| 00:07.32 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 00:07.49 | Micc | Is there a way to see who you called on an aastra 6757i? |
| 00:08.04 | Micc | I can see who called me, but isn't there a called list too? |
| 00:10.39 | NateHB | Micc, it is the redail button |
| 00:10.53 | NateHB | redail |
| 00:10.57 | *** join/#asterisk hopper75 (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 00:11.02 | NateHB | why cant i spell dial today |
| 00:11.46 | NateHB | [TK]D-Fender: figured out that shit |
| 00:12.30 | NateHB | [TK]D-Fender: I had some kind of intrusion |
| 00:13.36 | Micc | I know about the redial, but I want a list of number dialed and the time spent on the call. |
| 00:25.30 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 00:25.44 | nix8n82 | Micc, your asterisk log files should have the information |
| 00:27.00 | Micc | yeah, I know. my customer wants to be able to look it up on the phone. |
| 00:27.13 | Micc | they are a law office, they charge for every second you talk to them, you know. |
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| 00:30.33 | paulc | Micc: What about a script that emails them a call notification at the end of the call.. then they could assign it to a client code etc |
| 00:30.44 | TJNII | Why not offer to build a website the secretaries can use? It would probably be easier for you than making the phone do what they want and you can spin it as less trouble for the hot-shot lawyers. |
| 00:31.53 | TJNII | Ooh, I like the email idea, too. |
| 00:31.59 | Micc | paulc, thats an interesting idea. They only need it at the end of the day I think. |
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| 01:00.02 | paulc | Micc: delayed reply - people at my desk... if you can get them to punch account codes, you could send them a list of all calls, with account codes, at the end of the day.. ready to be handed to a lacky to enter into their billing system etc |
| 01:01.25 | TJNII | I think a intranet wobsite is better if you're going to pitch the lackey / secretary angle. Though the email doesn't open itself to "we want it to look THIS way" nearly as bad as a site does. |
| 01:05.36 | KavanS | TJNII, what? |
| 01:06.10 | TJNII | KavanS: Eh? |
| 01:06.37 | KavanS | sorry missed your convo - was trying to figure out what you were tryign to do |
| 01:06.39 | KavanS | just curious |
| 01:07.11 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 01:07.51 | TJNII | KavanS: Micc is setting up a phone system for a legal practice where calls are billable hours. He was wondering about how to get a phone to give a list of calls and durations. paulc and I were offering alternatives. |
| 01:08.16 | TJNII | Paul is pitching emails and I'm pitching a website with a call listing. |
| 01:14.41 | *** part/#asterisk nny (~Scott@cpe-071-076-058-253.sc.res.rr.com) |
| 01:21.24 | kc8pxy | wow.. never had that before.. had my console up, and it looks liek someone did the voip equivilent to a port scan.. looks like they tried to register all of the 4-digit possible extensions |
| 01:22.54 | TJNII | I've had that happe |
| 01:22.58 | TJNII | *happen |
| 01:23.08 | TJNII | 4 or 5 times. It is fairly common. |
| 01:23.27 | TJNII | Now my firewall only allows VoIP connections from approved subnets. |
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| 02:37.46 | boodu | re |
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| 02:41.14 | WowFactor | I have a VPS with spry.com , I am looking for a VOIP provider to make mostly domestic calls |
| 02:41.28 | WowFactor | I do not like Voicepulse at all, is there another alternative? |
| 02:42.03 | TJNII | ~itsp |
| 02:42.04 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 02:42.37 | WowFactor | ~itsplist-us |
| 02:42.38 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 02:43.35 | TJNII | I use Broadvoicem I've used Vitelity as a backup, and prople here are always chattering about voip.ms |
| 02:47.46 | WowFactor | What country is .ms ? |
| 02:48.46 | ChannelZ | Montserrat? |
| 02:49.11 | WowFactor | So voip.ms is in montserrat? |
| 02:49.18 | ChannelZ | no |
| 02:49.45 | TJNII | Looks like they have offices in Canadia and Mexico. |
| 02:50.11 | ChannelZ | Taxas |
| 02:50.14 | ChannelZ | err Texas |
| 02:50.14 | Nugget | Don't mess with Texas. |
| 02:50.47 | ChannelZ | but yes they have POPs all over |
| 02:51.09 | WowFactor | so their selection of .ms is just for vanity purposes? |
| 02:51.16 | TJNII | tom@eServer0 ~ $ whois voip.ms connect: Connection refused |
| 02:51.21 | TJNII | Well that's useful. |
| 02:52.46 | ChannelZ | I don't know what the significance of .ms is to them |
| 02:52.59 | *** join/#asterisk mpdavis73 (~mpdavis73@c-66-177-190-186.hsd1.fl.comcast.net) |
| 02:53.14 | ChannelZ | But people can sell domains in their TLD to anyone. Like .tv |
| 02:54.03 | TJNII | ChannelZ: Wait, are you implying that the internet transcends physical boundaries? |
| 02:54.08 | TJNII | I don't believe you. No sir. |
| 02:54.16 | *** join/#asterisk pabelanger-lap (~pabelange@206-248-185-92.dsl.teksavvy.com) |
| 02:55.10 | ChannelZ | And that countries can whore out their domain space to anyone for a buck |
| 02:55.30 | mpdavis73 | heh - .ly lybia |
| 02:55.35 | ChannelZ | The .tv domain might be their only industry... |
| 02:56.40 | ChannelZ | I'm surprised nobody has ladyga.ga |
| 02:57.29 | mpdavis73 | i'm considering asterisk to replace or current hosted internet+voip (smoothstone) |
| 02:57.39 | mpdavis73 | and have some basic ?s |
| 02:58.17 | TJNII | ~book |
| 02:58.17 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 02:58.25 | TJNII | Good starting point ^^^ |
| 02:58.38 | TJNII | You don't have to memorize it, but it is wise to at least skim it. |
| 02:59.03 | TJNII | If you've got any smarts you'll realize which parts are worth reading and which parts ... arn't. |
| 02:59.08 | jql | yes, definitely a good starting point |
| 02:59.11 | mpdavis73 | we currently have Cisco 7961 phones - will they work? |
| 02:59.18 | mpdavis73 | cool, thanks |
| 02:59.27 | jql | I know a scary amount of info from that book, and I still end up mostly fumbling about |
| 02:59.43 | jql | I can't imagine what I'd be like if I didn't know all that stuff. :) |
| 02:59.50 | TJNII | hehehe, but it does cover the basics. |
| 03:00.15 | mpdavis73 | my main concern was admin, we have about 60 users, and i am the only real sysadmin |
| 03:00.22 | jql | cisco phones need to be flashed for sip -- you got that covered? |
| 03:00.34 | mpdavis73 | and i am slammed now without any phone admin |
| 03:00.35 | jql | unless you're gonna try for skinny |
| 03:00.36 | jql | shivers |
| 03:00.54 | mpdavis73 | flashed - via tftp? |
| 03:01.00 | jql | yeah |
| 03:01.04 | mpdavis73 | i'm green in telephony |
| 03:01.15 | mpdavis73 | k, i know a bit of tftp |
| 03:01.21 | jql | in that case, you best hope you inherited phones with sip already |
| 03:01.24 | jql | :) |
| 03:01.39 | mpdavis73 | linksys routers and such |
| 03:02.04 | jql | my main caveat would be to not have any nat between the phones and asterisk |
| 03:02.10 | jql | cisco phones hate, hate, hate nat |
| 03:02.46 | mpdavis73 | that's a completely different concern - we have net through smoothstone as well as voip |
| 03:03.04 | mpdavis73 | mpls network for our various offices |
| 03:03.42 | jql | well... it's possible to make that work |
| 03:03.51 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 03:04.44 | mpdavis73 | we are way overspending right now, T1 to every office, some with only 1 user |
| 03:05.07 | jql | does it at least include internet? |
| 03:05.15 | jql | or is that T1 a telephony T1? heh |
| 03:05.19 | mpdavis73 | i like smoothstone as they handle all admin, but our ceo wants a cheaper alternative |
| 03:05.38 | jql | yeah, my company's working on mpls at the moment too. it's fun |
| 03:05.53 | jql | or working on AND working off |
| 03:05.58 | jql | shrugs |
| 03:06.12 | mpdavis73 | smoothstone has an alt route on the router at each office for phones |
| 03:09.34 | mpdavis73 | one odd thing i noticed with the cisco 7961's - if another device is connected through it, the device behind shows the phone's mac |
| 03:09.55 | mpdavis73 | on a network scan |
| 03:10.42 | mpdavis73 | hosed my wyse thin client admin software that id's by mac |
| 03:13.14 | *** join/#asterisk pabelanger-lap (~pabelange@206-248-185-93.dsl.teksavvy.com) |
| 03:14.25 | mpdavis73 | the good thing is i have already successfully implemented a few open source projects |
| 03:14.35 | mpdavis73 | and now accounting loves me :) |
| 03:19.10 | mpdavis73 | in book, number of channels - does that mean the max number of simultaneous calls? |
| 03:19.54 | mpdavis73 | for system req guidelines |
| 03:22.08 | mpdavis73 | thanks for the starting point, i'm sure i'll be back - luckily, i have some time, about 18 months, before our smoothstone contract is up |
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| 03:56.15 | Asinus1223 | hello |
| 03:56.45 | Asinus1223 | I cant get an analog phone to ring on an TMD400 FSX port |
| 03:57.04 | Asinus1223 | I get a dial tone but no ring when the extension is dialed |
| 03:57.09 | Asinus1223 | this is strange |
| 03:59.31 | TJNII | Asinus1223: Did you plug in the power cable to the card? |
| 04:00.30 | Asinus1223 | yes |
| 04:00.37 | Asinus1223 | the molex extension |
| 04:00.51 | TJNII | Welp, than I'm out of ideas. :P |
| 04:00.52 | Asinus1223 | the phone works but just...won't...ring |
| 04:01.08 | Asinus1223 | thanks tho |
| 04:01.09 | TJNII | Obvious question: Have you tried another phone? |
| 04:01.35 | TJNII | Really obvious question: Is the ringer on? |
| 04:06.15 | ChannelZ | Super obvious question: Is this actually a phone? |
| 04:06.18 | Asinus1223 | Bad phone :( |
| 04:06.21 | Asinus1223 | thanks all |
| 04:06.54 | TJNII | goes *Swish* |
| 04:07.18 | Asinus1223 | 3 days knocking myself out |
| 04:07.28 | Asinus1223 | over a $7 WalMart phone |
| 04:07.35 | TJNII | HALP! I've plugged a banna into my asterisk but It no call! Did Raffi lie to me!? |
| 04:12.35 | *** join/#asterisk kkm (~kkm@76.91.228.152) |
| 04:12.58 | drmessano | BANANA FONE? |
| 04:13.38 | jql | I think you need to use a hamburger phone |
| 04:13.44 | drmessano | ZOMG |
| 04:14.05 | coppice | $7 for a phone that leaves the factory at around $1 is a nice markup :-) |
| 04:14.09 | drmessano | http://www.sourcingmap.com/desk-top-corded-hamburger-telephone-yellow-p-2805.html?utm_source=google&utm_medium=froogle&utm_campaign=usfroogle |
| 04:14.15 | drmessano | HAMBURGER PHONE |
| 04:14.33 | jql | you can has |
| 04:14.39 | coppice | I liked the old duck phone that quacked for its ring |
| 04:14.52 | drmessano | Yeah, those were badass |
| 04:15.23 | drmessano | Duck phone + ATA = well planted hijinx |
| 04:15.45 | drmessano | "WUT IS DAT QUACKIN?" |
| 04:15.51 | drmessano | "I DOT NO?" |
| 04:15.55 | drmessano | :( |
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| 04:56.06 | ChannelZ | argh |
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| 07:15.04 | ascent | points to topic, beta2 for 1.8 is already available. |
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| 07:53.32 | redax | hi |
| 07:54.07 | redax | is it safe to make a cron script which simply deletes older voicemail messages than 1week ? |
| 07:54.10 | *** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net) |
| 07:54.43 | Diffen2 | good morning. when i call a queue in my asterisk i dont get any ring signal. as far as i can understand i should use r to get the asterisk to send ring signal instead of moh. my question is, what are the parameter name that should be infront of the r? |
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| 07:56.48 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 07:57.09 | kaldemar | Diffen2: "core show application Queue" will show you the syntax for the command. |
| 07:57.38 | Diffen2 | kaldemar thanks man |
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| 07:59.44 | *** join/#asterisk G_G (~G_G@41-132-229-142.dsl.mweb.co.za) |
| 08:00.21 | G_G | hi folks, i am having some trouble with asterisk :( what application can i use to play MP3 files? |
| 08:00.31 | G_G | at the moment i am using, Application: MP3Player |
| 08:00.56 | *** join/#asterisk Da-Geek (~Da-Geek@80.235.230.186) |
| 08:03.00 | redax | hm |
| 08:03.39 | redax | contrib/scripts/message-expire.pl is a good tool :D |
| 08:04.07 | G_G | at the moment when i put my .call file into the outgoing folder, it initates the call but hangs up after .5 second ringing |
| 08:04.11 | G_G | and it wont let me answer |
| 08:05.14 | Diffen2 | kaldemar hmmm i dont get it, i guess i should be using r but i cant find out the parameter name. sorry for sucking but i cant figure it out |
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| 08:07.54 | ChannelZ | Do you want to see my butt? Ñ |
| 08:08.18 | kaldemar | Diffen2: what do you mean by parameter name? just add the r option to your Queue command in your dialplan: Queue(yourqueue,r) |
| 08:09.56 | Diffen2 | kaldemar ahh ok i was in queue.conf and tried to find out where to set r |
| 08:10.14 | G_G | URG! |
| 08:10.18 | G_G | this is so annoying |
| 08:10.32 | G_G | can anyone suggest some troubleshooting tips for me? |
| 08:11.59 | *** join/#asterisk BenC[UK] (~BenC_UK_@cpc3-lock1-0-0-cust299.cos2.cable.ntl.com) |
| 08:12.28 | BenC[UK] | does the asterisk IVR system allow integration with third party scripts? |
| 08:13.07 | kaldemar | BenC[UK]: sure. |
| 08:14.12 | kaldemar | BenC[UK]: there really is no separate "IVR system", just dialplan that can do pretty much whatever you want, based on what the user does. |
| 08:14.30 | BenC[UK] | Ok cool.. can I get responses from scripts and store them? |
| 08:15.01 | BenC[UK] | I need to make a system where a customer can dial in, and get status from their account |
| 08:17.14 | Diffen2 | kaldemar thanks man i found it :) |
| 08:18.15 | kaldemar | BenC[UK]: yes you can |
| 08:18.53 | *** join/#asterisk coppice (~chatzilla@m121-202-72-232.smartone-vodafone.com) |
| 08:19.07 | BenC[UK] | thank, i've downloaded asterisk now.. just going to get it running in a virtual pc |
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| 08:27.29 | G_G | is g729 the only codec asterisk supports? |
| 08:28.14 | redax | `core show codecs' |
| 08:28.16 | ChannelZ | no it supports ulaw, alaw, gsm, ilbc... |
| 08:28.44 | ChannelZ | g729 is actually one it doesn't support out of the box, in the sense that you need to buy license(s) for it |
| 08:32.19 | ChannelZ | actually I guess ilbc is separate.. but anyway... sleepy time |
| 08:32.57 | Repzak | Hello, anyone have a clue why i don't get sound between my asterisk and my phone company?, it calls fine in both directions but no sound. when i enable dmz it also works with sound. i have forwarded the RTP ports specified in the RTP.conf. how can i figure out what other ports to open? |
| 08:33.22 | kaldemar | ~sipnat |
| 08:33.23 | infobot | rumour has it, sipnat is Quick guide on configuring Asterisk + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 08:33.29 | kaldemar | Repzak: ^^^ |
| 08:42.42 | *** join/#asterisk gamedna (~Adium@cpe-70-125-155-74.satx.res.rr.com) |
| 08:43.41 | gamedna | what is the preferred way of building asterisk on OS X? |
| 08:45.14 | gamedna | well, recommended way⦠|
| 08:45.31 | gamedna | i have it building under snow lep but lots of the deps are missing.. |
| 08:45.36 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 08:45.50 | gamedna | before i go and hunt everything down myself, wanted to see if there were any suggestions |
| 08:46.58 | Repzak | did as this page: http://www.aocomputing.net/?p=3 but lo luck |
| 08:47.39 | gamedna | huh? |
| 08:48.05 | gamedna | links to sip&nat |
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| 08:57.22 | Repzak | who is deciding the RTP port, the server or the client? |
| 08:57.51 | florz | both, obviously |
| 08:59.20 | Repzak | depending on what way the call goes or? |
| 08:59.23 | kaldemar | Repzak: enable verbosity and sip debug, and pastebin a failed call. |
| 08:59.27 | kaldemar | ~pb |
| 08:59.28 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 09:00.15 | Repzak | no rtp debug? |
| 09:00.30 | florz | Repzak: I said both, not either. |
| 09:01.22 | kaldemar | Repzak: the sip debug shows used rtp ports and addresses. no rtp debug. |
| 09:01.35 | florz | and it's pretty obvious that you can't allocate ports on a remote host |
| 09:02.38 | kaldemar | server decides the used server port and client the used client port. they just send the addresses and ports to each other via SIP. |
| 09:06.48 | Repzak | http://pastebin.com/Y7h2yKUJ |
| 09:10.30 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 09:11.51 | kaldemar | "Kasper" seems to be behind a nat. did you configure so in sip.conf? |
| 09:12.00 | Repzak | yes |
| 09:12.43 | Repzak | nat=yes externip= xxx localnet=10.16.1.0/24 |
| 09:13.01 | Repzak | under general |
| 09:13.18 | kaldemar | you thought you did, but you didn't. |
| 09:13.21 | Repzak | under my 100 extension i have nat=no |
| 09:13.25 | gamedna | anybody have suggestions for building asterisk on OSX? |
| 09:13.33 | kaldemar | pastebin your sip.conf, masking any secrets |
| 09:14.40 | kaldemar | gamedna: http://www.voip-info.org/tiki-index.php?page=Asterisk%20MacOSX%20Support |
| 09:15.29 | Repzak | http://pastebin.com/sQP19q7e |
| 09:15.31 | kaldemar | gamedna: http://www.voip-info.org/wiki/view/Building+Asterisk+on+MacOSX |
| 09:16.29 | gamedna | kaldemar: i have asterisk building on OSX, but i am wondering about how to handle all the deps for various featuers. basically w/o having to hunt down each lib and build them independently |
| 09:17.04 | gamedna | stuff like mysql, libxml, popt, etc |
| 09:17.06 | kaldemar | Repzak: what is 10.16.1.2? |
| 09:17.21 | Repzak | asterisk |
| 09:17.43 | Repzak | local ip |
| 09:18.15 | kaldemar | you need to show what happens before "-- Executing [100@incoming:1] Dial("SIP/46902283-081be728", "SIP/100") in new stack" |
| 09:18.20 | kaldemar | show the whole call. |
| 09:19.19 | *** join/#asterisk Bendbanks (bendbanks@60-241-59-77.tpgi.com.au) |
| 09:21.43 | Repzak | http://pastebin.com/kJUqutcp |
| 09:21.46 | Repzak | complete call |
| 09:25.16 | *** join/#asterisk nextime (~nextime@unaffiliated/nextime) |
| 09:25.53 | nextime | hello all. I have a wct4xxp card ( te405p first generation ) with 4 E1 of different operators plugged in |
| 09:26.15 | nextime | all 4 pri have the same issue: PRI span 1/0: Provisioned, Down, Active |
| 09:26.42 | nextime | loading dahdi modules with debug=1 show me |
| 09:26.43 | nextime | [252481.351981] wct4xxp: LOF/LFA detected on span 1 but debouncing for 2500 ms |
| 09:26.43 | nextime | [252481.352045] wct4xxp: LOS detected on span 1 but debouncing for 5000 ms |
| 09:26.53 | nextime | and also [252481.365112] Detected loss of E1 alignment on span 0! |
| 09:27.30 | nextime | the operators are insisting that they pri are working good, and also using they test device it work |
| 09:27.49 | nextime | any idea on what can be the cause of this lack of alignment of my pri? |
| 09:30.57 | Repzak | hmm.. The phone i am calling from is also on my IP.. must be 87.54.160.2:5060 |
| 09:31.43 | Repzak | it has number 46902283, and calls 46902283 |
| 09:31.54 | Repzak | my asterisk also registers to that number |
| 09:32.44 | Repzak | so it get the call on 87.54.160.2:5061 which my router changes to 10.16.1.2:5060 |
| 09:33.02 | Repzak | byt this seems to work correctly |
| 09:33.18 | Repzak | but the RTP packets seems lost in both directions |
| 09:33.45 | Repzak | i have set an option in musimi called RTP so it forces all data packets arround their server (i think) |
| 09:39.37 | *** join/#asterisk sulex (~sulex@office.blindata.ch) |
| 09:40.20 | *** part/#asterisk gamedna (~Adium@cpe-70-125-155-74.satx.res.rr.com) |
| 09:42.14 | Repzak | kaldemar.. any clues? |
| 09:45.24 | Repzak | it seems the data is on port 46156 when running in DMZ... |
| 09:46.35 | Repzak | how can i control that port, it is outside the RTP range i set in asterisk |
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| 10:11.50 | redax | is it possible to add a voicemail as a queue member ? |
| 10:16.50 | *** join/#asterisk sulex (~sulex@office.blindata.ch) |
| 10:19.56 | Repzak | users |
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| 10:32.36 | kerframil | nextime: I don't have an answer as such, but this may help: http://lists.digium.com/pipermail/asterisk-r2/2009-September/001228.html |
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| 11:03.29 | m1oluf | hi all |
| 11:04.06 | m1oluf | i need some help with hfc based car in dahdi: anyone? |
| 11:04.17 | m1oluf | car=card :) |
| 11:05.05 | *** join/#asterisk king313 (~king313@unaffiliated/king313) |
| 11:06.40 | m1oluf | i need some help with hfc based card in dahdi: anyone? |
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| 11:31.07 | redax | hm. |
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| 11:31.49 | redax | adding voicemail to queues is possible, although I loose the monitor possibilitÃy in queue... |
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| 11:52.42 | redax | don't you know any Asterisk manager event "grep" ? |
| 11:53.03 | redax | like filter some kind of events... |
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| 12:15.27 | Diffen2 | Hello, does anyone know if its possible to remove missed call from a phone when you have a missed call from a queue? Sample, one queue has two agents and when there are an incoming call to the queue it calls on both agents at the same time. One takes the call and the other are stucked with a missed call in the display. |
| 12:15.37 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 12:19.04 | beardy | Depends on the phone, and I think finding one that allow you to control such a thing with an interface other than its buttones will be hard to find.. |
| 12:19.14 | beardy | -e |
| 12:21.52 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 12:22.21 | *** join/#asterisk Thorn (~thorn@unaffiliated/thorn) |
| 12:23.03 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:23.06 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 12:23.11 | Thorn | hello |
| 12:24.07 | Thorn | I've got a queue with rrmemory strategy. is it possible to set a member to only receive calls when other members are busy? |
| 12:24.14 | *** join/#asterisk DogWater (~ddd@dhcp92.cmh.ee.net) |
| 12:24.15 | Thorn | I'm running 1.4 |
| 12:25.09 | [TK]D-Fender | Thorn: PauseQueueMemeber |
| 12:25.15 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 12:25.23 | [TK]D-Fender | Thorn: OOps, forget that |
| 12:25.33 | [TK]D-Fender | Thorn: Use member priorities |
| 12:25.45 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 12:26.19 | Thorn | will they work with rrmemory? documentation only talks about priorities in connection with ringall |
| 12:26.53 | Thorn | I mean this http://www.voip-info.org/wiki/view/Asterisk+call+queues |
| 12:26.53 | [TK]D-Fender | Thorn: Should work fine. |
| 12:27.06 | [TK]D-Fender | Thorn: And the WIKI it not the authority on * |
| 12:27.43 | Thorn | you mean member penalty? |
| 12:28.08 | Thorn | as in member => SIP/200,1 |
| 12:28.16 | [TK]D-Fender | Yes, that |
| 12:28.23 | [TK]D-Fender | mixed the wording |
| 12:28.30 | Thorn | ok I will try that thank you |
| 12:28.38 | Thorn | that page mixes it too |
| 12:28.39 | [TK]D-Fender | hasn't gotten to his first cup of coffee yet... |
| 12:29.11 | Thorn | what's the authoritative documentation on asterisk? always wondered about that :) |
| 12:29.20 | DogWater | offtopic: Is anyone familiar with software for windows called VoIPBox/VoipSwitch? |
| 12:29.52 | beardy | ~book |
| 12:29.53 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 12:29.59 | Chainsaw | DogWater: Running servers on the Windows operating system seems a rather silly idea. Don't you want high uptime for these things? |
| 12:30.50 | [TK]D-Fender | beardy: That book isn't either..... |
| 12:30.58 | Thorn | isn't the book somewhat outdated and non-comprehensive? |
| 12:31.02 | [TK]D-Fender | Thorn: What counts is what's in your tarball |
| 12:31.11 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 12:31.12 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 12:31.17 | [TK]D-Fender | Thorn: Like almost all * documentation :) |
| 12:31.32 | beardy | Then there is none, never, for nothing. |
| 12:32.30 | DogWater | Chainsaw: No I am not the user, we have some users who are installing this software onto our hosting servers (VPS, Dedicated) and they all seem to be getting compromised in short order the only common thread being that this particular software is installed on all of them, so I am trying to find people who have used this product or are aware of it. |
| 12:34.01 | [TK]D-Fender | DogWater: You're asking for a Whopper from a McDonalds with that ;) |
| 12:34.43 | DogWater | Okay, I was just asking since this seems to be the most voice centric place on Freenode that I have found, I appreciate your time. |
| 12:36.43 | drmessano | We hate Windows, in general |
| 12:36.52 | Chainsaw | DogWater: Windows boxes get owned all the time though. You really feel it's specific to that bit of software? |
| 12:37.36 | [TK]D-Fender | I think "Windows" is a common thread too ;) |
| 12:39.34 | DogWater | On a fully updated Windows 2008 R2 node it should be nearly impossible for someone to create a new administrator account and disable the default administrator account remotely without credentials and/or without Malware coming in via IE8/web. |
| 12:39.35 | *** join/#asterisk oktay (~oktay@78.183.151.198) |
| 12:39.57 | drmessano | "My Windows boxes keep getting compromised and they're all running...." "Windows?" "Ok, fine.." |
| 12:40.07 | oktay | anybody familiar with PSTN-To-VOIP on spa/sipura boxes ? |
| 12:40.16 | DogWater | But I don't want to get too much into an OS war here, but thanks for your time. |
| 12:40.20 | [TK]D-Fender | oktay: Many of us. |
| 12:40.25 | oktay | very good. |
| 12:40.33 | drmessano | DogWater: Except for a new vuln? |
| 12:40.43 | oktay | i know you can limit who can use that by caller id, and the rest will pass through |
| 12:41.07 | oktay | but what i want is to dedicate one pstn number for it (actually an extension on my legacy pbx) |
| 12:41.21 | oktay | that extension is 116.. which is the line attached to the spa |
| 12:41.25 | [TK]D-Fender | oktay: Still not too clear. |
| 12:41.39 | oktay | but when I call 130, it still gives me the beep for password to do pstn to voip |
| 12:42.22 | [TK]D-Fender | oktay: I doubt any of us have ever passworded the device itself... |
| 12:42.37 | oktay | you've done Caller ID matching? |
| 12:43.13 | oktay | actually. this has been a bit premature i think.. i will come back after trying a few things. |
| 12:43.17 | oktay | thanks. |
| 12:43.39 | [TK]D-Fender | oktay: Normally we let ALL calls out on it , and all calls in and let * handle everything |
| 12:44.07 | [TK]D-Fender | oktay: If yuo want the device to be smarter... I'd recommend visiting Voxilla's forums |
| 12:44.20 | oktay | will do. i am too confused right now :) |
| 12:47.40 | redax | can I copy files in dialplan? |
| 12:48.13 | *** part/#asterisk ascent (~ascent@schoot.org) |
| 12:48.58 | [TK]D-Fender | redax: System() |
| 12:49.16 | redax | ehh :) thanks |
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| 13:28.28 | silvestre_id | How can i get the agent number who answered a queue call? I'm using AGI with Queue app like: Queue(servicedesk|t||||crm_info.agi) |
| 13:34.30 | *** join/#asterisk mnicholson (~mnicholso@nat/digium/x-ofgasjythjuoaszu) |
| 13:35.30 | [TK]D-Fender | silvestre_id: Get it where? |
| 13:36.04 | silvestre_id | with agi |
| 13:39.16 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 13:39.17 | silvestre_id | If the Asterisk generate a variable indicating who answered the call, I know what it is to get the information by agi |
| 13:40.08 | [TK]D-Fender | silvestre_id: Do an AMI call to see who it is bridged to, or check the queue log. |
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| 13:45.55 | anonymouz666 | coppice: do you know if SIP/RTP can handle the modem communication (ContactID protocol) that uses the phase-shifted analog signals? |
| 13:46.22 | *** join/#asterisk khussein78 (~khussein7@188.225.192.238) |
| 13:46.33 | khussein78 | can i ask about Hylafax here ? |
| 13:46.54 | silvestre_id | I will do something to merge the information from queue_log to AGI. |
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| 13:50.56 | coppice | anonymouz666: what protocol is that? |
| 13:51.46 | anonymouz666 | It is used for panel alarms |
| 13:53.52 | anonymouz666 | there's a handshake between the client and remote panel |
| 13:54.09 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 13:54.52 | anonymouz666 | the handshake consists in: A burst of 1400 Hz. ±3% tone with a duration of 100 msec. ±5% A pause of 100 msec. ±5%. A burst of 2300 Hz. ±3% tone with a duration of 100 msec. ±5% |
| 13:55.52 | anonymouz666 | there's an old thread on asterisk-users that someone states that this protocol (ContactID) won't work with SIP/RTP at all |
| 13:56.00 | anonymouz666 | http://www.mail-archive.com/asterisk-users@lists.digium.com/msg125373.html |
| 13:57.31 | anonymouz666 | Using PAP2 indeed doesn't work. I just plugged into a legacy PBX (FXS without RTP) and everythings goes fine. |
| 13:59.15 | coppice | the message says they send DTMF |
| 14:00.49 | anonymouz666 | I do not understand. DTMF shouldn't be a problem. |
| 14:01.45 | anonymouz666 | if it was just the case, we detected the 50ms DTMF (sent by the client) emulate to 80ms and pass-through to the DAHDI channel |
| 14:03.26 | anonymouz666 | coppice: but that statement is true when it says that phase-shifted signals could be a problem to SIP/RTP? |
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| 14:12.14 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:12.18 | *** part/#asterisk c0rnoTa (~c0rnoTa@109.188.46.99) |
| 14:13.29 | coppice | that exchange about modems is garbage. nnetheless, the last message makes it sound like the real requirement is to carry DTMF, and some simple tone bursts |
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| 14:14.57 | *** join/#asterisk viraptor (~viraptor@212.11.65.66) |
| 14:16.18 | anonymouz666 | some simple tone bursts that shouldn't be a problem to PAP2 |
| 14:16.40 | anonymouz666 | and that tones differs from CED tones, so you have to explicity disable the EC |
| 14:16.54 | Kobaz | aughh switchvox is such a pain |
| 14:17.34 | anonymouz666 | coppice: that makes sense? |
| 14:17.55 | viraptor | hi, I've got a TDM fxo channel on a digium card on asterisk 1.4 - when a call comes in, I'm trying to reject it with hangup or congestion, but the line keeps ringing and asterisk tries to route the call again (as if it was a new one) - is there any specific way to reject calls? |
| 14:19.23 | [TK]D-Fender | viraptor: Show us the call |
| 14:19.44 | [TK]D-Fender | viraptor: And there is no such thing as "rejecting" an analog call |
| 14:20.01 | [TK]D-Fender | viraptor: There is either "answer" or "don't answer" |
| 14:20.34 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:20.34 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:22.36 | Kobaz | so |
| 14:22.37 | viraptor | [TK]D-Fender: so how do I signal "busy" for example? |
| 14:22.41 | tzafrir_laptop | viraptor, don't Answer() the call |
| 14:22.43 | Kobaz | i have this silly switchvox server |
| 14:22.49 | Kobaz | but this seems to be a sip problem |
| 14:22.53 | viraptor | is that possible from the client side? |
| 14:22.53 | Kobaz | http://pastebin.com/0JidnX96 |
| 14:23.11 | Kobaz | audiocodes is dialing in on sip... and asterisk never hits any dialplan |
| 14:23.13 | viraptor | what I want to achieve is quick hangup of unrecognised calleids |
| 14:23.18 | [TK]D-Fender | viraptor: You have an analog phone on an analog line. How do YOU make a "busy"? |
| 14:23.39 | [TK]D-Fender | viraptor: Busy is the telco saying you're on the phone. You do that by BEING on the phone. |
| 14:24.02 | [TK]D-Fender | viraptor: A quick hangup means you're answering and hanging up |
| 14:24.04 | Kobaz | SIP/2.0 484 Address Incomplete |
| 14:24.08 | Kobaz | i think that's the problem... right? |
| 14:24.54 | [TK]D-Fender | Kobaz: Looking for 1 in sip_provider_103 (domain 192.168.55.99) |
| 14:24.57 | [TK]D-Fender | ^^^ |
| 14:25.08 | [TK]D-Fender | INVITE sip:1@192.168.55.99;user=phone SIP/2.0 |
| 14:25.21 | viraptor | [TK]D-Fender: so I have to answer then... ok - I'm still new to analog lines in a way |
| 14:25.22 | [TK]D-Fender | Kobaz: Are you even LOOKING at what's coming in? :) |
| 14:25.28 | Kobaz | well |
| 14:25.32 | Kobaz | this config is really limited |
| 14:25.40 | Kobaz | i have it set up to match all |
| 14:25.52 | Kobaz | but it seems match all doesnt match that for some reason |
| 14:25.54 | [TK]D-Fender | Kobaz: O RLY? |
| 14:26.03 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 14:26.11 | Kobaz | <PROTECTED> |
| 14:26.17 | Kobaz | that's the rule that it made |
| 14:26.22 | [TK]D-Fender | Kobaz: that isn't "all" |
| 14:26.27 | Kobaz | no, it isn't |
| 14:26.50 | Kobaz | that doesn't match '1' does it |
| 14:26.57 | Kobaz | it's expecting at least two characters |
| 14:27.20 | [TK]D-Fender | Kobaz: nope... |
| 14:27.41 | [TK]D-Fender | Kobaz: Try again... |
| 14:27.41 | Kobaz | no? |
| 14:27.46 | [TK]D-Fender | Kobaz: NO <- |
| 14:27.48 | Kobaz | _[][] |
| 14:27.54 | Kobaz | one set followed by another set |
| 14:27.59 | [TK]D-Fender | (o)(o) |
| 14:28.18 | [TK]D-Fender | [10:27]<Kobaz>one set followed by another set <- close but no cigar |
| 14:28.31 | Kobaz | but but |
| 14:28.56 | [TK]D-Fender | [10:26]<Kobaz> '_[a-zA-Z0-9_+][a-zA-Z0-9_].' => 1. Set(is_provider=1) <- read it again. |
| 14:30.05 | Kobaz | '_[a-zA-Z0-9_+][a-zA-Z0-9_].' the first [] any char a-zA-Z... etc the second [a-zA-Z..etc] matches another.. with a '. |
| 14:30.10 | Kobaz | ' one or more match |
| 14:30.18 | Kobaz | oh... it's looking for a minimum of three |
| 14:32.19 | Kobaz | :) |
| 14:32.23 | *** join/#asterisk nix8n82 (~nate@63.162.27.14) |
| 14:32.52 | [TK]D-Fender | Kobaz: Doing a job right the first time is efficient. Doing it right after 37 tries is job security |
| 14:35.52 | *** join/#asterisk babyhuey (~jlyons@cpe-067-023-144-044.static.wadsnet.com) |
| 14:36.25 | babyhuey | I have a question with the Digium Asterisk Appliance. Does it come with root access by chance? |
| 14:37.10 | [TK]D-Fender | babyhuey: Which? |
| 14:37.16 | *** join/#asterisk ixyd (~denzs@carbon.gonicus.de) |
| 14:37.48 | Kobaz | haha |
| 14:39.56 | ixyd | hey guys, iam trying to send faxes via sip/t.38 through the 1.8-svn from yesterday and iam wondering that the asterisk is reinviting to alaw after a couple of pages...can you tell me why the asteirsk is doing so? any hints would be great! :) |
| 14:40.45 | [TK]D-Fender | ixyd: #asterisk-dev <- |
| 14:40.55 | ixyd | ah ok thank you! |
| 14:41.53 | drmessano | Maybe Asterisk gives up on T.38 like we all do |
| 14:42.28 | anonymouz666 | drfreeze: LOL |
| 14:42.44 | *** join/#asterisk timeshell_atwork (~chatzilla@gw.lusi.on.ca) |
| 14:42.51 | ixyd | hehe maybe...but the transfer is doing really well until the asterisk gets the idea to reinvite with t.38, it is just a stupid pass through :) |
| 14:43.32 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:46.18 | tzafrir_laptop | ixyd, so you were transferred back here? |
| 14:46.48 | ixyd | seems so :( |
| 14:47.14 | chazzam | ixyd: so you are trying to call SendFax from asterisk, or is this a passthrough fax via sip? |
| 14:47.22 | chazzam | are you using spandsp? |
| 14:47.36 | ixyd | it is just passthrough |
| 14:48.18 | anonymouz666 | anyone in here uses the DNSMASQ with Asterisk? |
| 14:48.19 | chazzam | so it starts in G.711 and sends a few pages, then asterisk does a re-invite to switch to t.38, then it fails and drops back to g.711? |
| 14:48.23 | ixyd | the asterisk is just doing the sip routing, faxdetection in sip.conf is outcommented which should then be "no" as the comment says |
| 14:48.25 | *** join/#asterisk ruyo (~psantos@a83-132-248-161.cpe.netcabo.pt) |
| 14:48.42 | *** join/#asterisk iratik (~itariki@74-84-99-12.client.mchsi.com) |
| 14:49.10 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
| 14:49.13 | wcselby | o/ |
| 14:49.14 | chazzam | ixyd, you realize 1.8 is still in beta right? have you tried explicitly setting faxdetect=no? |
| 14:49.40 | Katty | hugs wcselby |
| 14:49.46 | ixyd | it is starting with g.711 until one either the pstn-gateway or the ata detects the CED-tone and does the reinvite to t.38 which does very well until the asterisk decides to reinvite with g.711 |
| 14:49.46 | wcselby | morning Katty |
| 14:49.50 | iratik | Anyone ever have trouble using SPAN to monitor g729 traffic from asterisk on catalyst switches? The RTP becomes totally garbled as soon as i activate the destination port |
| 14:50.08 | ixyd | @chazzam you are right! i will set it to no right now |
| 14:50.11 | babyhuey | [TK]D-Fender: the aa50 s808 |
| 14:50.37 | [TK]D-Fender | babyhuey: IIRC You can SSH to it normally. It runs rPath (suck) |
| 14:51.00 | babyhuey | ok, as long as i am able to add files to the tftp and such for other phones |
| 14:51.19 | [TK]D-Fender | babyhuey: I'd call Digium sales first.. |
| 14:51.51 | chazzam | babyhuey: the aa50 runs uClinux, and you can enable SSH in the networking page in the GUI |
| 14:51.58 | babyhuey | [TK]D-Fender: ok, I was going to, just thought i may be able to ask in here first, thanks for your help! |
| 14:52.09 | ixyd | @chazzam but as i see faxdetection should lead the asterisk to jump to the fax extension which doesnt exist...but it doesn jump anywhere in the dialplan, it just does the reinvite... :( |
| 14:52.12 | babyhuey | chazzam: perfect, thanks |
| 14:52.42 | chazzam | its shell is.... limited though, so don't expect all the normal linux commands to work |
| 14:53.19 | chazzam | ixyd: well, it will forward the re-invite requests and re-write them |
| 14:53.25 | chazzam | do you have all the ports open for UDPTL? |
| 14:53.37 | chazzam | if that traffic isn't flowing through, then it will get cancelled |
| 14:53.56 | ixyd | yes, iam watching almost every packet in wireshark and everything looks good to me |
| 14:53.59 | wcselby | so i was playing around wtih realtime and adaptive odbc last night, the results weren't quite what I was expecting |
| 14:54.43 | wcselby | the adaptive odbc would create the tables, but it wouldn't really create the whole table. for that I had to pull from an example online |
| 14:54.45 | ixyd | the only things catching my eye are some rtcp packet from asterisk to the extensions which are dropped |
| 14:54.55 | chazzam | in the SDP information for the re-invites do the IP addresses and ports match up? the UDPTL should be flowing through asterisk then being passed on to the end-points. NOT going straight between the endpoints |
| 14:55.31 | ixyd | yes, it is passed through the asterisk |
| 14:56.28 | *** join/#asterisk adyn (~adyn@unaffiliated/adyn) |
| 14:56.38 | *** join/#asterisk timeshell_atwork (~chatzilla@gw.lusi.on.ca) |
| 14:57.33 | *** join/#asterisk Hmail (~henk@wolerized.xs4all.nl) |
| 14:59.53 | Hmail | Hi guys, I am having trouble connecting 2 voip lines to 2 extensions. At first, both lines connect to both extensions, but pretty much all of a sudden none of the extensions are available, and the ss-noservice message is played. I can't figure out why they aren't available, but I have tried with multiple phones and the problem remains the same. Do you have any idea where to look to figure out why this is happening? |
| 14:59.53 | Hmail | \ |
| 15:01.15 | *** join/#asterisk c0rnoTa (~c0rnoTa@80.251.113.56) |
| 15:01.17 | *** join/#asterisk moy (~moy@209.117.177.35) |
| 15:01.40 | *** join/#asterisk bent_screwdriver (~socain00@74.255.249.66) |
| 15:03.49 | iratik | On that note |
| 15:03.50 | *** part/#asterisk iratik (~itariki@74-84-99-12.client.mchsi.com) |
| 15:04.28 | ixyd | @hmail what do you see in the cli during that? |
| 15:05.38 | Hmail | @ixyd: A lot. I don't see something like: extension not found or something like that. |
| 15:06.02 | Hmail | @ixyd: I can paste it in pastebin or something if you want |
| 15:06.04 | ixyd | can you show it on pastebin? |
| 15:06.06 | ixyd | yeah |
| 15:06.08 | ixyd | :)\ |
| 15:06.11 | Hmail | ok, hold on\ |
| 15:07.22 | *** part/#asterisk c0rnoTa (~c0rnoTa@80.251.113.56) |
| 15:09.09 | wcselby | any suggestions for cdr analyzers that people here have worked with or seen and liked? |
| 15:09.14 | Hmail | @ixyd: this is part of the log, it looked like it was from one conversation (where it didn't work): http://pastebin.com/ek7aKg1M |
| 15:11.18 | ixyd | @Hmail is that asterisk@home? |
| 15:11.36 | Hmail | @ixyd: trixbox |
| 15:11.54 | ixyd | hm k iam not familiar with that.. |
| 15:12.20 | [TK]D-Fender | Hmail: It calls the device then it dies. We can't see why because you aren't looking at the SIP DEBUG <- |
| 15:12.44 | ixyd | - SIP/200-00000016 is ringing |
| 15:12.48 | wcselby | ixyd - trixbox is asterisk@home |
| 15:13.04 | Hmail | @[TK]D-Fender: Okay, i'll try to enable sip debug |
| 15:13.55 | ixyd | @wcselby iam neither familiar with trixbox nor aah ;) |
| 15:14.28 | [TK]D-Fender | ixyd: Then why did you ask about it in the first place? |
| 15:15.15 | ixyd | i wanted to know where the not-so-simple dialplan comes from and the first google matches for the used macros looked like aah |
| 15:15.25 | Hmail | @[TK]D-Fender: this is with debugging enabled: http://pastebin.com/pUhxPcTy |
| 15:15.28 | *** join/#asterisk SaiSoma (~chatzilla@client105.jdcc.edu) |
| 15:15.38 | *** part/#asterisk viraptor (~viraptor@212.11.65.66) |
| 15:15.49 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.86) |
| 15:16.38 | Hmail | (sip debugging that is) |
| 15:17.09 | ixyd | in the first pastebin it looked like the extensions 200 was ringing?! |
| 15:18.23 | Hmail | @ixyd: Well, I'm not really sure if that was the most recent log. The second pastebin is recent. And it definitively didn't ring then |
| 15:19.17 | Hmail | also, it looks like if asterisk reloads the configuration, it works again for some time (can't figure out how long). Then it stops working |
| 15:19.22 | [TK]D-Fender | Hmail: I don't see a COMPLETE call to debug here |
| 15:19.34 | ixyd | as you know i dont know trixbox but to me it looks like you are missing a mapping from the external number 858783641 to an internal extension? |
| 15:21.09 | Hmail | @[TK]D-Fender: I called, got the noservice message, hung up, and then copied the log from the console. Do you miss something from the start or from the end? |
| 15:21.27 | [TK]D-Fender | Hmail: this last call was an inbound call you didn't set up a DID for |
| 15:21.43 | Hmail | @ixyd: I guess. But I don't know why it does work sometimes |
| 15:21.53 | [TK]D-Fender | Hmail: This is GUI config issue, and it is NOT supported here |
| 15:22.58 | Hmail | @[TK]D-Fender: Erm, I have a inbound route for all DID's. So I should have it |
| 15:23.15 | [TK]D-Fender | Hmail: You FAILED |
| 15:23.24 | Hmail | @[TK]D-Fender: How do you know it's a GUI issue? |
| 15:23.30 | *** join/#asterisk Russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 15:23.39 | [TK]D-Fender | Hmail: Because the call was accepted and your DIALPLAN told them to GTFO |
| 15:23.50 | Russ | There isn't a way to dynamically create extensions, is there? |
| 15:23.58 | [TK]D-Fender | russyes there is |
| 15:24.06 | [TK]D-Fender | russ: yes there is |
| 15:24.44 | Russ | I want to run AGI to parse the CID of the extension against a nanpa database to make some entries for 10 digit rules |
| 15:25.20 | ixyd | for my above mentioned t.38 problem: http://pastebin.com/xhF0n74C it looks like a session timer?! expires, which makes the asterisk reinvite the call legs....any ideas on this? |
| 15:26.29 | jpmcallister | Hello. I'm trying to install codex_speex.so with asterisk16 from digium repos. When I try to load de module I compiled I get that message: Module 'codec_speex.so' was not compiled with the same compile-time options as this version of Asterisk. |
| 15:26.41 | Katty | hello my asterisk does not work at all how to fix pls?????? |
| 15:26.58 | Katty | it does not come on???? |
| 15:27.17 | nix8n82 | Katty, are you being funny or mocking someone? |
| 15:27.56 | *** join/#asterisk sgtpepper (d80ac116@gateway/web/freenode/ip.216.10.193.22) |
| 15:28.09 | Naikrovek | nix8n82: she does this all the time |
| 15:28.24 | Naikrovek | nix8n82: she's being cute. mocking everyone who doesn't know how to ask for helps |
| 15:28.26 | Naikrovek | help |
| 15:28.39 | sgtpepper | Hi guys... quick question... is there a way to avoid asterisk sending the 183 and 180 after an invite? |
| 15:29.17 | ixyd | @sgtpepper sure you dont want ringing? |
| 15:29.47 | sgtpepper | its a long story.. basically.. I'm routing that call by a trunk that might later give me a 503 |
| 15:29.54 | nix8n82 | That's what I thought, but then again she could of hit her head and went retarded. |
| 15:30.04 | sgtpepper | so I want the ringing when that trunk is actually ringing |
| 15:30.16 | sgtpepper | ixyd: I don't know if I'm making my point clear |
| 15:31.01 | Katty | hurrradduhhrrr |
| 15:31.58 | ixyd | hm i think asteirsk should only send 180 or 183 to the caller it already an prov-response was sent by the callee..?! |
| 15:32.10 | ixyd | or are you using the "r" option in your dial statement? |
| 15:32.11 | Kobaz | hmm |
| 15:32.42 | Kobaz | anyone know how to set the timzone on a spa-941 |
| 15:32.57 | sgtpepper | let me check |
| 15:33.01 | *** join/#asterisk modsaid (~modsaid@82.201.210.162) |
| 15:33.01 | [TK]D-Fender | sgtpepper: Run a SIP proxy in front of *, or "vi chan_sip.c" |
| 15:33.41 | [TK]D-Fender | [11:32]<ixyd>or are you using the "r" option in your dial statement? <- VERY wrong. |
| 15:33.56 | [TK]D-Fender | ixyd: That will ANSWER the caller. |
| 15:34.32 | raden | Morning Katty :D |
| 15:35.02 | sgtpepper | [TK]D-Fender: that was my fear... |
| 15:35.13 | sgtpepper | adding yet another box.. |
| 15:35.17 | raden | jpmcallister, ask in #linux or #c |
| 15:35.41 | Katty | hugs on raden |
| 15:35.47 | Katty | raden: ohai. howrechu today |
| 15:36.27 | raden | Katty, I'm well and how are thee ? |
| 15:36.51 | jpmcallister | raden: I'm using the asterisk pakhages from digium. I just want to add codex_speex. It is compiled, but it does not load in asterisk. |
| 15:38.10 | nix8n82 | jpmcallister, you might want to recompile asterisk with your speex codex |
| 15:38.15 | ixyd | @[TK]D-Fender in my asterisk 1.4 the call isnt answered even with the r-option |
| 15:39.41 | jpmcallister | nix8n82: That was what I was afraid... tank you anyway |
| 15:39.48 | p3nguin | jpmcallister: Try the proper spelling of codec_speex.so instead. |
| 15:39.58 | raden | jpmcallister, load the module ? |
| 15:41.04 | nix8n82 | Don't be scared it only hurts for a minute. |
| 15:41.08 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 15:41.53 | jpmcallister | raden: I'm using binary asterisk packages from digium. I compiled codec_speex.so with the same source version of asterisk. But the codec does not load. I get ther error: Module 'codec_speex.so' was not compiled with the same compile-time options as this version of Asterisk. |
| 15:41.54 | nix8n82 | or do like p3nguin and raden said and try to load it manually from the cli first. |
| 15:42.12 | jpmcallister | nix8n82: I'm trying to load from the cli |
| 15:42.20 | *** join/#asterisk hfb (~hfb@pool-98-112-109-237.lsanca.dsl-w.verizon.net) |
| 15:42.24 | p3nguin | You haven't shown us any evidence of that. |
| 15:45.39 | [TK]D-Fender | [11:35]<sgtpepper>adding yet another box.. <- don't need another box. Yuo can run it on the same |
| 15:49.22 | *** join/#asterisk eliel (~eliels@201.234.94.226) |
| 15:49.28 | *** join/#asterisk angavmx (~angav@189.140.240.155) |
| 15:50.38 | angavmx | Hello everyone. I'm looking information for a Voipsolutions ATA MTA-102. I can´t get it connected to asterisk using Elastix. Anyone has any experience with this equipment?. |
| 15:52.11 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 15:52.17 | [TK]D-Fender | angavmx: A brand no-one is likely to have even heard of let alone encountered... |
| 15:52.30 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 15:53.09 | *** join/#asterisk oej (~olle@95.209.106.251.bredband.tre.se) |
| 15:54.27 | angavmx | hahahaha kind of. I have found that these misterious boxes were distributed by several companies but the manufacturer is not clearley identified, so far all I know is that these use a Broadcom BCM1112 chip |
| 15:54.57 | gr0mit | anyone played with the Goip gsm sip gateways ? |
| 15:55.30 | Kobaz | hmm |
| 15:56.19 | angavmx | Still I havefound several posts (covered with dust) where people seem to have installed them with asterisk |
| 15:59.14 | [TK]D-Fender | angavmx: The only people who care what chip they use are those who actually write firmware... |
| 15:59.36 | [TK]D-Fender | angavmx: Again I can't imagine anyone who'd be in here would so much as touch it with a 10' pole |
| 16:00.06 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 16:03.19 | angavmx | :( I had a little, tiny, minuscule hope that someday I would find the way to set my annoying white boxes to astersik... Hey! for sale! cheap Voip ATAs! |
| 16:03.35 | [TK]D-Fender | angavmx: how much is "cheap"? |
| 16:04.05 | angavmx | mhhh... 45 USD? |
| 16:09.26 | p3nguin | It better be really good for that price. |
| 16:09.29 | drmessano | LOL |
| 16:09.52 | drmessano | You know that one can buy "unlocked" PAP2's on ebay for $30 or so, right? That includes shipping |
| 16:10.11 | drmessano | And you're asking $45 for your unknown, probably locked, whitebox ATA special? |
| 16:10.29 | drmessano | I will give you $5 each for them |
| 16:10.49 | p3nguin | I'll give you $45. |
| 16:10.50 | chuckf | but the extra $15 is for the game you get to play with it once it arrives! |
| 16:10.55 | p3nguin | for nine of them. |
| 16:10.59 | drmessano | ROFL |
| 16:11.55 | drmessano | I'll give you $50 |
| 16:11.58 | drmessano | For 10 |
| 16:13.12 | [TK]D-Fender | angavmx: Shit price for a shit product. FAIL |
| 16:13.47 | [TK]D-Fender | angavmx: http://www.telephonydepot.com/Catalog/Cisco-Analog-Adapters/Linksys-PAP2T-NA --- <$45 |
| 16:14.24 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 16:15.56 | angavmx | ok guys you force me to make my desition... now my boxes are destinated to hold my door open :P |
| 16:16.40 | [TK]D-Fender | angavmx: I doubt they are sturdy enough for that either... |
| 16:17.20 | angavmx | mhhh... U right ... :S |
| 16:17.27 | wcselby | lol |
| 16:17.43 | drmessano | $45 each lol |
| 16:18.28 | wcselby | hey [TK]D-Fender - you've answered this for me before but I failed to bookmark it. what's a good, web-based cdr analysis tool? |
| 16:19.37 | drmessano | angavmx: I will buy all you have for $5 each.. if I can unlock them and use them for Asterisk, you can keep the $5. Deal? |
| 16:25.46 | angavmx | drmessano: hahahahah I'm actually tring to set them up 4 Asterisk so... I guess I will accept the $5 from myself... but I'd offer you the $5 if you got me to connect them :P |
| 16:26.12 | Diffen2 | Hello, i want to setup one dynamic conferenceroom assigned to one phonenumber, i want only to promote one number and it should be able to dymically create conferencerooms of the pincode that the users enter. And it should be able to hold simultanious of conferencerooms. I have looked at the option D but that only create one conferenceroom and doesnt allow others to enter other pins. Does anyone have a clue on what o |
| 16:26.12 | Diffen2 | ption i might use? |
| 16:27.53 | drmessano | $5 to help you set up the ATAs? $25 each and you have a deal |
| 16:31.12 | angavmx | drmessano: U mean 5 times the valued cost of the box considering that I can get an Unlocked Lynksys PAP2 for 30 bucks including shipping?... I put $10 on the table (and I wont tell you helped me) |
| 16:31.41 | angavmx | hahahahhahaha |
| 16:32.30 | drmessano | If you're looking to hire a consultant to set up your ATA's, I will be glad to charge you 1/2 of my hourly rate, which is $25, and consider it a "charitable contribution to the spread of Asterisk". You should be thanking me. |
| 16:33.47 | angavmx | was just kidding, but I'll get serious then :) |
| 16:35.05 | [TK]D-Fender | wcselby: Areski |
| 16:35.14 | wcselby | [TK]D-Fender - thanks |
| 16:35.21 | drmessano | However, continued looking of said gifthorse in said mouth will only push my desired compensation much closer to my $100 an hour "overtime emergency rate" rather than simply charging the standard $50. I will also be forced to charge applicable sales tax, VAT, and SPLOST on said labor, as I would no longer be contributing to a charitable institution |
| 16:36.12 | drmessano | Now, I will fax over the appropriate IRS forms, which I need signed in triplicate and faxed back, preferably over SIP/T.38, and we can get started on those ATA's |
| 16:37.01 | angavmx | and you might add iternational rates too :D |
| 16:37.40 | drmessano | Also, I will need these faxed back in COLOR and all signatures in BLUE ink only, please.. and no felt tip pens, crayons, sharpies, or colored pencils, thanks |
| 16:38.09 | [TK]D-Fender | drmessano: Can I get that in cornflower blue? |
| 16:38.37 | angavmx | Come on... no crayons? |
| 16:39.47 | [TK]D-Fender | drmessano: I've had co-workers whose natural writing skills were comparable to 1 CPI Crayola :) |
| 16:41.16 | drmessano | lol |
| 16:43.39 | *** join/#asterisk bjhaid (~IceChat7@80.89.178.164) |
| 16:44.04 | paulc | Diffen2: You could use Read() to get a PIN/conf ID from the caller, then use the D option to create it dynamically |
| 16:45.45 | bjhaid | hello, please does anyone know of any open source sip phone for use on wifi enabled symbian phones (e.g nokia e70) |
| 16:47.53 | [TK]D-Fender | bjhaid: http://www.google.ca/#hl=en&source=hp&q=nokia+e70+sip+client&aq=f&aqi=&aql=&oq=&gs_rfai=&fp=e152ff60f580dc26 |
| 16:48.06 | [TK]D-Fender | JFGI |
| 16:48.14 | [TK]D-Fender | NEXT!!@!!!@! |
| 16:49.12 | Diffen2 | paulc ok thanks man i will take a look at it. Its too bad that the option D doesnt add more conferences when one is active |
| 16:51.31 | bjhaid | fender i have googled and not found something reasonable that's why i am asking |
| 16:52.39 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 16:53.14 | *** join/#asterisk b14ck (~rdegges@cpe-24-24-128-47.socal.res.rr.com) |
| 16:55.56 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 16:58.16 | Russ | so... [TK]D-Fender... generating extensions on the fly, how would one do that? |
| 17:05.14 | Russ | eg, _602NXXXXXX, _480NXXXXXX, _623NXXXXXX |
| 17:10.03 | *** join/#asterisk bjhaid (~IceChat7@80.89.178.164) |
| 17:10.43 | p3nguin | Why would those need to be generated on the fly? |
| 17:10.57 | p3nguin | _NXXNXXXXXX would take care of them all. |
| 17:12.15 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.168.54.dsl.dyn.forthnet.gr) |
| 17:13.45 | Russ | I want 7 digit and 10 digit dialing to work |
| 17:14.02 | Russ | _706XXXX is a valid 7 digit number |
| 17:14.04 | carrar | only allowed 6 |
| 17:14.07 | Russ | but 706 is also a NANPA area code |
| 17:14.18 | p3nguin | Write the extensions, then. |
| 17:14.44 | p3nguin | 7061234 does not match _NXXNXXXXXX |
| 17:14.50 | Russ | yes, then I have to write out all possible extension combinations for all possible source caler ids |
| 17:15.07 | carrar | why not have 2 |
| 17:15.08 | Russ | yes, but 7063331234 matches _706XXXX |
| 17:15.18 | p3nguin | no it doesn't. |
| 17:15.44 | Russ | as soon as I hit the 1 (7th digit) it will match the _NXXXXXX rule |
| 17:15.52 | p3nguin | You're only going to be dialing a few different area codes if you're using 7-digit dial. |
| 17:16.05 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 17:16.30 | *** join/#asterisk mnicholson (~mnicholso@nat/digium/x-bszczkpbvzhjghfq) |
| 17:16.31 | Russ | I'm trying to make a generic nanpa infastructure that keys off your source CID |
| 17:17.10 | carrar | So you need database of what a local call is depending on your caller ID? |
| 17:17.15 | Russ | suppose I could generate a context for each 10 digit dialing region |
| 17:17.45 | Russ | I need to check, but I don't think nanpa is mean enough to assign an NXX to a state with an area code matching it |
| 17:17.56 | Russ | so 50 contexts for the US |
| 17:19.02 | [TK]D-Fender | [13:15]<Russ>yes, but 7063331234 matches _706XXXX <- NO |
| 17:19.20 | carrar | Welcome to 2 mins ago TK!! :) |
| 17:19.22 | Russ | 7063331 certainly does |
| 17:19.38 | [TK]D-Fender | russ: That is a SEVEN digit fixed length pattern. It will NOT match a 10 digit dialed numebr |
| 17:19.51 | Russ | if I'm touch dialing on a POTS telephone it does |
| 17:19.53 | carrar | Russ, how so? |
| 17:20.11 | Russ | why would the extension matching rules wait for more digits, wouldn't it match right away? |
| 17:20.17 | [TK]D-Fender | russ: Connected to * how? |
| 17:20.22 | carrar | grabs the bunt from Russ, ENOUGH!! |
| 17:20.27 | Russ | *? |
| 17:20.34 | [TK]D-Fender | russbecause it isn't matching a 10 digit number... it is STOPPING after 7 <- |
| 17:20.54 | [TK]D-Fender | russ: * = ASTERISK |
| 17:21.50 | Russ | isn't that what I said? |
| 17:22.15 | [TK]D-Fender | [13:20]<Russ>*? <- |
| 17:22.29 | Russ | I mean before that, it stops at 7 |
| 17:22.43 | *** join/#asterisk bjhaid (~IceChat7@80.89.178.164) |
| 17:22.46 | carrar | 7 != 10 |
| 17:22.55 | Russ | I know, but you hit 7 on the way to 10 |
| 17:23.01 | *** join/#asterisk silvestre_id (~silvestre@200-204-158-49.dsl.telesp.net.br) |
| 17:23.13 | carrar | no you hit 7 and stop |
| 17:23.54 | Russ | ah, so if I have _602NXXXXXX and _NXXXXXX, depending on the sort order, it could easily use the _NXXXXXX extension |
| 17:24.08 | Russ | (if I dial 6023335555 |
| 17:24.49 | carrar | ~book |
| 17:24.49 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 17:24.55 | [TK]D-Fender | russ : There is no "could". * picks the most specific pattern that matches. |
| 17:25.04 | Russ | that means my contexts need to not only list a _<area_code>NXXXXXX for every planned 10 digit dialing area code, but also a _<nxx>XXXX for every NXX in the region |
| 17:25.34 | [TK]D-Fender | russ: Or a single open-ended pattern and you process it all yourself. |
| 17:25.43 | Russ | as long as no NXX codes overlap area codes in that 10 digit dialing region, I'm fine |
| 17:26.13 | Russ | yes, but then I don't get the same timeout and/or instant extension match behavior |
| 17:26.18 | *** join/#asterisk mpd (~chatzilla@CPE00121724e38f-CM00122542242c.cpe.net.cable.rogers.com) |
| 17:26.34 | silvestre_id | [TK]D-Fender: The channel variable received by AGI on the Queue is the real channel for the agent who answered the call. Now everything is working. |
| 17:26.55 | p3nguin | Dial more digits if you want 10. Dial less if you want 7. |
| 17:27.14 | *** part/#asterisk mpd (~chatzilla@CPE00121724e38f-CM00122542242c.cpe.net.cable.rogers.com) |
| 17:27.17 | Russ | I want it to match right away if I dial a 7 digit number with a valid NXX for my region |
| 17:28.09 | p3nguin | I recommend that you learn pattern matching. |
| 17:28.13 | p3nguin | ~pattern matching |
| 17:28.14 | infobot | i heard pattern matching is explained here: http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
| 17:28.22 | Kyosh | can asterisk real-time work with IAX in order to load the extensions dynamically from the database? if so, what would the table structuure look like? |
| 17:28.23 | [TK]D-Fender | russ: [13:25]<Russ>as long as no NXX codes overlap area codes in that 10 digit dialing region, I'm fine <- this will almost invariably happen... |
| 17:28.40 | *** part/#asterisk jpmcallister (~ec06113@200.242.28.231) |
| 17:28.41 | [TK]D-Fender | silvestre_id: Glad to hear |
| 17:28.43 | Russ | [TK]D-Fender, do you have an example? |
| 17:29.07 | [TK]D-Fender | 905-7000 |
| 17:29.46 | Russ | 700 isn't a valid NXX anywhere |
| 17:29.58 | [TK]D-Fender | russ : There. COULD be NXXXXXX or waiting for more to match NXXNXXXXXX |
| 17:29.59 | p3nguin | It would be pretty easy to rip the area code out of the caller ID, then stick it over on the Dial() command when dialing a 7-digit number. |
| 17:30.11 | Russ | that is why I have the nanpa databases |
| 17:30.19 | [TK]D-Fender | russ: 905-7223 <- ' |
| 17:30.37 | [TK]D-Fender | russ : the last 4 digits can be ANYTHING. FFS. |
| 17:30.42 | Russ | p3nguin, that is assuming you know that the user is doing 7 digit dialing |
| 17:30.49 | [TK]D-Fender | russSo yes, you're GOING to get overlap |
| 17:31.04 | carrar | Russ, 7 digit numbers don't have a NPA |
| 17:31.06 | p3nguin | I don't have to know what they are dialing -- the patterns take care of that for me. |
| 17:31.11 | [TK]D-Fender | [13:30]<Russ>p3nguin, that is assuming you know that the user is doing 7 digit dialing <- You DON'T know what they are dialing until they are DONE |
| 17:31.21 | Russ | p3nguin, if you are willing to wait till the stop dialing |
| 17:31.34 | [TK]D-Fender | russ: if 2 matches are possible (waiting for more or happy as we are) then yuo will WAIT |
| 17:31.44 | [TK]D-Fender | russ: That is the way of things |
| 17:32.41 | Russ | I'm still looking for the NXX area code ambiguity |
| 17:33.00 | Russ | when they dial the first three digits, you should be able to tell whether it is a valid NXX in your 10 digit area, or a valid area code in your 10 digit area |
| 17:33.04 | carrar | set a timeout |
| 17:33.07 | p3nguin | We don't care about the area code. The pattern matching still works the same. |
| 17:33.11 | carrar | use a lcads db |
| 17:33.18 | carrar | write something |
| 17:33.36 | Corydon76-dig | Russ: that was once true, but it's no longer true |
| 17:33.47 | Russ | Corydon76-dig, can you give an example? |
| 17:33.53 | Corydon76-dig | Russ: 414 |
| 17:34.34 | [TK]D-Fender | russ: I have 905 as a local exchange here NXX-XXXX as 7-digit, and it is ALSO the start of an AREA CODE. Thus it overlaps |
| 17:34.42 | Corydon76-dig | At one time, 414 was only an area code. But I know that in the 615 area code, 414 is also a legitimate NXX |
| 17:34.49 | *** join/#asterisk angavmx (~angav@189.140.240.155) |
| 17:34.50 | Russ | 414 is an area code in WI, and it is properly marked as UA in the nanpa database for all area codes in WI |
| 17:35.09 | Russ | [TK]D-Fender, an area code in your state? |
| 17:35.41 | [TK]D-Fender | russ: not mine personally. Next door, yes |
| 17:36.16 | Russ | damn canadians |
| 17:36.44 | Russ | suppose canadians won't get 10 digit dialing then |
| 17:37.01 | *** join/#asterisk bjhaid (~IceChat7@80.89.178.164) |
| 17:37.01 | *** join/#asterisk j4m3s (~j4m3s_@65.97.134.190.nw.nuvox.net) |
| 17:37.04 | [TK]D-Fender | russ: Wrong |
| 17:37.12 | [TK]D-Fender | russ: Had for many years in many cities |
| 17:37.15 | Russ | wait |
| 17:37.18 | [TK]D-Fender | FAIL |
| 17:37.34 | Russ | I think canada does have rules, you just need to know the regions |
| 17:37.43 | Russ | '226","905","","","Not Available","","Not available - existing Canadian NPA"' |
| 17:37.43 | [TK]D-Fender | russ: NANPA <- |
| 17:37.46 | carrar | no rules in CANADA |
| 17:37.56 | carrar | it's the last wild frontier!! |
| 17:37.58 | *** join/#asterisk j4m3s (~j4m3s_@65.97.134.190.nw.nuvox.net) |
| 17:38.05 | Corydon76-dig | The general rule is 'best effort' |
| 17:38.09 | p3nguin | What would be wrong with _NXXXXXX,1,Dial(SIP/itsp/${CALLERID(num):3}${EXTEN}) ? |
| 17:38.12 | Russ | so in 226, you need to dial 1 to get to 905 area code |
| 17:38.19 | Russ | er, wait other way around |
| 17:38.35 | Russ | er, yes, the first time |
| 17:38.51 | [TK]D-Fender | russ: The smarter you try to be, the dumber you may end up looking. Go ahead and fish for trouble. You will find it. The karmic wheel doesn't make a beeping sound when it backs up over you. |
| 17:38.54 | p3nguin | That would match when the person has dialed 7 digits, then take their own area code from their own CALLER ID number and dial the area code and 7 digits. |
| 17:39.01 | Russ | 226 and 249 both list 905 as reserved, so those area codes can 10 digit dial 905 area code |
| 17:39.22 | Russ | p3nguin, but it would break 10 digit dialing |
| 17:39.28 | p3nguin | Nope. |
| 17:39.43 | p3nguin | It doesn't care about 10 digits, since the pattern is only 7. |
| 17:40.04 | Russ | if I want 7 digit dialing to wait for the timeout |
| 17:40.38 | [TK]D-Fender | russ: What are your users dialing using? |
| 17:40.46 | Russ | POTS via dahdi |
| 17:41.12 | [TK]D-Fender | yuck |
| 17:41.54 | Russ | the nanpa database actually make this easy, if you have an NPA list another NPA as an NXX with "UA" or "Not Available", then they are part of a 10 digit dialing area |
| 17:44.10 | Russ | now I just need to make something that walks though the database and makes s/CID matching rules for each area code, within each one of those, a set of 10 digit dialing rules |
| 17:44.33 | j4m3s | anyone in here with good routes to latin america? |
| 17:45.43 | Russ | The only thing that would then suck is if I had to call out on POTS and figure out whether or not to dial a 1 |
| 17:47.35 | Russ | I don't actually have to be very smart, for any area code, list all UA (unassigned) NXX's that are valid NPAs as prefixes for 10 digit dialing |
| 17:49.04 | *** join/#asterisk bjhaid (~IceChat7@80.89.178.164) |
| 17:49.25 | Russ | thanks for helping me think though this and recognize the problems in my original, naive dialing plan rules |
| 17:53.48 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:56.42 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 17:57.13 | *** join/#asterisk punxos (~punxos@9.pool80-103-173.dynamic.orange.es) |
| 17:57.19 | punxos | He |
| 17:57.37 | mmlj4 | Them |
| 17:59.01 | [TK]D-Fender | We |
| 17:59.15 | [TK]D-Fender | cheers the start of "Personal Pronoun Week" |
| 18:10.05 | *** join/#asterisk nix8n82 (~nate@63.162.27.14) |
| 18:11.15 | *** join/#asterisk Jinxed- (~b0ot@198.99.129.129) |
| 18:13.47 | Jinxed- | how come the switchvox can only support a fraction of the concurrent calls that the regular version of asterisk can |
| 18:14.51 | [TK]D-Fender | Jinxed-: The number is a lie or is flatly artificail |
| 18:15.17 | [TK]D-Fender | Jinxed-: Feel free to specifically link us so we can tell which it is. |
| 18:15.33 | Jinxed- | [TK]D-Fender, I talked with a sales guy today |
| 18:15.51 | Jinxed- | it was something like 70 to 90 concurrent calls for switchvox |
| 18:16.03 | Jinxed- | and 500 concurrent calls with asterisk on a server |
| 18:16.30 | idespinner | Jinxed-, Switchvox dimensions very conservativley |
| 18:17.04 | idespinner | they claim 70-90 with full call recording while taking a backup, etc... |
| 18:17.15 | idespinner | running CDR reports |
| 18:18.27 | Jinxed- | idespinner, is the regular version of * able to do that with the 500 number they quoted? |
| 18:18.51 | idespinner | dunno, its all about hardware not software |
| 18:19.13 | [TK]D-Fender | Jinxed-: SHOW US |
| 18:19.20 | idespinner | ? |
| 18:20.31 | Jinxed- | ? |
| 18:20.48 | Jinxed- | show you what? |
| 18:21.18 | [TK]D-Fender | Jinxed-: Where these claims are made |
| 18:21.39 | Jinxed- | [TK]D-Fender, these were verbal claims made to me when I called Digium sales |
| 18:21.50 | [TK]D-Fender | Jinxed-: Its worth as much as you can show us... |
| 18:23.39 | Jinxed- | [TK]D-Fender, sorry I don't have a transcript of the chat |
| 18:23.54 | Jinxed- | I will probably get an email though |
| 18:23.55 | idespinner | Jinxed-, there are specs on the switchvox website |
| 18:24.10 | idespinner | the larger aa355 does spec out to about 70 concurrent calls |
| 18:24.18 | Jinxed- | it's alright, I like the custom asterisk solutions better anyway |
| 18:24.42 | idespinner | but that is a conservative estimate. You may want to jump in #switchvox to ask there... |
| 18:25.59 | carrar | http://www.switchvox.com/catalog/smb_bundles.php |
| 18:28.34 | [TK]D-Fender | Jinxed-: Show me this "switchvox" you were quoted about. |
| 18:29.10 | Jinxed- | they were the ones who talked about switchvox |
| 18:29.14 | *** join/#asterisk clintc (~clintc@n128-227-12-23.xlate.ufl.edu) |
| 18:29.27 | Jinxed- | well the nice thing about the asterisk is that you don't have to pay per subscriber |
| 18:29.33 | Jinxed- | you can just pay for the support |
| 18:29.41 | Jinxed- | I feel that it is more flexible |
| 18:29.44 | Jinxed- | than the turnkey solutions |
| 18:30.05 | [TK]D-Fender | Jinxed-: Ask yourself what this support covers and costs, and what you'll need. |
| 18:30.11 | Jinxed- | plus if your system was almost completly up and running but you just needed to add a couple of lines |
| 18:30.22 | [TK]D-Fender | Jinxed-: As well as what it takes to be "turnkey". |
| 18:30.25 | idespinner | The purpose of switchvox is just to be a pbx for those who cant/dont want to manage a linux server |
| 18:30.29 | Jinxed- | it would be very costly to have to pay the subscription on all your lines |
| 18:30.44 | raden_work | Jinxed-, wtf are you trying to accomplish ? |
| 18:30.45 | *** join/#asterisk jaxyeh (~jaxyeh@c-69-250-52-161.hsd1.md.comcast.net) |
| 18:31.23 | Jinxed- | I was just finding more information about Digium/Asterisk for possible solutions |
| 18:31.45 | *** join/#asterisk JamesHarrison (~jharr@hometree.mmmetrics.co.uk) |
| 18:31.50 | carrar | SwitchVox is a great product |
| 18:32.09 | Jinxed- | i never said it wasn't |
| 18:32.21 | [TK]D-Fender | Jinxed-: here's a good one : What exactly is this "switchvox" your were being informed about? |
| 18:32.32 | drmessano | Asterisk can handle 500+ concurrent calls on the RIGHT SIZE server |
| 18:32.48 | drmessano | You were spec'ed a SPECIFIC piece of hardware |
| 18:32.58 | carrar | or more on a LEFT SIZE server |
| 18:33.05 | drmessano | You're comparing apples and monkeys here |
| 18:33.10 | KavanS | Jinxed-, basically your best bet is to buy support from digium - asap |
| 18:33.32 | carrar | Millions of Monkeys, Millions of thumbs! |
| 18:33.40 | KavanS | <<<<--- not employed or affiliated by digium |
| 18:33.42 | KavanS | just sayin! |
| 18:33.50 | KavanS | by/with |
| 18:33.51 | Jinxed- | KavanS, if we ended up going with asterisk, that is what we would do |
| 18:34.13 | drmessano | That switchbox system that handles 70 calls likely can handle approx 70 calls if you reformatted and install a base OS and asterisk.. You spec'ed out HARDWARE and then asked support a general asterisk question, which they answered correctly |
| 18:34.14 | KavanS | Jinxed-, asterisk is awesome - I can't see using any other PBX for our SMB |
| 18:34.26 | drmessano | switchvox* |
| 18:34.33 | Jinxed- | [TK]D-Fender, this switchvox: http://www.digium.com/en/products/switchvox/ |
| 18:34.34 | KavanS | Jinxed-, 500 concurrent calls is a bit out of my experience though :) |
| 18:34.36 | carrar | Jinxed, if you want to use Asterisk you have a lot of learning that needs to happen, keep that in mind |
| 18:34.48 | carrar | there is a learning curve |
| 18:34.50 | Jinxed- | oh, I know |
| 18:34.55 | carrar | much larger then switchvox |
| 18:34.59 | carrar | much much |
| 18:35.12 | carrar | switchvox is plug and play |
| 18:35.17 | carrar | seriously |
| 18:35.26 | [TK]D-Fender | Jinxed-: Switchvox is HARDWARE <------------------- |
| 18:35.31 | Jinxed- | I am new to a lot of IP/VOIP world, but my background is EE/CE |
| 18:35.37 | [TK]D-Fender | Jinxed-: comparing it to software is folly |
| 18:35.41 | carrar | SwitchVox is Hardware & Software |
| 18:35.47 | drmessano | Jinxed-: If you pointed me to box A and I told you it could handle about 100 concurrent calls, then asked me "How many calls can Asterisk handle per server", I would tell you "500 or so" based on that being a best practice |
| 18:36.09 | Jinxed- | yeah, |
| 18:36.10 | [TK]D-Fender | Jinxed-: * scales to your hardware. Mind you the extra crap they shove in with the GUI etc will artifically limit you as well |
| 18:36.12 | drmessano | But the second query is NOT based on hardware specs, where the first one IS |
| 18:36.24 | Jinxed- | they said switchvox with 70 concurrent calls would be about 400 extentions |
| 18:36.25 | [TK]D-Fender | [14:35]<carrar>SwitchVox is Hardware & Software <- can you buy the software separately? |
| 18:36.43 | [TK]D-Fender | [14:36]<Jinxed->they said switchvox with 70 concurrent calls would be about 400 extentions <- BS statistic |
| 18:36.50 | carrar | we have |
| 18:37.00 | [TK]D-Fender | carrar: Can Joe Blow? |
| 18:37.04 | Jinxed- | [TK]D-Fender, exactly... switchvox from what they said is a commerical version of * meaning that the configuration files and everything are hidden from the user and it is a GUI configuration |
| 18:37.07 | carrar | probably not |
| 18:37.13 | carrar | but it does come with software |
| 18:37.17 | [TK]D-Fender | carrar: Lets just say "no" then :) |
| 18:37.20 | carrar | heh |
| 18:37.29 | carrar | it's a hardware & software solution |
| 18:37.35 | carrar | packaged |
| 18:37.37 | punxos | I'm want to install asterisk in gentoo, but I have this msg http://nopaste.info/d54d618f72.html I think that I need asterisk-sounds-1.6.XX but don't exist this ebuild |
| 18:37.51 | [TK]D-Fender | carrar: But not sealed for freshness :) |
| 18:38.08 | [TK]D-Fender | punxos: Go complain to your packager |
| 18:38.08 | carrar | If they include M&M's they might be sealed |
| 18:38.37 | Jinxed- | [TK]D-Fender, I thought you worked for digium |
| 18:38.40 | drmessano | Jinxed-: GUI doesn't mean "can handle less calls"... if you drop a bunch of extra running apps on there, that's one thing.. Your box that can handle about 70 calls likely couldnt handle much more if it was stripped down, and thats assuming theres any extra running processes on there that a bare * box wouldnt have |
| 18:38.48 | Jinxed- | last time I asked who here works for digium a bunch of people said they did |
| 18:38.52 | [TK]D-Fender | Jinxed-: Where would you get that idea? |
| 18:39.15 | [TK]D-Fender | Jinxed-: I said a bunch of people did. I never said I wasn one of them. |
| 18:39.20 | Jinxed- | last time I asked who here works for digium a bunch of people said they did <--- |
| 18:39.20 | punxos | A solution is to install asterisk-1.2 but I prefere 1.6 obviously |
| 18:39.21 | carrar | I work for Sustainable VoIP |
| 18:39.26 | raden_work | Jinxed-, what are your intentions how big of a company etc... ? |
| 18:39.32 | [TK]D-Fender | Jinxed-: Them != me :) |
| 18:39.38 | Jinxed- | haha |
| 18:39.48 | Jinxed- | I could just imagine calling you for help... |
| 18:39.50 | [TK]D-Fender | [14:39]<carrar>I work for Sustainable VoIP <- how long do you think that will last? ;) |
| 18:39.52 | drmessano | I work for Diguim.. Not to be confused with Digium |
| 18:39.55 | raden_work | drmessano, our single core P4 2.8's will handle 200 calls on pass through easily |
| 18:39.57 | carrar | TK, I made that up :) |
| 18:40.12 | drmessano | raden_work, Good for you.. |
| 18:40.20 | wcselby | drmessano - lol, Diguim |
| 18:40.23 | wcselby | the other Digium |
| 18:40.48 | Jinxed- | haha |
| 18:40.50 | carrar | VoIP will soon be replaced |
| 18:40.57 | raden_work | carrar, by what ? |
| 18:41.05 | wcselby | FoIP |
| 18:41.07 | carrar | VoT |
| 18:41.08 | [TK]D-Fender | VoCP <- |
| 18:41.22 | drmessano | raden_work: The query here isn't about what Asterisk can handle.. it's about looking at a SwitchVox box spec'ed to run about 70 concurrent calls, then asking "how many calls can Asterisk handle".. which are two different things |
| 18:41.23 | [TK]D-Fender | Carrier Pidgin > YOU |
| 18:41.24 | carrar | Voice over Telepathy |
| 18:41.32 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 18:41.40 | raden_work | drmessano, ahh gotcha |
| 18:41.49 | carrar | VoCP is ND backup plan |
| 18:42.05 | Jinxed- | Well I thought it was strange... but the two versions of * have to be different right? I mean a commerical version and the open source version... different licensing |
| 18:42.08 | [TK]D-Fender | Jinxed-: odds are any of those servers they sell could support a LOT more than that. Everything is variable however. |
| 18:42.15 | Jinxed- | I just found it odd that commerical version would be that much worse |
| 18:42.21 | [TK]D-Fender | Jinxed-: No, hardly different at all |
| 18:42.41 | [TK]D-Fender | Jinxed-: You pay for their GUI, their ahrdware, and their suppotr, and pay for it in $ and control. |
| 18:43.04 | drmessano | raden_work: He asked about a SwitchVox that's spec'ed for 70 calls and then was told Asterisk can handle 500 per server. Yes, my oranges are sweet, and my monkey drinks water.. What does one have to do with the other? |
| 18:43.13 | [TK]D-Fender | Jinxed-: You buy a solution that is about as locked as any other out there..... |
| 18:43.44 | drmessano | A WRT54G can handle a handful of concurrent calls with Asterisk.. does that mean Asterisk sucks and Trixbox doesnt? |
| 18:43.49 | citywok | I was unable to break 240 calls on my old dual xeon 3.6, :( |
| 18:43.50 | carrar | Jinxed, take the time to learn Asterisk from Source and build it yourself! |
| 18:43.54 | carrar | Thats what I recommend |
| 18:44.00 | carrar | ~book |
| 18:44.01 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 18:44.04 | carrar | read that |
| 18:44.08 | [TK]D-Fender | Jinxed-: And when you sell a product you don't want to get smashed for your product not working up to someones unrelaistic goals. So the (cl)aim low. |
| 18:44.09 | Jinxed- | haha, I have the book now |
| 18:44.26 | Jinxed- | and that book has a lot of stuff that is out of date |
| 18:44.42 | Jinxed- | yes i understand |
| 18:44.43 | [TK]D-Fender | Jinxed-: Correct, thats why there are docs in teh TARBALL |
| 18:44.45 | raden_work | there another book |
| 18:44.46 | carrar | You have to start someplace |
| 18:44.59 | [TK]D-Fender | [14:44]<raden_work>there another book <- several |
| 18:45.03 | drmessano | Jinxed-: Some of the CODE is out of date.. the other 80% of the book is ESSENTIAL reading |
| 18:45.21 | raden_work | Practical Asterisk 1.4 and 1.6: From Beginner to Expert |
| 18:45.26 | raden_work | that one really freaking good :) |
| 18:45.58 | carrar | Where is the PDF of it? |
| 18:46.13 | drmessano | I refuse to read any book that doesn't address 1.8 |
| 18:46.16 | raden_work | LOL there is not one |
| 18:46.21 | Jinxed- | anyone read this |
| 18:46.22 | Jinxed- | Learning Guide for Asterisk 1.6: Learn how to build a PBX using Asterisk in a week (Volume 1) |
| 18:46.22 | drmessano | Not reading OLD material |
| 18:46.25 | wcselby | raden_work - that's the book made from asteriskbook.org, right? |
| 18:46.27 | carrar | heh |
| 18:46.42 | drmessano | Jinxed-: You should read ANYTHING at this point |
| 18:46.48 | carrar | drmessano, you want my copy of Asterisk 2.0? |
| 18:47.00 | drmessano | Jinxed-: ANYTHING = better than what you've read currently |
| 18:47.01 | [TK]D-Fender | carrar: I remember that :) |
| 18:47.06 | wcselby | carrar - I think you can go to the-asterisk-book.com and get all the content of that book |
| 18:47.10 | drmessano | lol |
| 18:47.34 | drmessano | carrar: I have some pre-release Asterisk code too.. Wanna buy it? |
| 18:47.42 | carrar | Yes Please |
| 18:47.59 | drmessano | Post 1.8 kinda stuff.. all the source code |
| 18:48.03 | carrar | I want it to just work in 30 seconds when I click GO |
| 18:48.05 | wcselby | drmessano - I'll give you five bucks for it |
| 18:48.10 | raden_work | LMAO |
| 18:48.16 | wcselby | but you'll have to fill out a lot of IRS forms and stuff |
| 18:48.18 | wcselby | in blue ink |
| 18:48.20 | Jinxed- | carrar, you need my config files then |
| 18:48.22 | Jinxed- | :) |
| 18:48.26 | drmessano | wcselby: Paypal it, and I will send you a link |
| 18:48.38 | drmessano | lol |
| 18:48.39 | raden_work | 1.8 going to have SRTP ? |
| 18:48.51 | carrar | You didn't get the memo? |
| 18:49.00 | drmessano | 1.8 is going to have a SHITload of acronyms, raden_work |
| 18:49.05 | carrar | haha |
| 18:49.06 | wcselby | drmessano - i'll need the forms filled out first, they require social security number, birthdate, bank account numbers, bank account routing numbers, etc. if you could include a dna sample that'd be great as well |
| 18:49.12 | wcselby | for biometric security, of course |
| 18:49.19 | raden_work | LMAO |
| 18:49.26 | carrar | oh man, the more acronyms the better! |
| 18:49.43 | drmessano | TCP, SRTP, HTTP, IAX, VOIP, IPV6, and some SNMP amongst other things |
| 18:49.52 | [TK]D-Fender | drmessano: IIRC If you don't STFU with these acronyms ASAP you'll be DOA PDQ, AOK? |
| 18:50.07 | drmessano | ROFLMAO KTHXBYE |
| 18:50.11 | KavanS | lolz ^ |
| 18:50.24 | drmessano | Wait, that's an acronym and a meme |
| 18:50.25 | wcselby | [TK]D-Fender - IDK my BFF Jill? |
| 18:50.37 | KavanS | wcselby, BFF 4-EVUH? |
| 18:50.46 | wcselby | it's only a meme if you add the ^_^ |
| 18:50.55 | carrar | ^^) |
| 18:50.55 | drmessano | Sorry, I got my 1337-sp33k and my LOLCAT crossed again |
| 18:51.19 | carrar | points LASER EYES at drmessano |
| 18:51.33 | drmessano | CEILING CATS carrar |
| 18:51.55 | wcselby | hay guys my asteris server stopp, fix pls? ^)^ |
| 18:52.00 | drmessano | Don't make me invisible bicycle you |
| 18:52.06 | carrar | ok, that about wraps up my doy of work |
| 18:52.15 | carrar | day |
| 18:52.29 | *** join/#asterisk gamedna (~Adium@cpe-70-125-155-74.satx.res.rr.com) |
| 18:52.48 | drmessano | wcselby, oh hai, i can haz 1.8 srtp lol, iax u vrry much |
| 18:52.54 | drmessano | O RLY? YA RLY! |
| 18:53.21 | wcselby | KEKEKE ^_^ KTHXBAI ROFLCOPTER |
| 18:53.28 | wcselby | meh, time for work |
| 18:53.42 | wcselby | stupid having to pay bills and stuff |
| 18:54.21 | drmessano | So when is Digium bringing IAXtel back? |
| 18:54.25 | drmessano | I am waiting... |
| 18:55.48 | [TK]D-Fender | drmessano: Don't forget to hold your breath! |
| 18:57.23 | drmessano | http://www.google.com/search?q="mark+spencer"+mobile+number |
| 18:57.44 | drmessano | I would call him direct and ask him, but teh Google failed me |
| 18:58.27 | [TK]D-Fender | drmessano: I can tell you his last 4 digits ;) |
| 18:58.51 | Jinxed- | you guys ever hear of twillio |
| 18:59.03 | wcselby | voip in the cloud |
| 18:59.17 | KavanS | wcselby, do you run asterisk in the cloud? |
| 18:59.20 | wcselby | they presented at astricon last year |
| 18:59.25 | wcselby | no, twillio does, i think |
| 18:59.29 | KavanS | oh nice |
| 18:59.35 | KavanS | would be interested to hear their success |
| 18:59.45 | drmessano | How do you reboot a cloud? |
| 18:59.45 | wcselby | lot of cloud presentations last year, I think it had it's own track |
| 18:59.49 | drmessano | You people amaze me sometimes |
| 19:00.17 | KavanS | drmessano, whatever you call it - cloud is here to stay ;) |
| 19:00.18 | *** join/#asterisk kl4m (~kelam@gw1.sys-tech.net) |
| 19:00.32 | raden_work | shakes his head |
| 19:00.39 | KavanS | lol whatever... |
| 19:00.46 | KavanS | you guys take it as a joke |
| 19:00.50 | KavanS | I realize the cloud is a big marketing joke |
| 19:01.21 | KavanS | however - it's going to be an interesting proposition for businesses moving forward |
| 19:01.30 | KavanS | you can sit by and watch the train pass, or get with the program |
| 19:01.33 | KavanS | cloud is here to stay. |
| 19:01.37 | drmessano | KavanS: Yeah, to a point.. I haven't been convinced that realtime voice works with all my end user devices connecting to some remote server. |
| 19:01.59 | KavanS | drmessano, yeah I didn't think asterisk would fair well in a virtualized environment... |
| 19:02.06 | KavanS | just speaking of it as an abstract concept... |
| 19:02.15 | *** join/#asterisk nix8n82 (~nate@63.162.27.14) |
| 19:02.28 | KavanS | everytime you mention "cloud" in a tech channel - there is a bit of a scuffle ;) |
| 19:02.30 | raden_work | KavanS, SOME REMOTE SERVER @!!! |
| 19:02.42 | KavanS | raden_work, bacon or pork - all the same buddy ;) |
| 19:02.45 | raden_work | nothing wrong with cloud if it on your local network |
| 19:03.11 | gamedna | my old boss said my head was always in a cloud |
| 19:03.15 | raden_work | asterisk works fine in VM for me |
| 19:03.17 | gamedna | j/k |
| 19:03.18 | drmessano | I can see offloading a LOT to the cloud.. I am a big fan of Google Apps, for example.. and hosted CRM, and even online backup.. But I think there's enough client to client usage that warrants your phone switching being done onsite |
| 19:03.37 | raden_work | Totally agree |
| 19:03.54 | KavanS | yep, I can't see * being a solid solution in the cloud |
| 19:04.02 | KavanS | but I would love to see more info on it |
| 19:04.30 | raden_work | info on what ? |
| 19:04.35 | gamedna | in my experience, the problem w/ cloud services is not the provider or the service, its always the last mile to the end user |
| 19:05.06 | *** join/#asterisk Shaaan (~Un1x@CPE000024cccb7c-CM0014045acc3c.cpe.net.cable.rogers.com) |
| 19:05.07 | drmessano | Aretta has done some neat stuff with these stripped down, hosted FreePBX+Asterisk "Virtual PBX's".. It's insane how well they work with the specs they provision out for each VM |
| 19:05.11 | raden_work | gamedna, totally |
| 19:05.25 | Shaaan | has anyone ever run Asterisk with a large volume of calls 200+concurrent calls on HP Proliant Servers? |
| 19:05.27 | drmessano | gamedna: Exactly |
| 19:05.34 | Shaaan | or does anyone have any suggestions on hardware for 200 + concurrent calls? |
| 19:06.01 | gamedna | im a big fan of virtual + cloud but when you are deploying and enterprise or carrier grade app/service there is way too much outside of your control. |
| 19:06.07 | drmessano | raden_work runs a couple hundred concurrent calls, but I think he's using Vista |
| 19:06.36 | *** join/#asterisk Russ (~russ@206.29.188.232) |
| 19:07.06 | raden_work | Shaaan, dualcore w/ 4 GB ram |
| 19:07.39 | Shaaan | raden 200 concurrent calls + generating i would say additional 60 calls per minute or more its going to be a dialer.. |
| 19:07.53 | drmessano | Do you work for Obama? |
| 19:07.55 | raden_work | drmessano, Asterisk 1.6.0.10 built by root @ linux-zm7c on a i686 running Linux on 2009-07-27 20:11:39 UTC |
| 19:08.23 | drmessano | raden_work, http://en.wikipedia.org/wiki/Humour |
| 19:09.12 | gamedna | my take on cloud is that you really need some good partnerships w/ the last mile providers to be successful in the long run. Amazon / akamai work b/c they are so heavily interconnected. |
| 19:09.13 | raden_work | haha i hate windows never even used vista |
| 19:09.47 | gamedna | google also has been extremely innovative w/ this as well. |
| 19:09.53 | drmessano | gamedna: Yeah, nothing like rain taking out your cloud |
| 19:10.03 | gamedna | drmessano: hahaha |
| 19:10.21 | raden_work | Shaaan, all matters codecs etc too |
| 19:10.51 | Shaaan | well codecs will be g711 or ulaw |
| 19:11.13 | raden_work | drops head on desk |
| 19:11.26 | drmessano | Seriously though.. Put 25,000 users on Cloudmail.com and then explain to the CEO what "Backhoe Fade" is in layman's terms |
| 19:11.47 | raden_work | LMAO |
| 19:11.58 | raden_work | Shaaan, well g711 or ulaw ? |
| 19:12.11 | nix8n82 | I don't know what you mean by backhoe fade |
| 19:12.17 | gamedna | drmessano: hahahaha |
| 19:12.21 | Shaaan | g711 most likely |
| 19:12.37 | gamedna | i used to work in construction.. i know all about backhoe fade |
| 19:12.47 | raden_work | Shaaan, u sure u dont want to use ulaw ? |
| 19:12.53 | drmessano | Wikipedia ---> Backhoe fade or JCB fade is a humorous term coined by the telecommunications industry, referring to the accidental severing of a cable by a backhoe or similar construction activity. |
| 19:13.14 | drmessano | Deserves it's own entry, IMHO |
| 19:13.14 | Shaaan | okay lets base it on ulaw then... |
| 19:13.20 | raden_work | never done that 0:) |
| 19:13.20 | Shaaan | what are we looking in terms of hardware? |
| 19:13.42 | raden_work | Shaaan, you should really decide if your going to use ulaw or g711 |
| 19:13.50 | raden_work | also bandwith ? |
| 19:13.57 | gamedna | drmessano: in contrast to your comment, we have become "accepting" of backhoe fade w/ our home POTS service b/c there is a decent SLA attached to it |
| 19:14.33 | gamedna | drmessano: but we hate it if our internet goes down |
| 19:14.40 | Shaaan | raden_work, g711 and gigE |
| 19:14.48 | gamedna | b/c it could go down for no good reason and w/o an SLA |
| 19:15.09 | gamedna | i have 2 installs that use verizon FIOS |
| 19:15.22 | raden_work | what your WAN connection ? |
| 19:15.24 | wcselby | Shaaan - ulaw is g711, raden_work is just messing with you |
| 19:15.33 | gamedna | and i always get 1 phone line w/ it just so the SLA is there in case something goes wrong w/ the fiber line. |
| 19:15.37 | raden_work | wcselby, totally ruining my day dude :) |
| 19:15.48 | drmessano | gamedna: We've also become accepting of last mile fade on the near end because of occasional service outages.. which generally amount to the same SLA as you get from say something like Exchange run by a fully staffed department. But putting all your faith in cloudmail.com that its not run on a Comcast 12/2 connection takes a lot |
| 19:15.53 | Shaaan | thats what i thought i was a little confused, i was like hrmp something is wrong.. |
| 19:16.03 | raden_work | Shaaan, ok ill be nice so G711 on a dual core youll be able to handle roughly 250 calls going |
| 19:16.07 | raden_work | on a 2.5 |
| 19:16.08 | wcselby | Shaaan - for your 200+ concurrent calls, do you plan on doing this over digital trunks (T1 / E1, etc) or using an ITSP, because that will make a big difference |
| 19:16.12 | Shaaan | anyway can someone suggest a decent harware setup for lets say 200 concurrent calls per minute and another 100 being placed per minute |
| 19:16.27 | gamedna | drmessano: i agree⦠so then it boils down to the amount of risk your business can absorb.. |
| 19:16.29 | *** part/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 19:16.48 | Shaaan | wcselby, well the server is connected via GiGE, but the calls are being terminated via SIP |
| 19:16.56 | gamedna | IMHO |
| 19:17.00 | drmessano | gamedna: How the hell do I know RoxMyVox isn't running Trixbox on some GX240 with 512MB RAM? |
| 19:17.29 | wcselby | Shaaan - you'll need a very large pipe then. the server may have gige, do you mean to say that you've got 1000 MB/s bandwidth between your server and your provider? |
| 19:17.30 | gamedna | drmessano: i dont think it matters what they run, as long as the service they provide is good. |
| 19:17.43 | Shaaan | wcselby, yes |
| 19:17.43 | gamedna | that is the whole cloud mentality |
| 19:17.45 | raden_work | Shaaan, you actually have a full GIGE connection to your ISP ? |
| 19:17.46 | drmessano | gamedna: But I can start firing admins or buying new hardware when the in-house PBX doesn't work.. eventually something will budge enough that we're back up |
| 19:18.10 | gamedna | drmessano: right⦠but that is a business decision |
| 19:18.14 | raden_work | Shaaan, I find that hard to believe |
| 19:18.15 | drmessano | Yep |
| 19:18.19 | Shaaan | at my office, from my ISP wich is TiNEt |
| 19:18.34 | gamedna | here is another example. hosted PBX's for realtors |
| 19:18.40 | wcselby | Shaaan - nice. you still may want to look into a lower bit rate codec, such as g729 or whatever |
| 19:18.43 | gamedna | no office, no phones, no nothing |
| 19:18.44 | wcselby | but that adds to system overhead |
| 19:18.58 | gamedna | yet the system routes calls to each realtor's cell on the road |
| 19:19.07 | Shaaan | well hence i wanna get a decent server to handle the overhead |
| 19:19.08 | wcselby | grab yourself a nice dual quad-core server with 6+ gb of RAM, and you should be set |
| 19:19.11 | drmessano | Google just needs to start offering hosted PBX services and we can scratch that off the list of stuff we need to worry about it. |
| 19:19.17 | *** join/#asterisk ZeXr0 (~ZeXr0@ip216-239-95-218.vif.net) |
| 19:19.20 | Shaaan | but g729 is WAAAY to big of a resource hog |
| 19:19.28 | gamedna | there is a chance that mycloudpbx.fart all of a sudden gets hit by a meterorite and that whole business goes down |
| 19:19.35 | raden_work | g729 in passthrough is not a issue |
| 19:19.49 | carrar | g729 in hardware |
| 19:20.11 | gamedna | chances are that if you get a good virtualpbx in a good datacenter w/ a company that provides good service, you will be fine |
| 19:20.15 | wcselby | howlertech (I think) makes a g729 transcoding card that offloads the transcoding overhead associated with g729 |
| 19:20.33 | Shaaan | so your suggesting use g729 ? |
| 19:20.34 | wcselby | offloads from the cpu to the car |
| 19:20.36 | carrar | So does Digium |
| 19:20.47 | wcselby | Shaaan - if you find you're having bandwidth issues. |
| 19:21.01 | wcselby | rule of thumb is what, 100Kb/s per g711 call? |
| 19:21.19 | wcselby | 64Kb/s for the actual call, plus overhead |
| 19:21.34 | wcselby | you're talking 200-300 concurrent calls |
| 19:21.48 | Shaaan | in theory shouldn't a gigE have enough bandwidth to support it without having the need to compress it using g729 |
| 19:22.07 | wcselby | so 100Kb * 200 = 20,000Kb/s, or 20Mb/s |
| 19:22.10 | nix8n82 | I thought it was 64kb/s per rtp channel and isn't there two channels per call? |
| 19:22.22 | gamedna | shaaan: how many g711 can you realistically squeeze into a 1gigE line? |
| 19:22.30 | drmessano | gamedna: True, then you're also looking at "DIY" PBX + hardware hosting or a complete hosted turnkey.. Do you want a whole cloud, part of the cloud, or just the fluffy part? |
| 19:22.32 | wcselby | nix8n82 - one channel coming in, one channel to the phone, hopefully the phones aren't going over the same gigE pipe |
| 19:22.48 | Shaaan | i would think if its a burstable gigE |
| 19:22.58 | Shaaan | you should be able to get 20mb/s traffic to it no problem |
| 19:23.08 | wcselby | Shaaan - hopefully |
| 19:23.16 | wcselby | Shaaan - and my math may be off, it's been known to happen |
| 19:23.21 | gamedna | drmessano: i just want the silver lining =) |
| 19:24.02 | Shaaan | gigE is burstable according to my SLA upto 1000mB/s |
| 19:24.03 | carrar | Shaaan, you are getting a SLA for LINE RATE GigE? |
| 19:24.13 | wcselby | Shaaan - so again, like I said, a dual quad-core (2x quad core procs) box with 6+ gb of ram should be good. |
| 19:24.14 | Shaaan | err not SLA lost |
| 19:24.22 | *** join/#asterisk sahafeez (~sahafeez@65-119-47-100.dia.static.qwest.net) |
| 19:24.23 | carrar | GigE can really mean you can do 120Mbps |
| 19:24.27 | carrar | and no more |
| 19:24.33 | carrar | do you really know |
| 19:24.40 | nix8n82 | I'm a little confused, if you have an incoming sip call from your provider and it hits an ivr wouldn't that call take up 128kbs for that call as long as asterisk is playing the ivr? |
| 19:25.24 | carrar | g711 call is about 88kbps with overhead |
| 19:25.30 | Shaaan | so then what would my line capacity need to be according to your calculations i would need what like fios or fiber in there to handle that many calls ? |
| 19:25.43 | wcselby | ~book |
| 19:25.44 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 19:25.46 | wcselby | doh |
| 19:25.46 | carrar | 88kbps up and down |
| 19:25.50 | drmessano | or cable |
| 19:25.55 | raden_work | G.711 86.2 KBPS |
| 19:26.07 | carrar | .225 |
| 19:26.11 | drmessano | 50 meg, 100 meg DOCSIS 3.0 and lots of prayer |
| 19:26.15 | carrar | GET IT RIGHT! |
| 19:26.19 | raden_work | G.729 31.2 KBPS |
| 19:26.33 | Shaaan | lol |
| 19:26.44 | gamedna | carrar: you mean 120MB/s |
| 19:26.54 | raden_work | Shaaan, are you sure its a full GIGE connection ? |
| 19:26.54 | carrar | I mean Mbps |
| 19:27.01 | Shaaan | you guys are more confused then i am :| |
| 19:27.01 | raden_work | is it sync or async ? |
| 19:27.04 | Shaaan | radenn yes. |
| 19:27.16 | carrar | no one talks in bytes |
| 19:27.21 | raden_work | get a quad core 8 GB ram youll be fine |
| 19:27.25 | drmessano | G.726 (32 Kbps).......55.2 Kbps <-- PWN |
| 19:27.41 | Shaaan | ok |
| 19:27.44 | raden_work | where u get 55.2 from ? |
| 19:27.45 | drmessano | Also get lots of crack |
| 19:27.48 | raden_work | 47.2 |
| 19:28.00 | drmessano | G.726 (24 Kbps).......47.2 Kbps |
| 19:28.03 | Shaaan | Intel® Xeon® X7550 (8 core, 2.0GHz, 18MB, 130W) |
| 19:28.04 | drmessano | Reading, raden_work |
| 19:28.06 | drmessano | Reading |
| 19:28.12 | gamedna | carrar? gigE why only 120 megabits/sec |
| 19:28.15 | raden_work | Shaaan, overkill |
| 19:28.16 | drmessano | 32!! |
| 19:28.28 | carrar | gamedna, GigE can mean you can just burst over 100 Mbps |
| 19:28.34 | carrar | or it can mean LINE RATE |
| 19:28.38 | drmessano | Asterisk doesn't support 23 Kbps G.726, so fail |
| 19:28.41 | drmessano | or 24 |
| 19:28.43 | drmessano | typo |
| 19:28.51 | raden_work | Ah did not know that |
| 19:29.05 | carrar | lots of stuff can do "gige" but really can't sustain line rate gige throughput |
| 19:29.11 | raden_work | I dont understand why G.726 not more popular seems like a good codec never used it though |
| 19:29.32 | wcselby | how much bandwidth do the wideband codecs use? |
| 19:29.35 | drmessano | raden_work: Because you can bootleg G.729 |
| 19:29.37 | raden_work | Shaaan, and then you need a switch and a router that can handle that much through output |
| 19:29.42 | carrar | gige for general laymans term is it auto negotiates to 1000mbps |
| 19:29.55 | [TK]D-Fender | raden_work: because much lower BW options come close enough in quality. |
| 19:29.58 | carrar | not really a thruoghput rate of 1000Mbps |
| 19:30.22 | raden_work | [TK]D-Fender, makes sense and G.729 licensing is cheap in all reality |
| 19:30.26 | carrar | typically line rate gige is much more expensive |
| 19:30.31 | wcselby | why is everyone argueing with him about gige. |
| 19:30.41 | [TK]D-Fender | raden_work: GSM is also much lighter |
| 19:30.43 | carrar | give us something else to argue abo ut |
| 19:30.45 | carrar | !!! |
| 19:30.54 | wcselby | if he's supposed to be getting gige from his isp, then he should be getting it. he can take up specifics with them |
| 19:30.57 | raden_work | never played with gsm |
| 19:30.58 | Corydon76-dig | g729 is expensive, though, in terms of CPU usage |
| 19:31.01 | [TK]D-Fender | 13 vs 32 |
| 19:31.06 | drmessano | "Dark fiber" <--- Argue |
| 19:31.10 | wcselby | he asked about server hardware, which several people have given him suggestions for |
| 19:31.10 | carrar | OH MAN |
| 19:31.17 | Shaaan | lol |
| 19:31.21 | carrar | start your own ISP with DARK FIBER |
| 19:31.23 | raden_work | Corydon76-dig, only if you actually decording |
| 19:31.26 | drmessano | BUY AMD.. INTEL SUCKS |
| 19:31.27 | raden_work | right ? |
| 19:31.30 | carrar | Should do that |
| 19:31.33 | drmessano | WANNA FIGHT ABOUT IT? |
| 19:31.36 | carrar | YEAH |
| 19:31.39 | drmessano | Hows that? |
| 19:31.42 | raden_work | LMAO |
| 19:31.43 | carrar | Step outside pls |
| 19:31.45 | Corydon76-dig | raden_work: No, encoding also takes a lot of CPU |
| 19:31.52 | wcselby | drmessano - MS > MAC |
| 19:31.54 | Shaaan | Corydon76-dig, any suggestons from you and [TK]D-Fender and from you? |
| 19:31.55 | wcselby | there, fight |
| 19:32.00 | drmessano | lol |
| 19:32.07 | drmessano | Ubuntu IS INDEED the best distro |
| 19:32.11 | drmessano | ^^^ |
| 19:32.12 | carrar | heh |
| 19:32.14 | wcselby | MS > all OS |
| 19:32.15 | raden_work | Corydon76-dig, well yes both ways the translation but in pass through it doesnt put any extra load on the cpu does it ? |
| 19:32.17 | wcselby | now, fight |
| 19:32.22 | carrar | body slams drmessano |
| 19:32.23 | carrar | NO WAY |
| 19:32.30 | raden_work | drmessano, suse is and AMD sucks ! |
| 19:32.32 | carrar | SAY IT |
| 19:32.34 | Corydon76-dig | raden_work: if it's truly passthru, there's no extra load, no |
| 19:32.50 | wcselby | i actually prefer Mint to Ubuntu for desktops, and CentOS / Red Hat over Debian for servers, but it doesn't really matter all that much in the end |
| 19:32.58 | drmessano | hurls Edukubuntuxanadubuntu disks at carrar |
| 19:33.08 | raden_work | LMAO |
| 19:33.19 | carrar | Ubuntu is a good desktop |
| 19:33.29 | wcselby | but in the end, windows 7 > all |
| 19:33.31 | wcselby | :P |
| 19:33.32 | drmessano | Ubuntu is a good desktop, server, and coffee maker |
| 19:33.36 | carrar | great desktop actually |
| 19:33.44 | Shaaan | Corydon76-dig, any suggestions dude on hardware and line for a dialer thats going to have 200 concurrent calls? |
| 19:33.46 | gamedna | IBM PC JR all the way!!!! |
| 19:33.56 | carrar | FreeBSD FTW |
| 19:34.09 | drmessano | FreeBSOD is more like it |
| 19:34.15 | raden_work | Shaaan, dude a P4 will handle it get a freaking quad and 8GB of ram and youll seriously be fine |
| 19:34.29 | gamedna | drmessano: FreeBSOD = windows vista that came w/ your netbook |
| 19:34.52 | drmessano | hahah |
| 19:34.59 | raden_work | and id personally recommend running RAID 1 w/ spare drive on controller for automatic failover |
| 19:35.06 | gamedna | hehehe |
| 19:35.17 | Shaaan | raden_work its also the fact that i need to know take into consideration if i use g726 or g729 |
| 19:35.29 | *** join/#asterisk phix (~threat@123-243-44-131.tpgi.com.au) |
| 19:35.32 | raden_work | a quad will handle that much G729 |
| 19:35.43 | drmessano | BSD is for for guys who have the extra time not kissing girls to screw around with a *nix variant made intentionally difficult |
| 19:35.44 | raden_work | only time it taxing the CPU is when using IVR or doing translation |
| 19:35.47 | drmessano | THERE, I SAID IT |
| 19:36.21 | gamedna | drmessano: i thought that was AIX |
| 19:36.24 | KavanS | agrees :) with drmessano! |
| 19:36.38 | wcselby | freebsd is easier than openbsd to work with :P |
| 19:36.55 | drmessano | wcselby: I feel so sad for you, man |
| 19:36.59 | KavanS | I like *bsd core on osx |
| 19:37.13 | carrar | FreeBSD makes the girls hot |
| 19:37.16 | wcselby | heh |
| 19:37.24 | wcselby | that little devil penguin man |
| 19:37.28 | wcselby | does it every time |
| 19:37.45 | gamedna | wcselby: haha |
| 19:37.46 | wcselby | i think that's where p3nguin get's his attitude from |
| 19:37.52 | Corydon76-dig | bsd core on osx makes the developers punch holes in the wall |
| 19:37.58 | drmessano | wcselby, "I know one BSD variant from another so well I can point out their core differences" = "I really need a woman. Even a fat, ugly Waffle House waitress with no teeth" |
| 19:38.13 | wcselby | drmessano - hahaha |
| 19:38.17 | gamedna | HAHAHA |
| 19:38.37 | wcselby | life is good here in my mother's basement, what can I say |
| 19:38.51 | carrar | drmessano, move to Alaska |
| 19:39.01 | wcselby | i live in houston, i didn't know what a basement was until I was 13 |
| 19:39.05 | drmessano | wcselby: "Mooooom, knock before you enter my LAIR, PLEASE" |
| 19:39.05 | gamedna | wcselby: you just reminded me of will ferrel in wedding crashers |
| 19:39.25 | jdoe | off-topic, why can I not make a digitmap for a polycom phone that accepts 9[2-9]xxxxxxxxx and [2-9]xxxxxxxxx? It always matches the shorter so when I dial an initial 9 I get a truncated number :/ |
| 19:39.57 | raden_work | Shaan, get like a quad 2.5 , 8 GB RAM, Hardware based NIC, RAID 1 w/ 3rd drive for fail over you'll be set for a while |
| 19:40.19 | wcselby | you need a 4x 6-core server with 48GB of ram, dell makes them for their blade chassis |
| 19:40.31 | raden_work | LMAO |
| 19:40.33 | drmessano | Oh, and get one of those Dell hats |
| 19:40.34 | [TK]D-Fender | jdoe: Change the order they appear in the string |
| 19:40.48 | wcselby | i have a client with a few of those in their blade chassis |
| 19:40.52 | jdoe | [TK]D-Fender: I've tried both ways, neither works. |
| 19:40.54 | drmessano | I need to get servers with weatherproofing on them |
| 19:41.10 | gamedna | wcselby: your client must be poor now⦠j/k |
| 19:41.13 | wcselby | originally bought to be a vmware cluster, then they were going to be a sql cluster, now I think they're running MS Hyper-V |
| 19:41.22 | wcselby | i think the server is only like 9k |
| 19:41.42 | wcselby | they've probably got 250k in servers sitting in their datacenter |
| 19:41.51 | wcselby | servers / network equipmnet |
| 19:41.52 | drmessano | MS Hyper-V? OMG, they must be running Office 2010 |
| 19:42.06 | [TK]D-Fender | jdoe: Do what I do then.. screw specific patterns and jsut slap a timeout on EVERYTHING |
| 19:42.08 | wcselby | drmessano - some of them in their it department are, yes |
| 19:42.27 | drmessano | wcselby: 48GB RAM would be a minimum for Office, would it not? |
| 19:42.36 | gamedna | HAHAHA |
| 19:42.40 | wcselby | about an 80% ms shop, with about 20% linux / other |
| 19:43.01 | wcselby | although they did get juniper network gear because (seriously) one of their old network admins liked that it ran on BSD |
| 19:43.18 | wcselby | that, plus it was like half the cost of an equivalent cisco deployment |
| 19:43.24 | [TK]D-Fender | [15:39]<raden_work>Shaan, get like a quad 2.5 , 8 GB RAM, Hardware based NIC, RAID 1 w/ 3rd drive for fail over you'll be set for a while <- 4 drive RAID 6 |
| 19:43.25 | jdoe | [TK]D-Fender: yeah... sigh. Kinda wish they'd made the firmware suck less before they EOL'd it. |
| 19:43.30 | wcselby | although now none of their staff understand juniper gear |
| 19:43.51 | [TK]D-Fender | jdoe: Its list of shortcomings is comparatively small.... |
| 19:44.10 | raden_work | [TK]D-Fender, you need 5 drives to do raid 6 |
| 19:44.22 | drmessano | Juniper gear is just like anything else.. There's nothing you can't do with it once you pass 12 certification exams |
| 19:44.35 | raden_work | [TK]D-Fender, the performance increase would be tremendous though |
| 19:44.43 | raden_work | I personally run raid 10 w/ failover drive |
| 19:45.17 | jdoe | [TK]D-Fender: depends on how old your gear is. I'm stuck on 2.1.3, it has some especially annoying bugs. My favorite is how it forces the phone to wipe and reimage itself every time it restarts, so it takes a couple minutes every time to test the digitmap ;) |
| 19:45.23 | [TK]D-Fender | raden_work: "Diagram of a RAID 6 setup, which is identical to RAID 5 other than the addition of a second parity block" <- Raid5 + 1 MORE drive. So 3 drives for the % + 1 = 4 |
| 19:45.29 | [TK]D-Fender | 5* |
| 19:45.37 | [TK]D-Fender | (instead of %) |
| 19:45.55 | gamedna | TK, if you are going to deal w/ 2 drive failures w/ raid 6, you are better off w/ raid 10 |
| 19:46.02 | gamedna | more performance |
| 19:46.06 | drmessano | I like HP's RAID devices.. Lose a drive on a 3 drive RAID 5 array and it dynamically reallocates disk space across the other drives on the controller until you restore the drive |
| 19:46.24 | wcselby | who uses raid anymore, I just build my systems on 256gb thumb drives |
| 19:46.27 | [TK]D-Fender | gamedna: No, RAID 10 can fail if 2 drives fail. |
| 19:46.37 | [TK]D-Fender | gamedna: RAID 6 @ 4 drives survives |
| 19:47.00 | drmessano | RAID 5 @ 2 drives survives with an HP DL380 series lol |
| 19:47.47 | [TK]D-Fender | drmessano: Yes... in a 3-drive array. But that's nto the point :) |
| 19:48.01 | [TK]D-Fender | drmessano: That 2-drive solution doesn't survive 2 failures :) |
| 19:48.07 | drmessano | True |
| 19:48.14 | KavanS | raid? dude - booting from floppy works just fine! |
| 19:48.44 | citywok | gamedna: you aren't guaranteed 2 drive failures in raid10 |
| 19:48.59 | citywok | that's a real bad strategy to go with. if both drives in a mirror fail you're toast |
| 19:49.17 | gamedna | right, that is true |
| 19:49.27 | citywok | btw i've seen that happen. it's no fun. |
| 19:50.01 | gamedna | thanks for setting me straight citywok |
| 19:50.01 | gamedna | ;) |
| 19:50.30 | gamedna | i never use raid anymore, i just use cloud services |
| 19:50.43 | *** join/#asterisk Thorn_ (~thorn@unaffiliated/thorn) |
| 19:50.49 | citywok | you must not have lots of storage requirements, or you have a massive budget |
| 19:51.03 | drmessano | How do you reboot a cloud? |
| 19:51.10 | carrar | He had money belonging to him in Nigeria from some King |
| 19:51.14 | citywok | ec2-reboot-instance |
| 19:51.15 | drmessano | haha |
| 19:51.15 | carrar | he's rich |
| 19:51.16 | gamedna | citiwok: im just playing on the previous conversation |
| 19:51.28 | citywok | that should do the trick drmessano |
| 19:51.41 | citywok | ah, i didn't read that far back gamedna |
| 19:51.46 | gamedna | drmessano: dont reboot, be root. |
| 19:51.59 | drmessano | citywok: You're 0% fun, and that's with NO drive failures |
| 19:52.08 | KavanS | you don't need to reboot the cloud |
| 19:52.17 | [TK]D-Fender | [15:50]<gamedna>i never use raid anymore, i just use cloud services <- that means you have no idea where your actual data is or what its odds of survival are :) |
| 19:52.17 | KavanS | isn't that the entire idea? |
| 19:52.25 | citywok | hah. what happens if the cloud crashes? does it rain? |
| 19:52.44 | [TK]D-Fender | citywok: Oh I feel it coming down again.... |
| 19:52.46 | gamedna | [TK]D-Fender: im not serious about that comment⦠guess you missed the cloud debate a while back. |
| 19:52.53 | drmessano | KavanS: Clouds have no place to insert floppy disks.. Have you ever tried to stick a floppy in a cloud? |
| 19:53.11 | citywok | drmessano: not in a cloud, just an apple pie. but it wasn't very floppy at the time. |
| 19:53.11 | KavanS | drmessano, I'm floppying the cloud as we speak! |
| 19:53.12 | [TK]D-Fender | drmessano: I did. It tripped the water sensor |
| 19:53.22 | citywok | lmao |
| 19:53.25 | gamedna | hahaha |
| 19:53.42 | drmessano | What happens when the cloud says "Please insert disk in drive A:" ? |
| 19:53.53 | gamedna | you press the any key |
| 19:54.00 | drmessano | lol |
| 19:54.11 | [TK]D-Fender | looks for the "any" key |
| 19:54.18 | citywok | i have alt! |
| 19:54.29 | *** join/#asterisk philipp64|laptop (~chatzilla@75-92-150-245.war.clearwire-wmx.net) |
| 19:54.34 | drmessano | I tried to plug a thumb drive into a cloud once, and it fell straight down and hit me in the face. |
| 19:55.01 | raden_work | [TK]D-Fender, your correct Im so used to using it with a hot spare LOL 5 was the number in my head |
| 19:55.07 | drmessano | So much for telling Laslo to reboot 3 times |
| 19:55.17 | raden_work | LMAO |
| 19:55.33 | drmessano | ITS THE GREY ONE |
| 19:55.36 | drmessano | "They're all grey" |
| 19:55.41 | [TK]D-Fender | raden_work: RAID 6 witha hot spare? Thats like 3 non data drives... |
| 19:55.41 | gamedna | heh |
| 19:55.50 | [TK]D-Fender | raden_work: paranoid much? |
| 19:56.00 | drmessano | "Dude, you just rebooted the Exchange server" |
| 19:56.21 | raden_work | [TK]D-Fender, I dont like liability, I'm sure you can understand.... |
| 19:56.23 | gamedna | [TK]D-Fender, whats your chassis drive limit? |
| 19:57.19 | [TK]D-Fender | raden_work: Sure you can do it.. it just sounds extreme. RAID 6 was already a jump up on RAID5 + HOT because it survives 2 simultaneous without waiting to rebuild the hot spare. |
| 19:57.31 | drmessano | Could be worse.. could be a RAID .5.... that's one drive and a spare on the shelf |
| 19:57.48 | KavanS | raid is meh |
| 19:57.52 | [TK]D-Fender | gamedna: I don't run 6 here yet. I have 5+a currently |
| 19:57.57 | gamedna | drmessano: yea, and then you plug in the spare to find out it does not work |
| 19:58.19 | *** join/#asterisk ideaman (~ihaveapla@c-174-52-20-94.hsd1.ut.comcast.net) |
| 19:58.26 | raden_work | [TK]D-Fender, raid 6 nice.... i normally run 10 + hot spare |
| 19:58.27 | gamedna | <PROTECTED> |
| 19:58.31 | drmessano | gamedna: You're assuming I got to image the new drive from the bad drive before it died |
| 19:58.47 | raden_work | [TK]D-Fender, yes totally paranoid :( |
| 19:58.48 | KavanS | gamedna, for a couple bucks more you could have the entire system redundant - why stop at the drives? ;) |
| 19:59.12 | gamedna | kavans, that is what i use. |
| 19:59.18 | gamedna | redundant everything |
| 19:59.22 | KavanS | sweet |
| 19:59.27 | drmessano | He even studders in meetings |
| 19:59.36 | gamedna | yyou you you betcha |
| 19:59.42 | drmessano | Just so his words are redundant |
| 19:59.58 | gamedna | im so redundant i say the same thing twice |
| 20:00.09 | drmessano | I keep a spare fiance' in the spare bedroom just in case |
| 20:00.27 | gamedna | drmessano: i thought you used cloudfiance.com? |
| 20:00.34 | raden_work | gamedna, our asterisk servers run a round robin DNS with Mysql Master master both with raid 10 and redundant power supplies witch is probally a lil much |
| 20:00.45 | gamedna | sounds nice |
| 20:00.47 | drmessano | gamedna: Like I said earlier, have you ever tried to stick.... oh nevermind |
| 20:00.55 | gamedna | HAHAHA |
| 20:01.08 | raden_work | gamedna, its in testing for last 3 months working good so far |
| 20:01.11 | gamedna | raden_work: how do you handle registration? |
| 20:01.29 | drmessano | He doesnt have any phones |
| 20:01.33 | drmessano | He went over budget on the servers |
| 20:01.33 | gamedna | ah |
| 20:01.34 | raden_work | mysql, master master |
| 20:01.37 | gamedna | hahaha |
| 20:02.02 | drmessano | Sorry, had to.. |
| 20:02.05 | raden_work | lol |
| 20:02.20 | raden_work | gamedna, they basically clone each other |
| 20:02.22 | *** part/#asterisk kkm (~kkm@76.91.228.152) |
| 20:02.47 | drmessano | Ok, out for a few.. I will be back.. (Yes, that's a warning) |
| 20:02.51 | raden_work | gamedna, the master master makes the 2 mysql databases one |
| 20:03.03 | raden_work | throws drmessano outa the room :) |
| 20:03.18 | gamedna | drmessano: dont spend too much time in the clouds⦠|
| 20:03.21 | *** join/#asterisk pabelanger (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
| 20:03.23 | KavanS | oh snap |
| 20:03.25 | raden_work | LMAO |
| 20:03.31 | raden_work | ROFLMAO |
| 20:04.13 | gamedna | raden_work: so each asterisk server can register based on the mysql database |
| 20:04.22 | gamedna | you dont need a single registration server |
| 20:04.50 | raden_work | like how do we register to our ITSP ? |
| 20:05.14 | gamedna | right |
| 20:05.16 | raden_work | our how do phones resiter to asterisk ? |
| 20:05.20 | gamedna | well, both |
| 20:05.36 | gamedna | each server have a connection to your ITSP |
| 20:05.42 | raden_work | correct |
| 20:05.53 | raden_work | well we are a itsp but we connect to another one |
| 20:06.01 | gamedna | i figured w/ that setup |
| 20:06.02 | gamedna | :) |
| 20:06.11 | raden_work | ok here is how it works |
| 20:10.21 | *** join/#asterisk eppigy (~eppigy@c-76-105-72-69.hsd1.ga.comcast.net) |
| 20:12.35 | Shaaan | http://h10010.www1.hp.com/wwpc/us/en/sm/WF06b/15351-15351-3328412-241644-3328422-4142916-4160032-4160078.html |
| 20:12.40 | Shaaan | thats a NICE SERVER! |
| 20:13.59 | carrar | Sure is |
| 20:14.02 | carrar | PLS SEND |
| 20:14.45 | Shaaan | Carrar, i think that would alone handle the 200 calls per miute no problem :) |
| 20:14.49 | Shaaan | 4 x 6 cores |
| 20:15.12 | carrar | doesn't take much to handle 200 calls per min |
| 20:15.25 | Shaaan | well how about generating 200 calls |
| 20:15.31 | Shaaan | or even 100 calls per minutes |
| 20:15.53 | carrar | Just passing SIP? |
| 20:16.00 | carrar | 100 calls a second |
| 20:16.41 | Shaaan | maybe i should get a cheaper server and buy like 5 of them and do a cluster |
| 20:16.45 | Shaaan | much more effective i would think |
| 20:16.55 | carrar | What are you trying to do, you need to know that first |
| 20:17.53 | carrar | opensips states it's doign 900 per second |
| 20:18.17 | carrar | So if all you are doing is sip setup |
| 20:18.31 | carrar | perhaps I missed your original question |
| 20:19.22 | Shaaan | building a Dialer, and Terminating those calls via SIP |
| 20:19.27 | Katty | hugs carrar |
| 20:19.36 | carrar | KATTYHUGGLES Katty!! |
| 20:19.44 | carrar | Thats a special hug!! |
| 20:20.02 | Katty | :>> |
| 20:20.27 | carrar | Shaaan, dialer that runs on the users desktop or on the SIP Server? |
| 20:20.36 | carrar | or do you really need a full PBX? |
| 20:20.44 | Shaaan | need a full PBX |
| 20:20.51 | Shaaan | i was thinking asterisk with GNUDialer |
| 20:20.55 | carrar | Asterisk will handle your requirements |
| 20:20.57 | Shaaan | to do voice broadcasting |
| 20:21.08 | carrar | easy I am s ure |
| 20:21.08 | Shaaan | with live ttansfer to agents |
| 20:21.29 | carrar | You are only doing 3-4 calls per secodn |
| 20:21.50 | Shaaan | voice broadcasting with 20 agents your doing about i would say 60 calls per second easy |
| 20:22.06 | Shaaan | and alot of people wont even pick up or most will and wont press 1 |
| 20:22.09 | carrar | WHy voice broadcasting? |
| 20:22.14 | carrar | Why not overhead paging? |
| 20:22.18 | Shaaan | LOL! |
| 20:23.01 | carrar | Always start with the best machine you can get |
| 20:23.05 | gamedna | overhead paging = throwing something over the cube walls |
| 20:23.07 | leifmadsen | why not underchin paging?! |
| 20:23.21 | Shaaan | comon guys the conversation is going where its not supposed to |
| 20:23.23 | carrar | underchin causes cancer |
| 20:23.50 | gamedna | sorry shaan |
| 20:23.58 | carrar | Where is this suppose to go? |
| 20:24.22 | gamedna | (to the clouds)? |
| 20:25.05 | carrar | I imagine you could get hardware to run this on for under $300 |
| 20:25.11 | carrar | ebay |
| 20:25.34 | carrar | doesn't need to be a $12k HP |
| 20:25.41 | carrar | unless you want it to be |
| 20:25.50 | Shaaan | seems almost untrue how you are going to get 300 hardware to do about 200 concurrent calls and generating another atleast 100 per minute |
| 20:26.07 | carrar | now we are at 200 concurrent calls? |
| 20:26.10 | *** join/#asterisk jaxyeh (~jaxyeh@c-69-250-52-161.hsd1.md.comcast.net) |
| 20:26.11 | carrar | before it was 100 |
| 20:26.17 | Shaaan | no, i said 200 before too |
| 20:26.22 | carrar | 100 per min |
| 20:26.26 | Shaaan | 100 per min yes. |
| 20:26.27 | carrar | I guess |
| 20:26.49 | carrar | Does Asterisk have to proxy the RTP? |
| 20:27.02 | carrar | or will your firewall do that? |
| 20:27.21 | Shaaan | hrmp not decided on that one yet but i would be safe to guess Asterisk |
| 20:27.58 | carrar | and will Asterisk have T1 connected to it? |
| 20:28.01 | carrar | or 100% sip |
| 20:28.23 | carrar | Any transcoding? |
| 20:28.43 | ZeXr0 | tzafrir_laptop : Do you know if it's possible to use an Astribank using a virtual machine with CentOS on it ? Using a vmware server ESXi ? |
| 20:29.51 | Shaaan | it will be connected to a GigE with SIP termination so 100% sip |
| 20:31.00 | carrar | So you can do just do g.711 then |
| 20:31.53 | carrar | Let Asterisk off load the RTP over to a firewall or something |
| 20:32.22 | carrar | then it's all about just call setup for the most part |
| 20:32.35 | carrar | cept for playing recordings, vm and stuff |
| 20:32.42 | Shaaan | no vm |
| 20:32.45 | Shaaan | just recordings |
| 20:32.55 | carrar | lots of recordings? |
| 20:33.03 | Shaaan | no just 1 for everyone |
| 20:33.06 | carrar | also are you RECORDING the audio/calls? |
| 20:33.08 | citywok | if they are encoded in the same codec as the channel, it should be pretty simple |
| 20:33.43 | citywok | recording is a beast, i can make about 200 calls simul recorded on a dual xeon 3.6. not sure how high i can go w/out recording i haven't tried. |
| 20:34.19 | Shaaan | well i wont be recording the calls |
| 20:34.33 | Shaaan | we will be doing recording tho |
| 20:34.39 | Shaaan | like basicly last 1 minute of the conversation |
| 20:34.58 | carrar | You really don't need a huge machine then if you are NOT recording calls |
| 20:35.24 | carrar | put your audio files in a memory drive |
| 20:35.33 | carrar | or record to a memory drive |
| 20:35.37 | tzafrir_laptop | ZeXr0, not sure. I heard reports both ways |
| 20:35.39 | [TK]D-Fender | [16:34]<Shaaan>like basicly last 1 minute of the conversation <- doesn't work that way. |
| 20:35.55 | [TK]D-Fender | Shaaan: Can't have only the last minute if you don';t knwo when the call will end |
| 20:35.58 | citywok | i can record to disk w/ 200 calls (scsi), but it's not IO limited it's cpu lol. |
| 20:36.11 | [TK]D-Fender | Shaaan: You may only KEEP the last minute, but thats actually even MORE processing after |
| 20:36.15 | Shaaan | well |
| 20:36.27 | Shaaan | we would stop and start recording by doing *somethin and stoping it by *something |
| 20:36.43 | [TK]D-Fender | Shaaan: When the other end hangs up on you you'll have nothing. |
| 20:36.49 | citywok | why do you only need the last minute? |
| 20:36.51 | carrar | ugg |
| 20:36.56 | Shaaan | okay forget recordings |
| 20:36.58 | [TK]D-Fender | Ok, checkout time, BBIAB |
| 20:36.59 | Shaaan | we wont do recordings |
| 20:37.03 | carrar | heh |
| 20:37.07 | citywok | you can record only bridged calls to make sure it only records while people are talking to your agents |
| 20:37.11 | Shaaan | can just use third party then i assume |
| 20:37.32 | carrar | You can always us a softphone and record your calls :) |
| 20:37.34 | citywok | like most projects, i'd suggest you come up with a defined requirements list. |
| 20:37.38 | Shaaan | its okay forget recording ) |
| 20:37.45 | citywok | that's a GREAT starting place. knowing what you need. |
| 20:38.04 | carrar | well if you want recordigns keep it on the list |
| 20:38.15 | Shaaan | my starting place is trying to figure out my hardware and what software to use for the voicebroadcasting someone suggested GNUDialer |
| 20:38.26 | carrar | also then Asterisk will also then be proxing the RTP |
| 20:38.31 | carrar | that addes overhead |
| 20:38.39 | citywok | if you dont know your requirements how do you know how to size your hardware? |
| 20:38.39 | carrar | well handing the RTP |
| 20:39.16 | Shaaan | well asterisk is going to handle the RTP |
| 20:39.32 | carrar | IF you don't know your requirements yet then just get the fastest machine you can afford and hope for the best! |
| 20:39.48 | carrar | Shoot the moon! |
| 20:39.54 | citywok | that works :P |
| 20:40.18 | citywok | quad 6 core procs, 256gb of memory, and 8 256gb SSD's should suffice. |
| 20:40.25 | carrar | haha |
| 20:40.31 | Shaaan | lol |
| 20:40.49 | Shaaan | Fine i'll make a list of req and pastebin it just for you. |
| 20:40.58 | carrar | Thats HOT++ |
| 20:41.06 | citywok | a project is much more likely to succeed with a list of requirements. |
| 20:41.40 | *** part/#asterisk punxos (~punxos@9.pool80-103-173.dynamic.orange.es) |
| 20:42.20 | carrar | citywok, stop being practical!! |
| 20:42.37 | citywok | sorry. and would you believe i'm not even a project manager? |
| 20:44.23 | dohd | citywok: you're being practical, so yes, I'd believe that |
| 20:44.27 | *** join/#asterisk buttons840 (~buttons84@207.224.213.42) |
| 20:44.51 | citywok | lolol. sorry. i'm just a sysadmin / bad programmer :P |
| 20:45.42 | buttons840 | does anyone know what form the callerid field takes in the AMI originate command? |
| 20:48.47 | *** join/#asterisk dailylinux (~test@s21-00210.dsl.no.powertech.net) |
| 20:50.07 | *** join/#asterisk orioni (~chatzilla@95.107.225.207) |
| 20:50.31 | orioni | hi there , i have an incoming sip and i want to transfer with h323 , is this possible and how ? |
| 20:50.47 | *** join/#asterisk defsdoor (~andy@plingit.gotadsl.co.uk) |
| 20:50.57 | Shaaan | carrar, citywok, http://pastebin.org/449469 |
| 20:52.19 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:52.47 | gamedna | hey shaan: instead of putting your eggs in one basket, why not get a bunch of cheaper servers |
| 20:53.02 | gamedna | instead of one expensive one |
| 20:53.28 | gamedna | based on what i am seeing you are probably better off w/ a distributed system |
| 20:53.28 | carrar | Hows that distributed queue? |
| 20:53.56 | Shaaan | hrmp |
| 20:54.13 | gamedna | that way failiure is now acoss multple servers |
| 20:54.31 | leifmadsen | distributed queues are hard to do with Asterisk if you're expecting to have the same queue broken between several servers -- you can't transmit call position across servers. |
| 20:54.47 | carrar | yup |
| 20:54.52 | gamedna | dont do the queue in asterisk⦠|
| 20:55.03 | gamedna | have an app that uses a DB backend |
| 20:55.17 | gamedna | each server keeps polling for calls |
| 20:55.19 | gamedna | grabs 10 at a time |
| 20:55.23 | Shaaan | so your saying its easy to do a simple que but a distributed one cant be done? |
| 20:55.34 | carrar | "easy" |
| 20:55.35 | leifmadsen | not trivially |
| 20:56.20 | Shaaan | Okay updated. |
| 20:56.21 | Shaaan | http://pastebin.org/449475 |
| 20:56.24 | leifmadsen | if app_queue was modified to transmit queue position to calls across XMPP or something then yes, it could be done |
| 20:56.26 | gamedna | the outbound side is pretty easy |
| 20:56.35 | leifmadsen | everything "can" be done with code changes |
| 20:56.40 | leifmadsen | yay open source! |
| 20:56.49 | carrar | yay!! |
| 20:56.50 | Shaaan | finding a competent developer is the hard part. |
| 20:56.55 | leifmadsen | at the drop of a consultant you can have anything done |
| 20:57.12 | Shaaan | leifmadsen, are you volunteering to be a consultant? |
| 20:57.20 | carrar | yay for consultants!!! |
| 20:57.37 | leifmadsen | I am a consultant, but I don't do code changes, sorry |
| 20:57.41 | leifmadsen | point at pabelanger |
| 20:57.54 | Shaaan | why dont you do code changes? |
| 20:58.10 | leifmadsen | because I don't program in C |
| 20:58.19 | carrar | excuses!!! |
| 20:58.22 | gamedna | haha |
| 20:58.34 | Shaaan | lol |
| 20:58.40 | leifmadsen | I only write documentation and dialplan :) |
| 20:58.44 | pabelanger | waves |
| 20:58.47 | leifmadsen | and PHP when necessary |
| 20:58.54 | *** join/#asterisk myster (~myster@207.148.172.210) |
| 20:59.00 | leifmadsen | pabelanger: you should totally take up that consulting job and make app_queue distributable |
| 20:59.10 | leifmadsen | hawtness would abound |
| 20:59.47 | pabelanger | which? |
| 20:59.58 | leifmadsen | see scroll back |
| 21:00.07 | leifmadsen | <leifmadsen> if app_queue was modified to transmit queue position to calls across XMPP or something then yes, it could be done |
| 21:00.11 | leifmadsen | <Shaaan> leifmadsen, are you volunteering to be a consultant? |
| 21:00.42 | pabelanger | :) I just finished a contract today, so looking for new work |
| 21:00.52 | Shaaan | join the club :) |
| 21:01.04 | orioni | hi there , i have an incoming sip and i want to transfer with h323 , is this possible and how ? |
| 21:03.52 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:04.18 | carrar | man the blue angels are all over here |
| 21:04.31 | carrar | buzzin my house |
| 21:05.59 | *** join/#asterisk Shaaan (~Un1x@CPE000024cccb7c-CM0014045acc3c.cpe.net.cable.rogers.com) |
| 21:07.45 | *** join/#asterisk Russ (~russ@206.29.188.232) |
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| 21:13.03 | *** join/#asterisk buttons840 (~buttons84@207.224.213.42) |
| 21:14.15 | pabelanger | orioni: Yes, just bridge the channels within asterisk |
| 21:14.51 | pabelanger | orioni: CLI*> core show application Dial |
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| 21:17.37 | citywok | carrar: you in seattle? i'll be out there on saturday on the big boat :) |
| 21:17.40 | carrar | I am |
| 21:17.53 | carrar | (Bellevue) |
| 21:17.58 | citywok | ahh. Redmond :) |
| 21:19.18 | carrar | I did the whole boat thing one year, barely made it back to the dock and home, I leave town now when Seafair goes on :) |
| 21:19.32 | citywok | heh. our ceo owns a 95' tugboat. |
| 21:20.03 | carrar | Is that Bill |
| 21:20.20 | citywok | it's awesome b/c the blue angels use it as a landmark for turning, so they are routinely buzzing over the top of it. you can smile at the pilots if they are inverted. lol. |
| 21:20.28 | carrar | err guess it's Steve now |
| 21:20.36 | citywok | no, i don't work directly for msft |
| 21:22.05 | citywok | btw your resume says Currently working in renton, but it says april2004 - nov2006 |
| 21:22.54 | rootlinux | question: can i use the F of Dial comand ption on Asterisk 1.4? |
| 21:24.11 | bmoraca_work | rootlinux: "core show application dial" on 1.4.30 shows only a lowercase f option, no uppercase F |
| 21:25.41 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 21:27.17 | rootlinux | bmoraca_work, ok.. thanks i have the 1.4.24 and only see the f lowercase |
| 21:27.26 | carrar | what resume |
| 21:28.01 | carrar | oh |
| 21:28.03 | carrar | haha |
| 21:28.09 | carrar | currently in that time frame |
| 21:28.10 | carrar | :) |
| 21:28.45 | carrar | I'm not going anyplace soon |
| 21:28.52 | carrar | working from home is hard to beat |
| 21:30.42 | Shaaan | Carrar, did you see my paste bin? |
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| 21:31.00 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 21:31.01 | carrar | the 1st one |
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| 21:35.32 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:36.44 | p3nguin | I guess no one ever proof read that resume. |
| 21:40.49 | *** join/#asterisk kerpal (~tyler@63.135.231.207) |
| 21:41.17 | kerpal | my asterisk doesn't transcode and i see error reference to slin when i try to playback an mp3 via format_mp3.so |
| 21:41.35 | kerpal | if gsm (phone) -> ulaw (gizmo) no audio |
| 21:41.54 | kerpal | if ulaw(phone) -> ulaw (gizmo) audio |
| 21:42.30 | kerpal | MP3Player, and Playback work when called upon via extension |
| 21:42.49 | kerpal | Playback doesn't play mp3 though, even though I have the module loaded. |
| 21:43.40 | kerpal | [Aug 5 17:43:12] WARNING[26813]: app_playback.c:474 playback_exec: ast_streamfile failed on SIP/tyler-21dde548 for /tmp/Tool |
| 21:44.06 | kerpal | -- <SIP/tyler-21dde548> Playing '/tmp/Tool.slin' (language 'en') |
| 21:44.07 | kerpal | <PROTECTED> |
| 21:46.45 | [TK]D-Fender | kerpal: Doesn't look like it chose an MP3 file to play... |
| 21:48.24 | citywok | carrar: out of curiosity i checked your whois and went to your website :) |
| 21:49.07 | drmessano | Porn and Ducks, huh? |
| 21:49.22 | citywok | yea. donkeys. |
| 21:49.55 | carrar | h4X0r |
| 21:50.47 | drmessano | carrar: out of curiosity i checked your whois and posted your website on slashdot :) |
| 21:51.00 | drmessano | The good news is, nobody goes to slashdot anymore |
| 21:51.05 | carrar | haha |
| 21:53.00 | leifmadsen | slashdot still exists? |
| 21:53.03 | leifmadsen | crazy! |
| 21:53.40 | Russ | I think they have some script that just reruns random stories every few years |
| 21:53.56 | *** join/#asterisk Z_God (~julius@2001:610:1908:8000:21e:ecff:fe5d:679e) |
| 21:53.57 | kerpal | [Aug 5 17:53:11] WARNING[27053]: file.c:650 ast_openstream_full: File /tmp/Tool.mp3 does not exist in any format |
| 21:54.09 | drmessano | carrar: I also just posted your website on Google Wave and submitted it to Lycos |
| 21:54.33 | kerpal | i tried to rename to mp3, i heard if you have format_mp3.so you could load the file but you would have to remove the end from the file. ill try that desperately here in a sec. |
| 21:54.37 | Russ | you should send them his geocities site too |
| 21:55.02 | drmessano | Russ: No need, his angelfire site has much more content |
| 21:55.08 | kerpal | but really i want to know why ulaw->gsm doesn't work and why mp3player doesn't work via pbxspool (no sound) |
| 21:55.36 | kerpal | pbxspool calls |
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| 22:01.46 | [TK]D-Fender | kerpal: Don't jsut show use the error, show us the COMPLETE call, and the file itself |
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| 22:08.44 | kerpal | how does one call format mp3 using playback()? |
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| 22:18.47 | [TK]D-Fender | kerpal: How do we SEE you calling it? |
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| 23:14.12 | cmendes0101 | I'm having trouble registering sip with Broadvoice. Getting Unauthorized insip debug mode. Would that only be related to the user/pass? or could there be settings that would deny it? |
| 23:18.22 | *** join/#asterisk ecrane (~ecrane@o1-69-19-166-10.static.o1.com) |
| 23:28.36 | [TK]D-Fender | cmendes0101: pastebin the SIP debug of your failed attempts |
| 23:28.38 | [TK]D-Fender | ~pb |
| 23:28.39 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 23:50.11 | cmendes0101 | Hmm well I ended up reenting the sip information. I don't see an unathorized anymore but now it still doesn't register http://pastebin.com/8G76qwzk |
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