| 00:09.37 | *** join/#asterisk spiceycurry (~mcurry@2002:63f7:e93e:0:5ab0:35ff:fe71:4ac1) |
| 00:10.12 | *** join/#asterisk spiceycurry (~mcurry@proxy.hostopia.com) |
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| 00:15.03 | *** part/#asterisk ming_zym (~ming_zym@124.160.123.67) |
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| 00:38.45 | *** join/#asterisk darg (~djc@65.209.147.101) |
| 00:39.19 | darg | Ok. I'm at a loss to explain this. I have a Dial, that rings a SIP device. following that at the next prio, I have a voicemail |
| 00:40.02 | darg | what happens, is the device rings until it times out, and then instead of going to voicemail, it disconnects the channel |
| 00:40.19 | darg | the console never shows it even trying to run the vicemail |
| 00:40.50 | p3nguin | What is the timeout value in your Dial()? |
| 00:40.53 | darg | 35 |
| 00:41.18 | p3nguin | Just for the heck of it, turn it down to something obscenely short, like 12. |
| 00:41.31 | darg | ok one sec |
| 00:41.50 | p3nguin | I'm thinking maybe the phone is dumping the channel after an internal timeout. |
| 00:42.12 | darg | hrm |
| 00:42.20 | darg | ok, time to check the spa config then |
| 00:42.25 | darg | it went to vm that tim |
| 00:42.26 | darg | e |
| 00:42.33 | darg | ty |
| 00:43.27 | p3nguin | My Cisco 7912G has a ring timeout value in it, which can be used to handle voicemail or forwarding at the phone level rather than at the call control device. Your phone might have something similar. |
| 00:43.46 | darg | its a spa-200 |
| 00:43.53 | darg | phone is just an analog phone |
| 00:43.55 | darg | er |
| 00:43.56 | darg | 2000 |
| 00:44.23 | p3nguin | I don't have experience with that ATA, but there is probably some type of timeout setting that is causing the behavior. |
| 00:44.35 | *** join/#asterisk aidinb (~Aidin@ip70-187-172-87.oc.oc.cox.net) |
| 00:44.39 | darg | theres a bunch of 'timer' settings, but they have wonderfully useless names like |
| 00:44.47 | darg | "SIP timer B", "SIP timer H" |
| 00:45.33 | p3nguin | Look for some timer with a value of around 30 seconds or whatever time your call was falling apart. |
| 00:45.52 | darg | well, theres 5 of them set to 32 |
| 00:45.59 | p3nguin | oh no! |
| 00:46.00 | darg | F, D, B H and J |
| 00:46.08 | p3nguin | Time to dig out a manual. |
| 00:46.17 | darg | yeah im googling |
| 00:47.19 | *** join/#asterisk ming_zym (~ming_zym@121.0.29.237) |
| 00:47.56 | darg | well i found some info, but it isn't really useful |
| 00:47.57 | *** join/#asterisk dieno (773f883e@gateway/web/freenode/ip.119.63.136.62) |
| 00:48.04 | darg | B is 'invite time out" |
| 00:48.07 | dieno | hi every one |
| 00:48.14 | darg | F is 'non-invite time out'. |
| 00:48.20 | darg | I may just say fsck it and set them all to 40 |
| 00:48.21 | *** join/#asterisk pabelanger (~pabelange@CPE001fe2a8fd1d-CM0012254094b2.cpe.net.cable.rogers.com) |
| 00:48.39 | p3nguin | Or one at a time until you find the right one. |
| 00:48.54 | darg | but who cares if I add 8 seconds to some other timer? |
| 00:48.56 | darg | not gonna break anyway |
| 00:48.58 | darg | er |
| 00:48.59 | darg | anything |
| 00:49.25 | dieno | can any one please explain me from my log am i having pri configured correctly |
| 00:49.25 | dieno | http://pastebin.ca/1881456 |
| 00:49.33 | darg | either that O I suppose I could use 30 for the dial timeout |
| 00:49.39 | darg | lemme see how that works |
| 00:50.47 | darg | must be too close.. it dumped it |
| 00:51.22 | darg | i'll stay at 30 but change all those from 32 to 35 |
| 00:52.18 | darg | hrm |
| 00:52.20 | darg | still no joy |
| 00:54.43 | spiceycurry | darg: set asynchronous balanced mode extended |
| 00:54.45 | dieno | hmmm any one with pri experience please |
| 00:54.56 | spiceycurry | thats a problem |
| 00:55.07 | darg | spiceycurry, say what? |
| 00:55.19 | darg | thats a spa setting or something in *? |
| 00:55.22 | spiceycurry | the hopeful resolution is that you screwed up your timing config |
| 00:55.36 | spiceycurry | lets see your span configs |
| 00:55.48 | darg | oh yer talking to dieno |
| 00:55.52 | darg | his PRI |
| 00:55.57 | spiceycurry | ok lol |
| 00:56.14 | spiceycurry | dieno: lets see your span config lines- you prob screwed up your timing |
| 00:56.44 | darg | p3nguin, any way I can tell * to go fall through if theres no answer even if the spa dumps the call? |
| 00:57.04 | darg | to be honest, i'd want it to go to vm for any sort of error. |
| 00:57.07 | p3nguin | You could check sip debug and see what is going on. |
| 00:57.20 | darg | only time it shouldnt is if the call is actually answered |
| 00:57.29 | p3nguin | Generally, I don't want my phones or devices to make any of their own decisions. |
| 00:58.05 | darg | ok, sip sebug.. gonna run a call and see wht I get |
| 00:58.07 | spiceycurry | dieno: your timing settings are probably wrong, or your card is screwed. |
| 01:02.08 | Nugget | My friend just bought a new car, it only drives in reverse. It's a Dis lexus. |
| 01:02.34 | p3nguin | Reverse is better than nothing, I guess. |
| 01:03.05 | p3nguin | I need the help of an audiophile for my car's problem. |
| 01:03.21 | darg | http://pastebin.com/r1kEPnsU |
| 01:03.33 | darg | theres the sip debug, if anyone better at reading that than me wants to look |
| 01:03.51 | darg | ive obfuscated some phone#'s and IP's |
| 01:04.21 | *** join/#asterisk blaines (~blaines@75-171-121-6.phnx.qwest.net) |
| 01:05.18 | darg | im gonna see if 25 for the dial timeout works |
| 01:06.45 | dieno | spiceycurry can you please let meknow what configuration you will be want to see |
| 01:06.46 | darg | ok that seems to work.. but its really short than id like |
| 01:07.13 | darg | *er |
| 01:07.29 | *** join/#asterisk bjhaid (~IceChat7@41.220.68.4) |
| 01:07.42 | spiceycurry | dieno: are you using dahdi or zap? |
| 01:07.50 | dieno | spiceycurry if you want i will paste my chan_dahdi and dahd-channels.conf |
| 01:07.52 | dieno | dahdi |
| 01:08.13 | spiceycurry | dieno: pastebin /etc/dahdi/system.conf |
| 01:09.15 | dieno | spiceycurry there it is http://pastebin.ca/1881464 |
| 01:10.31 | spiceycurry | do you have a warrany on that card? |
| 01:10.38 | dieno | eyup |
| 01:10.46 | dieno | and surprisingly i just bought it |
| 01:10.51 | dieno | like a few hours ago |
| 01:10.53 | dieno | :) |
| 01:10.54 | spiceycurry | you'll want to call them up |
| 01:11.41 | dieno | OHKae |
| 01:11.41 | dieno | but what will would explain them any idea ? |
| 01:11.41 | spiceycurry | they'll do some loop tests |
| 01:11.41 | *** join/#asterisk SaiSoma (~chatzilla@216.109.13.105) |
| 01:11.41 | dieno | oh got it |
| 01:11.41 | spiceycurry | tell them your getting "set asynchronous balanced mode extended" |
| 01:11.46 | spiceycurry | thats a problem with the timing |
| 01:12.08 | dieno | ok got it thanks for your time mate |
| 01:12.40 | spiceycurry | no prb, good luck |
| 01:13.40 | dieno | spiceycurry btw one more question please |
| 01:13.55 | spiceycurry | sure |
| 01:14.18 | dieno | do you think thats the configuration will be used in outbound or is it going to out/in both like FXO |
| 01:15.15 | spiceycurry | dont follow you |
| 01:15.53 | dieno | does this configuration is good with outboung calls ? |
| 01:16.06 | dieno | if this card is replaced |
| 01:17.10 | spiceycurry | that config is only for the hardware... lets see your: /etc/asterisk/dahdi-channels.conf, /etc/asterisk/chan_dahdi.conf |
| 01:17.23 | spiceycurry | you using analog? |
| 01:17.49 | spiceycurry | hey- |
| 01:17.55 | bjhaid | I run asterisk on my ubuntu machine, I would want to know how I can make asterisk run at start-up so if I am not present someone with a poor knowledge of linux can get the asterisk box started by starting the ubuntu machine |
| 01:18.51 | dieno | spiceycurry please take a look at it http://pastebin.ca/1881468 |
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| 01:19.58 | *** part/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 01:21.01 | dlynes | bjhaid, if you installed asterisk from the binary on ubuntu 10.04 LTS, it would already be running when you reboot |
| 01:21.12 | spiceycurry | dieno: looks good. did you setup your extensions.conf? |
| 01:21.16 | p3nguin | darg: I guess SIP/2.0 487 Request Terminated is where the phone says "Stop calling me!" |
| 01:21.25 | dieno | yup all smooth with round robin |
| 01:21.27 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 01:22.15 | dieno | spiceycurry thanks again :) |
| 01:22.23 | *** join/#asterisk aidinb (~Aidin@ip70-187-172-87.oc.oc.cox.net) |
| 01:22.47 | spiceycurry | np |
| 01:23.36 | bjhaid | well dlynes, I use ubuntu 9.10 and it does not run until i initialise it from the terminal |
| 01:24.05 | spiceycurry | bjhaid: service asterisk start |
| 01:24.23 | darg | p3nguin, hrm.. ok.. but obviously that doesnt help figure out how to tell it to quit that |
| 01:24.40 | p3nguin | There has to be a manual somewhere. |
| 01:25.05 | bjhaid | spiceycurry where do i include service asterisk start? |
| 01:25.14 | darg | well, i have a manual.. but it doesnt really explain anything.. its just a reference |
| 01:25.18 | spiceycurry | you dont, that will start it |
| 01:25.39 | bjhaid | i type it at the terminal? |
| 01:25.58 | spiceycurry | bjhaid: yes... also- look for a file named asterisk in /etc/init.d |
| 01:26.32 | bjhaid | spiceycurry, what do i do to the file? |
| 01:26.42 | dlynes | bjhaid, /etc/init.d/asterisk start |
| 01:26.50 | drmessano | O.o |
| 01:27.03 | darg | http://corp.deltathree.com/productsandservices/manuals/sipura.pdf |
| 01:27.17 | bjhaid | spiceycurry: i change the name of the file or? |
| 01:27.28 | spiceycurry | no |
| 01:27.53 | spiceycurry | run it |
| 01:27.59 | darg | hrm.. |
| 01:28.16 | spiceycurry | if it is there, it will start when booted- so long as your configs are ok |
| 01:28.55 | bjhaid | Spiceycurry, I am running on windows right now, I would get that done later and if I have problems would check the channel tommorow, thanks |
| 01:29.17 | *** join/#asterisk ming_zym (~ming_zym@121.0.29.237) |
| 01:29.41 | drmessano | I could have sworn there was a decision that all installed "servers" in Ubuntu would run at startup |
| 01:30.26 | spiceycurry | p |
| 01:30.28 | spiceycurry | np |
| 01:30.54 | *** join/#asterisk Doc (~scott@2001:470:1:8::2) |
| 01:31.10 | Doc | anyone got any recommendations for a good/free softphone that supports multiple accounts? |
| 01:32.57 | shido6_ | zoiper |
| 01:34.21 | Doc | is it stable now days? i havent tried it for a fair while, but it had a habit of crashing a lot... |
| 01:34.46 | exothermc | I'm trying to load dahdi but I'm getting Running dahdi_cfg: DAHDI_SPANCONFIG failed on span 1: Invalid argument (22) |
| 01:35.00 | exothermc | hmm not sure why that inverted sorry. |
| 01:35.54 | darg | sigh |
| 01:35.56 | darg | still no joy |
| 01:37.57 | darg | found this: |
| 01:38.02 | darg | http://www.freepbx.org/book/export/html/7508 |
| 01:38.11 | darg | look for "more than one minute" |
| 01:38.23 | darg | but mine is stopping at 30s.. |
| 01:38.34 | darg | i checked those values, they are 60.. for grins I made them 90 |
| 01:38.35 | darg | no change |
| 01:39.17 | spiceycurry | exothermc: pastebin your dahdi config file /etc/dahdi/system.conf |
| 01:42.48 | exothermc | spiceycurry: http://pastebin.ca/1881476 |
| 01:43.51 | *** join/#asterisk zyphlar (~z@wsip-70-182-59-230.ph.ph.cox.net) |
| 01:43.58 | zyphlar | hey there my lovelies |
| 01:44.28 | zyphlar | [TK]D-Fender you there? i fixed that awesome freepbx pri issue thanks to digium |
| 01:45.03 | spiceycurry | exotermc: get rid of ",yellow" |
| 01:45.15 | exothermc | spiceycurry: ok ya it was before, but didn't matter. |
| 01:45.41 | zyphlar | question, i'm trying to use a dual t1 card to pass thru a PRI from my telco provider thru asterisk to another box |
| 01:45.54 | spiceycurry | you need to save the file, and at the CLI> prompt type... dahdi restart |
| 01:46.02 | zyphlar | should my 2nd span have channels 25-48, or should it be channels 1-24? |
| 01:46.11 | exothermc | spiceycurry: yup no dice. |
| 01:46.22 | spiceycurry | zyphlar: 25-48 |
| 01:46.33 | zyphlar | k thx |
| 01:48.31 | spiceycurry | exothermc: try stopping the asterisk service, than the dahdi service, and restart everything |
| 01:48.53 | exothermc | asterisk isn't even running, and I'm pretty sure it doesn't need to be yet. |
| 01:48.55 | *** join/#asterisk ming_zym (~ming_zym@121.0.29.237) |
| 01:49.01 | exothermc | restarting dahdi now. |
| 01:49.03 | spiceycurry | ok, stop the dahdi service |
| 01:49.05 | spiceycurry | k |
| 01:49.24 | exothermc | spiceycurry: same thing. |
| 01:49.36 | spiceycurry | repaste your new config |
| 01:49.37 | zyphlar | spiceycurry: know how i could troubleshoot that inner t1(pri) link not showing as connected? both cards are red despite looking ok in dahdi_cfg -vvv |
| 01:49.47 | zyphlar | it's set as pri_net as well |
| 01:50.19 | spiceycurry | zyphlar: what does dahdi_tool say? |
| 01:50.33 | zyphlar | span 1 is OK, span 2 is red |
| 01:50.44 | spiceycurry | is span 2 plugged in? |
| 01:50.51 | zyphlar | yep i'll triple check |
| 01:51.18 | spiceycurry | zyphlar: on the circuit (on the wall) are the lines all green? |
| 01:51.36 | darg | try swap the first and second spans, see if the problem is the card or the span itself? |
| 01:51.44 | exothermc | spiceycurry: http://pastebin.ca/1881482 |
| 01:52.33 | exothermc | http://pastebin.ca/1881483 |
| 01:52.45 | zyphlar | my guess is that span2 doesn't realize it's supposed to be pri_net |
| 01:53.13 | zyphlar | i used the same cable that was from telco->asteriskspan1 and plugged it into asteriskspan2->intertel and same deal |
| 01:53.26 | zyphlar | it's as tho asteriskspan2 isn't providing a signal which is what pri_net is supposed to do |
| 01:53.28 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 01:53.49 | zyphlar | both asteriskspan1 and intertel work with telco |
| 01:53.52 | exothermc | really odd that dahdi is complaining about the arguments of span 1. I'm 99% sure they are correct. |
| 01:53.55 | zyphlar | i'm inserting asterisk in between the two |
| 01:54.41 | darg | well I had a 52 page pdf.. now ive found a 137 page one.. maybe it has more useful info |
| 01:55.34 | spiceycurry | exothermc: weird, it works on mine- |
| 01:56.03 | spiceycurry | exothermc: Your timing is weird though- mine is 1,2,0 and 2,1,0 |
| 01:56.17 | spiceycurry | I also have 1 less channel on the second span |
| 01:56.28 | spiceycurry | as 48 is usually the next dchan |
| 01:56.46 | spiceycurry | zyphlar: could it be the cable? |
| 01:56.49 | exothermc | spiceycurry: you just use your second span as your primary timer where I use my first as the primary. |
| 01:57.09 | zyphlar | definitely not; again, same cable, works when connected from either PBX to my telco |
| 01:57.13 | zyphlar | just not from PBX-PBX |
| 01:57.16 | exothermc | spiceycurry: mine are configured with 1 dchan for both pris. |
| 01:57.19 | *** join/#asterisk coppice (~chatzilla@19.176.64.202.dyn.pacific.net.hk) |
| 01:57.32 | zyphlar | conclusion: asterisk span2 isn't providing a PRI |
| 01:57.37 | zyphlar | not sure how to troubleshoot |
| 01:58.03 | spiceycurry | exothermc: I would delete the file and recreate it, in the case there is some hidden char or something |
| 01:58.12 | spiceycurry | by chance did you "copy/paste" it? |
| 01:58.51 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 01:59.30 | spiceycurry | zyphlar, post your configs (pastebin) |
| 02:00.02 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 02:00.12 | darg | sigh.. nothing useful there either |
| 02:00.14 | darg | this sucks |
| 02:00.15 | zyphlar | spiceycurry: thanks http://pastebin.com/CAAa9t4T |
| 02:00.16 | exothermc | spiceycurry: I comment out the span lines and it loads just fine. |
| 02:00.25 | exothermc | from what I'm reading the card driver isn't loading. |
| 02:00.43 | darg | why the hell should this spa-2000 not let a phone ringer longer than 30s |
| 02:01.12 | zyphlar | spiceycurry: sorry http://pastebin.com/uVgjXt5a |
| 02:01.35 | darg | need to find a spa2000 irc channel |
| 02:01.58 | zyphlar | trying to specify timing as 0 on span2 |
| 02:03.25 | zyphlar | doesn't seem to help |
| 02:03.52 | darg | cisco bought sipura i see |
| 02:04.13 | spiceycurry | weird :O |
| 02:04.24 | spiceycurry | back soon, have to jet |
| 02:04.41 | xheliox | that happened like.. |
| 02:04.59 | xheliox | at least 4 years ago |
| 02:05.34 | xheliox | I hear the govt ordered a break up of AT&T |
| 02:05.36 | darg | ah.. well.. this rock I live under tends to filter out stuff like that |
| 02:05.55 | darg | :P |
| 02:06.20 | darg | so any idea why a spa-2000 might refuse to ring the phone longer than 30 seconds? |
| 02:06.32 | xheliox | Nope. |
| 02:06.51 | darg | incoming call, * dials the spa, it rings, if the timeout is about 30 or higher, the spa dumps the call, and * doesnt fall through to vm |
| 02:06.59 | zyphlar | AH! might be a t1 crossover |
| 02:07.19 | xheliox | Hmm. |
| 02:07.25 | xheliox | Ain't that interesting. |
| 02:07.30 | darg | the * dial() timeout, that is |
| 02:07.39 | darg | if I make it 25, it works fine.. but I really want 35 |
| 02:07.43 | xheliox | Surely there's a time out setting in the Supra? |
| 02:08.08 | darg | yeah.. i found some 'timer' values that were set at 32.. tried changing them to 40.. no dice |
| 02:08.22 | darg | thought I had it at first |
| 02:08.41 | xheliox | only way to know for sure is to look at the sip debug and determine who is requesting the hang up |
| 02:08.59 | darg | the spa is. |
| 02:09.17 | xheliox | Yeah, really couldn't say. |
| 02:09.17 | darg | p3nguin helped me find that |
| 02:09.31 | darg | the sip debig is here: |
| 02:09.35 | darg | http://pastebin.com/r1kEPnsU |
| 02:10.00 | zyphlar | woot! i had to make a t1 crossover cable in order to connect from asterisk pri to another local pri |
| 02:10.01 | darg | been googling and googling.. cant seem to find any explanation |
| 02:10.12 | zyphlar | i.e. pri_net doesn't work unless you've got a crossover cable |
| 02:10.31 | darg | zyphlar, all depends on the polarity of the t1 jack(s) you are connecting |
| 02:10.55 | darg | just like old ethernet ports before auto-mdi.. |
| 02:11.03 | darg | you cant connect a hub to a hub without a crossover |
| 02:11.06 | darg | ditt pc to pc |
| 02:11.28 | xheliox | darg: normally I'd dig deeper for you, but I can't offer any serious assistence at the moment. |
| 02:11.41 | darg | xheliox, np |
| 02:12.11 | darg | maybe the thing needs a cold reboot for those timers to take effect |
| 02:12.13 | darg | wtf its worth a try |
| 02:12.57 | p3nguin | I found something. |
| 02:13.00 | zyphlar | yeah i remember. just figured pri_net would configure the asterisk card to be opposite polarity |
| 02:13.13 | p3nguin | look under Distinctive Ring Patterns |
| 02:13.39 | exothermc | damn card was configured for E1 |
| 02:13.48 | p3nguin | darg: Look at the Ring Cadence settings. |
| 02:13.55 | darg | yeah i tried that already |
| 02:14.02 | p3nguin | darg: Tried which thing? |
| 02:14.05 | darg | they were set to 60.. for grins i changed them to 90 |
| 02:14.09 | darg | didnt change anything |
| 02:16.50 | darg | thanks though |
| 02:19.12 | *** join/#asterisk Trixboxer (~Trixboxer@115.124.115.69) |
| 02:21.05 | darg | well im done with it for tonight. |
| 02:21.16 | darg | thanks all for trying |
| 02:32.39 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.73) |
| 02:41.41 | zyphlar | dialplan question for you homies |
| 02:42.16 | zyphlar | if i want to Dial("DAHDI/g2/${EXTEN}") where ${EXTEN} equals the DID Asterisk received, how would I do that? |
| 02:42.35 | zyphlar | i.e. i want to transparently pass thru DIDs I receive that start with the number 6 |
| 02:43.01 | zyphlar | right now ${EXTEN} seems to be not passing the right things |
| 02:43.11 | zyphlar | not sure how to debug what it is tho |
| 02:43.32 | p3nguin | exten => _6X.,1,Dial(DAHDI/g2/${EXTEN}) |
| 02:43.36 | p3nguin | Like this? |
| 02:43.41 | zyphlar | hrm i'll try that |
| 02:47.10 | *** join/#asterisk DarkRift (~dark@modemcable015.68-200-24.mc.videotron.ca) |
| 02:52.30 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 02:53.21 | zyphlar | i'm using freepbx but nobody's responding in that channel... is there a way to add this trunk->trunk route in the gui, or do i hafta add it in a certain place? |
| 02:54.51 | p3nguin | ~freepbx |
| 02:54.51 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 02:54.54 | p3nguin | No clue. |
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| 03:17.10 | zyphlar | booya |
| 03:17.14 | zyphlar | got it |
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| 03:22.40 | Sedorox | Do variables in AEL get reset when moving between contexts? |
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| 03:46.13 | zyphlar | hey there seems to be static when calling from my inner pbx to the telco |
| 03:47.03 | zyphlar | is that a timing issue? if so what timing should i use? (it doesn't seem like my telco is providing timing since it always defaults to internally clocked) |
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| 03:57.59 | psilikon | zyphlar, i recently had an audio issue that was due to the tx gain being set too high that caused clipping. |
| 03:59.04 | psilikon | zyphlar, pri? |
| 03:59.12 | zyphlar | yep |
| 03:59.33 | zyphlar | i think it might actually be only due to echo cancellation since i'm talking to myself |
| 03:59.39 | zyphlar | inbound and outbound only seem fine |
| 03:59.49 | psilikon | what type of card? |
| 03:59.54 | zyphlar | just in case i set each span to timing=0 so it's internally clocked |
| 04:00.00 | zyphlar | uhhh te207p? |
| 04:00.04 | zyphlar | dual t1 digium |
| 04:00.30 | psilikon | never used that one. We mostly use sangoma A104's |
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| 04:05.04 | zyphlar | cool well thx all |
| 04:05.41 | zyphlar | btw i got that span-span forwarding done in freepbx by defining an inbound route to a custom destination. |
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| 06:14.40 | exothermi | anyone able to help me with a couple Qwest PRIs? |
| 06:17.45 | exothermi | All the channels keep going red no matter how I configure dahdi |
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| 06:25.22 | Jumpie | any recommendations of providers that have good rates for in/outbound AND sms forwarding? |
| 06:25.34 | Jumpie | lookin for diff 'one stop shop' options |
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| 07:40.31 | hariom | What is the best file format to record a file in asterisk? wav, gsm or WAV (wav49). My goal is to get the high quality recordings. |
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| 07:41.43 | Ziaeon | wav49 is good, good quality and a tenth the size of wav |
| 07:41.56 | Ziaeon | best quality and performance is wav of course |
| 07:42.04 | Ziaeon | but you'll fill up the drive quick, depending on volume |
| 07:42.20 | Ziaeon | I use a custom script to convert everything to mp3, but thats for a specific purpose... |
| 07:43.34 | hariom | Considering that I get about 25 calls per hour, what do you suggest? These recording should be of good quality (good in the sense audible difference compared to other) |
| 07:43.34 | florz | Ziaeon: hard disks have become larger in recent years, in case you haven't notices ... |
| 07:44.11 | hariom | I have been using sox to convert wav to gsm. But .gsm files are not very good when played back. |
| 07:44.12 | Ziaeon | florz: you are obviously underestimating the volume of my customers :) |
| 07:44.39 | Ziaeon | im talking about a 20,000 tollfree bill a month. |
| 07:44.43 | Ziaeon | debt consolidation call centers =/ |
| 07:44.50 | florz | Ziaeon: 20000 frogs? |
| 07:45.16 | Ziaeon | might as well be frogs with what USD is worth these days |
| 07:45.29 | florz | *g* |
| 07:45.30 | hariom | Ziaeon: do you suggest to record as .wav? |
| 07:45.46 | Ziaeon | if you have a big harddrive, and not a lot of volume, why not |
| 07:46.03 | florz | Ziaeon: that would be in the region of 2 mio minutes, then? |
| 07:46.14 | hariom | ok |
| 07:46.39 | hariom | Is it possible to record directly in Ogg or Mp3 format? |
| 07:46.50 | Ziaeon | no you have to save in wav and convert it to mp3 |
| 07:46.54 | Ziaeon | I have a script for such purpose |
| 07:46.59 | hariom | ok |
| 07:47.00 | Ziaeon | and you have to edit the dialagi |
| 07:47.39 | hariom | I use simply a shell script with sox to convert all wav to ogg or gsm |
| 07:47.44 | florz | hariom: that really only makes sense if you have to send them by email or somesuch, otherwise you are just creating complication and CPU overhead |
| 07:48.16 | hariom | Yes, I need to send recording back to my clients. |
| 07:48.29 | florz | Ziaeon: that would make ~ 1 TB of data to record all of it as uncompressed WAV ... !? |
| 07:48.36 | Ziaeon | just record in wav49 then, its good quality and emailable size |
| 07:49.00 | Ziaeon | i get so many recordings, that I cant delete them! |
| 07:49.03 | Ziaeon | if I do rm * |
| 07:49.06 | Ziaeon | it says "too many arguments" |
| 07:49.07 | Ziaeon | lol |
| 07:49.43 | Ziaeon | (i can delete the directory, though) |
| 07:50.11 | florz | then you obviously must first have deleted the files inside ... |
| 07:50.29 | Ziaeon | -rf |
| 07:50.50 | florz | yeah, still that first deletes the files inside ;-) |
| 07:51.19 | florz | find . -type f -maxdepth 1 -print0 | xargs -0 rm would do just as well |
| 07:52.26 | florz | (and probably would even be faster due to the pipe buffer in between the two |
| 07:52.27 | florz | ) |
| 07:52.39 | Ziaeon | yeah |
| 07:52.47 | Ziaeon | I use find also to delete files that are older than a month or two months etc |
| 07:52.48 | Ziaeon | for maintenance |
| 07:53.10 | Ziaeon | yeah but for some reason, doing rm -rf on the dir works, where as rm * in the dir gives error of too many arguments |
| 07:53.37 | Ziaeon | probably a linear argument instead of a batch job |
| 07:53.43 | florz | well, for the obvious reason that one of them has only two arguments, while the other has thousands? |
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| 07:55.44 | Ziaeon | right |
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| 08:09.17 | jblack | The key part is the "-r" which stands for "recursive" |
| 08:10.55 | florz | hmm? |
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| 08:52.37 | ChannelZ | ok WTF does 'context=from-trunk' mean? |
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| 09:02.05 | ChannelZ | nevermind.. I thought 'from-trunk' was maybe a special keyword or something but apparently not, these people just don't know what they're doing is all |
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| 10:24.16 | khussein78 | i need to convert cisco 7911 to SIP to connect to asterisk server ? where can i find the image and show to upgrade |
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| 10:36.02 | [sr] | yelolow |
| 10:36.05 | [sr] | ops |
| 10:36.08 | [sr] | yellow |
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| 10:41.46 | [sr] | im interested in know from where came the name "asterisk" to the asterisk project |
| 10:42.13 | BarthezZ | * |
| 10:42.28 | [sr] | yes but why chose that name |
| 10:42.44 | [sr] | it could be comma => , |
| 10:42.51 | [sr] | or dot, => . |
| 10:42.56 | [sr] | just curious |
| 10:43.42 | troffasky | nah |
| 10:43.52 | troffasky | comma and dot aren't on a 12 button keypad! |
| 10:45.11 | [sr] | i see |
| 10:45.16 | [sr] | but it could cardinal then => # |
| 10:45.16 | [sr] | :p |
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| 10:46.17 | [sr] | asterisk sounds better its a fact |
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| 11:21.31 | jblack | [sr]: irc is not twitter. You don't need to hit the enter key every 14 characters |
| 11:33.28 | *** join/#asterisk replifs (~ho@ip-70-160.dobrich.net) |
| 11:34.03 | replifs | Hi, I order VoIP number but the people told me that I need to send them VoIP Gateway information, to connect to the phone via SIP softphone |
| 11:34.19 | replifs | does I understand this right? :) |
| 11:35.01 | replifs | I need to get VoIP Gateway to connect to the phone with softphone? |
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| 11:36.33 | replifs | I install VoIP Gateway, set SIP settings, connect to the VoIP Gateway with setted SIP settings and finally VoIP Gateway connects to their servers? |
| 11:36.37 | replifs | is that how it works? :) |
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| 11:53.45 | [sr] | replifs: you order a number to use with the provider you use, maybe voipbuster? |
| 12:01.34 | replifs | I don't know they just give me a number |
| 12:01.44 | replifs | and they tell me - if I want to dial mobile phones |
| 12:01.47 | [sr] | but voipbuster? |
| 12:01.55 | replifs | or mobile phones call me |
| 12:01.59 | replifs | I need to use softphone |
| 12:02.06 | replifs | to use softphone I need to set SIP details |
| 12:02.20 | replifs | but I can't connect with them directly, I need to have VoIP Gateway |
| 12:02.24 | replifs | :) |
| 12:02.30 | replifs | I don't know just some voip number |
| 12:02.34 | replifs | what is voip buster? |
| 12:02.36 | [sr] | who are them? |
| 12:02.44 | replifs | BulgariaOpen |
| 12:02.47 | replifs | in my country |
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| 12:03.37 | [sr] | for sure u just need to use a sip client and enter the details, but the best is to ask them help |
| 12:03.51 | [sr] | but u'll hit the call center for sure and that is going to be painfull |
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| 12:15.15 | replifs | yes thank you |
| 12:15.25 | replifs | that's what I'm going to do :) |
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| 15:32.37 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 16:12.01 | KavanS | parsing "sip show channels" to see if an extension is in use - would it be safe to say that Rx: ACK is a safe assumption for the Last Message: of a sip channel that is in use? |
| 16:12.11 | KavanS | i.e. call in progress |
| 16:12.52 | *** join/#asterisk darkskiez (~dz@72-254-56-42.client.stsn.net) |
| 16:17.59 | [TK]D-Fender | KavanS: "core show hints" <- and set each one up with one. |
| 16:19.00 | [TK]D-Fender | KavanS: Or make a an extension that dumps "ChanisAvail" resulst out somewhere. Hints are more practical, and less work though and usable elsewhere |
| 16:19.23 | *** join/#asterisk residentgrey (~residentg@72-59-146-187.pools.spcsdns.net) |
| 16:21.01 | KavanS | [TK]D-Fender, during a call via core show hints - my SIP/501 is showing as "idle" |
| 16:23.09 | KavanS | I looked into ChanisAvail and also ExtensionState - chanisavail didn't seem to fit the bill |
| 16:23.25 | KavanS | ChanIsAvail is not a solution to tell you conclusively whether the channel is busy or not, it is primarily to tell you whether it would be possible to send a call there. Whether that call would end up being accepted or not is entirely up to the peer that we send the call to, and they could easily reject the call even though they do not appear to be 'busy'. |
| 16:23.55 | [TK]D-Fender | [12:21]<KavanS>[TK]D-Fender, during a call via core show hints - my SIP/501 is showing as "idle" <- set your peer up right |
| 16:23.56 | KavanS | basically I'm trying to determine if a line is in use/call is in progress for a certain SIP peer - if it is idle/ringing - I am not concerned about that |
| 16:24.11 | *** join/#asterisk ming_zym (~ming_zym@124.160.123.67) |
| 16:24.14 | [TK]D-Fender | [12:23]<KavanS>ChanIsAvail is not a solution to tell you conclusively whether the channel is busy or not <- incorrect |
| 16:24.18 | KavanS | ok - configuration error? :\ calls seem to work fine on |
| 16:24.34 | KavanS | ok - chanisavail definition I pulled from voip-info.org |
| 16:24.43 | [TK]D-Fender | KavanS: And you CAN'T know if the device will actively reject you. There is no dodging that bullet |
| 16:24.56 | KavanS | well I'm not worried if the device is going to reject me |
| 16:25.02 | KavanS | I just want to see if it is making a call over the system |
| 16:25.17 | KavanS | i.e. if someone is on the line |
| 16:27.27 | KavanS | peer I'm testing (501) is setup with qualify=yes |
| 16:29.31 | [TK]D-Fender | KavanS: menngless... |
| 16:29.41 | [TK]D-Fender | meaningless* |
| 16:29.46 | [TK]D-Fender | KavanS: Pastebin your peer |
| 16:30.01 | [TK]D-Fender | KavanS: nvm |
| 16:30.13 | [TK]D-Fender | KavanS: Should be : type=peer , call-limit=99 |
| 16:30.18 | [TK]D-Fender | kavait'd BETTER be |
| 16:32.33 | *** part/#asterisk peterosd (~Peter@c83-253-39-232.bredband.comhem.se) |
| 16:45.29 | KavanS | call limit is set |
| 16:47.25 | *** join/#asterisk coppice (~chatzilla@202.64.176.19) |
| 16:48.54 | [TK]D-Fender | KavanS: type=peer? |
| 16:49.38 | KavanS | whoa - missing that line for this peer |
| 16:49.39 | KavanS | :\ |
| 16:49.43 | KavanS | trying again |
| 16:50.06 | KavanS | <PROTECTED> |
| 16:50.13 | *** join/#asterisk BANSAL (~bansal@117.199.123.196) |
| 16:50.19 | KavanS | state is still idle with type=peer and call-limit=99 - via core show hints |
| 16:50.45 | [TK]D-Fender | KavanS: [general] limitonpeers=yes |
| 16:50.52 | *** join/#asterisk eliel (~eliels@186.18.131.44) |
| 16:50.56 | [TK]D-Fender | kava mod all, and reload SIP, and pastebin the lot.... |
| 16:51.47 | KavanS | ok - modded with limitonpeers |
| 16:52.24 | KavanS | ok - shows as inuse now |
| 16:52.35 | KavanS | via core show hints |
| 16:56.01 | *** join/#asterisk ManxPower (~manxpower@234.sub-75-235-252.myvzw.com) |
| 16:59.09 | [TK]D-Fender | \o/ |
| 17:06.15 | *** join/#asterisk prinzdezibel (~michaelje@p57B084BA.dip0.t-ipconnect.de) |
| 17:10.41 | KavanS | ok - getting some results here that are worthwhile |
| 17:11.14 | *** join/#asterisk ming_zym (~ming_zym@124.160.123.67) |
| 17:15.15 | *** join/#asterisk BANSAL (~bansal@117.199.123.196) |
| 17:22.52 | *** join/#asterisk kartik (~koolkarti@117.199.113.215) |
| 17:30.38 | *** join/#asterisk pabelanger (~pabelange@CPE001fe2a8fd1d-CM0012254094b2.cpe.net.cable.rogers.com) |
| 17:31.36 | *** join/#asterisk pabelanger (~pabelange@CPE001fe2a8fd1d-CM0012254094b2.cpe.net.cable.rogers.com) |
| 17:32.54 | *** join/#asterisk pabelanger (~pabelange@CPE001fe2a8fd1d-CM0012254094b2.cpe.net.cable.rogers.com) |
| 17:50.47 | *** join/#asterisk [sr] (~Unknowned@pa8-84-91-197-8.netvisao.pt) |
| 17:51.22 | [sr] | hi WIMPy, i think i finally have lcr and asterisk working! now i have a setup problem on the incoming route to send call's to my sip extension |
| 17:56.14 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 17:59.40 | *** join/#asterisk ming_zym (~ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 18:00.13 | [TK]D-Fender | [sr]: Perhaps you should show us your call and the problem with it |
| 18:01.22 | *** join/#asterisk kartik (~koolkarti@117.199.113.215) |
| 18:01.23 | [sr] | [TK]D-Fender: http://pastebin.com/r4v6qfj1 |
| 18:01.47 | [sr] | i see there "due to extension missmatch" |
| 18:01.52 | [sr] | (you're going to kill me now) |
| 18:01.53 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 18:02.14 | [sr] | i've setup an inbound route, without specifying any DID, and set the destination to a sip user |
| 18:04.43 | [sr] | i'm not getting the part of "extension missmatch" |
| 18:04.59 | [TK]D-Fender | [sr]: There clearly isn't a match in extensions.conf for exten=234377460 context=Ext_hfc4s_1 complete=no) |
| 18:05.21 | [TK]D-Fender | [sr]: It is telling you EXACTLY what it's looking for in extensions.conf... this isn't even a guess. |
| 18:08.16 | [sr] | [TK]D-Fender: going to try with the docs samples |
| 18:11.30 | *** join/#asterisk Pimmetje (~Pimmetje@83.119.156.35) |
| 18:12.05 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 18:13.12 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 18:13.23 | [sr] | [TK]D-Fender: a small help, extern=5,1,Dial(SIP/my_sip_number) |
| 18:13.30 | [sr] | exten sorry |
| 18:13.45 | *** join/#asterisk Alagar (~Administr@122.164.39.36) |
| 18:13.50 | [TK]D-Fender | [sr]: it isn't LOOKING for "5". It's looking for "234377460" |
| 18:14.01 | [TK]D-Fender | exten=234377460 <--------------------------------- |
| 18:15.04 | [sr] | lets see what is going to happen.... |
| 18:15.30 | [sr] | [TK]D-Fender: just perfect! :D |
| 18:16.20 | [sr] | perfect perfect!!!! |
| 18:18.19 | [sr] | [TK]D-Fender: ok so i see this cannot be some by freepbx |
| 18:18.22 | [sr] | no problem about that |
| 18:18.37 | [sr] | i have to this to each number, right? |
| 18:21.12 | [TK]D-Fender | ? |
| 18:21.31 | [TK]D-Fender | [sr]: This is a number that lands on your system.. It is one of YOUR numbers, right? not the callers? |
| 18:21.51 | [sr] | [TK]D-Fender: yap, my question was, i have to route each of my numbers |
| 18:23.57 | [sr] | [TK]D-Fender: ok so far so good, call you tell me or point me to what the arguments mean? exten=>number,1,dial(sip999) |
| 18:24.01 | *** join/#asterisk Netgeeks (~chris@gw1.netgeeks.net) |
| 18:24.04 | [sr] | what does "1" means? |
| 18:25.10 | [TK]D-Fender | [sr]: You don't seem to understand the basics of the dialplan. Setting up LCR to take calls is the SMALLEST portion of *. The other %95 is your DIALPLAN. Go read Chapter 5 <----- |
| 18:25.12 | [TK]D-Fender | ~book |
| 18:25.12 | 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 |
| 18:27.14 | *** join/#asterisk darkskiez (~dz@72-254-56-42.client.stsn.net) |
| 18:29.07 | *** join/#asterisk QaDeS (~mklaus@p4FC72579.dip0.t-ipconnect.de) |
| 18:32.34 | [sr] | [TK]D-Fender: i'm to take a look |
| 18:33.45 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 18:35.43 | Naikrovek | http://i.imgur.com/O90lz.jpg |
| 18:35.51 | Naikrovek | eventually you will see why that is funny |
| 18:36.26 | [sr] | the extra "eye"? |
| 18:36.30 | Naikrovek | no |
| 18:37.26 | [TK]D-Fender | :D |
| 18:37.47 | Naikrovek | when you see it you'll laugh |
| 18:38.01 | pabelanger | its looking right at me |
| 18:39.20 | Naikrovek | gotta be a photoshop job |
| 18:39.32 | [sr] | dont find it..heeh |
| 18:39.56 | [TK]D-Fender | ESL <- thats why |
| 18:41.45 | *** join/#asterisk dauergast (~sag@188-193-228-23-dynip.superkabel.de) |
| 18:42.23 | [sr] | esl = ? |
| 18:43.09 | [TK]D-Fender | [sr]: You're not reading between the lines.... |
| 18:45.27 | dauergast | hi, im using asterisk 1.6.2.8, compiled from source. Porblem is that asterisk sometimes takes ages to respond to calls, internal calls external calls, doesnt matter. After about 20 seconds == Using SIP RTP CoS mark 5 appears in the CLI and asterisk establishes the call, i cant find a solution, already reistalled ubuntu server, but i doesn't seem to change anything |
| 18:48.59 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 18:49.52 | pabelanger | dauergast: enable SIP debugs and trace your calls |
| 18:50.18 | dauergast | ok |
| 18:50.29 | pabelanger | ~collectdebug |
| 18:50.29 | infobot | collectdebug is probably a method of collecting logs allowing others help troubleshoot an issue. Refer to doc/HOWTO_collect_debug_information.txt or http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt |
| 18:50.59 | drmessano | At least ASTERISK logs something |
| 18:51.04 | drmessano | kicks ejabberd |
| 18:51.28 | drmessano | kicks ejabberd again |
| 18:51.41 | drmessano | hey ejabberd, look over there |
| 18:51.45 | drmessano | kicks ejabberd again |
| 18:52.16 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 19:00.02 | *** join/#asterisk d4rkstar (~bruno@93-41-29-219.ip79.fastwebnet.it) |
| 19:03.10 | dauergast | ok, here is the debug output, tried a few times to call internal applications, at the last try, calling 960, it hangs about 10 seconds http://pastebin.com/iV2VQgiN |
| 19:07.12 | *** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 19:09.27 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 19:10.02 | dauergast | it happens randomly, all calls come from the internal lan, same subnet |
| 19:11.13 | Deeewayne | anyone know if sending spam text messages to cell phones is legal? |
| 19:11.29 | Deeewayne | (I'm a receiver not a sender) |
| 19:12.03 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 19:13.27 | [sr] | at least in here, when people subscribe to sms service, it becames legal 'cause the user signed, the problem it to stop the service |
| 19:23.39 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
| 19:25.33 | pabelanger | dauergast: [Jun 12 20:53:09] WARNING[10929] res_musiconhold.c: Found no files in '/var/lib/asterisk/schleife/' |
| 19:26.07 | pabelanger | dauergast: you have audio file issues. |
| 19:26.37 | p3nguin | deeewayne: In the US? |
| 19:27.45 | Deeewayne | p3nguin, yes |
| 19:27.45 | dauergast | i know, currently im not yet finished copying all the sound files back, but I have had the problems with the delayed calls before, this shouldn't cause such problems? |
| 19:27.50 | p3nguin | deeewayne: http://www.fcc.gov/cgb/consumerfacts/canspam.html |
| 19:27.58 | Deeewayne | thanks |
| 19:28.40 | p3nguin | In short, yes it is illegal. |
| 19:28.51 | dauergast | pabelanger: do you need other logs / traces to determine the problem? |
| 19:28.52 | Deeewayne | I thought so |
| 19:30.47 | Deeewayne | goes off to BBQ |
| 19:31.04 | pabelanger | dauergast: you'll need to better describe what a 'delayed calls' means. IE: dead air, dtmf, what is the actual problem |
| 19:31.31 | *** join/#asterisk darkskiez (~dz@72-254-56-42.client.stsn.net) |
| 19:32.19 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 19:33.42 | dauergast | pabelanger: when i'm trying to make a call from my softphone, sometimes it keeps saying connecting... for a couple of seconds before asterisk recognizes that a client wants to make a call, this can take up to 30 seconds |
| 19:34.49 | pabelanger | dauergast: How do you not it is not the softphone that is the problem? |
| 19:34.53 | pabelanger | IE: resolving DNS |
| 19:36.22 | dauergast | using the snom190 hw phone doesn't make any difference, i think this is a generally configuration or network issue |
| 19:37.11 | dauergast | the asterisk is reached by its lan ip address, dns is not used |
| 19:38.14 | pabelanger | dauergast: if you are on the same LAN, why do you have NAT enabled? |
| 19:38.25 | pabelanger | dauergast: pb your sip.conf |
| 19:38.31 | dauergast | ok |
| 19:43.16 | pabelanger | dauergast: once the first INVITE message gets to Asterisk it takes less then .01 seconds for Asterisk to answer the channel. |
| 19:43.30 | pabelanger | IE: Asterisk is not the problem |
| 19:44.01 | dauergast | pabelanger: here is my the sip.conf http://pastebin.com/wgbAgS8v |
| 19:50.35 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-67-71-155.home.otenet.gr) |
| 19:51.09 | dauergast | ok, i will disable nat for the peers, and try again |
| 19:56.09 | *** join/#asterisk mnicholson (~mnicholso@207.111.163.221) |
| 20:07.05 | dauergast | after sending out nine invites to asterisk with no reply, the softphone ends the call 'No Connection' |
| 20:11.08 | dauergast | maybe 1000 port for RTP are too less? |
| 20:11.12 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-67-71-155.home.otenet.gr) |
| 20:14.16 | p3nguin | I use much less than that. |
| 20:15.24 | dauergast | hmm |
| 20:19.08 | dauergast | is it possible that the sip modules 'freezes' for whatever reason every x seconds? |
| 20:20.25 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 20:20.27 | *** join/#asterisk mnicholson (~mnicholso@nat/digium/x-ojxxhpbyblspnivm) |
| 20:28.43 | KavanS | is there a simple way to say "if callerid digits = 3" goto? |
| 20:28.50 | KavanS | gotoif I assume...googling... |
| 20:35.23 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 20:38.53 | dauergast | same issue with my asterisk in vm, after replacing only the sip.conf, i think there is something wrong |
| 20:43.47 | *** join/#asterisk Blackvel (~blackvel@dslb-088-065-077-238.pools.arcor-ip.net) |
| 20:44.23 | Blackvel | hi all. protected my asterisk local server with a fail2ban jail. is there any tool which tries to hack into to test it? |
| 20:45.08 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 20:45.45 | xheliox | a soft phone and the wrong password? :P |
| 20:46.12 | xheliox | is it on a public IP? I can try to register for you. |
| 20:51.30 | *** join/#asterisk MiserySoft (~lnd@95.145.212.39) |
| 21:02.21 | *** join/#asterisk p3nguin_ (gpz5GvdFkf@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 21:07.46 | KavanS | exten = s,n,GotoIf($["${CALLERID(num)}" = "[0-9][0-9][0-9]"]?begin) |
| 21:08.10 | KavanS | ^^^ what is wrong with that statement? - it shows that the caller ID is 501 - but it is not going to begin |
| 21:10.07 | KavanS | NoOp("SIP/501-000000xX", "501") |
| 21:11.43 | KavanS | ahh it's my regex :\ |
| 21:15.45 | Blackvel | anyone uses fail2ban? fail2ban-regex can not get a right match. i added a new regex for ACL error (permit/deny). what is wrong? date? |
| 21:15.45 | Blackvel | fail2ban-regex /var/log/asterisk/messages /etc/fail2ban/filter.d/asterisk.conf |
| 21:16.13 | xheliox | did you change the time stamp format in logger.conf? |
| 21:16.36 | Blackvel | yes, but i think there is still something wrong with it |
| 21:16.54 | Blackvel | even fail2ban-regex uses timestamp check? |
| 21:17.22 | Blackvel | 2010-06-12 23:01:13 NOTICE[7697] chan_sip.c: |
| 21:17.25 | Blackvel | is this any correct? |
| 21:18.17 | xheliox | My log says.. [2010-06-12 17:17:15] NOTICE[8656] |
| 21:22.00 | Blackvel | NOTICE.* .*: Registration from '.*' failed for '<HOST>' - ACL error (permit/deny) |
| 21:22.04 | Blackvel | something wrong with that? |
| 21:25.49 | [TK]D-Fender | [17:08]<KavanS>^^^ what is wrong with that statement? - it shows that the caller ID is 501 - but it is not going to begin <-- since when did you think you could just shove a patter/regex like that? "core show function REGEX" |
| 21:26.15 | KavanS | [TK]D-Fender, I know - I know |
| 21:26.18 | KavanS | I fixed it... |
| 21:43.55 | *** join/#asterisk x303 (~x303@97.100.255.188) |
| 21:49.21 | *** join/#asterisk ming_zym (~ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 21:49.42 | *** join/#asterisk Jumpie (n3rdz@ip68-230-28-186.ph.ph.cox.net) |
| 21:52.47 | *** join/#asterisk TheKing (~chatzilla@188.123.175.100) |
| 21:53.59 | TheKing | hi all. i m new in open source, and i want to know what is the difference between FreePBX and Asterisk? |
| 21:55.15 | TheKing | should i use Asterisk or FreePBX, i know that AsteriskNOW is using FreePBX but is that good to use for production environment? |
| 21:59.45 | *** join/#asterisk pabelanger (~pabelange@CPE001fe2a8fd1d-CM0012254094b2.cpe.net.cable.rogers.com) |
| 22:03.41 | ruben23 | hi guys |
| 22:04.00 | carrar | hi!! |
| 22:04.04 | carrar | How are you? |
| 22:04.49 | carrar | TheKing, are you still here? |
| 22:04.58 | TheKing | yes |
| 22:05.00 | carrar | "the difference between FreePBX and Asterisk?" |
| 22:05.04 | TheKing | i m fine thank u |
| 22:05.07 | carrar | One is a PBX01 software |
| 22:05.12 | carrar | One is a Operating System |
| 22:05.20 | carrar | err PBX01=PBX |
| 22:05.33 | carrar | two completely different items |
| 22:05.49 | carrar | Asterisk being the PBX software |
| 22:05.59 | TheKing | aha |
| 22:06.04 | carrar | Asterisk can run on FreeBSD |
| 22:06.13 | carrar | But I wouldn't recommend it |
| 22:06.30 | carrar | I would use CentOS 5.5 OS instead of FreeBSD |
| 22:06.40 | carrar | unless you are a FreeBSD guru |
| 22:06.45 | TheKing | i m using CentOS 5.5 right now |
| 22:06.50 | carrar | excellent |
| 22:06.55 | TheKing | no i m new in Linux |
| 22:07.04 | carrar | CentOS is good then |
| 22:07.18 | carrar | Asterisk works great in CentOS |
| 22:07.27 | TheKing | good |
| 22:07.53 | carrar | I wouldn't change it |
| 22:08.10 | carrar | unless you are not running Asterisk from the source and compiling it |
| 22:08.23 | carrar | then I would just compile from source instead of RPM's |
| 22:08.41 | carrar | but whatever works for you |
| 22:09.14 | TheKing | i used AstersikNOW but when i install it i could not find the IVR tab |
| 22:09.19 | nightwalk | carrar: What do you have against FreeBSD? Is driver support lacking for PBX-type cards or something? |
| 22:09.26 | carrar | I love FreeBSD |
| 22:09.31 | carrar | but for someone new |
| 22:09.37 | nightwalk | PC-BSD, then |
| 22:09.38 | carrar | and Asterisk |
| 22:09.44 | carrar | it's prbably not the best choice |
| 22:09.51 | nightwalk | PC-BSD seems just as easy as any linux distro |
| 22:10.14 | nightwalk | The only thing I don't like about FreeBSD are the non-obvious device naming scheme |
| 22:10.20 | nightwalk | schemes |
| 22:10.23 | *** join/#asterisk viq (~viq@unaffiliated/viq) |
| 22:10.26 | carrar | Speaking of FreeBSD |
| 22:10.26 | carrar | 8.0-RELEASE-p2 FreeBSD 8.0-RELEASE-p2 |
| 22:10.29 | carrar | I need to update that |
| 22:11.35 | TheKing | is it better than CentOS, did FreeBSD have a GUI, coz u know as a windows user i depend in that GUI in most of my operaitonsd? |
| 22:11.40 | TheKing | *operations |
| 22:12.00 | carrar | Are oranges better then apples? |
| 22:12.07 | carrar | CentOS has more support |
| 22:12.13 | nightwalk | On second thought, you're right, FreeBSD might not be the best for beginners. It's alot like slackware in that you're left to fend for yourself when it comes to resolving package dependencies and installing packages. No rpm/apt-get :/ |
| 22:12.17 | TheKing | it depends on ur taste :) |
| 22:12.42 | carrar | I would not recommend FreeBSD for Asterisk users |
| 22:13.03 | carrar | I would recommend FreeBSD for a DB server or web, like servers |
| 22:13.10 | TheKing | aha |
| 22:13.32 | TheKing | so u second my using of CentOS |
| 22:13.38 | carrar | yes |
| 22:13.47 | TheKing | they also recommend it in the website |
| 22:13.52 | carrar | however |
| 22:14.02 | carrar | I don't recommend AsteriskNOW |
| 22:14.13 | TheKing | that's it, i knew it |
| 22:14.19 | carrar | I recommend installing Asterisk from source files |
| 22:14.24 | nightwalk | Personally, I recommend linux as the *host* OS, because there are tons of virtualization technologies it supports. CentOS/RHEL isn't worth using until 6.0 in my opinion, though. VT support is lacking |
| 22:14.27 | TheKing | it's not the best choice |
| 22:14.36 | carrar | from here: http://www.asterisk.org/downloads |
| 22:14.47 | p3nguin_ | nightwalk: If you think FreeBSD doesn't do dependencies, you're using it wrong. |
| 22:15.25 | p3nguin_ | Time for you to learn how to use ports/packages the right way. |
| 22:15.31 | nightwalk | p3nguin_: I haven't done more than play with it yet. PC-BSD is the only thing that's gotten me that far -- the original FBSD installer is god-awful |
| 22:15.34 | carrar | Asterisk 1.6.2.8 + Asterisk Add-Ons 1.6.2.1 + Dahdi Complete & LibPri |
| 22:16.08 | p3nguin_ | PC-BSD is, from what I hear, a pile of crap compared to actual FreeBSD. |
| 22:16.30 | TheKing | does "Asterisk 1.6.2.8 + Asterisk Add-Ons 1.6.2.1 + Dahdi Complete & LibPri " it have GUI ? |
| 22:16.42 | carrar | GUI's are for punks |
| 22:17.05 | nightwalk | It doesn't matter if FBSD were the greatest thing on earth. If the installer is garbage, people aren't going to use it. The FBSD devs would rather bury their heads in the sand and ignore the reality of it, though |
| 22:17.13 | carrar | I don't use any GUI's |
| 22:17.22 | *** join/#asterisk CngZ (~cngz@melis.cngz.fr) |
| 22:17.28 | carrar | but I think there might be one in it but I don't know |
| 22:17.35 | CngZ | hello |
| 22:17.49 | nightwalk | That's why I support PC-BSD's efforts. I don't really *like* GUIs, but I *do* like things being fairly easy to install |
| 22:17.51 | carrar | The only GUI I use is on my iPhone |
| 22:18.01 | TheKing | i m trying to be a good one, but for the meanwhile i have to be a punk till i learn it very well ;) |
| 22:18.19 | carrar | nt gonna learn it using a GUI |
| 22:18.22 | carrar | not |
| 22:18.45 | nightwalk | TheKing: Don't feel bad. I learned the way I did because I'm old, and there WERE no good GUIs in the distros way back when. I'd have used GUIs at first too if they'd been there :) |
| 22:18.53 | carrar | All you are gonna know is a GUI and not how the configs really work |
| 22:19.36 | nightwalk | carrar: That statement operates on a fallacy |
| 22:19.40 | p3nguin_ | People aren't going to use FreeBSD? WHAT?! You realize how many people DO use FreeBSD? |
| 22:19.55 | nightwalk | p3nguin_: Not as many as use linux :) |
| 22:20.01 | *** join/#asterisk pyite (~dschreibe@unaffiliated/pyite) |
| 22:20.12 | p3nguin_ | Don't flatter your Linux-using self. |
| 22:20.29 | carrar | nightwalk, it's opinion, not fallacy |
| 22:20.41 | TheKing | carrar, i totally agree with u , can u tell me where i can start using Asterisk step by step? |
| 22:20.46 | nightwalk | Yeah, I use linux. I also use Solaris, and I plan to switch from Solaris to FreeBSD eventually (for ZFS support; linux can't have it) |
| 22:20.57 | carrar | TheKing, read this and use the source |
| 22:20.58 | carrar | ~book |
| 22:20.58 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 22:21.10 | carrar | You will be a guru in no time |
| 22:21.21 | nightwalk | I use whatever works for my purposes, so I have no reason to build up any one platform/technology |
| 22:21.37 | CngZ | i installed and configured and iaxmodem, and hylafax, i can send fax by a sip line with no problem, but i can't receive fax, the iaxmodem got the call but the sender is saying that it's failed |
| 22:22.32 | carrar | CngZ, look at your log files |
| 22:22.36 | carrar | debug files |
| 22:22.44 | carrar | find out where it's breaking at |
| 22:23.07 | carrar | find out what failed means |
| 22:23.14 | TheKing | ok, what about the reports u know that management always asks for reports, and some times they asked to have an access by themslefs |
| 22:23.14 | CngZ | hum do you know where is the log of hylafax ? i will search :p |
| 22:23.25 | p3nguin_ | I've been using FreeBSD almost as long as I have Linux, and I can't say that either is so great above the other that people are going to suddenly stop using the lesser of the two. |
| 22:23.43 | *** join/#asterisk darg_ (~djc@65.209.147.101) |
| 22:23.46 | carrar | TheKing, so create a unix login for them, not that they know unix |
| 22:24.04 | darg_ | knows eunichs |
| 22:24.04 | carrar | You can create reports out of the CDR's if you put them in a database like PostgreSQL |
| 22:24.10 | carrar | make your own queires |
| 22:24.22 | darg_ | ok not really |
| 22:24.30 | TheKing | aha, that's right |
| 22:24.33 | darg_ | p3nguin_, you alive |
| 22:24.44 | p3nguin_ | only in the flesh |
| 22:24.46 | carrar | They can then be crontab so they run automatically |
| 22:24.47 | *** join/#asterisk extnct (~extnct@unaffiliated/extnct) |
| 22:25.06 | darg_ | it turned out not to be the spa at all |
| 22:25.08 | TheKing | and save them in PDF format, u r right |
| 22:25.17 | darg_ | was a max ring setting in the DID provider |
| 22:25.23 | darg_ | set at 30 |
| 22:25.42 | TheKing | it needs a lot of works but it may work |
| 22:25.50 | carrar | TheKing, there is not limit what you can do, you can fly to the moon with UNIX |
| 22:25.58 | carrar | NASA did it |
| 22:26.01 | carrar | You can too! |
| 22:26.04 | nightwalk | Well, my point was that the linux culture seems more welcoming. FBSD people seem to have more of an narcissist-elitist mentality. Their Way is Perfect, and its blasphemous to even challenge Their Way. Whoa be it to anyone who'd ask them to CHANGE anything :) |
| 22:26.17 | carrar | granted you might need a ship |
| 22:26.21 | darg_ | itym "woe be to anyone" |
| 22:26.27 | TheKing | :D |
| 22:28.10 | TheKing | another question please, does these books teach u how to compile the source code, or i have to look some where else? |
| 22:28.13 | nightwalk | eh, whatever. My inner english instructor gets weekends off :P |
| 22:28.19 | p3nguin_ | darg_: That's pretty strange, considering I thought the ATA was sending the termination notice to Asterisk. |
| 22:28.59 | carrar | gives you the commands to compile it |
| 22:29.09 | carrar | walks you through it |
| 22:29.10 | carrar | go read it |
| 22:30.36 | CngZ | carrar: i have found "Failure to receive silence (synchronization failure)." in the xferfaxlogs file |
| 22:31.11 | carrar | Could be cause you are doing it over SIP |
| 22:31.12 | TheKing | carrar, thank u very much for ur help, i m a .NET developer and if u need any help there i ready, on another hand and as i m new in Linux products can u recommend book or a place i can go for to learn Linux ? |
| 22:31.18 | carrar | or could be a protocol issue |
| 22:31.20 | nightwalk | TheKing: Most well-established packages have an INSTALL file inside of their source tarball that gives you a run-down on what you have to do to compile. A lot of times it's just something like './configure && make && make install' unless you have special requirements |
| 22:31.22 | carrar | have to google it |
| 22:31.31 | CngZ | yes, i'm doing it :) thx |
| 22:32.07 | carrar | TheKing, hrmm anything that specific about the OS you are using, check here http://www.centos.org/ |
| 22:33.05 | TheKing | nightwalk, thanks very much |
| 22:33.14 | darg_ | p3nguin_, well, in the sip debug I was only debugging the SPA SIP, not the SIP traffic from the DID provider |
| 22:33.15 | carrar | http://www.linux-books.us/centos.php |
| 22:33.27 | darg_ | probably the SPA message was just acknowledging a message coming from the origin |
| 22:34.10 | p3nguin_ | I see. |
| 22:34.51 | darg_ | anyway, thats one thing on my list of things to worry about that I can tick off |
| 22:34.59 | TheKing | that's great, thank u all. |
| 22:37.35 | TheKing | last one please, when i try to open OpenOffice writer it shows on the window bar and then disappear, any idea? |
| 22:37.59 | nightwalk | TheKing: Run it from a terminal window. It's probably segfaulting |
| 22:38.29 | *** part/#asterisk ManxPower (~manxpower@234.sub-75-235-252.myvzw.com) |
| 22:38.31 | TheKing | by write, run "OpenOffice"? |
| 22:39.02 | p3nguin_ | The program's name is OpenOffice.org, and it is usually ran by the "soffice" command. |
| 22:40.35 | nightwalk | Yes, 'soffice' should work. If that doesn't work, try running it as 'openoffice.org' |
| 22:42.05 | *** join/#asterisk carrar (tim@osburn.com) |
| 22:42.05 | TheKing | thank u |
| 22:43.30 | CngZ | can you advise me a gui for sending fax via hylafax another than avantfax ? :) |
| 22:46.50 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 22:48.27 | *** join/#asterisk kruemeltee (~Maddin@port-92-198-62-82.static.qsc.de) |
| 22:48.33 | CngZ | think found |
| 22:48.49 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 22:57.58 | *** join/#asterisk f00bar80 (~f00bar80@41.234.168.27) |
| 22:58.09 | f00bar80 | I received a one ring call from this # 973-360-2026, when i checked , it belongs to New Jersey Kushner Co. I called back but the number is disconnected. thousands of reports/online forums when i googled they have the same happened to them , any comment ? here's the link, http://www.google.com.eg/search?hl=en&safe=off&q=9733602026+phone+number&cts=1276382036310&aq=f&aqi=&aql=&oq=&gs_rfai= |
| 22:59.41 | [psy] | maybe a wardailer? |
| 23:03.19 | *** join/#asterisk jksM (~jks@193.189.93.254) |
| 23:11.46 | *** join/#asterisk sky1975 (76f3d0db@gateway/web/freenode/ip.118.243.208.219) |
| 23:11.56 | sky1975 | Hi |
| 23:12.53 | sky1975 | Can anyboy help me to get h323 work |
| 23:16.36 | *** join/#asterisk QaDeS (~mklaus@p4FC72579.dip0.t-ipconnect.de) |
| 23:19.30 | *** join/#asterisk viq (~viq@unaffiliated/viq) |
| 23:20.02 | sky1975 | Anybody use h323? |
| 23:28.17 | nightwalk | CngZ: There's a list of third party GUIs on hylafax' website |
| 23:28.58 | f00bar80 | [psy], is it a wardailer or caller id spoofing ? |
| 23:29.14 | CngZ | yes, found already found the page, thx |
| 23:31.40 | sky1975 | What is the meaning of gatekeeper? |
| 23:33.49 | *** join/#asterisk pfn (pfnguyen@socal.hanhuy.com) |
| 23:34.02 | pfn | damnit, I need to figure out how to configure my 7960 again... |
| 23:35.09 | p3nguin_ | What do you need to know? |
| 23:35.49 | sky1975 | <p3nguin_> for me? |
| 23:37.10 | p3nguin_ | Please do not quote me. |
| 23:37.13 | pfn | I really just need a /tftpboot directory |
| 23:37.19 | p3nguin_ | Create one. |
| 23:37.26 | pfn | the contents for configuring 7960's |
| 23:37.33 | p3nguin_ | SCCP or SIP? |
| 23:37.36 | pfn | sip |
| 23:37.49 | pfn | my old asterisk server's hard drive bit the dust, and I didn't save the config anywhere |
| 23:38.15 | p3nguin_ | Do you have the firmware files? |
| 23:38.26 | pfn | not anymore |
| 23:38.27 | *** join/#asterisk eliel (~eliels@186.18.131.44) |
| 23:38.54 | p3nguin_ | Start by getting that much. |
| 23:39.09 | p3nguin_ | I'll get you the config examples. |
| 23:41.33 | pfn | yay for firmware links on voip-info |
| 23:41.39 | pfn | digging through cisco's technet would be a pita |
| 23:41.57 | p3nguin_ | Actually, getting the firmware right from Cisco.com is pretty simple. |
| 23:42.13 | sky1975 | Can somebody help me to get h323 working? |
| 23:42.31 | p3nguin_ | Unless you don't have the proper contract, then it's slightly more difficult to download the files. |
| 23:42.46 | pfn | yeah, I don't have a contract with cisco |
| 23:43.56 | p3nguin_ | http://www.loligo.com/asterisk/Cisco/79xx/current/SIPDefault.cnf |
| 23:44.09 | p3nguin_ | http://www.loligo.com/asterisk/Cisco/79xx/current/SIP0002B9EB0EF4.cnf |
| 23:44.43 | pabelanger | sky1975: Do you need to use H323, or can you SIP? |
| 23:45.01 | pabelanger | get h323 support is very rare |
| 23:45.06 | sky1975 | I need to user H323 |
| 23:45.16 | pabelanger | s/get/getting |
| 23:45.18 | sky1975 | Is that |
| 23:45.23 | pfn | p3nguin_, thanks |
| 23:45.37 | sky1975 | I found ooh323 already installed |
| 23:45.53 | p3nguin_ | Don't forget to put the right version number in OS79XX.TXT |
| 23:45.56 | exothermc | what are the cli commands to show more information about dahdi channels? |
| 23:47.00 | pfn | need to rebuild my entire asterisk dialplans, pita |
| 23:47.18 | p3nguin_ | What happened to your backups? |
| 23:47.46 | *** join/#asterisk x303 (~x303@97.100.255.188) |
| 23:47.54 | pfn | backups? what are those? ;-) |
| 23:48.01 | KavanS | backups? wtH?! |
| 23:48.03 | pabelanger | exothermc: dahdi show channels |
| 23:48.31 | p3nguin_ | Ah, yeah, those tend to take up disk space and time. I forgot that was why people don't use them. |
| 23:48.46 | exothermc | pabelanger: No such command 'dahdi show channels' (type 'core show help dahdi show' for other possible commands) |
| 23:49.30 | exothermc | pabelanger: module show like dahdi shows that chan_dahdi.so and 5 other modules with that name are loaded. |
| 23:49.35 | pabelanger | exothermc: then dahdi is not loaded |
| 23:49.44 | pabelanger | exothermc: module load chan_dahdi.so |
| 23:49.58 | *** join/#asterisk frek818 (~herman@rrcs-74-62-208-50.west.biz.rr.com) |
| 23:50.29 | exothermc | pabelanger: ahh I see it now, thanks |
| 23:51.10 | pabelanger | exothermc: edit your modules.conf if you want to load it each time |
| 23:51.16 | pabelanger | asterisk starts |
| 23:58.09 | exothermc | pabelanger: ok so now I have it coming up, but it puts the status of all my channels as "in use" when looking at /proc/dahdi/* |
