| 00:00.11 | oconshaw | I understand that I need to connect to a telco. I just want to make sure I have a solution that works. |
| 00:00.45 | oconshaw | POTS is analog -- that is where I started from in this dialog as I wish to have a solution that I know for sure will work. |
| 00:01.28 | ManxPower | oconshaw: You do understand that Asterisk is not a PBX right? |
| 00:01.30 | mosty | oconshaw, compare the cost of four analogue lines vs BRI isdn |
| 00:01.34 | oconshaw | As I undertand it: ISDN BRI, ISDN PRI can be used for regular dialup without VOIP right? Just a better signal that POTS. |
| 00:01.41 | ManxPower | It's a telephony toolkit that will allow you to build a PBX |
| 00:02.36 | oconshaw | ManxPower: Actually, no. My understanding was that Asterisk had that capability. |
| 00:02.40 | luke-jr_work | ManxPower, last I checked, Asterisk was advertised as a PBX =p |
| 00:02.43 | ManxPower | oconshaw: What country are you in? |
| 00:02.52 | oconshaw | US, TX |
| 00:03.06 | ManxPower | oconshaw: then don't even think about BRI ISDN |
| 00:03.15 | oconshaw | :) why? |
| 00:03.31 | *** join/#asterisk eltech (i=G00Ds@ool-457c9421.dyn.optonline.net) |
| 00:03.37 | ManxPower | oconshaw: because there is no support for USA ISDN BRI |
| 00:03.44 | luke-jr_work | oconshaw, never heard of ISDN in the US... |
| 00:03.45 | oconshaw | I suspect the cost is prohibitive. |
| 00:04.04 | oconshaw | Of course there is ISDN in the US. |
| 00:04.05 | ManxPower | If you hear of someone using BRI with Asterisk then they are not in the USA or Canada |
| 00:04.28 | diclophis-work | so... is it possible for the dialplan to go past a Hangup()? |
| 00:04.34 | luke-jr_work | oconshaw, news to me |
| 00:04.53 | ManxPower | diclophis-work: why? |
| 00:05.05 | oconshaw | OK, well anyway -- I'm not a big fan of ISDN so if we don't need to do ISDN to make this work then great. |
| 00:05.18 | luke-jr_work | oconshaw, why do you need analog? |
| 00:05.50 | ManxPower | oconshaw: ISDN PRI is the best, most supported, easiest to manage, PSTN connection for Asterisk |
| 00:06.41 | oconshaw | Well... maybe I don't. My thought was to start off with asterisk running as a PBX to handle regular pots lines and ALSO voip. I want a solution I know will work (hence the analog), but would like to go exclusively with the VOIP once I'm confident of the signal, etc. |
| 00:06.57 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 00:06.57 | *** mode/#asterisk [+o anthm] by ChanServ |
| 00:07.33 | *** join/#asterisk k-man__ (n=jason@ppp244-232.static.internode.on.net) |
| 00:08.05 | oconshaw | ManXPower, I hate to say this but what is ISDN PRI and PSTN? |
| 00:09.01 | mosty | PSTN is the regular telephone network. ISDN PRI is a digital connection to the PSTN |
| 00:09.20 | oconshaw | OH... |
| 00:09.28 | doug | ok |
| 00:09.30 | doug | http://ast.con.com |
| 00:09.35 | *** join/#asterisk angryuser (n=uk@i03v-213-44-169-43.d4.club-internet.fr) |
| 00:09.41 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 00:09.43 | doug | so, who wants the job? |
| 00:09.50 | doug | oh wait |
| 00:10.48 | oconshaw | I'll look into the ISDN PRI -- see what the cost is vs. the 4 POTS lines. |
| 00:11.34 | oconshaw | I have a feeling the ISDN Pri is going to be much more expensive. How would you do what I'm asking with the POTS lines and VOIP through DSL? |
| 00:12.00 | diclophis-work | also does anyone know what this means? Unexpected control subclass '14' |
| 00:12.14 | *** part/#asterisk vbasicguru (n=vbasicgu@201.255.163.219) |
| 00:12.42 | doug | okauy, now reload http://ast.con.com |
| 00:12.44 | doug | got some pix now |
| 00:13.22 | doug | Ubuntu/dapper-security |
| 00:13.23 | doug | nice |
| 00:14.22 | doug | X11; U; Linux i686; |
| 00:14.23 | diclophis-work | i am getting same crazy funky happenings |
| 00:14.25 | doug | who's that? good man. |
| 00:14.42 | doug | good american, at least. |
| 00:14.52 | doug | er, no. argentinian |
| 00:14.55 | doug | en-US user. |
| 00:15.22 | doug | zazzizza... |
| 00:16.04 | zazzizza | doug what's up? |
| 00:16.28 | *** join/#asterisk hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net) |
| 00:18.57 | doug | gonna take a shot at my dysfunction? |
| 00:19.12 | zazzizza | is it dead? |
| 00:19.19 | zazzizza | just curious |
| 00:19.55 | zazzizza | i had an issue myself today, just glad it is now sorted out. made me sick |
| 00:20.04 | diclophis-work | and is this normal ? returned normally even though call was hung up |
| 00:20.22 | diclophis-work | Extension conference, priority 4 returned normally even though call was hung up |
| 00:20.23 | diclophis-work | that is |
| 00:20.29 | *** join/#asterisk JustinWick (n=jwick@unaffiliated/jpl/jpl-justin) |
| 00:23.25 | ManxPower | diclophis-work: a hangup is normal |
| 00:24.18 | ManxPower | diclophis-work: What is the specific thing that is not working? |
| 00:24.25 | diclophis-work | i dont know |
| 00:24.33 | diclophis-work | i got this shit all crazy fucked up |
| 00:24.42 | diclophis-work | local channels with one leg in a conference and the other leg in an AGI |
| 00:26.12 | diclophis-work | technically two calls with the first being a local channel thats stuck in a conference, and the second pointed at an AGI thats running. then the second call in the same manner, but setup to dial through the agi, afterwhich the first call with make it through the agi, and then switch to a second agi, while hanging up the local channel, and then moving the AGI channel into the conference room |
| 00:27.40 | diclophis-work | and for some reason the AGI is fucking up on a STREAM FILE thats not returning -1 or crapping out when a call hangsup |
| 00:29.26 | zazzizza | is iaxtel back online? |
| 00:29.37 | diclophis-work | oh damnit, does asterisk 1.4 return -1 for the 'result' instead of the 'offset' when doing a streamfile on a hungup channel? |
| 00:43.08 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 00:43.12 | *** join/#asterisk CyberKnet (n=CyberKne@ip68-13-246-61.ok.ok.cox.net) |
| 00:45.10 | *** join/#asterisk CyberKnet (n=CyberKne@ip68-13-246-61.ok.ok.cox.net) |
| 00:45.38 | diclophis-work | finally figured it ou |
| 00:45.39 | diclophis-work | t |
| 00:45.45 | CyberKnet | bkw_: Is asterlink offering LNP and E911 in Tulsa yet? |
| 00:48.35 | *** join/#asterisk knarfly (n=knarfly@c-65-34-177-3.hsd1.fl.comcast.net) |
| 00:48.54 | knarfly | anyone running * on FC5-x86_64? |
| 00:52.12 | *** join/#asterisk sysreq (n=sysreq@142-217-128-85.telebecinternet.net) |
| 00:59.00 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 01:00.40 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 01:01.44 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 01:02.07 | knarfly | Anyone using FC5 ... I'm having trouble with zaptel build? |
| 01:03.10 | *** join/#asterisk DasTech (n=DasTech@ppp-71-128-113-209.dsl.irvnca.pacbell.net) |
| 01:03.45 | zazzizza | knarfly: what's going wrong? |
| 01:03.52 | DasTech | ok anyone here get the nv fax to compile on 1.4 ? |
| 01:04.03 | DasTech | or the spandsp fax ? |
| 01:04.31 | knarfly | zaptel wants the sources for the kernel ... but I can't get the sources for 2.6.15 |
| 01:04.49 | DasTech | yes you can |
| 01:04.56 | DasTech | kernel.org |
| 01:04.56 | zazzizza | why not? can't yum them? |
| 01:04.58 | knarfly | if I do yum install kern-devel it starts loading sources for 2.6.18 |
| 01:05.07 | zazzizza | go to kernel.org |
| 01:05.35 | knarfly | I installed 2.6.18 kernel and the machine went bonkers...I don't understand why so I'm trying to stick with 2.6.15 |
| 01:05.49 | DasTech | centos or ? |
| 01:05.56 | zazzizza | i guess this "easyness" of apt-get & yum is a drawback... |
| 01:06.02 | knarfly | DasTech, how do I do this? |
| 01:06.20 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 01:06.30 | DasTech | centos or other linux ? |
| 01:06.41 | *** join/#asterisk juice (i=1000@mo-76-0-46-199.dhcp.embarqhsd.net) |
| 01:06.47 | knarfly | DasTech, FC5-x86_64 ? But I prefer FreeBSD |
| 01:06.53 | zazzizza | go to kernel.org download your "kernel version", untar i /usr/src and make symlink as linux |
| 01:06.56 | zazzizza | that's it (a) |
| 01:07.03 | DasTech | asterisk works on bsd |
| 01:07.20 | DasTech | I have 1.4 in 6.2-pre |
| 01:07.21 | knarfly | DasTech, yes, quite well I might add... |
| 01:07.51 | knarfly | zazzizza, I'll try that but I don't think I follow the link part? |
| 01:07.59 | DasTech | I just cant figure out how to add apps to the 1.4 |
| 01:08.49 | zazzizza | knarfly: the sources will probably be untarred as linux-2.6....whatever |
| 01:09.24 | zazzizza | but to compile i guess you need to simply do it relative to /usr/src/linux |
| 01:09.43 | knarfly | zazzizza, kernel.org does not have 2.6.15, only 2.6.18 and higher |
| 01:09.53 | zazzizza | on /usr/src just do: ln -sf <the directory where the newly kernel is untarred...> linux |
| 01:09.54 | zazzizza | that's it |
| 01:10.07 | DasTech | well 2.6.18 is fine |
| 01:10.28 | knarfly | DasTech, even if I'm running 2.6.15 ? |
| 01:10.51 | DasTech | just make a new kernel |
| 01:10.57 | knarfly | zazzizza, that gives me an idea...be back later after I try a few things? |
| 01:11.04 | DasTech | install it reboot build zaptel and go |
| 01:11.26 | zazzizza | http://www.kernel.org/pub/linux/kernel/v2.6/linux-2.6.16.15.tar.gz |
| 01:11.38 | knarfly | DasTech, I'd love to except when I installed 2.6.18 it whacked my computer...lost NIC, lost DNS, ouch...had to rebuild to clean it up. |
| 01:11.56 | zazzizza | sorry |
| 01:11.58 | zazzizza | http://www.kernel.org/pub/linux/kernel/v2.6/linux-2.6.15.1.tar.bz2 |
| 01:12.33 | knarfly | zazzizza, I have to ask...I'm running x86_64...will this be the right kernel ? |
| 01:12.58 | DasTech | well back to work on getting this thing to compile nv and spandsp fax and then hunt down other needed modules |
| 01:13.18 | knarfly | DasTech, good luck and thanks.... |
| 01:13.45 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 01:14.21 | DasTech | ok np |
| 01:14.33 | DasTech | break time first |
| 01:27.46 | tzafrir | usually there is no need to generate /usr/src/linux link . Usually you will have the right link in /lib/modules/VERSION/build |
| 01:28.17 | tzafrir | Both with distro kernels and with properly installed self-built ones |
| 01:28.37 | iCEBrkr | tzafrir: In my experience, I've always had to symlink the kernel source to /usr/src/linux |
| 01:28.47 | iCEBrkr | tzafrir: But I'm a RedHat/Fedora fan :P |
| 01:28.52 | iCEBrkr | ok, maybe not a fan... |
| 01:29.02 | iCEBrkr | But it's what I grew up on and what I'm comfortable with. |
| 01:29.05 | zazzizza | hah ok |
| 01:29.31 | tzafrir | iCEBrkr, zaptel makefile looks at the "build" symlink first. And that link is usually generated/installed automatically |
| 01:29.37 | iCEBrkr | knarfly: You haven't gotten your problem straightened out?? |
| 01:29.41 | zazzizza | anyway, no offence but sign me up for the rh/fedora hater's list |
| 01:29.47 | iCEBrkr | tzafrir: ahh. |
| 01:30.00 | iCEBrkr | zazzizza: I hate them all really. I just know RH like the back of my hand. |
| 01:30.05 | tzafrir | iCEBrkr, if you use a distro-kernel there, that link is provided by the kernel-headers package on redhats |
| 01:30.16 | zazzizza | :-D |
| 01:30.22 | iCEBrkr | tzafrir: Yea, that just makes a Redhat directory, tho, no |
| 01:30.23 | iCEBrkr | ? |
| 01:30.29 | tzafrir | (and by kernel-headers/linux-headers on debians) |
| 01:30.59 | knarfly | iCEBrkr, no, I had to rebuild again...it's a pain...I may be on to something here that zazzizza suggested |
| 01:31.09 | tzafrir | iCEBrkr, it points to /usr/src/kernel-headers-VERSION or something similar |
| 01:31.10 | iCEBrkr | Ya know, I think you're right. I haven't had to do anything special once I installed kernel-devel and headers... |
| 01:31.24 | iCEBrkr | knarfly: Ahh, good, making progress then? |
| 01:31.45 | iCEBrkr | tzafrir: my machines are just so old, i forget :P |
| 01:31.47 | tzafrir | right. kernel-devel , not kernel-headers (on redhats) . kernel-headers is from older debian... |
| 01:31.59 | *** join/#asterisk NDT (n=noone@cpe-74-70-211-81.nycap.res.rr.com) |
| 01:32.02 | knarfly | iCEBrkr, well I wouldn't call it progress...three evenings now and all I have is FC5-x86_64 but no * |
| 01:32.09 | iCEBrkr | knarfly: :( |
| 01:32.16 | iCEBrkr | knarfly: You're learning tho, right? :D |
| 01:32.30 | iCEBrkr | learning you need more beer! |
| 01:32.51 | knarfly | iCEBrkr, learning that like Dire Straits.....I want my BSD ! |
| 01:32.59 | iCEBrkr | knarfly: I wish I had a 64bit machine, I'd try and walk ya through it |
| 01:33.04 | iCEBrkr | hahaha |
| 01:33.15 | mosty | if i want to receive sip calls from a server, i have a register line in sip.conf, and a user section which sends those calls to a particular context in my dialplan, but how does asterisk know which user section to use? |
| 01:33.28 | *** part/#asterisk NDT (n=noone@cpe-74-70-211-81.nycap.res.rr.com) |
| 01:33.41 | iCEBrkr | mosty: user section? you mean context? |
| 01:33.55 | knarfly | iCEBrkr, that seems to be the problem...I know I could run i386 and it would probably resolve but since this is an AMD64 I wanted to take advantage of the 64 bit stuff. |
| 01:34.07 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 01:34.13 | iCEBrkr | knarfly: Agreed |
| 01:34.27 | mosty | iCEBrkr, i'm talking about sip.conf |
| 01:34.56 | iCEBrkr | mosty: Ok, you have a register line in there to accept calls from some provider or whatever.. right? |
| 01:34.58 | joe | is there a place where you don't have to log in to get polycom boot roms and stuff? |
| 01:35.17 | *** join/#asterisk olinux (n=olinux@starbucks.wellspublishing.net) |
| 01:35.25 | joe | I have a rev F 301 that doesn't like the version on there ... |
| 01:35.33 | joe | on my ftp server that is |
| 01:35.41 | iCEBrkr | mosty: Then you need a context for that provider. |
| 01:35.57 | mosty | iCEBrkr, yes, but i don't understand how asterisk knows which context to use for that provider |
| 01:36.11 | iCEBrkr | joe: Send me one of your Polycoms, and I'll figure it out for ya :D (j/k) |
| 01:37.03 | [TK]D-Fender | joe : http://www.freedomphones.net/polycom/files/ |
| 01:37.34 | iCEBrkr | mosty: damn man, you're making me think about this.. I haven't mucked with my asterisk config in 3yrs. |
| 01:38.03 | joe | iCEBrkr: :P suuuuurrrrre |
| 01:38.09 | iCEBrkr | joe: hehehe |
| 01:38.09 | joe | [TK]D-Fender: thank you much! |
| 01:38.15 | [TK]D-Fender | joe : ywc |
| 01:38.25 | *** join/#asterisk DHuang (n=DHuang@mail.medec.com.au) |
| 01:39.25 | iCEBrkr | mosty: If I recall correctly, the register line is so you can make outbound calls... While the [context] are for inbound calls. |
| 01:39.38 | iCEBrkr | mosty: you're dialplan decides what to do with the call and where to send it |
| 01:39.40 | iCEBrkr | err your too |
| 01:39.53 | *** join/#asterisk NEZPERCECOUNTY (n=Nez@co.nezperce.id.us) |
| 01:40.11 | NEZPERCECOUNTY | Can I get some purchasing advice here? |
| 01:40.15 | mosty | iCEBrkr, yes but somewhere in sip.conf has to say in which context to start in the dialplan |
| 01:40.23 | *** join/#asterisk asouza (n=araujo@200-193-76-25.fnsce7006.dsl.brasiltelecom.net.br) |
| 01:40.50 | *** join/#asterisk Tili (n=tili@202.133.65.49) |
| 01:42.10 | olinux | nez just ask |
| 01:42.28 | [TK]D-Fender | iCEBrkr : No, the register line is just to tell your provider what IP & exten to send calls to. It does not handle anything else. |
| 01:42.53 | [TK]D-Fender | NEZPERCECOUNTY : Buy low, sell high! |
| 01:43.14 | iCEBrkr | [TK]D-Fender: Well it tells them you're 'on-line' and taking calls.. no? |
| 01:43.16 | NEZPERCECOUNTY | LOL |
| 01:43.34 | iCEBrkr | ok, so I had it backwards |
| 01:44.02 | [TK]D-Fender | iCEBrkr : Only if they use that information to send qualify packets at you. and well... its your system's continued responses that would indicate that, not the initial register |
| 01:44.34 | NEZPERCECOUNTY | We have a PBX already that is getting old; we are going to upgrade and I suggested Asterisk with a Diguim PRI card. What works best for a medium buinesses? |
| 01:44.40 | iCEBrkr | [TK]D-Fender: ahh |
| 01:44.41 | joe | [TK]D-Fender: ywc? |
| 01:44.52 | iCEBrkr | joe: You're welcome |
| 01:45.03 | joe | duh |
| 01:45.05 | joe | thanks |
| 01:45.08 | iCEBrkr | hehe |
| 01:45.11 | [TK]D-Fender | NEZPERCECOUNTY : define "medium", and when you say "works best", what aspect are you asking about? |
| 01:45.28 | asouza | Hi, somebody users Asterisk in the FreeBSD? |
| 01:45.31 | iCEBrkr | NEZPERCECOUNTY: hehe, Sangoma and a PRI :P |
| 01:45.53 | iCEBrkr | asouza: knarfly does |
| 01:45.56 | [TK]D-Fender | joe : If I don't STFU with all the acronyms, I'll be DOA PDQ, AOK? :) |
| 01:46.03 | NEZPERCECOUNTY | We have 250 end-users. We have two PRI connections currently going to our ancient PBX. We make no more than 30 sim phone calls. |
| 01:46.07 | iCEBrkr | [TK]D-Fender: lol |
| 01:46.33 | iCEBrkr | NEZPERCECOUNTY: That's kinda crazy.. 250 users and no more than 30 calls at a time? |
| 01:46.35 | asouza | iCEBrkr FreeBSD and Asterisk function very fine ? |
| 01:46.37 | knarfly | asouza, Yes...it works great... |
| 01:47.01 | [TK]D-Fender | iCEBrkr : I presume that would be 30 calls to PSTN |
| 01:47.08 | NEZPERCECOUNTY | That's what our Phone Company reports. |
| 01:47.24 | iCEBrkr | [TK]D-Fender: Yea, but that seems low for 250 people |
| 01:47.39 | asouza | knarfly Using FreeBSD and Asterisk inside server production? |
| 01:47.44 | NEZPERCECOUNTY | Okay let's assume my two pri's are maxed |
| 01:47.44 | JT | really depends on the sort of company |
| 01:47.58 | iCEBrkr | JT: Yea, I suppose. :P |
| 01:48.23 | knarfly | asouza, please clarify...I'm not sure what you're asking |
| 01:48.30 | JT | if those 250 people are all making stuff in a factory, it's hardly surprising |
| 01:48.37 | NEZPERCECOUNTY | Well I just want a little bit of direction before I make a final purchase. I figured what better place to ask. |
| 01:49.01 | NEZPERCECOUNTY | Which card should I buy? |
| 01:49.14 | NEZPERCECOUNTY | Can I get a PCI-X card? |
| 01:49.16 | iCEBrkr | NEZPERCECOUNTY: Sangoma |
| 01:49.17 | NEZPERCECOUNTY | Does it help? |
| 01:49.28 | NEZPERCECOUNTY | Not Diguim? |
| 01:49.28 | iCEBrkr | Oops, I said a bad word |
| 01:49.32 | NEZPERCECOUNTY | Hee hee |
| 01:49.49 | NEZPERCECOUNTY | Sangoma was on my list of possiblities. |
| 01:49.56 | iCEBrkr | NEZPERCECOUNTY: If your proposed server is on Digiums 'supported list' you may go with Digium :P |
| 01:50.22 | iCEBrkr | But I was dialing for dollars with a quad-port Sangoma card on a crappy dell 1u and it was fine. |
| 01:50.26 | asouza | knarfly Sorry, You use FreeBSD and Asterisk for simple tests or use FreeBSD to finally users ? |
| 01:50.30 | iCEBrkr | no IRQ issues, no spikes in CPU, etc, etc. |
| 01:50.49 | NEZPERCECOUNTY | *Nods* |
| 01:51.08 | NEZPERCECOUNTY | What is 'wrong' with Diguim cards? |
| 01:51.22 | iCEBrkr | NEZPERCECOUNTY: I hear they still have IRQ and compatibility issues |
| 01:51.28 | NEZPERCECOUNTY | From my research Sagnoma's can't share 'D-Channel' signalling? |
| 01:51.46 | olinux | is there an 'official' gui (web management) |
| 01:51.51 | JT | interesting |
| 01:51.53 | iCEBrkr | NEZPERCECOUNTY: Digium cards supposedly spew HDLC errors |
| 01:51.57 | JT | that's called NFAS, btw, NEZPERCECOUNTY |
| 01:51.58 | knarfly | asouza, I use it at my home...I'm only a hobbiest...I speak to others though who use it in production and commercial environments. Check out the Asterisk-BSD forum at http://asteriskpbx.org |
| 01:52.16 | NEZPERCECOUNTY | Thanks JT. |
| 01:52.29 | iCEBrkr | olinux: join #asterisk-gui ??? |
| 01:52.32 | asouza | knarfly OK, thanks... |
| 01:52.49 | olinux | ok :) |
| 01:53.57 | NEZPERCECOUNTY | Can I buy the SANGOMA A104 with 64-bit bus? (PCI-X)? |
| 01:54.22 | NEZPERCECOUNTY | (I'm assume 64-bit bus can only help) |
| 01:54.24 | iCEBrkr | NEZPERCECOUNTY: Check out their site :P |
| 01:54.31 | NEZPERCECOUNTY | LOL |
| 01:54.49 | iCEBrkr | NEZPERCECOUNTY: I dunno, a lot of people thought SMP would help, and there's reports that SMP is detrimental to an Asterisk system |
| 01:54.52 | [TK]D-Fender | NEZPERCECOUNTY : if you only plan on 2 PRI, then get the A102d instead. |
| 01:55.09 | NEZPERCECOUNTY | VoipSupply - "PCI 32 bit (5v and 64 bit (3.3v) compatible" |
| 01:55.37 | NEZPERCECOUNTY | I had a thought. |
| 01:55.50 | NEZPERCECOUNTY | Besides the extra echo cancellation - I thought I might m |
| 01:55.55 | NEZPERCECOUNTY | migrate off my current PBX |
| 01:55.56 | *** join/#asterisk Telamon (i=telamon@blk-222-23-213.eastlink.ca) |
| 01:55.59 | JT | i don't know of any 64bit pci pri cards for asterisk |
| 01:56.01 | NEZPERCECOUNTY | 2 pri in, 2 pri out |
| 01:56.05 | JT | 32bits is ample bandwidth |
| 01:56.10 | *** part/#asterisk asouza (n=araujo@200-193-76-25.fnsce7006.dsl.brasiltelecom.net.br) |
| 01:56.16 | *** part/#asterisk Telamon (i=telamon@blk-222-23-213.eastlink.ca) |
| 01:56.50 | [TK]D-Fender | NEZPERCECOUNTY : Howabout we look at this from a few steps back. What are the reasons for your migration? |
| 01:56.54 | iCEBrkr | NEZPERCECOUNTY: Echo cancelling == Must have! |
| 01:57.30 | NEZPERCECOUNTY | Okay. Any recommendations for Hardware? |
| 01:57.42 | NEZPERCECOUNTY | (I was thinking SuperMicro) |
| 01:57.48 | NEZPERCECOUNTY | Duo etc... |
| 01:57.59 | JT | brandname with redundant everything |
| 01:58.05 | JT | does not need the latest clockspeeds |
| 01:58.05 | NEZPERCECOUNTY | LOL |
| 01:58.16 | JT | does not even need to be new |
| 01:58.29 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 01:58.37 | NEZPERCECOUNTY | (*coughs* I also want video later *coughs*) |
| 01:58.54 | JT | what sort of video? |
| 01:59.26 | JT | but if you deal with 2 pris for 250 users with non-redundant hardware, you're insane :P |
| 01:59.34 | iCEBrkr | JT: Yea, really. my crappy dell 1u was only a 1Ghz I think? Maybe it was 800mhz |
| 01:59.59 | JT | iCEBrkr: which is sufficient for a lot of applications |
| 02:00.20 | [TK]D-Fender | NEZPERCECOUNTY : Again, what are you migrating FROM, and why? |
| 02:00.36 | *** join/#asterisk hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net) |
| 02:00.54 | hmmhesays | anyone remember how to pass arguments to a script in perl? |
| 02:01.31 | iCEBrkr | holyshit. knarfly is near by me. |
| 02:02.47 | zazzizza | hmmhesays just pass is it along you need to get it by $ARG[] |
| 02:07.23 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:08.45 | tzafrir | hmmhesays, [perl] script arg1 arg2 arg3 |
| 02:09.25 | tzafrir | hmmhesays, @ARGV will have (arg1 arg2 arg3) |
| 02:10.57 | DasTech | ok how do I find app_*.c files that are addons to asterisk I do a google and find nada |
| 02:11.12 | *** join/#asterisk slayer192 (n=slayer19@adsl-70-137-24-211.dsl.okcyok.swbell.net) |
| 02:12.25 | DasTech | did app_curl get dropped ? |
| 02:12.41 | DasTech | I cant find a current.c file for it |
| 02:13.39 | *** join/#asterisk _DAW (n=_DAW@adsl-157-54-57.msy.bellsouth.net) |
| 02:13.58 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 02:14.20 | hmmhesays | tzafrir something is wrong here |
| 02:14.40 | tzafrir | what? |
| 02:14.51 | *** join/#asterisk CyberKnet2 (n=CyberKne@ip68-13-246-61.ok.ok.cox.net) |
| 02:14.53 | tzafrir | a perl script that runs where? |
| 02:17.42 | Supaplex | is it just far fetched/paranoia that someone can+would spoof a caller id number while using dialup? And no, the NAS/POP in question isn't on VoIP. (real T1/PRI etc) |
| 02:23.19 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 02:24.03 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 02:24.53 | *** join/#asterisk _Vile (i=vile@bc181104.bendcable.com) |
| 02:28.02 | *** join/#asterisk asdx (n=diego@200.61.236.33) |
| 02:32.37 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 02:35.47 | *** join/#asterisk hieunm_vips (n=hieunm_v@210.245.57.162) |
| 02:37.10 | hieunm_vips | hi all |
| 02:37.23 | hieunm_vips | I'm testing Read function on Asterisk 1.2.12.1 |
| 02:37.36 | hieunm_vips | but it does not run right |
| 02:38.31 | iCEBrkr | define, not right? |
| 02:38.37 | hieunm_vips | no |
| 02:38.49 | Supaplex | what are you expecting, and what do you see? |
| 02:38.49 | hieunm_vips | I will copy here for you can check |
| 02:38.59 | iCEBrkr | noo |
| 02:38.59 | Supaplex | no flooding :) |
| 02:39.02 | iCEBrkr | ~pb |
| 02:39.06 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 02:39.09 | hieunm_vips | exten => 1001, 1, DigitTimeout(2) |
| 02:39.12 | hieunm_vips | yes |
| 02:39.20 | hieunm_vips | I'm a newbie |
| 02:39.49 | hieunm_vips | Wait me for a moment |
| 02:39.51 | Supaplex | 2 second timeout isn't much :) |
| 02:40.27 | [TK]D-Fender | Supaplex : Wait till he gives you more rope before you hang him.... |
| 02:40.38 | [TK]D-Fender | hieunm_vips : Pastebin the whole thing please. |
| 02:40.42 | [TK]D-Fender | ~pb |
| 02:40.44 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 02:41.10 | *** join/#asterisk jeebusmobile (n=jeebusmo@64.73.243.162) |
| 02:43.41 | hieunm_vips | http://pastebin.ca/243160 |
| 02:44.17 | hieunm_vips | I pasted as you recommended |
| 02:44.26 | hieunm_vips | could you take a look ? |
| 02:44.51 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 02:48.05 | [TK]D-Fender | hieunm_vips : Try doing 5 digits and see if it exits properly on the 4rd |
| 02:48.32 | *** join/#asterisk belamiru (n=mike@modemcable111.97-131-66.mc.videotron.ca) |
| 02:48.48 | hieunm_vips | yes, thank, I will try immediately |
| 02:49.52 | [TK]D-Fender | 4th* |
| 02:51.39 | hieunm_vips | the same problem still occurred |
| 02:52.07 | [TK]D-Fender | hieunm_vips : did it exit on the 4th digit immediately? |
| 02:52.23 | hieunm_vips | no |
| 02:52.36 | [TK]D-Fender | hieunm_vips : then your problem isn't Read. its your DTMF settings. |
| 02:52.42 | hieunm_vips | I could press more than 6 digits |
| 02:52.44 | hmmhesays | anyone near a fax machine I can send a test to? |
| 02:52.46 | [TK]D-Fender | hieunm_vips : Fix your device config |
| 02:52.58 | [TK]D-Fender | hmmhesays : PM |
| 02:53.07 | hmmhesays | k |
| 02:53.21 | hieunm_vips | but, I tested with Asterisk1.2.7.1 OK |
| 02:54.12 | [TK]D-Fender | hieunm_vips : if it doesn't see the 4th digi being pressed, its not reads fault. Read works EVERY time I've ever heard of. Check you config. AGAIN. |
| 02:54.49 | hieunm_vips | OK, I will try another soft, Xlite is OK ? |
| 02:56.08 | *** join/#asterisk axscode (n=axscode@203.213.217.123) |
| 02:57.29 | NEZPERCECOUNTY | Anyone done video with Asterisk |
| 02:57.50 | axscode | NEZ: what do you want video? as content or as peer-to-peer video call? |
| 03:03.25 | NEZPERCECOUNTY | For starters yes. |
| 03:03.36 | NEZPERCECOUNTY | Is there such a thing as a confrance of video? |
| 03:06.30 | CyberKnet2 | a video of confluence? |
| 03:07.06 | NEZPERCECOUNTY | LOL |
| 03:07.14 | NEZPERCECOUNTY | Multi streams to everyone. |
| 03:07.28 | Supaplex | what's to con france? they're gullible to begin with right? |
| 03:07.29 | iCEBrkr | Yeah it's called Cu-SeeME :D |
| 03:07.44 | axscode | hahaha |
| 03:08.00 | NEZPERCECOUNTY | BRB |
| 03:08.14 | Supaplex | back yet? |
| 03:08.24 | axscode | actually you just turn on the VideoSupport to sip.conf then make the codec available.. try using eyeBeam softphone to start with... |
| 03:09.06 | *** join/#asterisk Winkie (n=urmom@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com) |
| 03:09.29 | *** join/#asterisk undrdawg (n=ud@unaffiliated/underdawg) |
| 03:17.51 | *** join/#asterisk mosty (n=mostynm@203.143.64.82) |
| 03:18.56 | mosty | how can i use register with multiple sip accounts on a single server with asterisk 1.2? it seems to just use the last defined friend in my sip.conf |
| 03:20.21 | DasTech | I noticed alot of kpflemings modules have been pulled from 1.4 |
| 03:20.36 | DasTech | where they dropped or just not been ported |
| 03:20.44 | [TK]D-Fender | Hey, whats the command to test zaptel timing issues on analog chanels? |
| 03:20.54 | JT | zttest? |
| 03:20.58 | [TK]D-Fender | I've got some crackling,e tc, and have rebooted the box already to see if that'd clear it up |
| 03:21.09 | [TK]D-Fender | JT : Didn't say anything to me. |
| 03:21.27 | JT | it's not analog specific either |
| 03:21.31 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 03:21.31 | *** mode/#asterisk [+o denon] by ChanServ |
| 03:21.32 | mosty | does anyone know how i can setup asterisk to act as multiple sip clients from a particular host? |
| 03:21.41 | JT | it returned nothing? |
| 03:22.45 | [TK]D-Fender | JT : I even have a channel open and I see nothing at all. |
| 03:23.20 | JT | what card? |
| 03:23.25 | [TK]D-Fender | TDM2400P |
| 03:23.31 | [TK]D-Fender | 8 FXO |
| 03:24.03 | JT | that's really weird, haven't heard of zttest returning nothing |
| 03:25.01 | [TK]D-Fender | JT : was using zttool. zttest is "not found" |
| 03:25.19 | JT | well you need to use zttest |
| 03:25.28 | JT | which may mean installing and compiling it |
| 03:25.35 | [TK]D-Fender | from where? |
| 03:25.50 | JT | zaptel sources? |
| 03:26.29 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 03:28.59 | [TK]D-Fender | JT : This system is possibly running binaries parallel to SRPMS+compile. I'm going to try and manual build it. Could explain the problem. |
| 03:29.41 | JT | if you used a package management system, it should've installed zttest for you |
| 03:29.45 | JT | but oh well |
| 03:30.29 | [TK]D-Fender | my avg is 99.987793% |
| 03:30.53 | [TK]D-Fender | I did a manual build and though make bombed out the install, it ran from the source folderf. |
| 03:31.21 | JT | my system here seems to like 99.951172% :( |
| 03:33.44 | sevard | [TK]D-Fender: you're weird. |
| 03:34.51 | [TK]D-Fender | sevard : Flatter will get you far. For everything else...... MasterCard ;) |
| 03:37.13 | *** part/#asterisk hieunm_vips (n=hieunm_v@210.245.57.162) |
| 03:39.30 | *** join/#asterisk Hansin321 (n=chatzill@c-67-190-5-42.hsd1.co.comcast.net) |
| 03:41.43 | *** join/#asterisk alerios (n=alerios@190.24.97.148) |
| 03:46.36 | *** join/#asterisk techie (n=techie@c-67-182-22-174.hsd1.ca.comcast.net) |
| 03:47.17 | DasTech | format_au.so |
| 03:47.17 | DasTech | <PROTECTED> |
| 03:47.17 | DasTech | <PROTECTED> |
| 03:47.17 | DasTech | <PROTECTED> |
| 03:47.24 | DasTech | have wrong channel |
| 03:47.42 | JT | okay? |
| 03:48.31 | DasTech | all things I noticed have been dropped in 1.4 |
| 03:48.48 | DasTech | making a list of formats and apps not in 1.4 |
| 03:53.01 | *** join/#asterisk bmg505 (n=leon@c1-39-9.rndf.isadsl.co.za) |
| 03:58.08 | *** part/#asterisk techie (n=techie@c-67-182-22-174.hsd1.ca.comcast.net) |
| 03:58.42 | NEZPERCECOUNTY | Okay I'm back. |
| 03:58.53 | NEZPERCECOUNTY | Anyone have experiance with Video using Asterisk |
| 03:58.57 | NEZPERCECOUNTY | Peer-to-Peer |
| 03:59.02 | NEZPERCECOUNTY | X-Lite / Eyebeam |
| 04:12.59 | JT | it'd be good if ircds had an option "supress HydraIRC quit messages" |
| 04:15.36 | *** part/#asterisk DasTech (n=DasTech@ppp-71-128-113-209.dsl.irvnca.pacbell.net) |
| 04:19.39 | [TK]D-Fender | JT : Set your's to ignore them :) |
| 04:20.03 | JT | i don't want to ignore all quit messages |
| 04:20.12 | JT | just the stupid pompous ones from hydrairc |
| 04:22.33 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:42.39 | *** join/#asterisk saftsack (n=oliver@p54A7CAE9.dip.t-dialin.net) |
| 04:42.45 | *** join/#asterisk eltech (i=G00Ds@ool-457c9421.dyn.optonline.net) |
| 04:44.06 | *** join/#asterisk Jenna (i=CherryRe@gateway/tor/x-617d1af0c8a7f499) |
| 04:47.56 | *** join/#asterisk nortex (n=barracud@adsl-70-252-57-95.dsl.amrltx.sbcglobal.net) |
| 04:53.06 | nortex | Anybody around I can run a question by? |
| 04:54.03 | Jenna | hmm. I guess not |
| 04:54.08 | nortex | I have a cordless phone that fails to dial from time to time, it is one that allows you to dial then number in then press send to place the call. |
| 04:54.28 | nortex | Every other calls seems to fail. |
| 04:55.08 | nortex | Jenna: Apparently I'll have to remeber my question for tomorrow :) |
| 04:56.34 | Jenna | yeah. seems like most of are captivated by the elections thingies |
| 05:06.25 | *** join/#asterisk saftsack (n=oliver@p54A7C5E7.dip.t-dialin.net) |
| 05:07.38 | CunningPike | nortex: Is it a wifi phone? |
| 05:08.13 | nortex | Nope, a vtech cordless. |
| 05:08.17 | *** join/#asterisk xxoxx (n=xxoxxx@tor/regular/xxoxx) |
| 05:11.10 | nortex | I may need to swicth back to a 1.2 build and not the 1.4, I have other problems with the zap channel not completeing calls once it dials out to the pstn. |
| 05:11.47 | mitcheloc | hola nortex |
| 05:12.07 | nortex | Hey mitcheloc |
| 05:12.48 | mitcheloc | hey, grab the ff extension from the website please :), i'm not sure yours will auto-update |
| 05:12.56 | mitcheloc | you were testing beta |
| 05:13.49 | mitcheloc | this ones got a disable button too in case a webpage can't handle it |
| 05:14.40 | hads | mitcheloc: Is your extension platform dependant? |
| 05:15.00 | nortex | Cool, I'll update this pc and the one at work in the morning. |
| 05:15.30 | nortex | mitcheloc: Have you been getting the bug reports from the beta you sent? |
| 05:16.25 | mitcheloc | just "it doesn't work" |
| 05:17.31 | nortex | mitcheloc: I'll try to put some details together in the morning and email them to you. It has been stable sometimes and others not so stable. |
| 05:17.53 | mitcheloc | hads: nooo, it is, but not completely |
| 05:17.58 | mitcheloc | you need the snap part too |
| 05:18.12 | mitcheloc | nortex: cool thanks |
| 05:18.17 | mitcheloc | i'm out of here guys |
| 05:18.29 | nortex | later |
| 05:19.23 | nortex | hads: The extension sends the number to the snap program, which runs currently only in Windows I believe. |
| 05:19.37 | hads | I see. |
| 05:20.43 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 05:26.02 | *** join/#asterisk xxoxx (n=xxoxxx@tor/regular/xxoxx) |
| 05:30.19 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:36.17 | *** join/#asterisk JustinWick (n=jwick@unaffiliated/jpl/jpl-justin) |
| 06:02.45 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 06:02.45 | *** mode/#asterisk [+o mog] by ChanServ |
| 06:04.38 | *** join/#asterisk tengulre (n=tengulre@221.11.5.182) |
| 06:07.56 | *** join/#asterisk pipipi (n=suedoh@adsl-68-125-33-140.dsl.irvnca.pacbell.net) |
| 06:14.48 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:17.00 | *** join/#asterisk Marshall16 (n=Marshall@cpe-24-26-138-61.columbus.res.rr.com) |
| 06:17.16 | Marshall16 | what's a place that provides IAX services? |
| 06:19.13 | shellshark | shellshark.net :) |
| 06:20.38 | hads | atp.org.au does |
| 06:20.55 | shellshark | but he's in the US ;) |
| 06:21.19 | hads | He didn't say that. |
| 06:21.20 | hads | :) |
| 06:21.38 | shellshark | his hostname did ;) |
| 06:21.41 | mog | nufone |
| 06:21.46 | mog | voicepuls |
| 06:21.46 | mog | e |
| 06:21.47 | *** join/#asterisk ThaZZa (n=me@229.9.233.220.exetel.com.au) |
| 06:21.51 | mog | broadvox |
| 06:21.52 | shellshark | shellshark.net ;) |
| 06:21.53 | mog | mogtel |
| 06:22.01 | mog | freeworld dialup |
| 06:22.01 | ThaZZa | Hey all. |
| 06:22.05 | hads | shellshark: I know. People should really be more specific though. |
| 06:22.07 | Marshall16 | shellshark |
| 06:22.12 | Marshall16 | how long till i can use it? |
| 06:22.19 | shellshark | Marshall16: immediately :) |
| 06:22.19 | Marshall16 | after buying |
| 06:22.21 | Marshall16 | okay |
| 06:22.30 | Marshall16 | 1800 numbers? |
| 06:22.32 | Marshall16 | can i call? |
| 06:22.39 | shellshark | can you call toll-free? sure |
| 06:22.48 | shellshark | 900/976... no ;) |
| 06:22.57 | ThaZZa | Anyone got a Cisco CCO account? |
| 06:23.31 | ThaZZa | I need some old versons of 7960 firmware |
| 06:23.49 | shellshark | i have a CCO account, but it's so limited I can hardly do anything with it |
| 06:23.52 | Marshall16 | paypal |
| 06:23.54 | Marshall16 | is the only method? |
| 06:24.12 | shellshark | Marshall16: in /msg |
| 06:24.51 | ThaZZa | shellshark: any chance you can login and try and access the 7960 verison 5 firmware for me? |
| 06:24.56 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 06:24.56 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 06:26.07 | shellshark | ThaZZa: i can't download any firmwares at all |
| 06:26.11 | shellshark | ThaZZa: ask in #cisco |
| 06:26.17 | shellshark | ThaZZa: look for Catfive... |
| 06:26.36 | ThaZZa | shellshark: Thanks for trying. |
| 06:26.48 | shellshark | well, a lot of people in there could probably help you, but Catfive can definitely help you out :p |
| 06:39.29 | *** join/#asterisk aao_pwner (n=_s@c-24-21-91-140.hsd1.mn.comcast.net) |
| 06:39.44 | aao_pwner | erm guys, i just installed Trixbox |
| 06:40.01 | Strom_M | there's your first mistake ;) |
| 06:40.01 | aao_pwner | and it reboots ... |
| 06:40.07 | aao_pwner | LoL someone told me to. |
| 06:40.11 | aao_pwner | I was like umm at first. |
| 06:40.21 | aao_pwner | but it's a spare computer with nothing on it so i was like mnus well try it |
| 06:40.26 | aao_pwner | but every time it starts up it just shuts down. |
| 06:40.36 | aao_pwner | after it trys and loads all the modules... |
| 06:41.14 | aao_pwner | ok |
| 06:41.16 | aao_pwner | it's up |
| 06:48.55 | aao_pwner | errrm |
| 06:48.59 | aao_pwner | in Trixbox how do i remove a user? |
| 06:49.02 | aao_pwner | i'm so used to FreeBSD |
| 06:49.07 | aao_pwner | so im like trying rmuser and stuff lol |
| 06:52.06 | shellshark | #trixbox |
| 06:52.17 | shellshark | we dont support asterisk@home, freepbx, or trixbox here |
| 06:52.28 | hads | #trixbox |
| 06:52.37 | *** join/#asterisk adorah (n=admin@87.68.145.189.cable.012.net.il) |
| 06:52.46 | hads | erm. Yeah |
| 06:53.16 | hads | I believe #freepbx is the one. |
| 06:54.42 | shellshark | that too ;) |
| 06:54.44 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 06:56.16 | litage | what type of connection/cable hooks up to a TDM2400P? is it an RS-232 breakout box? |
| 06:56.38 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
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| 07:02.06 | *** mode/#asterisk [+o mog] by ChanServ |
| 07:06.04 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 07:09.05 | *** join/#asterisk oej (n=oej@p54BEC8F4.dip0.t-ipconnect.de) |
| 07:09.49 | *** join/#asterisk h3x0r4t0r (n=hex@ip68-224-236-92.lv.lv.cox.net) |
| 07:12.14 | *** join/#asterisk petit-toon (n=petit-to@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 07:14.04 | *** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org) |
| 07:14.06 | Corydon76-home | litage: it's an Amphenol adapter. Typically for a 66block |
| 07:14.44 | litage | thanks Corydon76-home |
| 07:15.34 | *** join/#asterisk tengulre11 (n=tengulre@221.11.5.182) |
| 07:16.11 | litage | Corydon76-home: are there any particular brands that are recommended for use with the TDM2400P? |
| 07:17.27 | Corydon76-home | 66 blocks are commodity hardware |
| 07:17.55 | Corydon76-home | Graybar typically carries them |
| 07:18.13 | Corydon76-home | or pick your favorite contractor supply house |
| 07:18.14 | Strom_M | $6 a piece |
| 07:18.52 | Corydon76-home | Strom_M: $6 sounds like the price for one wired without the Amphenol connector |
| 07:19.02 | Strom_M | right, thats what im talking about |
| 07:19.15 | Strom_M | buy the amphenol cable and punch the damned thing down yourself :) |
| 07:19.33 | Corydon76-home | Heh |
| 07:20.35 | Strom_M | makes for a cleaner install when you don't have extra cable coiled up all over the place |
| 07:22.35 | Strom_M | although, as a friend of mine observed, "When you have purple wire with orange dots on it, you know you have too many damned wires" |
| 07:22.37 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 07:30.30 | *** join/#asterisk S^P (n=ss@203.81.196.20) |
| 07:36.54 | *** join/#asterisk bon (n=bon@193.93.72.11) |
| 07:37.16 | bon | how'd you analyze the mysql realtime connection? |
| 07:37.33 | bon | seems like my cdr_mysql is not working anymore and i don't know why |
| 07:41.32 | bon | extensions.ael what is it |
| 07:42.09 | *** join/#asterisk af_ (n=af@ip-179-179.sn1.eutelia.it) |
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| 07:44.27 | tengulre11 | app_adsiprog.c:923: internal compiler error: Segmentation fault ?? |
| 07:47.03 | Rhizome | That a question? |
| 07:58.55 | tengulre11 | Rhizome: I got it when I compile asterisk-1.4.0-beta3. |
| 08:02.13 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 08:02.39 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) |
| 08:05.40 | aao_pwner | SIP/2.0 404 Not found |
| 08:05.43 | aao_pwner | what does that mean? |
| 08:05.46 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 08:05.46 | *** mode/#asterisk [+o denon] by ChanServ |
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| 08:54.35 | lat1234 | hello |
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| 09:20.47 | *** join/#asterisk knarfly (n=knarfly@c-65-34-177-3.hsd1.fl.comcast.net) |
| 09:21.24 | knarfly | can anyone help me with getting kernel sources linked so that zaptel will compile? |
| 09:22.46 | *** join/#asterisk knarfly (n=knarfly@c-65-34-177-3.hsd1.fl.comcast.net) |
| 09:23.19 | lat1234 | hello |
| 09:23.45 | knarfly | can anyone help me link kernel sources so that zaptel will compile on FC5-x86_64 ? |
| 09:23.49 | lat1234 | anyone knows what is better sip or iax in terms of bandwidth |
| 09:24.14 | monsted | lat1234: should be about the same |
| 09:24.29 | monsted | lat1234: the real issue is your choice of codecs |
| 09:24.41 | S^P | hi how to use A(x) option of Dial. |
| 09:25.22 | S^P | is there is any additional option to make it work? |
| 09:28.39 | S^P | I gave valid value of X but no annoucement |
| 09:37.44 | *** join/#asterisk oej_ (n=oej@80.187.222.83) |
| 09:38.32 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 09:38.32 | tzafrir | lat1234, iax can have an atvantage if you happen to have a number of calls to the same trunk |
| 09:41.30 | pif | how can I prevent asterisk to attempt sip native bridging? |
| 09:41.46 | pif | I added canreinvite=no to all sip.conf involved |
| 09:42.39 | lat1234 | thanks tzafrir |
| 09:44.06 | tzafrir | pif, make sure both parties have no common codec. Asterisk will have to do transcoding :-( |
| 09:44.19 | *** join/#asterisk Tili (n=tili@202.133.65.98) |
| 09:44.36 | pif | only way? |
| 09:44.53 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 09:46.52 | pif | what does allowguest=yes in sip.conf do ? |
| 09:48.04 | *** join/#asterisk [hC] (n=hardcore@S01060018394c6be3.vc.shawcable.net) |
| 09:51.15 | *** join/#asterisk henrique (n=henrique@201-1-130-79.dsl.telesp.net.br) |
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| 09:53.27 | angryuser | if someone can help me... i have a strange "BZZZZZZ" between intrenal sip/iax calls, calling party can hear, but another only near BZZZZZZ;) help |
| 09:54.19 | angryuser | sip tp sip does the same, and externam sip provider, same thing |
| 09:54.31 | *** join/#asterisk hieunm_vips (n=hieunm_v@210.245.57.162) |
| 09:54.34 | *** join/#asterisk lorinc (n=ang@caracas-2642.adsl.interware.hu) |
| 09:54.41 | hieunm_vips | hi |
| 09:54.59 | hieunm_vips | how could I benchmark asterisk's performance |
| 09:55.51 | hieunm_vips | Is there anyone can teach me ? |
| 09:57.01 | qdk | hieunm_vips: try sipp, siksak or similar. |
| 09:57.38 | hieunm_vips | Can they test media streaming performance ? |
| 09:59.00 | *** join/#asterisk BurtyB (n=chris@alxa.camelnetwork.com) |
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| 10:03.08 | *** join/#asterisk elshaa (n=elshaa@polar.es6.egwn.net) |
| 10:03.10 | elshaa | hi. |
| 10:03.39 | *** join/#asterisk my007ms (n=my007ms@217.139.224.194) |
| 10:04.00 | elshaa | I'm looking for a linux softphone, which one do you advise ? |
| 10:04.08 | *** part/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 10:04.28 | *** join/#asterisk apardo (n=apardo@87.217.144.70) |
| 10:04.56 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 10:05.11 | jmesquita | Hello yall |
| 10:05.38 | my007ms | elshaa, xlite |
| 10:06.00 | BurtyB | xlite is linux too? |
| 10:06.12 | jmesquita | Does anyone knows how to know if an agent has ignored/reject a call on CallBackLogin? |
| 10:06.22 | *** join/#asterisk mfroes (n=froes@200-162-218-81.corp.ajato.com.br) |
| 10:07.50 | tzafrir | twinkle is a nice linux client |
| 10:07.58 | tzafrir | Not to mention gnomemeeting |
| 10:08.54 | elshaa | ok. thanks. I think I'll go for twinkle. |
| 10:12.10 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 10:12.20 | *** join/#asterisk jm|work (n=jamiem@dilbert.jamiem.com) |
| 10:12.30 | jmesquita | Guys, plz, I need some help! :'( |
| 10:13.06 | *** join/#asterisk marexz (n=marexz@195.20.127.222) |
| 10:13.56 | jmesquita | <PROTECTED> |
| 10:14.15 | jmesquita | I have agents logged in, but they are probably with the DO NOT DISTURB on |
| 10:14.22 | jmesquita | Meaning they wont take the call |
| 10:14.24 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 10:15.44 | jmesquita | How do I generate reports saying that the agent had DND on? |
| 10:18.42 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) |
| 10:19.30 | my007ms | exten => 100,1,<some code> can i make something els run b4 that line for example exten => 100,<XX>,<some code> my Q is what is this XX |
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| 10:24.17 | qdk | elshaa: sjphone |
| 10:24.57 | qdk | my007ms: why would you advise x-lite? I would like to know the reason for it. |
| 10:25.46 | my007ms | coz i try it and it was have very good |
| 10:26.31 | my007ms | i try some onet but xlite is easy lite from his name say |
| 10:26.43 | my007ms | and if the clinet will be windows he can use xten |
| 10:27.06 | my007ms | but this not mean that othere is not well also |
| 10:27.46 | qdk | my007ms: ok, so you dont care about the added latency in x-lite? |
| 10:28.51 | my007ms | it's do very well with me |
| 10:29.04 | my007ms | and eyeBeam is most great |
| 10:29.46 | qdk | I guess that's a 'yes'. |
| 10:37.41 | bXi | yo |
| 10:37.48 | bXi | how is voicemail 2 email handled? |
| 10:37.56 | bXi | does asterisk expect sendmail in the path or so? |
| 10:40.21 | *** part/#asterisk hieunm_vips (n=hieunm_v@210.245.57.162) |
| 10:41.31 | *** join/#asterisk xnon (i=xnon@200.8.30.50) |
| 10:42.05 | jmesquita | Guys, does anyone have problems with agents rejecting calls on callcenters here? |
| 10:42.40 | lat1234 | hi im new in asterisk can anyone help me setup the sip |
| 10:43.01 | lat1234 | it keeps on prompting Request timedout |
| 10:43.59 | ghenry | can you have 2 tdm400p in one server, or is it best to get the tdm2400p? |
| 10:45.13 | jm|work | ghenry: I am lead to believe that 2 x TDM400P would work |
| 10:45.45 | jm|work | isn't TDM2400P T1 anyway? |
| 10:46.07 | ghenry | was look at http://www.voipon.co.uk/digium-tdm2411e-4-fxs-4-fxo-with-echo-cancellation-p-184.html |
| 10:46.37 | jm|work | ghenry: are you connecting to digital or analogue lines? |
| 10:46.49 | ghenry | analogue |
| 10:46.56 | ghenry | looks liek you need this too: http://www.voipon.co.uk/breakout-box-tdm2400p-p-174.html |
| 10:47.19 | ghenry | need 4 in and 4 out analogue |
| 10:47.20 | jm|work | o.O |
| 10:48.01 | jm|work | "The TDM400P takes the place of an expensive channel bank and brings the system price point to the lowest in the industry. To scale this solution simply add additional modules or TDM cards as required." |
| 10:48.12 | jm|work | that last sentence |
| 10:48.22 | ghenry | so mayeb a channel is best? |
| 10:48.24 | jm|work | I would do 2+2 + 2+2 |
| 10:48.31 | *** join/#asterisk mattfletcher (n=matt@heysham-mail.nshc.co.uk) |
| 10:48.39 | jm|work | so that if one card goes pop you still have in and out available |
| 10:48.44 | jm|work | without having to re-jig the modules |
| 10:49.33 | ghenry | yeah. Just not sure of the best way to go. |
| 10:49.45 | jm|work | anyone using POTS + AUX for one number to two lines? |
| 10:50.02 | jm|work | I take it that it's quite straight forward |
| 10:50.08 | jm|work | and the telco does the routing ... |
| 10:50.14 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 10:50.49 | ghenry | we've just got 4 incoming lines, but need about 4 out, but the rest 20 IP phones will all be through sip in and out |
| 10:51.38 | jm|work | I'm just figuring out one of our branches at, |
| 10:51.40 | jm|work | atm* |
| 10:51.45 | jm|work | it currently has BT featureline |
| 10:51.47 | mattfletcher | Hello. I have set up spandsp for faxing, but I am having a problem. Whenever I receive a fax, the server recognises it, switches to the "fax" extension, and then * promptly crashes. there is nothing in the console, nothing in /var/log/asterisk/mesages, and only the fact that asterisk exited exit code 15 in /var/log/messages. Does anyone have any ideas what would cause the system to crash like this? |
| 10:52.21 | knarfly | mattfletcher, yes, not using pastebin for starters... |
| 10:53.26 | mattfletcher | there is nothing there that would require me to use pastebin! there's no code! |
| 10:53.41 | reza_ | send a link to an empty pastebin then |
| 10:53.42 | reza_ | duh |
| 10:53.50 | mattfletcher | no error messages even |
| 10:54.18 | knarfly | mattfletcher, now your getting it...! |
| 10:54.39 | reza_ | so does your log look like this : http://pastebin.ca/243352 ? |
| 10:55.10 | mattfletcher | knarfly, i am confused. what would you like me to do with pastebin? i have no output to show you. it just crashes. |
| 10:55.44 | knarfly | mattfletcher, nothing...just don't type in more than 4 lines at a time...like the motd instructs |
| 10:56.13 | *** part/#asterisk jonta (n=jonta@c-f86ae155.238-1-64736c20.cust.bredbandsbolaget.se) |
| 10:56.14 | knarfly | mattfletcher, it causes everyone to have to scroll by your message to see what they're reading... |
| 10:56.16 | BurtyB | knarfly buy a wider monitor ! |
| 10:56.18 | reza_ | matt - i dont see a problem with doing that - some people just like to complain |
| 10:56.22 | jmesquita | Guys, does anyone have problems with agents rejecting calls (such as DND) on callcenters here? |
| 10:56.49 | reza_ | scroll by? |
| 10:56.52 | reza_ | what do you mean by that |
| 10:57.26 | knarfly | or worse...the whole chatroom logs everyone out |
| 10:58.15 | knarfly | nothing...I just like to complain...! |
| 10:58.25 | reza_ | thought so |
| 10:58.28 | reza_ | ok, time for bed, night |
| 10:58.59 | knarfly | no it's 6 AM here in Miami....time for work |
| 10:59.51 | lat1234 | could somebody help me... my client would not register to asterisk using sip... it says request timedout 408 |
| 11:01.59 | knarfly | lat1234, what does your sip.conf entry for this client look like? |
| 11:02.15 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 11:05.50 | *** join/#asterisk phacka_ (n=phacka@chello082119111107.chello.sk) |
| 11:12.08 | mattfletcher | Here is a pastebin knarfly. It is some of my config, and the output from the * console: http://pastebin.ca/243359 |
| 11:15.42 | af_ | how could I have doorbell connected to *? |
| 11:15.46 | *** join/#asterisk oej_ (n=oej@80.187.210.16) |
| 11:21.34 | BurtyB | af_ some form of IO plugged into an AGI script? |
| 11:21.58 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 11:23.36 | af_ | well mostly I am puzzling how to wire things and to what pc parts could I connect i/o and audio from the doorbell. |
| 11:24.04 | af_ | for example: could just attach the doorbel mike to a pc mike - in? |
| 11:29.44 | *** join/#asterisk skirmisha (n=vk@213.144.225.179) |
| 11:31.50 | *** join/#asterisk oej_ (n=oej@80.187.210.16) |
| 11:33.20 | pif | when an asterisk server bridges two sip asterisk legs, no sound is heard |
| 11:33.38 | pif | even if no native bridging occurs |
| 11:34.04 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:35.52 | puzzled | morning |
| 11:36.10 | jm|work | anyone got a photo/image of an ISDN30 terminator interface sockets thingy? |
| 11:36.17 | jm|work | is it just std RJ45? |
| 11:36.22 | pif | yep |
| 11:36.53 | jm|work | so it I were to have 24 channels on my ISDN30, I'd just need one interface on my PRI card? |
| 11:37.02 | jm|work | rj45 <---> rj45 |
| 11:37.05 | pif | ack |
| 11:37.13 | jm|work | syn |
| 11:37.30 | jm|work | nop |
| 11:37.34 | jm|work | rst :( |
| 11:38.19 | tzafrir | jm|work, wow! |
| 11:38.25 | jm|work | wow? |
| 11:38.41 | *** part/#asterisk henrique (n=henrique@201-1-130-79.dsl.telesp.net.br) |
| 11:38.46 | tzafrir | you managed to kick ghenry ;-) |
| 11:38.54 | jm|work | lol |
| 11:39.00 | jm|work | Norton Firewall ;) |
| 11:40.53 | puzzled | jm|work: rj45 connectors but the pin-out is not of a standard LAN cable |
| 11:41.05 | jm|work | ok |
| 11:41.12 | jm|work | so I need an "ISDN cable" |
| 11:41.16 | jm|work | the orange ones? |
| 11:41.18 | jm|work | heh |
| 11:41.38 | jm|work | I could do do look at someones config for ISDN30 |
| 11:41.53 | puzzled | jm|work: you need T1/E1 pin-out. google should have it |
| 11:42.02 | jm|work | cool |
| 11:45.29 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 11:45.41 | pif | E1 pin-out only for crossed cables, |
| 11:46.00 | pif | depending on the telco you might only need a straight one |
| 11:46.45 | pif | are there still issues with iax2 reliability in 1.2.13 ? |
| 11:48.52 | tzafrir | a standard ethernet cable will do for a nob-crossed one, right? |
| 11:49.11 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 11:49.16 | tzafrir | (standard = with all 8 wires) |
| 11:49.37 | pif | yes |
| 11:55.19 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 11:56.29 | pif | anyone uses sip to route calls between asterisk's ? |
| 11:59.15 | *** join/#asterisk aadilismail (n=adilisma@41-247-154-202.wol.net.pk) |
| 12:00.43 | puzzled | pif: there are always issues with iax reliability :) |
| 12:00.43 | qdk | pif: no, but thinking about using SIP all the way, coz reinvite in backend might be a good way to get some kind of failover and SIP is also a more used protocol than IAX if i want to interconnect different switch/pbx software. |
| 12:01.36 | *** join/#asterisk russel1234 (n=Miranda@218.111.142.229) |
| 12:02.06 | puzzled | qdk: 1.2 svn got a few more chan_sip fixes since 1.2.13 so sip should be getting better |
| 12:02.15 | aadilismail | hi |
| 12:02.41 | russel1234 | to all the asterisk expert out there...exactly is there a max. user limitation for concurrent sip signaling for asterisk ? |
| 12:04.52 | qdk | puzzled: well I expect to use just about anything but * at the time of production, so its not that important. :-) though improvements is always good. |
| 12:05.12 | puzzled | qdk: hehe |
| 12:05.18 | qdk | russel1234: limits in software or hardware? |
| 12:05.27 | mattfletcher | I'm still having problems receiving faxes. I've worked out by trial and error that it is the rxfax application that is causing the server to crash. But still no output on any error logs that I can see |
| 12:06.21 | mattfletcher | The app shows up when I do a show applications, but calling it crashes the server. How can I make further tests? |
| 12:06.25 | aadilismail | new born: whats it Swap file ".extensions.conf.swp" already exists! |
| 12:06.26 | aadilismail | [O]pen Read-Only, (E)dit anyway, (R)ecover, (Q)uit, (D)elete it: |
| 12:06.28 | russel1234 | qdk: i believe it should be software...the asterisk sip proxy itself |
| 12:06.42 | puzzled | russel1234: it depends on so many variables that the only way to figure that out is to setup your desired setup and start poiunding it with calls |
| 12:07.01 | puzzled | russel1234: and asterisk is not a sip proxy. get openser for that |
| 12:07.46 | russel1234 | puzzled: how exactly different is asterisk with ser? i've played with ser before but not asterisk :( |
| 12:07.57 | russel1234 | puzzled: isnt it more or less the same? |
| 12:08.45 | qdk | russel1234: SER doesnt handle the media stream, asterisk does. |
| 12:08.51 | puzzled | russel1234: asterisk is not a sip proxy. asterisk is a b2bua (hope I got that term right) |
| 12:09.35 | puzzled | russel1234: and openser scales about a gazillion times better than asterisk. so better test your particular setup and find out where the limits are |
| 12:09.38 | russel1234 | eeks..i see...then i assume asterisk has all the nat support within itself since media stream is going through it |
| 12:10.17 | puzzled | it can cope with nat. don't know if it can cope with all sorts of nat but at least some |
| 12:10.41 | russel1234 | of course as ser only handles the signaling portion...not very heavy...but is there a rough limit for asterisk in terms on software? |
| 12:10.54 | russel1234 | let say i am using a standard xeon intel processor with 2gb ram |
| 12:11.16 | qdk | russel1234: are we talking media or no media through *? |
| 12:11.53 | russel1234 | media |
| 12:12.21 | qdk | russel1234: * can handle 3-400 concurrent calls at best i guess. |
| 12:12.31 | puzzled | russel1234: just pass through of sip session? transcoding? any services on the box like ivr, meetme? |
| 12:12.51 | puzzled | qdk: I'd say that's a bit on the high side |
| 12:12.56 | russel1234 | nope...just conference maybe... |
| 12:12.57 | qdk | russel1234: that number being much lower with transcoding and similar stuff. |
| 12:13.31 | qdk | puzzled: perhaps, but you could probably tweak it to do it on some good hardware. |
| 12:14.32 | puzzled | russel1234: with conferencing on your uni proc box I would guess around 100 depending on how many in the conf and how many sip pass through |
| 12:14.56 | pif | puzzled : what issues where fixed with sip in 1.2-svn ? |
| 12:14.57 | puzzled | qdk: yeah but he mentioned a rather unimpressive uniproc box :) |
| 12:15.05 | qdk | russel1234: btw. the RAM in that machine is probably overkill, but anyway i guess you could expect about 100... well what puzzled said. .-) |
| 12:15.15 | *** join/#asterisk evilmnky (n=evilmnky@adsl-75-42-28-214.dsl.hstntx.sbcglobal.net) |
| 12:15.26 | qdk | puzzled: True |
| 12:15.59 | puzzled | pif: destroy the right dialog and remove the dialog properly at module unload |
| 12:16.10 | russel1234 | hmm okay how scaleable is asterisk ? |
| 12:16.24 | russel1234 | if i am expecting like 1k users... |
| 12:16.30 | *** join/#asterisk mfroes (n=froes@200-162-218-81.corp.ajato.com.br) |
| 12:16.54 | puzzled | russel1234: use openser in front of asterisk. will save you a lot of trouble |
| 12:17.14 | puzzled | russel1234: then use asterisk for services like conferencing and voicemail |
| 12:17.38 | puzzled | jm|work: congrats |
| 12:17.42 | jm|work | puzzled: :P |
| 12:18.07 | jm|work | ... which is why I've been looking to chroot |
| 12:19.21 | jm|work | so why would I need a multi channel PRI card? |
| 12:19.28 | jm|work | for >60 channels? |
| 12:19.50 | qdk | russel1234: there is like no scalability in *.. but you could place several beside each others and target them individual through some frontend... like SER (or just another * doing SIP reg/proxy only.) |
| 12:20.30 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:21.35 | jmesquita | Have anyone ever had calls on queue_log that does not show in cdr? |
| 12:22.39 | jmesquita | And vice-versa |
| 12:24.16 | russel1234 | hmm whats the differences between OpenSER and SER? gee...so noob :( |
| 12:26.06 | jmesquita | russel1234: I heard the SER is a lot better coping with NAT then Asterisk |
| 12:26.22 | jmesquita | russel1234: As a matter of fact, NAT and Asterisk are pretty tuff on each other |
| 12:26.31 | russel1234 | hmm SER or OpenSER? |
| 12:26.41 | russel1234 | eventually they're both different :( |
| 12:26.57 | jmesquita | russel1234: OpenSER is the open source version and SER is the company's version |
| 12:27.36 | russel1234 | aiks SER no longer free? ever since bought over by iptel? |
| 12:27.36 | lat1234 | could somebody help setup my asterisk... |
| 12:27.37 | lat1234 | using sip |
| 12:27.48 | lat1234 | i could not make a call although im registered |
| 12:27.49 | jmesquita | pingtel always owned ser |
| 12:28.01 | qdk | jmesquita: what issues do you have with * and NAT? I have like non, so i would like to know WHAT should/could be the problem. |
| 12:28.09 | jmesquita | It was it that opened it |
| 12:29.03 | jmesquita | qdk: I haven't really debugged a lot but I have had some problems with small routers such as linksys/dlink ADSL modems |
| 12:29.37 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 12:29.48 | jmesquita | qdk: RTP is the big problem in that case |
| 12:30.01 | jmesquita | qdk: Asterisk can handle the rest |
| 12:31.42 | tcseke | hi |
| 12:31.58 | tcseke | is there a know bug about app_meetme? |
| 12:32.01 | tcseke | Nov 8 13:22:28 WARNING[22773]: app_meetme.c:1541 conf_run: Unable to write frame to channel: No such file or directory |
| 12:32.44 | tcseke | i receive a lot of warning like this |
| 12:33.34 | qdk | jmesquita: well SER doesnt handle the RTP part, so how does it help? |
| 12:34.15 | jmesquita | qdk: That's why I said I heard SER can handle it better |
| 12:34.15 | qdk | jmesquita: how does your problem show itself? no sound? no connection? what? |
| 12:34.38 | russel1234 | i dont think SER can handle NAT without rtp going through it |
| 12:34.53 | russel1234 | but OpenSER claims to do it well |
| 12:35.54 | russel1234 | # used a load balancer in stateless mode, OpenSER can handle over 5000 call setups per second |
| 12:35.54 | russel1234 | # on systems with 4GB memory, OpenSER can serve a population over 300 000 online subscribers |
| 12:36.12 | russel1234 | i dont think the figures here are for media? |
| 12:36.22 | russel1234 | its a little HUGE...dont u think? |
| 12:37.02 | myiagy | ser itself doesn't handle rtp, but mediaproxy does.. so, SER/Mediaproxy is a viable solution for nat problems |
| 12:37.08 | qdk | it stil does shit for failover... though the "stateless" part doesnt scream RTP. .-) |
| 12:37.17 | guilherme-jorge | <PROTECTED> |
| 12:37.17 | guilherme-jorge | ks fine. Any idea? |
| 12:38.09 | qdk | guilherme-jorge: use nohup? |
| 12:38.32 | *** join/#asterisk billy-jo (n=billy-jo@AClermont-Ferrand-251-1-42-181.w81-251.abo.wanadoo.fr) |
| 12:39.15 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 12:39.15 | *** mode/#asterisk [+o denon] by ChanServ |
| 12:39.20 | russel1234 | hmm how about this...can i use SER/openser with a SBC in front? |
| 12:39.37 | russel1234 | a session border controller to solve all the nat/firewall/security |
| 12:40.04 | puzzled | russel1234: I thought openser has some modules that make it act like an SBC. at least fix nat problems etc. |
| 12:40.26 | russel1234 | seems like so...but never know till i get my hands dirty to try it... |
| 12:40.46 | *** join/#asterisk loopt (n=pt@catv54033E44.pool.t-online.hu) |
| 12:41.44 | guilherme-jorge | qdk: I already test with nohup and it doesn't work fine! When daemon goes down asterisk goes down too!! |
| 12:46.49 | qdk | whats the purpose of a SBC? |
| 12:47.11 | qdk | guilherme-jorge: screen perhaps? |
| 12:48.44 | guilherme-jorge | qdk: Right now I discovered that asterisk goes down when I kill the daemon with CTRL+C, but when I kill the daemon with kill -9 or -15, asterisk server continues up |
| 12:49.58 | *** join/#asterisk oej_ (n=oej@80.187.148.18) |
| 12:51.20 | qdk | guilherme-jorge: what if you either shutdown correctly or just make an infinite loop? |
| 13:00.16 | jmesquita | Have any of you had problems of having calls on queue_log that does not show on cdr? |
| 13:02.21 | *** join/#asterisk arava (n=phani@ppp-69-218-205-5.dsl.wotnoh.ameritech.net) |
| 13:03.00 | *** join/#asterisk _PauloS_ (n=PauloS@mail.eletrodireto.com.br) |
| 13:03.11 | _PauloS_ | Hi all. |
| 13:03.26 | arava | Hi ineed some help in setting up an asterisk server. Iam not sure whether i have done iit right. I have started asterisk, installed freeepbx, configured two extensions and then was trying to login using EKiga, It says registration failed |
| 13:03.34 | arava | any help will be apprecizated |
| 13:04.21 | *** join/#asterisk sysreq (n=sysreq@142-217-128-85.telebecinternet.net) |
| 13:04.24 | _PauloS_ | arava, just share yout your doubts |
| 13:04.38 | _PauloS_ | arava, just share your doubts |
| 13:04.40 | arava | What ports does asterisk listen on ? |
| 13:04.44 | arava | i mean the server ? |
| 13:05.26 | _PauloS_ | arava, try "asterisk -r" on the console |
| 13:05.32 | arava | And why am I not able to my asterisk server ? anything iam missing |
| 13:06.32 | arava | what does that do ? |
| 13:06.40 | _PauloS_ | arava, there is an asterisk management interface you can connect from network, but its not meant for human interaction. |
| 13:07.05 | _PauloS_ | arava, this will give you the asterisk console |
| 13:07.21 | _PauloS_ | ~seen coppice |
| 13:07.37 | jbot | coppice <n=chatzill@229.166.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 3d 18h 39m 4s ago, saying: 'rusellb: I'd say you're much more paranoid now than 2 years ago'. |
| 13:09.57 | arava | _PauloS_, Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 13:10.24 | arava | iam running asterisk and this is the err when i tried 'asterisk -r' |
| 13:10.59 | arava | satinfo@satinfo-desktop:~$ ps ax | grep asterisk |
| 13:10.59 | arava | 18657 pts/1 Sl+ 0:00 asterisk -vvvvv -g ddddd -c |
| 13:10.59 | arava | 18838 pts/2 S+ 0:00 grep asterisk |
| 13:11.14 | _PauloS_ | arava, do you see a line for asterisk when you type "ps ax | grep asterisk" (by the way, what OS are you running asterisk on?) |
| 13:11.25 | arava | Ubuntu |
| 13:11.37 | _PauloS_ | arava, ok... let me think... |
| 13:11.45 | *** join/#asterisk CMike (i=daemon@c-614071d5.116-1-64736c10.cust.bredbandsbolaget.se) |
| 13:11.52 | Modcuts | Is there away of running the cli and outputing its content to a file? |
| 13:12.28 | arava | and btw that file exists |
| 13:13.07 | _PauloS_ | Modcuts, you can issue a command like "asterisk -r -x 'show channels' > output.txt" |
| 13:13.18 | Modcuts | lovely |
| 13:13.19 | myiagy | does your asterisk.conf says: astrundir => /var/run/asterisk |
| 13:13.20 | myiagy | ? |
| 13:13.21 | _PauloS_ | arava, weird... |
| 13:13.21 | Modcuts | cheer |
| 13:13.49 | arava | is it with something with the permissions ? I mean i ran the asterisk with root |
| 13:13.50 | *** join/#asterisk zaswk (n=zas@LSt-Amand-152-33-30-86.w82-127.abo.wanadoo.fr) |
| 13:14.57 | _PauloS_ | arava, Ubuntu is great for desktops, but I think you will get better support with Debian for an asterisk server... |
| 13:15.58 | _PauloS_ | arava, some people thinks Ubuntu is a poor choice for server. |
| 13:16.31 | arava | _PauloS_, Ya i understand that. but aint it using the same deb packages |
| 13:18.36 | _PauloS_ | arava, I cant understand why you cant connect to your asterisk if it is running. Try shuting it down, then run "asterisk -vvvvvcp |
| 13:18.38 | arava | Modcuts, yes it does |
| 13:18.54 | arava | as a root user ? |
| 13:18.59 | _PauloS_ | yeps. |
| 13:19.30 | _PauloS_ | arava, that way asterisk will not go in background |
| 13:19.46 | arava | ya i checked that out , but the same problem |
| 13:20.06 | arava | btw iam able to connect it thru freepbx and able t configure |
| 13:20.36 | arava | satinfo@satinfo-desktop:/var/run/asterisk$ cat /var/run/asterisk/asterisk.ctl |
| 13:20.36 | arava | cat: /var/run/asterisk/asterisk.ctl: No such device or address |
| 13:21.48 | _PauloS_ | arava, do you have ztdummy loaded? |
| 13:22.07 | arava | _PauloS_, iam not sure what is that |
| 13:22.14 | _PauloS_ | arava, modprobe ztdummy |
| 13:23.05 | _PauloS_ | arava, are you pure voip or do you have some telephony interface, like digium or sangoma cards? |
| 13:23.24 | arava | right now we dont have any cards |
| 13:23.47 | arava | but we are planning to put some in the near future if we get this up and running |
| 13:23.58 | arava | but we have some sipura devices for testing |
| 13:24.22 | _PauloS_ | Im not sure, but I think you need the ztdummy driver loaded to provide clock for asterisk |
| 13:24.45 | _PauloS_ | check that out on the wiki. |
| 13:24.58 | arava | iam installing that now lemme check |
| 13:27.05 | _PauloS_ | arava, where are you from? |
| 13:27.27 | arava | Iam in US, but my company is in India |
| 13:28.21 | _PauloS_ | arava, nice |
| 13:28.41 | _PauloS_ | I'm in Brazil. |
| 13:28.44 | arava | so iam trying to have asterisk to cut on my phone bill |
| 13:29.13 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 13:29.43 | arava | _PauloS_, nice |
| 13:29.50 | _PauloS_ | if you are willing to give me ssh access I can help you better. |
| 13:31.19 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.162) |
| 13:31.20 | *** join/#asterisk tom007 (n=tom@p54A88AD0.dip0.t-ipconnect.de) |
| 13:31.46 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 13:31.55 | backblue | hi, anyone know how can i restart a polycom phone? |
| 13:32.59 | monsted | yank the power? :) |
| 13:33.02 | tom007 | hi, using the ENUMLOOKUP app, I get the message " No application 'ENUMLOOKUP' for extension " - what's going wrong here? |
| 13:33.19 | backblue | monsted: well, remotly! :) |
| 13:33.57 | monsted | backblue: call it and ask whoever picks it up to yank the power? :) |
| 13:34.00 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 13:34.56 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 13:36.16 | backblue | monsted: heheh i'm seeing here one way, with sip notify stuff |
| 13:37.01 | *** join/#asterisk delphus (n=delphus@mail.delphus.org) |
| 13:38.23 | *** join/#asterisk svenna (n=svenna@p548D3A26.dip0.t-ipconnect.de) |
| 13:40.39 | jm|work | pots+aux, two lines one number: anyone got this setup? |
| 13:42.49 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 13:42.49 | *** join/#asterisk dattolie (n=edgardat@r200-40-173-118.ae-static.anteldata.net.uy) |
| 13:42.50 | monsted | i have 450 lines with one number, but that's all ISDN |
| 13:43.34 | dattolie | hi all .. I'm looging for a "nice" (maybe commercial) IAX Client. can we helpme ? |
| 13:44.07 | *** join/#asterisk oej_ (n=oej@80.187.209.39) |
| 13:44.18 | tcseke | is it possible that although i call MeetMe(,E), i won't join to an empty conference? |
| 13:44.47 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 13:45.20 | *** join/#asterisk Givur (n=mail@vpngw.de.callforcenetwork.eu) |
| 13:45.26 | Givur | Hello all |
| 13:48.21 | qdk | monsted: Does you provide loadbalancing on your ISDN30 lines if i have more that one? and of course dont send to a line which is down (for whatever reason)? |
| 13:52.21 | iCEBrkr | yo yo |
| 13:52.43 | qdk | s/does/do/ |
| 13:53.14 | qdk | s/that/than/ |
| 13:53.26 | qdk | this is going well. |
| 13:53.38 | qdk | lol |
| 13:53.47 | qdk | s/Does/do/ |
| 13:54.12 | monsted | qdk: yes, we do |
| 13:54.19 | monsted | multihomed, too |
| 13:55.46 | monsted | it's based on cisco call manager (sorry) which will pick a random available channel from the gateway (not sure how it chooses) and the PSTN people load balance the calls coming in over all of the available PRI connections |
| 13:56.42 | qdk | monsted: why sorry? its your money. :-P |
| 13:56.46 | monsted | i think i'm up to about 80 E1 lines now |
| 13:56.58 | *** join/#asterisk apardo (n=apardo@87.217.144.94) |
| 13:57.00 | monsted | no, it's the customers money :) |
| 13:57.11 | qdk | monsted: so true. hehe |
| 13:58.50 | qdk | monsted: but i guess i can put up as many boxes as i have lines then, which will provide me with a nice loadbalancing and failover going to and from PSTN. |
| 14:01.01 | *** join/#asterisk juanjoc (n=juanjoc@201.216.212.113) |
| 14:03.16 | *** join/#asterisk zotz (n=zotz@24.244.133.107) |
| 14:03.29 | tom007 | can anyone tell me what I have to do to use the enumlookup function? |
| 14:04.47 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 14:04.48 | *** join/#asterisk ManxPower (n=manxpowe@71-8-11-111.dhcp.leds.al.charter.com) |
| 14:05.19 | *** join/#asterisk alerios (n=alerios@190.24.97.148) |
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| 14:14.40 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 14:14.44 | supjigatr | Hi. |
| 14:15.21 | *** join/#asterisk Cinen (n=Cinen@208.70.20.33) |
| 14:17.06 | Cinen | Hello |
| 14:17.28 | supjigatr | Hi. |
| 14:18.05 | tom007 | hi |
| 14:18.24 | bXi | how would i set the global language for asterisk sound files? |
| 14:18.46 | supjigatr | Where in zapata should echo stuff go to enable echo on channels? Right before channels? |
| 14:21.28 | Qwell | enable echo? |
| 14:21.58 | *** part/#asterisk _PauloS_ (n=PauloS@mail.eletrodireto.com.br) |
| 14:24.32 | supjigatr | Well I have one box that works great but the other has PRI and a bunch of FXO/FXS and it has echo on and I think i have echo commands in wrong order or something. |
| 14:27.33 | *** join/#asterisk olivier__ (n=olivier@82.239.116.113) |
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| 14:29.28 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:33.08 | pif | what does you zap channel return as HANGUPCAUSE for an non-existent number? |
| 14:35.32 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.219.Dial1.SanJose1.Level3.net) |
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| 14:39.33 | kiddy | can anybody tell how astfax is installed ? |
| 14:49.45 | *** join/#asterisk Cinen (n=Cinen@208.70.20.33) |
| 14:50.11 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 14:50.47 | *** join/#asterisk marv[work] (n=timr@br0.asteriasgi.com) |
| 14:52.00 | *** join/#asterisk russel1234 (n=Miranda@60.51.49.255) |
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| 14:58.28 | *** join/#asterisk asdx (n=asdx@200.85.40.178) |
| 14:58.45 | jmesquita | Folks, do any of you have calls in queue_log that are NOT in cdr? |
| 15:03.05 | AndyCap | anyone used one of the iax2 hardphones? |
| 15:03.19 | wasim | AndyCap: sure |
| 15:03.28 | AndyCap | any recommendations or particular suckiness? |
| 15:04.08 | wasim | nah, its generally ok |
| 15:04.10 | AndyCap | wasim: which one did you try? |
| 15:04.49 | *** join/#asterisk spaghetty (n=user@lugbari/people/spaghetty) |
| 15:05.21 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
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| 15:08.11 | mercestes | THe only suckiness I would be concerned about with an IAX Hardphone would be the near requirement to do codec translations to go to most carriers, unless you are using PRI or an IAX VoIP provider. |
| 15:08.42 | mercestes | With Ulaw phones it's pretty easy to stick with Ulaw from phone to provider. |
| 15:08.46 | AndyCap | mercestes: BRI actually. I'm my own provider. :P |
| 15:08.59 | mercestes | Sweet. Should be pretty solid then. |
| 15:11.31 | AndyCap | wondering if I should use my isdn base with an isdn card in the asterisk box or buy the Siemens C450 ip. :-) |
| 15:12.36 | backblue | AndyCap: siemens ip phone? |
| 15:13.06 | AndyCap | backblue: they have a SIP DECT basestation (with one handset) yes |
| 15:13.27 | AndyCap | backblue: leafing through their manual interestingly enough there was a copy of the lgpl there |
| 15:13.34 | AndyCap | seems they use libosip |
| 15:13.51 | AndyCap | had a downloadable toolchain + source for libosip |
| 15:14.58 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.162) |
| 15:15.05 | my007ms | can i use SetGroup to limit outgoing call |
| 15:15.09 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 15:15.10 | backblue | AndyCap: buy a polycom phone. |
| 15:15.29 | my007ms | it's work finr for incoming but can not make it work for outgoing |
| 15:15.54 | AndyCap | backblue: why? |
| 15:16.12 | AndyCap | nobody ever got fired for buying polycom? |
| 15:16.16 | backblue | AndyCap: who it's siemens, in a voip market? :o |
| 15:17.48 | tom007 | hi. seems like compiling asterisk 1.4 beta3 didn't build the ENUMLOOKUP application. what can I do to achieve this? |
| 15:19.30 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:19.30 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:19.34 | wasim | AndyCap: any of the PA168 ones |
| 15:20.40 | AndyCap | wasim: heh, ok. off to see if someone sells them. :) |
| 15:20.42 | *** join/#asterisk ambriento (n=ambrient@201-27-81-188.dsl.telesp.net.br) |
| 15:21.03 | BurtyB | just dont buy a atcom at320.. else your bin will be full :p |
| 15:21.14 | AndyCap | Hehe |
| 15:21.17 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 15:21.31 | wasim | BurtyB: whats wrong with the at320? |
| 15:22.10 | BurtyB | wasim sounds as though its distorting at the other end so no one can hear what your saying |
| 15:22.46 | AndyCap | Ah, no, that's one of the stupidly designed phones, with massive waste of buttons |
| 15:22.57 | BurtyB | looking in the forum its better with older firmware.. which i'm waiting for from china.. but who knows when/if that will arrive |
| 15:23.20 | *** join/#asterisk suma (n=suma@cm53.omega182.maxonline.com.sg) |
| 15:23.36 | *** join/#asterisk jeffg (n=jeffg@pdpc/supporter/active/jeffg) |
| 15:23.42 | suma | hi i bought e400p from digium long back and i could not configure it, can anyone please help me ? |
| 15:23.58 | BurtyB | AndyCap it has a button to say the IP.. makes my life easier since i'm forever in the admin page ;) |
| 15:24.07 | *** part/#asterisk jeffg (n=jeffg@pdpc/supporter/active/jeffg) |
| 15:24.25 | AndyCap | I'm sorry, but I do not want buttons for local ip, router ip etc. |
| 15:24.41 | AndyCap | BurtyB: hehe, different needs I guess. |
| 15:25.17 | mog | <PROTECTED> |
| 15:25.20 | BurtyB | AndyCap "cheap with display" in reality.. tho i'm wishing we had something else as i'm sick of em :) |
| 15:25.21 | AndyCap | better display than the grandscreen budgetone I guess |
| 15:26.06 | AndyCap | whoever implemented the managment interface on the budgetone must be insane. :) |
| 15:26.20 | BurtyB | i was looking at some of the linksys ones since we have some spa901 which work fine |
| 15:27.47 | my007ms | can i use SetGroup to limit outgoing call |
| 15:27.51 | mercestes | Grandstream Budgetone is to VoIP what Gremlin is to vintage italian cars. |
| 15:28.38 | iCEBrkr | I love my Budgetone. |
| 15:28.43 | iCEBrkr | I works and works well. |
| 15:28.53 | iCEBrkr | Of course, I only have 1. |
| 15:29.50 | suma | what is pri_net and pri_cpe ? |
| 15:29.55 | mercestes | iCEBrkr: Oh, so it is getting more one on one supervision. Get 20 of them and then they'll start slacking off and forming unions so you can't fire them. |
| 15:30.18 | my007ms | suma, this is typ if PRI line |
| 15:30.55 | myiagy | suma i don't know if this is the most correct way to explain but.. net would be like the server (net = network), or the carrier.. and cpe would be the other end (your pbx) |
| 15:31.15 | Qwell[] | myiagy: pretty much |
| 15:31.24 | iCEBrkr | mercestes: yea, I don't have that administrative nightmare :P |
| 15:31.37 | suma | my007ms, ic , i have connected ISDN to one of my asterisk box, and it is not able to receive calls can you please help me |
| 15:31.55 | iCEBrkr | mercestes: But I do have 3 Sipura2k's |
| 15:32.07 | file | Qwell[]: YOU |
| 15:32.18 | suma | my007ms, checking with ztcfg -vv says Channel 01: Clear channel (A-law) (Slaves: 01) |
| 15:32.25 | iCEBrkr | Not entirely sure where I got 3 from. I think I left them a dark warm closet or something. |
| 15:32.30 | *** join/#asterisk pincushion (n=anon@rrcs-65-34-29-126.se.biz.rr.com) |
| 15:32.53 | my007ms | suma, check with ut PSTN the setting |
| 15:33.35 | suma | my007ms, did not get you |
| 15:33.36 | asdx | what card can i get? |
| 15:35.52 | spaghetty | how can I disable all kind of cdr logging and take just one into db ? |
| 15:35.54 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-6.rcmdva.fios.verizon.net) |
| 15:36.45 | spaghetty | my asterisk try to write cdr on Master.csv file |
| 15:37.06 | spaghetty | and onto an account file with the name of the user |
| 15:38.04 | intralanman | spaghetty: you looked into cdr_odbc.conf? |
| 15:38.15 | my007ms | suma, wait i will find the link for u |
| 15:38.25 | intralanman | or cdr_mysql if that's more your style |
| 15:38.37 | spaghetty | intralanman: yep cdr to db work well |
| 15:38.51 | spaghetty | i would disable other logging features |
| 15:39.25 | *** join/#asterisk petit-toon (n=petit-to@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 15:39.45 | myiagy | does anyone know if unicall is still under development, or has it been completely droped? |
| 15:39.47 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) |
| 15:40.20 | suma | my007ms, thanks, i will wait |
| 15:41.15 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 15:42.34 | russel1234 | may i ask something about digium ? |
| 15:42.54 | mog | shoot |
| 15:43.33 | russel1234 | apart from pluggin in digium and configuring asterisk to do tdm/pstn calls out...is there any sdk for digium whereby you can develop niche applications ontop of it? |
| 15:43.39 | hoobastooba | Is it possible to have a voice mail box that does not exist on a physical device, present a message indicator light on a physical device? for example i have a company voice mail box which is ext 4000, all of the physical extensions are between 3000 and 3999. When a voice mail is left in 4000 i want it to show up on the phone at extension 3717. Can this be done? |
| 15:43.49 | russel1234 | e.g. building an IVR? |
| 15:44.47 | *** join/#asterisk hohum (n=dcorbe@jomama.interceltelecoms.net) |
| 15:44.47 | mog | russel1234, all of that is in asterisk |
| 15:44.59 | hoobastooba | russel1234: we did our IVR using php and connected it to our database using AGI |
| 15:45.09 | mog | for example building an ivr is easy via dialplan |
| 15:45.11 | mog | or agi |
| 15:45.16 | russel1234 | or develop a calling card prepaid IVR...picks up calls from the E1 line...has some biz logic on it...then terminate the call out via VoIP... |
| 15:45.20 | mog | or writing your own in line asterisk application to do so |
| 15:45.24 | mog | russel1234, astcc |
| 15:45.34 | backblue | mog: ael works better for ivr. |
| 15:45.38 | russel1234 | i see...so u gotta do in asterisk? |
| 15:45.38 | backblue | i think... |
| 15:45.52 | russel1234 | digium is just like an enabler for TDM/PSTN for asterisk? |
| 15:46.07 | mog | is an enabler |
| 15:46.19 | mog | we also maintain and help drive asterisk development |
| 15:46.46 | my007ms | suma, http://kb.digium.com/entry/26/93/ |
| 15:46.58 | russel1234 | hmm okay...cuz i'm using other CTI board at the moment...and for 1 moment was actually looking at digium cuz its much cheaper |
| 15:47.44 | mog | well supporting digium does directly support development on asterisk, pays my paycheck ^_^ |
| 15:47.58 | sivana | mog: does AEL still resolve to extensions logic? |
| 15:48.26 | suma | my007ms, thanks for the link |
| 15:48.52 | mog | yes |
| 15:49.09 | sivana | is there plans for it's own interpreter? |
| 15:49.14 | *** join/#asterisk skirmisha (n=vk@213.144.225.179) |
| 15:49.23 | skirmisha | anyone familar with perl |
| 15:49.26 | mog | why? |
| 15:49.34 | sivana | just curious :) |
| 15:49.43 | mog | i dont see the point |
| 15:50.04 | skirmisha | if i have something like this <sip:2000@1.2.3.4> how can i take the number only using perl regexp |
| 15:50.23 | intralanman | anyone know of any projects working on VXML for asterisk? i heard that there are a couple around |
| 15:50.29 | bXi | skirmisha: the cheapest would be to explode twice |
| 15:50.31 | mog | there are too |
| 15:50.32 | mog | er two |
| 15:50.34 | bXi | once on : and once on @ |
| 15:50.36 | mog | one is a pay project |
| 15:50.40 | mog | the other is opensource |
| 15:50.47 | intralanman | open is good |
| 15:50.49 | mog | both rely on agi, and manager |
| 15:51.21 | intralanman | cheap is good too |
| 15:51.25 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 15:51.33 | intralanman | but philosophically.... open is good |
| 15:51.57 | suma | my007ms, i configured as in the page, but still i'm not receiving incoming calls with my ISDN line, can you please help ? |
| 15:53.23 | my007ms | are u sure from all information from your telco |
| 15:53.55 | pincushion | Suma, can you make outgoing calls? |
| 15:55.23 | suma | pincushion, i did not try that, since i want to get the incoming calls get through |
| 15:56.07 | suma | my007ms, i'm based in singapore and e1 line and configured the zaptel, having loadzone=uk, defaultzone=uk |
| 15:56.13 | ManxPower | suma: the most common reason for that is that the MSN is wrong in the capi.conf file. |
| 15:57.11 | spaghetty | yep |
| 15:57.25 | pincushion | Suma, for me, on my first setup, making outgoing calls worked before I got the incomming calls working. |
| 15:57.42 | suma | pincushion, i c, i will check it out |
| 15:57.56 | suma | ManxPower, i will check into that |
| 15:58.24 | pincushion | To make incomming calls, I had to configure the context correctly and configure an inbound route. |
| 15:58.48 | pincushion | You make also want to try simulating an incomming call if you have that setup. |
| 16:00.55 | suma | pincushion, i made the context to point out to default, i think should not be a problem |
| 16:01.03 | *** join/#asterisk Lord_Pall (i=Flerb@cpe-66-68-44-56.austin.res.rr.com) |
| 16:01.22 | Lord_Pall | Hiyo. Anyone able to answer some pretty rudimentary questions? |
| 16:01.30 | *** part/#asterisk russel1234 (n=Miranda@60.51.49.255) |
| 16:01.39 | aydiosmio | I'd be pretty disappointed if we couldn't. |
| 16:01.44 | Lord_Pall | Ha. |
| 16:01.51 | Qwell[] | can't answer a question that hasn't been asked |
| 16:02.06 | aydiosmio | thanks sophocles |
| 16:02.09 | *** join/#asterisk syzygyBSD_ (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 16:02.26 | Lord_Pall | I'm debating between biting off a full asterisk server, or simply using a spa-3102. I think asterisk is overly feature rich for what I'd like to actualy do, but I'm not familiar enough with this stuff to actually tell |
| 16:03.16 | spaghetty | what you think about something like "unable to create channel of type 'SIP'" |
| 16:03.29 | Lord_Pall | Essentially, I want to setup seamless routing between a telephone number in the UK, and the states. Dialing into a router/asterisk box, which routes out to voip, to a phone number in the states, as well as calling a local number in the states, routing over voip to my router, which dials rings locally, and then forwards to a mobile or other location. |
| 16:03.32 | *** join/#asterisk kumbalae (n=suma@cm53.omega182.maxonline.com.sg) |
| 16:03.39 | *** join/#asterisk plonky (i=dom@80.82.16.205) |
| 16:03.53 | Lord_Pall | I'm pretty sure the 3102 will do this without much fuss, but I'm not totally sure.. |
| 16:04.36 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-42.ph.ph.cox.net) |
| 16:04.39 | Lord_Pall | any commentary or suggestions? |
| 16:04.59 | aydiosmio | you want to use 2 SPA-3102s? |
| 16:05.03 | pincushion | Suma, you may also want to log into the asterisk console "asterisk -rvvv" and see any message that fly across. |
| 16:05.25 | Lord_Pall | I'd prefer not to use 2 |
| 16:05.48 | Lord_Pall | I know voicestick will setup a local access number to dial out, so I can use that on the american side for dialing |
| 16:05.49 | *** join/#asterisk andrebarbosa (n=andrebar@83.240.149.163) |
| 16:06.20 | aydiosmio | oh um |
| 16:06.40 | aydiosmio | so theother side of this is an asterisk box for certain? |
| 16:07.00 | andrebarbosa | hi guys |
| 16:07.01 | Lord_Pall | No, that's the question. Is asterisk overkill for this? I think the 3102 can handle this, but I'm not sure |
| 16:07.18 | aydiosmio | then I'm afraid I don't understand your setup |
| 16:07.20 | andrebarbosa | question: is there a way to change default keyword for parking? |
| 16:07.32 | kumbalae | pincushion, it says BLUE / RED ALARM, what does it mean ? |
| 16:07.34 | andrebarbosa | for example if i dialout using Dial() with T option |
| 16:07.42 | *** part/#asterisk mattfletcher (n=matt@heysham-mail.nshc.co.uk) |
| 16:07.47 | andrebarbosa | i can't use "#" on IVR menus anymore.. |
| 16:07.55 | andrebarbosa | it would park my call |
| 16:07.59 | Lord_Pall | aydio - I don't have a setup yet. that's the point :). I'm debating whether to buy a single 3102 for usage in the UK to handle the routing of incoming and outgoing call, or building an asterisk box to handle the same. |
| 16:08.03 | kumbalae | pincushion, i'm suma my irc got disconnected came with another name |
| 16:08.47 | ManxPower | kumbalae: BLUE alarm means "Call your telco and say "I have a blue alarm." |
| 16:09.34 | aydiosmio | Lord_Pall: I mean what you want to accomplish--so you're just terminating a voip line? an ATA would be fine for that |
| 16:09.34 | pincushion | kumbalae, I have no idea. I would try googling it. |
| 16:09.56 | kumbalae | ManxPower: is it due to the problem with the configuration or problem with the line ? |
| 16:10.18 | ManxPower | blue alarm is normally a TELCO PROBLEM |
| 16:10.21 | aydiosmio | I thought you were setting up a remote ATA so you could move an international call to a local termination across the internet |
| 16:10.30 | Lord_Pall | yeah. That's what I want to do |
| 16:10.32 | *** join/#asterisk syzygyBSD_ (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 16:10.43 | aydiosmio | but |
| 16:10.51 | aydiosmio | then you need two sip devices |
| 16:10.59 | aydiosmio | one remote and one to temrinate |
| 16:11.04 | aydiosmio | terminate |
| 16:11.13 | Lord_Pall | Even if there's a remote number already setup to route one way? |
| 16:11.40 | Lord_Pall | So essentially, voicetick sets up the remote US #. I setup all the hardware on the UK side. |
| 16:13.10 | aydiosmio | oh so your provider is the termination |
| 16:13.21 | aydiosmio | you aren't trying to terminate to POTS |
| 16:13.41 | Lord_Pall | just on one end. |
| 16:14.04 | aydiosmio | by terminate I mean where you connect to the telephone network |
| 16:14.08 | aydiosmio | not your office phones |
| 16:14.12 | *** join/#asterisk SplasPood (n=jwb@nat-loopback.lga4.us.corp.voxel.net) |
| 16:14.13 | Lord_Pall | yeah. That's only on one end. |
| 16:14.18 | Lord_Pall | Here's a specific example. |
| 16:14.28 | aydiosmio | you aren't terminating to POTS, your provider is |
| 16:14.38 | *** join/#asterisk syzygyBSD_ (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 16:14.44 | aydiosmio | an ATA will be fine |
| 16:14.53 | Lord_Pall | aha. Got it. |
| 16:14.57 | Lord_Pall | 3102 a decent one? |
| 16:15.12 | pincushion | I am having problems with outgoing caller id with a t1 and a te207p. It does work with fine with outgoing VoIP calls. Our telco says we aren't sending the cid data. Anyone know what I can try? |
| 16:15.32 | aydiosmio | yeah it's got lots of good features |
| 16:16.27 | Lord_Pall | great. Thank you for the information. Much appreciated |
| 16:17.08 | aydiosmio | no problem |
| 16:18.41 | bXi | the spa3102 ? |
| 16:18.57 | kumbalae | pincushion, how do i know whether the isdn line is connected to my card ? |
| 16:19.17 | pincushion | Do you have lights on the back of the card? |
| 16:19.24 | bXi | all SPA's have 1 issue as far as i know |
| 16:19.33 | Lord_Pall | bxa - yeah |
| 16:19.45 | bXi | the RTP payload length should be 0.020 instead of the default 0.030 |
| 16:19.52 | kumbalae | pincushion, oh, my server is in the data center, is it not possible through the ssh console ? |
| 16:19.55 | Lord_Pall | i dont know what that actually means |
| 16:20.55 | pincushion | kumbalae, I am not sure. I had problems with configuring my t1. I had to use the lights because my telco didn't give me the right settings the first time. |
| 16:21.14 | kumbalae | i c |
| 16:23.03 | *** join/#asterisk tigger27 (n=tigger27@71-8-56-41.dhcp.leds.al.charter.com) |
| 16:25.58 | Lord_Pall | reboot time! |
| 16:29.10 | variable_office | is there a way to fax stuff through asterisk over g.711? |
| 16:29.19 | aydiosmio | hahaha |
| 16:29.37 | jmesquita | Guys, anyone has problems with missing entries on CDR? |
| 16:29.42 | aydiosmio | variable_office: the short answer is yes, the long answer is it's hard to get right. |
| 16:30.03 | *** join/#asterisk anthonyl (i=anthonyl@nat/digium/x-16633c041c998715) |
| 16:30.07 | variable_office | aydiosmio such as what? |
| 16:30.24 | *** join/#asterisk Teeli (n=tili@202.133.65.165) |
| 16:31.07 | *** join/#asterisk timscott (n=tim@S01060016b6e5e09b.ed.shawcable.net) |
| 16:31.19 | aydiosmio | variable_office: look it up, you'll find a bunch of stuff on it |
| 16:31.40 | variable_office | searching "voip fax" didnt help any thats for sure |
| 16:31.57 | aydiosmio | there are some people here who have done VoIP faxing with much frustration and if you hang around I'm sure they'll have some good advice |
| 16:32.02 | dasuberdavid | <PROTECTED> |
| 16:32.47 | aydiosmio | did you try asterisk g711 fax? |
| 16:32.47 | aydiosmio | or are you not using asterisk? |
| 16:32.48 | aydiosmio | oh nm you are |
| 16:32.48 | variable_office | yes, i am using asterisk |
| 16:32.55 | aydiosmio | "Asterisk and fax calls. Fax over IP. In short: Across the Internet even a G.711 codec fax transmission is unpredictable..." |
| 16:32.59 | variable_office | aydiosmio ill definatly hand around, been in here for days, dont plan on leaving :) |
| 16:33.01 | aydiosmio | www.voip-info.org/wiki-Asterisk+fax. |
| 16:33.03 | variable_office | *hang |
| 16:33.05 | aydiosmio | first result:) |
| 16:33.21 | docelmo | aydiosmio I just sent a 4 page fax over asterisk |
| 16:33.26 | docelmo | to digium.. |
| 16:33.31 | docelmo | it works just fine |
| 16:33.37 | variable_office | docelmo what kindof settings do you need for that? |
| 16:33.44 | docelmo | ULAW |
| 16:33.45 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.172.139) |
| 16:33.52 | *** join/#asterisk spaghetty (n=user@lugbari/people/spaghetty) |
| 16:33.53 | aydiosmio | docelmo: I'm not saying it hasn't been done |
| 16:34.00 | variable_office | so just use g.711 and that it? |
| 16:34.03 | aydiosmio | I'm saying it can be a hassle |
| 16:34.12 | *** part/#asterisk timscott (n=tim@S01060016b6e5e09b.ed.shawcable.net) |
| 16:34.13 | docelmo | Well in my config I am using a linksys ATA w/ fax detection and asterisk with a ZAP tel card connected to a vonage ATA |
| 16:34.25 | docelmo | aydiosmio never had a fax fail yet.. |
| 16:34.27 | variable_office | docelmo do you know of any softfax programs that run with voip |
| 16:34.33 | variable_office | ? |
| 16:34.39 | AndyCap | docelmo: what does the fax detection do? |
| 16:35.02 | docelmo | Routes the call internally and forces the ULAW codec out |
| 16:35.21 | docelmo | this way I am not using g729 to try and send a fax which doesnt work worth a shit |
| 16:35.39 | docelmo | you can do a exten => fax,1,bla.. |
| 16:35.43 | variable_office | humm, so you are using an ata as opposed to some kindof softfax program? |
| 16:35.54 | docelmo | but it only works with ZAPtel cards.. I have a TDM400P |
| 16:36.22 | docelmo | bingo.. I have spandsp setup but I dont use it that often unless I am leaving the house |
| 16:36.26 | variable_office | docelmo well for me it wouldnt be going out over pots from my machine, it gets sent somewhere else over g.711 first |
| 16:36.28 | docelmo | this way I can get my faxes remotely |
| 16:36.35 | aydiosmio | I was hearing how the digium cards have clocks that aren't accurate enough for reliable faxing |
| 16:36.49 | docelmo | aydiosmio works fine for me |
| 16:36.52 | aydiosmio | to POTS anyway... that was crazy |
| 16:37.00 | spaghetty | I need to forward a sip call to my asterisk installation to a voip carrier |
| 16:37.03 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 16:37.11 | docelmo | variable_office this is fine but you have to make sure its ULAW 100% of the time to the PSTN.. |
| 16:37.19 | docelmo | if it goes g729 you will NOT be able to fax |
| 16:37.32 | variable_office | docelmo ok, i am reasonably certain about that |
| 16:37.42 | spaghetty | I've use Dial(SIP/skypho/12345,60,r) |
| 16:37.57 | spaghetty | but asterisk cannot open sip channel |
| 16:38.00 | docelmo | if your not you can use me in a week.. I am 100% ULAW and 1 hop to the PSTN |
| 16:38.14 | spaghetty | someone can give me some hits? |
| 16:38.27 | docelmo | spaghetty try DIAL(SIP/12345@skypho,60,r) |
| 16:38.32 | *** join/#asterisk frenzy (n=frenzy@196.46.104.37) |
| 16:38.38 | frenzy | hellowwww all... |
| 16:38.46 | frenzy | cant compile asterisk... /usr/bin/ld: cannot find -lssl |
| 16:39.00 | docelmo | make sure skypho exists as a sip endpoint in sip.conf |
| 16:39.15 | docelmo | frenzy aparently you didnt search GOOGLE! install LIBSSL! |
| 16:39.20 | docelmo | well openssl actually |
| 16:39.22 | spaghetty | yep skypho exists |
| 16:39.26 | docelmo | yer good then |
| 16:39.39 | spaghetty | ok I try to change extension |
| 16:39.56 | frenzy | I have openssl installed |
| 16:40.03 | docelmo | are you sure? |
| 16:40.12 | docelmo | cause if you did it would find it. |
| 16:40.14 | aydiosmio | make sure make sure the config knows where you installed it |
| 16:40.17 | frenzy | openssl-0.9.7f-7.10 |
| 16:40.19 | frenzy | thats what i got |
| 16:40.20 | docelmo | maybe your ld.so.conf isnt setup right |
| 16:40.38 | frenzy | include ld.so.conf.d/*.conf |
| 16:40.57 | docelmo | ahh but was openssl installed in one of the *.conf places? |
| 16:41.02 | docelmo | your new at this right? |
| 16:41.12 | frenzy | yap |
| 16:41.13 | BurtyB | frenzy try installing openssl-devel |
| 16:41.15 | docelmo | figures |
| 16:41.40 | docelmo | thanks BurtyB that was my next question |
| 16:41.51 | frenzy | :P |
| 16:42.00 | xheliox | site* |
| 16:42.05 | xheliox | wrong window :-[ |
| 16:42.17 | *** join/#asterisk BrokenNoze (n=BKNNoze@host86-144-75-200.range86-144.btcentralplus.com) |
| 16:42.45 | jmesquita | Guys, anyone has problems with missing entries on CDR and queue_log?? |
| 16:42.59 | spaghetty | I've try with new extension but the result is the same |
| 16:43.01 | xheliox | Oh I am, just not today. ;) |
| 16:43.03 | mog | <PROTECTED> |
| 16:43.06 | jeremy_g | jmesquita:no if i have one, i alloc new cdrs for them manually |
| 16:43.13 | docelmo | what is the error you get back from Dial? |
| 16:43.31 | spaghetty | unable create channel of type SIP (cause 3 -No route to destination)0000000000000 |
| 16:43.33 | spaghetty | sorry |
| 16:43.40 | docelmo | well there's your problem |
| 16:43.42 | jmesquita | jeremy_g: What do you mean jeremy_g ? |
| 16:43.56 | docelmo | what do you have for host= in your sip.conf for that entry? |
| 16:44.01 | BrokenNoze | When a Dial is successful, if there anyway to carry on in the context when the called party hangs up? |
| 16:44.05 | jmesquita | jeremy_g: I have a problem that I have a lot of entries in queue_log that are not shown in cdr |
| 16:44.13 | jmesquita | jeremy_g: Is that what you are talking about? |
| 16:44.19 | docelmo | BrokenNoze yes use h as your exten |
| 16:44.32 | spaghetty | docelmo: I've it but I use odbc configuration for sip_buddies |
| 16:44.38 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 16:44.39 | spaghetty | can depend on that ? |
| 16:44.43 | docelmo | yes.. |
| 16:44.53 | spaghetty | hehehe |
| 16:44.53 | docelmo | what does extconfig.conf say? |
| 16:45.01 | docelmo | cause realtime is a whole different ball of wax |
| 16:45.02 | BrokenNoze | docelmo: Ah so e.g. 1000,h,<application> |
| 16:45.07 | docelmo | no.. |
| 16:45.17 | iCEBrkr | docelmo: shut up! |
| 16:45.19 | docelmo | h,priority,application |
| 16:45.21 | docelmo | fuck you |
| 16:45.25 | docelmo | :) |
| 16:45.29 | docelmo | and the horse you road in on |
| 16:45.34 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 16:45.36 | spaghetty | just binding to the db tables |
| 16:45.42 | *** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net) |
| 16:45.43 | spaghetty | the configuration should work |
| 16:45.46 | BrokenNoze | so one h exten per context? |
| 16:45.50 | docelmo | yep |
| 16:45.54 | docelmo | BrokenNoze yes |
| 16:45.57 | spaghetty | the user that do the call is register into sip table |
| 16:46.07 | docelmo | spaghetty do you have caching turned on? |
| 16:46.59 | BrokenNoze | brill. thanks. anywhere got a really good dial plan resource for this stuff, have book and voip-info... |
| 16:47.13 | docelmo | bill? |
| 16:47.14 | spaghetty | docelmo: no |
| 16:47.36 | docelmo | spaghetty you should this way you can type sip show peers and see what the peer name is |
| 16:47.42 | docelmo | once registered |
| 16:48.15 | spaghetty | docelmo: no peers are shown |
| 16:48.20 | spaghetty | I try that |
| 16:48.31 | spaghetty | so what the solution for that ? |
| 16:48.32 | caio1982 | does someone knows anything about (or have tested) the res_conv module for audio format conversion inside asterisk? the site seems to be offline so here's the cache: http://216.239.51.104/search?q=cache:Wq7pbtQ8DgUJ:redice.krisk.org/+http://redice.krisk.org/&hl=pt-BR&lr=&client=firefox-a&strip=1 |
| 16:48.43 | caio1982 | would be awesome to teste such automatica audio conversion |
| 16:49.33 | spaghetty | I should preload the peers into sip_buddies table ? |
| 16:49.34 | caio1982 | i wonder if someone here still have a tarball of it in disk :( |
| 16:49.38 | docelmo | spaghetty your realtime is broke |
| 16:49.40 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 16:50.33 | docelmo | kris companies.. I know the guy who runs that site.. I can ask him.. |
| 16:50.48 | docelmo | next time I see him |
| 16:50.58 | spaghetty | docelmo: but i can register myself on asterisk through realtime |
| 16:50.59 | caio1982 | docelmo: great :) |
| 16:51.24 | spaghetty | and extension are keep from table |
| 16:51.29 | docelmo | if you are getting no route to destination then you have something broken |
| 16:51.32 | *** join/#asterisk _PauloS_ (n=PauloS@mail.eletrodireto.com.br) |
| 16:51.58 | caio1982 | docelmo: any idea if that yahoo.com mail is still up for contacting him? |
| 16:52.43 | docelmo | I dont know |
| 16:52.47 | docelmo | I usually just call him |
| 16:52.58 | docelmo | he lives in Sarasota, Florida |
| 16:53.14 | iCEBrkr | I'll go knock on his door |
| 16:53.27 | docelmo | iCEBrkr not before diving 2 hours |
| 16:53.35 | docelmo | driving* |
| 16:53.38 | iCEBrkr | docelmo: I'll take the motorcycle. |
| 16:53.42 | docelmo | sigh |
| 16:53.44 | iCEBrkr | lol |
| 16:53.49 | docelmo | road pancake |
| 16:53.56 | iCEBrkr | gotta catch me first |
| 16:54.16 | iCEBrkr | oh, how I miss the Howard-Franklin bridge. |
| 16:54.23 | docelmo | hehe |
| 16:54.32 | docelmo | Sunshine Expressway |
| 16:54.33 | docelmo | :P |
| 16:54.46 | iCEBrkr | and Bayshore... Bayshore is pretty vacant around 11:00pm |
| 16:55.15 | *** join/#asterisk Waverly360 (n=mirc@209.12.249.243) |
| 16:55.16 | docelmo | unless a cop is lerking |
| 16:55.16 | iCEBrkr | Sunshine is shitty man.. Too many crazy people |
| 16:55.16 | docelmo | i.e. YOU |
| 16:55.16 | iCEBrkr | Cops rarely lurk on Bayshore |
| 16:55.18 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 16:55.18 | Waverly360 | Hey guys, it's been awhile. |
| 16:55.19 | docelmo | shit could have fooled me |
| 16:55.33 | *** join/#asterisk zotz (n=zotz@24.244.133.107) |
| 16:55.41 | iCEBrkr | docelmo: errr, wait, I'm retarded, I meant Bay Side Bridge |
| 16:55.48 | iCEBrkr | From Clearwater to St. Pete |
| 16:56.20 | iCEBrkr | I took Regan for a ride on the bike on one of our first dates. I went rip'n across Bay Side Bridge about 130mph and she started being the shit outta me. |
| 16:56.21 | Waverly360 | Has anyone had any trouble with using * or # to do things like transfer in asterisk? |
| 16:56.51 | Waverly360 | I currently have *5 setup to do blind xfers, and it works great if you're pretty fast in hitting the keys |
| 16:56.51 | docelmo | nope |
| 16:56.57 | docelmo | if everything is setup right |
| 16:57.14 | Waverly360 | but if you hit *, and then wait a second or so, and hit 5, I lose audio on both connections |
| 16:57.26 | Waverly360 | and the call doesn't actually get transferred. |
| 16:57.42 | iCEBrkr | I've never been impressed with Asterisk's transfer |
| 16:57.51 | *** join/#asterisk spaghetty (n=user@lugbari/people/spaghetty) |
| 16:57.57 | Waverly360 | well..I think this goes beyond the transfer feature.. |
| 16:58.08 | *** join/#asterisk ellisdee (n=ellisdee@69.15.174.114) |
| 16:58.11 | _PauloS_ | iCEBrkr, what motorcycle? |
| 16:58.14 | Waverly360 | I can set them up where they can use * for transfer..they need # for something else. |
| 16:58.24 | iCEBrkr | _PauloS_: GSX-R 750 |
| 16:58.29 | *** join/#asterisk jaike (i=jaike@124.106.188.3) |
| 16:59.03 | caio1982 | docelmo: now i AM retarded... that module has been merged into asterisk trunk. thanks anyway :) |
| 17:00.09 | Waverly360 | but it seems wasteful to do that, cause that means I won't be able to setup *1, *2, *3, etc for other things |
| 17:00.19 | _PauloS_ | Nice bike. I used to ride sport bikes. |
| 17:00.47 | iCEBrkr | _PauloS_: It's a 1989 GSX-R 750 :D Oldskewl baby! |
| 17:00.50 | Waverly360 | is there a setting or something that I'm missing where I can change the time asterisk waits for you to hit the right key combination? |
| 17:01.04 | _PauloS_ | Now I have a Yamaha Dragstar |
| 17:01.25 | russellb | Waverly360: featuredigittimeout in features.conf |
| 17:01.49 | iCEBrkr | _PauloS_: http://www.cyberdyne.org/~icebrkr/pics/Motorcycles/Mine/dsc00137.jpg |
| 17:02.14 | iCEBrkr | CunningPike: How metro :D |
| 17:02.29 | CunningPike | :P |
| 17:02.57 | CunningPike | I went from spending $15 a day to get to work to $5 a week |
| 17:03.04 | iCEBrkr | Yeah |
| 17:03.17 | iCEBrkr | I can get about 15 trips to/from work on my motorcycle for $10 |
| 17:03.44 | iCEBrkr | That's if I don't tear off the line at every light :D which is hard NOT to do. |
| 17:03.56 | _PauloS_ | iCEBrkr , and a lot of fun... |
| 17:04.02 | iCEBrkr | approx 130mpg on 4gals |
| 17:04.09 | iCEBrkr | err |
| 17:04.15 | iCEBrkr | 130miles with 4gals |
| 17:04.26 | iCEBrkr | I think I can squeeze more out if I ride normal. |
| 17:04.38 | docelmo | iCEBrkr schmuck |
| 17:04.39 | *** join/#asterisk |Johny| (n=gomesper@bacus.corp.fccn.pt) |
| 17:04.40 | Waverly360 | russellb: is that in seconds? |
| 17:04.43 | iCEBrkr | docelmo: eat one. |
| 17:04.51 | docelmo | Not my job |
| 17:05.02 | iCEBrkr | docelmo: Here fillout this app :P |
| 17:05.14 | docelmo | You know there is a Asterisk Users Group in Philadelphia Area now |
| 17:05.18 | *** join/#asterisk buckrey (n=buck@218.111.232.41) |
| 17:05.26 | iCEBrkr | docelmo: Is this your fault? |
| 17:05.34 | monsted | can someone lend me a shotgun? |
| 17:05.37 | vader-- | docelmo really? |
| 17:05.41 | docelmo | yes |
| 17:05.46 | vader-- | whats the info on it? |
| 17:05.52 | vader-- | thats near me |
| 17:05.54 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 17:05.54 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:05.58 | monsted | i need to shoot the bastard who dug through our ISDN lines... |
| 17:06.00 | iCEBrkr | docelmo: I was thinking about starting one here in the Ft. Lauderdale area. :D |
| 17:06.13 | vader-- | i work in wilmington de and i used to live in philly now i live in nj |
| 17:06.16 | vader-- | hehe |
| 17:06.17 | iCEBrkr | docelmo: I'll just travel the country starting Asterisk Users Groups |
| 17:06.27 | docelmo | http://asteriskpbx.meetup.com/4/ |
| 17:06.31 | iCEBrkr | 4? |
| 17:06.37 | iCEBrkr | Who started the other 2? |
| 17:06.38 | docelmo | Vader I live in Newark |
| 17:06.39 | *** join/#asterisk slayer192 (n=slayer19@66.138.39.225) |
| 17:06.44 | docelmo | DE |
| 17:06.54 | vader-- | oh ok cool |
| 17:06.55 | russellb | Waverly360: milliseconds i believe |
| 17:06.59 | iCEBrkr | haha that's awesome |
| 17:07.00 | vader-- | i work for a private school |
| 17:07.04 | docelmo | nice |
| 17:07.04 | vader-- | you probably heard of them |
| 17:07.07 | docelmo | I doubt it |
| 17:07.09 | russellb | Waverly360: default is 500 ms |
| 17:07.10 | vader-- | sallies? |
| 17:07.11 | docelmo | I just moved it |
| 17:07.13 | docelmo | here |
| 17:07.13 | iCEBrkr | 2 is NYC and 3 is NorCal |
| 17:07.22 | docelmo | 4 is me and 5 is orlando |
| 17:07.29 | Waverly360 | oh, ok. Lemme see if that helps my problem. |
| 17:07.44 | iCEBrkr | docelmo: and who started #1 and got asteriskpbx on Meetup? |
| 17:07.47 | iCEBrkr | <== I DID! I DID |
| 17:07.51 | iCEBrkr | :D |
| 17:07.51 | iCEBrkr | lol |
| 17:07.57 | docelmo | uhh piss off wanker |
| 17:07.57 | intralanman | anyone happen to know right off.... can sipp send dtmf? |
| 17:08.01 | iCEBrkr | haha |
| 17:08.11 | iCEBrkr | docelmo: Like I said, I'll just get them started, other people can run them. |
| 17:08.16 | iCEBrkr | *cough*Matt*cough* |
| 17:08.32 | docelmo | I was for the longest time.. actually I still am.. |
| 17:08.34 | docelmo | :) |
| 17:08.54 | iCEBrkr | ha |
| 17:09.24 | docelmo | ahh well |
| 17:09.33 | iCEBrkr | That's crazy 55 people signed up |
| 17:09.36 | iCEBrkr | for the St. Pete one |
| 17:09.38 | docelmo | but yes.. There is one in philly well Wilminton |
| 17:09.51 | docelmo | once our office is setup we will host it in Newark, DE |
| 17:09.57 | docelmo | its fairly close for everyone |
| 17:10.09 | docelmo | hell we had guys driving 4 hours to come to the one in St. Pete |
| 17:10.19 | iCEBrkr | Yea, well.. ALan is a nutjob |
| 17:10.27 | docelmo | no comment |
| 17:10.35 | *** part/#asterisk jaike (i=jaike@124.106.188.3) |
| 17:10.42 | iCEBrkr | When he called me telling me he was driving all the way to St. Pete.. I was like WHAAAAA? |
| 17:11.06 | docelmo | ya.. well he wanted to be around some of the grand masters of Asterisk |
| 17:11.19 | iCEBrkr | docelmo: I'm still not sure where I fit in with all this crap. Him and Tom want to invest some money in something and I felt like I was being interviewed the night he took Regan and I out to dinner |
| 17:11.45 | docelmo | hehe |
| 17:11.46 | iCEBrkr | docelmo: Good thing Matt was there, otherwise Alan would have been disappointed. :P |
| 17:11.53 | docelmo | piss off |
| 17:11.56 | iCEBrkr | lol |
| 17:12.00 | docelmo | wanna be asterisk boy |
| 17:12.17 | iCEBrkr | Wanna be? |
| 17:12.24 | iCEBrkr | Pfft, I make Asterisk do jumping jacks. |
| 17:12.43 | docelmo | ya.. uhh ok |
| 17:12.52 | iCEBrkr | Well as far as dialplans go. |
| 17:13.07 | docelmo | you know if people didnt know us they would think we were fighting |
| 17:13.08 | docelmo | haha |
| 17:13.11 | mercestes | funny, I don't recall you being all that helpful, iCEBrkr. |
| 17:13.17 | iCEBrkr | docelmo: Yea, really. :D |
| 17:13.17 | docelmo | well they probably dont know.. |
| 17:13.23 | BurtyB | users suck... they complain "it doesnt work" yet dont explain why... i'm going to get my chainsaw... |
| 17:13.33 | iCEBrkr | docelmo: Well, you know how friends are, they constantly give each other shit |
| 17:13.39 | docelmo | yep.. |
| 17:13.42 | docelmo | check your cell phone |
| 17:13.43 | docelmo | haha |
| 17:13.45 | iCEBrkr | mercestes: I didn't say anything about being helpful :P |
| 17:13.55 | mercestes | lol....Good thing..;) |
| 17:13.56 | iCEBrkr | docelmo: when I get some $$ to pay the fucker off |
| 17:14.05 | docelmo | ohh its shut off |
| 17:14.12 | iCEBrkr | yeah |
| 17:14.15 | docelmo | well shit |
| 17:14.27 | *** part/#asterisk _PauloS_ (n=PauloS@mail.eletrodireto.com.br) |
| 17:14.27 | docelmo | you need to quit yer job and move back to tampa |
| 17:14.27 | iCEBrkr | docelmo: I got screwed on selling my old phone to a friend |
| 17:14.41 | iCEBrkr | docelmo: I intend on moving back to Tampa if things don't shape up over here. |
| 17:14.48 | iCEBrkr | I can't even call this place home, it just doesn't feel right |
| 17:14.53 | iCEBrkr | It feels very temporary |
| 17:14.54 | docelmo | I hear orlando might be hiring |
| 17:14.55 | docelmo | soom |
| 17:15.04 | docelmo | soon |
| 17:15.10 | iCEBrkr | F Orlando man.. |
| 17:15.14 | iCEBrkr | I couldn't live there |
| 17:15.16 | iCEBrkr | TOOOOO many people |
| 17:15.39 | docelmo | hello tampa? |
| 17:15.48 | iCEBrkr | Tampa is more spread out |
| 17:15.59 | iCEBrkr | If you have to go from one side of Orlando to the other, you're screwed. |
| 17:16.05 | docelmo | I dunno.. Tons of people in Town & Country |
| 17:16.06 | iCEBrkr | Traffic there is worse than it is here. |
| 17:16.14 | iCEBrkr | I want to go back to Tampa |
| 17:16.17 | iCEBrkr | I liked it there |
| 17:16.29 | Waverly360 | russelb: It doesn't seem to be changing anything. if I wait more than about a second to hit 5 after the *, I lose audio, and the ability to transfer or anything on that call. |
| 17:16.38 | docelmo | DrkShdw I used to live 2 miles from the Airport |
| 17:16.47 | Waverly360 | does it go into the featuremap section? or a different part? |
| 17:16.48 | docelmo | I now live 30 minute from the Philadelphia airport |
| 17:17.10 | iCEBrkr | DrkShdw: I used to live off 4th St and Gandy, just down the street from Howard-Franklin bridge. |
| 17:17.33 | iCEBrkr | DrkShdw: Oh, Ellenton area? or however you spell it |
| 17:17.42 | DrkShdw | iCEBrkr: ahh, cool. we used to ride every tuesday night across the franklin bridge, the causeway, and a few others |
| 17:17.44 | iCEBrkr | oh wait. Hour and a half.. I read that as Half-an-hour |
| 17:17.53 | iCEBrkr | DrkShdw: yup yup! |
| 17:17.57 | DrkShdw | iCEBrkr: Port Charlotte, it's in between sarasota and fort myers |
| 17:18.20 | iCEBrkr | DrkShdw: I road with the Clearwaterbiker folks and then later Seductive Cycles guys |
| 17:18.26 | DrkShdw | small retirement town, short drive from either of the others |
| 17:18.54 | docelmo | DrkShdw ya.. where all the old people live |
| 17:19.13 | DrkShdw | iCEBrkr: cool! I haven't gone in like 4+ years.. I wasn't with any group.. but about 15-20 of us would hit tampa/stpete/clearwater and join up with the massive groups |
| 17:19.27 | iCEBrkr | DrkShdw: Yeah, it's crazy |
| 17:19.54 | DrkShdw | I remember some nights, when it literally looked like bikeweek in daytona.. 300+ bikes |
| 17:20.00 | iCEBrkr | haha |
| 17:20.26 | DrkShdw | iCEBrkr: did you just ride, or were you one of the stunters? |
| 17:20.36 | iCEBrkr | DrkShdw: 'The Lot' in Clearwater on Saturday nights is like that.. 200 bikes just show up and hang out in a strip-mall parking lot.. Then around 11 they roll out 15-20 bikes at a time. |
| 17:20.58 | iCEBrkr | DrkShdw: I just rode. It was always a social event for me. Just to get out, ride a bit and meet and talk with people |
| 17:21.18 | docelmo | or find your better half] |
| 17:21.26 | iCEBrkr | docelmo: No, that's what MySpace is for. :P |
| 17:21.33 | docelmo | sigh |
| 17:21.35 | DrkShdw | thats cool. we have a small group of like 50-80 bikes that get together each week locally.. but the cops chase us off nearly every week. there is a small pack of like 5 stunters, and they ruin it for the rest |
| 17:22.25 | iCEBrkr | welp. lunch time. |
| 17:22.30 | DrkShdw | we meet at a hooters in venice, and roll out usually as 2-3 subgroups. fun times |
| 17:22.32 | iCEBrkr | DrkShdw: take care man.. ttyl |
| 17:22.35 | DrkShdw | enjoy |
| 17:23.14 | *** part/#asterisk tigger27 (n=tigger27@71-8-56-41.dhcp.leds.al.charter.com) |
| 17:25.13 | *** join/#asterisk notmebutHim (n=dpobanz@h162-040-102-198.adsl.navix.net) |
| 17:27.29 | Waverly360 | Oh..I do have one more quick question regarding the Polycom 501/601s. |
| 17:28.09 | Waverly360 | On a 601, I know how to order the list on the sidecar, but is there a way to customize the order of the list that shows up when you hit the Buddies softkey? |
| 17:28.18 | Waverly360 | By default it looks like it just orders people by their last names. |
| 17:30.48 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 17:34.27 | russellb | Waverly360: that sounds like a bug in an older version ... you probably need to update |
| 17:36.35 | Waverly360 | russellb: Are you referring to the blind xfer question from earlier? |
| 17:37.23 | *** join/#asterisk Falconix (n=anders@194-237-185-7.customer.telia.com) |
| 17:37.41 | russellb | yes, where you get no audio .. |
| 17:41.09 | Waverly360 | hmm |
| 17:41.14 | *** join/#asterisk [hC] (n=hardcore@S01060018394c6be3.vc.shawcable.net) |
| 17:41.28 | Waverly360 | hah. |
| 17:42.49 | *** join/#asterisk _alex_mx_ (n=alex@200.78.229.18) |
| 17:44.29 | *** join/#asterisk bkw_ (n=brian@ppp-70-128-126-144.dsl.tulsok.swbell.net) |
| 17:48.39 | *** join/#asterisk klibza (n=klibza@bny92-5-82-232-58-135.fbx.proxad.net) |
| 17:49.35 | *** join/#asterisk adorah (n=admin@87.68.145.189.cable.012.net.il) |
| 17:56.50 | Waverly360 | russellb: had the featuredigittimeout in the wrong place. It seems to have fixed my problem. Thanks a ton :) |
| 17:58.34 | frenzy | I need some help with my new tdm 2400p.. I've got it all configured but I cant make calls or receive... |
| 17:58.52 | frenzy | I've got 2 quad FXO's on slot 5 and six and using a break out box |
| 17:59.07 | frenzy | i've configured channels 17-24.... correct? |
| 17:59.42 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 18:03.04 | *** join/#asterisk olinux (n=olinux@starbucks.wellspublishing.net) |
| 18:04.34 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 18:06.38 | frenzy | ? |
| 18:07.23 | *** join/#asterisk rbw (n=rbw@c-a3ace253.360-1-64736c10.cust.bredbandsbolaget.se) |
| 18:08.47 | *** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it) |
| 18:10.28 | *** join/#asterisk eltech (i=G00Ds@ool-457c9421.dyn.optonline.net) |
| 18:11.10 | *** join/#asterisk JustinWick (n=jwick@unaffiliated/jpl/jpl-justin) |
| 18:13.06 | frenzy | anyone around? |
| 18:14.21 | frenzy | I have configured TDM 2400p per the digium site guides however i cannot make or receive calls |
| 18:16.07 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 18:17.04 | Waverly360 | sorry..I don't know anything about that frenzy |
| 18:18.16 | frenzy | never mind.. just rebooted machine and seems to be wroking |
| 18:19.33 | *** join/#asterisk Jamez^7 (n=martini@modemcable131.214-131-66.mc.videotron.ca) |
| 18:26.42 | *** join/#asterisk tumyp (n=tumyp@222-33.ip.tps.uz) |
| 18:27.49 | BrokenNoze | anyone able to help me with a sipgate problem? |
| 18:28.00 | jmesquita | Guys, anyone has problems with missing entries on CDR and queue_log?? |
| 18:28.09 | tumyp | Hi |
| 18:28.33 | tumyp | is there anyone who can help me with zaptel and answer detection ? |
| 18:29.05 | file | tumyp: on analog? |
| 18:29.24 | tumyp | yep |
| 18:29.40 | tumyp | I've wctdm24 |
| 18:29.45 | file | reliable answer detection that works for every country? |
| 18:29.48 | tumyp | from digium |
| 18:30.15 | tumyp | equipment is in US |
| 18:30.28 | tumyp | usual analog lines |
| 18:30.35 | tumyp | (sorry for my english) |
| 18:30.36 | file | well, for the most part there is no answer detection... no call progress detection... |
| 18:30.46 | file | there is experimental capability but it may hang up your calls randomly or not work properly |
| 18:30.56 | tumyp | yes, I saw it |
| 18:31.08 | tumyp | it answers after second "Hello" :) |
| 18:31.43 | tumyp | is there any other way to detect answers ? |
| 18:31.55 | Strom_C | not unless you order answer supervision on your analog lines |
| 18:32.20 | tumyp | is it real service ? |
| 18:32.29 | Strom_C | yes, it's a real service |
| 18:32.36 | tumyp | how to explain it to phone company ? |
| 18:32.38 | *** join/#asterisk LoneShadow (n=duh@c-67-188-235-220.hsd1.ca.comcast.net) |
| 18:32.49 | Strom_C | "Hi, I'd like to order answer supervision on my analog lines" |
| 18:32.52 | tumyp | they are don't understand me :) |
| 18:33.16 | tumyp | they are even do not know what is disconnect supervision ... |
| 18:33.30 | tumyp | thanks Strom_C |
| 18:33.32 | tumyp | :) |
| 18:33.43 | tumyp | I'll ask them |
| 18:33.54 | *** join/#asterisk xnon (i=xnon@200.8.30.50) |
| 18:33.59 | *** join/#asterisk riksta (n=rick@89.240.128.13) |
| 18:34.18 | tumyp | but how it works ? |
| 18:34.29 | tumyp | through polarity switch ? |
| 18:34.36 | tumyp | or somehow else ? |
| 18:35.41 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 18:40.28 | *** part/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 18:40.39 | tumyp | does anyone used this "answer suprvision" |
| 18:40.47 | tumyp | is it expensive service ? |
| 18:41.14 | *** join/#asterisk cbm11211 (n=Administ@66.28.182.170) |
| 18:41.59 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
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| 18:47.20 | Strom_C | tumyp: how many telephone lines do you have |
| 18:47.29 | tumyp | 24 |
| 18:47.32 | jeremy_g | :D |
| 18:47.38 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:47.49 | jeremy_g | tumyp:one sures has a lot of options if it comes to calling u |
| 18:47.51 | Strom_C | tumyp: you should really replace those with an ISDN PRI line |
| 18:48.09 | Strom_C | that will solve your answer supervision problem and wind up saving you money |
| 18:48.33 | tumyp | with my pleasure .... but I can;t |
| 18:49.06 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 18:50.22 | tumyp | Strom_C: I can not change this, I should solve this situation with analog lines |
| 18:50.22 | *** join/#asterisk Bpedersen (n=bp@82-192-171-74.sk.dsl.struer.net) |
| 18:50.22 | Strom_C | why can't you change it? |
| 18:51.22 | tumyp | it's hard and long to explain :) |
| 18:51.22 | Strom_C | give me the short short version |
| 18:51.22 | CunningPike | North American ring cadence is 6 seconds, right? |
| 18:51.22 | tumyp | money |
| 18:51.22 | Strom_C | tumyp: usually a PRI is /cheaper/ than 24 analog lines |
| 18:51.23 | tumyp | :) |
| 18:51.23 | Strom_C | CunningPike: typically 2 seconds on, 4 seconds off |
| 18:51.48 | CunningPike | Strom_C: OK - thanks. That's what I thought..... |
| 18:58.33 | Bpedersen | Is there a way to get the original callerid over when you the bristuff command pickupchan? |
| 18:59.28 | Bpedersen | like if i pickup a call on extension 504 from another extension right now i get 504 as callerid |
| 19:02.04 | notmebutHim | I have 2 * servers with identical .conf files. Why can I sip phones to register with 1 but not the other? I use to be able to get them to register with either server. |
| 19:04.32 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 19:04.44 | mercestes | notmebutHim: Sip.conf or mysql realtime? |
| 19:04.55 | *** join/#asterisk shellshark (n=x86@get.hooked.on.voip.with.shellshark.net) |
| 19:04.58 | notmebutHim | Sip.conf |
| 19:05.35 | mercestes | notmebutHim: Not sure, then, honestly. |
| 19:05.52 | *** join/#asterisk klapzin (n=esdras@200.228.53.40) |
| 19:05.57 | klapzin | hi all |
| 19:06.03 | klapzin | how i change the flash time ? |
| 19:06.10 | Qwell[] | flash time? |
| 19:06.32 | klapzin | i add a line flash = 150 in zapata.conf and dont works right |
| 19:07.23 | *** part/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 19:08.21 | notmebutHim | sip is using port 5060. How could I tell if it was closed or in use by somethings else? |
| 19:10.06 | mercestes | nmap perhaps. |
| 19:12.58 | *** join/#asterisk Luke-Jr (n=luke-jr@CPE-24-31-245-218.kc.res.rr.com) |
| 19:13.34 | shellshark | nmap -sU -P0 -T INSANE 127.0.0.1 |
| 19:13.37 | *** join/#asterisk kilobit2001 (n=locid@206-248-158-84.dsl.teksavvy.com) |
| 19:13.38 | shellshark | should suffice :) |
| 19:14.00 | notmebutHim | I'll try it. |
| 19:14.27 | *** join/#asterisk champster (n=asterisk@AH.tescogroup.com) |
| 19:14.33 | kilobit2001 | How can I wait 5sec for keypress in dialplan? the wait command does not listen to keypress |
| 19:14.54 | *** part/#asterisk Teeli (n=tili@202.133.65.165) |
| 19:15.05 | Qwell[] | waitexten? |
| 19:15.13 | *** join/#asterisk dasenjo (n=dasenjo@201.228.128.10) |
| 19:15.46 | champster | Me TE411P has the VPM400 echo canceller. What is Digium's policy regarding upgrading it to VPM450M? Any costs? |
| 19:15.49 | notmebutHim | Should nmap be listing port 5060? |
| 19:16.04 | mercestes | Yes.... |
| 19:16.34 | Qwell[] | champster: I recommend calling sales and asking them. |
| 19:17.07 | mog | champster, we have a policy for it |
| 19:17.12 | mog | but im not familiar with it |
| 19:17.20 | mog | like Qwell said sales has got your back |
| 19:17.20 | champster | ;-) |
| 19:18.01 | notmebutHim | I'm only seeing 4 ports listed. 111 631 732 and 735. However I am ssh into on port 22 which is not listed so!!!! |
| 19:18.42 | mercestes | hmm. |
| 19:18.50 | mercestes | Give me yoru IP I can Nmap you...lol |
| 19:19.07 | notmebutHim | wow. thoughtful! |
| 19:19.19 | mercestes | What? It'd be my pleasure..:) |
| 19:21.03 | notmebutHim | I'll do a little reading about the nmap options. Maybe I specified something wrong. |
| 19:21.12 | mercestes | As you wish. |
| 19:23.10 | *** join/#asterisk ikey (i=ikey@220.226.59.241) |
| 19:23.16 | *** join/#asterisk henrique (n=henrique@201-1-130-79.dsl.telesp.net.br) |
| 19:24.05 | Bpedersen | does anyone know if its posible to keep the original callerid when using the Pickup command or bristuff pickupchan? |
| 19:27.40 | mercestes | kilobit2001: background(silence/10) tends to work too |
| 19:30.15 | dattolie | all... somebody have a Digium TDM2400 ? |
| 19:31.07 | *** join/#asterisk RoyK (n=roy@ti211310a080-6757.bb.online.no) |
| 19:33.24 | kilobit2001 | in the dialplan, if i press 2 , and that extension does not exist, the call is hangup. How can i create an extension 2, that does nothing |
| 19:36.24 | kristalino | is it possible to have a free phone number ? |
| 19:36.28 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 19:37.12 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 19:38.53 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 19:40.44 | mercestes | kilobit2001: Why do you want an extension 2 that doesn't do anything? |
| 19:44.56 | iCEBrkr | kilobit2001: create an 'i' extension... |
| 19:45.29 | monsted | ooh, 8-port FXO SIP gateway :) |
| 19:45.29 | iCEBrkr | kilobit2001: 'i' stands for invalid. So anytime an invalid 'extension' is entered, it'll land in 'i' and you can do your logic there |
| 19:45.32 | mercestes | an I extension with a goto preferably. |
| 19:45.42 | monsted | gotta love people who buy stuff they can't actually use ;) |
| 19:45.43 | mercestes | maybe back to teh beginning of yoru menu. |
| 19:45.44 | iCEBrkr | mercestes: exactly. :P |
| 19:45.56 | mercestes | Or i,1,goto(i,1) that would do nothing. |
| 19:45.58 | mercestes | for a very long time. |
| 19:46.03 | iCEBrkr | haha |
| 19:46.54 | mercestes | Here is a code snippet. i,1,NoOp(I was pwned by Mercestes) i,2,goto(i,1) |
| 19:47.02 | mercestes | problem solved. |
| 19:47.08 | iCEBrkr | oh dear. |
| 19:48.18 | Nugget | heh |
| 19:48.21 | *** join/#asterisk cekc (n=cekc@66-17-9-220.biz.bkfd.arrival.net) |
| 19:48.33 | cekc | help! I have someone stuck in a queue how do I get them out from the CLI? |
| 19:48.48 | mercestes | Or i,1,Dial(Zap,G1,2023243000) i,2,goto(i,1) |
| 19:49.44 | *** join/#asterisk topping (n=topping@natip.kink.com) |
| 19:49.49 | *** join/#asterisk alerios (n=alerios@190.24.97.148) |
| 19:50.05 | *** join/#asterisk _Vile (i=vile@bc181104.bendcable.com) |
| 19:55.34 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 19:57.18 | *** join/#asterisk RogueCmdr (n=RogueCmd@ip72-201-30-198.ph.ph.cox.net) |
| 19:58.23 | RogueCmdr | I am wondering who is using the grandstream GXP 2000 phones.. Yes i know they are bad but its what my coomapny wants to use.. I need to find out how to make it show when the line is in use |
| 19:59.08 | cekc | I am using the GXP-2000 and I love them. They are a big headache sometimes, but for the price you can't beat them |
| 19:59.24 | RogueCmdr | yea well could ya help me.. |
| 19:59.33 | RogueCmdr | anychance you can PM me? |
| 19:59.35 | cekc | when I was using Trixbox the lights worked fine, but I haven't figured it out on plain asterisk yet |
| 19:59.45 | hads | hints |
| 19:59.48 | RogueCmdr | yea im using trix box right now |
| 19:59.58 | Bpedersen | then it works out of the box |
| 20:00.02 | RogueCmdr | its just i had a bitch session just now |
| 20:00.06 | cekc | my problem is I don't fully understand contexts and stuff yet |
| 20:00.36 | cekc | RogueCmdr: go to their web control panel thingy and setup extensions on each button and set it to BLF |
| 20:00.38 | RogueCmdr | and i got line one on another pbx and line 2 on the old systems is rolled over to the new system.. but they are complaining (the women) about the issues of that |
| 20:00.41 | RogueCmdr | and other stuff |
| 20:01.13 | *** join/#asterisk xnon_ (i=xnon@200.8.30.50) |
| 20:01.29 | cekc | I was getting a lot of bitching and then dumped trixbox and put in plain asterisk and haven't had problems since. unfortunately all the cool stuff I could do with trixbox I can't figure out with asterisk |
| 20:02.12 | RogueCmdr | had no issues with ether just trix box was easlily setup |
| 20:02.30 | cekc | I just learned AGI and now I have PHP scripts that can talk to asterisk, it's pretty neat |
| 20:02.46 | cekc | RogueCmdr: are you using that feature already where the phones get their configuration from the server? |
| 20:03.09 | cekc | RogueCmdr: also, you can probably get better help in #trixbox |
| 20:03.18 | cekc | or wait |
| 20:03.21 | cekc | it's #freepbx |
| 20:03.30 | RogueCmdr | alright thanks |
| 20:07.26 | *** join/#asterisk malcolmd (i=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 20:07.31 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 20:09.22 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 20:12.56 | *** join/#asterisk tbic (n=tbic@protious.fciautomation.com) |
| 20:13.09 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:13.41 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-228-131.dc.res.rr.com) |
| 20:13.58 | blitzrage | is it possible to disable the ability to dump a call with the star (*) key in the AgentLogin() app? |
| 20:21.16 | *** join/#asterisk zotz (n=zotz@24.244.133.107) |
| 20:21.49 | *** part/#asterisk henrique (n=henrique@201-1-130-79.dsl.telesp.net.br) |
| 20:23.18 | jeremy_g | computability theory |
| 20:24.59 | *** join/#asterisk pids (n=pids@dsl081-072-084.sfo1.dsl.speakeasy.net) |
| 20:26.31 | *** join/#asterisk ManxPower (n=manxpowe@41.sub-75-200-83.myvzw.com) |
| 20:27.31 | notmebutHim | No wonder my sip phones would not register. port 5060 and 5061 are closed. (Thanks shellshark and mercestes!) |
| 20:28.09 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:28.25 | mercestes | no problem. |
| 20:28.36 | notmebutHim | I'm running CentOS 4. can someone give me a pointer on how to open ports? |
| 20:28.51 | ManxPower | I'm trying to find a SIP Softphone for Win32 that sucks less than the rest. |
| 20:28.57 | ManxPower | Any ideas other than X-Lite? |
| 20:29.00 | mercestes | iptables or pf? |
| 20:29.46 | ManxPower | notmebutHim: one of the following "service ipchains stop" or "service iptables stop" or "service firewall stop". |
| 20:33.45 | blitzrage | iptables -I RH-Firewall-1-INPUT -p udp --dport 5060 -j ACCEPT |
| 20:33.49 | blitzrage | service iptables save |
| 20:34.11 | blitzrage | don't forget RTP |
| 20:34.26 | blitzrage | iptables -I RH-Firewall-1-INPUT -p udp --dport 10000:20000 -j ACCEPT |
| 20:34.39 | blitzrage | modify range as required in rtp.conf |
| 20:35.01 | blitzrage | 1 simultaneous call = 4 rtp ports (in/out RTP, in/out RTCP == 4) |
| 20:36.51 | shellshark | unless you use IAX2, where everything is combined :) |
| 20:36.54 | blitzrage | of course -- but then you have to deal with IAX2 :) |
| 20:37.27 | ManxPower | Since Asterisk doesn't support RTCP.... |
| 20:37.51 | *** join/#asterisk RoyK (n=roy@ti211310a080-6757.bb.online.no) |
| 20:38.13 | pids | Anyone have any way to make automon start recording automatically and to allow the user to stop recording by pressing the start stop key? |
| 20:39.16 | pids | Using senddtmf doesnt work since it only sends to the audio channels and not * itself. |
| 20:39.28 | blitzrage | ManxPower: yet :) |
| 20:44.44 | *** join/#asterisk postel (n=jp@wikimedia/Postel) |
| 20:49.44 | *** join/#asterisk dioedu (n=dioedu@201.7.117.114) |
| 20:49.52 | dioedu | hello all |
| 20:50.11 | blitzrage | hoi hoi |
| 20:50.18 | dioedu | i have a queue with agentcallbacklogin |
| 20:50.30 | blitzrage | haha -- nice, I'm teaching that right now :) |
| 20:50.43 | blitzrage | login with exten, pass, then hit # when you hear "extension..." to logout |
| 20:50.46 | blitzrage | is that your Q? :) |
| 20:50.58 | file | blitzrage: pfft - you don't teach things... you just yell at random people on the street |
| 20:51.05 | blitzrage | file: sometimes true |
| 20:51.11 | dioedu | but i had problem when the agents use the DND button of softphone |
| 20:51.19 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 20:51.27 | blitzrage | chan_agent has some ways to go before its really ready... |
| 20:52.27 | dioedu | i don't know what i need to do with the call in this case.. i know that asterisk receive a BUSY state |
| 20:52.49 | blitzrage | if thats the case, then you should be able to parse on the BUSY and perform DP logic |
| 20:52.53 | blitzrage | ~dp |
| 20:52.54 | jbot | from memory, debianplanet is http://www.debianplanet.org: Free Beer for the users of the Free Linux of a Free Planet. Debian community site. Founders: robster + Netsnipe Editors: agl, solomon, flugh Uberguru: ZaNaGa, or DP present no content, in a tactless manner that is un-paralleled by other site that lack talent |
| 20:53.31 | dioedu | but in this context, i don't have the 'dst' anymore |
| 20:54.08 | hoobastooba | i am trying to add a queue member to a queue dynamically. but, the call to the agent goes out across IAX to another server that dials the number on a local T1... i have add queue member Zap/g1/<my cell number> to helpdesk. that works. but zap/g1 sends out on the ld t1 on that server. what would the equivalent be using iax? when normally dialing a local number this is my exten => exten => _NXXXXXX,n,Dial(IAX2/REC_SERVER_Local_Progrexion_Outbound:sphero |
| 20:54.19 | hoobastooba | i have tried a few things and am getting nowhere. |
| 20:54.32 | dioedu | what is DP logic ? |
| 20:54.47 | blitzrage | dialplan |
| 20:54.58 | blitzrage | i.e. extensions.conf |
| 20:56.09 | *** join/#asterisk SomethingISODD (n=dan@h109.42.63.69.cable.ottr.cablerocket.net) |
| 20:56.11 | SomethingISODD | Hello all |
| 20:56.14 | dioedu | http://pastebin.ca/243731 |
| 20:56.29 | SomethingISODD | does anyone know of any good tut`s on setting up Freeradius and setting up asterisk to work with it |
| 20:56.52 | dioedu | this is the part of my extensions that treat the calls received in my queues |
| 20:57.01 | intralanman | anybody in here know ssh really well? |
| 20:57.17 | SomethingISODD | why whats the question |
| 20:57.17 | intralanman | i asked a question in #openssh, but that channel sucks |
| 20:57.38 | hoobastooba | intralanman: |
| 20:57.40 | hoobastooba | what do you want to know |
| 20:57.52 | intralanman | any reason why ssh wouldn't work over a vpn connection only when a dsa key is offered? |
| 20:58.00 | intralanman | vpn is pptp |
| 20:58.04 | intralanman | ms chap |
| 20:58.17 | intralanman | actually mschapv2 i think |
| 20:58.29 | SomethingISODD | intralanman sorry i am no help I don`t even know vps |
| 20:58.31 | SomethingISODD | vpn |
| 20:59.03 | intralanman | eh, no biggie.... i got an rsa key generated now... and it works just fine |
| 20:59.20 | intralanman | it's weird though.... only doesn't work over the VPN |
| 21:00.01 | *** join/#asterisk bkw__ (n=brian@ppp-70-128-126-144.dsl.tulsok.swbell.net) |
| 21:01.19 | dioedu | blitzrage: can you help me ? |
| 21:03.29 | dioedu | if i receive the BUSY state in this call, i don't have lose this one. I want to put this call in the queue again... |
| 21:05.19 | dioedu | but i don't have any variable that tell me what is the 'dst' (or queue) of this call |
| 21:14.28 | ghenry | would you recommend a single or dual cpu server for around 50 users all SIP based? |
| 21:15.00 | ghenry | and aorund 4 fxo and 4fxs (tdm2400p) |
| 21:15.15 | ghenry | 2gb RAM? |
| 21:16.17 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 21:16.32 | shellshark | asterisk can't utilize SMP systems, so dual procs / dual core / hyper threading / etc dont benefit asterisk at all |
| 21:17.23 | ghenry | cool |
| 21:17.33 | ghenry | what's a good calc for working out ram? |
| 21:17.46 | ghenry | so, amd64 or xeon64. Doesn't matter |
| 21:18.46 | h3x0r4t0r | asterisk is threaded |
| 21:18.57 | h3x0r4t0r | so thats not true |
| 21:19.18 | justinu|laptop | shellshark: h3x0r4t0r is right, what's your basis for saying asterisk can't utilize SMP? |
| 21:19.26 | ghenry | ah, ok . That's what I thought |
| 21:19.26 | h3x0r4t0r | asterisk has always been threaded |
| 21:19.42 | shellshark | asterisk's SIP support is not threaded |
| 21:19.44 | ghenry | 2gb ram enough for 50 users? |
| 21:19.45 | h3x0r4t0r | amd64 is definately better for I/O so if you are using zaptel interfaces |
| 21:19.55 | ghenry | yeah |
| 21:20.03 | h3x0r4t0r | shya like sip uses tons of cpu ... the rest of the modules do |
| 21:20.46 | *** join/#asterisk jm|home (n=jamiem@dilbert.jamiem.com) |
| 21:21.47 | shellshark | h3x0r4t0r: SIP can use a lot of CPU, especially if your translating codecs ;) |
| 21:21.56 | shellshark | s/your/you're/ |
| 21:22.37 | h3x0r4t0r | sip dosent translate codecs |
| 21:22.50 | h3x0r4t0r | that is an rtp function |
| 21:23.04 | jm|home | so I want a web based 'directory' |
| 21:23.05 | ManxPower | Codecs are translated by the Codec Faeries. |
| 21:23.08 | jm|home | that will place calls for me |
| 21:23.09 | h3x0r4t0r | yes!!!! |
| 21:23.11 | h3x0r4t0r | like bkw! |
| 21:23.14 | h3x0r4t0r | NEXT! |
| 21:23.41 | *** part/#asterisk RogueCmdr (n=RogueCmd@ip72-201-30-198.ph.ph.cox.net) |
| 21:23.45 | ManxPower | jm|home: Let us know when youve written one. |
| 21:23.50 | jm|home | :/ |
| 21:23.55 | jm|home | is there call for this?! |
| 21:23.58 | h3x0r4t0r | jm|home: to call customers or what |
| 21:24.03 | ghenry | so rule of thumb for RAM? |
| 21:24.09 | jm|home | h3x0r4t0r: at this point, I'm not sure |
| 21:24.13 | jm|home | I just wanted to play with the API |
| 21:24.23 | ManxPower | Um, there is no API. |
| 21:24.26 | jm|home | although - for this instance it looks like I can just drop .call files into /outgoing/ |
| 21:24.26 | h3x0r4t0r | well check out snapanumber.com |
| 21:24.29 | carrar | 2gb is fine for 50 users |
| 21:24.38 | h3x0r4t0r | you can place calls from web pages, outlook, etc anywhere theres a phone number |
| 21:24.43 | jm|home | ooh |
| 21:24.52 | jm|home | doesn't help me learn: but I'll look :) |
| 21:25.04 | h3x0r4t0r | there is also astapi or something which does tapi to asterisk manager |
| 21:25.13 | h3x0r4t0r | and some straight out outlook apps |
| 21:25.25 | h3x0r4t0r | also the vTiger and SugarCRM's have integrations with asterisk as 3rd party modules |
| 21:25.31 | h3x0r4t0r | to speed dial and do screen pops from their CRM databases |
| 21:25.32 | jm|home | ok ok |
| 21:25.39 | jm|home | {overload} |
| 21:25.45 | h3x0r4t0r | well you didnt specify! |
| 21:25.50 | jm|home | SugarCRM is open source |
| 21:25.52 | jm|home | I could steal! |
| 21:25.55 | h3x0r4t0r | vTiger is better |
| 21:25.57 | h3x0r4t0r | its based on sugar |
| 21:26.01 | ghenry | cheers carrar |
| 21:26.30 | h3x0r4t0r | the gpl vtiger has just a few less features than the commercial version of sugar |
| 21:26.41 | jm|home | fewer features, eh ... |
| 21:26.41 | h3x0r4t0r | the oss sugar sucks |
| 21:26.50 | h3x0r4t0r | open source sugar dosent do crap |
| 21:27.10 | h3x0r4t0r | coz they are trying to sell it |
| 21:27.14 | shellshark | works fine for me ;) |
| 21:27.37 | h3x0r4t0r | you can migrate to vtiger easily |
| 21:28.34 | h3x0r4t0r | en.vtiger.com is the demo of the new version 5 |
| 21:28.46 | h3x0r4t0r | yay it has draggable panels now |
| 21:29.52 | shellshark | does vtiger support sugarcrm addons? |
| 21:30.12 | shellshark | i use the asterisk addon for outbound dialing |
| 21:32.39 | mercestes | woah, vtiger CRM has an asterisk addon for dialing?? |
| 21:32.43 | *** join/#asterisk Maxxed (i=foobar@65.59.245.122) |
| 21:32.52 | Maxxed | oi :) |
| 21:32.54 | mercestes | er..Sugarcrm I mean. |
| 21:33.08 | Maxxed | im trying to define a global variable, how is it done again? |
| 21:33.15 | mercestes | [global] |
| 21:33.24 | Maxxed | right |
| 21:33.25 | Maxxed | AnalystCallFileName=/var/spool/asterisk/monitor/Analyst-${TIMESTAMP}-${CALLERID(num)} |
| 21:33.36 | Maxxed | should i stick some commas in here or somthing? |
| 21:33.43 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 21:33.46 | mercestes | how is that a global variable? |
| 21:33.50 | Maxxed | asterisk dies when i reload the config with that wackyness |
| 21:34.12 | mercestes | that would be a contextual variable that woudl be iterated when you access it from a dialplan. |
| 21:34.20 | Maxxed | ah |
| 21:34.30 | Maxxed | so now way to like have that as an alias or what evrs |
| 21:34.36 | mercestes | unless....your pattern match happens to be under [global] but I am not even sure you can do that. |
| 21:34.37 | Maxxed | sry, im way rusty with the good ol asterisk |
| 21:34.48 | Maxxed | ah |
| 21:35.08 | mercestes | Why does it need to be global? |
| 21:35.24 | Maxxed | s,1,SetVar(AnalystCallFileName=/var/spool/asterisk/monitor/Analyst-${TIMESTAMP}-${CALLERID(num)}) i guess? |
| 21:35.25 | mercestes | you can assign the file name directly before monitor or before calling hte macro, or even in the macro |
| 21:35.34 | Maxxed | well i would have to put it in the config like 99 times |
| 21:35.35 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 21:35.41 | mercestes | or.....write a macro. |
| 21:35.51 | hoobastooba | i am trying to install zaptel on my computer and I am getting this output: http://pastebin.ca/243776 |
| 21:35.55 | hoobastooba | can anyone please help? |
| 21:35.57 | Maxxed | um, iv never jacked with the macros but that sounds like what i can do |
| 21:36.08 | Maxxed | thx for the pointer :) |
| 21:36.18 | mercestes | [macro-recordstuff] s,1,setVar(blah) s,2,Monitor(blah) |
| 21:36.19 | hoobastooba | i have the files i think it is telling me i dont have and i have tried a new dowload to make sure that it is not corrupted. |
| 21:36.29 | mercestes | and then you just have 99 liens of Macro(recordstuff) |
| 21:36.37 | Maxxed | yeah, that looks like the trick :) |
| 21:36.44 | mercestes | :) Your welcome. |
| 21:37.17 | X-Rob | ~centosbug |
| 21:37.23 | jbot | centosbug is probably a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h" |
| 21:41.52 | aydiosmio | spinlock eh? |
| 21:41.59 | aydiosmio | that doesn't sound nefarious. |
| 21:43.52 | *** join/#asterisk mivck (i=1000@ip-70-228.telesat.com.co) |
| 21:44.00 | *** join/#asterisk linlin (n=will@c-71-194-70-13.hsd1.il.comcast.net) |
| 21:48.04 | _alex_mx_ | hoobastooba, fc6 does not use config.h anymore...you can try changing the include in the makefile to autoconf.h. this has worked for me with other pacakges but not sure it will with zaptel |
| 21:48.25 | _alex_mx_ | *packages |
| 21:48.48 | hoobastooba | _alex_mx_: thanks dude, will try |
| 21:49.05 | tehdely | howdy kiddos |
| 21:49.12 | tehdely | quick extensions.conf question regarding match order |
| 21:49.22 | tehdely | let's say i want to send all calls that start with 1 out to my primary outbound provider |
| 21:49.26 | tehdely | EXCEPT for toll free calls, which i'd like to divert |
| 21:49.33 | tehdely | how do i set things up so that it tries to match against _1866. first |
| 21:49.37 | tehdely | before matching against _1X. |
| 21:50.16 | *** join/#asterisk RoyK (n=roy@ti211310a080-3120.bb.online.no) |
| 21:51.27 | *** join/#asterisk Lord_Pall (i=Flerb@cpe-66-68-44-56.austin.res.rr.com) |
| 21:52.57 | Maxxed | what the duce! |
| 21:52.57 | Maxxed | Nov 8 15:52:37 WARNING[11217]: app_macro.c:151 macro_exec: Context 'macro-AnalystMonitoring' for macro 'AnalystMonitoring' lacks 's' extension, priority 1 |
| 21:53.08 | Maxxed | but it is there!? |
| 21:53.22 | Maxxed | [macro-AnalystMonitoring] |
| 21:53.23 | Maxxed | s,1,SetVar(AnalystCallFileName=/var/spool/asterisk/monitor/Analyst-${TIMESTAMP}-${CALLERID(num)}) |
| 21:53.23 | Maxxed | s,2,Monitor(wav,${AnalystCallFileName},m) |
| 21:53.26 | Maxxed | bug? |
| 21:53.39 | *** join/#asterisk anthonyl (i=anthonyl@nat/digium/x-05c412886e7b4460) |
| 21:53.39 | Maxxed | or should i like s,1,noop ? |
| 21:54.31 | Maxxed | oh! wait a sec |
| 21:54.40 | Maxxed | its prob how im calling the macro? |
| 21:54.53 | Maxxed | exten => 2604,3,Macro(AnalystMonitoring) |
| 21:54.54 | *** join/#asterisk scurb (n=scurb@c-b7ade355.14-16-64736c13.cust.bredbandsbolaget.se) |
| 21:55.07 | Maxxed | should i like exten => 2604,3,Macro(AnalystMonitoring,2604) ? |
| 21:55.34 | mercestes | Maxxed: shouldn't be. |
| 21:55.48 | Maxxed | not sure what im missing :/ |
| 21:55.52 | mercestes | Maxxed: Are they lowercase S's? Is that a copy paste or a retype?? |
| 21:56.05 | Maxxed | lowercase yes |
| 21:56.10 | Maxxed | and i pasted in vi |
| 21:56.12 | Maxxed | didtn work |
| 21:56.16 | Maxxed | so i typed it out |
| 21:56.26 | Maxxed | thinking it was invisable cars or whatver |
| 21:56.37 | Maxxed | oh!! |
| 21:56.41 | Maxxed | i need exten => |
| 21:56.44 | Maxxed | gdamnit :p |
| 21:56.52 | mercestes | lol |
| 21:56.56 | mercestes | nice. |
| 21:57.13 | Maxxed | pssh.. |
| 21:57.17 | Maxxed | i worry about my self somtimes :p |
| 21:57.34 | mercestes | Hey, I'm the one helping you and I didn't even notice it. |
| 21:57.41 | mercestes | and I'm in the process of programing extensions.conf now too. |
| 21:57.44 | reza_ | are there any services where i can just get a did to forward to a pots? |
| 21:57.52 | Maxxed | hah :) |
| 21:57.58 | Maxxed | well thanks again, the macro did the trick! |
| 21:58.14 | mercestes | :) welcome. |
| 22:02.02 | *** join/#asterisk supjigatr (n=syslod@152.53.17.26) |
| 22:03.06 | supjigatr | Anyone know how to debug echo with a working a104d with HWEC enabled? |
| 22:04.14 | devolish | is it right that asterisk should return something other than 5060 in the contact header field on a 200 when asterisk answers a call? |
| 22:04.30 | devolish | port 5060, of course |
| 22:05.47 | *** join/#asterisk CunningPike_ (n=CunningP@204.239.12.189) |
| 22:08.50 | dattolie | hi all... I've a big problem.. I've * 1.2.13, Linksys SPA-922.942, and a Digium TDM2400 x 12FXO. If I make a call from de hardphone to the PSTN, I listen noice in the hardphne but the person at the PSTN listen clear sound |
| 22:10.25 | *** join/#asterisk kupesoft (n=kupesoft@CPE004010100002-CM0013718cb08a.cpe.net.cable.rogers.com) |
| 22:10.35 | kupesoft | If I set up an asterisk server, can I make free long distance calls? |
| 22:11.17 | iCEBrkr | kupesoft: Nothing in life is free. |
| 22:11.30 | iCEBrkr | kupesoft: But you can make cheaper long distance. |
| 22:11.37 | kupesoft | What service do I have to buy? |
| 22:12.22 | iCEBrkr | kupesoft: There's a few out there.. nufone, voicepluse, asterlink. |
| 22:14.24 | aydiosmio | packet8 vonage broadboice |
| 22:14.37 | aydiosmio | broadvoice* |
| 22:14.46 | justinu|laptop | nufone? lol |
| 22:15.16 | iCEBrkr | aydiosmio: none that work with Asterisk |
| 22:15.50 | aydiosmio | vonage softphone accounts work fine with asterisk |
| 22:15.58 | justinu|laptop | broadvoice BYOD works fine too |
| 22:16.09 | iCEBrkr | I had problems with Broadvoice |
| 22:16.09 | *** join/#asterisk aao_pwner (n=_s@c-24-21-91-140.hsd1.mn.comcast.net) |
| 22:16.19 | aydiosmio | boradvoice does indeed work with * |
| 22:16.21 | iCEBrkr | Then again, this was like 3yrs ago |
| 22:16.30 | aydiosmio | they enven have a support page for it |
| 22:16.33 | aydiosmio | http://www.broadvoice.com/support_install_asterisk.html |
| 22:16.46 | justinu|laptop | we use it all the time... users are happy with the service |
| 22:16.52 | iCEBrkr | aydiosmio: You their spokes person or something :D |
| 22:16.56 | justinu|laptop | they love the "unlimited" international |
| 22:17.13 | iCEBrkr | justinu|laptop: So Broadvoice seems stable these days? |
| 22:17.14 | aydiosmio | and I've heard Packet 8 works with * but haven't verified |
| 22:17.39 | aydiosmio | iCEBrkr: no, just makign up for your lack of knowledge in the subject |
| 22:17.44 | justinu|laptop | iCEBrkr: in my experience yeah, it's ok... however, bv has a miami pop, and this particular server i'm talking about is in miami |
| 22:17.48 | justinu|laptop | 5ms latency to their pop |
| 22:18.06 | iCEBrkr | justinu|laptop: Cool. I'm in Boca. |
| 22:18.14 | justinu|laptop | should be pretty good for you too |
| 22:18.32 | iCEBrkr | aydiosmio: Broadvoice was JUNK awhile ago.. so I wrote them off.. |
| 22:19.07 | aydiosmio | oh cool |
| 22:19.17 | iCEBrkr | I may have to check them out again tho. |
| 22:19.20 | aydiosmio | packet 8 has a free plan where you can call other packet8 users |
| 22:19.38 | justinu|laptop | packet8 allows byod now? |
| 22:20.35 | iCEBrkr | Did the FWD<=>Vonage interchange ever get built? |
| 22:20.57 | justinu|laptop | i think it got built, then unbuilt |
| 22:21.04 | iCEBrkr | haha |
| 22:22.00 | *** join/#asterisk oej (n=oej@p54BEB408.dip0.t-ipconnect.de) |
| 22:22.49 | aydiosmio | I'm not seeinf BYOD for Packet8 |
| 22:23.01 | aydiosmio | SIPGate though |
| 22:23.03 | aydiosmio | they do BYOD |
| 22:23.05 | *** part/#asterisk tehdely (i=tehdely@cement.frap.net) |
| 22:23.41 | aydiosmio | (UK) |
| 22:26.33 | *** join/#asterisk luke-jr_work (n=luke-jr@adsl-70-128-250-253.dsl.ksc2mo.swbell.net) |
| 22:30.40 | Givur | Ist there a List avaible for the OID used in res_snmp? |
| 22:30.59 | *** join/#asterisk rev365 (n=log@70.88.158.113) |
| 22:32.15 | rev365 | channel.c: Didn't get a frame from channel: SIP/203-0986b638 - Does that mean anything special or is it just some gerneic message, it seems to pop up everytime someone gets a dropped call |
| 22:32.25 | rev365 | generic* |
| 22:35.03 | *** join/#asterisk dsfr (i=dsfr@pdpc/sponsor/digium/dsfr) |
| 22:36.46 | rev365 | hello? |
| 22:38.25 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-82-81-190-140.red.bezeqint.net) |
| 22:47.15 | Givur | Ahh, I have found the OIDs :) |
| 22:58.11 | *** join/#asterisk CyberKnet (n=CyberKne@ip68-13-246-61.ok.ok.cox.net) |
| 23:03.42 | *** join/#asterisk predder (n=predder@203.220.55.70) |
| 23:07.25 | *** join/#asterisk BhaalWK (i=bhaal@freenode/unconfirmed/bhaal) |
| 23:26.50 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 23:28.40 | *** join/#asterisk eduardov (n=tester@adsl-065-007-205-234.sip.mia.bellsouth.net) |
| 23:29.22 | eduardov | how well does voip handle data? we have a DS3 line. |
| 23:30.04 | Qwell[] | voip is over data... |
| 23:30.58 | eduardov | ok let me rephrase, how well does a modem perform over voip? |
| 23:31.08 | bkw__ | um good |
| 23:31.17 | eduardov | so there are no error correction issues? |
| 23:31.39 | JT | bkw__: good?? |
| 23:32.22 | *** join/#asterisk Marshall16 (n=Marshall@d60-65-11-228.col.wideopenwest.com) |
| 23:32.29 | bkw__ | voip over the public internet is iffy anyway |
| 23:32.43 | JT | very |
| 23:33.44 | *** join/#asterisk Star568 (i=Star@69-167-150-120.lmdaca.adelphia.net) |
| 23:33.52 | smark | my understanding is that as long as you a) use u-law codec and b) have 0 packet loss it is fine |
| 23:34.11 | smark | but I may be off base here |
| 23:34.20 | reza_ | smark - and low latency |
| 23:34.35 | JT | or A-law codec |
| 23:35.10 | Nugget | I recently upgraded my cisco 7960s to 8.4 (yes, Qwell[], the SIP image) and I'm having problems where the phones will stop accepting calls. |
| 23:35.12 | *** join/#asterisk sysreq (n=sysreq@142-217-128-85.telebecinternet.net) |
| 23:35.16 | Qwell[] | welcome to SIP |
| 23:35.18 | Nugget | shows as "circuit-busy" in the console |
| 23:35.25 | Star568 | Hi folks, where i can get more info about how to bridge 2 calls? |
| 23:35.31 | Nugget | I just brought one back down to 7.2 to see if it fixes it. |
| 23:35.57 | *** part/#asterisk dasenjo (n=dasenjo@201.228.128.10) |
| 23:36.04 | Nugget | I tried SCCP a few months ago but it caused me problems with NAT and I never got the remote phones really working. |
| 23:36.25 | file | Nugget! |
| 23:36.25 | Qwell[] | chan_skinny should work fine with NAT in 1.4 |
| 23:36.38 | Nugget | cool, I'll reconsider it then when we move to 1.4 |
| 23:38.54 | bkw__ | wideband or death? |
| 23:38.59 | bkw__ | wideband please... |
| 23:40.55 | CyberKnet | evening bkw |
| 23:41.15 | eduardov | so you literally have to have 0 packet loss? |
| 23:41.27 | eduardov | its not un-normal to see small amounts |
| 23:41.35 | *** join/#asterisk CyberKnet2 (n=CyberKne@ip68-13-246-61.ok.ok.cox.net) |
| 23:41.41 | Nugget | clearly you need non-un-normal connectivity. |
| 23:41.43 | CyberKnet2 | ack. |
| 23:42.27 | *** join/#asterisk fr0z3n (n=email@CPE0016b64d8af3-CM000f9fa6b664.cpe.net.cable.rogers.com) |
| 23:42.57 | CyberKnet | Either that... or face douple plus not ungoodness. |
| 23:43.05 | Nugget | heh |
| 23:43.30 | fr0z3n | hello everyone...i am trying run asterisk for the first time (newbie!!) and i am getting the following error msg: chat.us.freenode.net |
| 23:43.32 | fr0z3n | err |
| 23:43.32 | fr0z3n | lol |
| 23:43.44 | Nugget | that's mIRC, not asterisk! ;) |
| 23:43.52 | fr0z3n | haha i know |
| 23:43.54 | fr0z3n | sorry copied it wrong |
| 23:44.19 | fr0z3n | it says make: *** [makeopts] Error 1 |
| 23:44.27 | Nugget | if it's a large paste please use pastebin. |
| 23:44.53 | Nugget | we'll probably need to see more than just that last line to make any sense of the error. |
| 23:45.12 | Nugget | try http://pastebin.ca/ |
| 23:45.25 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 23:45.27 | fr0z3n | i did make install and it is asking me "The configure script must be excecuted before running "make" |
| 23:45.57 | fr0z3n | still want me to do pastebin? i am guessing i need configure script first? how? |
| 23:46.26 | Nugget | you should only have to run "make" and then "make install" as the README describes. |
| 23:46.38 | Nugget | if "make" is failing, yeah, paste the last bits and maybe someone here will spot the problem. |
| 23:47.10 | fr0z3n | make says the same thing as well..... |
| 23:47.18 | fr0z3n | its only 2 lines, there nothign else it tries |
| 23:47.25 | Nugget | that's weird. what operating system? |
| 23:47.43 | intralanman | also, what version of asterisk? |
| 23:47.45 | fr0z3n | redhat |
| 23:47.51 | fr0z3n | its the latest beta |
| 23:48.00 | fr0z3n | and sorry redhat 9 |
| 23:48.00 | intralanman | try a more stable version |
| 23:48.05 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 23:48.14 | Nugget | if this is your first shot at asterisk you'd be a lot better off running the stable 1.2 release. |
| 23:48.18 | intralanman | betas aren't generally good for newbies |
| 23:48.21 | Nugget | it will actually be documented. |
| 23:48.44 | fr0z3n | k i'll try the stable version....by the way, do the zaptel and libpri versions differ for asterisk versions? or are they generic? |
| 23:49.01 | Nugget | do you even need either? |
| 23:49.13 | intralanman | everyone needs zaptel |
| 23:49.18 | Nugget | that's not true. |
| 23:49.21 | intralanman | ok |
| 23:49.33 | intralanman | lol |
| 23:50.37 | intralanman | i guess it depends on your definition of the word "need" |
| 23:52.20 | Nugget | plenty of people run asterisk on operating systems where zaptel isn't even available. :) |
| 23:52.23 | Nivex | zaptel comes in handy for ztdummy, which one needs for meetme |
| 23:52.45 | Supaplex | I'm a ztdummy still. ;) |
| 23:52.50 | fr0z3n | i need it for the sangoma card to work |
| 23:53.03 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 23:53.05 | fr0z3n | or dont i? |
| 23:53.20 | fr0z3n | atleast everything i have read tells me to install zaptel and libpri |
| 23:53.24 | Supaplex | not for the card itself I suspect. don't it have its own driver? |
| 23:53.52 | Nugget | libpri is for the t1/e1 interface cards. |
| 23:54.11 | fr0z3n | ummm i was just trying to follow the instructions from the sangoma website |
| 23:54.29 | JT | well, libpri is for pretty much any digital card, although some don't use it, true |
| 23:54.39 | Nugget | I'm not familiar with sangoma hardware, so if they say you need it you probably do. |
| 23:55.20 | intralanman | doesn't sangoma use wanpipe or something to make the zaptel drivers work with it? |
| 23:55.36 | intralanman | or is that just a vicious rumor i heard? |
| 23:55.37 | fr0z3n | well how about the original questions....any version of zaptel and libpri work with asterisk or what versions are recommended for the latest stable build? |
| 23:55.45 | fr0z3n | yes sangoma does use wanpipe |
| 23:56.57 | fr0z3n | all this is for a school project, i have to asterisk up with the sangoma card....and 2 ata's... |
| 23:57.45 | fr0z3n | so any recommendations on the versions of libpri and zaptel? |
| 23:57.49 | intralanman | from asterisk.org i'd use the latest stable of everything |
| 23:57.59 | fr0z3n | k cool |
| 23:58.19 | fr0z3n | thank u! |
| 23:59.12 | *** part/#asterisk cbm11211 (n=Administ@66.28.182.170) |
