17:42.15 | drmessano | "We will NOT tolerate knockoffs being made in our own plants" |
| 17:42.30 | drmessano | "K O, I go down street now for smoke..be back in 8 hour" |
| 17:42.42 | pkunkra | i think i need to read my EE books again. |
| 17:43.09 | pkunkra | took electrical 101 in college. hated it. was fine until we got to phasors. |
| 17:43.41 | denon | drmessano: well, when they have the complete shadow plants, I think usually there's just one mole working both sides.. I dont think the legit plant is alway in on it |
| 17:43.53 | drmessano | pkunkra: All you need to remember is that you can change the color of a wire with a piece of tape |
| 17:44.19 | drmessano | denon: surely not.. hence the 8 hour smoke break :) |
| 17:44.25 | denon | hehe |
| 17:44.32 | pkunkra | drmessano. that would be a horrible prank to play on someone who relies on those colors. |
| 17:44.45 | denon | wonder what the feds do with $79M of cisco gear .. |
| 17:44.51 | denon | we see it on ebay in a few months |
| 17:44.52 | denon | ? |
| 17:44.56 | Entr4nced | great firewall of USA |
| 17:44.57 | drmessano | Sell it on eBay for $200 |
| 17:45.03 | denon | govt auctions |
| 17:45.04 | denon | heh |
| 17:45.22 | drmessano | Doesn't a new PIX 501 run like $18,000 now |
| 17:45.24 | denon | I suppose if they were to un-cisco-ize it, it'd be ok to sell |
| 17:45.25 | drmessano | Oh sorry |
| 17:45.31 | drmessano | That's with 50 licenses |
| 17:45.36 | Entr4nced | lol |
| 17:45.45 | Entr4nced | i was gonna say. |
| 17:45.59 | drmessano | Slight exaggerated, of course :) |
| 17:46.00 | denon | ah well, gotta run |
| 17:46.03 | denon | catch ya lates |
| 17:46.12 | denon | dont sell all the c1cso gear without me |
| 17:46.16 | drmessano | lata |
| 17:46.17 | drmessano | lol |
| 17:46.23 | *** join/#asterisk envisean (n=envisean@wsip-98-173-0-162.oc.oc.cox.net) |
| 17:46.27 | budol | hello |
| 17:46.31 | pkunkra | drmessano, not a pix 501.. ;-) |
| 17:46.35 | envisean | hi budol |
| 17:46.39 | pkunkra | maybe a pix 535 :-) |
| 17:46.49 | envisean | anyone here have a good QoS script? |
| 17:46.59 | sevard | pees and cornbread |
| 17:47.04 | budol | i did config call barging using ChanSpy |
| 17:47.07 | envisean | i've tried the one here: http://www.howtoforge.com/voip_qos_traffic_shaping_iproute2_asterisk but it didn't seem to work out so well |
| 17:47.12 | envisean | or anyone have a good packetshaping app? |
| 17:47.13 | pkunkra | i have a pix 515e. $6,000 for an active/failover pair of those. |
| 17:47.20 | drmessano | nice |
| 17:47.26 | budol | envisean : i did config call barging using ChanSpy |
| 17:47.35 | pkunkra | evisean, "QoS script" is a rather vague term. |
| 17:47.43 | envisean | pkunkra: QoS solution :-P |
| 17:47.56 | envisean | budol: chanspy, will do a google on it |
| 17:48.01 | pkunkra | still a vague term. sorry. ;-) |
| 17:48.17 | scooby2 | does centos work with * or is there a better flavor to use? |
| 17:48.26 | budol | here is my sample -->> exten => 1234,1,Chanspy(SIP/${EXT}) |
| 17:48.26 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:48.39 | pkunkra | scooby2, asterisk works just fine with centos |
| 17:48.48 | envisean | pkunkra: I'm trying to make sure my voice packets get highest priority over any other traffic |
| 17:49.05 | envisean | pkunkra: currently when I download/browse the net and have an upload going in the office, it just obliterates the phone connections |
| 17:49.08 | envisean | pkunkra: mad lag |
| 17:49.11 | budol | envisean : i did but i can't work on it |
| 17:50.07 | pkunkra | envisean, i suggest you do it at the router level. if you use cisco product, you'll need to know IOS |
| 17:50.18 | envisean | pkunkra: I have the centos box |
| 17:50.23 | envisean | handling the routing and ip forwarding |
| 17:51.28 | pkunkra | envisean, let me get this straight. the linux box is doing all of your routing for your network? do voice packets pass through any other devices? |
| 17:51.42 | envisean | nope, jsut the phones |
| 17:51.52 | envisean | we have a 10-15 person office |
| 17:52.02 | envisean | so we have one server, handles the routing & voip packets |
| 17:52.13 | pkunkra | envisean. there's your problem. |
| 17:52.26 | pkunkra | buy a dedicated networking device for routing. |
| 17:52.34 | pkunkra | and good high quality one. |
| 17:52.49 | pkunkra | linux machines are very slow at passing packets. |
| 17:52.59 | pkunkra | routers have specialized network stacks to do that. |
| 17:53.30 | pkunkra | a linux box takes about 20 ms to pass a packet usually. |
| 17:53.36 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:53.47 | pkunkra | most routers do it at a speed of 0.1 ms |
| 17:54.06 | pkunkra | and most hardware phones have a jitter buffer of 30-50ms |
| 17:54.21 | pkunkra | so that linux box ate up most of your time already. |
| 17:54.52 | pkunkra | mind you, jitter means packets arriving out of order.... |
| 17:55.16 | pkunkra | but still. that's a lot of time to waste in one machine. leads to jitter quite frequently. |
| 17:55.26 | drmessano | BSD FTW |
| 17:55.29 | drmessano | j/k |
| 17:55.36 | pkunkra | hah |
| 17:55.47 | stansmith | lo? |
| 17:56.30 | Havokmon | shit I can't get asterisk to reply to my aastra registrations in 5 seconds :P Maybe it should be compiled for IOS ;) |
| 17:56.30 | drmessano | BSD has the best network stack out there, pkunkra.. It actually removes so much latency, it needs to add some in to pad the CRC checkers |
| 17:56.33 | drmessano | ~sike |
| 17:56.33 | jbot | [Sike!] The preceeding statement was intented to be humourous, and may have involved false statements used in a humorous manner, or an exaggeration of a real life situation with humourous intent, but in no way should be taken as pure fact or not considered questionable at best. |
| 17:56.35 | envisean | pkunkra: I see what you're saying, but I should still have hte linux box handle the routing? |
| 17:56.46 | *** join/#asterisk Urge4Vert (n=kbmetz@raccoon.omeninc.com) |
| 17:57.12 | Urge4Vert | I'm running asterisk 1.6, I'm trying to figure out how I can route calls on Zap channels 1-4 to ring group 600 and Zap channels 5-6 to ring group 700 without using DID or CallerID |
| 17:57.21 | envisean | pkunkra: err I mean voice packets? |
| 17:57.29 | pkunkra | envisean, no. for "significant" networks, routing should always be handled by a dedicated device. |
| 17:57.39 | *** join/#asterisk Bananaskin (n=Banana@user-54415fea.l5.c5.dsl.pol.co.uk) |
| 17:57.54 | [TK]D-Fender | Urge4Vert: Set each of your zaptel channels into whatever context you want in zapata.conf |
| 17:57.55 | envisean | pkunkra: I feel ya, but in the mean time, is there a way I can perform qos on the network? |
| 17:58.06 | pkunkra | now, the voice packets will eventually go to the pbx since its the pbx. but it shouldn't be making routing decisions. |
| 17:58.29 | Urge4Vert | [TK]D-Fender, then route the contexts I'm guessing? |
| 17:58.37 | pkunkra | you can buy a cheaper dedicated device for $200. |
| 17:58.47 | pkunkra | don't go with the cheapo home routers. |
| 17:58.58 | pkunkra | those are just linux kernels in a fancy box |
| 17:59.06 | pkunkra | they have the same issue. |
| 17:59.21 | pkunkra | but, you can do qos on linux. |
| 17:59.23 | [TK]D-Fender | Urge4Vert: You don't "route" contexts. for analog cards calls come in against "s" on whatever context you specify. From there do whatever you want with them. |
| 17:59.36 | pkunkra | i don't remember how to do it off the top of my head. |
| 18:00.02 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 18:00.13 | pkunkra | a good pix 501/506 is $400-$900. |
| 18:00.14 | envisean | pkunkra: what's one that you've used in the past that has worked well? |
| 18:00.17 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 18:00.19 | envisean | pix? |
| 18:00.22 | envisean | is that a cisco router? |
| 18:00.28 | [TK]D-Fender | PIX? GOOD? |
| 18:00.29 | Urge4Vert | [TK]D-Fender, gotcha. Thanks for the info |
| 18:00.30 | [TK]D-Fender | lol |
| 18:00.35 | pkunkra | that's probably one of the best choices you can make for a small branch office/home network. |
| 18:00.43 | pkunkra | a pix is a firewall |
| 18:01.07 | pkunkra | tkd-fender, what's wrong with a pix? |
| 18:01.22 | pkunkra | my 515e is one of the best damn devices i ever got. |
| 18:01.32 | pkunkra | never had one issue in nearly three years of operation |
| 18:02.05 | envisean | this is some serious equipment we're getting into |
| 18:02.07 | [TK]D-Fender | pkunkra: Horrible with NAT & VoIP |
| 18:02.38 | pkunkra | tkd-fender. |
| 18:02.42 | pkunkra | i have no issues with it. |
| 18:03.07 | envisean | pkunkra: alright, this device looks amazing, is there anything we can do in the meantime |
| 18:03.13 | envisean | pkunkra: before we get a dedicated routing machine? |
| 18:03.18 | *** join/#asterisk bkw_ (n=brian@adsl-70-234-180-217.dsl.tul2ok.sbcglobal.net) |
| 18:03.39 | pkunkra | envisean, it depends on what would be a satisfactory solution. |
| 18:03.50 | envisean | pkunkra: simply giving priority to voice packets during routing |
| 18:04.02 | envisean | pkunkra: I applied some tc rules |
| 18:04.08 | envisean | pkunkra: but it doesn't seem to help out too much |
| 18:04.08 | pkunkra | how overloaded is your linux box? |
| 18:04.19 | pkunkra | is it lightly loaded? |
| 18:04.28 | envisean | pkunkra: extremely lightly loaded |
| 18:04.42 | envisean | pkunkra: we have a 2mbps/2mbps connection in the office, only around 10 connections |
| 18:05.00 | envisean | pkunkra: and on average we're not maxing out bandwidth, only when ppl here have to download large attachments for media work |
| 18:05.07 | pkunkra | envisean, then qos won't help you. qos is mean to give priority to traffic when there is a load. |
| 18:05.13 | pkunkra | meant |
| 18:05.26 | pkunkra | but since there is no load, qos won't make much of a difference. |
| 18:05.27 | envisean | pkunkra: I understand that, our voip packets are fine when ppl aren't downloading though |
| 18:05.38 | envisean | pkunkra: which is why I'm thinking qos is the way to go |
| 18:05.38 | *** join/#asterisk amir_ (n=amir@gentoo/developer/amir) |
| 18:05.55 | *** part/#asterisk amir_ (n=amir@gentoo/developer/amir) |
| 18:06.00 | pkunkra | what kind of "downloading" are we talking about here? |
| 18:06.11 | pkunkra | like normal web usage for 10 people? |
| 18:06.20 | envisean | pkunkra: no, large 1gb files |
| 18:06.25 | envisean | pkunkra: maxing the 2mbps down |
| 18:06.31 | pkunkra | oh |
| 18:06.38 | pkunkra | yes. qos will probably help you then. |
| 18:06.54 | pkunkra | i don't know how to do it with ipchains. |
| 18:06.54 | Havokmon | Or use getright and limit your dl speed to 1.5 |
| 18:06.54 | envisean | plus, people in here love youtube |
| 18:07.15 | envisean | Havokmon: hahaha I'd love to hear the complaints in the office after making that move |
| 18:07.20 | pkunkra | right, you can also have your folks rate limit the download on their own boxes. |
| 18:07.38 | pkunkra | that's probably the easiest and least technical solution. |
| 18:07.42 | envisean | Havokmon: 1.5mbps ? |
| 18:07.53 | Havokmon | envisean: yeah :) |
| 18:07.55 | pkunkra | but involves the most amount of "police" work |
| 18:08.00 | envisean | pkunkra: but i still want full bandwidth when possible |
| 18:08.01 | Havokmon | If it's just you, regulate yourself ;) |
| 18:08.09 | envisean | Havokmon: lol it's not just me |
| 18:08.13 | envisean | Havokmon: i wish i could do that |
| 18:08.54 | pkunkra | google: qos ipchains |
| 18:09.21 | Havokmon | grab arnos firewall script, there's some basic QOS in there |
| 18:09.41 | Havokmon | and it's a decent example overall |
| 18:10.44 | envisean | pkunkra: ok, browsing around |
| 18:11.03 | pkunkra | envisean, keep in mind that regular boxes like linux and windows are horrible inefficient at things like passing packets |
| 18:11.30 | drmessano | I will say this |
| 18:11.41 | *** join/#asterisk jmacz (n=jmacz@190.90.35.212) |
| 18:11.46 | drmessano | Switching from some OTHER linux solutions to OpenWRT was a HUGE increase here |
| 18:12.07 | pkunkra | for a linux box, raising an interrupt when a packet arrives is a very very expensive. 1 gb files will have the machine spending more time in interrupt than doing actual work. |
| 18:12.10 | envisean | pkunkra: hmm unfortunately no good results |
| 18:12.14 | drmessano | So even in the same "family", there's differences |
| 18:13.29 | pkunkra | envisean, google: qos iptables ? |
| 18:13.33 | envisean | pkunkra: so you're saying that end result is that bandwidth will be affected regardless? |
| 18:13.44 | envisean | pkunkra: if I use qos on the same router box? |
| 18:14.10 | Havokmon | <PROTECTED> |
| 18:14.13 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 18:14.15 | *** join/#asterisk angasule (n=angasule@190.49.223.115) |
| 18:14.57 | drmessano | RTP is going to be your issue |
| 18:14.57 | pkunkra | bandwidth probably won't be affected *too* much. tcp allows a large window before it starts retransmitting packets, the linux box will have responded in time to those packets. however... |
| 18:15.14 | pkunkra | running voice through that box will result in a noticable difference. |
| 18:15.24 | pkunkra | you may be able to mitigate it someone with qos |
| 18:15.37 | *** part/#asterisk angasule (n=angasule@190.49.223.115) |
| 18:15.43 | pkunkra | but i highly doubt you will be able to completely solve it |
| 18:16.10 | pkunkra | someone -> somewhat* |
| 18:17.46 | Havokmon | I've got an * 1.2.26 box CentOS4, that randomly doesn't respond to aastra re-registrations within 5 seconds, which causes the aastras to say 'No Service' until the * decides to respond in another second or two. 11 phones, no calls, no load. Any guesses? |
| 18:18.43 | envisean | pkunkra: ok, so I think i know that for the longterm solution we should definitely get a separate routing/firewall box |
| 18:18.56 | envisean | pkunkra: i guess for today we need to get qos on the ipchains |
| 18:19.09 | envisean | pkunkra: have you ever done this yourself? |
| 18:19.43 | pkunkra | envisean, no. i have not done it myself. |
| 18:20.10 | pkunkra | never bothered with learning it. |
| 18:20.42 | pkunkra | i went straight to the cisco equiment for my needs. |
| 18:21.10 | Havokmon | envisean, get the arno iptables firewall script |
| 18:21.26 | *** join/#asterisk tobias (n=tobias@cpe-066-026-085-055.nc.res.rr.com) |
| 18:21.28 | Havokmon | There's qos in there |
| 18:22.54 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:22.59 | envisean | arno iptables ok, googling |
| 18:27.19 | envisean | Havokmon: I just glanced thru the site, doesn't mention anything w/ qos |
| 18:32.23 | Havokmon | envisean: It's in there - I run it to get things started, and modify it when needed |
| 18:32.35 | Havokmon | I added this: $IPTABLES -t mangle -A OUTPUT -o $interface -p udp --dport 5060 -j TOS --set-tos Maximize-Throughput |
| 18:32.44 | Havokmon | There's about 15 of those lines in the script |
| 18:35.31 | ccvp | what's a good, high quality, app to virtually mount bin/cue/iso files in ubuntu? |
| 18:38.35 | Katty | O HAI i upgraded your pbx |
| 18:38.59 | *** join/#asterisk markgreene (n=markgree@130.160.45.30) |
| 18:39.15 | ccvp | http://www.slysoft.com/en/virtual-clonedrive.html |
| 18:39.17 | ccvp | may run that in wine |
| 18:39.21 | markgreene | Is anyone in here running a Redfone Fonebridge? |
| 18:41.16 | *** join/#asterisk esaym (n=user@72.183.198.134) |
| 18:46.19 | *** join/#asterisk hfb (n=hfb@pool-72-67-142-193.lsanca.dsl-w.verizon.net) |
| 18:46.37 | *** join/#asterisk b1shop (n=b1shop@c-24-7-202-70.hsd1.il.comcast.net) |
| 18:47.13 | ccvp | wtf, i just randomly remembered a weird dream i had last night . . . odd.......... i walked into my basement, and saw a secret trap door, that was covered by cob webs, and below that basement, was like a basement from the 1700's.....old ancient valuables......there were 250 yr old vultures , that were albino and still alive . . . .besides valuable stamps and other items, that were long lost.... |
| 18:48.31 | ccvp | a hidden basement, below a basement |
| 18:49.18 | Katty | did you find cisco equipment down there too? |
| 18:49.37 | ccvp | heh |
| 18:49.53 | Katty | i'm trying to understand what an 'agent' is. |
| 18:50.00 | envisean | Havokmon: can't I just run that command without arno's proj though? |
| 18:50.42 | Katty | jbot: agent? |
| 18:50.42 | jbot | Newsflex has a free-agent like interface, but without requiring Gnome or KDE, you can find it at freshmeat |
| 18:50.55 | Katty | jbot: you, sir, are not helpful. |
| 18:50.56 | jbot | Katty: I think you lost me on that one |
| 18:51.01 | ccvp | cdemu.sourceforge.net - a decent package? |
| 18:51.03 | Katty | jbot: clearly. |
| 18:51.32 | *** join/#asterisk yahy1 (n=yahya@ner-as4884.alshamil.net.ae) |
| 18:51.58 | Katty | jbot: /AGENT |
| 18:52.02 | Katty | jbot: AGENT/ |
| 18:52.11 | Katty | jbot: updatedb! |
| 18:52.16 | Katty | jbot: bad :< |
| 18:52.16 | jbot | Bad :<, bad! No cookie for you! |
| 18:53.06 | *** join/#asterisk guillote_GNU (n=guillote@host158.190-31-83.telecom.net.ar) |
| 18:53.34 | Katty | Qwell: teach jbot agent = http://www.voip-info.org/wiki/index.php?page=Asterisk+Agent+channels |
| 18:55.10 | yahy1 | Hi, I have problem in my dial plan. ${CALLERIDNUM} is not being set in a sip channel, although I have usecallerid=yes and hidecallerid=no in sip.conf |
| 18:55.54 | Katty | yahy1: can you confirm you're recieving it? |
| 18:56.13 | yahy1 | I put this in my dial plan: exten => 611,n,SayDigits(99${CALLERIDNUM}88) and it just says 9988 |
| 18:56.31 | *** join/#asterisk quigon (n=matias@32.59.64.130) |
| 18:56.41 | yahy1 | I'm dialing between two local sip extensions |
| 18:56.57 | _ShrikE | yahy1: try ${CALLERID(num)} |
| 18:57.13 | yahy1 | ok |
| 18:58.26 | yahy1 | _ShrikE: no, still the same thing. is it enough if I type 'reload' at the console, or do I have to restart asterisk? |
| 18:58.59 | _ShrikE | reload is fine |
| 18:59.05 | *** join/#asterisk dexpdx (n=dexpdx@66-162-134-242.static.twtelecom.net) |
| 18:59.36 | dexpdx | anyone feel like giving me a little guidance for compiling thrid party applications into asterisk |
| 19:00.21 | Katty | what's everyone's opinon of Agents.. |
| 19:00.30 | Katty | i would think, after awhile, having to log into a phone would get horribly annoying. |
| 19:00.37 | dexpdx | As long as they are not named "Smith" |
| 19:01.03 | Katty | the timeout default is what, 15 minutes? |
| 19:01.05 | WilliamK | dexpdx, you been watching Matrix to much :) |
| 19:01.06 | Katty | i've not made a call in 3 hours |
| 19:01.30 | Katty | perhaps more for a call center, where you get to go pick a room each day |
| 19:01.39 | Katty | everyone gets issued an ID and password |
| 19:01.44 | dexpdx | Katty: having agents actually log into a phone provides metrics on agent productivity |
| 19:01.44 | _ShrikE | yahy1: can you pastebin the relevant portions of your sip.conf? |
| 19:01.55 | dexpdx | if you don't care about those sort of things then it doesn't really matter |
| 19:01.57 | Katty | dexpdx: more so than accountcodes? |
| 19:02.00 | yahy1 | ok |
| 19:02.08 | Katty | dexpdx: oh but we do. |
| 19:02.21 | Katty | dexpdx: they want to do some call barging and phone login stuff for our telemarketing team |
| 19:02.24 | *** join/#asterisk jameswf-home (n=james@mobile-166-214-065-249.mycingular.net) |
| 19:02.25 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 19:02.30 | dexpdx | then have them log in and implement call states like "ACW" |
| 19:02.31 | Katty | dexpdx: but i'm thinking it will be more annoying than productive |
| 19:02.55 | dexpdx | that way you can measure how much time they spend working/closing a call in relation to how much time they are actually talking on the phone |
| 19:03.39 | Katty | what do you mean "working/closing"? |
| 19:04.08 | Katty | i don't think i follow. |
| 19:04.13 | dexpdx | account managers won't think it's annoying when they want a report telling them how much time their agents spent on break, bathroom, etc |
| 19:04.23 | dexpdx | ACW = After Call Work |
| 19:04.33 | Katty | hmm. |
| 19:04.40 | dexpdx | it's the time after an agent hangs up with a customer and before they are ready to go back into their inbound queue |
| 19:04.42 | Katty | so it tells you the time spent between calls too |
| 19:04.50 | dexpdx | "paperwork" |
| 19:04.51 | Katty | oh |
| 19:04.58 | Katty | this works with queues too |
| 19:05.01 | dexpdx | no the time between calls would be queue time |
| 19:05.04 | dexpdx | or idle time |
| 19:05.05 | Katty | that never even crossed my mind. |
| 19:05.10 | Katty | all our calls are outgoing, not incoming. |
| 19:05.26 | Katty | that makes sense. |
| 19:05.35 | Katty | the info goes into the regular call log stuff? |
| 19:05.57 | dexpdx | outbound call centers don't really need logins unless they are using it to match against payroll |
| 19:06.15 | Katty | here our telemarketers must make 25 calls an hour |
| 19:06.20 | Katty | or they get set back to minimum wage |
| 19:06.25 | yahy1 | _ShrikE: here it is http://pastebin.ca/923651 |
| 19:06.33 | Katty | but they all have their own desk |
| 19:06.37 | dexpdx | Katty: that sounds pretty shitty |
| 19:06.37 | Katty | but they work in shifts too... |
| 19:06.50 | Katty | dexpdx: yeah, it is kinda crappy.... but they get other perks |
| 19:06.58 | Katty | dexpdx: for every appointment they make, they get 50 bucks |
| 19:06.59 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 19:07.21 | Katty | dexpdx: and comissions off stuff, if it sells too |
| 19:07.54 | Katty | it would be nice for the girls to be able to share a phone, but not a voicemail account... |
| 19:08.36 | dexpdx | well then you would have to setup logins |
| 19:08.42 | Katty | yeah. |
| 19:08.46 | [TK]D-Fender | Katty: use separate line appearances or fudge the CID and give 2 extensions that ring that phone |
| 19:08.55 | dexpdx | the fact that you are running in shifts without dedicated logins + voicemail means you have todo it |
| 19:08.57 | Katty | [TK]D-Fender: but iw ant to play with agents |
| 19:09.12 | dexpdx | [TK]D-Fender: that won't work if they move desks everyday |
| 19:09.13 | [TK]D-Fender | Katty: Doesn't matter. |
| 19:09.30 | Katty | [TK]D-Fender: of course it matters. |
| 19:09.36 | Katty | [TK]D-Fender: agents would be perfect. |
| 19:09.44 | [TK]D-Fender | Katty: You log in your agent and that dials the appropraite resource. that the phone is shared can be accounted for in your dialplan |
| 19:09.54 | Katty | exten => 110,Dial(AGENT/Katty!) |
| 19:10.32 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 19:11.01 | [TK]D-Fender | Katty: Yes, and that Agent will trigger ANOHER channel which will do something else real.... that is the point where you can still send the call to a "shared" phone. |
| 19:11.34 | Katty | well at that point you're really not sending it to a phone |
| 19:11.41 | Katty | you're sending it to an agent |
| 19:11.45 | Katty | the agent can be anywhere |
| 19:12.03 | Katty | it'd be like me changing my SIP logins everytime i moved. |
| 19:12.08 | Katty | except hella easier. |
| 19:12.30 | grandpapadot | hella |
| 19:12.42 | Katty | yes. |
| 19:12.42 | jameswf-home | ~hella |
| 19:12.43 | jbot | i guess hella is lame |
| 19:12.43 | Katty | hella. |
| 19:12.50 | jameswf-home | hmmm |
| 19:13.02 | jameswf-home | lmao |
| 19:13.24 | cpm | hella lame |
| 19:14.01 | jameswf-home | whew I am on my way home made it through South Carolina and no one lost an eye.... |
| 19:14.26 | Katty | ..yet |
| 19:14.31 | Katty | you keep poking fun of me and you might loose one. |
| 19:14.37 | Katty | OR WORSE |
| 19:14.55 | cpm | hella worse |
| 19:15.10 | jameswf-home | cpm did it |
| 19:15.25 | Katty | jameswf-home: i hope you have children. |
| 19:15.32 | [hC] | jameswf-home: hey, you're coming back from the freepbx conference right? |
| 19:15.32 | jameswf-home | I didnt make fun btw it was jbot |
| 19:15.41 | grandpapadot | CP/M |
| 19:15.46 | grandpapadot | My C128 ran that. |
| 19:15.58 | grandpapadot | It was hella cool. |
| 19:16.01 | jameswf-home | yes sitting in the charleston thing they call an airport |
| 19:16.07 | *** join/#asterisk BBHoss (n=hoss@c-71-207-220-138.hsd1.al.comcast.net) |
| 19:16.22 | Katty | [TK]D-Fender: so... |
| 19:16.27 | Katty | [TK]D-Fender: if you have a SIP phone registered as 110 |
| 19:16.28 | [hC] | jameswf-home: did you see the "microsoft response point" emulated "magic button" demoed by ethans? |
| 19:16.45 | Katty | [TK]D-Fender: you can have an AGENT as 110 too right? |
| 19:16.46 | iCEBrkr | [hC]: I take home a Starbox tonight. whooope! |
| 19:16.48 | iCEBrkr | ie |
| 19:16.58 | [hC] | iCEBrkr: nice. i want to talk to you tonight then :) |
| 19:17.03 | jameswf-home | noh that was after lunch and I had to leave to here due to security and all :( |
| 19:17.08 | iCEBrkr | haha |
| 19:17.11 | iCEBrkr | About? |
| 19:17.25 | [hC] | jameswf-home: aw, that sucks. I really want a copy of it, and supposedly it was being given out at the conference but i couldnt make it. |
| 19:17.27 | iCEBrkr | [hC]: You just want to steal our technologiez! |
| 19:17.32 | grandpapadot | WTF is this Microsoft SIP server anyway? Is it even a phone system? I swear I can't figure it out by their website ... |
| 19:17.32 | [hC] | iCEBrkr: I'm curious about whats inside. |
| 19:17.36 | iCEBrkr | lol |
| 19:17.39 | [TK]D-Fender | Katty: Sure |
| 19:17.46 | [hC] | iCEBrkr: man, we're not even in the same country! |
| 19:17.55 | [hC] | iCEBrkr: I share secrets with other people even in my city. the pie is big enough :) |
| 19:17.56 | iCEBrkr | [hC]: Ohhhh! Here we go! |
| 19:18.16 | [hC] | I just finished speaking with aastra about a new product they're releasing |
| 19:18.18 | [hC] | Very interesting stuff |
| 19:18.23 | jameswf-home | grandpapadot: Its the best thing ever ask em... it will revolutionize the industry like no other product has oh wait... |
| 19:18.27 | iCEBrkr | f Astra |
| 19:18.43 | iCEBrkr | Astra needs to work on quality before they get a new product. |
| 19:18.56 | [hC] | Thats what i just finished talking to them about |
| 19:18.57 | [hC] | :) |
| 19:18.58 | grandpapadot | About 1 out of every 15 we see are DOA, Aastra 480's. |
| 19:18.59 | jameswf-home | aastra has this quad dect phone thing that looks neat |
| 19:19.00 | iCEBrkr | haha |
| 19:19.10 | [hC] | Our rep is very cool |
| 19:19.12 | iCEBrkr | grandpapadot: exactly. |
| 19:19.17 | [hC] | He gives me prerelease firmware and beta products so i can rip them apart |
| 19:19.23 | [hC] | I bitched about the rubber buttons. alot. |
| 19:19.27 | iCEBrkr | hehe |
| 19:19.36 | [hC] | and alot of the stupid bugs |
| 19:19.45 | iCEBrkr | [hC]: Now that I'm back in the Asterisk scene, I'm going a head and writing my desktop call manager. |
| 19:19.56 | [hC] | iCEBrkr: can i give you feedback on it and test it? |
| 19:19.58 | iCEBrkr | [hC]: Especially now since I have the 'hard' stuff done. |
| 19:20.03 | iCEBrkr | [hC]: Sure. |
| 19:20.04 | jameswf-home | I met an astra guy at the confrence.. hi car was broken in to and they stole his gps.... nice litle town.. |
| 19:20.10 | [hC] | iCEBrkr: or is it going to be owned by s2s and all hush hush? |
| 19:20.19 | [hC] | iCEBrkr: I should really start a conversation with kristian about sharing ideas. |
| 19:20.24 | iCEBrkr | [hC]: I had a prototype in the works years ago, But HD crashed, I lost the HD, it was in VB6 (gayness). |
| 19:20.27 | iCEBrkr | etc. etc. |
| 19:20.31 | Katty | if a phone is registered as a SIP extension, how do you force an agent to login before they can dial out? |
| 19:20.32 | iCEBrkr | [hC]: I still have screenshots of the prototype though. |
| 19:20.40 | Katty | just tell the SIP phone it has no dialout abilities in sip.conf? |
| 19:20.49 | grandpapadot | ilbc sounds sooooo much better than g729 it seems. |
| 19:21.06 | iCEBrkr | [hC]: The conversations about desktop software for clients is all negative. |
| 19:21.13 | iCEBrkr | [hC]: I believe this call manager will be my own baby. |
| 19:21.31 | [hC] | iCEBrkr: cool. I have some ideas for how to make desktop software -really- cool. |
| 19:21.34 | iCEBrkr | [hC]: Which I think is a big mistake. |
| 19:21.50 | [hC] | iCEBrkr: at the end of the day, im sure you will be able to show s2s and be like.. oh yeah? dumb idea? really? and they'll bite. |
| 19:21.54 | iCEBrkr | [hC]: Yeah, I have some ideas and a good base to work from. |
| 19:21.56 | grandpapadot | We have an Ajax based call manager that works pretty good w/AMI on 1.2 |
| 19:22.01 | iCEBrkr | [hC]: hahaha |
| 19:22.06 | iCEBrkr | [hC]: I'll license it to them! |
| 19:22.06 | [hC] | grandpapadot: public? |
| 19:22.19 | grandpapadot | Part of our service offering ... |
| 19:22.29 | [hC] | iCEBrkr: good idea :) I have put a lot of thought into what a call manager would be like, but IANAC :( |
| 19:22.53 | iCEBrkr | Web/Flash based Call Managers = Epic fail. |
| 19:23.08 | [hC] | well, they could be good.. but the existing ones suck. |
| 19:23.10 | iCEBrkr | People need to give up the idea of runing a browswer to see shit going on |
| 19:23.22 | iCEBrkr | FOP = lame. |
| 19:23.29 | iCEBrkr | Totally lame. |
| 19:23.37 | [hC] | FOP is only lame because of the UI, i think. It -could- be nice. |
| 19:23.39 | iCEBrkr | What enterprise company would use such a thing? |
| 19:23.44 | Katty | i like it. |
| 19:23.46 | Katty | it's pretty. |
| 19:23.50 | Katty | i even have a cute puppy picture on it. |
| 19:23.57 | Katty | but it's prone to crashing. |
| 19:24.03 | [hC] | especially if you did some sort of Air platform, where you could run it as a desktop app too, or a web app |
| 19:24.05 | Katty | and even deadlocking *, and that makes me sad. |
| 19:24.05 | *** join/#asterisk kannan (n=kannan@219.64.114.217) |
| 19:24.20 | [hC] | Ive never had a FOP die like that |
| 19:24.24 | [hC] | I have an install of 200+ seats using it |
| 19:24.26 | Katty | i have! |
| 19:24.29 | [hC] | its not my favorite, but its what ive got |
| 19:24.34 | Katty | [hC]: me too. |
| 19:24.41 | Katty | [hC]: tho, i am playing around a bit with HUDlite |
| 19:24.51 | Katty | [hC]: having problems with the install :/ |
| 19:24.57 | iCEBrkr | [hC]: It can be so much better. |
| 19:25.00 | [hC] | hudlite also sucks in its own regard. |
| 19:25.07 | [hC] | the dependency on an irc server is... wtf. |
| 19:25.22 | [hC] | and the UI is at first glance "wooo shiny!" but after you try to use it, its ridiculous |
| 19:25.28 | grandpapadot | FOP and hudlite do suck. |
| 19:26.17 | cpm | I thought hudlite was a fonality specific thingee |
| 19:27.10 | Havokmon | envisean: You should be able to, I wasn't sure if TOS was a script-created jump to point. This script has / |
| 19:27.18 | kannan | hi all. I hve a digium e-1 card for PRI line. I have 1 primary number from the telco , and 5 DIDs. I am able to pass the callerid of the DID from the respective internal phnes from a hardware Samsung PBX , but in Asterisk not able to. I have used Set(CALLERID(num)=60607070) in the outbound calling context in extensions.conf |
| 19:27.29 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 19:27.49 | Qwell | kannan: chances are, your Samsung isn't sending all 8 digits |
| 19:27.52 | Havokmon | envisean, not has, uses, Maximize-Throughput or Minimize-Delay |
| 19:28.51 | kannan | Qwell :i see ok , i will call the telco and ask for the ID length then? |
| 19:29.41 | cpm | Katty, does it actually work? |
| 19:29.44 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 19:30.22 | Katty | cpm: baroo? |
| 19:30.29 | Katty | cpm: FOP? |
| 19:30.33 | cpm | hudlite |
| 19:30.33 | *** part/#asterisk pwd5 (n=feng@MTRLPQ02-1177745966.sdsl.bell.ca) |
| 19:30.45 | cpm | I know fob works, |
| 19:30.50 | cpm | fop rather |
| 19:30.59 | Katty | cpm: i have problems with the install: http://angela.sleekgeek.org/2008/02/28/hudlite-installation/ |
| 19:31.03 | cpm | I just wasn't all that impressed iwth it. |
| 19:31.07 | Katty | cpm: despite the fact i don't know what i'm doing... |
| 19:31.22 | Katty | cpm: that's about as far as i got before getting distracted by agents., |
| 19:32.15 | cpm | aiie! agents! hella bad! |
| 19:32.16 | Katty | cpm: it told me to copy files over to the asterisk directory. i couldn't find the files they spoke of. they also talked about users.xml, which was not on my machine. at all. |
| 19:32.19 | cpm | well, there's the hud server |
| 19:32.21 | Katty | mayhaps i was supposed to make it. |
| 19:32.27 | Katty | oh well. |
| 19:32.34 | Katty | a topic for another time. |
| 19:32.39 | cpm | which seems, , well, it isn't exactly clear, looks like a nasty kludge |
| 19:32.52 | Katty | there's not much documentation |
| 19:32.55 | Katty | and what i could find, was for redhat |
| 19:32.56 | jameswf-Airport | yum over 3g holy crap |
| 19:34.08 | outtolunc | cntrl-c see if you can hit a faster repo |
| 19:34.33 | outtolunc | (it isn't always your end that is slow) |
| 19:36.22 | *** join/#asterisk DrBadly (n=retloc@199-117-163-66.dia.static.qwest.net) |
| 19:36.55 | DrBadly | How do I set in my dialplan if a number is 303 or 720 to use desired trunk |
| 19:36.56 | Katty | hmm. |
| 19:37.03 | *** join/#asterisk gormux (n=julien@was59-1-82-226-112-70.fbx.proxad.net) |
| 19:37.03 | Katty | so does an Agent have to wait for a call to go to them? |
| 19:37.05 | gormux | hello |
| 19:37.11 | Katty | or can they login, and make outgoing calls? |
| 19:37.38 | gormux | i have asterisk installed, but i've had big problems on the server after an update |
| 19:37.39 | *** join/#asterisk SteveTotaro (n=Elizabet@c-69-243-124-5.hsd1.md.comcast.net) |
| 19:37.47 | gormux | and nobody can login |
| 19:37.59 | *** join/#asterisk kannan (n=kannan@219.64.114.217) |
| 19:38.04 | gormux | sip show peers tells me that there are no users |
| 19:38.28 | gormux | so it seems the users from sip.conf are not added |
| 19:38.34 | SteveTotaro | reload |
| 19:38.37 | gormux | and i have nothing in the logs |
| 19:38.52 | gormux | i already reloaded |
| 19:38.57 | *** join/#asterisk nirz (n=nir@IGLD-83-130-229-206.inter.net.il) |
| 19:39.13 | gormux | the logs show nothing at all, no new lines since this morning |
| 19:39.23 | SteveTotaro | reboot |
| 19:39.36 | gormux | hum, why not |
| 19:39.43 | DrBadly | where can I find documentation on 91NXXNXXXXXX patterns? |
| 19:39.54 | *** join/#asterisk stansmith (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 19:43.02 | gormux | hum no, still the same problem |
| 19:43.17 | [TK]D-Fender | DrBadly: What do you mean by "91NXXNXXXXXX" patterns? |
| 19:44.24 | gormux | hum, strange, sip port is opened |
| 19:44.32 | gormux | but asterisk does not start |
| 19:45.03 | gormux | well, it starts, but dont write logs |
| 19:45.03 | [TK]D-Fender | gormux: Running any other SIP osftware on your server? |
| 19:45.06 | gormux | nope |
| 19:45.08 | DrBadly | Dial Patterns |
| 19:45.23 | DrBadly | im trying to say if CID = 303 or 720 areacode use this trunk |
| 19:45.26 | *** part/#asterisk Bananaskin (n=Banana@user-54415fea.l5.c5.dsl.pol.co.uk) |
| 19:45.35 | [TK]D-Fender | DrBadly: Go read the chapter called "Dialplan Basics" : |
| 19:45.37 | [TK]D-Fender | ~book |
| 19:45.37 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 19:45.40 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^ |
| 19:45.42 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 19:45.45 | *** join/#asterisk ruied (n=ruied@bl10-127-164.dsl.telepac.pt) |
| 19:46.28 | [TK]D-Fender | DrBadly: You don't check for a CID pattern match in the pattern, you'd do it in the code of the exten itself |
| 19:46.42 | [TK]D-Fender | DrBadly: Via GotoIf |
| 19:47.01 | stansmith | how would a young developer go about detecting when a user keys "#"? |
| 19:47.12 | stansmith | this is all hypothetical of course |
| 19:47.20 | Katty | [TK]D-Fender: i understand that agents are usually waiting for a call from the queue.. |
| 19:47.22 | envisean | Havokmon: do you have any good arno tutorials? |
| 19:47.27 | Katty | [TK]D-Fender: is there a way to...not put them in a queue? |
| 19:47.39 | Katty | [TK]D-Fender: or put them in an outgoing queue only? where they just get to dial out |
| 19:47.39 | envisean | Havokmon: i'm reading the help section right now, but doesn't mention anything about qos w/ the commands u wrote earlier |
| 19:47.52 | Katty | [TK]D-Fender: i dunno if that makes any sense. |
| 19:48.21 | Katty | [TK]D-Fender: i want the telemarketers to login to their phone after picking an office, and making their outgoing calls. but they won't be waiting for any incoming ones... and new office next day. |
| 19:48.36 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:49.27 | [TK]D-Fender | Katty: That would not have anything to do with "Agents" then. You'd do your own code to ID the phone they are using as "logged in as them". |
| 19:49.40 | [TK]D-Fender | Katty: And your own code to log out, etc. |
| 19:49.50 | [TK]D-Fender | Katty: And could restrict whatever you felt like for them. |
| 19:49.57 | [TK]D-Fender | Katty: All jsut basic dialplan |
| 19:52.39 | SteveTotaro | Ok, I have done the impossible. I have a quad port Junghanns BRI card and a Sangoma A200 (FXO) in the same box, this is in the US. everything works except dialing out the BRI I always get " Ext: 1 Cause: Call Rejected (21), class = Normal Event (1) ] |
| 19:52.39 | SteveTotaro | 1 -- Processing IE 8 (cs0, Cause) |
| 19:52.39 | SteveTotaro | 1 q931.c:4482 q931_receive: call 136 on channel 1 enters state 0 (Null) |
| 19:52.40 | SteveTotaro | 1 -- Restarting T203 counter |
| 19:52.40 | SteveTotaro | <PROTECTED> |
| 19:53.17 | stansmith | is there a log file for this channel some where online? |
| 19:53.32 | stansmith | i posted a pastebin link last night from home, deleted the code when i got to work, and i need to look at it again |
| 19:53.41 | SteveTotaro | it get's indexed so it must be |
| 19:53.56 | SteveTotaro | comes up in google searches sometimes |
| 19:54.51 | JenniferAkemi | <PROTECTED> |
| 19:55.48 | stansmith | JenniferAkemi: OMG THANK YOU |
| 19:56.04 | JenniferAkemi | stansmith: np. i just scrolled up ;) |
| 19:56.14 | *** join/#asterisk Tourinho (n=tourinho@201.37.118.16) |
| 19:56.16 | SteveTotaro | anyone know why i would get this " Ext: 1 Cause: Call Rejected (21), class = Normal Event (1) ]" on a bri? |
| 19:56.22 | Tourinho | good day all |
| 19:57.06 | Tourinho | can Asterisk place a call (using dial app) and send audio through that channel (and get dtmf too)? |
| 19:58.16 | lirakis | later all |
| 19:58.26 | *** part/#asterisk lirakis (i=lirakis@pr0tected.us) |
| 19:59.26 | mvanbaak | SteveTotaro: I have no idea |
| 19:59.39 | mvanbaak | SteveTotaro: bri is a mistery to me |
| 20:00.25 | SteveTotaro | it is so silly! |
| 20:00.39 | SteveTotaro | bri intense debug span 1 |
| 20:01.51 | SteveTotaro | ~pb |
| 20:01.51 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 20:02.22 | SteveTotaro | http://pastebin.ca/923713 |
| 20:03.13 | Havokmon | envisean: I don't have any tutorials for arno's script.. I've bee using iptables/ipchains (not advanced) for years, and I just inserted that line into the script, based on the other qos lines that were already in there. |
| 20:03.37 | JenniferAkemi | is there anyway to know what .conf files i need? i want to clean up my installation |
| 20:03.58 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 20:04.12 | dexpdx | Quick: what's everyones favorite TTS engine? |
| 20:04.39 | [hC] | cepstral |
| 20:04.52 | SteveTotaro | depends what modules you are loading |
| 20:04.56 | SteveTotaro | jennifer |
| 20:05.05 | [hC] | there's an allison cepstral module now |
| 20:05.07 | [hC] | kinda creepy! |
| 20:05.08 | Katty | [TK]D-Fender: neat. |
| 20:05.19 | dexpdx | Cepstral 8khz sounds a lot better on the website then through Swift |
| 20:05.20 | JenniferAkemi | SteveTotaro: right now I have autoload = yes. is that generally bad? |
| 20:05.24 | Katty | [TK]D-Fender: so you put a group of SIP phones with the ability to own dial 1 number. their login number. |
| 20:05.29 | SteveTotaro | she has a nice module |
| 20:05.39 | Katty | [TK]D-Fender: they login as... |
| 20:05.53 | Katty | [TK]D-Fender: i guess that would set their accountcode |
| 20:05.57 | SteveTotaro | jennifer, autoload is not "generally bad" |
| 20:06.15 | Katty | [TK]D-Fender: how would set a user's accountcode change their ability to dialout? |
| 20:06.17 | envisean | Havokmon: alright i'm going to poke through the script |
| 20:06.24 | Katty | [TK]D-Fender: do you literally change the phones context on the fly or something? |
| 20:06.34 | envisean | Havokmon: I definitely don't want to just throw this script up today, it seems like I have to disable a few things on my system in order for this to work |
| 20:06.35 | SteveTotaro | but "best practice" would be to autoload but noload the modules you don't use |
| 20:06.40 | JenniferAkemi | SteveTotaro: ok |
| 20:06.49 | JenniferAkemi | SteveTotaro: I'll try to figure out which ones i don't need :D |
| 20:06.59 | Havokmon | envisean: yeah I would definitely do this after hours :) |
| 20:07.11 | SteveTotaro | just rename the confs to conf.old and try to start asterisk |
| 20:07.22 | JenniferAkemi | yeah good idea. |
| 20:07.25 | SteveTotaro | if you don't use iax, don't load it |
| 20:09.31 | envisean | Havokmon: haha k |
| 20:09.36 | *** join/#asterisk VaNNi (n=VaNNi@216.70.165.200) |
| 20:09.51 | JenniferAkemi | if i have autoload=yes, do i need to have load => res_musiconhold.so ? |
| 20:10.01 | JenniferAkemi | isn't that included in the autoload? |
| 20:10.26 | envisean | Havokmon: crap, I don't have a log of that iptables line u wrote up now |
| 20:10.29 | justdave | Is the zaptel 1.4.9.2 tarball really missing zaptel.h or did I somehow get a bad copy? |
| 20:11.42 | justdave | It seems to compile fine on its own, which makes me suspect it was purposely removed, but it breaks things that build on top of it (like wanpipe) |
| 20:12.31 | SteveTotaro | the new wanpipe does not touch zaptel |
| 20:12.38 | SteveTotaro | zaptel 1.4 that is |
| 20:12.44 | Havokmon | I added this: $IPTABLES -t mangle -A OUTPUT -o $interface -p udp --dport 5060 -j TOS --set-tos Maximize-Throughput |
| 20:12.46 | jameswf-Airport | autoload=yes ; set this to yes and Asterisk will load any modules it finds in /usr/lib/asterisk/modules |
| 20:12.56 | justdave | wanpipe 3.2.3 is current, right? |
| 20:13.05 | justdave | it's asking me for zaptel source when I try to install it |
| 20:13.31 | jameswf-Airport | zaptel.h should be put in /usr/include/zaptel |
| 20:14.06 | RypPn | justdave: wanipe wont build with anything higher than 1.4.8 atmo |
| 20:14.28 | justdave | it's finding my old zaptel 1.4.5 source, but refusing to touceh 1.4.9.2 |
| 20:14.28 | justdave | yeah, what ryppsaid I guess |
| 20:14.32 | Havokmon | envisean, $IPTABLES -t mangle -A OUTPUT -o $interface -p udp --dport 5060 -j TOS --set-tos Maximize-Throughput |
| 20:14.52 | *** join/#asterisk ddunavant (n=David@pool-71-191-30-72.washdc.east.verizon.net) |
| 20:14.55 | Katty | [TK]D-Fender: i don't suppose you have a wiki page that would show some sort of useful information, if Agent isn't what i'm looking for |
| 20:15.01 | jameswf-Airport | RypPn: as far as i am aware sangoma did the patches for 1.4.8 all the major vendors got notice and corected like a month ago |
| 20:15.08 | justdave | while I'm at it, I hate the wifi on Mac OS X 10.5.2 |
| 20:15.21 | RypPn | which is why I said higher |
| 20:15.24 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 20:15.28 | generalhan | hey all ! |
| 20:15.28 | dexpdx | anyone have any outgoing spool generaters that would allow me to keep a pri saturated for Nth amount of time? |
| 20:15.30 | Havokmon | envisean, replace --dport 5060 with --dport 10000:20000 for rtp as well |
| 20:15.31 | justdave | laggy as hell when I'm 2 feet from the access point :) |
| 20:15.43 | _ShrikE | anyone using the IP04? |
| 20:15.46 | stansmith | ~t |
| 20:15.47 | jbot | t is probably ibot's idiotic master from hell that uses Wind0ze... |
| 20:15.48 | dexpdx | ./out_loadtest -channels 10 -t 3600 |
| 20:16.00 | jameswf-Airport | ~tsa |
| 20:16.23 | justdave | I see a wanpip-2.3.4-16 on the sangoma website, but it's listed as old for some reason, despite the higher version number than the "current" one |
| 20:16.31 | jameswf-Airport | ooh I should add switches to my load tester.... |
| 20:16.33 | stansmith | michels? |
| 20:17.09 | jameswf-Airport | just a thought maybe faster to call sangoma support... |
| 20:17.25 | generalhan | i am building a 1.4.18 test machine right now to upgrade my 1.2.x machine, and i want to know what i can do with dynamic queues, or whatever it is referred to in 1.4, can some one point me in the right direction so i can read more about how to accomplish this, and what all i can do with it ? |
| 20:17.31 | jameswf-Airport | tats kinda what they dp |
| 20:17.38 | jameswf-Airport | *thats |
| 20:17.43 | jameswf-Airport | *do |
| 20:17.44 | jameswf-Airport | bah |
| 20:18.31 | Katty | Does anyone know of a wiki page that kinda shows how a user would login and authenticate from a SIP phone and then be able to make outgoing calls with an accountcode attached to it? |
| 20:19.13 | SteveTotaro | a combination of authenticate, disa, and account code |
| 20:19.48 | Katty | SteveTotaro: and it would be like logging in as an agent, but not waiting for a call from the queue? |
| 20:20.11 | SteveTotaro | disa gives you a dialtone |
| 20:20.18 | Katty | neat. |
| 20:20.26 | Katty | i'll wiki that |
| 20:21.14 | SteveTotaro | make sure it is secure, disa is a great way to get toll frauded |
| 20:21.16 | jameswf-Airport | `disa |
| 20:21.24 | jameswf-Airport | ~disa |
| 20:21.24 | jbot | it has been said that disa is direct inward system access. show application disa |
| 20:21.44 | Katty | SteveTotaro: the examples from wiki are from what looks like from the PSTN |
| 20:21.52 | Katty | SteveTotaro: i only want internal going out... |
| 20:21.54 | envisean | Havokmon: do I have that in a second line? |
| 20:21.56 | Katty | SteveTotaro: but i'll keep digging |
| 20:22.08 | Katty | SteveTotaro: but i guess... |
| 20:22.12 | SteveTotaro | you just have to put it in the right place |
| 20:22.20 | Katty | SteveTotaro: it doesn't have to originate from the PSTN, i could keep it totally internal. |
| 20:22.23 | Katty | SteveTotaro: yeah (= |
| 20:22.32 | SteveTotaro | you got it |
| 20:22.48 | Katty | but an agent can login and it times out after 15 minutes, and logs them out |
| 20:22.51 | SteveTotaro | now help me with my bri issue |
| 20:22.55 | SteveTotaro | http://pastebin.ca/923713 |
| 20:23.04 | Katty | with disa and accountcodes, wouldn't you have to put in your password for every single numer you dialed? |
| 20:23.31 | SteveTotaro | maybe i am not clear on what you are trying to do |
| 20:23.45 | Katty | random telemarketer picks an office and picks a phone. |
| 20:23.51 | Katty | they login to that phone with a userid and password. |
| 20:23.51 | SteveTotaro | please help me |
| 20:24.26 | [TK]D-Fender | Katty: No, there is no guide for what you want, this is jsut you using probably some sort of DB (I might use AstDB personally) in all of your outbound extens that your "agents" will be using to dial. |
| 20:24.27 | Katty | the operator can dial them by their 'agent' number. tho apparently what i'm wanting isnt' an agent. |
| 20:24.32 | [TK]D-Fender | Katty: Really simple stuff. |
| 20:24.50 | Katty | and then they just make phone calls out till they logout |
| 20:24.56 | generalhan | anyone in here using RealTime Queues ? im looking for a document somewhere to show me a sample table of how their queue is setup in the table |
| 20:25.27 | JenniferAkemi | sounds like the hotdesk feature in the book? |
| 20:26.16 | Havokmon | envisean: Yes. What I pasted was a command from the script. $IPTABLES should be replaced with the path to iptables if you're running it in the shell |
| 20:26.28 | [TK]D-Fender | JenniferAkemi: "hotdesk" isn't a feature IMO, its just dialplan flow decision. |
| 20:26.37 | JenniferAkemi | ok |
| 20:26.54 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:26.56 | JenniferAkemi | but if katty reads it through it could help |
| 20:26.57 | Havokmon | envisean: I would separate each rule, rather than trying to bundle a bunch into a single rule (not even sure if you could) |
| 20:26.58 | Katty | well maybe next year i'll know enough to tackle that one then. |
| 20:27.10 | JenniferAkemi | i guess odbc_func is the feature |
| 20:27.54 | Katty | i think i'm just going to give up for the day. |
| 20:29.10 | SteveTotaro | katty the answer is easy |
| 20:29.19 | Katty | to me it's not (= |
| 20:29.26 | Katty | i'm not even remotely familiar with databases |
| 20:29.30 | Katty | so it looks like this one's out for awhile. |
| 20:29.50 | jameswf-Airport | wooohooo off to ATL |
| 20:30.14 | Katty | SteveTotaro: and the only hotdesk reference i could find uses agi. and i'm not sure how to use those either. |
| 20:31.17 | JenniferAkemi | you dont need the db |
| 20:31.51 | JenniferAkemi | have a good flight james |
| 20:34.39 | stansmith | "t" extension sucks, no? |
| 20:35.12 | mvanbaak | it's for 'timeout' |
| 20:35.21 | *** join/#asterisk bkruse (n=bkruse@216.207.245.1) |
| 20:35.21 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 20:36.09 | stansmith | i thought it stood for "time to leave" |
| 20:36.24 | mvanbaak | yeah, that's what you can call it as well |
| 20:36.46 | ccvp | Is there a way to limit the amount of digits dialed? For example I have a * box set up for the department of corrections for prisoners to call home. It has the Digium 2 FXO/ 2 FXS card in it. I have two Lines brought in to the fxo ports and 2 standard 2500 analog sets for the prisoners to use to dial out. Everything seems to work great. We use * to record all the calls. After the prisoner dials the original number and their PIN, I do not want them to be able to se |
| 20:36.55 | ccvp | anymore DTMF tones. |
| 20:37.18 | ccvp | The PIN number is not processed by *. It is processed by the LEC's switch. |
| 20:37.55 | SteveTotaro | PIN number is redundant |
| 20:38.43 | SteveTotaro | i do not see any way you can limit the number of dtmf digits sent though |
| 20:39.29 | ccvp | heh, I saved a prison in a near town |
| 20:39.32 | ccvp | $250,000 |
| 20:39.36 | ccvp | instead of going with a bellsouth phone system |
| 20:39.47 | Katty | JenniferAkemi: i don't suppose you have any sort of documentation, wiki link, or anything on this, do you? |
| 20:40.10 | JenniferAkemi | katty: on the hotdesk thing from the book? |
| 20:40.27 | stansmith | ~book |
| 20:40.27 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 20:40.28 | Katty | JenniferAkemi: i don't know what that is. in reference to my 'agent' login and get to dial out sort of thing. |
| 20:40.31 | *** join/#asterisk lisandropm (n=lisandro@190.1.22.76) |
| 20:40.37 | stansmith | i forgot my book at home, sorry |
| 20:41.10 | JenniferAkemi | read from page 274 |
| 20:41.34 | Katty | thanks. |
| 20:41.42 | JenniferAkemi | it probably does more than you need |
| 20:41.46 | JenniferAkemi | but its' good reading anyways :) |
| 20:41.48 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 20:45.15 | *** join/#asterisk etfonhomey (n=chatzill@74-143-192-66.static.insightbb.com) |
| 20:48.17 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:48.22 | LemensTS | Anyone read Asterisk Future of Telephony 1st edition AND 2nd edition? |
| 20:48.57 | stansmith | leif madsen maybe has |
| 20:49.11 | Qwell | stansmith: nah |
| 20:49.18 | stansmith | well nevermind then |
| 20:49.27 | stansmith | is there some trick to get the "t" extension to work? |
| 20:49.58 | LemensTS | stansmith: If it timesout it will look for it |
| 20:50.42 | stansmith | yea, thats what i thought, but this is my first time actually trying to use it and it isnt getting called |
| 20:50.48 | [TK]D-Fender | stansmith: It works based on the timeout you set |
| 20:51.00 | [TK]D-Fender | stansmith: PASTEBIN is your friend. |
| 20:51.02 | LemensTS | stansmith: you must set the timeout in that context first |
| 20:53.34 | stansmith | http://www.pastebin.ca/923757 |
| 20:53.39 | *** join/#asterisk gutzb4ll (n=mark@gateway.meteor-web.com) |
| 20:53.42 | gutzb4ll | hello |
| 20:54.11 | gutzb4ll | this is driving me crazy, why is the uniqueid variable as referenced in my dialplan not containing the same data as the uniqueid variable in my CDR records? |
| 20:54.44 | gutzb4ll | erm, my bad. i see |
| 20:54.51 | JenniferAkemi | why? |
| 20:55.54 | justdave | ok, I successfully compiled wanpipe against zaptel 1.4.8, then installed zaptel 1.4.9.2 overtop of it |
| 20:55.57 | justdave | seems to work fine |
| 20:56.05 | [TK]D-Fender | stansmith: And your test call? |
| 20:56.18 | justdave | as someone mentioned earlier, wanpipe appears to not even touch zaptel now, it just wants to know that it's there |
| 20:56.27 | gutzb4ll | well, then i guess my dilemma is that there appears to be no way for me to correlate between the a CDR record and the Unique ID of the call in the dialplan |
| 20:56.29 | stansmith | runs s,1 -> s,8 |
| 20:56.32 | gutzb4ll | is there a way? |
| 20:56.34 | justdave | so based on that premise I was assuming it would probably be okay |
| 20:56.34 | JenniferAkemi | is wanpipe onl7y for sangoma? |
| 20:56.55 | justdave | dunno, sangoma is what I'm using it on. |
| 20:57.18 | justdave | I hate their setup script. |
| 20:57.51 | justdave | only have the one asterisk box with a sangoma card in it, the rest are all Digium cards |
| 20:57.58 | [TK]D-Fender | stansmith: get rid of "autofallthrough=yes" , and S,6 - S,8, and put your count check into "t". Also set your timeouts at the start of your "s" |
| 20:58.03 | justdave | kernel upgrades are so much easier on the boxes with Digium cards |
| 20:58.39 | Katty | JenniferAkemi: so the phones tell the database their location. |
| 20:58.45 | tzanger | justdave: true, but honestly how often are you upgrading kernels on production machines? |
| 20:58.49 | Katty | JenniferAkemi: and asterisk consults the database before sending the call? |
| 20:58.59 | tzanger | Katty: get your PRI up? |
| 20:59.06 | [TK]D-Fender | JenniferAkemi: Yes |
| 20:59.12 | justdave | tzanger: about once a month for the last year or so. |
| 20:59.13 | Katty | tzanger: no cable yet. |
| 20:59.17 | JenniferAkemi | katty: yeah. your guys don't really need that part do they? they're doing outbound only? |
| 20:59.29 | justdave | they keep finding exploits in the kernel, and these boxes are internet facing |
| 20:59.30 | tzanger | you should try connecting them with good vibrations |
| 20:59.30 | JenniferAkemi | katty: the authenticataion etc isuseful for you though i think |
| 20:59.45 | JenniferAkemi | katty: sorry typos... hands are SO COLD |
| 21:00.03 | Katty | JenniferAkemi: yes. outbound only. I know how to use authenticate...but, only in the area of everytime the a call is placed, prompting for a password. |
| 21:00.04 | tzanger | justdave: "they keep finding exploits in the kernel" -- I can't think of a single remote exploit that was affected by the kernel |
| 21:00.29 | gutzb4ll | does anyone know the answer to my question? |
| 21:00.31 | tzanger | one or two local privelige escalations, yes |
| 21:00.32 | lnx | can u tell me please which variable contains the return code {-1;0} example in case of $AGI->send_text('You've got mail!'); ? |
| 21:00.41 | tzanger | but not a signle remote exploit (root or other user) |
| 21:00.49 | justdave | most of the recent stuff has been local privilege escalation, but there's other things besides asterisk on these, too, and there are shell users |
| 21:00.53 | gutzb4ll | is there any good way to corrilate between the uniqueid of a call as expressed in the dialplan during a call and the corrisponding call's CDR record? |
| 21:01.23 | justdave | regardless, I like to stay up-to-date anyway. |
| 21:01.29 | Katty | JenniferAkemi: after reading this, i still don't understand how they could login and make phone calls, etc. |
| 21:01.33 | JenniferAkemi | katty: Also, the boss wants to be able |
| 21:01.33 | JenniferAkemi | to track when they are in the office, and control calling privileges from those phones |
| 21:01.33 | JenniferAkemi | when no one is there |
| 21:01.36 | stansmith | lnx: -1,0 is what gets returned from "$AGI->send_text(..)" |
| 21:01.38 | Katty | JenniferAkemi: it seems like it would have to change the SIPs info on the fly. |
| 21:01.40 | justdave | and since these are office servers, it's easy enough to reboot them at night when nobody's in the office :) |
| 21:02.07 | stansmith | so its up to you to assign it, or just use that statement in an If statement (as an example) |
| 21:02.32 | lnx | stansmith: can i put in a variable? like $? in bash ? |
| 21:02.34 | [TK]D-Fender | Katty: No, you just make an exten they dial to "log in" to that phone. |
| 21:03.00 | [TK]D-Fender | Katty: When they do that you set the DB value that will permit the phone to actually make it to the outside. This involves no SIP trickery whatsoever |
| 21:03.03 | stansmith | lnx: what language are you writing the AGI in? |
| 21:03.10 | lnx | stansmith: perl |
| 21:03.16 | stansmith | uhm, yea you should be able to |
| 21:03.25 | Katty | [TK]D-Fender: but it involves Databases. |
| 21:03.27 | stansmith | $status = $AGI->send_text(..) |
| 21:03.29 | cesar_CR | [hC], how are you man!! |
| 21:03.33 | Katty | [TK]D-Fender: and that's even harder than sip trickery for me. |
| 21:03.35 | lnx | stansmith: k, ty |
| 21:03.37 | stansmith | should work..i think...in php it would |
| 21:03.37 | [TK]D-Fender | Katty: In some capacity, yes |
| 21:03.39 | [hC] | cesar_CR: good! did you get your thing working? |
| 21:03.56 | JenniferAkemi | katty: you can do it :) |
| 21:03.58 | [TK]D-Fender | Katty: This is about 1/2 dozen lines of dialplan TOTAL |
| 21:04.15 | JenniferAkemi | katty: just read it over and over and understand every lien that the dialplan is doing |
| 21:04.52 | Katty | i'll look at it again tomorrow. my brain is fried and i don't understand half of what i'm reading. |
| 21:04.53 | JenniferAkemi | katty: that chapter (12) also has step by step instructions to get a database working (that will store if the agent is logged in or not) |
| 21:05.19 | cesar_CR | [hC], nop |
| 21:07.07 | [hC] | guys, when you save a file in ulaw format, is that file not essentially a 8khz mono wav? |
| 21:07.15 | stansmith | [TK]D-Fender: thanks a lot ... i actually had to keep the WaitExten() |
| 21:07.28 | [TK]D-Fender | stansmith: Not the way I was suggesting. |
| 21:08.24 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 21:08.37 | teknoprep | does anyone here have a cisco net contract for 79xx phones ? |
| 21:08.40 | teknoprep | i need 8.8 firmware |
| 21:09.05 | stansmith | http://www.pastebin.ca/923772 <-- works, albeit some tidying up i need to do...really though, thanks for the style pointers |
| 21:09.15 | stansmith | [TK]D-Fender: ^ |
| 21:11.09 | *** join/#asterisk mitcheloc (n=mitchel@216.207.245.1) |
| 21:11.35 | [TK]D-Fender | stansmith: Oh, you don't need waitexten,... what you needed to do was set "autofallthrough =no". Not to remove the line itself, jsut stop saying "yes" |
| 21:12.29 | stansmith | o... i know removing it defaults to "yes", i was kinda wondering why to do that |
| 21:12.31 | [TK]D-Fender | stansmith: And you should do "Set(TIMEOUT(response)=10)" , "Set(TIMEOUT(digit)=3)" and the like for this IVR |
| 21:12.44 | *** part/#asterisk mitcheloc (n=mitchel@216.207.245.1) |
| 21:13.31 | stansmith | :-/, im replacing existing IVR's that use Voxeo with asterisk+cepstral...im trying to follow exactly what we already have to an exact T..even though i know there are better ways to do it |
| 21:14.17 | [TK]D-Fender | stansmith: I wasn't suggesting changing the functional flo so much as putting the code in the right place for the ends you want out of it |
| 21:15.49 | stansmith | i see what your saying now, by declaring those in the entry context, it will be available throughout the dialplan, unless overridden later on |
| 21:22.19 | [TK]D-Fender | stansmith: Should only be done in an IVR itself |
| 21:22.54 | stansmith | thats all * is doing for me |
| 21:26.01 | [TK]D-Fender | ok, well its checkout time over here, BBL |
| 21:36.39 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 21:48.52 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:52.50 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
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| 21:58.49 | justdave | sangoma just emailed me a pre-release version of wanpipe that correctly recognizes zaptel 1.4.9.x |
| 22:00.30 | justdave | so it exists, guess you just have to ask for it currently |
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| 22:03.27 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 22:04.43 | lnx | can you check http://pastebin.com/m7db06ad6 please. I got syntax from wiki :] please help me to correction |
| 22:23.09 | scooby2 | why does * make me want to go postal? |
| 22:25.26 | Qwell | speaking of postal... |
| 22:25.34 | LemensTS | 200 concurrent G729 calls over internet. 18kbps * 200 = 3600kbps. 3.6mb * 8 = 28megabit/s.......does that sound right??? |
| 22:25.56 | Qwell | my cards were deliver. yay! |
| 22:26.00 | Qwell | thanks for reminding me scooby2 :D |
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| 22:27.04 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 22:54.48 | TJNII | LemensTS: Use 1024 instead of 1000 when converting for computing applications. |
| 22:56.12 | TJNII | Also are you sure the 18kbps isn't already in bits? |
| 22:58.44 | *** join/#asterisk angryuser (n=angyuser@df01t2-212-194-105-124.d4.club-internet.fr) |
| 22:58.48 | angryuser | hi |
| 22:59.09 | TJNII | 200 G729E calls: 12.4 kilo-bits/second * 200 = 2480 kilo-bits/second / 1024 kilo/mega approx= 2.43 mega-bits per second |
| 22:59.26 | angryuser | fender you here ? |
| 22:59.41 | TJNII | Bitrate found at http://www.adaptivedigital.com/product/vocoders/g729.htm |
| 23:00.01 | TJNII | angryuser: I think he's AFK. |
| 23:01.20 | angryuser | heh, has anyone configured sipproxd and asterisk ? |
| 23:02.32 | angryuser | my head is going to blow, asterisk and sip proxy is such a pain |
| 23:05.14 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-183-109.hsd1.or.comcast.net) |
| 23:07.05 | drmessano | lol |
| 23:08.08 | angryuser | http://sourceforge.net/forum/message.php?msg_id=4520699 read this drmessano have fun :) |
| 23:09.05 | angryuser | i have exact problem, and i finally found the cause, do you remember i tol you i had 3 account on same server ? |
| 23:09.38 | drmessano | yeah |
| 23:09.44 | *** part/#asterisk Hyphenex (n=Hyphenex@60-241-73-120.tpgi.com.au) |
| 23:09.55 | drmessano | Tell me again why you're using Siproxd |
| 23:09.55 | angryuser | firs register all teh rest bye bye |
| 23:10.43 | angryuser | ah, well replacing my fvs 338 with unstable vpn + load balancing and traffic shaping |
| 23:11.17 | drmessano | ok |
| 23:12.04 | angryuser | pfsense got a good traffic shaping utils, openvpn integrated, ect but uses masquarade firewall (port changing) |
| 23:12.28 | angryuser | i got 3 peers only, an it will be a painnnnnn |
| 23:13.25 | angryuser | at leas i am sure it's not me |
| 23:13.26 | drmessano | I just know others using Asterisk + Pfsense without siproxd |
| 23:13.30 | angryuser | *now* |
| 23:13.49 | angryuser | yes, with only one sip device ? |
| 23:13.54 | *** join/#asterisk coolhp (n=Toshiba@modemcable239.6-80-70.mc.videotron.ca) |
| 23:13.59 | coolhp | Hi all... |
| 23:14.24 | angryuser | or one, outbound peer .. |
| 23:14.27 | drmessano | So the 3 devices are on the inside and connect to a remote asterisk box? |
| 23:14.41 | coolhp | Just out of curiosity, is anyone have problems with the "hit # for transfer" feature when passing "Tt" as options in the Dial command ? |
| 23:15.06 | coolhp | I no longer seem to be able to trigger the transfer feature... I can still trigger automon though. |
| 23:15.38 | angryuser | i got ~20 users,10 astribank, 10 sip, internal lan >>asterisk >>outbound peers(3) |
| 23:16.04 | drmessano | So the issue is the outbound peers |
| 23:16.05 | drmessano | ? |
| 23:16.16 | angryuser | each peer = 2 comms=6 comms max, inbound from misdn |
| 23:16.42 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 23:17.29 | angryuser | yes, as it it written in post, asterisk mess up, "Asterisk does the matching of SIP traffic to its SIP peers" like sip proxy so.. |
| 23:17.37 | drmessano | ok |
| 23:18.05 | angryuser | hm, i will install ipcop, whatever |
| 23:18.15 | angryuser | but pfsense is better |
| 23:18.34 | drmessano | I dunno.. I know a few guys that run pfsense that had multiple ITSP accounts, etc, on the outside with no siproxd |
| 23:19.00 | drmessano | and had remote phones connecting from the outside into asterisk |
| 23:19.09 | angryuser | i will test it tomorrow again (people angry :) ) who knows |
| 23:19.22 | *** join/#asterisk puzzled (n=patrick@53533DDB.cable.casema.nl) |
| 23:19.52 | drmessano | Alright.. time for dinner.. bbiab |
| 23:20.04 | angryuser | write a feedback maybe, pfsense manual incomplete..... |
| 23:22.33 | coolhp | do I have to set DYNAMIC_FEATURES=blindxer for blind transfer to work when passing Tt do Dial ? |
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