| 00:01.09 | bluregard | EmleyMoor: the FWD user in sip.conf has context=from-sip right? |
| 00:02.31 | EmleyMoor | Yes |
| 00:04.07 | bluregard | what's your FWD # again? |
| 00:04.16 | EmleyMoor | 794933 |
| 00:05.14 | bluregard | I'm still getting 603 declined |
| 00:05.16 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 00:06.40 | EmleyMoor | asterisk says it's connected |
| 00:10.15 | mosty | is it possible to get Queue() to not ring an extension that is in use? for example, if an agent makes an outgoing call, the queue seems to still want to call that agent if a call comes in and they are next in line to answer |
| 00:10.37 | *** join/#asterisk MGSsancho (n=user@70.135.185.201) |
| 00:10.43 | bluregard | can you dial out? |
| 00:11.14 | EmleyMoor | Yes |
| 00:16.49 | EmleyMoor | Time I was preparing for bed |
| 00:16.54 | bluregard | can you dial other fwd? |
| 00:19.29 | EmleyMoor | Yes |
| 00:19.40 | bluregard | <PROTECTED> |
| 00:20.14 | EmleyMoor | "I am sorry, that is not a valid extension. Please try again." |
| 00:21.10 | bluregard | uuhhh, that's weird |
| 00:21.14 | EmleyMoor | Hmmm |
| 00:21.15 | bluregard | that's my number |
| 00:21.49 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 00:22.05 | *** join/#asterisk Mad_guy (n=austin@ip68-1-213-212.dl.dl.cox.net) |
| 00:22.10 | CrashHD | anyone have any info on faxing with asterisk? |
| 00:22.16 | CrashHD | good site with info or something? |
| 00:22.28 | EmleyMoor | I'd like that info too... |
| 00:22.31 | pdunkel | CrashHD: Yes, don't do it ;) |
| 00:22.37 | CrashHD | :) |
| 00:22.39 | bluregard | <PROTECTED> |
| 00:22.43 | EmleyMoor | Yes |
| 00:23.03 | EmleyMoor | (I have nothing else apart from internal at present) |
| 00:23.06 | CrashHD | well we are looking at eliminating the needs for pots lines |
| 00:23.12 | CrashHD | even if it's voip fax machines or something |
| 00:23.15 | CrashHD | bypassing asterisk |
| 00:23.18 | CrashHD | any ideas? |
| 00:23.19 | pdunkel | Fax@asterisk ist nice except don't try to do it with VoIP |
| 00:23.30 | EmleyMoor | I reached a friend who was on it earlier with no problems |
| 00:23.34 | CrashHD | what about any other voip fax solutions? |
| 00:23.36 | EmleyMoor | (5-digit number though) |
| 00:23.56 | bluregard | <PROTECTED> |
| 00:24.06 | EmleyMoor | I do - never fails! |
| 00:24.12 | pdunkel | The problem ist that at this stage * does not have full T.38 support yet |
| 00:24.41 | CrashHD | ok, now asterisk out of the picture |
| 00:24.47 | CrashHD | is there another solution? |
| 00:24.51 | pdunkel | What you could do is find yourself a T.38 provider though and connect directly to that via an T.38 capaple ATA (Sipura) |
| 00:24.52 | CrashHD | like a carrier |
| 00:24.57 | CrashHD | to work direct with a voip fax machine |
| 00:24.58 | CrashHD | or something |
| 00:25.06 | CrashHD | to use until asterisk comes around |
| 00:25.08 | pdunkel | Exactly |
| 00:25.17 | CrashHD | ok |
| 00:25.24 | CrashHD | any recommendations on T.38 providers |
| 00:25.26 | CrashHD | :) |
| 00:25.38 | InfraRed | T.38 is pita too |
| 00:25.41 | pdunkel | Actually you buy an adapter, connect your regular faxmachine and Ethernet. And go |
| 00:25.54 | InfraRed | on my tests it fai |
| 00:25.59 | CrashHD | sweet, so provider |
| 00:26.11 | CrashHD | any thoughts on provider to offer T.38? |
| 00:26.13 | InfraRed | it worked 70% on g711u |
| 00:26.18 | pdunkel | @InfraRed: With what ata did you test? |
| 00:26.19 | bluregard | <PROTECTED> |
| 00:26.27 | InfraRed | mediatrix |
| 00:26.30 | bluregard | see if my calll actually reaches your box |
| 00:26.47 | EmleyMoor | How do I turn that on? |
| 00:26.56 | *** part/#asterisk kuku5 (n=kuku5@c-71-201-219-72.hsd1.il.comcast.net) |
| 00:27.05 | InfraRed | g729 failed 90% of the time |
| 00:27.10 | InfraRed | this was over a lan |
| 00:27.17 | InfraRed | not even the internet |
| 00:27.43 | bluregard | try mine again 808235 |
| 00:27.52 | bluregard | I think that was my fault |
| 00:27.57 | pdunkel | @InfraRed: Which equppment? |
| 00:28.54 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 00:29.31 | EmleyMoor | bluregard: Still no go |
| 00:29.55 | *** join/#asterisk iCEBrkr (i=icebrkr@69.9.167.70) |
| 00:29.55 | InfraRed | pdunkel: Meditrix 1104 |
| 00:30.22 | pdunkel | Ok, now I know to sty away from that. I had 99.99999& Success with Sipura |
| 00:31.00 | EmleyMoor | Try mine again - sip debug on |
| 00:31.21 | bluregard | do you see my call trying to come in? |
| 00:31.35 | EmleyMoor | Nothing seen yet |
| 00:32.42 | EmleyMoor | Hmmm... |
| 00:33.19 | bluregard | you're not using iax2 at all? |
| 00:33.26 | EmleyMoor | Something happened then |
| 00:33.32 | EmleyMoor | Only internally for the phones |
| 00:33.50 | *** join/#asterisk Katty (n=Administ@24-182-147-74.dhcp.cpgd.mo.charter.com) |
| 00:33.57 | EmleyMoor | Tried it with fwd but it was hopeless - couldn't make calls at all |
| 00:34.09 | Katty | hihi. |
| 00:34.22 | pdunkel | hihi |
| 00:35.01 | bluregard | hmm |
| 00:35.02 | EmleyMoor | Think it's only a register attempt I see |
| 00:35.18 | bluregard | I can't even get a call from the FWD Call me button |
| 00:35.28 | EmleyMoor | I had the same problem |
| 00:35.53 | *** join/#asterisk angler_ (i=angler@nat/digium/x-785f9b26329583a5) |
| 00:36.47 | EmleyMoor | Switching to SIP solved it |
| 00:36.57 | *** join/#asterisk [shodan] (n=shodan@ip215.96-113-216.pppoe1.joliette.intermonde.net) |
| 00:37.27 | bluregard | I had it working fine the other day though |
| 00:37.42 | bluregard | and I can call their echo and time extension just fine |
| 00:38.45 | EmleyMoor | Yes - I had likewise |
| 00:38.53 | [shodan] | I'm having a weird problem, sometime I receive a call and I want to transfert it, I dial #101 for example, then hangup , 101's phone rings, he picks up , the caller and 101 can talk , but after maybe 3-4 seconds they get cut off , sometime my phone rings again and I can talk with the caller again , but sometime the caller just gets hungup, why is that happening ? |
| 00:38.54 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 00:40.33 | *** join/#asterisk fromm (n=fromm@fromm.omnis.com) |
| 00:41.08 | andresmujica | e |
| 00:41.09 | andresmujica | exit |
| 00:41.11 | andresmujica | exit |
| 00:41.59 | *** join/#asterisk MGSsancho (n=user@70.135.185.201) |
| 00:42.02 | pdunkel | EmleyMoor: So helpful :) |
| 00:47.36 | [shodan] | hmm , didn't realize call transfering was such a big can of worms :\ |
| 00:47.50 | pdunkel | @shodan |
| 00:48.16 | mosty | shodan: i believe it's a bug |
| 00:48.25 | pdunkel | It isn't, however I would not use the * features for it. Try doing it via the stuff in your phone |
| 00:49.41 | [shodan] | pdunkel yeah but I have different phone models and brands , I don't want to have to deal with each kinks of each phones :\ |
| 00:50.08 | [shodan] | looks like right now call transfert works only because of my spa3k2 happens to works with "#" |
| 00:50.14 | pdunkel | shodan: Believe me these are just little kinks in comparison |
| 00:50.21 | [shodan] | I thought it was * doing this |
| 00:50.24 | [shodan] | ah ok :) |
| 00:51.21 | pdunkel | EmleyMoor: I didn't. Goole -> Modle Ages Lower class |
| 00:51.29 | [shodan] | EmleyMoor, it's a member of the Yeomanry, duh ! |
| 00:51.31 | pdunkel | But I gues you didn't mean that |
| 00:52.27 | EmleyMoor | In telephone terms, it's an old telephone - any one of a number of models available from the British GPO, 1967-1981, and BT, for a few years from 1981 |
| 00:52.48 | [shodan] | EmleyMoor , there's 9 modern definition for yeoman , and none of these mean your widgetthingy ;) |
| 00:52.50 | *** join/#asterisk Lord_Pall (i=Flerb@cpe-66-69-213-131.austin.res.rr.com) |
| 00:53.00 | *** part/#asterisk Lord_Pall (i=Flerb@cpe-66-69-213-131.austin.res.rr.com) |
| 00:53.00 | EmleyMoor | Large body, bell, a few models had tone dialling but most were pulse (or even rotary), wall versions existed |
| 00:53.22 | [shodan] | closest is probably the origin of the nickname of your phone "A signal communications officer in the British Royal Navy." |
| 00:53.39 | pdunkel | Now all you need is some "petrol" or "gas" and some matches you get a fine bonfire |
| 00:53.55 | pdunkel | :) |
| 00:53.58 | EmleyMoor | At one time, there were only two models of telephone in the UK - Yeoman and Trimphone |
| 00:54.12 | *** join/#asterisk zeppelin_ (n=zeppelin@201-34-96-24.paemt700.dsl.brasiltelecom.net.br) |
| 00:54.15 | EmleyMoor | (except in Hull where they had their own) |
| 00:54.21 | *** join/#asterisk bjohnson (n=bjohnson@209.195.80.69) |
| 00:54.59 | pdunkel | Where the Hell is Hull? |
| 00:55.19 | [shodan] | eh , this is fun , rotary phone to cell phone conversion http://www.sparkfun.com/tutorial/Port-O-Rotary/portable-rotary.htm |
| 00:55.20 | EmleyMoor | (Kingston-upon-)Hull, a city in East Yorkshire |
| 00:55.34 | CunningPike | Home of The Housemartins |
| 00:56.01 | pdunkel | I am waxing les and less certain I really wanted to know all that. |
| 00:58.15 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 01:00.15 | mosty | what does the retry setting in queues.conf do? the wiki is vague |
| 01:07.59 | bluregard | <PROTECTED> |
| 01:08.25 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 01:09.17 | EmleyMoor | 120 ms or so |
| 01:10.13 | *** join/#asterisk CyberKnet (n=CyberKne@ip68-13-246-61.ok.ok.cox.net) |
| 01:15.16 | bluregard | I can call myself through FWD |
| 01:19.38 | carrar | mosty, it has to do with the previous setting abov eit |
| 01:22.36 | bluregard | <PROTECTED> |
| 01:27.25 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 01:27.38 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 01:27.47 | [hC] | Is there any way to easily tell how long asterisk has been running for? |
| 01:27.52 | *** join/#asterisk lule (i=lule@host32.200-82-89.telecom.net.ar) |
| 01:28.12 | *** part/#asterisk lule (i=lule@host32.200-82-89.telecom.net.ar) |
| 01:29.06 | JT | show uptime |
| 01:31.16 | [hC] | ... duh.. :P |
| 01:31.17 | [hC] | thanks |
| 01:31.23 | *** part/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 01:34.06 | mosty | carrar, i guessed that much, but i still don't have much idea of what it does exactly |
| 01:36.35 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 01:41.26 | *** join/#asterisk hieunm_vips (n=hieunm_v@210.245.57.162) |
| 01:41.42 | hieunm_vips | hi |
| 01:42.07 | hieunm_vips | I want to know more about Asterisk Architecture, design. |
| 01:42.40 | hieunm_vips | where could I found this information? |
| 01:43.35 | JT | http://www.asterisk.org/architecture |
| 01:43.51 | hieunm_vips | I have read it! |
| 01:44.14 | hieunm_vips | But I want more information so that I can customize asterisk |
| 01:44.32 | hieunm_vips | and control asterisk :d |
| 01:44.56 | JT | do you need to understand the internals to use it? |
| 01:45.21 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:45.50 | hieunm_vips | If I don't understand asterisk I can not create the best product |
| 01:46.55 | *** join/#asterisk hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net) |
| 01:47.52 | JT | then try reading the source code? |
| 01:49.24 | hieunm_vips | It takes so much time! |
| 01:49.43 | pdunkel | Without time you cannto create the best product! |
| 01:50.28 | pdunkel | s/cannto/cannot/ |
| 01:51.10 | hieunm_vips | OK, time is the most valuable thing |
| 01:51.31 | pdunkel | I guess your product will be really expensive then! ;) |
| 01:52.00 | hieunm_vips | pdunkel, I think so |
| 01:52.24 | pdunkel | Well then again, you could just hire someone who has the expertiese you need! |
| 01:52.26 | hieunm_vips | I am trying to find your help |
| 01:54.10 | pdunkel | Thou hast sought and thou hast found, yet to utilize thou shalt askspecific questions and not "I want to know more about Asterisk Architecture" :) |
| 01:54.33 | *** join/#asterisk BZBW (n=wlwzhang@ip67-153-142-110.z142-153-67.customer.algx.net) |
| 01:54.36 | pdunkel | What are you trying to do? |
| 01:55.00 | hieunm_vips | I'm trying to know more about asterisk ! |
| 01:55.10 | [hC] | anyone have analog pstn lines from telus? |
| 01:55.12 | hieunm_vips | How is it created |
| 01:55.14 | pdunkel | Then nread the source. :( |
| 01:55.19 | [hC] | (coming in to zaptel) |
| 01:55.38 | pdunkel | (going out to sleep) |
| 01:55.43 | hieunm_vips | pdunkel, Are there another way? |
| 01:56.26 | hieunm_vips | "hire someone who has experties" is an impossible thing in my country |
| 01:57.58 | hmmhesays | lol |
| 01:58.03 | hmmhesays | what country are you in? |
| 01:58.12 | hieunm_vips | I'm from Vietnam |
| 01:58.54 | hieunm_vips | There are not much persons that has knowledge about Asterisk |
| 01:59.02 | hmmhesays | I see |
| 01:59.08 | hmmhesays | weell that is the beauty of the internet |
| 01:59.32 | hieunm_vips | Oh, yes. This is the reason that why I'm here |
| 02:00.05 | JT | hieunm_vips: have you tried reading the book? |
| 02:00.47 | hieunm_vips | I have read "Asterisk The Future Of Telephony", "Building Telephony System with Asterisk" |
| 02:00.54 | JT | hmm |
| 02:01.34 | hieunm_vips | It's only "How to use" book, isn't it ? |
| 02:01.58 | JT | i guess |
| 02:02.07 | JT | the low level stuff is not of interest to most |
| 02:02.26 | hieunm_vips | Could you recommend me another ? |
| 02:02.35 | JT | not a book |
| 02:02.59 | hieunm_vips | It's OK. Anything can help me know more about asterisk! |
| 02:03.01 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 02:03.10 | JT | voip-info.org |
| 02:03.17 | EyeCue | jaytee |
| 02:04.33 | hieunm_vips | voip-info is also a "how to use" site. |
| 02:04.59 | JT | it has all sorts of info there |
| 02:05.07 | hieunm_vips | I really want a "how does asterisk make every thing" docs |
| 02:05.15 | JT | as people have already said, if it that important you know lower level stuff, read the source |
| 02:05.18 | jart | hieunm_vips: are you a developer? |
| 02:05.34 | hieunm_vips | Yes, I'm a developer |
| 02:05.36 | jart | oh wait, we spoke earlier |
| 02:07.33 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:08.13 | *** join/#asterisk CyberKnet2 (n=CyberKne@ip68-13-246-61.ok.ok.cox.net) |
| 02:09.20 | hieunm_vips | JT, pdunkel, thank for your kind. I back to my way (read source) :( |
| 02:12.37 | *** part/#asterisk hieunm_vips (n=hieunm_v@210.245.57.162) |
| 02:21.23 | [hC] | hmm |
| 02:21.34 | [hC] | I guess this might be the cause of weird echo and dropped calls... |
| 02:21.46 | [hC] | zttest drops down to 99.975 sometimes and as low as 99.96 when i pingflood the lan |
| 02:24.41 | JT | [hC]: damn |
| 02:25.04 | [hC] | im gonna have to get in there and replace that box |
| 02:37.18 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 02:37.18 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:41.43 | hmmhesays | this house is freezing |
| 02:45.14 | Supaplex | is your house running windows? ;) |
| 02:45.16 | *** join/#asterisk CyberKnet2 (n=CyberKne@ip68-13-246-61.ok.ok.cox.net) |
| 02:45.30 | hmmhesays | no |
| 02:52.09 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 03:01.03 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 03:01.03 | *** mode/#asterisk [+o mog] by ChanServ |
| 03:06.42 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-166-226.buff.east.verizon.net) |
| 03:06.50 | SuPrSluG | hello |
| 03:13.09 | *** join/#asterisk tengulre (n=tengulre@221.11.5.182) |
| 03:16.54 | SuPrSluG | i am trying 1.4 beta 3 and when i call my number i don't get sound. no ringing and at the cli it says playing my greeting menu but no sound. any ideas? |
| 03:20.31 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
| 03:21.00 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 03:23.20 | mosty | codec issue? |
| 03:26.44 | *** join/#asterisk terrapen (n=cjs@208.64.89.90.utahbroadband.com) |
| 03:26.46 | terrapen | howdy |
| 03:26.53 | terrapen | does anybody have a friend at Sangoma? |
| 03:27.13 | *** join/#asterisk krausen (n=krausen@cpe-70-123-186-177.stx.res.rr.com) |
| 03:27.14 | krausen | hello |
| 03:28.28 | krausen | is there a good source of information re-writing my various .conf files (extension.conf, sip.conf, etc) translating from the 1.2 series to the 1.4 series? |
| 03:32.58 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 03:33.01 | shmaltz | hi everyone |
| 03:33.10 | russellb | krausen: just read the UPGRADE.txt in 1.4 |
| 03:33.28 | russellb | there shouldn't be much you have to change |
| 03:33.29 | krausen | ok |
| 03:33.45 | russellb | but that talks about various issues you should be aware of when upgrading |
| 03:33.51 | krausen | stupid question, but did the jingle stuff make it in the beta? |
| 03:33.55 | russellb | yes |
| 03:33.59 | krausen | ah cool |
| 03:34.06 | krausen | because I noticed it wasn't in beta 1 |
| 03:34.13 | JT | i love it, what a good old pci swaparoo game can give you |
| 03:34.13 | russellb | there is a CHANGES file you can look at to see what new stuff has been added |
| 03:34.14 | JT | --- Results after 327 passes --- |
| 03:34.15 | JT | Best: 100.000000 -- Worst: 99.987793 -- Average: 99.988448 |
| 03:34.23 | JT | up from 99.95% |
| 03:34.24 | russellb | krausen: it's called chan_gtalk |
| 03:34.42 | [hC] | JT: how was your results at 99.95%? |
| 03:34.54 | JT | [hC]: different pci slot |
| 03:35.01 | JT | or you mean how did it work |
| 03:35.05 | ThaZZa | Hey all. |
| 03:35.09 | russellb | sine apparently, operating with google talk is quite the same as the jingle spec |
| 03:35.27 | ThaZZa | Can anyone help me with a problem i am having with a SIP IP phone? |
| 03:35.59 | JT | [hC]: do you mean how did i find it working at 99.95? |
| 03:37.58 | shmaltz | ThaZZa, shoot |
| 03:38.39 | ThaZZa | shmaltz: I have a Cisco 7960 with SIP firmware and it is having major issues talking to a remote asterisk box. |
| 03:39.03 | [hC] | JT: yeah. |
| 03:39.08 | [hC] | JT: any problems? |
| 03:39.09 | shmaltz | ThaZZa, like what issues? do you have the same issues if Asterisk is NOT remote? |
| 03:39.16 | JT | [hC]: absolute crap, to put it nicely |
| 03:39.25 | JT | horrible crackling and poping noises on the line |
| 03:39.31 | JT | d channel couldn't stay synced |
| 03:39.35 | JT | calls dropped out |
| 03:39.38 | [hC] | JT: any echo or dropped calls? oh.. its a pri |
| 03:39.40 | JT | tons of PRI errors on the console |
| 03:39.44 | ThaZZa | shmaltz: Pretty much i can't see any requests for it to register on the asterisk box. |
| 03:40.00 | [hC] | JT: i have a sangoma a200d fxo card that drops calls and has echo "randomly" |
| 03:40.01 | JT | yeah crackling and poping on a digital line |
| 03:40.09 | shmaltz | ThaZZa, whats the network topology? |
| 03:40.10 | [hC] | and it does 99.975 99.96 alot |
| 03:40.26 | JT | [hC]: yeah that's not really a good enough zttest result |
| 03:40.40 | JT | needs to be >99.8 most of the time |
| 03:40.50 | JT | might handle one or two drops to 99.97 |
| 03:41.06 | ThaZZa | shmaltz: Pretty much 1 DSL at the phone end, with a debian box doing nat and 2 sip phones.. the 7960 and a GXP2000.. the GXP2000 is having no issues. |
| 03:41.18 | [hC] | JT: yeah this pretty solidly drops to 99.975 |
| 03:41.26 | JT | hrm |
| 03:41.29 | [hC] | JT: i havent for AGES been able to find the cause of these peoples dropped calls and echo |
| 03:41.33 | shmaltz | ThaZZa, you have to tell the Cisco phone that it is natted |
| 03:41.33 | JT | is the box doing only asterisk? |
| 03:41.39 | [hC] | Yes. |
| 03:41.50 | JT | try swapping pci slots |
| 03:41.56 | ThaZZa | shmaltz: I have done this in the SIP<mac>.cnf file |
| 03:42.16 | JT | or even better, remove all pci cards that aren't absolutely necessary and try differennt slots |
| 03:42.17 | shmaltz | what does it look like? your sipmac.cnf file? |
| 03:42.22 | JT | then try repopulating the other cards |
| 03:42.53 | shmaltz | ThaZZa, it should be: |
| 03:42.55 | shmaltz | nat_enable:1 |
| 03:42.56 | [hC] | Im going to replace the box entirely |
| 03:42.58 | [hC] | Just incase. |
| 03:43.05 | shmaltz | nat_address: ipaddress |
| 03:43.08 | [hC] | Its been like that for months now with no reason |
| 03:43.10 | tengulre | hi,all |
| 03:43.11 | [hC] | i cant take any more chances |
| 03:43.22 | [hC] | the hardware echo canceller only works about 97% of the time |
| 03:44.27 | JT | fair enough |
| 03:44.44 | JT | it might just turn out to be it doesn't like the pci card arrangement |
| 03:44.51 | [hC] | im glad that i found that though, ive been scratching my head like you have no idea. |
| 03:45.03 | JT | nothing i did in software made as noticable a difference as moving cards around |
| 03:45.10 | [hC] | not to mention that someone introduced the possibility of sip to sip echo to me, and i was like.. what the...? how is that even possible |
| 03:45.23 | ThaZZa | schurzi: http://pastebin.ca/244754 |
| 03:45.24 | JT | yeah it's good, giagnosis leads you on the way to a cure :) |
| 03:45.42 | ThaZZa | shmaltz: http://pastebin.ca/244754 |
| 03:46.15 | ThaZZa | shmaltz: Also have tried to set the nat_address to the wan static ip of the debian nat box. |
| 03:46.30 | [hC] | i dont know how lost interrupts could cause sip to sip echo even, unless there was some sort of zaptel issue with that happening |
| 03:46.34 | *** join/#asterisk blebleble (n=ble@d149-67-99-160.col.wideopenwest.com) |
| 03:47.14 | JT | [hC]: yeah i'm not really sure how that'd be possible |
| 03:47.19 | blebleble | is there any good papers on a distributed asterisk system? looking for a small distributed system like 3 boxes all sitting nex to each other to distribute load and prevent against failure |
| 03:47.57 | JT | blebleble: what are the client devices, what are the outside trunks? |
| 03:48.23 | blebleble | clients are scattered, sip/ata, all sip trunks on the outside |
| 03:48.37 | [hC] | JT: do you have apic on or off, btw? I have it on... i debated turning it off |
| 03:48.43 | JT | so it's pretty mich all voip |
| 03:48.57 | JT | [hC]: on, it made no difference on or off for me |
| 03:48.58 | blebleble | oh sorry , yah 100% |
| 03:49.05 | [hC] | JT: I see. |
| 03:49.08 | blebleble | its a hosted pbx solution |
| 03:49.21 | JT | [hC]: i tried switching smp off, but my system refused to boot then |
| 03:50.24 | [shodan] | how can I set the verbose level for the log file /var/log/asterisk/messages |
| 03:50.46 | blebleble | logger.conf |
| 03:53.08 | *** join/#asterisk bmg505 (n=leon@c1-73-3.rndf.isadsl.co.za) |
| 03:58.32 | ThaZZa | shmaltz: Any thoughts? |
| 04:03.00 | JT | [hC]: still there? |
| 04:06.05 | [hC] | yep, sorry. |
| 04:06.10 | [hC] | was away for a minute. |
| 04:07.15 | [hC] | JT: my box isnt smp, its just a dell |
| 04:07.49 | *** join/#asterisk sbingner (n=thanotos@pdpc/supporter/sustaining/sbingner) |
| 04:08.19 | JT | not sure what the brand has to do with it, i have a few SMP dells here :P |
| 04:08.26 | JT | been doing some more expeimenting |
| 04:08.31 | JT | experimenting |
| 04:08.58 | JT | and i'm thinking zap accuracy has less to do with the slot the cards is in, but more the irq it is assigned |
| 04:09.17 | JT | although slot could still be a factory, i have yet to test that out with my new line of thinking |
| 04:18.39 | shmaltz | why am I getting an error when running sox that: sox: error while loading shared libraries: libvorbisfile.so.3: cannot open shared object file: No such file or directory |
| 04:18.47 | shmaltz | and I do have libvorbis installed |
| 04:20.56 | JT | wrong symlinks or something? |
| 04:24.51 | JT | [hC]: i got it working at 99.98 again after re-enabling HT, too |
| 04:25.39 | [hC] | well, im pretty sure slots have preferred irq's unless you hard specify them in the bios |
| 04:25.44 | [hC] | (if your bios lets you) |
| 04:25.49 | [hC] | but most motherboard have a default assignment per slot |
| 04:27.56 | JT | yeah which is why i'm seeing what happens when i hard specify them |
| 04:28.13 | JT | i've noticed some weird behaviour though |
| 04:28.37 | JT | in a scenario that usually gives 99.98, the odd boot up gives average results of 99.97 instead |
| 04:34.09 | [shodan] | thanks blebleble |
| 04:38.36 | nosbig | How do I set up Asterisk to bridge two Zaptel channels? I want to provide call forwarding to cell phones as an option to simply leaving a message on voice mail... |
| 04:39.16 | nosbig | When I try this from my analog phone, it rings busy... |
| 04:39.36 | nosbig | Calling cell-->Asterisk-->another cell |
| 04:40.06 | [hC] | JT: odd.. |
| 04:41.02 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-86-116.lsanca.dsl-w.verizon.net) |
| 04:41.32 | [hC] | well, I'm gonna head home now... I'll touch base with you tomorrow JT and see how your testing came up |
| 04:41.50 | terrapen | sangoma a104 is a pain in my ass right now |
| 04:42.34 | mog | you could get a nice digium card terrapen ^_^ |
| 04:42.48 | russellb | mog: :D |
| 04:43.02 | nosbig | The Digium cards are nice.... |
| 04:43.07 | terrapen | this is actually for TCP/IP over T1, not TDM |
| 04:43.17 | russellb | Digium cards support that, too. |
| 04:43.20 | mog | you can do tcp/ip with digium cards |
| 04:43.53 | terrapen | i'm trying to do four point-to-point interfaces under OpenBSD and trunk them |
| 04:43.53 | terrapen | don't think digiums are supported under OpenBSD |
| 04:43.53 | terrapen | if they were, i'd certainly try them |
| 04:43.57 | russellb | no, they are not. |
| 04:43.58 | russellb | oh well |
| 04:44.02 | terrapen | doesn't look like the sangoma drivers under openbsd support trunking |
| 04:44.25 | terrapen | yeah, my father's company had a lot better luck with the digium cards for voice |
| 04:44.34 | terrapen | but at my company we are on 100% openbsd |
| 04:44.45 | terrapen | for network equip, anyway |
| 04:44.58 | russellb | perhaps you should use a real operating system, that is, one with a linux kernel |
| 04:45.08 | terrapen | suuuuuure |
| 04:45.15 | russellb | hehe |
| 04:45.18 | file | Vista, now that's a real operating system |
| 04:45.21 | russellb | file: you win |
| 04:45.23 | mog | beginith the distro war has |
| 04:45.32 | file | speaking of which... you just gave me an idea |
| 04:45.39 | russellb | no i didn't |
| 04:48.00 | mog | huntsville was on daily show tonight |
| 04:48.07 | file | mog: yup |
| 04:49.17 | sevard | mog: you're your own best friend? |
| 04:49.35 | mog | yes as well as being from final fantasy and star trek |
| 04:49.38 | mog | and a cat |
| 04:49.57 | sevard | eff ff and st, eff cats too. spaceballs ftw. |
| 04:50.04 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 04:50.08 | sevard | also, mr. oogie boogie man ftw. |
| 04:50.17 | mog | well i think Matt OGorman for the winn |
| 04:50.30 | mog | one day i will be number 1 in wikipedia for mog |
| 04:50.35 | sevard | what about john candy? |
| 04:50.42 | h3x | i wish the wiki had a decent example of how to add remote agents |
| 04:51.02 | sevard | h3x: the wiki is pretty bad, please write something better. |
| 04:51.04 | h3x | do you have to use Local/ ? |
| 04:52.00 | Qwell | mog: one day? That can be arranged |
| 04:53.12 | *** join/#asterisk Dovid (n=Dovid@line134-239.adsl.actcom.co.il) |
| 04:53.52 | Dovid | i just saw the chage log on asterisk from 1.2 to 1.4. I have a smile from ear to ear :) :) :) :) |
| 04:55.12 | russellb | hehe |
| 04:55.12 | russellb | yay |
| 04:55.24 | russellb | Qwell wrote all of it |
| 04:55.29 | sevard | I haven't read it yet, or tried out the beta |
| 04:55.37 | russellb | every new feature, all him. yep. |
| 04:55.48 | sevard | suckup. |
| 04:55.55 | h3x | Dovid: my friend here is watching my screen saying "I don't know why that guy is smiling, I am sure they broke a bunch of shit..." |
| 04:55.59 | h3x | hahah |
| 04:56.07 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 04:56.13 | Dovid | hehe. |
| 04:56.15 | russellb | h3x: that hurts! |
| 04:56.21 | Dovid | i goto learn all the new toys. |
| 04:56.34 | Dovid | time for a new server, boys its play time |
| 04:56.44 | h3x | hahah |
| 04:56.50 | sevard | I goto? your heads been stuck far too long in a dialplan, my friend. |
| 04:56.59 | h3x | the only thing that would be funnier is if kram wasnt a corporate whore and was around. |
| 04:57.19 | russellb | i know something that would be funnier |
| 04:57.29 | h3x | if bkw_ said NEXT !!! |
| 04:57.30 | russellb | me banning you for being a troll! |
| 04:57.36 | *** mode/#asterisk [+b %h3x!*@*] by russellb |
| 04:57.45 | sevard | kekeke. |
| 04:58.02 | Dovid | Qwell: how hard would it be to code asterisk so that real time would be "real time" so we wouldnt need the switch command ? |
| 04:58.12 | bkw_ | h3x whats up? |
| 04:58.18 | Dovid | as well as being able to use includes in real time ? |
| 04:58.41 | *** mode/#asterisk [-b %h3x!*@*] by russellb |
| 04:58.52 | h3x | i can feel for me but i just cant quite reach me |
| 04:59.08 | sevard | i tried out asterisk realtime, i think it needs major improvement as it turns out to be a glorified text editor. |
| 04:59.24 | justinu|laptop | lol |
| 04:59.31 | russellb | h3x: pretty much |
| 04:59.35 | russellb | :-p |
| 04:59.55 | sevard | justinu|laptop: sup baby-cakes |
| 05:00.06 | sevard | erm, what's happening vanilla face? |
| 05:00.17 | justinu|laptop | hey sevard |
| 05:00.47 | Dovid | i am outa here |
| 05:00.59 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Zaptel 1.2.11 released (Nov. 10, 2006) -=- Join #freepbx for freepbx/trixbox support. -=- Join #asterisk-gui to learn about the new Asterisk GUI framework |
| 05:01.14 | Dovid | its 7:00 AM and all anoying clinets are off the hone. |
| 05:01.30 | Qwell | russellb: subtract 2 minutes. :) |
| 05:01.42 | Dovid | ? |
| 05:01.43 | russellb | lol |
| 05:01.53 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Zaptel 1.2.11 released (Nov. 9, 2006) -=- Join #freepbx for freepbx/trixbox support. -=- Join #asterisk-gui to learn about the new Asterisk GUI framework |
| 05:01.56 | Qwell | :) |
| 05:02.01 | russellb | Qwell: pwned by staying up late |
| 05:02.11 | Qwell | we've all done it |
| 05:02.32 | Dovid | been nightly for the last week |
| 05:02.54 | Dovid | i use to be in plain IT with an incompitent boss. i have been thru the drill more than once |
| 05:03.08 | Qwell | file: 30%? |
| 05:03.09 | sevard | Does anyone know the actual difference between asterisk that us normal people get and "asterisk buisness edition?" does the "stability" stem from the fact that they use like.. asterisk 1.0? |
| 05:03.10 | Dovid | pulled an over night on avg. once every 2 weeks |
| 05:03.16 | file | I'm sitting 10 feet away from the wireless ap and it was not at full... lame |
| 05:03.43 | Qwell | sevard: BE is heavily tested |
| 05:04.00 | sevard | so, it's an entirely different release? |
| 05:04.09 | Dovid | i am connecting at excelent thru a repeater that is connected to a router. my pc is behind 4 heavy conceate walls ;) |
| 05:04.24 | russellb | yes, it is an entirely different release |
| 05:04.27 | russellb | and it is not based on 1.0 :) |
| 05:04.33 | file | it's based on 1.2... |
| 05:04.36 | file | well, 'B' is |
| 05:04.39 | russellb | right. |
| 05:05.11 | russellb | heavily regression tested. Also, when BE bugs get reported, they get #1 priority to get fixed |
| 05:05.16 | file | binary modules are also included by default, like Lumenvox, and you get a certain guarantee that your problems will get handled quickly and if you find a problem it'll try to get fixed asap |
| 05:05.22 | sevard | that's confusing. I thought they used us peons for beta testing on the 1.2, fixed the bugs, and gave the buisness customers an earlier, bug free, release. |
| 05:06.07 | russellb | same idea ... though anything we fix for BE does go into the open source versions as well |
| 05:06.21 | Qwell | if appropriate |
| 05:06.22 | sevard | So, if the buisness edition is an entirely differnent release, is it GPL? |
| 05:06.35 | russellb | no, it is commercially licensed. |
| 05:07.06 | sevard | so they don't publish any code |
| 05:07.07 | russellb | i'm sure this stuff is all on digium.com :) |
| 05:07.15 | russellb | it's the same code as open source asterisk |
| 05:07.24 | russellb | except for the pieces that *have* to be proprietary |
| 05:07.30 | sevard | russel: all their information says is "it's asterisk! it costs money" |
| 05:07.34 | russellb | which is the pieces that interface with proprietary addons |
| 05:08.15 | russellb | your benefits come in regression testing, commercial license so you can get proprietary 3rd party addons, and you get priority on getting bugs fixed |
| 05:09.04 | russellb | and you get some extra support, i believe |
| 05:09.17 | sevard | it'd be interesting if they published more information, the benifits, of these 'proprietary addons' |
| 05:09.17 | russellb | it has been a while since I have looked at all of the details |
| 05:09.32 | russellb | well, it's things like the lumenvox speech recognition engine |
| 05:10.03 | file | (already included) |
| 05:10.10 | russellb | cepstral text to speech engine ... |
| 05:10.26 | russellb | in the near future, dialogic channel driver |
| 05:10.48 | russellb | these are all things that the only way to use it with Asterisk is with a version of Asterisk that is commercially licensed |
| 05:10.54 | russellb | due to *their* licensing restrictions, not ours |
| 05:13.38 | file | and the conversation dies |
| 05:13.59 | russellb | heh, there were no more questions to answer :) |
| 05:14.07 | russellb | and on that note, i need to pass out |
| 05:14.10 | russellb | i am so tired that it hurts |
| 05:27.19 | *** join/#asterisk linlin (i=will@71.194.70.13) |
| 05:31.44 | *** join/#asterisk gerphimum (n=trekkie@cpe-68-206-83-62.satx.res.rr.com) |
| 05:34.30 | JT | benjk: about? |
| 05:37.00 | *** join/#asterisk groogs__ (n=greg@d38-54-164.commercial1.cgocable.net) |
| 05:39.01 | gerphimum | i would like to set up asterisk for a small business that has 2 lines. i have a spare computer floating around, but i dont have any modem cards. i have a good deal with my phone service, also, so i really dont want to have to switch to any kind of internet-based phone service. can i go and buy a couple modems from newegg, install the cards, plug the pstn lines into the modems, and then get some new sip phones to talk back to the p |
| 05:39.47 | Qwell | gerphimum: not unless you (or somebody else) write drivers for the modems - if it's even possible |
| 05:40.11 | Qwell | gerphimum: it's best to get a normal FXO, either a PCI card, or something like an SPA3000 (or equiv) |
| 05:40.58 | mistermocha | gerphimum: I agree with Qwell ... modems aren't FXO cards |
| 05:41.08 | gerphimum | what does FXO stand for |
| 05:41.11 | gerphimum | and mean o.O |
| 05:41.14 | Qwell | ~fxo |
| 05:41.15 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 05:41.53 | mistermocha | gerphimum: are you using ip phones or pots phones? |
| 05:42.22 | gerphimum | some of the phones will be in areas of the office that i wont be able to cheaply rewire |
| 05:42.34 | gerphimum | but i want some of the sweet features of voip |
| 05:43.00 | mistermocha | gerphimum: if you're building an asterisk system, you'll be able to exploit all the features of voip |
| 05:43.03 | gerphimum | so im planning on shopping around for some voip phones and use both in tandum |
| 05:43.15 | *** join/#asterisk remmo (n=chatzill@smack.isp.net.au) |
| 05:43.25 | mistermocha | but you won't be able to experience the joys until you get decent ip phones |
| 05:43.29 | mistermocha | what's your budget like? |
| 05:43.53 | gerphimum | well, ill probably have to buy that spa-3000, which is... $? |
| 05:43.54 | Qwell | mistermocha: same as most other small businesses - $20 :) |
| 05:44.01 | gerphimum | haha |
| 05:44.06 | mistermocha | that's a lot! |
| 05:44.13 | Qwell | gerphimum: I'm biased, but I'd look at a Digium TDM400p, with 2 FXO modules |
| 05:44.20 | mistermocha | you won't be able to get a good modem for $20 |
| 05:44.47 | gerphimum | Qwell >> i was looking at that earlier, but i got confused about those extra module cards... |
| 05:44.47 | Qwell | and if you've got a decent budget, I'm liking the Polycom IP430's |
| 05:45.01 | Qwell | gerphimum: You need FXO modules for phone lines |
| 05:45.09 | mistermocha | Qwell: have you seen the polycom 501 yet? |
| 05:45.16 | Qwell | mistermocha: sure |
| 05:45.42 | mistermocha | aastra makes a good phone for the money too |
| 05:45.45 | gerphimum | ~fxs |
| 05:45.55 | jbot | somebody said fxs was foreign exchange system - or the type of port you need to connect a analog device (phone, fax machine) to a pbx |
| 05:45.55 | gerphimum | :( |
| 05:45.55 | Qwell | ~fxofxs |
| 05:45.57 | jbot | from memory, fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 05:46.16 | mistermocha | you can probably pick one up for about half the cost of a polycom |
| 05:46.38 | mistermocha | the 1.4 firmware version is very stable too |
| 05:46.59 | gerphimum | fxo == from pstn, fxo == to a pots phone |
| 05:47.03 | gerphimum | right ? |
| 05:47.05 | Qwell | yes |
| 05:47.11 | Qwell | basically |
| 05:47.15 | mistermocha | correction |
| 05:47.22 | mistermocha | fxs == to pots phone |
| 05:47.26 | Qwell | oh, heh |
| 05:47.35 | Qwell | yeah, obviously |
| 05:47.42 | gerphimum | not a phone - it could be a whole string of em... |
| 05:48.16 | mistermocha | if they're all serial-connected to the same port, then they'd all ring and pick up the same line... just like your phones at home |
| 05:48.28 | gerphimum | right. |
| 05:48.29 | mistermocha | fxs is one line per port |
| 05:48.38 | mistermocha | fxo is one line per port too |
| 05:48.44 | Qwell | "line" |
| 05:48.48 | mistermocha | what's your broadband connection like? |
| 05:49.05 | gerphimum | road runner - 800kBps down, 40kBps up |
| 05:49.18 | mistermocha | DSL then |
| 05:49.23 | Qwell | cable |
| 05:49.39 | Qwell | unless RR does DSL now |
| 05:49.40 | mistermocha | how many phone lines do you need? |
| 05:49.53 | gerphimum | they bumped it up to 800 down recently... i was torrenting slackware (which i found isnt what i want...) at 800. i was like whoa |
| 05:49.55 | gerphimum | 2 |
| 05:49.57 | gerphimum | minimum |
| 05:50.16 | gerphimum | and yes, its cable |
| 05:50.25 | mistermocha | you don't need to run analog lines at all if you're running voip |
| 05:50.43 | mistermocha | and if you're not running analog, then you don't need any fxo ports |
| 05:50.53 | Qwell | completely offtopic: this "Philips Ambilight" TV seriously bugs me |
| 05:51.11 | mistermocha | sorry to hear it |
| 05:51.17 | mistermocha | I think you should drop-kick it |
| 05:51.33 | Qwell | well, just watching commercials of it |
| 05:51.33 | mcab | I think you should throw things at it |
| 05:51.41 | gerphimum | the hulk one / |
| 05:51.42 | gerphimum | ?* |
| 05:51.44 | Qwell | yeah |
| 05:51.54 | gerphimum | it really isnt that green irl |
| 05:51.59 | mistermocha | wtf am I doing in here? I should be working |
| 05:52.06 | Qwell | "Lighting effect simulated." |
| 05:52.10 | gerphimum | yeah lol |
| 05:52.31 | Qwell | Does that TV really not have those lights? |
| 05:52.43 | Qwell | because if not, I totally missed the point of "ambilight" |
| 05:52.52 | Qwell | erm, if so...stupid double negatives |
| 05:53.09 | gerphimum | have you never seen it in action |
| 05:53.15 | Qwell | nope |
| 05:53.19 | gerphimum | its kinda cool |
| 05:53.32 | gerphimum | but not as intense as they show it on the hulk commercial |
| 05:53.45 | Qwell | so it actually DOES make ambient light behind it? |
| 05:53.48 | gerphimum | yes. |
| 05:54.00 | gerphimum | they have a whole bank of leds behind it |
| 05:54.13 | gerphimum | it shines a light glow of light based on whatever is on the screen at the current time |
| 05:54.24 | gerphimum | so if youre watching aeon flux, itll mostly be blue |
| 05:54.57 | *** join/#asterisk PakiPenguin (i=wifigeek@linuxpakistan/admin/pakipenguin) |
| 05:55.02 | PakiPenguin | hello everyone |
| 05:55.12 | gerphimum | or if youre watching the hulk, itll be green |
| 05:55.17 | gerphimum | PakiPenguin >> howdy |
| 05:55.41 | gerphimum | anyways.. |
| 05:55.57 | *** join/#asterisk VibroMax (n=phisto@83.229.70.154) |
| 05:56.19 | PakiPenguin | :) |
| 05:57.25 | gerphimum | i dont want to have to change my phone service provider, but if itll save me $700 on that txotxs card, i might look into it |
| 05:57.39 | gerphimum | what are some good voip providers ? |
| 05:58.45 | Qwell | gerphimum: 2 fxo modules on a TDM400p is nowhere near $700 |
| 05:59.14 | gerphimum | i must have been looking in the wrong place.. how much would they be (around) |
| 05:59.23 | Qwell | $200ish |
| 05:59.29 | Qwell | total |
| 05:59.44 | gerphimum | so.. those things have 4 rj11 ports |
| 05:59.54 | Qwell | 4 ports, but you need a module per port |
| 06:00.00 | gerphimum | ill have to buy 2x txo modules |
| 06:00.05 | gerphimum | and 1x txs |
| 06:00.06 | Qwell | fxo, but yes |
| 06:00.11 | gerphimum | fxo |
| 06:00.19 | gerphimum | right (new terminology, sry) |
| 06:00.54 | gerphimum | the 1x fxs for the existing pots phones around the office that wont recieve the benefit of having their own extension |
| 06:01.23 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) |
| 06:02.22 | gerphimum | looks like thats close to $300.. what else in the way of voip providers are there ? |
| 06:07.40 | [shodan] | hmm , the feature that lets me transfert a call when I press "#" , I'm suppose to configure that in features.conf right ? |
| 06:08.19 | *** join/#asterisk aadilismail (n=adilisma@41-247-154-202.wol.net.pk) |
| 06:08.38 | [shodan] | because I'm looking at my features.conf , and there's nothing about transfer that is explicitely set in there , yet if I hit (as the callee) # then my extension (most of the time) I get to transfer my call to that extention |
| 06:09.09 | [shodan] | and when I hit "#" it says "transferring" so I guess it's * doing the transfert and not my ip phone/ATA |
| 06:09.35 | [TK]D-Fender | [shodan] : thats what happens when you use "t" and "T" to allow transfers. |
| 06:09.42 | [TK]D-Fender | [shodan] : What ATA? |
| 06:09.55 | [shodan] | I'm using SPA-2k2 and 3k2 |
| 06:10.04 | [shodan] | works great after some fiddling |
| 06:10.17 | [TK]D-Fender | [shodan] : then you shouldn't be using features.conf for transfers. use the interal method |
| 06:10.38 | [TK]D-Fender | [shodan] : DTMF transfers are lame and should be avoided |
| 06:11.28 | [shodan] | oh but then transfering call will be done differently on the polycom phone than the other phones , that'll be hard to explain to the family :\ |
| 06:13.23 | [TK]D-Fender | [shodan] : you use DTMF transfers on Polycom's?! OMG |
| 06:14.19 | [shodan] | try explaining to a 70 year old receptionnist how she has to press more then tsfr then dial the extention then press send compared to "#101" |
| 06:14.33 | [shodan] | the #101 seemed so much more attractive to me ;) |
| 06:14.53 | Strom_C | [shodan]: that's why you write up a document detailing how to do things and give it to your clients |
| 06:15.34 | [TK]D-Fender | [shodan] : so you risk mucking up IVR's DTMF because you're afraid of explaining that you have to hit a but that explicitly has the word TRANSFER on it> Retarded |
| 06:15.51 | [TK]D-Fender | Strom_C : Polycom makes a user's guide already :) |
| 06:15.58 | Strom_C | this is true |
| 06:16.16 | Strom_C | but printing up a preexisting user guide isn't billable time in my book |
| 06:16.28 | [shodan] | that would be a great way of letting them know that it's "not my problem" |
| 06:17.06 | [shodan] | the worst part is that the #$extension part almomst kinda work sortof perfectly |
| 06:17.25 | [shodan] | 'cept some call get lost in the void for some reason :\ |
| 06:18.44 | [TK]D-Fender | [shodan] : Yeah, the "loses customers" bit can be a downer ;) |
| 06:19.43 | [shodan] | it's doctor's office so they don't really care about losing patients ;) it's when some medical specialist gets hungup that I get yelled at ;) |
| 06:20.39 | [TK]D-Fender | [shodan] : Just train her to push the "blatently labeled button" and be done with it. |
| 06:20.56 | [TK]D-Fender | [shodan] : and ATA's should transfer with hook-flash using the ATA's internal features. |
| 06:21.17 | [TK]D-Fender | [shodan] : I for one LIKE the SPA's |
| 06:22.23 | [TK]D-Fender | [shodan] : Dependency on DTMF transfers and the like is the reason I tell people to avoid Zaptel FXS (not to mention server load, hardware concerns, reduced functionality, etc) |
| 06:22.39 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 06:23.01 | [shodan] | hmm , the SPA way is quite tastefully done , and a lot less risk of losing calls |
| 06:23.22 | [shodan] | I'll think I'll give her some analog phone with an SPA and I'll keep the polycom for me heheheheh |
| 06:24.10 | [shodan] | [TK]D-Fender, you're from the mtl aera right ? |
| 06:24.28 | [TK]D-Fender | file : I still think about picking up an AL2216 for myself, jsut can't seem to validate the expense :) |
| 06:24.33 | [TK]D-Fender | [shodan] : Yup |
| 06:24.47 | file | [TK]D-Fender: just do it! :D |
| 06:25.06 | [shodan] | k, you're the one who recommended me gentek , thank , but just FIY , when you give them an email they sell your email to real spammers |
| 06:25.28 | [TK]D-Fender | [shodan] : take your word for it.... |
| 06:25.35 | PakiPenguin | i know its offtopic , but does anyone know where i can get my hands on a gsm modem ( siemens / nokia / wavecom ? ) |
| 06:25.55 | [TK]D-Fender | PakiPenguin : Thats not very OT here... |
| 06:26.17 | [shodan] | I subscribed to their website using gentek@mydomainname.something and now that gentek@ address is getting spam, I think someone is stealing they're customer list |
| 06:26.26 | *** join/#asterisk Guest^DJ (n=me@211.24.146.11) |
| 06:26.43 | [shodan] | still, they offer convenient service so thanks for the mentionning them :) |
| 06:26.44 | Guest^DJ | hey, anyone try out TDM-GSM Module from http://www.asteriskgsmmodule.com/ yet |
| 06:26.50 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 06:27.08 | PakiPenguin | Guest^DJ: i have got one gateway from www.voismart.net they are good! |
| 06:27.42 | file | that is sort of cool... |
| 06:28.04 | Tebi_ | 2N Stargate and VoiceBlue works too... |
| 06:28.06 | gerphimum | do you plug a sim card into it or what |
| 06:28.10 | Guest^DJ | PakiPenguin: www.voismart.net require username and password ? |
| 06:28.14 | [TK]D-Fender | [shodan] : Good choice using a "disposable" box name for them. |
| 06:28.19 | file | it would be cooler if I could see what GSM module they used |
| 06:28.43 | PakiPenguin | oh www.voismart.it try that |
| 06:28.43 | PakiPenguin | http://open.voismart.it |
| 06:29.48 | PakiPenguin | www.voismart.it doesnt need password |
| 06:29.50 | [shodan] | how much is that vGSM card worth ? |
| 06:30.21 | [TK]D-Fender | ok, definately time to head to bed. Later all. |
| 06:30.33 | file | ah |
| 06:30.35 | PakiPenguin | [shodan] : i am not sure, i've got a complete machine with 5 cards in it ( 4 gsm modules each ) |
| 06:30.40 | file | it's a Siemens |
| 06:31.19 | Un1x | hey filehow you tonite? |
| 06:31.29 | file | I'm good, sleepy... soon going |
| 06:31.31 | file | you? |
| 06:32.04 | Guest^DJ | PakiPenguin: it would leave my pocket a big hole by just buying 1 gsm module |
| 06:32.20 | PakiPenguin | :) |
| 06:32.22 | JT | anyone know how to set the outgoing MSN number (not callerid) on a pri acting as network? |
| 06:33.07 | file | huh... they must have the ARM CPU doing the GSM modem handling... interesting |
| 06:33.37 | Un1x | good going to watcvh dead president soon |
| 06:33.40 | *** join/#asterisk hieunm_vips (n=hieunm_v@210.245.57.162) |
| 06:33.47 | Un1x | *watch |
| 06:33.49 | [shodan] | I wouldn't pay more than 100$usd per gsm module :\ |
| 06:34.02 | Un1x | [shodan] you mean GSM GATEWAY? |
| 06:34.20 | file | the actual GSM modem/module can be expensive depending on manufacturer |
| 06:34.25 | Guest^DJ | [shodan] : someone quoted me Euro 9,000.00 for 4 GSM |
| 06:34.36 | Un1x | hmm |
| 06:34.42 | Un1x | GSM module? |
| 06:34.49 | Guest^DJ | GSM gateway |
| 06:34.52 | JT | Un1x: junghanns.net |
| 06:34.55 | [shodan] | Guest^DJ for 9000 , you could just get 4x Nokia E70 and pocket the remaining 7000 euro |
| 06:34.56 | JT | you can buy gsm cards |
| 06:34.57 | Un1x | ahh okay well you can get one for 200$ |
| 06:35.02 | Un1x | but its only one SIm per module |
| 06:35.05 | JT | and from a few different places too |
| 06:35.07 | Un1x | its pretty good i got one :) |
| 06:35.24 | Un1x | heh file |
| 06:35.27 | Guest^DJ | i am now using 4 FXO and 4 X Nokia 32 |
| 06:35.41 | Un1x | anyone have a callback.agi script |
| 06:35.55 | Un1x | for when you wanna Call youre asterisk and the box says its busy and then it calls you back |
| 06:36.04 | JT | Un1x: funny you should ask, there's one on junghanns.net too |
| 06:36.07 | JT | callback |
| 06:36.29 | Un1x | yea i seen it but it doesn't really explain how to set it all up tho :( |
| 06:37.51 | [shodan] | damn , for 9000 euro , I'd get a USRP, a software defined radio, cost about 1000$, with a sim card reader , then pay 2000-3000 euro for a programmer to make a GSM TX and RX for the USRP and pocket the difference and I'd have unlimited concurent calls over GSM (to bandwidth limit of GSM) |
| 06:38.02 | PakiPenguin | hmmms |
| 06:38.25 | Un1x | fuck i dont like GSM |
| 06:38.35 | Un1x | because its not 3g technology and the speeds are SLOOOOWWW! |
| 06:38.40 | Un1x | datawise at least |
| 06:38.45 | JT | you can get stuff for much cheaper than 9000 euro |
| 06:38.48 | file | GPRS you mean? |
| 06:38.50 | JT | i guess it depends what it does |
| 06:38.52 | file | and EDGE? |
| 06:38.58 | Un1x | http://www.junghanns.net/en/callback.html |
| 06:39.07 | Un1x | doesn't have frigen callback.agi on there |
| 06:39.07 | JT | Un1x: people usually use gsm for voice so they have cheap voice calls |
| 06:39.28 | Un1x | well i was wondering if i only have one SIP porovider |
| 06:39.32 | Un1x | once asterisk cals me |
| 06:39.40 | Un1x | how is it going to provide a second dailtone for me to dail out? |
| 06:39.43 | JT | Un1x: calm down |
| 06:39.50 | JT | Un1x: the second code box |
| 06:39.53 | JT | is the php script |
| 06:40.01 | Guest^DJ | for now, i am quite happy with FXO - GSM with the exception that asterisk records the call is ANSWERED when FXO answered and dial out |
| 06:40.09 | JT | php is only one way of implementing it, mind you |
| 06:40.26 | JT | Un1x: app_disa |
| 06:40.30 | Un1x | JT i'm looking for the agi callback.agi |
| 06:40.41 | JT | do you know what agi is, Un1x ? |
| 06:40.42 | Un1x | and i assume i will have to get a Second provider in order to use callback.agi |
| 06:40.52 | Un1x | since 1 provider wont let me dail out 2 calls probably |
| 06:41.02 | JT | some sip providers do |
| 06:41.11 | *** join/#asterisk yxa (n=lonari@58.185.90.101) |
| 06:41.39 | Un1x | I See. |
| 06:41.40 | *** join/#asterisk oej_ (n=oej@apollo.webway.se) |
| 06:41.46 | JT | i'll ask again |
| 06:41.51 | JT | do you know what agi is? |
| 06:42.27 | Un1x | Yes |
| 06:42.40 | JT | then you'd realise the second code box is the agi |
| 06:43.00 | Un1x | AGI is supposed to be in perl tho not php far as i rember |
| 06:43.05 | JT | wrong |
| 06:43.22 | JT | it can be in any programming language on the face of the planet that can do unix STDIN and STDOUT pipes |
| 06:43.37 | Un1x | yea i figured that out just now |
| 06:43.41 | Un1x | i always assumed perl.. |
| 06:43.56 | Un1x | since its what majourity of the scripts are in.. and since i only know perl and not much of php |
| 06:44.04 | JT | assumptions are bad, especially when you're going to get hot under the collar over a "missing .agi" |
| 06:44.05 | hmmhesays | heh, yes you can bash script an agi |
| 06:44.19 | file | it's a hmmhesays! |
| 06:44.24 | hmmhesays | hello file |
| 06:44.28 | hmmhesays | i can't sleep |
| 06:44.45 | hmmhesays | i was thinking about playing some super mario brothers |
| 06:44.59 | Guest^DJ | PakiPenguin: there is no mention in price. how much did you buy the 4 GSM ? |
| 06:45.00 | shellshark | heh |
| 06:45.06 | shellshark | hmmhesays: snes9x? :) |
| 06:45.12 | hmmhesays | zsnes |
| 06:45.16 | shellshark | ah |
| 06:45.22 | JT | Guest^DJ: are you after a gsm standalone gateway box or a gsm card for a pc? |
| 06:45.32 | shellshark | hmmhesays: does it have intarweb support? |
| 06:45.55 | Guest^DJ | just GSM card for asterisk |
| 06:45.56 | shellshark | hmmhesays: perhaps we could get a game of super mario kart going ;) |
| 06:45.58 | hmmhesays | point to point, I knos snes9k has a kaillera client |
| 06:46.05 | hmmhesays | *know |
| 06:46.13 | shellshark | kaillera? |
| 06:46.25 | JT | Guest^DJ: ok |
| 06:46.38 | hmmhesays | kaillera has been compiled into many emulators for network support |
| 06:46.44 | hmmhesays | kaillera.com maybe? |
| 06:47.10 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:47.13 | JT | Guest^DJ: 4 port GSM card from junghanns.net has a list price of EUR$1600 |
| 06:47.35 | shellshark | hmmhesays: well either way... is it possible for us to get a game on? :) |
| 06:47.41 | Un1x | heh why does he want to go GSM Err people man i dont understand you guys |
| 06:47.47 | Un1x | unless you get fre calls to other mobile phones |
| 06:47.53 | Un1x | from the same or other gsm networks |
| 06:48.02 | hmmhesays | yeah but not right now, I need some mindless gaming I think |
| 06:48.05 | Un1x | then id save you money or else its going to cost pretty much the same ammount |
| 06:48.09 | JT | Un1x: in a lot of countries it is cheaper to call mobiles from mobiles than any landlines |
| 06:48.09 | Guest^DJ | JT: how much is that is US$ ? |
| 06:48.16 | JT | Guest^DJ: xe.net/ucc |
| 06:48.18 | shellshark | Un1x: you can recieve calls from GSM networks into an asterisk box :) |
| 06:48.43 | Un1x | I know that, but i just ment it makes no sense unless you plan on routing gsm calls to gsm gateway |
| 06:49.00 | shellshark | or receive from the gsm network |
| 06:49.01 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 06:49.02 | Un1x | assuming you have a script that checks dailed numbers by users and then once sees a specific area code directs the call via apropriate channel |
| 06:49.03 | JT | err if you make lots of outbound calls to mobiles it can be worth it |
| 06:49.08 | *** join/#asterisk lostpig (n=gabe@160.7.251.57) |
| 06:49.13 | Un1x | if its a mobile phone via the gsm gateway making the call free to you |
| 06:49.15 | Un1x | and cheap to customer |
| 06:49.19 | Un1x | if thats what you plan on doing.. |
| 06:49.40 | shellshark | Un1x: you can also have all your outbound mobile calls from within your office route out via your asterisk box, possibly to a VoIP provider where the call costs you practically nothing... |
| 06:49.48 | shellshark | Un1x: all transparent to the end user |
| 06:49.48 | Guest^DJ | approx US$ 2000 |
| 06:49.54 | JT | i hardly understand what you're saying, Un1x but it sounds like you're talking about Least Cost Routing, which can be implemented in the Asterisk dialplan |
| 06:50.06 | hmmhesays | t.38 hates me |
| 06:50.12 | JT | Guest^DJ: if that's what the currency convertor says |
| 06:50.37 | shellshark | hmmhesays: check out openpbx... it supports t.38 out of the box |
| 06:50.49 | lostpig | hey all - I'm having issues with DTMF and SIP over an OpenVPN tun using RFC2833. |
| 06:50.52 | hmmhesays | shellshark: haha, I have |
| 06:50.55 | shellshark | hmmhesays: I heard SER does as well, but i've not tried it |
| 06:50.59 | Guest^DJ | JT: for now, i will stick to my FXO+GSM |
| 06:51.11 | JT | what is your fxo + gsm? |
| 06:51.11 | hmmhesays | shellshark: ser doesn't touch the media unless you intentionally proxy it |
| 06:51.34 | lostpig | I don't know why it would work. The client is ekiga. It works if go over eth and not tun to a local Asterisk server. |
| 06:52.05 | hmmhesays | and I'm in the process of ironing out some openpbx issues |
| 06:52.19 | Guest^DJ | JT: just TDM400 --> RJ 11 --> Nokia GSM Terminal |
| 06:52.36 | JT | how much did the nokia terminal cost? |
| 06:52.39 | Un1x | according to youre comment of recieving calls from gsm networks u can already do that on asterisk just get a DID... |
| 06:52.57 | lostpig | It's late. I don't know why it WOULDN'T work. Just changes the meaning of the whole sentence. |
| 06:53.02 | Guest^DJ | 400 |
| 06:53.17 | Un1x | eh |
| 06:53.24 | Un1x | its 2am here lol |
| 06:53.30 | hmmhesays | i see digitech released a new line of effects pedals |
| 06:53.39 | JT | Un1x: why are you trying to fight people using gsm gateway, almost? there are a lot of different applications and scenarios in telephony, and gsm gateways fill a niche in a lot of cases |
| 06:53.56 | JT | Un1x: gsm gateways allow either cheap incoming or outgoing calls |
| 06:54.01 | JT | that's they main advantage |
| 06:54.10 | JT | perhaps redundancy from wired networks, too |
| 06:54.14 | Un1x | JT im not figting i'd just like some information so i refer to it for future needs |
| 06:54.38 | Un1x | JT you do know Landlines offer free incoming/outgoing calls as long as there... local |
| 06:54.39 | JT | main advantage is cost savings |
| 06:54.52 | Un1x | i guess it all depends on scenerio and country |
| 06:54.57 | JT | Un1x: calls to mobiles aren't considered local calls in most places |
| 06:55.17 | JT | and in north america, a lot of people have to pay to receive phone calls as well as make them |
| 06:55.23 | Un1x | they are in usa/canada/south asia/japan/norway/uk |
| 06:55.56 | JT | well mobile calls are more expensive than landlines in a lot of places |
| 06:56.10 | Un1x | lol if you go to south asia |
| 06:56.14 | Un1x | compettition is fierce |
| 06:56.20 | Un1x | less then 1cent per minute |
| 06:56.25 | JT | try australia |
| 06:56.31 | Un1x | the calls down there are literaly free |
| 06:56.38 | Un1x | asia is like IT sector of the world |
| 06:56.53 | Un1x | just people down ther ehave a different trend and way of life. |
| 06:56.53 | JT | uhh ok |
| 06:57.01 | Un1x | i was there in 2004 lol |
| 06:57.03 | Un1x | its how i know |
| 06:57.04 | JT | ok, but this is relevant, how? |
| 06:57.08 | JT | awesome |
| 06:57.15 | Un1x | because it depends on geographical location... |
| 06:57.17 | Un1x | and scenerio |
| 06:57.29 | JT | that's right |
| 06:57.40 | Un1x | but i just wanted to know what his scenerio was in order to maybe help or learn from him... and you said i was fighting people with gsm gateways |
| 06:57.44 | JT | i thought you were giving a tourism speel about south asia for a minute there |
| 06:58.00 | Un1x | ^^^ |
| 06:58.08 | JT | gsm gateways do have disadvantages |
| 06:58.17 | JT | but in some cases, they're economical to use |
| 06:58.20 | JT | oh |
| 06:58.20 | JT | and |
| 06:58.29 | JT | you can get these cheap little consumer gsm gateways |
| 06:58.35 | Un1x | yea i have one |
| 06:58.38 | hmmhesays | every experience i've had with gsm gateways has ended badly |
| 06:58.40 | hmmhesays | very badly |
| 06:58.53 | Un1x | i use it tho because i got free calling between the 2 sim cards |
| 06:58.55 | JT | that allow you to plug your normal mobile phone into the gateway, and into asterisk or what not |
| 06:59.05 | JT | so you can receive your mobile calls with asterisk |
| 06:59.06 | Un1x | and i dont have to pay provider extra 10$ a month for frigen free incoming calls |
| 06:59.10 | JT | or call out at mobile rates |
| 06:59.29 | Un1x | i just insrt my sim into my gsm gatewat |
| 06:59.32 | Un1x | it cost me like 200$ |
| 06:59.33 | Un1x | not even |
| 06:59.36 | Un1x | 150 i think it was |
| 06:59.46 | JT | some allow you to put the whole phone in |
| 07:00.07 | Un1x | eek theen you should be able to use cdma phones |
| 07:00.12 | Un1x | tdma too |
| 07:00.13 | Un1x | :P |
| 07:00.34 | JT | yeah i prefer cdma to gsm |
| 07:00.49 | Un1x | hehe, there is these littlew cdma modules |
| 07:00.58 | Un1x | theyve got the imei num etc all the info like a celly |
| 07:01.03 | Un1x | u just clal in and rtegister it |
| 07:01.09 | Un1x | no need to buy a sep phone or anytrhing |
| 07:01.10 | Un1x | pretty neet |
| 07:01.20 | JT | what are they for? |
| 07:02.02 | [shodan] | any sources for those cheap consumer gsm gateway except ebay ? |
| 07:02.08 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 07:02.17 | Un1x | JT |
| 07:02.22 | Un1x | there for making calls and stuff |
| 07:02.25 | hmmhesays | i used to use reliant gsm gateways |
| 07:02.26 | Un1x | its like gsm gateway |
| 07:02.35 | JT | so the same thing as a gsm gateway |
| 07:02.36 | JT | right |
| 07:02.38 | Un1x | except it doens't need sim and stuff and runs on CDMA |
| 07:02.39 | hmmhesays | plugged into e1 cards |
| 07:02.56 | JT | sims are better for getting cheap call rates though |
| 07:03.33 | Un1x | well there is alot of ways to get free calling and super super low rates |
| 07:03.37 | Un1x | just gotta know where and how |
| 07:03.44 | Un1x | if ur worried about money |
| 07:03.50 | Un1x | and for quality there is things you can do as well |
| 07:03.55 | hmmhesays | damnit firefox crashed AGAIN |
| 07:04.01 | gerphimum | speaking of which - what are some ways to get free/super low rated calling ? |
| 07:04.13 | Un1x | lol |
| 07:04.19 | Un1x | gerphimum depends where you are |
| 07:04.23 | [shodan] | hmmhesays good thing there's this new "Restore session" thingy |
| 07:04.25 | Un1x | i only know most ways in USA/canada |
| 07:04.33 | JT | seriously, if you can get everyday calling rates via SIP or IAX of under AUD20c/min to an australian mobile phone, i'd be pretty impressed |
| 07:04.33 | Un1x | i dont know any in other countrys |
| 07:04.34 | gerphimum | sry, i was reading yalls convo for... since you started the gsm debate |
| 07:04.43 | JT | so that doesn't really help me, Un1x |
| 07:04.49 | gerphimum | << texas |
| 07:04.53 | [shodan] | Un1x, what about in Canada ? ;) |
| 07:05.05 | Un1x | JT if i helped you with that what will you help me with ;P |
| 07:05.11 | Un1x | low rates man |
| 07:05.13 | Un1x | nothing else.. |
| 07:05.20 | JT | Un1x: i doubt you could help me with it |
| 07:05.25 | JT | i've looked around a fair bit |
| 07:05.42 | Un1x | heh i got a freind AU he gets his local calling for free in AU |
| 07:05.48 | Un1x | :/ i dont bother asking him how tho |
| 07:06.00 | Un1x | because i have no need for it :) |
| 07:06.06 | gerphimum | well |
| 07:06.14 | gerphimum | all the more reason to move to the down under |
| 07:06.16 | JT | Un1x: local calls |
| 07:06.17 | JT | big deal |
| 07:06.28 | JT | calls to mobiles is the only price killer |
| 07:06.42 | Un1x | lol |
| 07:06.47 | Un1x | no need to move down under |
| 07:06.57 | Un1x | i like canada its above everyone else :P |
| 07:07.01 | *** join/#asterisk dacleric (n=dacleric@p548212AF.dip0.t-ipconnect.de) |
| 07:07.09 | JT | except perhaps iceland |
| 07:07.10 | gerphimum | depends on how you look at the world o.O |
| 07:07.10 | gerphimum | o0oo |
| 07:07.11 | [shodan] | in AU I'd use an HAM transceiver hehe |
| 07:07.12 | Un1x | yea haha |
| 07:07.20 | JT | [shodan]: why? |
| 07:07.22 | JT | ... |
| 07:07.36 | JT | i bet you've never even been to Australia |
| 07:07.37 | gerphimum | but in the usa, what are some ways to get that there cheap/free phone service you were talking about ? |
| 07:07.45 | [shodan] | better reception in the outbacks no doubt |
| 07:08.06 | JT | [shodan]: ok, and how much of the population lives in "the outbacks", then? |
| 07:08.23 | [shodan] | how would I know , I've never been |
| 07:08.27 | gerphimum | lol |
| 07:08.31 | JT | then stop sprouting crap |
| 07:08.45 | gerphimum | JT is the channel almanac |
| 07:08.55 | JT | heh |
| 07:08.57 | JT | howso? |
| 07:09.06 | aadilismail | new born: .. how to dial further extensions after dialing some digits or phone numbers??? |
| 07:09.32 | Un1x | lol |
| 07:09.34 | gerphimum | these facts about population and ham transceiver range are quickly debunked |
| 07:09.35 | Un1x | you guys are jokes |
| 07:09.41 | gerphimum | i think its fun :) |
| 07:09.42 | Un1x | imma go watch death of a president |
| 07:09.47 | gerphimum | hey wait |
| 07:10.01 | gerphimum | :') |
| 07:10.05 | [shodan] | aadilismail, I've just done that , it's ugly believe me |
| 07:10.06 | gerphimum | :'(* |
| 07:10.09 | JT | Un1x: your english is a joke, if you're from canada :P |
| 07:11.08 | Un1x | no it's not a joke i'm just a lazy typist :P |
| 07:11.14 | Un1x | anyway i gotta go tho, cya |
| 07:11.16 | hmmhesays | cheers to that |
| 07:11.18 | gerphimum | gnight |
| 07:11.26 | [shodan] | is everyone ok with me pasting 10 lines or do I have to get my browser ? |
| 07:11.46 | gerphimum | sure, those last 10 lines were useless anyways |
| 07:11.53 | JT | it's more difficult to understand than a lot of the people who come here from non-english speaking countries, to be honest |
| 07:12.58 | [shodan] | I'm not hearing a no , so aadilismail , here is how you can dial some numbers after you first dialing , first is a macro |
| 07:13.05 | [shodan] | [macro-dialmacro] |
| 07:13.05 | [shodan] | exten => s,1,SIPDTMFmode(inband) |
| 07:13.05 | [shodan] | exten => s,2,GotoIF($[${LEN(${ARG1})} = 0]?8:3) |
| 07:13.06 | [shodan] | exten => s,3,Set(COUNT=0) |
| 07:13.06 | [shodan] | exten => s,4,Wait(0.3) |
| 07:13.06 | [shodan] | exten => s,5,SendDTMF(${ARG1:${COUNT}:1}) |
| 07:13.08 | [shodan] | exten => s,6,Set(COUNT=$[${COUNT} + 1]) |
| 07:13.10 | [shodan] | exten => s,7,GotoIF($[${LEN(${ARG1})} = ${COUNT}]?8:4) |
| 07:13.12 | [shodan] | exten => s,8,NoOp |
| 07:13.13 | Un1x | canada is all english.... we dont have any other language except french as a second langauge tho... |
| 07:13.14 | Un1x | i gotta go |
| 07:13.35 | JT | Un1x: i know, hence why i was surprised when you said you were from canada :P |
| 07:13.41 | [shodan] | aadilismail, you call it like that exten => 1,1,Dial(${TRUNK}/${PREFIX}${ARG1},180,nM(dialmacro,ww${ARG2})) |
| 07:14.50 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-14-65.cybersurf.com) |
| 07:18.24 | *** join/#asterisk petit-toon (n=petit-to@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 07:24.24 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 07:36.48 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-16-82.cybersurf.com) |
| 07:39.40 | EmleyMoor | Canadians could have enjoyed: American know-how, English Government and French culture |
| 07:40.11 | EmleyMoor | Instead, they got: English know-how, French Government and American culture |
| 07:41.35 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:41.38 | EmleyMoor | There is currently no NAT at all here, just in case bluregard is still here |
| 07:41.57 | *** join/#asterisk tengulre (n=tengulre@221.11.5.182) |
| 07:55.40 | *** join/#asterisk mega (i=mega@gateway/tor/x-4c10c6c16dd16859) |
| 07:59.50 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 08:09.33 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 08:13.00 | *** part/#asterisk hieunm_vips (n=hieunm_v@210.245.57.162) |
| 08:23.36 | *** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.35) |
| 08:24.22 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 08:27.08 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 08:27.22 | *** join/#asterisk linlin (i=will@71.194.70.13) |
| 08:33.03 | *** join/#asterisk UlbabraB (n=UlbabraB@host-84-222-47-242.cust-adsl.tiscali.it) |
| 08:38.45 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 08:40.16 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 08:54.20 | *** join/#asterisk sergee (i=opera@195.94.224.197) |
| 09:02.52 | *** join/#asterisk baconbuttie_uk (n=bob@host-84-9-143-221.bulldogdsl.com) |
| 09:08.50 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:14.22 | *** join/#asterisk sumasuma (n=sumasuma@61.14.86.23) |
| 09:14.50 | sumasuma | hi, i want to play my file when a person join in a conference, where i need to specify the file for the same ? |
| 09:24.04 | *** join/#asterisk linlin (i=will@c-71-194-70-13.hsd1.il.comcast.net) |
| 09:25.14 | *** join/#asterisk MGSsancho (n=user@adsl-67-126-141-216.dsl.irvnca.pacbell.net) |
| 09:25.30 | aadilismail | hi |
| 09:26.08 | MGSsancho | morning |
| 09:26.36 | aadilismail | good afternoon |
| 09:33.22 | sergee | mbranca: Matteo Brancaleoni? |
| 09:33.34 | mbranca | yes? |
| 09:37.31 | EmleyMoor | Other than SIP and IAX2, are there any other kinds of IP phone worth bothering with for a domestic setup? |
| 09:41.05 | EmleyMoor | (if buying all kit new, that is...) |
| 09:41.55 | sergee | mbranca: i found your email in google :) http://lists.digium.com/pipermail/asterisk-video/2006-July/000225.html |
| 09:42.12 | sergee | mbranca: can you share your sip.conf and extensions.conf? |
| 09:42.28 | mbranca | exten is just a dial |
| 09:42.34 | sergee | mbranca: i can't make GXV3000 to work with echo. |
| 09:42.48 | mbranca | sip is a standard entry with videosupport=yes and allow=h264 |
| 09:43.00 | mbranca | (videosupport also in global section) |
| 09:43.03 | mbranca | nothing more :) |
| 09:43.48 | sergee | mbranca: exten => 222,1,Echo() ? |
| 09:44.41 | mbranca | put an answer before |
| 09:48.32 | sergee | mbranca: will try now |
| 09:49.02 | EmleyMoor | New asterisk supports "n" as a priority - but what happens if you use "n" as the first? |
| 09:49.09 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 09:49.37 | mbranca | it does not work (at least on 1.2), first must always be 1 |
| 09:49.47 | EmleyMoor | Fair enough |
| 09:50.13 | sergee | mbranca: i can hear myself but i can't see myself (allow=all, but videosupport not in general..) |
| 09:50.23 | sergee | mbranca: i'm using latest trunk |
| 09:50.36 | mbranca | <mbranca> (videosupport also in global section) |
| 09:50.50 | EmleyMoor | Any protocol for phones worth me experimenting with for a domestic setup, other than IAX2 and SIP? |
| 09:51.01 | *** join/#asterisk LoneShadow (n=duh@c-67-188-235-220.hsd1.ca.comcast.net) |
| 09:51.39 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 09:52.55 | bXi | are there any good documents to read about faxing ? |
| 09:53.03 | sergee | mbranca: you mean [general] section in sip.conf? |
| 09:53.07 | mbranca | yes |
| 09:54.10 | saam4u | new born: if i dial _44208XXXNXXX... asterisk will ask to another extension number ... like dial extension 23 , 24 , 26 etc??? |
| 09:55.01 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) |
| 09:59.09 | EmleyMoor | saam4u: What do you mean by "ask to"?# |
| 10:02.20 | *** join/#asterisk lorinc (n=ang@caracas-2642.adsl.interware.hu) |
| 10:02.45 | EmleyMoor | Do you mean you want to set up a "To call Fred press 1..." thing? |
| 10:05.13 | saam4u | i mean ... i want to dial ... extensions numbers simply? |
| 10:05.59 | EmleyMoor | saam4u: Do you mean you want people to dial in and be able to then dial internally? |
| 10:06.01 | saam4u | further extensions |
| 10:06.07 | saam4u | yep |
| 10:06.31 | saam4u | yes |
| 10:07.10 | EmleyMoor | Well, there is a way but you have to be careful that it does not permit them to dial out... I think there;s an idea for this in the book |
| 10:07.29 | sergee | mbranca: sucess! |
| 10:07.41 | mbranca | :) |
| 10:07.41 | *** join/#asterisk mfroes (n=froes@200-162-218-81.corp.ajato.com.br) |
| 10:07.53 | sergee | mbranca: 2 gxv3000 works well with current trunk |
| 10:08.26 | mbranca | yes , why it should not :) ? |
| 10:08.44 | sergee | mbranca: someone should update wiki |
| 10:09.56 | mbranca | go for it |
| 10:10.06 | saam4u | sorry to say ... what is "book"? |
| 10:10.41 | *** join/#asterisk af_ (n=af@ip-179-179.sn1.eutelia.it) |
| 10:10.44 | EmleyMoor | Asterisk: The Future of Telephony |
| 10:18.14 | *** join/#asterisk tsurk0 (n=tsurko@80.72.68.86) |
| 10:19.20 | *** join/#asterisk olivier__ (n=olivier@obs92-4-82-239-116-113.fbx.proxad.net) |
| 10:20.04 | EmleyMoor | If I hadn't read it, I doubt I would have got set up quite so quickly |
| 10:27.01 | sergee | mbranca: thank you for support :) |
| 10:29.51 | *** join/#asterisk hello2007 (n=test@mail.splendor.net) |
| 10:29.55 | hello2007 | <PROTECTED> |
| 10:31.34 | [shodan] | is it just me or Polycoms 301 have glitchy blind transfer ? , I a caller that I want to blind transfer , I hit "More", then "Trnsfr", then "Blind" , it says "Blind Transfer to:" , and if I hit for example 103 , sometime it'll just get "10" and fail, leaving the caller on hold and the phone about to open a line on "3" even if you do it very fast it writes 10 then appears to hang for a quarter of a second then writes t |
| 10:31.43 | mbranca | de nada |
| 10:32.19 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 10:32.23 | aadilismail | new born... how to dial 44208878XXXX followed by access number like 2020??? means before dialing the destination number i want to dial access number ...how to do ? |
| 10:35.22 | *** join/#asterisk apardo (n=apardo@eu85-87-2-82.clientes.euskaltel.es) |
| 10:38.26 | JT | aadilismail: stop saying new born already :P |
| 10:39.47 | janisg | anyone have some exp with openser her e? |
| 10:40.03 | [shodan] | damnit aadilismail, I gave you the answer to that already , if you can't understand it, then stop asking |
| 10:41.04 | *** join/#asterisk UlbabraB (n=UlbabraB@host-84-222-47-242.cust-adsl.tiscali.it) |
| 10:42.25 | aadilismail | sorry for all |
| 10:53.39 | *** part/#asterisk sergee (i=opera@195.94.224.197) |
| 10:55.47 | infernix | i have horrible audio on outgoing calls (I have one SIP trunk for incoming and outgoing) and i'm trying to figure out why. i've tried twinkle and ekiga so far, both have their annoyances. what is the recommended softphone for linux? |
| 11:05.03 | *** join/#asterisk zeppelin_ (n=zeppelin@201-34-96-24.paemt700.dsl.brasiltelecom.net.br) |
| 11:08.31 | *** join/#asterisk adamowitz (n=adamowit@ip68-14-27-224.ri.ri.cox.net) |
| 11:10.04 | infernix | it's become more complicated now. i turned on canreinvite=no for the SIP trunks (both the voip provider and my local softphone). incoming calls work fine. outgoing calls don't seem to send any voice packets to the outgoing SIP trunk to the provider. |
| 11:10.42 | shellshark | firewall? |
| 11:10.53 | infernix | isnt canreinvite=no supposed to avoid that my local softphone (in a 192.168.0.x network) connects straight to the sip provider on the 'net? |
| 11:11.09 | shellshark | i'd be willing to bet it is sending out voice packets just fine, but not letting them back in ;) |
| 11:11.18 | infernix | what I want is to have asterisk handle all outgoing and incoming traffic, and it now seems that that isnt working. |
| 11:11.34 | infernix | hang on, i'll flush iptables incoming and outgoing for a sec. i doubt that's it hto |
| 11:12.23 | infernix | nope, its not the firewall. |
| 11:13.18 | infernix | actually, on outgoing calls I can hear myself talking on the other end (calling to my cellphone) - i hear myself on the cell, but what the cell is receiving isn't getting through to my softphone |
| 11:13.27 | infernix | incoming works just fine. |
| 11:13.49 | shellshark | what does tcpdump say? |
| 11:14.00 | shellshark | also what codec are you using? |
| 11:14.11 | infernix | tried with gsm, alaw, ulaw. |
| 11:14.23 | infernix | lemme check tcpdump. |
| 11:14.51 | shellshark | ulaw is probably the safest to try with |
| 11:15.53 | SheriF_SpacE | infernix: and hte huge bandwidth eater |
| 11:16.11 | infernix | my softphone is sending to asterisk. e.g. 192.168.0.163 (softphone) talks to 192.168.0.1 (asterisk) |
| 11:17.06 | infernix | if i do an incoming call vs an outgoing call, i dont seem to see a difference in traffic flows on the softphone end. |
| 11:17.31 | bXi | what softphone are you usng? |
| 11:18.18 | *** join/#asterisk UlbabraB (n=UlbabraB@host-84-222-47-242.cust-adsl.tiscali.it) |
| 11:19.05 | infernix | ekiga atm. |
| 11:19.31 | infernix | is it possible that due to an error, asterisk is telling the sip trunk on the provider end to send its packets to 192.168.0.163? |
| 11:19.44 | infernix | thats the only explanation i have really |
| 11:20.05 | infernix | if i remove canreinvite=no, it works but the incoming audio on outgoing calls is horrible |
| 11:21.59 | infernix | sigh. it worked fine last night. |
| 11:22.39 | infernix | even if i switch back to canreinvite=yes, the traffic flows seem the same to me |
| 11:22.45 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) |
| 11:22.46 | infernix | but obviously it's not |
| 11:23.10 | infernix | oh. actually, i stand corrected. |
| 11:23.43 | infernix | with canreinvite=yes, there's traffic from the sip provider going straight over the NAT to my softphone |
| 11:23.54 | infernix | but only on incoming calls |
| 11:24.28 | infernix | on outgoing calls, the softphone only talks with asterisk |
| 11:28.49 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 11:32.57 | *** join/#asterisk ghenry (n=ghenry@82-69-192-46.dsl.in-addr.zen.co.uk) |
| 11:32.59 | infernix | i've lost it. |
| 11:34.07 | infernix | i can't figure out why this isnt working. to me, it looks as if the audio which is coming in from the provider SIP trunk gets routed through asterisk, and that's where the audio jitter starts. if I play with canreinvite=no I lose all incoming audio on outgoing calls |
| 11:34.21 | infernix | incoming calls always seem to work, no matter what the setting. |
| 11:38.34 | Winkie | so does 1.4 have anything radical that wasn't in 1.2 that i should look out for? |
| 11:41.54 | Winkie | infernix: I don't have any particular experience with this but I assume you have nat=yes or similar in your provider def? personally i'd ensure SIP was forwarded to the asterisk box and disable reinvites |
| 11:41.56 | *** join/#asterisk dongs (n=HPUX@h193019.ppp.asahi-net.or.jp) |
| 11:41.58 | Winkie | SIP being 5060? |
| 11:42.05 | dongs | hey guys have FWD been dead all day today? |
| 11:42.33 | infernix | Winkie: i've tried all possible flag combos for nat and canreinvite now, it only works if i leave canreinvite=yes and then incoming sound on outgoing calls is horrifying. |
| 11:42.37 | infernix | i'm trying kiax now. |
| 11:42.55 | dongs | HALP |
| 11:43.06 | Winkie | dongs: lol hpux |
| 11:43.27 | infernix | and I don't really think I need nat=yes. asterisk runs on the linux firewall, which has the internet external IP on eth0 and the 192.168.0.1 ip on eth1. the softphone is connected through eth1 on 192.168.0.163 so i dont really think i need nat=yes |
| 11:43.29 | dongs | yeah, very funny is FWD down or what |
| 11:43.43 | dongs | ive got my asterisk console spammed with FWD timeout messages |
| 11:44.03 | Winkie | i have no idea i don't use fwd, i've asked in another channel for you |
| 11:44.18 | Winkie | infernix: is asterisk listening on eth0? |
| 11:44.47 | infernix | Winkie: its listening on 0.0.0.0. note the fact that incoming calls to the voip providers sip trunk work perfectly. |
| 11:45.17 | dongs | what other channel |
| 11:45.19 | dongs | is there a fwd channel |
| 11:45.27 | infernix | meaning, i dial the landline number with my cellphone, it goes to asterisk and then to my softphone. sounds perfect. |
| 11:45.59 | mfroes | how can i set the chan->language variable ? i want to overwrite it on the source code |
| 11:46.24 | Winkie | infernix: ok so incoming calls are flawless? you wouldn't need nat on the provider from what i can see |
| 11:46.47 | dongs | omfg |
| 11:47.10 | infernix | Winkie: exactly. |
| 11:47.49 | infernix | Winkie: with nat=no and canreinvite=yes on both provider and local softphone, all is fine except horrible incoming audio on outgoing calls |
| 11:48.24 | Winkie | infernix: horrible? aka jittery? |
| 11:48.36 | Winkie | dongs: what's the fwd gateway? |
| 11:48.49 | infernix | Winkie: jittery is the word, yes. |
| 11:48.50 | dongs | it pings |
| 11:48.54 | dongs | i duno what the fuck is up |
| 11:48.57 | dongs | its been running for years literally |
| 11:49.23 | dongs | but this morning I started getting Nov 10 20:44:45 NOTICE[10283]: chan_sip.c:5289 sip_reg_timeout: -- Registration for 'blah' timed out, trying again |
| 11:49.39 | dongs | i have another sip account not with fwd, and it works fine |
| 11:49.44 | infernix | Winkie: actually thats wrong. |
| 11:49.46 | dongs | so its not that SIP is blocked or somethign all of a sudden |
| 11:50.05 | Winkie | infernix: and with canreinvite=no on the sip peer you don't get any inbound audio? |
| 11:50.11 | infernix | Winkie: i need to set nat=yes on the softphone to get the jittery incoming audio on outgoing calls |
| 11:50.23 | oa | does anyone know of a free soft phone, which is able to handle G.729? |
| 11:50.25 | *** join/#asterisk oej_ (n=oej@apollo.webway.se) |
| 11:50.48 | Winkie | infernix: that's interesting what happens without nat=yes? |
| 11:50.49 | infernix | Winkie: correct. |
| 11:51.07 | Winkie | sorry i'm not exactly clued up with asterisk's nat support but it seems very strange you'd get no inbound audio |
| 11:51.31 | dongs | lol nat |
| 11:51.43 | Winkie | could you capture the actual packets being sent and check out whether it's reinviting or something like that? |
| 11:52.10 | *** join/#asterisk UlbabraB (n=UlbabraB@host-84-222-47-242.cust-adsl.tiscali.it) |
| 11:52.18 | infernix | Winkie: with provider on nat=no & canreinvite=no and softphone on nat=no & canreinvite=yes, all works except i have no incoming audio on outgoing calls |
| 11:53.01 | Winkie | infernix: what happens with all of those set to no? |
| 11:53.30 | *** join/#asterisk oej_ (n=oej@apollo.webway.se) |
| 11:54.08 | infernix | Winkie: same problem. |
| 11:54.15 | infernix | kiax is having no problems whatsoever, somehow. |
| 11:56.23 | dongs | oh man |
| 11:56.29 | dongs | fwd 'knowledge base' is fucking worthless |
| 11:57.49 | Winkie | infernix: sorry minor emergency here |
| 11:58.08 | infernix | if it werent for the fact that kiax uses dsp i would stick with that, but i really prefer alsa |
| 11:58.16 | Winkie | you mean oss? |
| 11:58.20 | infernix | yea |
| 11:58.21 | Winkie | and yes alsa is far superior |
| 11:58.32 | Winkie | i haven't had much use of ekiga yet to be honest |
| 11:58.34 | Winkie | although it seems promising |
| 11:58.54 | *** join/#asterisk HaMYaI (n=HaMYaI@ppp-58.8.8.197.revip2.asianet.co.th) |
| 11:58.55 | infernix | well i tried twinkle, same problem |
| 11:59.13 | HaMYaI | ohh Zaptel 1.2.11 is released! |
| 11:59.13 | Winkie | i don't know whether you're insulting my name or naming a client ;) |
| 11:59.36 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 11:59.44 | HaMYaI | good news but where can I get it? |
| 11:59.56 | infernix | oh man! now i'm losing it |
| 11:59.59 | infernix | twinkle works fine now |
| 12:00.03 | infernix | (its a client ;)) |
| 12:00.25 | infernix | i need a break. i just wasted a full hour |
| 12:00.53 | InfraRed | HaMYaI: ftp.asterisk.org ? |
| 12:01.19 | infernix | Winkie: thanks for your help though! |
| 12:01.42 | infernix | this nat business with SIP isnt easy |
| 12:01.47 | dongs | HELLO LOL DOES ANYONE HERE USE FWD |
| 12:01.52 | dongs | and whty hte fuck is it down |
| 12:02.00 | dongs | i demand QUALITY SERVICE for a FREE SERVICE |
| 12:02.08 | dongs | OR I WILL TAKE MY $0 BACk |
| 12:02.09 | infernix | dongs: are you on drugs? |
| 12:02.24 | dongs | no but i asked 5 minutes ago and got NO REPLIES |
| 12:02.28 | dongs | in a supposedly SUPPORT CHANNEL |
| 12:02.37 | infernix | well, did you pay for support? |
| 12:02.46 | macTijn | lol |
| 12:02.58 | dongs | OF COURSE NOT!!! THATS WHY IM ANGRY |
| 12:02.59 | infernix | if you didnt, you might want to try to just be polite and patient |
| 12:03.09 | infernix | it helps when it comes to people helping you for free. |
| 12:03.14 | macTijn | dongs: you sound just like a fwd user ;) |
| 12:03.31 | macTijn | infernix: do you understand the concept of "sarcasm" ? |
| 12:03.44 | infernix | uhh, sure. i think i missed it tho:P |
| 12:03.50 | macTijn | I think so :) |
| 12:04.29 | dongs | ugh |
| 12:04.30 | Winkie | infernix: sorry i wish i could help more |
| 12:04.31 | dongs | this is fucking bullshit |
| 12:04.36 | dongs | im missing mission-critical phone calls |
| 12:04.37 | dongs | because of this |
| 12:04.38 | macTijn | dongs: I sure hope for your sake it really was sarcasm ;) |
| 12:04.57 | dongs | and i cant be arsed to figure out how to do fwd over iax which I am sure works |
| 12:05.06 | infernix | Winkie: as i said it seems to work with Twinkle just fine now, might be an Ekiga thing. too bad i dont know what's actually going wrong :) |
| 12:05.06 | dongs | but this shit has worked since like 2-3 years ago |
| 12:05.10 | dongs | why the shit would it stop now |
| 12:05.22 | macTijn | dongs: why use a free service and make your core business depend on it? |
| 12:05.29 | macTijn | that's bad practice, man. |
| 12:05.32 | Winkie | infernix: we are having major issues with oracle here so there's a lot of drama |
| 12:05.36 | HaMYaI | InfraRed: did you mean "http://ftp.digium.com/pub/zaptel/releases/"? |
| 12:05.44 | Winkie | also dongs hey if you want business quality voip buy a business quality voip service :) |
| 12:05.46 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:05.47 | HaMYaI | InfraRed: it's not there |
| 12:05.49 | infernix | Winkie: go repair tables :) i am no priority:) |
| 12:06.01 | dongs | macTijn: LOL BECAUSE THE IDIOTS HERE CHARGE PER MINUTE FOR VOIP |
| 12:06.04 | Winkie | infernix: it's worse than that, it's a rogue oracle process sucking up memory and preventing other queries |
| 12:06.11 | infernix | Winkie: slay it |
| 12:06.14 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 12:06.25 | Winkie | i did, but that's not a solution :) |
| 12:06.35 | *** join/#asterisk zotz (n=zotz@24.244.133.107) |
| 12:06.36 | Winkie | i need to find what this process is and why it is doing what it's doing |
| 12:06.37 | infernix | aah well, at least for now it is :P |
| 12:06.55 | macTijn | dongs: stop using caps. does this look like the FWD helpdesk ? |
| 12:06.58 | infernix | good luck with tracing. i am off for lunch:) |
| 12:07.12 | macTijn | infernix: apparently, I was mistaken. |
| 12:07.14 | ScottyTM | can anybody suggest a usable and free sip client for windows? |
| 12:07.16 | *** join/#asterisk SheriF_SpacE (n=sherif@chatty.shellshark.net) |
| 12:07.29 | macTijn | infernix: dongs really is a Mad User[tm] |
| 12:07.45 | dongs | ScottyTM: uh xlite is pretty good |
| 12:07.49 | dongs | or whatever the shit its called now |
| 12:07.55 | infernix | macTijn: my gut feeling rarely lets me down ;) |
| 12:07.59 | dongs | critical path somethign. googling for xlite will get it to you thogh |
| 12:08.03 | macTijn | infernix: I had good hopes :) |
| 12:08.10 | *** join/#asterisk knarfly (n=knarfly@c-65-34-177-3.hsd1.fl.comcast.net) |
| 12:08.15 | dongs | sjphone doesnt suck either |
| 12:08.20 | dongs | if you cna deal with the asshat ui |
| 12:08.21 | macTijn | dongs: can you please stop swearing? go fix your problem or so. |
| 12:08.31 | dongs | macTijn: nobody has suggested a solution yet |
| 12:08.33 | ScottyTM | dongs: thx |
| 12:08.47 | knarfly | okay I have *-1.4 loaded and running...how to I get to this gui I've been hearing about? |
| 12:08.51 | macTijn | dongs: because no one is willing to help you. now I politely ask you to leave. |
| 12:09.54 | macTijn | dongs: another choice is that you calm down, stop swearing and properly tell us what your problem is. |
| 12:10.05 | macTijn | so it's up to you :) |
| 12:10.11 | dongs | macTijn: it doesnt work |
| 12:10.15 | dongs | it worked for years. |
| 12:10.17 | dongs | nothing changed. |
| 12:10.21 | dongs | been getting timeout errors all morning today. |
| 12:10.25 | macTijn | dongs: omg, you're such a user |
| 12:10.26 | dongs | another account works fine. |
| 12:10.47 | macTijn | "it doesn't work" does not do for a proper diagnose |
| 12:11.02 | dongs | i already pasted the error. |
| 12:11.03 | macTijn | do you get error messages? |
| 12:11.07 | macTijn | if so, which one ? |
| 12:11.12 | macTijn | did you try and contact FWD ? |
| 12:11.23 | dongs | 20:48 < dongs> its been running for years literally |
| 12:11.23 | dongs | 20:49 < dongs> but this morning I started getting Nov 10 20:44:45 NOTICE[10283]: chan_sip.c:5289 sip_reg_timeout: |
| 12:11.26 | dongs | <PROTECTED> |
| 12:11.27 | macTijn | ask them if they may have problems with one of their nodes or so |
| 12:11.27 | dongs | how hte hell do I contact them if they dont work |
| 12:11.44 | macTijn | cell. phone. |
| 12:12.12 | knarfly | how do I load the gui in * 1.4 |
| 12:12.26 | macTijn | and like you said, you have a working account. set up that account and try to reach them over that. |
| 12:12.56 | dongs | thats a different provider |
| 12:12.58 | dongs | and it works just fine. |
| 12:13.06 | macTijn | then use that :) |
| 12:13.10 | dongs | for what? |
| 12:13.12 | macTijn | problem solved |
| 12:13.13 | macTijn | NEXT! |
| 12:13.20 | dongs | i ahve calls regisered to the FWD shit |
| 12:13.22 | dongs | that are being missed |
| 12:13.24 | dongs | !!!!!! |
| 12:13.57 | macTijn | dongs: do I have a sign "helpdesk" on my forehead ? |
| 12:14.04 | dongs | no but youre the only one talking |
| 12:14.07 | macTijn | or "FWD" |
| 12:14.09 | macTijn | yeah |
| 12:14.14 | macTijn | and if you continue to be rude |
| 12:14.18 | macTijn | I won't be talking any more. |
| 12:14.31 | knarfly | dongs, FWD is often down for no reason...give it a while it might come back online |
| 12:14.35 | macTijn | yelling won't get you anywhere |
| 12:14.42 | dongs | i havent noticed too many problems |
| 12:14.45 | dongs | its never been down for 12+ hours |
| 12:14.47 | dongs | like it is now |
| 12:14.58 | macTijn | give them a call, I say |
| 12:15.07 | HaMYaI | anyone knows how to get the Zaptel 1.2.11? |
| 12:15.07 | macTijn | they probably have "real" numbers as well |
| 12:15.07 | knarfly | dongs, there's a first time for everything... |
| 12:15.28 | HaMYaI | it's not at http://ftp.digium.com/pub/zaptel/releases/ |
| 12:18.02 | intralanman | HaMYaI: did you look in old realeases? |
| 12:18.08 | *** join/#asterisk CyberKnet (n=CyberKne@ip68-13-246-61.ok.ok.cox.net) |
| 12:18.26 | *** join/#asterisk UlbabraB (n=UlbabraB@host-84-222-47-242.cust-adsl.tiscali.it) |
| 12:19.16 | intralanman | is there an old-releases for zaptel? |
| 12:21.14 | *** join/#asterisk Ebola (n=Ebola@host86-136-130-111.range86-136.btcentralplus.com) |
| 12:22.34 | intralanman | HaMYaI: btw, don't mind me.... it's really really early for me and i didn't realize what i was talking about ;) |
| 12:23.39 | *** join/#asterisk p0g0 (n=pogo@madwifi/support/p0g0) |
| 12:27.50 | HaMYaI | intralanman: =) |
| 12:28.16 | *** join/#asterisk ambriento (n=ambrient@201-27-81-188.dsl.telesp.net.br) |
| 12:32.31 | inspired | hi, where is the karma hall of shame? |
| 12:32.42 | inspired | I found the hall of fame, but I want to see the hall of shame, as I think I might be leading it |
| 12:32.49 | inspired | thanks |
| 12:34.05 | *** join/#asterisk FreezeS (n=Gladius@82.208.156.94) |
| 12:34.16 | FreezeS | hey guys |
| 12:34.17 | FreezeS | I've got a question |
| 12:34.28 | FreezeS | what are the big differences between 1.2 and 1.4 ? |
| 12:34.49 | monsted | FreezeS: 0.2 |
| 12:35.19 | intralanman | heheh, 1.4 has many more bugs |
| 12:35.40 | FreezeS | what are the extra features ? |
| 12:36.10 | intralanman | wounldn't that sort of thing be in a changelog or something? |
| 12:36.11 | EmleyMoor | I can't get on FWD at the moment (over SIP) |
| 12:36.14 | EmleyMoor | Is it down? |
| 12:36.30 | FreezeS | is somewhere a comparison table ? |
| 12:37.13 | intralanman | probably not until it's released.... although i could be wrong, wouldn't be the first time this AM |
| 12:38.25 | FreezeS | I found something |
| 12:38.31 | FreezeS | quite a lot of changes |
| 12:39.20 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 12:39.33 | intralanman | FreezeS: what did you find? |
| 12:40.26 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 12:40.38 | FreezeS | http://svn.digium.com/view/asterisk/branches/1.4/CHANGES?view=markup |
| 12:41.12 | intralanman | oh, cool |
| 12:42.09 | *** join/#asterisk kiddy (n=kiddy@59.93.32.220) |
| 12:42.21 | FreezeS | can't wait for it to be ready for production |
| 12:45.58 | intralanman | yeah, that'll be nice |
| 12:48.19 | Zordrak | Who wants to give me a compelling reason or two why someone should choose Asterisk over OpenPBX? |
| 12:48.22 | intralanman | not that it matters to me too much, i'm still trying to talk management into 1.2 on some of our machines |
| 12:50.16 | FreezeS | Zordrak: I see on the OpenPBX page that they just released a release candidate |
| 12:50.40 | Zordrak | .. |
| 12:50.56 | intralanman | it's a fork of asterisk? |
| 12:51.22 | FreezeS | yes |
| 12:51.33 | FreezeS | and most of its features are in 1.4 |
| 12:52.08 | Zordrak | And so, to the original question.. ? |
| 12:52.32 | FreezeS | that's it, I don't see a real reason |
| 12:53.09 | Zordrak | Well, the primary reason I can see is the use of POSIX timing to prevent you having problems when you don't buy a Digium PCI board |
| 12:53.42 | FreezeS | yes, asterisk is centered around digium cards |
| 12:53.51 | FreezeS | they need to sell them, you know... |
| 12:54.07 | FreezeS | but there are patches to switch the timing method as I remember |
| 12:55.03 | Zordrak | FreezeS: "they need to sell them, you know..." that's hardly a reason for building software that lacks direct support for other hardware - selling hardware should be done based on the quality of the hardware, its own internal firmware, and the support provided for it |
| 12:55.26 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 12:57.14 | Zordrak | But I don't want to drop into argument about the merits of proprietary building.. what I want is someone to give direct reasons for Asterisk over OpenPBX... so far I haven't really found any. But I have been given many reasons for OpenPBX over Asterisk. I don't exactly have balanced information at the moment. So far all I've been told about why to use Asterisk is because it was the originator. Which doesn't really mean much. |
| 12:58.28 | cpm | Zordrak, been using asterisk for a couple of years, haven't used openpbx |
| 12:58.33 | cpm | that's reason enough for me. |
| 12:58.33 | intralanman | Zordrak: who's backing openpbx? i don't really know much about it |
| 12:58.46 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 12:58.48 | Zordrak | Everyone I have spoken to so far. |
| 12:58.54 | intralanman | a real company (digium) is backing asterisk |
| 12:58.56 | cpm | having high hopes for freeswitch |
| 12:59.15 | kiddy | I was trying to make astfax working and but its not sending a fax |
| 12:59.26 | Zordrak | intralanman: indeed.. and that means profit is part of the decision making process |
| 12:59.27 | kiddy | can anybody help me to solve the problems with it ? |
| 12:59.40 | aadilismail | how to dial 44208878XXXX followed by access number like 2020??? means before dialing the destination number i want to dial access number ...how to do ? |
| 12:59.57 | FreezeS | Zordrak: and that's why we have zaptel timing |
| 13:00.05 | intralanman | Zordrak: and it also means that they will continue to commit resources to it's continued development and not just "get tired" of writing updates |
| 13:00.51 | intralanman | i've seen a lot of good open-source projects fall to the wayside because their creators and developers just couldn't commit time and money to continuing |
| 13:02.14 | Zordrak | So.. are we saying that on a pure software basis, there's no reason to use Asterisk over OpenPBX. Only use Asterisk if you are worried about the future? |
| 13:02.57 | cpm | If you want to use Asterisk, use it. If you want to use OpenPBX, use it. |
| 13:03.16 | cpm | what are you hunting for? |
| 13:03.26 | intralanman | Zordrak: i personally am saying i've never used openpbx and don't see a reason to change from asterisk.... so i can't give you an unbiased or knowlegable opinion |
| 13:04.21 | Zordrak | cpm: I am trying to build a reasoned argument to present to those in the company who decide where the money goes. Hence when a decision is made I need to present pros and cons for both options. I currently have a major list of pros for OpenPBX and cons for Asterisk.. but no-one can give me any real pros for Asterisk! |
| 13:05.12 | intralanman | Zordrak: maybe you can give us some background info then.... how long has the openpxb project been around? |
| 13:05.28 | intralanman | asterisk and digium have been around for a few years now |
| 13:05.48 | *** join/#asterisk davixx (n=davixx@staff-davixx-portable.euroweb-network.com) |
| 13:05.53 | Zordrak | OpenPBX forked at v1.0 |
| 13:05.55 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 13:05.55 | intralanman | if this is for a company.... i'd think that "the future" would be a pretty big concern |
| 13:05.58 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) |
| 13:06.18 | *** join/#asterisk alerios (n=alerios@190.24.97.148) |
| 13:06.59 | Zordrak | intralanman: indeed it IS a big concern... especially if Digium decide to close the Asterisk source - which they have the capability, and no doubt the will, to do - which is not something to worry about with OpenPBX |
| 13:07.45 | Zordrak | http://wiki.openpbx.org/tiki-index.php :: Most important differences between OpenPBX and Asterisk |
| 13:08.04 | Zordrak | That's an exmaple of "Reasons for OpenPBX" -- I have no such examples for Asterisk |
| 13:10.39 | intralanman | heheh, i wonder.... does openpbx have agi? that's kinda lame if it does.... |
| 13:10.59 | intralanman | i'd hope they at least renamed it if it does |
| 13:11.10 | intralanman | OGI or something |
| 13:11.32 | Zordrak | This is really quite concerning - no-one can tell me why Asterisk is still the way to go above all else? I was expecting at least a few hardcore fanboys who'd be backing it in every corner regardless of the reasons |
| 13:12.11 | davixx | hi |
| 13:12.28 | *** join/#asterisk UlbabraB (n=UlbabraB@host-84-222-47-242.cust-adsl.tiscali.it) |
| 13:12.58 | Zordrak | Maybe if I try mid-evening Eastern Standard Time.. |
| 13:13.04 | intralanman | Zordrak: you probably came in on the wrong timezone ;) hang around for a couple hours |
| 13:13.07 | intralanman | lol |
| 13:13.21 | Zordrak | tis 13:13 where I am :p |
| 13:13.35 | intralanman | UK or somewhere? |
| 13:14.07 | Zordrak | indeed |
| 13:14.10 | intralanman | that's right about GMT isn't it? |
| 13:14.10 | intralanman | lol |
| 13:14.16 | *** join/#asterisk drega (n=drega@mail.amcat.co.uk) |
| 13:14.24 | Zordrak | about 5 minutes west of GMT right now |
| 13:14.39 | Zordrak | if that |
| 13:15.49 | davixx | what is the way to follow about asterisk and voice synthesis ? |
| 13:16.42 | *** join/#asterisk juanjoc (n=juanjoc@201.216.212.113) |
| 13:19.26 | MrChimpy | AGI Rx << GET DATA "afltest/AstCallTV//10/45//126" 3000 12 |
| 13:19.26 | MrChimpy | <PROTECTED> |
| 13:19.41 | MrChimpy | doesn't play anything, and i can see strace managing to open the file |
| 13:19.48 | MrChimpy | no return |
| 13:20.00 | MrChimpy | just carries on to next AGI call |
| 13:23.27 | cpm | Zordrak, some of us just don't care. I'm not particularly keen on Asterisk, but it works, and has been working for quite some time. I like Digium, and I have no reason at all to see them as a 'bad guy', they've always been good to me. As freeswitch matures, I expect I'll replace some of my asterisk stuff with it. |
| 13:24.52 | cpm | As for Digium closing the source, they can't close what is already out here, they can only close what comes next. So what. They've done pretty well by the open community, I don't expect that to change. But yes, it could, but so what? |
| 13:26.02 | aadilismail | E45: 'readonly' option is set (add ! to override) .... what shud i do next? |
| 13:27.48 | intralanman | aadilismail: that looks like a vim error |
| 13:28.15 | aadilismail | so what shud i do ? |
| 13:28.31 | cpm | add write perms to the file, so you can write to it. |
| 13:28.37 | Zordrak | cpm: Thing is.. if I was already running Asterisk I would see no major reason to change. However I am about to start implementing a brand new PBX Server to replace an ageing proprietary PBX with a huge maintenance contract.. and which I start with is very important. Everything I've seen and read so far tells me I should go with OpenPBX (or indeed freeswitch depending on what they can do in the coming months)- but I'm looking for valid reasons to actually |
| 13:28.39 | cpm | or open it as a user who has perms |
| 13:28.49 | kumbalae | how will i get the dial status of the call dialled through spool ? |
| 13:29.10 | cpm | Zordrak, got a lot of vendors who support OpenPBX do ya? |
| 13:29.22 | Zordrak | Yes |
| 13:29.35 | cpm | Go for it then. |
| 13:29.40 | Zordrak | Especially as the main core is pretty much the same as Asterisk |
| 13:29.58 | Zordrak | I think I will actually be getting a Sangoma card. |
| 13:30.00 | cpm | Pretty much the same is meaningliess |
| 13:30.13 | kumbalae | how will i get the DIALSTATUS of the call dialled through spool ? |
| 13:30.59 | cpm | Zordrak, clues, |
| 13:31.08 | Zordrak | But then, it's not as important that the cards support the software as the software can be made to support the cards. And the fix/change process in OpenPBX is much faster simpler and less effort than trying to get support built into Asterisk |
| 13:31.33 | cpm | seen one asterisk installation, you've seen one asterisk installation. There are as many variations as there are folks who use it. |
| 13:31.55 | cpm | My question again was, Do you have vendors who will support OpenPBX? |
| 13:32.01 | Zordrak | In any case.. I only care about support for one card. The Sangoma AFT101U - that is laomst certainly what the server will be built around |
| 13:32.13 | Zordrak | My answer again is Yes |
| 13:32.26 | Zordrak | the following remark started "especially" not "because" |
| 13:32.29 | cpm | So, you will be planning, implementing, installing, and maintaining this set up yourself? |
| 13:32.38 | Zordrak | Along with one or two other people |
| 13:32.54 | cpm | Then do whatever the hell you want. It's going to be your show. |
| 13:33.44 | cpm | OpenWRT and RHEL can be said to have the same core, that doesn't mean supporting one implies support for the other |
| 13:33.58 | cpm | :) |
| 13:34.07 | Zordrak | It's not about what I want to do, it's about what's best for the company - hence the need for a reasoned argument for my decisiopn |
| 13:34.18 | cpm | I think you want to use OpenPBX, so go for it. I hope it does you proud, and I expect it will |
| 13:35.15 | cpm | being done with the old iron beast and it's attendant support, is going to be beautiful |
| 13:35.27 | Zordrak | cpm: That's really not what I said or meant. I have been researching the two.. and I can find many people who will give me reasons for OpenPBX.. and no-one who will give me reasons for asterisk other than "OpenPBX might not last" - it's that simple.. if I can be given good reasons for Asterisk, I may well go with Asterisk |
| 13:36.01 | MrChimpy | dur |
| 13:36.11 | *** join/#asterisk dasenjo (n=dasenjo@63.245.86.80) |
| 13:37.27 | *** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.35) |
| 13:37.55 | kumbalae | anyone please help ! |
| 13:38.20 | *** join/#asterisk xmws (n=me@213.219.175.223.adsl.dyn.edpnet.net) |
| 13:41.08 | xmws | when will * suport T38 termination? In the final 1.4? |
| 13:42.05 | *** join/#asterisk Splat (n=Splat@220-253-135-77.TAS.netspace.net.au) |
| 13:42.47 | *** join/#asterisk oliviert (n=oliviert@81.56.248.247) |
| 13:43.37 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 13:46.03 | oliviert | hi guys |
| 13:46.49 | oliviert | is someone may have a look at those log, it concern a SIP connexion normaly an inbound connexion, and i do have difficulties to understand the problem in the log |
| 13:46.50 | oliviert | http://pastebin.ca/244979 |
| 13:47.12 | aadilismail | can xpro and xlite communicate with each other by peer calling ? |
| 13:52.51 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:53.05 | kumbalae | aadilismail: yes |
| 13:53.07 | puzzled | morning |
| 13:57.16 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 14:08.08 | *** join/#asterisk lilwookie (n=jw@modemcable219.9-70-69.static.videotron.ca) |
| 14:08.11 | lilwookie | hi folks |
| 14:08.29 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:09.24 | lilwookie | quick questions, if I have in sip.conf under [4000] nat=yes when I d "show sip peers" should I see a Y under the col Nat ? because I see only N |
| 14:10.08 | *** join/#asterisk spr1te (n=spr1te@213.227.193.75) |
| 14:10.11 | EmleyMoor | lilwookie: Is that phone currently NATted? |
| 14:11.04 | EmleyMoor | I've got no outside line at the moment - FWDis down |
| 14:12.00 | lilwookie | Yes, it is.. forward 5060-5066 and 16384-16484 |
| 14:12.34 | EmleyMoor | I have no working experience... let me try something... |
| 14:12.46 | lilwookie | ah cool |
| 14:13.14 | EmleyMoor | Ah - N in that column means "NATted" |
| 14:13.22 | EmleyMoor | Nothing shows when not NATted |
| 14:13.29 | lilwookie | ah excellent.. :) thanks :) |
| 14:14.00 | lilwookie | it wierd I have a phone here in Canada on a natted network to a public * box, it worked I send it to India and it doesnt.. |
| 14:14.07 | lilwookie | *shake head* |
| 14:14.37 | aadilismail | want to dial some digits like 9147XXXXXX followed by access number(2222).... how can i ? |
| 14:14.52 | lilwookie | I like idefisk (iax), xten is nice too |
| 14:15.15 | EmleyMoor | I have moziax - but want to be used to SIP as well as IAX |
| 14:15.45 | EmleyMoor | (I will probably be getting hardphones at some stage and) there's more SIP than IAX |
| 14:16.11 | EmleyMoor | Hmmm... something funny happened with those brackets |
| 14:16.44 | lilwookie | hardphones I use right now are SPA941/2 and GXP2000's both work well on * |
| 14:17.25 | EmleyMoor | Much more urgent, though, is getting some old PSTN phones on (and the line) |
| 14:20.17 | *** join/#asterisk Chicago (n=Chicago@c-67-186-94-7.hsd1.in.comcast.net) |
| 14:25.02 | *** join/#asterisk Explisit (n=explisit@85.187.160.57) |
| 14:27.05 | Explisit | Hello, I want to start my Asterisk 1.2.12 with Alsa but when I dial 500 only some noise is made (with OSS no problem) |
| 14:27.28 | Explisit | I use Debian unstables with kernel 2.6.17.1 |
| 14:27.39 | Explisit | Can anyone help me ? |
| 14:28.15 | EmleyMoor | Explisit: What's 500? |
| 14:28.40 | Explisit | test number " dial 500 " |
| 14:29.11 | EmleyMoor | What have you set it up to do? Or, what is that number part of? |
| 14:29.31 | Explisit | it's part just of the default config |
| 14:29.38 | Explisit | to check if everything is running |
| 14:29.49 | Explisit | it plays some demo record |
| 14:30.23 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 14:30.33 | EmleyMoor | Oh, the Asterisk demo site? |
| 14:31.03 | EmleyMoor | I'd check that sound file if I were you |
| 14:32.11 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-166-226.buff.east.verizon.net) |
| 14:33.52 | EmleyMoor | Explisit: Not sure where you go from there - I don't use a sound card on my Asterisk box |
| 14:37.06 | *** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it) |
| 14:37.27 | Explisit | To be honest I'm not perfectly clear that I need sound card. :) |
| 14:37.44 | Explisit | But I think I will need it for Vmail |
| 14:38.00 | EmleyMoor | Explisit: Why particularly? I have voicemail |
| 14:38.33 | Rhizome | You never really need a sound card ;) |
| 14:40.15 | monsted | not in the server, no |
| 14:40.24 | EmleyMoor | A sound card (or headset) is only necessary on computers running softphones - the PABX doesn't actually need one |
| 14:40.32 | Explisit | well so my efforts to hear this " dial 500 " sound are really pointles :)) |
| 14:40.40 | monsted | but if you do a "call <number>" from the console, it'll use a sound card as a "soft phone" |
| 14:41.08 | monsted | err, "dial <number>" :) |
| 14:41.22 | Explisit | I see now. So if I use oss or alsa is then really matters? |
| 14:43.03 | monsted | no idea :) |
| 14:43.23 | monsted | <PROTECTED> |
| 14:43.28 | monsted | seems mine likes OSS |
| 14:45.31 | *** join/#asterisk slayer192 (n=slayer19@66.138.39.225) |
| 14:46.02 | Explisit | Well let me check it again to be sure. So in the Asterisk server I never need Sound card, because it really just route the voice to different providers and destinations. Ok but when I made a codec transformation G711 -> g729 , gsm-> g729 etc. ? |
| 14:46.26 | monsted | it does that in software |
| 14:46.35 | monsted | you'll never need a sound card in a server |
| 14:48.23 | Explisit | ok and the purpose of the digium card is to have connect me to for example ISDN. I mean in my company we will use Digium T110P (1 Pri port) |
| 14:48.39 | Explisit | and that's it. nothing more nothning less. |
| 14:48.41 | Explisit | ? |
| 14:49.07 | Explisit | (I'm pretty beginner in the Voice world) :) |
| 14:49.07 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:51.41 | *** join/#asterisk af_ (n=af@ip-179-179.sn1.eutelia.it) |
| 14:52.41 | monsted | server, software, some kind of link to the phone system, correct |
| 14:52.56 | monsted | you link to the phone system could be a SIP or IAX trunk, allowing you to skip the PRI too |
| 14:52.59 | monsted | your, even |
| 14:54.27 | monsted | hmm, my Cisco SIP phones don't seem to want to register if i use host=dynamic - any ideas what i can do? |
| 14:54.37 | monsted | Name/username Host Dyn Nat ACL Port Status |
| 14:54.41 | monsted | 101/101 (Unspecified) D 0 Unmonitored |
| 14:54.58 | Explisit | I can't skip PRI because of the network topology and also I must use 2-3 phone systems SIP, G729 |
| 14:55.04 | monsted | the ones where i put in the IP of the phone register perfectly |
| 14:59.41 | *** join/#asterisk delphus (n=delphus@mail.delphus.org) |
| 15:04.10 | *** join/#asterisk W9SH (n=Steve@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 15:04.16 | *** join/#asterisk zx6r (n=appak@ottawa-hs-206-191-33-80.d-ip.magma.ca) |
| 15:05.00 | zx6r | hey everyone |
| 15:09.23 | *** join/#asterisk dsfr (i=dsfr@pdpc/sponsor/digium/dsfr) |
| 15:13.54 | zx10r | hey little brother ;) |
| 15:15.03 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.25.Dial1.SanJose1.Level3.net) |
| 15:15.04 | monsted | (no, i actually drive an XV750) |
| 15:18.08 | vader-- | is there a way to detect a busy signal on the other end of a line? |
| 15:20.05 | *** join/#asterisk burd (n=burd@70-57-200-16.hlna.qwest.net) |
| 15:21.18 | burd | I'm trying to get the debian sarge vmail.cgi package working.. I made sure I had perl-suid installed and everything, i can run the script from the terminal but apache errors out with premature end of headers error.. any ideas or hints maybe? |
| 15:22.46 | *** join/#asterisk EFI-VJ (n=vj@ool-45756cb1.dyn.optonline.net) |
| 15:23.11 | EFI-VJ | I am getting a 404 , can anyone help me? |
| 15:24.33 | mercestes | Monsted: Qualify=yes |
| 15:24.54 | *** join/#asterisk marv[work] (n=timr@br0.asteriasgi.com) |
| 15:25.55 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 15:25.55 | zx6r | monsted: hehe |
| 15:26.16 | mercestes | vader--: ${DIALSTATUS} = BUSY |
| 15:27.07 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-42.ph.ph.cox.net) |
| 15:27.58 | monsted | mercestes: it's not NATted, but that still applies? |
| 15:29.19 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 15:30.41 | EFI-VJ | i am getting a 404 can someone help me? |
| 15:32.54 | zx6r | onto my question, i suppose ;) i'd like to give users the ability to select their hold music, once they've been put on hold. any idea of the best way to do this? i'm messing with the applicationmap of features.conf but i don't think it'll work completely... |
| 15:35.33 | monsted | mercestes: no joy - [Nov 10 16:34:44] WARNING[21816]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 15:35.50 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 15:36.21 | vader-- | ya im trying to create something that dials out to a place and if it gets a busy signal then it hangs up and calls again |
| 15:37.37 | mercestes | monsted: Qualtify=yes still applies. |
| 15:37.55 | mercestes | vader--: you telemarketer you....no help for you....die now pleas.e |
| 15:38.21 | mercestes | ./kick vader-- No telemarketing help! *muahahaha* |
| 15:39.04 | mercestes | monsted: if your phones are not registering then I am going to suggest it has nothing to do with host = dynamic v/s host=ip. |
| 15:39.12 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:39.12 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:41.23 | shellshark | mercestes: how do you know he's not doing outbound IVR, in an application such as a doctor's office appointment notification system? |
| 15:41.26 | vader-- | na im not hehe you actually have my wrong |
| 15:41.31 | vader-- | im trying to win a radio contest :) |
| 15:41.47 | *** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it) |
| 15:41.48 | shellshark | vader--: hahaha |
| 15:41.52 | *** part/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 15:42.14 | vader-- | :) |
| 15:42.42 | *** join/#asterisk riksta (n=rick@89.241.50.139) |
| 15:42.55 | vader-- | normally my questions are work related |
| 15:42.59 | monsted | mercestes: well, i have two phones here with the exact same setup, one using host=ip.of.the.phone and one using host=dynamic. The one with the IP works perfectly, the one with dynamic can make calls but not receive them |
| 15:43.05 | vader-- | but this one is for personal use to see if asterisk can actually do this |
| 15:44.11 | burd | this is just driving me nuts, gott be a solution as to why I would get Premature end of script headers on vmail.cgi |
| 15:44.34 | burd | i mean the script runs fine from the terminal |
| 15:44.41 | burd | or seems to anyhow |
| 15:44.46 | burd | but apache doesnt like it |
| 15:44.56 | burd | and apache isnt being very verbose about why |
| 15:45.27 | monsted | burd: that problem makes you a pirate with a steering wheel on your penis? |
| 15:45.49 | monsted | "Yarr! It's drivin' me nuts!" |
| 15:45.55 | burd | monsted: possibly |
| 15:46.15 | *** join/#asterisk quidpro (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 15:46.24 | shellshark | vader--: [elradio] exten => s,1,Dial(Zap/${EXTEN}|100|t) exten => _X.,n,GotoIf($["${DIALSTATUS}"="NOANSWER"||"${DIALSTATUS}" = "BUSY"]?redial) exten => s,n(redial),1,Goto(elradio|s|1) |
| 15:46.45 | shellshark | vader--: change that "_X." to just "s" |
| 15:47.26 | shellshark | vader--: that could be optimized a little bit, but it should work |
| 15:47.39 | vader-- | doesn't the s,1 have to _X.,1? |
| 15:47.44 | vader-- | so it can being the dialplan |
| 15:47.52 | shellshark | no you jump context from your main dialplan |
| 15:47.54 | vader-- | because DIALSTATUS would be = to "" |
| 15:48.34 | *** part/#asterisk aigroine (n=qzxcd@paranoia.coldev.org) |
| 15:48.46 | shellshark | [default] exten => 72346,1,Goto(elradio|s|1) |
| 15:49.05 | shellshark | where 72346 is numeric for "radio" :) |
| 15:50.24 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-228-131.dc.res.rr.com) |
| 15:50.36 | shellshark | always better to break out your dialplan into distinct contexts, so it's easier to debug or modify at a later date |
| 15:50.55 | quidpro | Why does Asterisk drive to 100% CPU (with no call load) sometimes after using SIP debug? |
| 15:51.15 | shellshark | quidpro: what version? |
| 15:52.12 | vader-- | :) |
| 15:53.41 | quidpro | 1.2.12.1 |
| 15:54.21 | quidpro | I did it after doing some SIP debugging yesterday (and after SIP NO DEBUG)... I've had it happens other times before, using SIP IAX2 or AGI debug |
| 15:54.30 | quidpro | Cant' remember which for the earlier incidents |
| 15:54.34 | mercestes | vader: on a busy signal ${dialstatus} would = busy so you would go to s-busy then goto s,1 again. |
| 15:54.49 | mercestes | vader: But shellshark is right...and I was just picking on you..;) |
| 15:55.46 | mercestes | vader: And if you have a PRI available you can do 24 consecutive dial(zap/g1/#&dial(zap/g1#... and entirely hammer the radio station. |
| 15:56.04 | *** join/#asterisk dsfr (i=dsfr@pdpc/sponsor/digium/dsfr) |
| 15:57.12 | vader-- | ya the only other issue is sometimes it comes up and says all circiuts are busy |
| 15:57.49 | mercestes | vader--: s-answered: Goto s,1 |
| 15:58.07 | mercestes | vader--: _s-.,1,goto(s,1) lol |
| 15:58.10 | *** join/#asterisk busster (i=Busster@69.15.193.2) |
| 15:58.18 | vader-- | ya we have a 23 channel pri line :) |
| 15:58.42 | burd | what is a nice frontend to manage asterisk remotely? |
| 15:59.21 | quidpro | burd: SSH. :) |
| 16:00.05 | burd | quidpro.. wow awesome suggestion |
| 16:00.31 | busster | Anyone have experience installing ooh323 add on? I keep getting an error that the module is incompatible with the version of asterisk compiled |
| 16:01.06 | quidpro | burd: Yeah, it's quiet portable between platforms. ;) |
| 16:02.14 | burd | quidpro.. I have seen lists of al kinds of management tools for the daemon.. I just dont have time to go through 20 of them or so to find that one or maybo none are worth a crap.. IE similar to my vmail.cgi script which is supposedly in Debian Stable yet doesn't work .. :| |
| 16:04.24 | mercestes | burd: I think he's quite serious. ssh is, honestly, the best way to do it. between the config files and the asterisk CLI (which works over ssh quite nicely) you can do most anything. |
| 16:04.51 | mercestes | burd: Unless your looking for something ssh just isn't offering you. |
| 16:05.18 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 16:06.30 | burd | mercestes, what I was more looking for was some interafce to manage extensions and users, add/remove voicemail withotu having to edit 5 files and write a damn flowchart for each user lol.. not that i am unable to do it, but there must be an easier way of managing all them config files |
| 16:06.47 | xmws | that will be vi |
| 16:07.31 | mercestes | burd: Your hoping for a gui maybe? |
| 16:07.47 | burd | mercestes, yah more like a web interface |
| 16:08.01 | shellshark | burd: freepbx! |
| 16:08.02 | burd | IE "front end" |
| 16:08.15 | mercestes | burd: Might I suggest metaswitch? or freepbx? |
| 16:08.29 | mercestes | burd: Or you could use mysql realtime and write yourself some php's with css or xml output. |
| 16:08.30 | shellshark | metaswitch... that's a new one by me |
| 16:08.41 | *** join/#asterisk _alex_mx_ (n=alex@200.78.229.18) |
| 16:08.44 | mercestes | It's a business class softswitch. |
| 16:09.05 | burd | if i need to i will move to mysql and start writing the stuff but once again that could pose to be time consuming |
| 16:10.35 | mercestes | Indeed. |
| 16:11.05 | mercestes | at that point, however, you could expose it ot the user with certain levels of "access" and have them change their own damn extensions, and then use scripts. |
| 16:11.24 | *** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org) |
| 16:12.08 | mercestes | We had dropdowns for Ring phone for <dropdown> seconds, and then forward phone to {voicemail,extension,phonenumber}. |
| 16:12.31 | mercestes | and then a script that read the Mysql data and constructed a AGI dial string based upon what information they gave. |
| 16:15.43 | *** join/#asterisk xxoxx (n=xxoxxx@tor/regular/xxoxx) |
| 16:17.08 | *** join/#asterisk syzygyBSD_ (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 16:17.28 | *** part/#asterisk Chicago (n=Chicago@c-67-186-94-7.hsd1.in.comcast.net) |
| 16:20.11 | *** join/#asterisk Chris-NB (n=chris@argos.campus-sbg.at) |
| 16:22.08 | *** part/#asterisk quidpro (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 16:22.12 | vader-- | ok i got it redialing but the problem is after about 5 times verizon picks up and says no good |
| 16:22.23 | vader-- | i tihnk i need to put a wait in there |
| 16:23.30 | zx6r | hrm. i think i've stumbled upon the unanswerable question. :P |
| 16:24.05 | mercestes | zx6r: What women are thinking?? |
| 16:24.31 | zx6r | mercestes, no, i know what they're thinking. they're thinking, "how can we screw up guys this time around?" |
| 16:26.06 | Ox0F0-0FF | Women are THINKING ???!!! |
| 16:26.44 | zx6r | long ago i gave up any and all hope of ever winning. just give up right away, it frees more time for other stuff ;) |
| 16:26.59 | *** join/#asterisk tecnico (n=tecnico@24.96.146.69) |
| 16:27.06 | mercestes | zx6r: Man has the last word in every arguement. It's "yes dear." |
| 16:27.27 | vader-- | mercestes ya i tried a 5 second wait and that didn't work |
| 16:27.30 | vader-- | after about 4 tries |
| 16:27.31 | *** join/#asterisk tao2 (n=tao@69.63.74.254) |
| 16:27.39 | vader-- | i get a verizon message saing call can't be completed as dialed |
| 16:27.59 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 16:28.12 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) [NETSPLIT VICTIM] |
| 16:28.12 | *** join/#asterisk kore (i=kore@mindwipe.org) [NETSPLIT VICTIM] |
| 16:28.12 | *** join/#asterisk Vegar (i=vegar@unaffiliated/vegar) [NETSPLIT VICTIM] |
| 16:28.12 | *** join/#asterisk Rhizome (n=Rhizome@host-81-191-147-145.bluecom.no) [NETSPLIT VICTIM] |
| 16:28.24 | tao2 | is there any one here that can give me a good provider for hoping out in South Africa. |
| 16:28.28 | vader-- | i wonder if verizon just freaks out about being called soo fast |
| 16:29.42 | zx6r | *scrolls and finds his question* |
| 16:30.14 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) [NETSPLIT VICTIM] |
| 16:31.49 | *** join/#asterisk xxoxx (n=xxoxxx@tor/regular/xxoxx) [NETSPLIT VICTIM] |
| 16:31.49 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) [NETSPLIT VICTIM] |
| 16:31.49 | *** join/#asterisk zx6r (n=appak@ottawa-hs-206-191-33-80.d-ip.magma.ca) [NETSPLIT VICTIM] |
| 16:31.49 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) [NETSPLIT VICTIM] |
| 16:31.49 | *** join/#asterisk olivier__ (n=olivier@obs92-4-82-239-116-113.fbx.proxad.net) [NETSPLIT VICTIM] |
| 16:31.49 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) [NETSPLIT VICTIM] |
| 16:31.49 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-16-82.cybersurf.com) [NETSPLIT VICTIM] |
| 16:31.49 | *** join/#asterisk groogs__ (n=greg@d38-54-164.commercial1.cgocable.net) [NETSPLIT VICTIM] |
| 16:31.49 | *** join/#asterisk angler_ (i=angler@nat/digium/x-785f9b26329583a5) [NETSPLIT VICTIM] |
| 16:31.49 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) [NETSPLIT VICTIM] |
| 16:31.50 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) [NETSPLIT VICTIM] |
| 16:31.50 | *** join/#asterisk ThaZZa (n=me@229.9.233.220.exetel.com.au) [NETSPLIT VICTIM] |
| 16:31.50 | *** join/#asterisk bkw_ (n=brian@ppp-70-128-126-144.dsl.tulsok.swbell.net) [NETSPLIT VICTIM] |
| 16:31.50 | *** join/#asterisk predder (n=predder@203.220.55.70) [NETSPLIT VICTIM] |
| 16:31.50 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) [NETSPLIT VICTIM] |
| 16:31.50 | *** join/#asterisk Simplix (n=loic@LSt-Amand-152-31-13-31.w82-127.abo.wanadoo.fr) [NETSPLIT VICTIM] |
| 16:31.50 | *** join/#asterisk JonR800 (i=jon@p1mp.org) [NETSPLIT VICTIM] |
| 16:31.50 | *** join/#asterisk smark (n=mdierolf@38.114.159.11) [NETSPLIT VICTIM] |
| 16:31.50 | *** join/#asterisk aydiosmio (n=aydiosmi@65.213.70.43) [NETSPLIT VICTIM] |
| 16:31.50 | *** join/#asterisk cavediver (i=jonas@trimix.eklof.eu) [NETSPLIT VICTIM] |
| 16:31.50 | *** join/#asterisk skeeziks (i=tim@66.23.208.6) [NETSPLIT VICTIM] |
| 16:31.51 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) [NETSPLIT VICTIM] |
| 16:31.51 | *** join/#asterisk L-info (n=linfo@g0962184.demon.co.uk) [NETSPLIT VICTIM] |
| 16:31.51 | *** join/#asterisk mrluring (i=mats@asswipe.temp.no) [NETSPLIT VICTIM] |
| 16:31.51 | *** join/#asterisk SwK (n=Silik0nJ@12-214-191-109.client.mchsi.com) [NETSPLIT VICTIM] |
| 16:31.51 | *** join/#asterisk DrkShdw (n=scorpio@unaffiliated/drkshdw) [NETSPLIT VICTIM] |
| 16:31.51 | *** join/#asterisk jontow (i=jontow@hijacked.us) [NETSPLIT VICTIM] |
| 16:31.51 | *** join/#asterisk oa (n=oal@62.61.133.90.generic-hostname.arrownet.dk) [NETSPLIT VICTIM] |
| 16:31.51 | *** join/#asterisk mixi (n=mixi@the.one.and.only.iammixi.de) [NETSPLIT VICTIM] |
| 16:31.51 | *** join/#asterisk devel (n=devel@wiggum.digitalcoven.com) [NETSPLIT VICTIM] |
| 16:33.24 | *** join/#asterisk Chris-NB (n=chris@argos.campus-sbg.at) [NETSPLIT VICTIM] |
| 16:33.24 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-228-131.dc.res.rr.com) [NETSPLIT VICTIM] |
| 16:33.24 | *** join/#asterisk EFI-VJ (n=vj@ool-45756cb1.dyn.optonline.net) [NETSPLIT VICTIM] |
| 16:33.24 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) [NETSPLIT VICTIM] |
| 16:33.24 | *** join/#asterisk juanjoc (n=juanjoc@201.216.212.113) [NETSPLIT VICTIM] |
| 16:33.24 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) [NETSPLIT VICTIM] |
| 16:33.24 | *** join/#asterisk dongs (n=HPUX@h193019.ppp.asahi-net.or.jp) [NETSPLIT VICTIM] |
| 16:33.24 | *** join/#asterisk CtRiX (n=CtRiX@ray.navynet.it) [NETSPLIT VICTIM] |
| 16:33.24 | *** join/#asterisk Luke-Jr (n=luke-jr@user-0c93tin.cable.mindspring.com) [NETSPLIT VICTIM] |
| 16:33.24 | *** join/#asterisk shellshark (n=x86@get.hooked.on.voip.with.shellshark.net) [NETSPLIT VICTIM] |
| 16:33.24 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) [NETSPLIT VICTIM] |
| 16:33.24 | *** join/#asterisk Inez (i=faceoff@devel4.net) [NETSPLIT VICTIM] |
| 16:33.24 | *** join/#asterisk iq (n=iq@unaffiliated/iq) [NETSPLIT VICTIM] |
| 16:33.25 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) [NETSPLIT VICTIM] |
| 16:33.25 | *** join/#asterisk pfn (n=pfnguyen@netblock-66-245-252-239.dslextreme.com) [NETSPLIT VICTIM] |
| 16:33.25 | *** join/#asterisk cromo (i=cromo@klej.net) [NETSPLIT VICTIM] |
| 16:33.25 | *** join/#asterisk hypnox (n=dan@lleuad.ocq.omnicea.net) [NETSPLIT VICTIM] |
| 16:33.25 | *** join/#asterisk Arnar (n=arnarb@landi.oddi.is) [NETSPLIT VICTIM] |
| 16:33.25 | *** join/#asterisk fholmes (n=fholmes@cpe-68-201-200-58.houston.res.rr.com) [NETSPLIT VICTIM] |
| 16:33.25 | *** join/#asterisk Mavvie (n=edwin@ppp40-206.lns2.syd6.internode.on.net) [NETSPLIT VICTIM] |
| 16:33.25 | *** join/#asterisk reza_ (i=reza@abort.boom.net) [NETSPLIT VICTIM] |
| 16:33.25 | *** join/#asterisk vader-- (n=johndoe@204.183.88.101) [NETSPLIT VICTIM] |
| 16:33.25 | *** join/#asterisk zamba (i=marius@flage.org) [NETSPLIT VICTIM] |
| 16:33.25 | *** join/#asterisk JunK-Y (n=junky@modemcable198.14-83-70.mc.videotron.ca) [NETSPLIT VICTIM] |
| 16:33.25 | *** join/#asterisk RaYmAn-Bx (i=rayman@kbhn-vbrg-sr0-vl212-213-185-15-16.perspektivbredband.net) [NETSPLIT VICTIM] |
| 16:33.26 | *** join/#asterisk bsdfreak (i=alex@ninja.paranode.net) [NETSPLIT VICTIM] |
| 16:33.26 | *** join/#asterisk Sponge_bob (n=None@cpe-66-27-162-13.socal.res.rr.com) [NETSPLIT VICTIM] |
| 16:33.26 | *** join/#asterisk Falle (n=falle@falle.se) [NETSPLIT VICTIM] |
| 16:33.26 | *** join/#asterisk tlp (i=tlp@71-33-103-74.bois.qwest.net) [NETSPLIT VICTIM] |
| 16:33.26 | *** join/#asterisk grantm (n=grantm@207.88.78.2) [NETSPLIT VICTIM] |
| 16:34.46 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-04ba434827c91001) [NETSPLIT VICTIM] |
| 16:34.46 | *** join/#asterisk tao2 (n=tao@69.63.74.254) [NETSPLIT VICTIM] |
| 16:34.46 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-42.ph.ph.cox.net) [NETSPLIT VICTIM] |
| 16:34.46 | *** join/#asterisk delphus (n=delphus@mail.delphus.org) [NETSPLIT VICTIM] |
| 16:34.46 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) [NETSPLIT VICTIM] |
| 16:34.46 | *** join/#asterisk slayer192 (n=slayer19@66.138.39.225) [NETSPLIT VICTIM] |
| 16:34.46 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) [NETSPLIT VICTIM] |
| 16:34.46 | *** join/#asterisk spr1te (n=spr1te@213.227.193.75) [NETSPLIT VICTIM] |
| 16:34.46 | *** join/#asterisk lilwookie (n=jw@modemcable219.9-70-69.static.videotron.ca) [NETSPLIT VICTIM] |
| 16:34.47 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) [NETSPLIT VICTIM] |
| 16:34.47 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) [NETSPLIT VICTIM] |
| 16:34.47 | *** join/#asterisk oliviert (n=oliviert@81.56.248.247) [NETSPLIT VICTIM] |
| 16:34.47 | *** join/#asterisk drega (n=drega@mail.amcat.co.uk) [NETSPLIT VICTIM] |
| 16:34.47 | *** join/#asterisk davixx (n=davixx@staff-davixx-portable.euroweb-network.com) [NETSPLIT VICTIM] |
| 16:34.47 | *** join/#asterisk adamowitz (n=adamowit@ip68-14-27-224.ri.ri.cox.net) [NETSPLIT VICTIM] |
| 16:34.47 | *** join/#asterisk mfroes (n=froes@200-162-218-81.corp.ajato.com.br) [NETSPLIT VICTIM] |
| 16:34.47 | *** join/#asterisk psk (n=psk@golia.caltanet.it) [NETSPLIT VICTIM] |
| 16:34.47 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) [NETSPLIT VICTIM] |
| 16:34.47 | *** join/#asterisk VibroMax (n=phisto@83.229.70.154) [NETSPLIT VICTIM] |
| 16:34.47 | *** join/#asterisk remmo (n=chatzill@smack.isp.net.au) [NETSPLIT VICTIM] |
| 16:34.47 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) [NETSPLIT VICTIM] |
| 16:34.48 | *** join/#asterisk CharlesR (n=charlesr@cpe-76-188-71-88.neo.res.rr.com) [NETSPLIT VICTIM] |
| 16:34.48 | *** join/#asterisk _Vile (n=vile@bc182112.bendcable.com) [NETSPLIT VICTIM] |
| 16:34.48 | *** join/#asterisk Druken (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) [NETSPLIT VICTIM] |
| 16:34.48 | *** join/#asterisk Qwell[] (i=qwell@nat/digium/x-f0635d971f4c61ba) [NETSPLIT VICTIM] |
| 16:34.48 | *** join/#asterisk jpablo (n=jpablo@linuxuanl.org) [NETSPLIT VICTIM] |
| 16:34.48 | *** join/#asterisk kumbalae (n=suma@cm53.omega182.maxonline.com.sg) [NETSPLIT VICTIM] |
| 16:34.48 | *** join/#asterisk synthetiq (n=syntheti@vonmail.vonworldwide.com) [NETSPLIT VICTIM] |
| 16:34.48 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-155-98-24.red.bezeqint.net) [NETSPLIT VICTIM] |
| 16:34.48 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) [NETSPLIT VICTIM] |
| 16:34.48 | *** join/#asterisk Kapsel (i=linknet@irc.thinkgeek.dk) [NETSPLIT VICTIM] |
| 16:34.48 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) [NETSPLIT VICTIM] |
| 16:34.48 | *** join/#asterisk Maxxed (i=foobar@65.59.245.122) [NETSPLIT VICTIM] |
| 16:34.49 | *** join/#asterisk jm|home (n=jamiem@dilbert.jamiem.com) [NETSPLIT VICTIM] |
| 16:34.49 | *** join/#asterisk marexz (n=marexz@195.20.127.222) [NETSPLIT VICTIM] |
| 16:34.49 | *** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) [NETSPLIT VICTIM] |
| 16:34.49 | *** join/#asterisk Winkie (n=urmom@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com) [NETSPLIT VICTIM] |
| 16:34.49 | *** join/#asterisk gmfm (i=gmfm@rtr.enterprisemtg.net) [NETSPLIT VICTIM] |
| 16:34.49 | *** join/#asterisk eivindtr (n=eivindtr@062016176152.customer.alfanett.no) [NETSPLIT VICTIM] |
| 16:34.49 | *** join/#asterisk devolish (i=ole@devolish.no) [NETSPLIT VICTIM] |
| 16:34.49 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) [NETSPLIT VICTIM] |
| 16:34.49 | *** join/#asterisk schurzi (n=schurzi@static.88-198-13-149.clients.your-server.de) [NETSPLIT VICTIM] |
| 16:34.49 | *** join/#asterisk Mw3_ (i=mw3@national.t-error.hu) [NETSPLIT VICTIM] |
| 16:34.49 | *** join/#asterisk `Sauron (i=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) [NETSPLIT VICTIM] |
| 16:34.49 | *** join/#asterisk tillo (i=tillo@85-218-17-14.dclient.lsne.ch) [NETSPLIT VICTIM] |
| 16:34.50 | *** join/#asterisk rajiv|work (n=rajiv@gentoo/developer/rajiv) [NETSPLIT VICTIM] |
| 16:34.50 | *** mode/#asterisk [+oo Cresl1n Qwell[]] by irc.freenode.net |
| 16:34.52 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) [NETSPLIT VICTIM] |
| 16:34.52 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) [NETSPLIT VICTIM] |
| 16:34.52 | *** join/#asterisk _alex_mx_ (n=alex@200.78.229.18) [NETSPLIT VICTIM] |
| 16:34.52 | *** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it) |
| 16:34.52 | *** join/#asterisk xmws (n=me@213.219.175.223.adsl.dyn.edpnet.net) [NETSPLIT VICTIM] |
| 16:34.52 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) [NETSPLIT VICTIM] |
| 16:34.52 | *** join/#asterisk FreezeS (n=Gladius@82.208.156.94) [NETSPLIT VICTIM] |
| 16:34.52 | *** join/#asterisk ambriento (n=ambrient@201-27-81-188.dsl.telesp.net.br) [NETSPLIT VICTIM] |
| 16:34.53 | *** join/#asterisk linlin (i=will@c-71-194-70-13.hsd1.il.comcast.net) [NETSPLIT VICTIM] |
| 16:34.53 | *** join/#asterisk sumasuma (n=sumasuma@61.14.86.23) [NETSPLIT VICTIM] |
| 16:34.53 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) [NETSPLIT VICTIM] |
| 16:34.53 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) [NETSPLIT VICTIM] |
| 16:34.53 | *** join/#asterisk sysreq (n=sysreq@142-217-128-85.telebecinternet.net) [NETSPLIT VICTIM] |
| 16:34.53 | *** join/#asterisk eltech (i=G00Ds@ool-457c9421.dyn.optonline.net) [NETSPLIT VICTIM] |
| 16:34.53 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) [NETSPLIT VICTIM] |
| 16:34.53 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.89) [NETSPLIT VICTIM] |
| 16:34.53 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) [NETSPLIT VICTIM] |
| 16:34.54 | *** join/#asterisk razu (n=rasmus@tln-kontor.norby.ee) [NETSPLIT VICTIM] |
| 16:34.54 | *** join/#asterisk InfraRed (n=bigboss@198.63.211.131) [NETSPLIT VICTIM] |
| 16:34.54 | *** join/#asterisk jets (n=jets@root.ownsu.com) [NETSPLIT VICTIM] |
| 16:34.54 | *** join/#asterisk ast_freak (n=jesse@h69-130-167-5.69-130.unk.tds.net) [NETSPLIT VICTIM] |
| 16:36.44 | *** join/#asterisk Falconix (n=anders@194-237-185-7.customer.telia.com) [NETSPLIT VICTIM] |
| 16:36.44 | *** join/#asterisk BrokenNoze (n=BKNNoze@host86-144-75-200.range86-144.btcentralplus.com) [NETSPLIT VICTIM] |
| 16:36.44 | *** join/#asterisk BurtyB (n=chris@alxa.camelnetwork.com) [NETSPLIT VICTIM] |
| 16:36.44 | *** join/#asterisk S^P (n=ss@203.81.196.20) [NETSPLIT VICTIM] |
| 16:36.44 | *** join/#asterisk litage (n=nick@203.220.55.70) [NETSPLIT VICTIM] |
| 16:36.44 | *** join/#asterisk enots (i=dimka@freelsd.net) [NETSPLIT VICTIM] |
| 16:36.44 | *** join/#asterisk AndyCap (n=aoy@pdpc/supporter/sustaining/AndyCap) [NETSPLIT VICTIM] |
| 16:36.44 | *** join/#asterisk Tebi_ (n=rantis@gw.aller.fi) [NETSPLIT VICTIM] |
| 16:36.44 | *** join/#asterisk roo9 (i=roo9@is-god.net) [NETSPLIT VICTIM] |
| 16:36.45 | *** join/#asterisk _polto_ (n=polto@d83-189-153-155.cust.tele2.ch) [NETSPLIT VICTIM] |
| 16:36.45 | *** join/#asterisk ScottyTM (n=ScottyTM@marlin.42h.de) [NETSPLIT VICTIM] |
| 16:36.45 | *** join/#asterisk carrar (i=tim@osburn.com) [NETSPLIT VICTIM] |
| 16:36.45 | *** join/#asterisk tessier (n=treed@kernel-panic/sex-machines) [NETSPLIT VICTIM] |
| 16:36.45 | *** join/#asterisk murf (n=steve_mu@216.166.159.235) [NETSPLIT VICTIM] |
| 16:36.45 | *** join/#asterisk h3x (n=h3xor@64.192.116.17) [NETSPLIT VICTIM] |
| 16:36.45 | *** join/#asterisk switch (n=switch@saya.attrition.jp) [NETSPLIT VICTIM] |
| 16:36.45 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) [NETSPLIT VICTIM] |
| 16:36.45 | *** join/#asterisk macTijn (i=martijn@linda.net.insecure.nl) [NETSPLIT VICTIM] |
| 16:36.45 | *** join/#asterisk Corydon76-home (i=silver@pdpc/supporter/sustaining/Corydon76-home) [NETSPLIT VICTIM] |
| 16:36.45 | *** join/#asterisk LnxPrgr3 (n=LnxPrgr3@br0.asteriasgi.com) [NETSPLIT VICTIM] |
| 16:36.45 | *** join/#asterisk So3kris (n=jan-will@ids.netland.nl) [NETSPLIT VICTIM] |
| 16:36.46 | *** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM] |
| 16:36.46 | *** join/#asterisk joe (n=nnnjsaue@66.107.33.195) [NETSPLIT VICTIM] |
| 16:36.46 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) [NETSPLIT VICTIM] |
| 16:36.46 | *** join/#asterisk infinity1 (i=foobar@208.184.76.100) [NETSPLIT VICTIM] |
| 16:36.46 | *** join/#asterisk tzanger (n=tzanger@mixdown.ca) [NETSPLIT VICTIM] |
| 16:36.46 | *** mode/#asterisk [+oo anthm Corydon76-home] by irc.freenode.net |
| 16:36.57 | zx6r | weee |
| 16:36.59 | *** join/#asterisk denon (i=denon@synapse.subneural.net) [NETSPLIT VICTIM] |
| 16:36.59 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) [NETSPLIT VICTIM] |
| 16:36.59 | *** join/#asterisk Juggie (n=Juggie@wiley-459-29776.roadrunner.nf.net) [NETSPLIT VICTIM] |
| 16:36.59 | *** join/#asterisk djflux (n=djflux@mm.shermfin.com) [NETSPLIT VICTIM] |
| 16:36.59 | *** join/#asterisk MrChimpy (n=MrChimpy@212.158.8.162) [NETSPLIT VICTIM] |
| 16:36.59 | *** join/#asterisk plonky (i=dom@80.82.16.205) [NETSPLIT VICTIM] |
| 16:37.00 | *** join/#asterisk Cinen (n=Cinen@208.70.20.33) [NETSPLIT VICTIM] |
| 16:37.00 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) [NETSPLIT VICTIM] |
| 16:37.00 | *** join/#asterisk puddingpimp_work (n=puddingp@gateway.quickcircuit.co.nz) [NETSPLIT VICTIM] |
| 16:37.00 | *** join/#asterisk glitchz (i=1045@unaffiliated/glitchz) [NETSPLIT VICTIM] |
| 16:37.00 | *** join/#asterisk X-Gen (n=X-Gen@dsl-242-22-228.telkomadsl.co.za) [NETSPLIT VICTIM] |
| 16:37.00 | *** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net) [NETSPLIT VICTIM] |
| 16:37.00 | *** join/#asterisk awannabe (n=gti@ip24-251-135-202.ph.ph.cox.net) [NETSPLIT VICTIM] |
| 16:37.00 | *** join/#asterisk bXi (i=bluepunk@irssi.co.uk) [NETSPLIT VICTIM] |
| 16:37.00 | *** join/#asterisk ApEtc (i=apetc@ip70-162-197-214.ph.ph.cox.net) [NETSPLIT VICTIM] |
| 16:37.00 | *** join/#asterisk delmar (n=delmar@ip-58-28-158-154.ubs-dsl.xnet.co.nz) [NETSPLIT VICTIM] |
| 16:37.00 | *** join/#asterisk monsted (n=monsted@rootweiler.dk) [NETSPLIT VICTIM] |
| 16:37.01 | *** join/#asterisk Barnes (n=barnes@c193-150-214-238.bredband.comhem.se) [NETSPLIT VICTIM] |
| 16:37.01 | *** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de) [NETSPLIT VICTIM] |
| 16:37.01 | *** join/#asterisk SkymeyeR (n=SkymeyeR@d5152F111.access.telenet.be) [NETSPLIT VICTIM] |
| 16:37.01 | *** join/#asterisk JT (n=jon@unaffiliated/jt) [NETSPLIT VICTIM] |
| 16:37.01 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) [NETSPLIT VICTIM] |
| 16:37.01 | *** join/#asterisk Champi (i=Champi@damn.e-leet.be) [NETSPLIT VICTIM] |
| 16:37.01 | *** join/#asterisk mcab (n=mb@66.195.193.151) [NETSPLIT VICTIM] |
| 16:37.01 | *** join/#asterisk Corydon-w (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) [NETSPLIT VICTIM] |
| 16:37.01 | *** join/#asterisk dusan2 (i=dusan@209-223-47-160-static.oplink.net) [NETSPLIT VICTIM] |
| 16:37.01 | *** join/#asterisk X-Rob (n=Rob@dsl-202-173-151-24.qld.westnet.com.au) [NETSPLIT VICTIM] |
| 16:37.02 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) [NETSPLIT VICTIM] |
| 16:37.02 | *** mode/#asterisk [+oo denon Corydon-w] by irc.freenode.net |
| 16:37.02 | vader-- | mercestes ya i tried a 5 second wait and that didn't work |
| 16:37.02 | vader-- | after about 4 tries |
| 16:37.02 | vader-- | i get a verizon message saing call can't be completed as dialed |
| 16:37.02 | vader-- | i wonder if verizon just freaks out about being called soo fast |
| 16:37.04 | zx6r | ah, the question: i'd like to give users the ability to select their hold music, once they've been put on hold. any idea of the best way to do this? i'm messing with the applicationmap of features.conf but i don't think it'll work completely... |
| 16:37.05 | *** join/#asterisk techman97_andy (n=me@74-38-83-212.dr01.rsmt.mn.frontiernet.net) |
| 16:37.58 | techman97_andy | hey all - I'm having a terrible problem. My users are being "locked" out of voicemail. People can leave messages, but they never get saved. How I'm working around it now is every morning, I'm SSH'ing in and deleting large amounts of .lockxxxxxx files out of each user's /voicemail/default/<ext>/INBOX folder. ANy ideas on what is causing this?? |
| 16:37.58 | Qwell[] | techman97_andy: It's the .lockxxx and not .lock? |
| 16:37.58 | Qwell[] | techman97_andy: What version of *? |
| 16:37.59 | techman97_andy | it's both actually |
| 16:37.59 | techman97_andy | I have to delete them all for VM to work again for that user |
| 16:37.59 | Qwell[] | That's...umm...funky |
| 16:37.59 | techman97_andy | yeah, tell me about it |
| 16:37.59 | Qwell[] | what version? |
| 16:37.59 | mercestes | techman97_andy: Check your permissions. |
| 16:38.00 | techman97_andy | Asterisk 1.2.9.1 |
| 16:38.00 | techman97_andy | check them how and to what? |
| 16:38.00 | mercestes | techman97_andy: well, asterisk is running as what user/group?? |
| 16:38.00 | Qwell[] | He wouldn't have been able to create the lock files in the first place, if it were permissions |
| 16:38.02 | techman97_andy | well, I could possibly have create only permissions, but not modify? |
| 16:38.37 | techman97_andy | I dunno |
| 16:38.37 | Qwell[] | nah |
| 16:38.37 | Qwell[] | unlikely |
| 16:38.38 | mercestes | yea, not in *nix. |
| 16:39.14 | Qwell[] | especially if it works "sometimes" |
| 16:39.14 | techman97_andy | well, it always works until these files start piling up |
| 16:39.14 | techman97_andy | as soon as I delete them, VM works again |
| 16:39.36 | zx6r | maybe it's not getting a hangup signal, and just keeps recording? do the files keep getting bigger? |
| 16:39.36 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 16:39.36 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:39.58 | techman97_andy | nope |
| 16:39.58 | techman97_andy | files stay the same size |
| 16:40.38 | Qwell[] | mog: techman97_andy here is having a VM locking issue... |
| 16:40.38 | Qwell[] | techman97_andy: How often would you say it happens? |
| 16:41.03 | techman97_andy | hmmm |
| 16:41.03 | techman97_andy | I'm thinking about that... |
| 16:41.20 | techman97_andy | you know, I have one user that gets the MOST voicemail...and he has never had an issue. |
| 16:41.20 | techman97_andy | the extensions that are affected are on a ring group, and a call queue. |
| 16:41.27 | mog | oh awesome |
| 16:41.31 | mog | can we work on your box? |
| 16:41.34 | Qwell[] | mog: but, get this |
| 16:41.35 | mog | how often does it happen |
| 16:41.42 | Qwell[] | mog: the .lock-random files are there also |
| 16:41.46 | techman97_andy | yeah |
| 16:42.01 | Qwell[] | That's the part that threw me |
| 16:42.19 | mog | that happens too if i recall |
| 16:42.21 | Qwell[] | ok |
| 16:42.22 | techman97_andy | I'm trying a test quick |
| 16:42.45 | techman97_andy | I'm calling our number and am going to go the "no answer -> leave message for this affected extension" option. |
| 16:43.06 | Qwell[] | heh |
| 16:43.11 | Qwell[] | I could totally see Allison... |
| 16:43.29 | Qwell[] | "Please leave a message for this user, who may not ever get it." |
| 16:44.32 | techman97_andy | heheheh |
| 16:44.36 | techman97_andy | ugh |
| 16:44.41 | techman97_andy | forgot the timeout was 5 minutes - |
| 16:45.11 | *** join/#asterisk ikey (n=ikey@125.22.106.119) |
| 16:45.21 | *** join/#asterisk gerphimum (n=trekkie@cpe-68-206-83-62.satx.res.rr.com) |
| 16:46.30 | *** join/#asterisk fromm (n=fromm@fromm.omnis.com) |
| 16:47.08 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 16:47.35 | techman97_andy | well crap |
| 16:47.38 | *** join/#asterisk adorah (n=admin@87.68.159.96.cable.012.net.il) |
| 16:47.41 | techman97_andy | that didn't pan out the way I was expecting |
| 16:48.16 | *** join/#asterisk Qain_SK (n=qain@adsl-072-149-043-253.sip.bct.bellsouth.net) |
| 16:48.27 | techman97_andy | but I did verify that the extensions being affected are static agents in a queue, and in a ring group - with the "timeout" option going to the "Core: Voicemail Box <ext>" |
| 16:48.43 | techman97_andy | I wonder if instead of hitting the mailbox, I send it to the extension natively. |
| 16:49.54 | tao2 | what you you recomend for using connecting skype to my trixbox ? |
| 16:50.13 | Nivex | *sigh* |
| 16:50.18 | adorah | hehe |
| 16:50.20 | techman97_andy | hehehehhe |
| 16:50.21 | Nivex | SKYPE IS PROPRIETARY! |
| 16:50.48 | adorah | well one can still use uplink to connect..if one insists |
| 16:50.51 | tao2 | i dont really care what i use then just need a good place to hope out in South africa |
| 16:50.51 | techman97_andy | yeah, there is indeed a Skype bridge-thing available out there for Asterisk, but it involves another machine completely seperate from Asterisk that acts as a "proxy" |
| 16:50.53 | techman97_andy | it's terrible |
| 16:51.01 | Qwell[] | tao2: how about using...skype? |
| 16:51.18 | Nivex | try gizmoproject.com. It's SIP based and has better interop. |
| 16:52.37 | tao2 | this is most prob a dum q but is there a code that i can dail if i hear a ext rining an i would like to pick it up from my phone ? |
| 16:53.11 | Qain_SK | *8# as long as you are in the call group |
| 16:53.53 | tao2 | thanks |
| 16:54.29 | Qain_SK | tao2: it's in the zapata.conf sample file |
| 16:55.07 | *** join/#asterisk toxap (n=toxap@213.227.193.75) |
| 16:55.08 | tao2 | it gives me error 503 |
| 16:55.17 | toxap | Hello |
| 16:56.32 | *** join/#asterisk natmlt (n=natmlt@ip68-109-184-159.ph.ph.cox.net) |
| 16:57.09 | toxap | How can I do it is busy, what in Q.931 there was an error 17? |
| 16:57.19 | Qain_SK | I am having an interesting issue with my TDM400P cards, and having an incompatibility with some local PSTN equipment. Something to do with a pair gain system and the newer eq my phones work fine, with the older eq my TDM picks up the phone, dials, and hangsup immediately upon connecting to my 7912G SIP enabled IP phones. anyone have any thoughts. |
| 16:57.23 | toxap | I have a 21 |
| 16:58.36 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 16:58.46 | CrashHD | where can I find better documentation on group()? |
| 16:58.48 | *** join/#asterisk Tili (n=tili@202.133.65.61) |
| 16:58.48 | CrashHD | or some examples? |
| 16:59.17 | Tili | is there any callback i can register in asterisk to get channel status information |
| 16:59.20 | Tili | in C modules |
| 16:59.21 | Qain_SK | crashHD: where have you looked?? |
| 16:59.36 | CrashHD | voip-info |
| 16:59.44 | CrashHD | google with "asterisk group()" |
| 17:00.19 | Qain_SK | crashHD: do you have "Asterisk The Future of Telephony" |
| 17:00.29 | CrashHD | I've read it |
| 17:00.48 | fromm | anyone able to get the current config_res_ldap branch from svn to compile? |
| 17:01.18 | *** join/#asterisk anthonyl (i=anthonyl@nat/digium/x-cd36e14dae739bec) |
| 17:01.21 | *** part/#asterisk Explisit (n=explisit@85.187.160.57) |
| 17:01.39 | mercestes | exactly what does "group" do supposedly? |
| 17:03.13 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 17:04.05 | *** join/#asterisk fr0z3n (n=email@CPE0016b64d8af4-CM0011aec7a550.cpe.net.cable.rogers.com) |
| 17:05.36 | *** join/#asterisk sb_mx (n=sb_mx@dsl-200-67-149-101.prod-empresarial.com.mx) |
| 17:06.36 | *** join/#asterisk ManxPower (n=manxpowe@71-8-11-73.dhcp.leds.al.charter.com) |
| 17:13.04 | *** join/#asterisk fr0z3n (n=email@CPE0016b64d8af4-CM0011aec7a550.cpe.net.cable.rogers.com) |
| 17:13.11 | kumbalae | why the rejected calls are not in the CDR ? |
| 17:14.41 | tao2 | where do i enable call waiting |
| 17:15.10 | *** join/#asterisk sb_mx (n=sb_mx@dsl-200-67-149-101.prod-empresarial.com.mx) |
| 17:15.32 | *** join/#asterisk saftsack (n=saftsack@pD9E052F7.dip.t-dialin.net) |
| 17:15.40 | saftsack | hi |
| 17:15.46 | tao2 | hi |
| 17:16.15 | Qain_SK | tao2: did you actually read any of the sample files?? |
| 17:16.16 | saftsack | which port do i need that asterisk works? i opened 10000 - 20000 udp and 5060 - 5070 tcp/udp but there is no seech in calls |
| 17:17.42 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 17:19.14 | ManxPower | saftsack: is Asterisk behind NAT or just the phone behind NAT? |
| 17:19.25 | saftsack | asterisk is behind NAT |
| 17:19.31 | vader-- | shellshark/mercestes the dialplan doesn't work against the radio station they must not send a busy signal they must just play a sound or something |
| 17:20.24 | ManxPower | saftsack: then you need to forward ports 5060/UDP and whatever ports you list in rtp.conf. If you don't have an rtp.conf then forward ports 10000-20000/UDP. Then set localnet= and externip= in sip.conf [general] |
| 17:21.03 | Qain_SK | is there a more specific Zaptel Channel?? |
| 17:21.14 | ManxPower | Qain_SK: not really. |
| 17:21.20 | ManxPower | What is your specific question? |
| 17:21.30 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 17:22.27 | saftsack | ManxPower, can externip be a domainname? |
| 17:22.35 | kumbalae | ManxPower: I enabled mysql cdr, but i also get duplicates in the Master.csv and also in the accountcode.csv file, will disabling cdr.conf will avoid duplication ? |
| 17:22.38 | Qain_SK | ManxPower: I have an issue with the connection between my TDM400P and my PSTN. |
| 17:23.22 | Qain_SK | I have one line sitting on a PairGain 35 system which uses an older piece of equipment. and 4 lines connected to a PairGain 45 system |
| 17:23.36 | Qain_SK | which uses some newer equipment. |
| 17:23.46 | Qain_SK | this is all on the ServiceProviders side. |
| 17:24.13 | Qain_SK | when I try and make an outgoing call on that line that is on the 35, it picks up the phone, dials, and hangsup immediately |
| 17:24.24 | CrashHD | so nobody can point me to some examples of group() usage? or documentation? |
| 17:25.41 | Qain_SK | CrashHD: i'm still confused as to where you are trying to use it. |
| 17:26.04 | *** join/#asterisk dasenjo (n=dasenjo@201.228.128.10) |
| 17:26.07 | *** join/#asterisk barros (n=barros@zaphod.inexo.com.br) |
| 17:27.21 | barros | Hi.. i've upgraded asterisk to version 1.2.13 and noticed a strange behavior in the CDR log... |
| 17:27.27 | Qain_SK | ManxPower: I had one of my lines on PG44, and they moved it to PG45 and it works fine. The Bellsouth Guys thought it may have to do with the line voltage and it not being recognized when making outgoing calls. |
| 17:27.40 | ManxPower | saftsack: no, externip must be an ip. I believe there is some external hostname option, but I don't recall the specific option name. |
| 17:27.51 | jeremy_g | hi, show app progress says it will report inband progress of the call to the calling channel, what the hell does that supposed to mean? |
| 17:28.04 | jeremy_g | reporting progress |
| 17:28.08 | ManxPower | Qain_SK: put the CLI output of a failed call on pastebin.ca |
| 17:28.09 | barros | for outgoing Zap calls, src fields used to show whith SIP started to call, now it shows the Caller ID, that is, the external number that was used in the call |
| 17:28.10 | jeremy_g | while a call is being made |
| 17:28.58 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-162.sw.biz.rr.com) |
| 17:29.09 | ManxPower | jeremy_g: "progress" usually means "report busy, congestion, answer, etc". |
| 17:29.20 | ManxPower | It never works right anyway |
| 17:29.34 | Qain_SK | ManxPower: there really isn't any output, looks like the phone call is made and hung up. |
| 17:29.46 | ManxPower | Qain_SK: analog or digital ports. For the most part I don't care about what is on the far end, just what type of connection you have to it. |
| 17:29.54 | *** join/#asterisk andresmujica (n=andresmu@201.244.244.253) |
| 17:29.59 | Qain_SK | ManxPower: analog |
| 17:30.00 | ctooley | Has anyone implemented any of the VoiceXML engines on top of Asterisk? |
| 17:30.05 | andresmujica | hello there! |
| 17:30.11 | ManxPower | Qain_SK: paste your Dial line |
| 17:30.13 | barros | all this change seems to be ok, but I noticed some "WRONG" entried in the CDR files |
| 17:30.45 | jeremy_g | ManxPower:but thats what asterisk already does by default, u dial a damn user and it returns with the status in $DIALSTATUS <-- is that call progress |
| 17:30.50 | barros | like, showing the number I called in the src and clid fields.. that is completely wrong in my point of view.. and worst, those numbers are wrong too.. |
| 17:30.52 | andresmujica | i need to change the language for an perl AGI script.. anyone knows which is the parameter fpr perl agi? |
| 17:30.57 | barros | anyone noticed something similar?? |
| 17:31.06 | ManxPower | jeremy_g: no, it does not. Not for analog FXO ports it doesn't. |
| 17:31.23 | ManxPower | barros: You didn't see a similar issue talked about on the mailinglists? |
| 17:31.41 | jeremy_g | ManxPower:so if its only used for sip, then it already does report progress right? |
| 17:31.43 | barros | ManxPower, hmm.. didnt looked that.. :D |
| 17:31.49 | CrashHD | what is the difference between << >> and []? |
| 17:32.22 | Qain_SK | ManxPower: exten => _9NXXXXXX,1,Dial(${OUTBOUNDTRUNKBH}/ww${EXTEN:1}) This? |
| 17:32.25 | ManxPower | CrashHD: in what context |
| 17:32.51 | ManxPower | Qain_SK: and that is why I need the CLI output to see what the substitutions end up being. |
| 17:32.52 | Qain_SK | ok |
| 17:32.59 | CrashHD | it may just be documentation purposes but I was looking at group examples from http://www.voip-info.org/wiki/index.php?page=Asterisk+func+GROUP |
| 17:33.01 | CrashHD | in the user comments |
| 17:33.06 | ManxPower | Qain_SK: work with me. I've been doing this for a while. |
| 17:33.16 | CrashHD | I have never seen << >> used |
| 17:34.05 | ManxPower | CrashHD: if you look at the comment he says he can't figure out how to make the wiki accept [ ] in his text |
| 17:34.12 | CrashHD | oh |
| 17:34.13 | CrashHD | I'm sorry |
| 17:34.38 | CrashHD | ManxPower: you wouldn't happen to have any other suggestions on group() documentation or examples? |
| 17:34.55 | CrashHD | s/suggestions on/suggestions on where I can look for |
| 17:35.01 | ManxPower | CrashHD: not really. In my experience I have never needed to use Group(). |
| 17:35.12 | CrashHD | ok thank you |
| 17:35.21 | ManxPower | When I tried to use Group() I ended up realizing that there were far easier ways to do what I need. |
| 17:35.39 | fr0z3n | is there anyone here who can help me install the sangoma A200 card on redhat? i have followed there instructions but it is unable to probe the card |
| 17:35.42 | CrashHD | aww |
| 17:35.54 | CrashHD | well group is really powerful in my opinion, just want to know more about how to use it |
| 17:36.31 | ManxPower | CrashHD: Group() works like a lock or semaphore in programming languages. |
| 17:36.43 | ManxPower | I just have never needed to lock something in my dialplan. |
| 17:37.04 | barros | ManxPower, can u point me the maillist archive? |
| 17:37.10 | ManxPower | ~mailinglist |
| 17:37.24 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 17:37.24 | CrashHD | I really just can't figure out what the difference between group() and group(something) is |
| 17:37.24 | CrashHD | lame eh? |
| 17:37.43 | CrashHD | I'll try the archives |
| 17:37.48 | ManxPower | CrashHD: group(something) just increments the usage count for "something" |
| 17:37.50 | *** join/#asterisk SplasPood (n=jwb@nat-loopback.lga4.us.corp.voxel.net) |
| 17:37.57 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool3-10.mtco.com) |
| 17:38.03 | *** part/#asterisk ctooley (n=ctooley@rrcs-24-227-212-162.sw.biz.rr.com) |
| 17:38.04 | CrashHD | so when using just group() |
| 17:38.08 | CrashHD | what is it incrementing? |
| 17:38.23 | ManxPower | It is incrementing the usage count" |
| 17:38.25 | barros | ManxPower, thanks |
| 17:38.40 | CrashHD | and if it is just doing counts, what is the reason you set it to something [eg. set(group()=something)] |
| 17:38.47 | fr0z3n | can anyone here help me with sangoma a200 installation? |
| 17:38.55 | ManxPower | CrashHD: what if you want to count more than one thing |
| 17:39.19 | CrashHD | but what is the point of setting it to something? |
| 17:39.25 | ManxPower | remember this is a GLOBAL usage count. |
| 17:39.32 | *** join/#asterisk haryv (n=sales@S010600a0c93f6f7e.vs.shawcable.net) |
| 17:39.34 | Qain_SK | ManxPower: http://pastebin.ca/245159 |
| 17:39.36 | *** join/#asterisk anthonyl (i=anthonyl@nat/digium/x-c28a9232825c34a0) |
| 17:39.37 | ManxPower | CrashHD: something is just a label |
| 17:40.04 | ManxPower | Qain_SK: did you manually replace the real numbers with N? |
| 17:40.09 | Qain_SK | yes |
| 17:40.09 | CrashHD | guess I don't know what label means? |
| 17:40.23 | CrashHD | I thought the label was group(label) |
| 17:40.48 | ManxPower | CrashHD: Lets say that you want to make sure that no more than 4 calls go to extension 1234 and you don't want more than 2 calls go to extension 6666 |
| 17:40.50 | Qain_SK | ManxPower: they are formed exactly as they look on any of the other outbound calls. |
| 17:41.07 | CrashHD | good scenario |
| 17:41.27 | ManxPower | You could use group(1234) and group(6666) so when there are mulpiple calls happening at the same time Asterisk can figure out what group name to imcrement |
| 17:41.55 | andresmujica | how can oi pass the agi_language variable to an agi script just beforlaunching it |
| 17:42.09 | ManxPower | Qain_SK: Good. Do you have any busydetect= or callprogress= lines in /etc/asterisk/zapata.confd |
| 17:42.20 | CrashHD | but the syntax I saw was set(group(1234)=something) |
| 17:42.29 | CrashHD | something isn't being stored in a var space? |
| 17:42.34 | ManxPower | CrashHD: I would have to look it up. |
| 17:42.39 | Qwell[] | It's a function |
| 17:43.17 | CrashHD | hmm |
| 17:43.24 | CrashHD | look it up where? short of looking at the source? |
| 17:43.25 | CrashHD | heh |
| 17:43.37 | CrashHD | just point me |
| 17:44.28 | Qwell[] | show functions like group? |
| 17:45.15 | CrashHD | I guess I don't understand the name vs category |
| 17:46.04 | Qain_SK | ManxPower: Tried with and without same problem |
| 17:46.40 | ManxPower | Qain_SK: you need them not set at all or set to no |
| 17:47.02 | Qain_SK | ManxPower: alright. |
| 17:47.14 | ManxPower | Qain_SK: now, in the priority after Dial put Noop(DIALSTATUS=${DIALSTATUS} and HANGUPCAUSE=${HANGUPCAUSE}) |
| 17:47.24 | ManxPower | try the call again and put the CLI result on pastebin.ca |
| 17:48.29 | *** join/#asterisk cloud9 (n=nothing@smtp.a-cti.com) |
| 17:49.09 | ManxPower | Qain_SK: Oh, and what model of analog card are you using? |
| 17:50.01 | ManxPower | Qain_SK: also, what country are you in? |
| 17:50.28 | cloud9 | hello, I've searched google extensively to solve this problem, with no resolution. I thout I'd check here.. I've got a basic asterisk install. I've got some extensions setup, and some SIP phones logged in. The SIP phones seem to timeout after about 5 minutes. Anyone seen this problem? |
| 17:50.39 | Qwell[] | timeout how? |
| 17:51.18 | Qain_SK | ManxPower: USA TDM400P |
| 17:51.19 | cloud9 | well for example, when the phone originaly logs in, I can call it's extension. but after about 5 minutes.. if I try to call the extension, asterisk output says it's calling, but the phone never rigns |
| 17:51.47 | ManxPower | cloud9: your phones are behind nat |
| 17:52.17 | cloud9 | ManxPower : hmm, yes they are. |
| 17:52.23 | fromm | anyone had any luck with res_config_ldap? |
| 17:52.28 | ManxPower | cloud9: then you need to deal with NAT issues |
| 17:52.37 | cloud9 | ManxPower : I added nat=yes is sip.conf |
| 17:52.55 | ManxPower | cloud9: the easiest and most reliable way to fix this is to configure the phones to register every 60 seconds |
| 17:53.06 | *** join/#asterisk ellisdee (n=ellisdee@69.15.174.114) |
| 17:53.11 | cloud9 | ManxPower : cool, thanks bud |
| 17:53.17 | ManxPower | because the NAT routers that the phones are behind are closing the NAT translation because it sees no traffic on that port. |
| 17:53.24 | ellisdee | anyone a fancy utility that modifies echo cancellation, and does a test for you? =P |
| 17:53.35 | Qwell[] | What, no qualify? |
| 17:53.37 | cloud9 | ManxPower : thanks for the brain! :-) |
| 17:53.40 | jeremy_g | I receive a call in an extension, I dial another user, i get a response like 404 not found, how do i send it back to the call that landed in my extension |
| 17:53.44 | Qwell[] | Doesn't that keep the translation tables open? |
| 17:54.00 | ManxPower | Qwell[]: qualify causes all sorts of bogus lagged and unreachable things |
| 17:54.03 | *** part/#asterisk cloud9 (n=nothing@smtp.a-cti.com) |
| 17:54.09 | Qain_SK | ManxPower: are the ${DIALSTATUS} & ${HANGUPCAUSE} CLI set variables, or do I have to set them? |
| 17:54.19 | ManxPower | Qain_SK: they are set by the Dial command |
| 17:54.23 | Qwell[] | true |
| 17:54.25 | Qain_SK | ok |
| 17:54.44 | ManxPower | they will tell us more about the cause of the failure |
| 17:55.03 | Qain_SK | ManxPower: Nov 10 12:51:59 WARNING[11915]: app_dial.c:1191 dial_exec_full: Invalid timeout specified: 'Noop(DIALSTATUS=|HANGUPCAUSE=0' |
| 17:56.34 | ManxPower | Qain_SK: it needs to be the NEXT line. |
| 17:56.51 | Qain_SK | gotcha |
| 17:56.52 | ManxPower | and next priority. |
| 17:57.00 | Qain_SK | ok |
| 17:57.07 | ManxPower | Dial exits, the dialplan will continue and execute the noop |
| 17:57.39 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 17:57.52 | Qain_SK | use the "and" or the "&" |
| 17:58.15 | ManxPower | AND |
| 17:58.41 | jeremy_g | is there an appliaction that sends the SIP response 404 not found |
| 17:58.43 | *** join/#asterisk file (n=file@neutrino.joshua-colp.com) |
| 17:58.43 | *** mode/#asterisk [+o file] by ChanServ |
| 17:58.43 | Qain_SK | gotcha |
| 17:58.58 | jeremy_g | NotFound() ...sends not found back to the user |
| 17:59.07 | jeremy_g | is there an appliaction that sends the SIP response 404 not found |
| 17:59.09 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 17:59.18 | ManxPower | jeremy_g: no there isn't. |
| 17:59.35 | ManxPower | jeremy_g: 404 means there were no extension matches for the context the call was sent to. |
| 17:59.47 | jeremy_g | i know what it means |
| 18:01.08 | fr0z3n | sangoma totally sucks! arghhh!! |
| 18:01.15 | jeremy_g | scenario is that i receive a call in an extension, dial another number, and want to report to caller what response i received. basically routing the call but also be able to route the response back |
| 18:01.20 | haryv | ManxPower ever provided consulting services and never been paid? |
| 18:01.43 | jeremy_g | shut up haryv, u make me remind of bad times |
| 18:01.53 | jeremy_g | remove make me ;( |
| 18:02.08 | ManxPower | haryv: every day on here. |
| 18:02.22 | ManxPower | haryv: or do you mean not paid when I expected to be paid? Not really. |
| 18:02.23 | haryv | I was not paid recently we agreed on a price he was deperate :) |
| 18:02.42 | ManxPower | I don't accept desperate clients. |
| 18:02.53 | ManxPower | at least poor desperate clients. |
| 18:02.59 | haryv | yea thats probebly true. |
| 18:03.24 | haryv | I am thinking of creating a html form where thay can fill out before I render my services. |
| 18:03.27 | ManxPower | I have 3 semi-regular clients. Client 1: consulting them for 10 years, Clienr 2: 15 years, Client 3: 2 years |
| 18:03.32 | Qain_SK | ManxPower: only difference is it moves to the congestion and gives a busy signal no output from Noop unless it is stored somewhere else. |
| 18:03.51 | ManxPower | Qain_SK: put that part of your dialplan on pastebin.ca |
| 18:03.57 | Qain_SK | ok |
| 18:04.23 | *** join/#asterisk timscott (n=tim@S01060016b6e5e09b.ed.shawcable.net) |
| 18:05.24 | Qain_SK | ManxPower: http://pastebin.ca/245175 |
| 18:06.31 | *** join/#asterisk henrique (n=henrique@200-153-196-111.dsl.telesp.net.br) |
| 18:07.07 | ManxPower | Qain_SK: remove the 103 priority that will make it just continue to the next priority. Also priority jumping (like 103) is not recommended as of 1.2 |
| 18:07.32 | Qain_SK | good to know. |
| 18:08.28 | ManxPower | Qain_SK: the correct way is check the value of DIALSTATUS or HANGUPCAUSE and determine what to do based on those values. See the std-exten-macro in extensions.conf.sample in the Asterisk source tree. |
| 18:08.54 | Qain_SK | ok |
| 18:08.58 | ManxPower | but at this point I need to know what the values of those are after the dial. |
| 18:09.04 | Qain_SK | ok |
| 18:09.12 | *** join/#asterisk rene1 (n=rene1@189.140.234.203) |
| 18:09.37 | rene1 | hello |
| 18:10.34 | rene1 | in zap and unicall channels there is the possibility to play audio before answering the channels, and therefore avoiding call establishment costs, IIRC there is something called Early audio in SIP? |
| 18:10.55 | ManxPower | rene1: in zap if you have a PRI you can. |
| 18:11.14 | ManxPower | "show application playback" pay special attention to the "noanswer" option |
| 18:12.00 | rene1 | ManxPower: i have done it in PRI, i wonder if it was possible to do it in SIP trunks |
| 18:12.32 | *** join/#asterisk evilstoy (i=HydraIRC@83-131-20-186.adsl.net.t-com.hr) |
| 18:12.32 | ManxPower | rene1: you should be able to do it with SIP, but that would also depend on the actual SIP device |
| 18:12.52 | rene1 | so it exists but i may / may not be implemented |
| 18:13.09 | ManxPower | correct. BTW, there is no such thing as a "SIP trunk" |
| 18:13.40 | rene1 | ManxPower: do you know if it is possible for the caller to send DTMF if the call has not been established/answered ? |
| 18:13.50 | ManxPower | rene1: no you cannot. |
| 18:13.58 | rene1 | i see |
| 18:14.01 | ManxPower | that would be "two way audio" and the telco won't allow that |
| 18:14.09 | rene1 | yeah you are right |
| 18:16.13 | Qain_SK | ManxPower: http://pastebin.ca/245183 |
| 18:18.18 | rene1 | ManxPower: do you think a P4/3.4 2G RAM box should be able to fill a 30 channel pri (E1) while recording all channels. SIP extensions and no transcoding? i am seeing some dropped calls and chopped audio at 18~ calls, the guy before me used to use @home and that configuration only yielded 12 calls with decent quality |
| 18:19.45 | ManxPower | Qain_SK: CHANUNAVAIL means Asterisk thinks the line is in use. |
| 18:19.53 | ManxPower | or is not configured. |
| 18:20.03 | ManxPower | rene1: I don't know. |
| 18:20.13 | *** join/#asterisk svenna (n=svenna@p548D3C2F.dip0.t-ipconnect.de) |
| 18:20.16 | ManxPower | Qain_SK: pastebin the output of "ztcfg -vvv" |
| 18:20.19 | *** join/#asterisk spr1te (n=spr1te@213.227.193.75) |
| 18:20.24 | rene1 | those things are only trial and error i guess |
| 18:20.37 | ManxPower | also the ourput of "zap show channels" and "show channels" |
| 18:20.51 | Qain_SK | ok |
| 18:21.42 | *** join/#asterisk implicit (n=implicit@lgb-static-66.18.140.106.mpowercom.net) |
| 18:21.52 | tao2 | ok i am going made i have looked but what file do i edit to set up call waiting for internal transfers |
| 18:21.53 | jeremy_g | ManxPower:what if i set exten 101,1 => Progress () 2. Dial (sip/somewhere) 3 |
| 18:22.08 | ManxPower | jeremy_g: I don't know. |
| 18:22.27 | jeremy_g | ManxPower:cant i use Progress() to route back whatever response i received after dialling somewhere back to the dude who called 101 |
| 18:22.52 | jeremy_g | ManxPower:what is Progress() there for? |
| 18:23.07 | Qain_SK | ManxPower: http://pastebin.ca/245190 |
| 18:23.20 | ManxPower | jeremy_g: I have no idea. I've never used it |
| 18:23.49 | ManxPower | Qain_SK: you have two TDM400P cards with all FXO ports on them? |
| 18:24.01 | Qain_SK | yes sir |
| 18:24.15 | ManxPower | Qain_SK: and your have the ports plugged into FXS lines on the PBX? |
| 18:24.23 | *** join/#asterisk Holos (n=asdf@204.101.26.106) |
| 18:24.46 | *** join/#asterisk hohum (n=dcorbe@jomama.interceltelecoms.net) |
| 18:25.14 | Qain_SK | ManxPower: Straight from the POTS lines |
| 18:25.36 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 18:25.36 | ManxPower | Qain_SK: POTS lines from the telco |
| 18:25.38 | ManxPower | ? |
| 18:25.40 | Qain_SK | yes |
| 18:26.11 | Qain_SK | ManxPower: I have 4 lines working just one that is not, tried all sorts of wire configurations too. |
| 18:26.50 | ManxPower | Qain_SK: do you have a standard analog phone available? |
| 18:27.00 | Qain_SK | yup.. works fine |
| 18:27.04 | Qain_SK | no problems at all |
| 18:27.09 | ManxPower | standby |
| 18:27.44 | ManxPower | Qain_SK: just for the hell of it put the output of "cat /proc/interrupts" on pastebin.ca |
| 18:27.46 | *** join/#asterisk CharlesR (n=charlesr@adsl-75-24-18-2.dsl.yntwoh.sbcglobal.net) |
| 18:27.52 | Qain_SK | alright. |
| 18:28.00 | ellisdee | ManxPower, heh. you have alot of time on your hands. you getting paid for all of this? |
| 18:28.15 | ellisdee | what kernel you running Qain_SK? |
| 18:28.46 | Qain_SK | ellisdee: Linux version 2.4.33.3 (root@tree) (gcc version 3.4.6) #1 Fri Sep 1 01:48:52 CDT 2006 |
| 18:28.51 | ellisdee | aah, oaky. |
| 18:28.52 | ellisdee | okay |
| 18:29.07 | ellisdee | do a ls -al on /proc/zap |
| 18:29.17 | ellisdee | pm me what you got in there. |
| 18:29.24 | ManxPower | ellisdee: If I got paid for this I would have to be nice to everyone, even the ones that that are idiots. |
| 18:29.35 | ellisdee | ManxPower, lol, i dont have time to help anyone =P |
| 18:29.57 | tao2 | please please i am going mad i have looked but what file do i edit to set up call waiting for internal transfers |
| 18:30.17 | ManxPower | tao2: /etc/asterisk/zapata.conf |
| 18:30.26 | ManxPower | for transfers on analog ports. |
| 18:30.40 | tao2 | i need it for sip ext |
| 18:30.50 | ellisdee | Qain_SK, which one of those pastebin urls is for the zapata.conf? |
| 18:30.58 | ManxPower | tao2: that would be TOTALLY up to the SIP device. All that stuff is handled on the actual SIP device. |
| 18:31.11 | ellisdee | Qain_SK, you have 4 pots lines going into one zapcard. what is the purpose of the 2nd zap card? |
| 18:31.12 | ManxPower | you could enable that ugly DTMF transfer hack. See features.conf |
| 18:31.43 | *** join/#asterisk oej_ (n=oej@apollo.webway.se) |
| 18:32.04 | haryv | SetCID num can override the customers user account name? I had that the other day on a sip call in. 000000 was on my CID display |
| 18:32.04 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) |
| 18:33.58 | *** join/#asterisk ryderblue (n=ryderblu@208.49.103.100) |
| 18:34.14 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 18:35.37 | Holos | Anyone know how to disable the call foward button on a Polycom phone? I have static agents, and someone forwarded their phone to an configuration extension and messed up the queue. |
| 18:35.49 | ellisdee | sip.cfg |
| 18:37.46 | haryv | which model has the call forward button on them? |
| 18:38.10 | Holos | haryv: Polycom 30x, 50x, 60x's all have a foward soft button |
| 18:38.51 | haryv | okay the soft button |
| 18:38.54 | haryv | I dont use that |
| 18:39.47 | Holos | haryv: I use it occasionally, my clients for some reason are drawn to it... |
| 18:40.06 | haryv | Thats important to know |
| 18:42.17 | Corydon-w | Holos: set divert.fwd.x.enabled="0" |
| 18:43.04 | *** join/#asterisk VibroMax (n=phisto@83.229.70.154) |
| 18:44.43 | Holos | Corydon-w: I have that in the mac.cfg, but it doesnt seem to be working, do I have my xml wrong? http://pastebin.ca/245206 |
| 18:45.37 | *** join/#asterisk zotz (n=zotz@24.244.133.107) |
| 18:45.50 | Corydon-w | Holos: can you confirm that the phone picked up that config during its last reboot? |
| 18:46.56 | Holos | Corydon-w: No.. because it is looking in a different folder... (I have a PlcmSpIp1 user for testing 2.0.x)...Dooh' |
| 18:49.15 | hmmhesays | my redirects are not working right |
| 18:49.16 | hmmhesays | damnit |
| 18:50.22 | Holos | Corydon-w: That divert.fwd.x.enable="0" worked obviously. Once I edited the correct files... |
| 18:51.28 | tao2 | GOT it i just made a queue for each ext and put there ext in the ring group that way if a call it put though to them they are put on hold untill the line is free |
| 18:53.06 | haryv | holos, are you disabeling it? |
| 18:55.01 | *** part/#asterisk rene1 (n=rene1@189.140.234.203) |
| 18:55.18 | *** join/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 18:55.48 | *** join/#asterisk rene1 (n=rene1@189.140.234.203) |
| 18:55.49 | Holos | haryv: Yes, I disabled call forwarding for the people in queues. |
| 18:55.57 | rene1 | i have been using the MYSQL app |
| 18:56.27 | rene1 | but i want to connect to tables that live in different databases in the same database server |
| 18:56.50 | rene1 | i wonder if i can reuse the connection and access the table that lives outside by means of dbname.dbtable ? |
| 18:57.02 | rene1 | permissions would allow it |
| 18:57.18 | rene1 | i wonder if asterisk 's mysql would play nicely with it |
| 18:58.02 | Zordrak | Ok.. I had no luck at lunchtime.. let's see what happens now: |
| 18:59.54 | Zordrak | I am evaluating Asterisk and OpenPBX for a new PBX server for our company to replace an ageing Lucent PBX. I'm looking for reasons to choose one over the other. I have been given many reasons to use OpenPBX over Asterisk.. but the only reason I've been given so far for using Asterisk instead of OpenPBX is that Asterisk has a company behind it and is more likely not to stop development. Can someone - ANYONE - give me more reason than that for choosing Aster |
| 19:00.16 | *** join/#asterisk drkludge (n=drkludge@ip68-231-40-97.ph.ph.cox.net) |
| 19:00.40 | Qwell[] | Zordrak: Asterisk is actively developed by a dedicated group of people, not just "in their spare time" - although, we also of course have community members who do stuff in their spare time |
| 19:01.23 | Qwell[] | and there are also various methods of support (paid, or free, such as in here) |
| 19:01.56 | Zordrak | Yes, that's pretty much what has already been covered.. it's developed by different people in a different way.. but what about the program itself? Can anyone tell me that Asterisk is a better product than OpenPBX or that it has better options, or that it's easier to use or anything that Finance and CEO could understand? |
| 19:02.33 | Zordrak | Something other than "OpenPBX is developed by a disparate Open Source community" ? |
| 19:02.37 | Qwell[] | sure, in my opinion, Asterisk is a better product than OpenPBX |
| 19:02.54 | Zordrak | OK. Why? |
| 19:03.13 | Qwell[] | partially, for the aforementioned reasons |
| 19:03.28 | *** join/#asterisk knarfly (n=bcartwri@c-65-34-177-3.hsd1.fl.comcast.net) |
| 19:03.29 | Qwell[] | The direction is understood, the future is also known |
| 19:03.44 | Qwell[] | with OpenPBX, (again, in my opinion), you never really know what you're gonna get |
| 19:03.50 | Zordrak | Forget the future. What about now. The product in its current state. |
| 19:04.40 | xmws | any idea when can we do t.38 with *? I would like to have t.38 termination |
| 19:04.45 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 19:04.45 | *** mode/#asterisk [+o mog] by ChanServ |
| 19:04.48 | Qwell[] | well, what features have they added, and what bugs have they fixed, since they forked off? |
| 19:05.04 | file | I would suggest finding someone who has used both and see what their perspective is... but I can't think of anyone off the top of my head |
| 19:05.06 | justinu|laptop | T.38 |
| 19:05.07 | Qwell[] | I can tell you the features we've added - just look at the 1.4 beta release notes stuff |
| 19:05.16 | justinu|laptop | T.38 gateway ,not just passthru |
| 19:05.30 | CunningPike | Zordrak: I think an important consideration is vendor certification - hardware manufacturers like Polycom and Audiocodes have certified their equipment on Asterisk |
| 19:05.35 | Zordrak | For example.. the OpenPBX homepage has a list of at least 5 specific reasons why OpenPBX is a better product and is going to be better for you. Tell me why they're wrong and that Asterisk is what I should use. |
| 19:05.36 | Qwell[] | justinu|laptop: What else? :) |
| 19:05.46 | xmws | justinu|laptop: is ther really t38 gateway? |
| 19:05.48 | Qwell[] | Zordrak: let me look at those reasons. |
| 19:05.50 | justinu|laptop | t.38 gateway is pretty major |
| 19:05.51 | Zordrak | CunningPike: good. There's at least a good suggestion of a reason |
| 19:05.59 | justinu|laptop | not a trivial thing to implement |
| 19:06.11 | Zordrak | http://wiki.openpbx.org/tiki-index.php |
| 19:06.40 | CunningPike | Zordrak: I think you're going to find many more consultants etc to pick from also - Asterisk is almost semi-commercial in the level of support, training etc |
| 19:07.20 | CunningPike | Zordrak: There is a 500+ attendee user conference..... |
| 19:07.22 | Qwell[] | okay, 8 reasons I see |
| 19:07.32 | mog | Zordrak, asterisk 1.4, supports, jabber, jingle, imap, whisper paging, ael, etc |
| 19:07.35 | mog | look at the changelog |
| 19:07.38 | mog | for 1.4 |
| 19:07.51 | mog | almost all of the features have been added here and not there |
| 19:07.57 | Zordrak | CunningPike: But then, OpenPBX is so close to Asterisk.. most of the support that you can get from non-developers goes for both. |
| 19:07.59 | mog | not to say they couldnt use such features |
| 19:08.01 | mog | just dont |
| 19:08.11 | mog | openpbx is similar to asterisk 1.0-1.2 |
| 19:08.13 | justinu|laptop | universal jitter buffer and t.38 gateway are the biggest things, and neither of those are trivial to implement |
| 19:08.26 | Qwell[] | justinu|laptop: 1.4 has a generic jitter buffer |
| 19:08.29 | mog | we have universal jttter buffer |
| 19:08.32 | Qwell[] | So, they should remove that, IMO :) |
| 19:08.52 | xmws | waaao, universal jitter buffer? that's good for the SIP call quality |
| 19:09.03 | *** join/#asterisk slappin (n=slappin@c-68-34-143-173.hsd1.mo.comcast.net) |
| 19:09.11 | Zordrak | As well as the lack of dependency on proprietary timing.. POSIX timing instead so you don't get problems when you dont buy a digium card |
| 19:09.21 | Qwell[] | Zordrak: we have ztdummy |
| 19:09.27 | Qwell[] | You don't need hardware with that |
| 19:11.00 | mog | ztdummy is kernel specific, but it does run on freebsd, linux, and i belive solaris kernels now |
| 19:11.55 | Qwell[] | and the VMware thing is kinda...meh. Some people use Asterisk (Damin for instance) under it just fine |
| 19:12.14 | rene1 | OpenPbx claims better stability, but as it has been said we need an unbiased opinion of a knowledgeable user of both systems |
| 19:12.34 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 19:12.44 | drkludge | sqllite3 verses Berkeley DB? I don't believe that's that big of deal. There's all kinds of extensions for mysql, etc. |
| 19:13.51 | rene1 | i think they also do googletalk |
| 19:15.38 | *** join/#asterisk Qain_SK (n=qain@adsl-072-149-043-253.sip.bct.bellsouth.net) |
| 19:15.57 | *** join/#asterisk quidpro (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 19:17.38 | quidpro | Hmm... anybody here have a Linksys SPA-3102? |
| 19:24.15 | knarfly | 8-) |
| 19:30.49 | rene1 | is the digium/sangoma echo canceling digital TDM hardware worth it? they are $1000 over regular non echo canceling versions.. say you have good quality IP stations, long distance calls should be echo cancelled by the telco, local calls to analog numbers could be handled by software, would offloading this echo canceling to the card save enough cpu power to justify the 1000 investment? |
| 19:31.20 | mercestes | rene1: Yes. |
| 19:31.34 | mercestes | rene1: If you are running any real call volume. |
| 19:31.37 | mog | rene1, you should try software first in my opinion |
| 19:31.38 | Cresl1n | rene1: set it up without it. You'll know very soon whether or not you needed it. |
| 19:31.40 | mog | it might be enough for you |
| 19:31.48 | mercestes | rene1: Compare that to dumping about 20g's into a server that cna handle it instead of the card. |
| 19:31.49 | mog | then you can always upgrade the card |
| 19:32.20 | rene1 | mercestes: predictive dialing, aka dial as much as you can per card/line |
| 19:32.24 | mercestes | I agree with them tho....IF your not having echo problems I wouldn't worry about hardware echocancellers. |
| 19:32.55 | rene1 | so i should explicitly disable echo canceling in software and see if it goes well |
| 19:33.09 | Cresl1n | no, don't disable it |
| 19:33.11 | CunningPike | rene1: Try software first, as mog suggests. In our experience (albeit with the older VPM), the hardware e/c won't cancel any echo that the software can't - in other words, it's cpu offload only. We went for an external e/c box in the end |
| 19:33.13 | Cresl1n | keep it enabled |
| 19:33.28 | mercestes | rene1: I know a guy who said that a PRI/T1 should not have echo at all and therefore echocancellers were unnecessary and merely a "cover up" for problems elsewhere. |
| 19:33.44 | Cresl1n | well the new VPM can cancel quite a bit better than the software can |
| 19:33.54 | mog | in a pure digital situation you shouldnt have echo mercestes |
| 19:34.01 | mog | echo is from analog links |
| 19:34.04 | CunningPike | Cresl1n: Yes - I believe that's tru |
| 19:34.04 | rene1 | mercestes i am aware of that but dialed number is most of the time analog and therefore prone to echo |
| 19:34.51 | rene1 | mog: should the telco cancel any echo coming tru their analog lines? |
| 19:35.23 | mog | yes, they should echo cancel their edge devices |
| 19:35.44 | mog | just like if you have a pri to telco , you should echo cancel your edge devices if they are analog |
| 19:36.02 | Cresl1n | rene1: but that doesn't always happen in reality |
| 19:36.03 | rene1 | they are all sip |
| 19:36.26 | rene1 | Cresl1n: the telcos echo canceling all their devices? |
| 19:36.47 | *** join/#asterisk spackle (n=spackle@ip207-199-243-35.static.ishsi.com) |
| 19:36.57 | mog | yeah they dont always do it rene1 |
| 19:37.11 | *** part/#asterisk spackle (n=spackle@ip207-199-243-35.static.ishsi.com) |
| 19:37.29 | rene1 | i think the book talked about they always canceling in long distance calls but not always in local calls |
| 19:38.15 | rene1 | the thing is that for the time being i am stuck with a low performing clone that needs to do recording at a 20-25 simultaneous channels level |
| 19:38.46 | rene1 | i get dropped audio and calls at the 18 calls level |
| 19:39.19 | Cresl1n | what, your T1/E1 card you mean? |
| 19:39.22 | Cresl1n | what kind of card is it |
| 19:40.34 | rene1 | a 2 port |
| 19:40.37 | rene1 | E1/T1 card 5v |
| 19:40.54 | rene1 | made by digium |
| 19:41.04 | rene1 | the host is a pentium 4 3.4 2 Gb ram |
| 19:41.47 | rene1 | somebody was using the same config but with @home and they could only get like 12 sym calls with good audio quality |
| 19:42.11 | rene1 | so many AGIs and the FOP backend make for a slower setup |
| 19:42.29 | Cresl1n | that seems pretty ridicuous |
| 19:42.41 | rene1 | Cresl1n: what part? |
| 19:43.44 | rene1 | that the server is underpowered? or my claims that it is under powered? or the @home part |
| 19:44.08 | Cresl1n | only 12 sym calls seems pretty low |
| 19:44.31 | *** part/#asterisk BurtyB (n=chris@alxa.camelnetwork.com) |
| 19:44.46 | haryv | some thing is seriosly grabbing cycles then. |
| 19:44.51 | haryv | did that person do a top? |
| 19:44.54 | *** join/#asterisk Amilcar_ (n=xxxxx@201.34.202.17) |
| 19:45.24 | *** join/#asterisk Tclp (n=tcalp@d198-166-31-102.abhsia.telus.net) |
| 19:46.32 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 19:47.01 | rene1 | yeah |
| 19:47.16 | rene1 | the system is like 70% idle |
| 19:47.22 | rene1 | but there are dropped calls |
| 19:47.48 | rene1 | we are doing recording |
| 19:47.51 | rene1 | and that can be a bitch |
| 19:48.45 | Tclp | hey all, I know this isn't isn't 100% asterisk related, but you guys are generally the goto guys for telephony. My company is attempting to setup a national VOIP call center (having agents in different areas of the country). Who would you guys recomend for a 1800 based phone number... Optimally I'd like free inbound calling, or at least very good Per/min rates with free local inbound. |
| 19:51.31 | quidpro | If you can find a 1-800 with free inbound, sign me up! |
| 19:53.26 | rene1 | hehe |
| 19:53.59 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 19:54.40 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 19:55.21 | *** join/#asterisk boryalic (n=chatzill@c-71-233-249-251.hsd1.ma.comcast.net) |
| 19:55.46 | Tclp | quid .. hehe I know .. still had to aks though :P ... |
| 19:56.16 | *** part/#asterisk boryalic (n=chatzill@c-71-233-249-251.hsd1.ma.comcast.net) |
| 19:57.40 | *** part/#asterisk natmlt (n=natmlt@ip68-109-184-159.ph.ph.cox.net) |
| 19:57.41 | hmmhesays | heh you won't get a free 1800 inbound |
| 19:57.57 | hmmhesays | who would get paid for providing that service if they gave it to you for free? |
| 19:57.59 | hmmhesays | makes no sense |
| 19:58.45 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 19:59.02 | Tclp | well I meant where you pay X amount for unlimited |
| 19:59.11 | hmmhesays | won't happen |
| 19:59.19 | hmmhesays | not on an toll free number |
| 19:59.33 | rene1 | there are 1cts ~ 2cts incoming providers |
| 19:59.47 | Tclp | ohh |
| 19:59.52 | Tclp | 1cts ~ 2cts .. :) |
| 19:59.55 | Tclp | that's decent |
| 20:00.00 | Tclp | best I've seen so far is 3.9 |
| 20:00.50 | *** join/#asterisk cloud9 (n=nothing@smtp.a-cti.com) |
| 20:01.37 | *** join/#asterisk knobenheimer (n=knobenhe@c-71-205-184-144.hsd1.mi.comcast.net) |
| 20:01.39 | rene1 | nufone |
| 20:01.55 | rene1 | havent used tho |
| 20:02.51 | cloud9 | I have a dial plan question. I've read that you can use variables in a dial plan. Is there an easy way to make a variable for extensions. for example, something like a line that looks like exten => ${foo},1,Goto(internal,${foo),1)? thanks in advance |
| 20:03.16 | quidpro | Anybody using LinkSys/Sipura adaptors? I need one that doesn't "hiss". |
| 20:03.17 | Tclp | http://nufone.net/ ? |
| 20:04.27 | *** join/#asterisk brc_ (n=brc___@pdpc/supporter/basic/brc) |
| 20:05.12 | hads | cloud9: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf |
| 20:06.56 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:07.29 | rene1 | yep |
| 20:11.24 | cloud9 | hads : good effort lad. |
| 20:11.43 | cloud9 | I was actually already looking at that exact URL. |
| 20:12.17 | cloud9 | I can't seem to find what I'm looking for. :-/ |
| 20:12.27 | hads | So you are wanting something like... |
| 20:12.28 | brc_ | anybody seen the primary d-chan go down, then get this warning " chan_zap.c:2290 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway!" |
| 20:12.46 | brc_ | telco claims everything checks out on their end |
| 20:12.50 | *** join/#asterisk aptura (n=sales@S010600a0c93f6f7e.vs.shawcable.net) |
| 20:13.04 | hads | cloud9: exten => _4XX,1,Goto(internal|${EXTEN}) |
| 20:13.08 | cloud9 | I understand how to use a variable in my dialplan, but I need a way for asterisk to detect what the caller just entered as an extension, and the put that into a variable, and then call that variable. |
| 20:13.23 | hmmhesays | show application read |
| 20:13.28 | cloud9 | oh really? let me try that |
| 20:13.47 | hads | That would match 400-499 |
| 20:13.57 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 20:14.06 | hmmhesays | i need a fax number to send a test fax to |
| 20:14.07 | hads | The EXTEN variable is populated with what was dialled |
| 20:14.38 | rene1 | brc_ that happens when the line goes under |
| 20:14.46 | rene1 | as in red alarm |
| 20:14.51 | brc_ | k |
| 20:15.27 | rene1 | possible causes: a) telco (very likely) b) cpu high load (somewhat) c) bad card (rare) |
| 20:15.42 | rene1 | check the logs and see if it happens recurrently |
| 20:15.51 | rene1 | check if it happens when the pbx is idle |
| 20:16.00 | cloud9 | hads : you're very cool |
| 20:16.03 | cloud9 | thank you |
| 20:16.10 | brc_ | rene1, k, thanks |
| 20:16.19 | rene1 | np :) |
| 20:16.28 | brc_ | it's happening almost everytime you attempt to place a call |
| 20:16.33 | hads | cloud9: No worries :) |
| 20:16.35 | brc_ | and the card was already replaced |
| 20:16.39 | brc_ | telco's probably lying :\ |
| 20:16.45 | rene1 | try this: |
| 20:16.51 | rene1 | take your box to the telco premises |
| 20:17.00 | hads | cloud9: There is a page somewhere on voip-info that explains dialplan pattern matching in more detail. |
| 20:17.04 | rene1 | and have it connected to their switch |
| 20:17.09 | rene1 | see if you have this issues |
| 20:17.16 | rene1 | they probably have some problem down their network |
| 20:17.54 | brc_ | I'm offsite and trying to help out a friend with a box I put in last year, but he's been taking care of it since... |
| 20:18.03 | rene1 | have him do it |
| 20:18.15 | rene1 | what you can do is the pattern test |
| 20:18.20 | rene1 | look in the zaptel sources |
| 20:18.42 | brc_ | don't recall that...do you know where I'd find it? |
| 20:18.53 | rene1 | patlooptest or something |
| 20:18.59 | rene1 | look in yoir zaptel src |
| 20:19.03 | brc_ | k |
| 20:19.56 | *** join/#asterisk spr1te (n=spr1te@213.227.193.75) |
| 20:20.54 | CunningPike | What's the difference between CALLERID(num) and CALLERID(ani) - do I need to set both? |
| 20:22.42 | CunningPike | I've read about the CDR changes in 1.4 that requires (ani) to be set - but do I need to set (num) as well? |
| 20:23.01 | quidpro | Hmm... what happened to #trixbox? |
| 20:23.32 | Nivex | I eated it. |
| 20:24.13 | rene1 | hehe |
| 20:24.13 | rene1 | burp |
| 20:24.15 | hmmhesays | who's gonna turn down a junior mint |
| 20:24.20 | quidpro | me |
| 20:24.23 | quidpro | mint's suck |
| 20:24.30 | quidpro | along with Peppermint Patti's |
| 20:24.53 | Nivex | York 4tw! |
| 20:25.11 | rene1 | there should be some mints coke flavored |
| 20:25.28 | hmmhesays | lol |
| 20:25.38 | hmmhesays | my intarweb connection is crap right now |
| 20:25.40 | hmmhesays | just awful |
| 20:26.13 | hmmhesays | I don't want to know your name, I just want bang bang bang |
| 20:27.26 | *** join/#asterisk _securez_ (n=securez@121.Red-80-33-36.staticIP.rima-tde.net) |
| 20:27.36 | _securez_ | Hello |
| 20:27.51 | *** join/#asterisk trelane_ (n=trelane@unaffiliated/trelane) |
| 20:28.18 | Tclp | aside from nuphone, anyone have recomendations for cheap inbound 1800 service ? |
| 20:28.58 | quidpro | try les.ent |
| 20:29.00 | quidpro | les.net |
| 20:29.19 | _securez_ | Anybody can point me, if the ADSL split can entroduce echo on the line, i have two analog lines, one with a ADSL other onlt phone, i have 20-30%, 2% echo respectly |
| 20:29.31 | _securez_ | i have used the last fxotune from CVS |
| 20:29.42 | aptura | sixtel is now offering inbound 1800 and is expanding. |
| 20:29.45 | rene1 | yeah lines with adsl are tricky |
| 20:29.57 | *** join/#asterisk WGFreewill (n=chatzill@69-170-244-239.atlsfl.adelphia.net) |
| 20:30.02 | _securez_ | can i solve the problem with the echo |
| 20:30.10 | *** join/#asterisk zotz (n=zotz@24.244.133.107) |
| 20:30.10 | rene1 | use the soft echo canceller |
| 20:30.16 | rene1 | in zaptel |
| 20:30.35 | *** join/#asterisk alerios (n=alerios@190.24.97.148) |
| 20:30.36 | hads | rene1: What does ADSL do that makes it tricky? |
| 20:30.40 | WGFreewill | is the new jitterbuffer enabled by default in asterisk 1.2 latest cvs, or how do I enable it |
| 20:30.56 | InfraRed | ] |
| 20:32.18 | WGFreewill | I see this stuff in chan_iax2.c |
| 20:32.20 | WGFreewill | #define NEWJB |
| 20:33.12 | WGFreewill | I need PLC badly |
| 20:33.46 | Strom_M | WGFreewill, have you tried jitterbuffer=yes? |
| 20:34.06 | WGFreewill | yes I wasn't sure if I was getting the old jitterbuffer or the new one |
| 20:35.03 | *** join/#asterisk Lawbringer (n=Lawbring@84-45-215-247.no-dns-yet.enta.net) |
| 20:35.48 | Lawbringer | hey, if i get an fxo card can i use it to route faxes? |
| 20:35.58 | WGFreewill | There are presently two jitterbuffer implementations available for Asterisk and chan_iax2; the classic and the new, channel/application independent implementation. These are controlled at compile-time. |
| 20:36.08 | Lawbringer | or can a modem route voice calls? |
| 20:36.12 | WGFreewill | there should be some Makefile flags or something |
| 20:38.14 | _securez_ | rene1 i try echo canceler MARK2 and MG2 |
| 20:38.22 | *** join/#asterisk IANster (n=ian@cpe-72-129-17-182.socal.res.rr.com) |
| 20:38.35 | _securez_ | but there is a considerably echo |
| 20:38.43 | *** part/#asterisk Cresl1n (i=matt@nat/digium/x-04ba434827c91001) |
| 20:41.18 | *** join/#asterisk brif8 (n=brif8@67.78.24.178) |
| 20:43.21 | _securez_ | lawbringer fxo connect to wall, fxs to phone/fax, if you have both you can route the call |
| 20:47.10 | hmmhesays | bah fax |
| 20:47.11 | hmmhesays | sucks |
| 20:47.32 | WGFreewill | fax works great with eicon cards |
| 20:51.45 | aptura | WGFreewill how so |
| 20:51.59 | *** join/#asterisk backblue (n=moo@87-196-3-20.net.novis.pt) |
| 20:52.02 | WGFreewill | you stick your eicon card in your server |
| 20:52.06 | WGFreewill | configure it for CAPI |
| 20:52.16 | WGFreewill | and the t.30 is handled in hardware on the eicon |
| 20:52.30 | WGFreewill | I use hylafax instead of asterisk |
| 20:52.36 | aptura | Right |
| 20:52.40 | aptura | well known app |
| 20:52.46 | CunningPike | Anyone get this running ./configure on RHEL with beta3: configure: error: C++ preprocessor "/lib/cpp" fails sanity check |
| 20:52.54 | aptura | im looking at there cards and see thay are wan cards. |
| 20:53.01 | WGFreewill | I have a box with spandsp and digium 405 |
| 20:53.10 | WGFreewill | 80% of inbound works great |
| 20:53.17 | WGFreewill | outbound is less reliable |
| 20:53.22 | aptura | of the fax calls? |
| 20:53.26 | WGFreewill | yeah |
| 20:53.38 | aptura | clients are only interested in 100% if possible |
| 20:53.42 | WGFreewill | inbound to rxfax then a script to PDF |
| 20:53.47 | Tclp | anyone know much about CallWave ? |
| 20:53.48 | *** part/#asterisk slappin (n=slappin@c-68-34-143-173.hsd1.mo.comcast.net) |
| 20:53.53 | aptura | remember, competing against fax machines |
| 20:53.56 | WGFreewill | but once you check out the eicon/hylfax solution, hehehe |
| 20:54.12 | WGFreewill | you forget spandsp if you can get that eicon card |
| 20:54.28 | *** join/#asterisk zamba (i=marius@flage.org) |
| 20:54.30 | aptura | i dont see the fax cards. |
| 20:54.36 | rene1 | CunningPike: get g++ |
| 20:54.42 | WGFreewill | I use a DivaServer PRI-30M |
| 20:54.48 | WGFreewill | I think is the one |
| 20:54.53 | WGFreewill | works with asterisk voice chan_capi |
| 20:54.59 | CunningPike | rene1: THought I had it already - guess not...... :) |
| 20:55.05 | WGFreewill | should be some way to share 1 card between asterisk and hylafax |
| 20:55.06 | aptura | okay diva server then |
| 20:55.30 | Tclp | they aparently offer unlimited SIP 1800 service for $10.95 unlimited incoming .. 60min max per call |
| 21:00.56 | WGFreewill | I wish someone here know about the new jitterbuffer in 1.2 |
| 21:01.45 | *** join/#asterisk JustinWick (n=jwick@unaffiliated/jpl/jpl-justin) |
| 21:04.17 | *** part/#asterisk brif8 (n=brif8@67.78.24.178) |
| 21:06.42 | *** join/#asterisk pfn (n=pfnguyen@netblock-66-245-252-239.dslextreme.com) |
| 21:07.29 | *** join/#asterisk lawbringer (n=Lawbring@84-45-215-247.no-dns-yet.enta.net) |
| 21:07.45 | lawbringer | gah why did i have to throw all my win modems away |
| 21:08.21 | lawbringer | now i only have hard modems and can't use them as fxo cards |
| 21:09.18 | hmmhesays | modems suck |
| 21:09.45 | lawbringer | well i only need one line mostly for fax but also for backup reasons |
| 21:10.30 | _securez_ | bye |
| 21:12.20 | *** join/#asterisk scratchme (n=kvirc@nc-76-3-205-133.dhcp.embarqhsd.net) |
| 21:12.39 | lawbringer | ohh i might be able to use that ISA port after all! |
| 21:12.47 | scratchme | What's a good and cheap FXS port either as PCI card or separate small box? |
| 21:13.20 | scratchme | And do you think it's worth having a fail-over to a land-line? |
| 21:13.49 | CunningPike | Anyone getting "Install ncurses to use the menu interface!" when running 'make menuselect' in beta3? |
| 21:14.01 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 21:14.42 | hmmhesays | sounds like you need to install ncurses |
| 21:15.17 | scratchme | Has anyone installed asterisk and ipcop into the same box before? |
| 21:15.44 | lawbringer | scratchme, i do think "fail over" is worth it in a production system |
| 21:16.27 | scratchme | lawbringer: It's for my house. |
| 21:16.42 | *** join/#asterisk brif8 (n=brif8@67.78.24.178) |
| 21:17.04 | scratchme | I'm one of those anal people who wants complete control over things like his telephone, and people being able to call with private/blocked IDs and such. |
| 21:17.16 | scratchme | I want them to get through, but I want them to verify they aren't automated calls. |
| 21:17.24 | lawbringer | scrachme, then it up to you, if you keep a old phone around next to the socket for emergency |
| 21:17.57 | scratchme | True, I guess I could just hook up an old phone instead, although if it failed over automatically, it would "just work" for outbound calls. |
| 21:18.08 | lawbringer | i'm putting in a fxo card for fax and failover |
| 21:18.27 | lawbringer | if you think the effort is worth it then do it |
| 21:18.32 | lawbringer | if not don't |
| 21:18.35 | scratchme | Yeah, currently I need inbound modem ability atm too, so I guess I probably should do the same. |
| 21:18.44 | scratchme | Modem over voip is kinda lame. |
| 21:19.05 | CunningPike | hmmhesays: ncurses is already installed - did a 'make distclean' - still getting the error |
| 21:19.15 | rene1 | ncurses dev |
| 21:19.25 | rene1 | or whatever is called in your system |
| 21:19.38 | rene1 | the headers for ncurses |
| 21:21.50 | *** join/#asterisk dasenjo (n=dasenjo@201.228.128.10) |
| 21:22.02 | dasenjo | Hi, I have a doubt |
| 21:22.05 | CunningPike | hmmhesays: That did the trick - thanks |
| 21:22.29 | dasenjo | does _X. match the 9 (or any single digit) extension? |
| 21:22.41 | dasenjo | it is not working for me |
| 21:22.58 | CunningPike | dasenjo: That matches _everything_ - you shouldn't really use it |
| 21:23.14 | CunningPike | dasenjo: _X matches a single digit |
| 21:23.31 | *** part/#asterisk brif8 (n=brif8@67.78.24.178) |
| 21:23.42 | dasenjo | I need it .. I need to match everithig to redirect it to another context |
| 21:23.59 | dasenjo | CunningPike, thanks |
| 21:24.27 | *** join/#asterisk droops (n=droops@adsl-074-245-001-031.sip.jan.bellsouth.net) |
| 21:24.36 | CunningPike | dasenjo: So, what's not working |
| 21:24.37 | hmmhesays | np |
| 21:24.48 | mercestes | _X I think..matches at the least 2 digits, I don't think it will pick up one digit unless I'm terribly mistaken. |
| 21:24.53 | CunningPike | My favorite 3 words: "It's not working...." |
| 21:25.07 | dasenjo | CunningPike, my context has an _X. but 9 does no match |
| 21:25.26 | mercestes | _X. rather, matches 2 digits or more. |
| 21:25.29 | mercestes | _X matches any one digit. |
| 21:25.34 | hmmhesays | dasenjo: what mercestes sai |
| 21:25.37 | hmmhesays | *said |
| 21:25.57 | scratchme | Is something like this good? http://cgi.ebay.com/Zoom-5801-ATA-VOIP-SIP-FXS-FXO-Asterisk-VOIP-to-PSTN_W0QQitemZ160049820052QQihZ006QQcategoryZ61840QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 21:25.59 | mercestes | and no, I don't think _. is a valid extension... |
| 21:26.13 | mercestes | I could be wrong again, but IIRC, it's not happy that way |
| 21:26.20 | rene1 | it used to be valid |
| 21:26.27 | dasenjo | I think _X. should match one digit extensions |
| 21:26.31 | hmmhesays | should still be, but you shouldn't use it |
| 21:26.33 | mercestes | Why? |
| 21:26.46 | rene1 | deprecation |
| 21:26.57 | hmmhesays | cause it matchs extension h |
| 21:27.09 | mercestes | X is a single digit, and . is any number of digits...so you are saying "Match any one digit, then any number of digits after that." |
| 21:27.25 | mercestes | Soryr, I was asking dasenjo "why" Not you guys |
| 21:27.42 | mercestes | as in, "why should _X. match one digit when syntatically it asks for 2 or more." |
| 21:28.05 | mercestes | lol...extension h. |
| 21:28.16 | mercestes | that would be amusing. |
| 21:28.22 | hmmhesays | what about it? |
| 21:28.42 | mercestes | isn't h an "on hangup" extension? |
| 21:28.49 | hmmhesays | yes |
| 21:28.54 | dasenjo | why I should not use it? I need to redirect all the incoming calls on a trunk context with a Goto |
| 21:29.04 | hmmhesays | if you have an h extensions and a _. h won't get called on hangup |
| 21:29.06 | scratchme | http://www.zoom.com/techsupport/voip/voip_5801.shtml <== is that any good, or should I move on and look at something else? |
| 21:29.06 | dasenjo | I can't imagine another way |
| 21:29.26 | mercestes | so it's like _.,1,Dial(${Exten}) Call mom....."hi mom." *hang up* *call mom again with null call* |
| 21:29.44 | hmmhesays | dasenjo: you have one extensions _X and one extensions _X. |
| 21:30.07 | mercestes | dasenjo: Yea, you do have copy paste figured out in vim, right??? It can be what, at most, 2 lines?? |
| 21:30.48 | hmmhesays | i think i'm going to go watch an episode of battlestar galactica |
| 21:30.51 | dasenjo | i have to include my _X line .. thanks a lot |
| 21:31.02 | mercestes | Your welcome. |
| 21:31.03 | mercestes | happy redirecting. |
| 21:31.09 | hmmhesays | goto'ing |
| 21:31.17 | mercestes | That too. |
| 21:31.35 | mercestes | s,1,Goto(away) lol |
| 21:32.31 | aptura | hmmhesays I am setting up some external sip phone to pbx accounts and want to know why my callerid=("cidname"<number>) in sip.conf is not showing up on my ip display? |
| 21:32.40 | dasenjo | :) |
| 21:32.47 | dasenjo | :XD |
| 21:32.50 | aptura | or if anyone else cares to answer that :) |
| 21:32.57 | mercestes | lol...I was kidding. |
| 21:33.06 | mercestes | just struck me as randomly funny |
| 21:33.25 | CunningPike | What does codec_zap do? |
| 21:33.43 | mercestes | aptura: SOme carries simply don't transfer caller Id correctly. |
| 21:34.01 | aptura | this is sip to my box..no carrier involved |
| 21:34.10 | aptura | its direct |
| 21:34.11 | Corydon-w | CunningPike: it permits no translation in software to be done for channels which use the hardware codec translation |
| 21:34.42 | CunningPike | Corydon-w: So, bridged zap channels? |
| 21:34.54 | Corydon-w | CunningPike: not just zap channels, but yes |
| 21:35.12 | CunningPike | OK - just trying to figure out if I need it or not..... |
| 21:35.16 | Corydon-w | CunningPike: the codec translation uses the zaptel layer |
| 21:35.47 | Corydon-w | CunningPike: not unless you have the codec card |
| 21:35.58 | CunningPike | Corydon-w: Ah - OK |
| 21:36.32 | scratchme | lawbringer: What are you using to have both fxs and fxo failover? |
| 21:45.06 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 21:52.43 | *** join/#asterisk brif8 (n=brif8@67.78.24.178) |
| 21:53.18 | brif8 | Is there a better Distro that links with Asterisk better? or What Distro is Asterisk Developed on ? |
| 21:54.04 | ellisdee | i prefer centos |
| 21:54.19 | ellisdee | compile asterisk, libpri, zap, etc from scratch |
| 21:54.46 | ellisdee | things tend to just work |
| 21:54.50 | ellisdee | centos or gentoo |
| 21:54.53 | ellisdee | both are great. |
| 21:55.17 | rene1 | anything dpkg based is grat |
| 21:55.22 | ellisdee | just try to free up as many irq's as possible by disabling devs in the bios. |
| 21:55.22 | rene1 | great |
| 21:55.39 | rene1 | i am interested in gentoo |
| 21:55.43 | ellisdee | if your running a 2.6 kernel |
| 21:55.43 | rene1 | tho |
| 21:55.57 | WGFreewill | debian you can just apt-get install asterisk |
| 21:56.02 | ellisdee | make sure you compile zaptel with 2.6 parameters |
| 21:56.06 | ellisdee | i learned the hard way |
| 21:56.08 | WGFreewill | and get 1.0.7 running in 5 seconds |
| 21:56.09 | ellisdee | on a production pbx =P |
| 21:56.37 | CunningPike | What is imap_tk? |
| 21:56.45 | CunningPike | Is it in addons? |
| 21:57.00 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 21:57.57 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-228-131.dc.res.rr.com) |
| 21:59.22 | brif8 | how about with Asterisk 1.4 ? |
| 21:59.29 | brif8 | still use gentoo or centos |
| 21:59.37 | ellisdee | yep |
| 21:59.54 | ellisdee | thats just my preference |
| 21:59.57 | ellisdee | others may vary |
| 21:59.59 | *** join/#asterisk rubble (n=rubble@196-47-23-173.access.uunet.co.za) |
| 22:00.15 | ellisdee | i am more familiar with centos/redhat |
| 22:00.33 | drkludge | ellisdee, What are the 2.6 options that you compile zaptel with |
| 22:00.34 | ellisdee | so i prefer it over gentoo |
| 22:00.43 | ellisdee | drkludge, make linux26 |
| 22:00.53 | ellisdee | drkludge, make install |
| 22:01.09 | ellisdee | if not your zap channels will fill up |
| 22:01.23 | ellisdee | and have problems hapnging u |
| 22:01.24 | drkludge | ellisdee, Thanks. I am wondering if that's why I am having some problems |
| 22:01.25 | ellisdee | hanging up |
| 22:01.39 | ellisdee | i had to manually restart my wctdm/zap |
| 22:01.42 | brif8 | thanks |
| 22:01.48 | ellisdee | modules |
| 22:01.51 | ellisdee | after this problem occured. |
| 22:01.56 | *** join/#asterisk justinu|laptop (n=Justin@cpe-76-170-91-201.socal.res.rr.com) |
| 22:02.18 | ellisdee | all that spam in /var/log/asterisk/messages. |
| 22:02.28 | ellisdee | had events where i recieved, and just couldn't hangup |
| 22:02.50 | ellisdee | did verbose logging on the zap activity to confirm this until all channels filled. |
| 22:02.52 | aptura | frick snap crashed |
| 22:02.54 | ellisdee | then i couldn't make outbound or inbound |
| 22:05.54 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 22:05.58 | PakiPenguin | hello everyone |
| 22:06.12 | *** join/#asterisk icel (n=dan@63.78.162.73) |
| 22:07.20 | icel | anyone use a digital interface card from digium? |
| 22:08.11 | ellisdee | yes |
| 22:08.38 | [TK]D-Fender | icel : Nope. Digium is a manufacturer of fictitious hardware. Oh... and you're not really here, and this conversation isn't happening. |
| 22:08.40 | icel | i have a te405p, i was wondering if you need a data T1 or a voice T1 |
| 22:08.52 | icel | Fender: how enlightening |
| 22:09.03 | [TK]D-Fender | icel : You're welcome ;) |
| 22:09.21 | ellisdee | well icel |
| 22:09.23 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 22:09.36 | [TK]D-Fender | icel : You should be getting a PRI (Voice T1 w/ D channel for signalling) if at all possible |
| 22:09.57 | ellisdee | since fender replied |
| 22:10.16 | ellisdee | i will discard my comment as it was actually not helpful at all = |
| 22:10.16 | [TK]D-Fender | icel : You can get a partial PRI (or voice T1 but thats not the ideal choice), and use the rest for data if you wish |
| 22:10.17 | ellisdee | =P |
| 22:10.41 | icel | Thanks |
| 22:11.04 | *** part/#asterisk drkludge (n=drkludge@ip68-231-40-97.ph.ph.cox.net) |
| 22:11.51 | [TK]D-Fender | icel : You bought your card before finding out what kinds of circuits work with it? |
| 22:11.55 | icel | A regular PRI is the best choice then? |
| 22:12.17 | icel | Fender: I didn't buy it. Someone bought it and I am allowed to figure out how it works |
| 22:12.31 | ellisdee | hehe |
| 22:12.38 | icel | lucky me |
| 22:12.48 | *** part/#asterisk brif8 (n=brif8@67.78.24.178) |
| 22:12.56 | ellisdee | what your setup be a pots config. |
| 22:13.04 | ellisdee | which will require you to pickup a fxs to replace that card |
| 22:13.05 | [TK]D-Fender | icel : "Allowed to figure out how it works"...... WOW. A whole new approach to business! ;) |
| 22:13.29 | icel | yeah, i'm not sure what the previous guy had in mind. |
| 22:13.49 | ellisdee | pri is nice though |
| 22:13.50 | icel | I set up asterisk though and have been playing with SIP. Now I wanna get a T1 and make local calls |
| 22:13.54 | [TK]D-Fender | icel : yes, PRI is the best choice, though you don't need to use all 23 B channels for voice if you don't need that many. You can use the remainder for dtat. |
| 22:13.55 | ellisdee | quality is so great |
| 22:14.08 | icel | k |
| 22:14.22 | *** part/#asterisk rene1 (n=rene1@189.140.234.203) |
| 22:14.36 | icel | thanks you guys |
| 22:14.39 | ellisdee | my telco handles all my requests. |
| 22:14.40 | ellisdee | there iad |
| 22:14.42 | ellisdee | =P |
| 22:14.45 | ellisdee | i just tell them what i need. |
| 22:14.47 | ellisdee | they provide it to me. |
| 22:14.56 | ellisdee | and i config my pbx's |
| 22:14.59 | [TK]D-Fender | icel : Well vs SIP (which with ITSP's is usually terminated to PRI anyways) is not really too different in the big-picture except that being a direct interface its more reliable. |
| 22:16.10 | icel | k |
| 22:16.35 | ellisdee | sell that card icel |
| 22:16.40 | ellisdee | use all data channels |
| 22:16.43 | ellisdee | and go straight sip |
| 22:16.43 | ellisdee | ; |
| 22:16.45 | ellisdee | ;) |
| 22:16.55 | ellisdee | actualy |
| 22:17.02 | ellisdee | have a couple of those b channels for voice |
| 22:17.05 | ellisdee | backup =P |
| 22:17.29 | CunningPike | Does anyone know where to get imap_tk? |
| 22:17.46 | CunningPike | It's required for IMAP storage of voicemail |
| 22:18.07 | WGFreewill | anybody know if the new jitterbuffer is enabled in 1.2 |
| 22:18.12 | icel | ellisdee: so you can just use the ztdummy kernel module and just give the asterisk box an outside IP address and use SIP for everything as well, right? |
| 22:18.18 | WGFreewill | by default |
| 22:18.35 | ellisdee | why even go so far to do that. |
| 22:18.40 | ellisdee | isolate the pbx along with phones. |
| 22:18.44 | ellisdee | from the pc network |
| 22:18.49 | ellisdee | setup a vpn |
| 22:18.54 | ellisdee | tunnel |
| 22:18.55 | *** part/#asterisk Tili (n=tili@202.133.65.61) |
| 22:19.01 | ellisdee | and wallah |
| 22:19.16 | [TK]D-Fender | icel : You don't necessarily even need ZTDUMMY. But SIP works just fine for me at home, and I use a PRI at the office. |
| 22:19.23 | ellisdee | and no ztdummy |
| 22:19.25 | ellisdee | just wctdm |
| 22:19.26 | ellisdee | and zaptel |
| 22:19.26 | jeremy_g | :D |
| 22:20.14 | icel | k. The most difficult part so far about asterisk is trying to come up with a good design |
| 22:20.24 | ellisdee | hmm |
| 22:20.28 | icel | every time i think i have something figured out i find out something new |
| 22:20.30 | ellisdee | setup dhcp |
| 22:20.42 | ellisdee | setup a dir on your pbx |
| 22:20.45 | *** join/#asterisk kph100 (n=kph100@206-248-156-242.dsl.teksavvy.com) |
| 22:20.47 | ellisdee | have all phones pull its config files. |
| 22:20.50 | kph100 | hi, |
| 22:20.53 | ellisdee | along with ip from pbx |
| 22:20.56 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 22:21.19 | ellisdee | i have seen alot of really knarly voip setups |
| 22:21.42 | ellisdee | have not seen a working failover asterisk's setup yet tho |
| 22:21.43 | icel | knarly as in good or bad or just plane insane |
| 22:21.53 | ellisdee | knarly as it awesome |
| 22:21.58 | ellisdee | works great |
| 22:22.36 | *** join/#asterisk Peaceful (n=Peaceful@70.98.162.62) |
| 22:22.55 | ellisdee | pri asterisk setups can be a pain tho |
| 22:22.58 | ellisdee | expecially if you have fax lines |
| 22:23.06 | ellisdee | most stressful thing in the world to work with |
| 22:23.20 | Peaceful | I've got two mpg123 processes running at 100% processor usage on my asterisk box...even when no calls are going on. |
| 22:23.23 | Peaceful | Is that common? |
| 22:23.28 | kph100 | anyone knows how I can prevent loops from occuring in the dialplan? ie. menu1 goto menu2 gotomenu1. |
| 22:23.43 | ellisdee | hangup() |
| 22:23.44 | ellisdee | ? |
| 22:24.00 | Peaceful | I'm using asterisk 1.2.10 and mpg123 0.59s-r11 ... if that makes any difference |
| 22:24.11 | icel | ellisdee: thanks, i will let you know how it goes |
| 22:24.27 | ellisdee | have seperate lines run for faxes |
| 22:24.34 | ellisdee | make sure you got all your irq shit sorted. |
| 22:24.43 | [TK]D-Fender | Peaceful : You must use only mpg123 .059r |
| 22:24.52 | ellisdee | zap/pri should not share same bus or irq |
| 22:25.06 | ellisdee | i have had isntances of hearing hard drive platters spinning on a phone call =P |
| 22:25.08 | [TK]D-Fender | Peaceful : Then again, I suggest avoiding mpg123 altogether and using Native MoH |
| 22:25.43 | icel | wow |
| 22:26.12 | [TK]D-Fender | kph100 : Its your dialplan.... it'll do whatever you tell it to. |
| 22:35.01 | *** join/#asterisk Peaceful (n=Peaceful@70.98.162.62) |
| 22:37.47 | aosdonovan | has anyone heard of an ATA that does SIP and IAX and has a web configuration interface |
| 22:38.50 | aosdonovan | The digium IAXY works great but it will not work for our needs, we need something that can be modified in the field. The common user can not change a server ip user/pss in the field. |
| 22:39.15 | aosdonovan | Also, it does not not support SIP or the common sip codecs |
| 22:39.28 | aosdonovan | Any thoughts? |
| 22:40.25 | [TK]D-Fender | aosdonovan : There is nothing decent that supports IAX that I've ever heard of. |
| 22:40.26 | hads | The Virbiage ATA does AFAIK. I haven't used one myself and have heard mixed reports. |
| 22:40.41 | mercestes | I've got a polycom phone hooked to an * box that continually picks up "10 new messages" and "11 old" but there are no voicemails in his folder. The mwi contact is correct, the voicemail.conf looks correct, I've deleted and recreated the voicemail box folder.... |
| 22:40.57 | mercestes | IT's setup the same as every other phone here....latest firmware. Asterisk 1.0.9 in this particular instance.... |
| 22:41.01 | mercestes | why is it screwing up? |
| 22:41.01 | hads | and their site doesn't resolve from here... |
| 22:41.25 | Qwell[] | mercestes: because it's 1.0.9 :p |
| 22:41.52 | ellisdee | lol |
| 22:43.04 | mercestes | Qwell[]: Workign on the upgrade but my HR is screaming at me to "fix his phone." |
| 22:43.18 | ellisdee | http://www.atcom.cn/En_products_AG188.html |
| 22:43.20 | ellisdee | you used that ata? |
| 22:43.23 | mercestes | Qwell[]: Any other clues?? |
| 22:43.25 | ellisdee | it supports sip/iax |
| 22:43.46 | ellisdee | mercestes, check sip.cfg |
| 22:43.55 | ellisdee | mercestes, make sure he is pointd to the appropriate mailbox |
| 22:44.11 | mercestes | elisdee: Hmm...didn't check that..figured it'd be wmi in the polycom config file..lemme check |
| 22:44.11 | ellisdee | mercestes, recheck voicemail.conf make sure you have appropriate nfo set in there for that ext as well |
| 22:44.35 | ellisdee | /etc/asterisk/sip.cfg |
| 22:44.39 | mercestes | Holy cow....ellisdee....... |
| 22:44.51 | mercestes | Didn't you know that you were my hero.........you are everything I want to be...... |
| 22:44.51 | ellisdee | all the peers reside in that file |
| 22:44.59 | mercestes | I can fly higher than an eagle........you are the wind beneath my wings. |
| 22:45.06 | ellisdee | lol |
| 22:45.12 | icel | how romantic |
| 22:45.13 | mercestes | Thank you! :) |
| 22:45.18 | ellisdee | np |
| 22:45.25 | ellisdee | paypal ellisdee@paypal.com kthx |
| 22:45.29 | implicit | hahaha |
| 22:45.30 | implicit | no |
| 22:45.34 | ellisdee | jk ;P |
| 22:45.49 | implicit | if u knew what u were talking about maybe :-) |
| 22:45.50 | implicit | ;) |
| 22:45.52 | implicit | heheh |
| 22:46.08 | implicit | jk |
| 22:46.11 | ellisdee | :( |
| 22:46.14 | ellisdee | i try |
| 22:46.25 | ellisdee | im new to asterisk. |
| 22:46.27 | ellisdee | =P |
| 22:47.27 | mercestes | lmao |
| 22:47.41 | mercestes | You rock, Ellisdae...don't let these jokers tell you differently. |
| 22:47.52 | ellisdee | ;) |
| 22:47.52 | mercestes | man, 4888 and 4880 look alot alike in vim in a sip.conf too. |
| 22:50.28 | aosdonovan | I am having trouble getting calls to go to sip ata's consistently. It is definelty a NAT issue. Is there a stun server that outperforms other stun servers? |
| 22:51.41 | brad_mssw | anyone got the new SPA-3102NA (which is supposedly replacing the SPA-3000) ? Wondering if there are any limitations over the SPA-3000 |
| 22:54.15 | aosdonovan | I have many spa-3000, havent seen that model yet |
| 22:54.22 | aosdonovan | is there any improvements? |
| 22:54.39 | brad_mssw | that's why I was asking :/ information seems short on it |
| 22:55.00 | *** join/#asterisk Peaceful (n=Peaceful@70.98.162.62) |
| 22:55.18 | Peaceful | stupid wireless internet... |
| 22:55.34 | *** join/#asterisk CharlesR (n=charlesr@cpe-76-188-71-88.neo.res.rr.com) |
| 22:56.06 | Peaceful | let's try this again: Is it common to have two mpg123 processes sucking 100% CPU each (even when no calls are on the box)? |
| 22:56.09 | Peaceful | I'm using asterisk 1.2.10 and mpg123 0.59s-r11 ... if that makes any difference |
| 22:56.25 | aosdonovan | has 2 rj45 connections... |
| 22:56.43 | [TK]D-Fender | Peaceful : I guess you missed my response.... |
| 22:56.46 | [TK]D-Fender | Peaceful : You must use only mpg123 .059r |
| 22:56.50 | [TK]D-Fender | Peaceful : Then again, I suggest avoiding mpg123 altogether and using Native MoH |
| 22:58.44 | aosdonovan | The 3102 has a built in router and extra memory space according to another forum |
| 23:02.10 | Peaceful | [TK]D-Fender: yes, I missed the response. We're having intermittent access problems from where I am right now. |
| 23:02.25 | Peaceful | [TK]D-Fender: Thanks for the response, I'll go downgrade to .059r |
| 23:02.53 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-86-116.lsanca.dsl-w.verizon.net) |
| 23:03.02 | hads | aosdonovan: Yah, apart from that they are similar. |
| 23:03.17 | hads | More memory for more features to be added to the firmware. |
| 23:03.20 | Peaceful | [TK]D-Fender: do you have any idea if we'll be able to use future versions? (like .061?) |
| 23:05.02 | CunningPike | Peaceful: Forget mpg123 and use native moh |
| 23:05.06 | CunningPike | ~moh |
| 23:05.07 | jbot | [moh] Music On Hold. Good information about how to set it up in the various possible ways can be found at http://www.voip-info.org/wiki/index.php?page=Asterisk+config+musiconhold.conf |
| 23:06.04 | [TK]D-Fender | Peaceful : I highly doubt it. You should be using Native MoH anyways and forego external programs. |
| 23:07.27 | Peaceful | CunningPike: [TK]D-Fender: Ok, I never made any explicit choice to use mpg123 anyways back when I set up this server. I'll investigate migroting to native MoH. |
| 23:07.46 | CunningPike | Peaceful: It's really simple |
| 23:08.33 | dasenjo | I like madplay for playing mp3 moh |
| 23:11.10 | Peaceful | so...my configs don't say anything about mpg123 OR native MoH |
| 23:11.28 | Peaceful | ...and I notice it requires asterisk-addons, which I don't think I have installed |
| 23:11.47 | Peaceful | If I install asterisk-addons and restart asterisk, would that do it? |
| 23:13.28 | [TK]D-Fender | Peaceful : And change your mode line to "files" |
| 23:15.07 | Peaceful | Ok, so mode=files, directory=/what/ever, and no application= line. |
| 23:15.16 | Peaceful | and install asterisk-addons, I suppose |
| 23:16.54 | *** join/#asterisk CyberKnet (n=CyberKne@ip68-13-246-61.ok.ok.cox.net) |
| 23:19.25 | Peaceful | hmm... |
| 23:19.39 | Peaceful | I installed asterisk-addons, changed mode to files, and did a 'restart now' |
| 23:19.52 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 23:20.01 | *** join/#asterisk ManxPower (n=manxpowe@71-8-11-73.dhcp.leds.al.charter.com) |
| 23:20.05 | Peaceful | ...and now I have THREE mpg123 process hogging 100% processor |
| 23:20.08 | mercestes | try a full shutdown and restart. |
| 23:20.19 | mercestes | and uninstall mpg123 >.> |
| 23:20.21 | *** part/#asterisk ManxPower (n=manxpowe@71-8-11-73.dhcp.leds.al.charter.com) |
| 23:20.23 | Peaceful | heh |
| 23:20.27 | *** join/#asterisk ManxPower (n=manxpowe@71-8-11-73.dhcp.leds.al.charter.com) |
| 23:20.33 | mercestes | and shake a chicken wing over it for good measure. |
| 23:20.46 | Peaceful | (sigh) there's three calls going on now |
| 23:20.58 | Peaceful | It's rare I get a time with no one on like I just saw |
| 23:21.34 | dasenjo | Peaceful, stop when convenient and keep on eye on the server |
| 23:21.38 | mercestes | unload chan_zap.so && unload chan_sip.so && unload chan_iax2.so && unload chan_capi.so |
| 23:21.42 | mercestes | that tends to help clear out a server. |
| 23:22.06 | Peaceful | would that drop current calls? |
| 23:22.21 | mercestes | peaceful: yes....but stop when convienient wouldn't. |
| 23:22.30 | mercestes | peaceful: So go with dasenjo's plan. |
| 23:22.45 | Peaceful | I'll give it a try. Thanks, guys. |
| 23:26.47 | *** join/#asterisk ten_novals (n=mood@87.255.1.74) |
| 23:26.51 | ten_novals | hi all =) |
| 23:27.04 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 23:27.06 | mercestes | hi, ten-novals. |
| 23:27.19 | ten_novals | May i ask about 1 question... intresting this bug or feature? =) |
| 23:27.34 | mercestes | you just burnt your 1 question, sorry. |
| 23:27.49 | ten_novals | =))) |
| 23:27.51 | mercestes | but you can ask another one because I'm feeling giving today. |
| 23:28.30 | ten_novals | if i set call-limit and user use line - why Subscribe status - Idle? |
| 23:28.52 | mercestes | ??? |
| 23:29.01 | ten_novals | i set call-limit=1 |
| 23:29.17 | mercestes | call-limit and Subscribe status shouldn't have much to do with each other AFIAK. |
| 23:31.05 | ten_novals | hmm.. |
| 23:31.33 | CunningPike | No-one has heard of imap_tk? |
| 23:36.28 | *** join/#asterisk metafore (n=metafore@adsl-75-18-238-125.dsl.pltn13.sbcglobal.net) |
| 23:48.23 | *** join/#asterisk aptura (n=sales@S010600a0c93f6f7e.vs.shawcable.net) |