| 00:00.41 | *** join/#asterisk file (~jcolp@mctn1-0841.nb.aliant.net) |
| 00:00.49 | file | HOME! |
| 00:00.58 | `Sauron | congrats file |
| 00:01.04 | file | finally home |
| 00:01.19 | opus___ | he2976999 |
| 00:01.22 | opus___ | hey |
| 00:01.57 | opus___ | topping - did you figure it out? |
| 00:02.33 | opus___ | ~ |
| 00:02.40 | opus___ | ~wrt iana |
| 00:02.49 | opus___ | ~wtf iana |
| 00:03.04 | opus___ | ~wrt imohomo |
| 00:03.26 | opus___ | ~wtf imho homo |
| 00:03.45 | opus___ | insert must be broken on my keyboard |
| 00:09.27 | topping | opus___: not yet, getting there |
| 00:14.33 | *** join/#asterisk IQ (~IQ@70-59-164-164.omah.qwest.net) |
| 00:15.00 | *** join/#asterisk ZX81 (matt@222-153-16-58.jetstream.xtra.co.nz) |
| 00:16.26 | Katty | you've all insaned |
| 00:16.44 | topping | i feel insaned |
| 00:16.54 | IQ | me three |
| 00:17.06 | Moc | Katty, alot of guys are gay in * |
| 00:17.20 | Katty | is there something wrong with being gay, Moc? |
| 00:17.35 | *** join/#asterisk brettnem (~mive29@user-0ccsr2l.cable.mindspring.com) |
| 00:17.37 | *** join/#asterisk imagmo (~imagmo@c-24-20-249-117.client.comcast.net) |
| 00:18.04 | Moc | Katty, hehe no, just that if you thought straight are always fixated on sex, gay are worst hehe |
| 00:18.25 | topping | who was talking about sex? |
| 00:18.33 | Katty | geeks are the perviest of all, i think |
| 00:18.38 | Katty | and being one, i heartily agree |
| 00:18.59 | Moc | Katty, that would explain a few thing ;) |
| 00:19.16 | *** join/#asterisk NewSole (david@i216-58-44-245.avalonworks.net) |
| 00:19.37 | Hmmhesays | lol |
| 00:19.38 | ZX81 | anyone know if Digium's E1 cards have CE Certificate of Conformity |
| 00:19.47 | Hmmhesays | i think geeks are just more "frustrated" |
| 00:19.50 | Katty | Hmmhesays: hi gorgeous |
| 00:19.54 | Moc | ZX81, I guess they are, check with digium |
| 00:19.54 | *** part/#asterisk Smythe (~Smythe@spock.cbcag.edu) |
| 00:19.57 | Moc | maybe in the PDF |
| 00:19.58 | ZX81 | yah |
| 00:20.04 | ZX81 | nah not in there |
| 00:20.05 | Hmmhesays | Hiya |
| 00:20.11 | ZX81 | give em a bell I guess |
| 00:20.19 | ZX81 | just don't want to stop this damn SCP |
| 00:20.27 | ZX81 | and QOS not set up on this net |
| 00:20.32 | ZX81 | :) |
| 00:20.38 | *** join/#asterisk BurnedOutGeek (nobody@res-66-191-172-161.spa.sc.charter.com) |
| 00:20.41 | ZX81 | 14Gb to go |
| 00:20.42 | ZX81 | hehe |
| 00:20.50 | Moc | ish hee |
| 00:21.04 | Katty | gosh, evil eyes? |
| 00:21.18 | Hmmhesays | yeah the big blue ones |
| 00:21.28 | Katty | oh, puppy dog eyes? |
| 00:21.37 | Hmmhesays | kind of |
| 00:21.40 | Katty | i don't understand how you can be evil |
| 00:21.48 | Hmmhesays | how I personally can? |
| 00:21.58 | Katty | that's what i said, bunny bread |
| 00:22.12 | ZX81 | bunny bread? |
| 00:22.22 | ZX81 | what kind of bread does a bunny eat? |
| 00:22.26 | Katty | ZX81: it's a brand. |
| 00:22.31 | ZX81 | oh |
| 00:22.33 | ZX81 | :) |
| 00:22.40 | ZX81 | :) |
| 00:22.45 | Hmmhesays | hmmm how about the casual glance eyes |
| 00:22.59 | Katty | your casual glances are anything but casual |
| 00:24.03 | Katty | and in other news, i'm still not out of bed. heh |
| 00:24.13 | Hmmhesays | lol |
| 00:24.20 | Katty | that's what i get for hiking this morning in cold weather. |
| 00:24.27 | *** join/#asterisk dooder (~nateputna@66.241.89.77) |
| 00:24.32 | Hmmhesays | define "cold" |
| 00:24.53 | ZX81 | noone avail at digium |
| 00:24.55 | ZX81 | heh |
| 00:25.00 | ZX81 | nevermind |
| 00:25.01 | Essobi | BORK BORK BORK |
| 00:25.05 | Katty | as in after 10 minutes your tail is numb. |
| 00:25.18 | Katty | the wind was also blowing, which didn't help matters. |
| 00:25.33 | Hmmhesays | sounds like you were in fargo this morning, lol |
| 00:25.39 | Essobi | mmm |
| 00:25.42 | Essobi | cheeese soup |
| 00:25.50 | Darien | best kind |
| 00:25.51 | Katty | Hmmhesays: heh |
| 00:26.20 | Hmmhesays | soon it will be warm.... maybe |
| 00:27.00 | Katty | soon is all relative |
| 00:27.38 | Hmmhesays | soon defined as within the next month |
| 00:27.59 | *** join/#asterisk NewSole (david@i216-58-44-245.avalonworks.net) |
| 00:29.37 | Darien | Katty->setGone(); |
| 00:29.43 | Hmmhesays | lol |
| 00:30.06 | opus___ | has anyone worked with java's sip communicator |
| 00:30.40 | topping | opus___: do you have a link? |
| 00:31.27 | topping | ah got it |
| 00:31.40 | topping | JAIN looks crazy |
| 00:31.47 | Bacon | opus___: Did you get broavoice working? |
| 00:32.57 | opus___ | no, .. |
| 00:33.06 | hcclnoodles | Anyone know any good Linux IRC channels ?? |
| 00:33.23 | topping | hcclnoodles: #fedora is active |
| 00:33.29 | Hmmhesays | #debian #redhat #fedora |
| 00:34.01 | NewSole | can anyone tell me why I am get this |
| 00:34.03 | hcclnoodles | cheers |
| 00:34.16 | Hmmhesays | #linuxhelp is large |
| 00:34.22 | NewSole | Rejected connect attempt from 216.X.x.X, requested/capability 0x1/0x10f incompatible with our capability 0xf800. |
| 00:34.54 | topping | sounds like an incompatibility |
| 00:35.16 | hcclnoodles | is anybody here using mIRC ?? |
| 00:35.47 | NewSole | I know its incompatibility but why |
| 00:35.55 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 00:36.00 | hcclnoodles | is there anyway of automating the iDENTITY stuff (i hate doing it each time) |
| 00:36.21 | Hmmhesays | umm.. the ident options in preferences? |
| 00:36.24 | ariel_ | NewSole, I think you have the wrong codec |
| 00:36.57 | NewSole | ariel_: only using G729 |
| 00:37.12 | ariel_ | do you have g729 installed on asterisk |
| 00:37.21 | NewSole | ya |
| 00:39.08 | hcclnoodles | Hmmhesays> ok, i cant see a preferences only tools-> options ...and in there i cant see an IDENT section |
| 00:39.09 | *** join/#asterisk JmanA9 (~josh@h97.181.40.69.ip.alltel.net) |
| 00:39.19 | opus___ | <PROTECTED> |
| 00:39.20 | opus___ | <PROTECTED> |
| 00:39.30 | opus___ | .,.. and just hangs when I call out on broadvoice |
| 00:39.48 | *** join/#asterisk DyOS (~me@ip68-2-152-242.ph.ph.cox.net) |
| 00:39.54 | opus___ | exten=_9NXXNXXXXXX, 1, Dial(SIP/1${EXTEN:1}@sip.broadvoice.com) |
| 00:39.54 | opus___ | exten=_9NXXNXXXXXX, 2, congestion() ; No answer, nothing |
| 00:39.54 | opus___ | exten=_9NXXNXXXXXX, 3, busy() ; Busy |
| 00:39.55 | NewSole | hcclnoodles... thats not it |
| 00:39.59 | opus___ | is anything wrong with that? |
| 00:40.31 | opus___ | i'm dialing 1 + local area code + number @ sip.broadvoice.com, and I am currently registered and can take incoming calls.. |
| 00:40.54 | hcclnoodles | can you ellaborate NewSole |
| 00:41.00 | NewSole | hcclnoodles... do an "Alt+R" and when windows pops up add "on 1:CONNECT:/msg nickserv IDENTIFY 101maxi |
| 00:41.18 | NewSole | example |
| 00:41.28 | hcclnoodles | k ill give it a try |
| 00:41.49 | *** join/#asterisk masuda (~masuda@pcp04490438pcs.brmngh01.mi.comcast.net) |
| 00:41.55 | opus___ | any idea? |
| 00:41.57 | *** part/#asterisk hcclnoodles (~hcclnoodl@hcclnoodles.plus.com) |
| 00:41.57 | *** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net) |
| 00:42.06 | NTJOCK | hey gang, what is the separator if I want an extension to ring multiple phones? |
| 00:42.13 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 00:42.21 | NTJOCK | i.e. exten=> 2,2,dial(sip/525,sip/526) |
| 00:42.35 | NewSole | ariel_... you have any idea |
| 00:42.48 | masuda | how does fax with with asterisk? can phones and fax machines call the same Asterisk-hosted number and be given the dial-menu or receive fax based on the calling device? |
| 00:42.57 | *** join/#asterisk hcclnoodles (~hcclnoodl@hcclnoodles.plus.com) |
| 00:43.06 | NTJOCK | masuda: yes, it works, if you can get spandsp running |
| 00:43.08 | NewSole | hcclnoodles... did it work |
| 00:43.21 | masuda | spandsp eh? |
| 00:43.23 | NTJOCK | yea |
| 00:43.25 | hcclnoodles | newsole that works great thankyou, is there a way to automatically join this channel aswell ? |
| 00:43.30 | Darwin35 | NTJOCK no luck yet |
| 00:43.41 | masuda | using asterisk@home, looks like it's fairly easy to activate the fax feature. |
| 00:43.42 | Darwin35 | spandsp is going to be a piece of work |
| 00:43.43 | NTJOCK | Darwin35: no, still won't load the modules |
| 00:43.52 | NTJOCK | :( |
| 00:43.53 | masuda | does asterisk@home make it easier? |
| 00:43.57 | NTJOCK | I'd be happy to regress |
| 00:44.01 | NTJOCK | masuda: I think it does |
| 00:44.02 | Darwin35 | I am going to rest and work on it more tomarroe |
| 00:44.09 | Darwin35 | Kram |
| 00:44.09 | masuda | looks like it's almost just a matter of checking a box and clicking apply |
| 00:44.10 | NTJOCK | cool |
| 00:44.25 | Darwin35 | I need to talk to you about the fbsd g729 codec |
| 00:44.32 | Darwin35 | when you have time |
| 00:44.32 | kram | okay |
| 00:44.37 | kram | i'm on the phone |
| 00:44.37 | *** join/#asterisk brc__ (~brian@brc.base.supporter.pdpc) |
| 00:44.41 | Darwin35 | ok |
| 00:44.54 | NewSole | can anyone tell me why I am get this |
| 00:44.56 | NewSole | Rejected connect attempt from X.X.X.X, requested/capability 0x1/0x10f incompatible with our capability 0xf800. |
| 00:45.11 | *** part/#asterisk hcclnoodles (~hcclnoodl@hcclnoodles.plus.com) |
| 00:45.56 | NTJOCK | off to burn meat on the grill... bbiaf |
| 00:46.09 | Darwin35 | NJ ooooooouch |
| 00:46.18 | Darwin35 | NT sorry |
| 00:47.10 | *** part/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com) |
| 00:47.15 | *** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com) |
| 00:47.17 | r0d3nt|m | opps |
| 00:47.36 | NTJOCK | steak and potatoes... hmmmm |
| 00:49.49 | riksta | sipgate down AGAIN |
| 00:49.51 | riksta | rediculous |
| 00:50.17 | Darien | ridiculous |
| 00:50.30 | riksta | my bad |
| 00:50.41 | riksta | i always get that wrong |
| 00:50.54 | *** join/#asterisk muesli (~muesli@mail.muehlhaeuser.de) |
| 00:51.35 | JmanA9 | if i can make outgoing calls through the console, but not through my softphone, what would that be a misconfiguration of? |
| 00:52.50 | WhiteWlf | your dialplan |
| 00:53.35 | JmanA9 | so i need to change stuff in my extensions.conf file? |
| 00:53.49 | *** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 00:53.49 | *** mode/#asterisk [+o twisted] by ChanServ |
| 00:54.08 | WhiteWlf | mhummm... sounds like your missing an include |
| 00:54.34 | JmanA9 | i think i am, since i can call all local extensions fine, but i just can't make outgoing calls to fwd or sipphone on my softphone |
| 00:54.40 | JmanA9 | they work just fine in the terminal |
| 00:54.57 | shido6 | sip softphone? |
| 00:54.59 | WhiteWlf | then for almost sure you're missing an include in your [local] or [default] contexts |
| 00:55.09 | kram | okay, what's your q? |
| 00:55.15 | kram | darwin35... |
| 00:55.27 | JmanA9 | ok, i'll check that out, thanks :) |
| 00:56.32 | NewSole | kram... got a question about your TDM cards if you got a moment |
| 00:57.05 | kram | newsole: is it a question you think i'm the only person qualified to answer? |
| 00:57.58 | NewSole | I was wondering if there was a setup doc for that Wildcard TE405P |
| 00:58.40 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 00:59.33 | JmanA9 | whitewlf: i got it working, i didn't add any includes in the [default] context :) |
| 01:00.31 | opus___ | jam - how can you make calls from the console |
| 01:03.33 | opus___ | Got SIP response 481 "Call/Transaction Does Not Exist" back from 147.135.0.128 |
| 01:03.40 | opus___ | when calling out on broadvoice |
| 01:04.04 | Inv_arp | context "A" is my incoming IVR "B" is internal sipphones and "C" is outside pstn line ; A includes => B and B includes => C, how can i prevent context "A" from accessing pstn "C"? |
| 01:04.12 | Inv_arp | the problems is during my incoming IVR "A" some1 can dial 91NXXNXXX and call anywhere on pstn "C" |
| 01:05.17 | riksta | can anyone recommend a reliable outgoing pstn service for the UK, sipgate is always going down |
| 01:07.26 | Inv_arp | riksta: i prefer iax providers myself any in the UK? |
| 01:07.57 | Flash_ | may i ask, what is this IAX thing ? |
| 01:08.07 | riksta | Inv_arp: not sure, trying to find out myself :) |
| 01:08.34 | riksta | i just want something for home use, no huge minutes |
| 01:09.57 | Inv_arp | Flash_: protocol establishes voip connections (like SIP)... werks perf behind nat etc... |
| 01:10.25 | riksta | Flash_: more info on the wiki |
| 01:10.36 | riksta | allows for trunking etc |
| 01:12.36 | *** join/#asterisk w0w0 (~w0w0@80-28-166-80.adsl.nuria.telefonica-data.net) |
| 01:12.51 | Inv_arp | heh i always called IAX I-A-X ... the dev conf made me realize its called "eekss" |
| 01:13.16 | NewSole | Question.... those TDM400P its and RJ45 plug... and they are single line cards.... can I plug a RJ11 into it |
| 01:13.23 | riksta | i still say I A X, because the latter sounds GAY |
| 01:13.35 | Inv_arp | riksta: lol |
| 01:13.59 | riksta | its like SQL, "sequel" just sounds gay |
| 01:14.44 | *** join/#asterisk zignig (~simon@203.217.15.10) |
| 01:14.45 | riksta | my sequel..... you aren't spielberg |
| 01:14.46 | riksta | :) |
| 01:16.03 | opus___ | hmmm, when i try a tcpdump when calling out on broadvoice i see the sip "unauthorized" error go by in a packet.. |
| 01:16.35 | *** join/#asterisk Rick_Hunter (~rhunter@bgp959110bgs.derbrh01.mi.comcast.net) |
| 01:16.50 | opus___ | but i can get incoming calls and it shows as registered |
| 01:17.06 | Bacon | opus___: Have you been able to get broadvoice working directly with any softphone? |
| 01:17.40 | opus___ | yes, i can call in and answer the call on multiple sip phones |
| 01:17.56 | opus___ | softphone, hardphone, softphones over the wan, etc |
| 01:21.54 | opus___ | bacon - in your register => broadvoice/yada/yada do you have a extension? The last config line |
| 01:24.34 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 01:25.13 | Inv_arp | opus___: has yer BV ever worked? |
| 01:25.45 | opus___ | yes, incoming works fine |
| 01:25.56 | opus___ | I've never got outoging to work |
| 01:26.20 | opus___ | http://www.mail-archive.com/asterisk-users@lists.digium.com/msg81284.html |
| 01:26.25 | opus___ | seems like this guy has the same problem |
| 01:27.01 | opus___ | he mentions: "I did an |
| 01:27.01 | opus___ | Ethereal trace on the softphone call setup and compared it to a trace of the |
| 01:27.01 | opus___ | call attempt from *. I noticed that the softphone sends an Authorization: line |
| 01:27.03 | opus___ | in the INVITE header, while * does not." |
| 01:27.15 | Inv_arp | opus___: sip show peers is it registered? |
| 01:27.22 | opus___ | inv_arp - yes |
| 01:27.47 | Essobi | stream_selectstream_selectstream_selectstream_selecthh |
| 01:27.47 | Essobi | *** tux_rulez is now known as tux_rulez|eating |
| 01:27.47 | Essobi | <opus___> bacon - in your register => broadvoice/yada/yada do you have a extension? The last config |
| 01:27.48 | Essobi | <PROTECTED> |
| 01:27.48 | Essobi | *** lilo [lilo@levin-pdpc.staff.freenode] has joined #asterisk |
| 01:27.48 | Essobi | <Inv_arp> opus___: has yer BV ever worked? |
| 01:28.22 | opus___ | hmm |
| 01:28.23 | NewSole | anyone got a sec to answer a question about the TDM400P cards |
| 01:28.26 | Inv_arp | opus___: what version of * u run? |
| 01:29.09 | tux_rulez|eating | Essobi: is that a fax pax here I got use to being on a smaller IRC net |
| 01:29.47 | Inv_arp | i havent chged * since 1.0.5 i hear to many issues with newer versions |
| 01:30.36 | tux_rulez|eating | I sould have signed of of the rooms I was not active it. There was a guy installing a kew kernel he had to reboot And I wanted him to know why I was not answering |
| 01:31.31 | opus___ | inv -- i just rm -rf /usr/src/asterisk and /var/lib/asterisk/ and downloaded the latest from cvs |
| 01:32.25 | opus___ | inv_arp, where can I get 1.0.5 ? :) Seems to be gone now |
| 01:33.09 | opus___ | oh . there are archive |
| 01:33.44 | Inv_arp | nope |
| 01:33.53 | opus___ | awesome |
| 01:33.57 | Inv_arp | 1st to come with patch |
| 01:34.17 | Inv_arp | opus___: give us an update if it fixes issue |
| 01:34.24 | opus___ | will do |
| 01:34.52 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 01:35.10 | *** join/#asterisk TheEmperor (~mattn@202.187.40.2) |
| 01:36.33 | *** join/#asterisk Juxt (~Juxt@sfl-dsl-64-135-113-4-cust.host.net) |
| 01:36.43 | Juxt | good afternoon |
| 01:37.00 | Juxt | i am trying to install asterisk in a path that is different from the deault linux path |
| 01:37.07 | Juxt | but there's no configure... so how do i do it? |
| 01:38.12 | Darwin35 | just look at the Makefile |
| 01:38.23 | Darwin35 | you will figure it out |
| 01:38.42 | *** join/#asterisk ocnarfid (~ocnarfid@24-193-58-88.nyc.rr.com) |
| 01:38.47 | Darwin35 | where do you want to install it |
| 01:38.53 | Darwin35 | what dir |
| 01:39.39 | Juxt | looks like i found the INSTALL_PREFIX |
| 01:39.51 | Juxt | i want to install it into /usr/local/asterisk |
| 01:43.45 | Darwin35 | are you on linux or fbsd ? |
| 01:44.22 | *** join/#asterisk Frantic (~ab@24-193-46-85.nyc.rr.com) |
| 01:44.26 | *** join/#asterisk kks (~kks@203.115.210.253) |
| 01:46.06 | *** join/#asterisk jeffik (~jeffik@maf7636d0.tmodns.net) |
| 01:47.07 | *** join/#asterisk caesar2 (caesar@pD9E4E731.dip.t-dialin.net) |
| 01:47.47 | opus___ | no luck with 1.0.5, same issue |
| 01:47.58 | Inv_arp | bah |
| 01:48.21 | Inv_arp | i know BV chge something in their servers thats been having * users go crazy |
| 01:48.43 | opus___ | proxy.lax.broadvoice.com is what I use now, is that a bad one? |
| 01:49.02 | *** join/#asterisk captmeridian (~mwood@bdsl.66.12.196.104.gte.net) |
| 01:49.05 | opus___ | i mean sip.lax.broadvoce.com |
| 01:49.55 | opus___ | i've tried other servers.. |
| 01:50.07 | Inv_arp | Juxt: just install normally, than mv to new location and have asterisk.conf point to new locations |
| 01:50.55 | Inv_arp | i use proxy.mia.broadvoice.com but just incoming |
| 01:51.29 | Inv_arp | opus___: under sip show registry what your refresh time? |
| 01:52.16 | riksta | can someone call me over IAX to test something ? |
| 01:52.25 | riksta | am getting echo speaking to someone, wanna make sure it's not this end |
| 01:53.00 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 01:55.39 | opus___ | inv_arp -- 3186 |
| 01:55.50 | opus___ | Host Username Refresh State |
| 01:55.51 | opus___ | sip.broadvoice.com:5060 5037150059@s 3186 Registered |
| 01:56.05 | Inv_arp | thats pretty high ... mine is 105 |
| 01:56.13 | opus___ | what does 5037150059@s mean? the @s part is curious |
| 01:56.25 | opus___ | What is the refresh variable for? |
| 01:57.03 | Inv_arp | no idea but i noticed everyon that has had BV probs has had high refresh rates |
| 01:57.30 | opus___ | strange |
| 01:57.44 | opus___ | I'm not running the time clock module, perhaps thats my problem |
| 01:58.08 | opus___ | what is it called again? |
| 01:58.29 | Inv_arp | rtc? |
| 01:58.40 | opus___ | yes, but the zap one? |
| 01:59.00 | captmeridian | can someone give me some quick help with *@home and CIsco sets? |
| 01:59.02 | *** join/#asterisk lately (~dougb@chi.econ.usyd.edu.au) |
| 01:59.56 | riksta | can someone call IAX2/guest@pbx.packetoverdose.com please |
| 02:00.00 | shido6 | @s |
| 02:00.01 | riksta | for a very brief test |
| 02:00.06 | shido6 | u have a s, somewhere |
| 02:00.12 | shido6 | "starting" a context |
| 02:00.35 | shido6 | i smell a "friend" in your iax.conf tho ... |
| 02:00.43 | shido6 | err |
| 02:00.47 | riksta | anyone? |
| 02:00.54 | shido6 | riksta, stdby |
| 02:00.57 | riksta | ta |
| 02:01.02 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 02:01.40 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 02:02.26 | riksta | thanks for that |
| 02:02.26 | shido6 | click |
| 02:02.31 | shido6 | no problem |
| 02:05.40 | *** join/#asterisk CoaxD (coax@shell1.cornernet.com) |
| 02:06.46 | lately | Newbie question: If my phone supports protocol X and my provider only supports protocol Y, can asterisk on my gateway convert between protocols so that I can communicate with my provider? (assuming asterisk supports both X and Y) |
| 02:07.34 | shido6 | yeshhhh |
| 02:07.43 | shido6 | sip, mgcp, iax, h323 |
| 02:07.48 | shido6 | sccp |
| 02:07.48 | Juxt | look into transcoding |
| 02:07.55 | Qwell | Do you mean protocol, or codec? |
| 02:08.01 | Qwell | The answer to both is yes, but... |
| 02:08.03 | lately | Qwell: ah. codec |
| 02:08.06 | shido6 | u can do both |
| 02:08.09 | Qwell | much different question |
| 02:09.23 | *** join/#asterisk trig_hm (~jb@home.monkeypr0n.org) |
| 02:11.31 | lately | Another newbie question: Are there any encryption protocols/tools for VoIP? eg TLS in SMTP. |
| 02:12.02 | Juxt | probably not, encryping a real time protocol is very cpu intensive |
| 02:12.16 | Qwell | Tunnel IAX over ssh. heh |
| 02:12.49 | Sedorox | IAX over PPP over SSH..... yuck |
| 02:13.08 | lately | Hrm. Is this avenue being researched? |
| 02:13.17 | *** join/#asterisk Koshatul (~evangelio@CPE-138-217-191-19.qld.bigpond.net.au) |
| 02:13.32 | *** part/#asterisk zignig (~simon@203.217.15.10) |
| 02:14.01 | *** part/#asterisk tux_rulez|eating (~sfeil@m415e36d0.tmodns.net) |
| 02:14.02 | NTJOCK | hey gang, anyone here written nice, complicated IVR scripts that have multiple levels in your tree? |
| 02:14.20 | NTJOCK | I have a progression question that I haven't found an answer to in the wiki |
| 02:15.48 | NTJOCK | lately, why not just run it via VPN |
| 02:15.51 | NTJOCK | that offers encryption |
| 02:15.56 | mog_home | wow fun i the sun |
| 02:15.57 | mog_home | but not |
| 02:16.10 | mog_home | ask away |
| 02:16.59 | NTJOCK | if you have your normal "hello thank you for calling" menu under s,etc... and option 1 is for blah and option 2 is for yack..... |
| 02:17.10 | NTJOCK | is it best to send 1 to it's own context and 2 to it's own context? |
| 02:17.16 | NTJOCK | i.e. 1 for customers, 2 for vendors? |
| 02:17.23 | NTJOCK | customers get alot more routing |
| 02:17.28 | NTJOCK | vendors have a very short tree |
| 02:17.36 | lately | NTJOCK: Cannot. I'm planning on getting an 802.11e (wireless) connection to the net |
| 02:17.44 | opus___ | ntjock -- goto ? |
| 02:17.48 | bjohnson | put each submenu in it's own context |
| 02:17.53 | NTJOCK | lately: why not? it's just a connection |
| 02:17.54 | lately | NTJOCK: So my VPN would have to extend to the SIP provider |
| 02:18.02 | bjohnson | brief example in the sample confs |
| 02:18.03 | NTJOCK | well, if your not doing an in house system, yea |
| 02:18.14 | NTJOCK | I don't honestly see the issue with ti... but some folks are paranoid |
| 02:18.15 | bjohnson | you will need to make custom voice prompts |
| 02:18.26 | NTJOCK | bjohnson ok... i'm on the right track then. |
| 02:18.36 | lately | NTJOCK: Hrmm. Maybe I'm missing something. Can IAX to encryption? |
| 02:18.36 | NTJOCK | bjohnson I already set up an extension to record the stupid things |
| 02:18.40 | NTJOCK | and then I rname them. |
| 02:18.46 | bjohnson | yep |
| 02:18.48 | lately | s/to/do/ |
| 02:19.02 | bjohnson | lately: no |
| 02:19.04 | NTJOCK | lately I don't know much with encryption... I'm replacing a KSU system at my small business... so everything is inside the network |
| 02:19.20 | bjohnson | it would need to be inside sonething else that provides encryption |
| 02:19.20 | NTJOCK | and our remote worker is connected via VPN to ensure connectivity and security |
| 02:19.50 | NTJOCK | I think if you want to do encryption you have to do it at the handset level from handset to handset |
| 02:19.52 | lately | NTJOCK: I'm talking about wireless internet connection. Not wireless on my LAN (even though that will be present too) |
| 02:20.03 | NTJOCK | yes... and it's a point to point connection |
| 02:20.08 | NTJOCK | NTJock = KD5FIU |
| 02:20.26 | NTJOCK | the beam is pretty narrow and you're really unlikely to have issues with packet snooping. |
| 02:20.33 | NTJOCK | unless of course your the FDIC or something |
| 02:20.47 | NTJOCK | I don't want to seem rude, but nobody probably cares about your phone calls..... |
| 02:20.57 | NTJOCK | they'd be more likely to sniff your credit card # and even that is a long shot. |
| 02:21.09 | lately | NTJOCK: totally agree but I would still like to know. |
| 02:21.25 | NTJOCK | if someone wanted to listen in they'd be better off to spend $50 at rat-shack for a telescopic microphone. |
| 02:21.26 | mikegrb | mikegrb = kd5pvw |
| 02:21.31 | NTJOCK | and point it at your window. |
| 02:21.38 | NTJOCK | mikegrb cool! |
| 02:21.38 | bjohnson | native encryption is currently not possible in voip |
| 02:21.51 | shido6 | um |
| 02:22.00 | shido6 | not publicly |
| 02:22.03 | NTJOCK | lately most folks are worried about just making VOIP work... let alone encrypting it. |
| 02:22.05 | lately | NTJOCK: 802.11e is city wide wireless |
| 02:22.21 | NTJOCK | but it's still point to multi-point at the worst |
| 02:22.30 | NTJOCK | i.e. a radial antenna somewhere high and a bunch of directionals pointed at it |
| 02:22.41 | NTJOCK | and if it's like the rest of the 802.11 racket it's a shared bandwidth |
| 02:22.50 | NTJOCK | I think you will have a very nasty VOIP experience |
| 02:23.02 | NTJOCK | unless you spring for the .729 codec to get your bandwidth way way down. |
| 02:23.30 | shido6 | office space |
| 02:23.30 | lately | The link is mean to be 256/64 down/up |
| 02:23.33 | shido6 | waay... waayy down |
| 02:23.40 | NTJOCK | VOIP one call can use 64K easily |
| 02:23.49 | NTJOCK | and it doesn't tolerate variable bandwidth |
| 02:23.51 | n3tar | 7 |
| 02:23.53 | lately | Yeah. I'm told that's the minimum |
| 02:24.03 | NTJOCK | we discovered that we had a congestion issue by trying out vonage. |
| 02:24.04 | NTJOCK | hehe |
| 02:24.07 | NTJOCK | 2 years ago |
| 02:24.19 | NTJOCK | I have a Gigabit switched network with like 15 computers |
| 02:24.31 | NTJOCK | slowest speed on the network is 100mb switched to the slow machines |
| 02:24.42 | NTJOCK | one cable segment is cat 1 wire because it's old. |
| 02:24.49 | NTJOCK | we figure that connectivity there is 10mb or less |
| 02:24.57 | NTJOCK | the phones on that segment sound like sh*t |
| 02:25.06 | NTJOCK | everything else is pretty well behaved |
| 02:25.15 | NTJOCK | and yes, we run a damn fast network.... :) |
| 02:26.08 | NTJOCK | one of my buddies sitting here is running 1 extension across a 54 mb point to point wireless connection and he's having issues |
| 02:26.15 | *** join/#asterisk Mike (~mike@201.135.48.217) |
| 02:26.20 | NTJOCK | he's only jumping about 100 feet and using parabolic antennae |
| 02:26.29 | *** join/#asterisk fmenard (fmenard@109-75.tr.cgocable.ca) |
| 02:26.40 | NTJOCK | so he's switching to a G.729 codec to get the bandwidth down so when trucks pull through it won't drop calls |
| 02:26.42 | bjohnson | on the other hand .. people use voip over staellite |
| 02:26.50 | NTJOCK | true |
| 02:26.54 | fmenard | any idea why latest cvs crashes on start |
| 02:26.55 | bjohnson | depends what you're willing to settle for |
| 02:27.00 | fmenard | is it stable? |
| 02:27.16 | NTJOCK | I'm not trying to scare ya off... I just want you to have a reasonable expectation of what you'll get and to not be mad at VOIP when you try to make it work across almost-free internet |
| 02:27.24 | fmenard | if I do a make clean, does it really clean all binaries compiled? |
| 02:27.44 | NTJOCK | to make a packet network perform like a switched network requires alot of bandwidth..... |
| 02:27.47 | *** join/#asterisk fqwer333 (ben@85.48.15.128) |
| 02:27.51 | Inv_arp | fmenard: rm -rf really cleans :) |
| 02:28.06 | NTJOCK | so if you have bandwidth to chew.... it's a good idea. |
| 02:28.27 | lately | NTJOCK: I'm in the market for a phone now. I have ADSL but will be moving shortly. I hardly ever use the landline so I thought getting unwired.com.au product and going with VoIP |
| 02:28.35 | NTJOCK | or if you are like us, you need features and would rather code them |
| 02:28.39 | NTJOCK | lately It might work then |
| 02:28.44 | Juxt | anyone have experience running asterisk realtime with postgres? |
| 02:28.47 | NTJOCK | lately: I'd keep my wireless phone handy |
| 02:29.04 | NTJOCK | at any rate... * is a wickedly cool toolkit |
| 02:29.06 | lately | NTJOCK: You mean wired phone? |
| 02:29.09 | Inv_arp | Juxt: nope but heard cdr_pqsql works |
| 02:29.12 | NTJOCK | no I meant your cell phone |
| 02:29.17 | lately | ah |
| 02:29.17 | NTJOCK | as a backup |
| 02:29.25 | NTJOCK | most folks down-under have cell phones |
| 02:29.35 | fqwer333 | Hey Guys, |
| 02:29.35 | fqwer333 | I've noticed something interesting, I hear an echo on incoming calls, but when I'm the one calling I don't hear an echo at all. |
| 02:29.41 | lately | yeah... hehe. My wife is pregnant! don't want my project failing at the crucial time! |
| 02:29.45 | NTJOCK | right |
| 02:29.50 | NTJOCK | she might not forgive you easily |
| 02:29.51 | NTJOCK | :) |
| 02:30.05 | NTJOCK | but as a chat-line or a toy to play with it's great. |
| 02:30.11 | NTJOCK | you could probably order pizza across it |
| 02:30.16 | NTJOCK | I just wouldn't do business over it all day |
| 02:30.34 | NTJOCK | one thing VOIP is very good for is avoiding international toll rates |
| 02:30.35 | NTJOCK | ;) |
| 02:31.19 | NTJOCK | I just like to toss a dose of reality in when I see folks are new to this..... alot of people think they can get * installed, cut their phone bill and avoid the local telco and have better service quality. |
| 02:31.34 | NTJOCK | it aint quite there yet |
| 02:32.26 | lately | we pretty much have only one telco here. It ownz all the friggen cables so you must go through them. I'd love to be rid of them |
| 02:32.27 | Damin | NTJOCK: That is interesting, considering I do like 750,000 minutes a day of Local and Long Distance service through Asterisk. |
| 02:32.52 | NTJOCK | Damin: what's your connection? |
| 02:32.59 | *** join/#asterisk Darwin[laptop] (~Darwin@c-24-3-241-22.client.comcast.net) |
| 02:33.09 | Damin | NTJOCK: I don't do it VoIP.. I dump it directly to carriers.. |
| 02:33.14 | NTJOCK | Damin: I'm referring to folks with el-cheapo Joy internet $6.95 sorta DSL |
| 02:33.18 | fqwer333 | hello i am new in asterisk and i am testing it with a PRI card. I have noticed that when i have an incoming calls, the echo canceller does not work |
| 02:33.29 | Damin | NTJOCK: All VoIP is done accross QOS links back to my clients.. Cisco on both ends.. |
| 02:33.47 | NTJOCK | that's not a amateur setup |
| 02:33.48 | NTJOCK | ;) |
| 02:33.54 | Damin | NTJOCK: Well, yeah.. Voice over the Public Internet might work.. but in most cases it is just stupid. |
| 02:34.03 | NTJOCK | exactly |
| 02:34.13 | NTJOCK | or cool while it works |
| 02:34.16 | lately | Does asterisk mark packets correctly with QoS ? |
| 02:34.29 | NTJOCK | I've just seen a handful of people lately looking at * as a free phone system........ |
| 02:34.35 | Damin | lately: It's not Asterisk's job reall, but it can set TOS bits.. |
| 02:34.35 | NTJOCK | anyhow |
| 02:34.59 | NTJOCK | I think * is a huge feature set for any business and it can level the playing field in terms of telephone-presence. |
| 02:35.12 | NTJOCK | <PROTECTED> |
| 02:35.24 | n3tar | bye |
| 02:35.36 | NTJOCK | and because it makes good use of all lines you can get better utilization with fewer "dedicated" lines. |
| 02:35.39 | NTJOCK | anyhow |
| 02:35.41 | NTJOCK | off to code |
| 02:35.42 | NTJOCK | :) |
| 02:35.45 | fmenard | ok, that's really weird ... I rm-f * then rebuild and it works... |
| 02:35.54 | NTJOCK | welcome to linux |
| 02:35.57 | NTJOCK | ;) |
| 02:36.02 | fmenard | with respect to sip ua registration failures... |
| 02:36.25 | fmenard | does asterisk rely on configuration entries outside sip.conf to reject registration requests? |
| 02:36.26 | Inv_arp | fmenard: told ya :) |
| 02:36.27 | NTJOCK | fmenard you probably had a mismatched set of drivers i.e. zaptel and libpri |
| 02:36.52 | lately | Damin: Yeah. Thats all it can do is set the TOS bits. How it is routed is done elsewhere. |
| 02:37.07 | fmenard | I get registration failed messages |
| 02:37.22 | Inv_arp | fmenard: BV outgoing i bet |
| 02:37.59 | fmenard | for intance, if entries in sip.conf with contexts that do not exist in extensions.conf, does it matter? |
| 02:38.03 | fqwer333 | Could anyone help me with echo cancell with incoming calls? |
| 02:38.05 | lately | Well... I'm thinking of buying a Snom 190 IP Desk Phone. Anyone know of anything wrong with this? |
| 02:38.06 | fmenard | BV? |
| 02:38.26 | fmenard | ne1 tried granstream gxp2000? |
| 02:38.27 | Inv_arp | fmenard: nvm |
| 02:38.33 | fmenard | nvm? |
| 02:38.35 | lately | Any gotchas that I want to know when buying a phone? |
| 02:38.52 | Inv_arp | fmenard: Broadvoice/nevermind |
| 02:39.14 | Inv_arp | lately: how much is it? |
| 02:39.19 | fmenard | no no, its sinply a mediatrix 2102 registering into my * box |
| 02:39.36 | Juxt | can asterisk realtime get a dialplan out of a db? |
| 02:39.48 | Inv_arp | fmenard: oh sound like misconfigured sip.conf |
| 02:40.12 | lately | Inv_arp: About $280 in US dollars |
| 02:40.13 | WilliamK | anyone seen this? |
| 02:40.14 | WilliamK | http://telephonyonline.com/voip/news/von_voip_consumers_030805/ |
| 02:40.16 | Inv_arp | Juxt: with AGI yea |
| 02:40.24 | Darwin[laptop] | <PROTECTED> |
| 02:40.37 | Juxt | Inv_arp: point me in the right direction |
| 02:40.45 | fqwer333 | Could anyone help me to make work ocho canceller on incoming calls? Thanks |
| 02:41.04 | *** join/#asterisk mrgoby (~mrgoby@pcp05308058pcs.wanarb01.mi.comcast.net) |
| 02:41.18 | mesi | With one special provider "web.de" I get the error message: handle_response: Failed to authenticate on INVITE to |
| 02:41.33 | mrgoby | ~seen jerjer |
| 02:41.41 | jbot | jerjer <~JerJer@dsl-106-170.che.centurytel.net> was last seen on IRC in channel #asterisk, 14d 10h 22m 46s ago, saying: 'mrgoby: sure'. |
| 02:41.50 | mrgoby | ha |
| 02:41.55 | mrgoby | ~seen JerJer |
| 02:41.56 | jbot | jerjer <~JerJer@dsl-106-170.che.centurytel.net> was last seen on IRC in channel #asterisk, 14d 10h 23m 1s ago, saying: 'mrgoby: sure'. |
| 02:42.17 | fqwer333 | Could anyone help me to make work ocho canceller on incoming calls? Thanks |
| 02:42.31 | mrgoby | ok, i'm having freakin wierd issues with iax lately |
| 02:42.40 | mrgoby | anyone using nufone ? |
| 02:42.51 | Juxt | yeah |
| 02:42.54 | Juxt | nufone is pretty solid |
| 02:43.02 | mrgoby | i have thought so too |
| 02:43.10 | mrgoby | maybe you could run an experiment for me ? |
| 02:43.18 | mrgoby | i've tried 2 of their switches |
| 02:43.23 | mrgoby | and two of my boxes |
| 02:43.27 | mrgoby | and 3 of my phones |
| 02:43.39 | Juxt | what do you want me to do |
| 02:43.40 | mrgoby | i cant get dtmf to my * over iax |
| 02:43.44 | fqwer333 | Anyone know how to make work echo canceller on incoming calls? |
| 02:43.45 | mrgoby | make a call file |
| 02:43.51 | mrgoby | and hit your phone with it |
| 02:43.57 | mrgoby | and then connect it to an IVR |
| 02:44.03 | mrgoby | and try to dial an extension |
| 02:44.12 | mrgoby | that is what i cant seem to do |
| 02:44.12 | Inv_arp | fqwer333: incoming pstn ZAP? |
| 02:44.15 | mrgoby | using iax |
| 02:44.27 | fqwer333 | Inv: yes |
| 02:44.46 | Juxt | i think you just went over my head :-) |
| 02:44.55 | mrgoby | there are several echo options in the zap conf file, have you tried them ? |
| 02:45.01 | fmenard | asterisk keeps sending back 401 UNAUTHORIZED... |
| 02:45.09 | fmenard | on my register requests from my 2102 |
| 02:45.29 | fqwer333 | Inv: i have been for a week trying to make it works but i dont know where to look for |
| 02:45.34 | *** join/#asterisk ctooley (~ctooley@65.166.25.111) |
| 02:45.36 | Inv_arp | fqwer333: zapata.conf think theres echo cancel lines and jitter buffer lines u can play with |
| 02:46.04 | Inv_arp | or what mrgoby said :) |
| 02:46.27 | fqwer333 | Inv: but the problem is that for outcoming calls it works, but doesn't for incoming calls |
| 02:47.40 | fmenard | now its sending 403 FORBIDDEN after I set the realm on my 2102 to the same in the matching entry in sip.conf |
| 02:48.06 | ctooley | There isn't anywhere that app_voicemail stores just the person's name is there? |
| 02:48.11 | Inv_arp | fmenard: post yer sip.conf in pastebin.ca |
| 02:48.29 | fqwer333 | Inv: I can see "Echo Cancellation: 128 taps, currently ON" for outcoming calls |
| 02:48.59 | ctooley | Like a "record your name for the company directory" so that app_directory could be somewhat useful |
| 02:49.08 | fqwer333 | Inv: and "Echo Cancellation: 128 taps, currently OFF" for incoming |
| 02:49.29 | Inv_arp | fqwer333: try echotraining=yes in zapata.conf |
| 02:50.02 | Inv_arp | i suck at ZAP chans im all SIP/IAX :) |
| 02:50.05 | fqwer333 | Inv: It is done |
| 02:50.15 | Inv_arp | fqwer333: reload call |
| 02:50.57 | fqwer333 | Inv: reload call? |
| 02:51.38 | fqwer333 | Inv: i am using 1.0.2 |
| 02:51.53 | *** join/#asterisk bonez41 (~aint@c-67-166-77-14.client.comcast.net) |
| 02:52.11 | Inv_arp | fqwer333: type reload in console |
| 02:52.36 | fmenard | done |
| 02:52.40 | fqwer333 | Inv: I have done, but it doesnt work |
| 02:52.53 | fmenard | kill it after... I forgot to take out my fwd password |
| 02:52.57 | mrgoby | ok, this is killin me |
| 02:53.16 | mrgoby | Juxt helped me out, and it works for him....must be something with my account |
| 02:53.24 | mrgoby | or her (sorry) |
| 02:53.27 | mrgoby | ;-) |
| 02:53.52 | fqwer333 | Inv: if i write "zap show channel 1", i see that echo canceller doesnt work for incoming calls |
| 02:54.58 | *** join/#asterisk ApEtc (apetc@ip68-99-136-197.ph.ph.cox.net) |
| 02:59.50 | *** join/#asterisk NormAst (~NormAst@70.49.170.113) |
| 02:59.52 | *** join/#asterisk ben_amezcu (ben@85.48.15.128) |
| 02:59.53 | fmenard | \ |
| 03:00.05 | *** part/#asterisk _570RM_ (~storm@80.172.149.171) |
| 03:00.11 | NormAst | Hi All.. |
| 03:01.22 | brc_ | mrgoby, is the dtmf getting to your asterisk box? |
| 03:01.38 | mrgoby | i dont think so.... iax2debug shows nothing |
| 03:01.48 | brc_ | well there's your problem |
| 03:01.51 | brc_ | sip phones? |
| 03:01.54 | mrgoby | i know |
| 03:01.55 | mrgoby | iax |
| 03:02.05 | mrgoby | shouldnt be an issue |
| 03:02.09 | brc_ | iax phone => your box => nufone => pstn? |
| 03:02.09 | mrgoby | i have no idea |
| 03:02.42 | mrgoby | no, call file => my * box => iax => nufone => pstn |
| 03:02.58 | brc_ | eh? |
| 03:03.03 | brc_ | oh |
| 03:03.10 | mrgoby | i'm using a call file to generate an iax call to nufone |
| 03:03.21 | brc_ | which connects to an ivr on your box? |
| 03:03.27 | mrgoby | then connecting it to my extensions, exactly |
| 03:03.30 | brc_ | what happens when you call your ivr form a cellphyone? |
| 03:03.41 | brc_ | (pstn) |
| 03:03.48 | mrgoby | if i dial in via my fxo module on the box, it works great |
| 03:04.06 | mrgoby | i think it is nufone, but Juxt just tried it for me using their network and it worked fine |
| 03:04.19 | brc_ | odd |
| 03:04.22 | mrgoby | indeed |
| 03:04.25 | mrgoby | has me stumped |
| 03:04.29 | mrgoby | i've tried 2 boxes |
| 03:04.29 | brc_ | file a support ticket |
| 03:04.33 | mrgoby | i have |
| 03:04.34 | brc_ | support@nufone.net |
| 03:04.36 | brc_ | oh |
| 03:04.38 | mrgoby | :-) |
| 03:04.42 | brc_ | heh.. |
| 03:04.44 | mrgoby | they take too long |
| 03:04.48 | brc_ | heh.. |
| 03:04.50 | mrgoby | i'm seizing my own destiny |
| 03:05.16 | mrgoby | well, i did write today, and it is sunday, so... i'll give em a while before i start bitching |
| 03:05.24 | mrgoby | ;-) |
| 03:05.55 | mrgoby | Juxt: which host are you using with nufone ? i wanna see if that makes a difference |
| 03:06.02 | *** join/#asterisk Inv_arp (junya@adsl-3-237-170.mia.bellsouth.net) |
| 03:06.08 | ben_amezcu | Do anyone know how to make work echo canceller on incoming calls? It works for outcoming but not for incoming |
| 03:06.46 | mrgoby | ben_amezcu: do you have the echo cancellation vars set ? |
| 03:07.45 | mrgoby | i know i asked you that before, but i'm too lazy to scroll |
| 03:07.49 | *** join/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
| 03:07.55 | *** part/#asterisk Juxt (~Juxt@sfl-dsl-64-135-113-4-cust.host.net) |
| 03:07.56 | ben_amezcu | mrgoby: sorry, what is that? |
| 03:08.42 | ben_amezcu | mrgoby: echocancell=yes? |
| 03:08.49 | mrgoby | yes |
| 03:09.01 | ben_amezcu | mrgoby: it is done |
| 03:09.17 | mrgoby | echotraining=yes |
| 03:09.17 | mrgoby | <PROTECTED> |
| 03:10.30 | ben_amezcu | mrgoby: when i type "zap show channel 1" in an incoming call in this channel, i see echo canceller OFF |
| 03:10.48 | mrgoby | did you unload and reload the modules ? |
| 03:10.59 | mrgoby | what type of device, fxo ? |
| 03:11.13 | ben_amezcu | mrgoby: Tor2 |
| 03:11.18 | ben_amezcu | PRI |
| 03:11.32 | mrgoby | ah.... i dont really know those devices |
| 03:11.37 | mrgoby | are you bridging ? |
| 03:11.51 | ben_amezcu | no |
| 03:11.53 | mrgoby | all pstn, or pstn to voip ? |
| 03:12.07 | ben_amezcu | pstn to voip |
| 03:12.19 | ben_amezcu | pstn <--> voip |
| 03:12.45 | mrgoby | and you reloaded modules after enabling echo cancel in the conf file ? |
| 03:12.53 | mrgoby | and restarted * ? |
| 03:12.57 | ben_amezcu | yes |
| 03:13.01 | ben_amezcu | all done |
| 03:13.12 | mrgoby | hm |
| 03:13.22 | ben_amezcu | restart the server too!! :) |
| 03:13.32 | ben_amezcu | But it doesnt work |
| 03:13.37 | ben_amezcu | it is strange! |
| 03:13.50 | mrgoby | no idea then... yeah, i'm having a bit of confusin myself |
| 03:14.10 | ben_amezcu | Thanks at all! |
| 03:15.28 | *** part/#asterisk ben_amezcu (ben@85.48.15.128) |
| 03:15.58 | *** part/#asterisk mrgoby (~mrgoby@pcp05308058pcs.wanarb01.mi.comcast.net) |
| 03:16.44 | nestAr | bah |
| 03:16.50 | nestAr | my beer is all foamy |
| 03:22.32 | bonez41 | a beer's what I need right now.. |
| 03:22.40 | bonez41 | nestAr, can you email me some. |
| 03:23.14 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivflf1.dialup.mindspring.com) |
| 03:23.24 | bonez41 | is there any online resource that can explain to a newbie how the pstn system works...from one end to the other..and how asterisk fits in the picture? |
| 03:24.50 | nestAr | look for stuff on "what is a pbx" |
| 03:24.57 | nestAr | because that's what asterisk is |
| 03:25.07 | *** join/#asterisk Essobi (kstone@75.137.26.216.host.teledvance.com) |
| 03:25.26 | bonez41 | ok |
| 03:25.54 | nestAr | generally a PBX is a hardware box... asterisk is just software |
| 03:28.14 | Moc | well PBX is hardware that run a software |
| 03:28.50 | Moc | it just in a different package (and interface card, traditional pbx use switching) |
| 03:37.59 | *** join/#asterisk CoolAcid (~jk@216.99.98.39) |
| 03:42.51 | nestAr | yeah yeah yeah |
| 03:42.54 | nestAr | semantics |
| 03:50.38 | bonez41 | nestAr, with asterisk..and digium card or sipura 2100 et al...combined with VOIP service to provide a line or two incoming...am I limited to just 4 to 5 incoming calls, or will running asterisk allow many more to come in..and be managed, when it's busy? |
| 03:51.39 | CoolAcid | Anyone know of a good Java Based Embedable Web/softphone? |
| 03:51.50 | nestAr | asterisk can only accept as many incoming calls as you have lines.. |
| 03:52.31 | nestAr | if you're using a voip provider for DID's/incoming calls, then you're limited by the number of simul calls they're allowed |
| 03:58.09 | bonez41 | nestAr, ok...what advantage do I then gain, for example, at my office, I have a standalone DSL line, and run VOIP from Vonage..with two lines...by running asterisk instead? I still have to have vonage or broadvoice or SOMEONE to provide the number to me.....right? |
| 04:00.06 | shido6 | . |
| 04:00.19 | nestAr | yes |
| 04:00.19 | shido6 | bonez39 |
| 04:00.24 | nestAr | asterisk is a pbx |
| 04:00.48 | shido6 | nufone allows as many simultaneous calls as your bandwidth will allow |
| 04:00.58 | shido6 | but other providers may limit you |
| 04:01.27 | nestAr | it's the difference between having a pots line with a plain phone hooked up and having a phonesystem to provide IVR, voicemail system, extenesion to extension dialing, conference calling, call parking, etc. |
| 04:01.37 | habakuk | shido6, for non 800#s as well? |
| 04:02.00 | shido6 | Michigan DIDs we allow 5 or so... just dont abuse it... |
| 04:02.10 | habakuk | so its not unlimited |
| 04:02.23 | shido6 | no such thing as unlimited |
| 04:02.39 | shido6 | your bandwidth will limit your calls even on 8xx from nufone |
| 04:02.47 | shido6 | from or to |
| 04:03.13 | habakuk | not if I have a 100MB unlimited connection |
| 04:03.29 | denon | if you have a 100MB connection, then you'll be limited to 100MB |
| 04:03.37 | denon | unless you mean 100Mb.. then you'll be limited to that |
| 04:03.49 | nestAr | TANSTAAFL |
| 04:04.21 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 04:05.48 | *** join/#asterisk jeffik (~jeffik@m549236d0.tmodns.net) |
| 04:05.48 | habakuk | denon how many calls can you have with 100Mb? My point is when providers make these comments they really can't handle "as much as your bandwidth allows" |
| 04:06.15 | mikegrb | habakuk: they so can |
| 04:07.15 | denon | habakuk: I doubt you'd have a problem with nufone, for example, pushing ~250 simultaneous calls. Anything over that, you'd certainly want to talk to the provider and make sure they allocate that much capacity to you |
| 04:07.45 | denon | heck, anything over 50 simul calls, I'd talk to them .. for both of your sakes |
| 04:07.58 | denon | in most cases, they'd be more than happy to add the capacity, if you're going to use it |
| 04:08.22 | habakuk | well I don't deal with nufone myself. but you're right its best to talk with them. So they don't have any surprises |
| 04:09.13 | *** join/#asterisk fmenard_ (fmenard@109-75.tr.cgocable.ca) |
| 04:09.30 | fmenard_ | ne1 can help me with a failure to register in sip problem? |
| 04:09.46 | *** join/#asterisk km- (~pgrace@aeneas.fierymoon.com) |
| 04:09.46 | denon | habakuk: I've had great success with nufone |
| 04:10.00 | km- | howdy! |
| 04:10.11 | fmenard_ | asterisk pumps out 403 FORBIDDEN in response to register requests |
| 04:11.29 | habakuk | denon, they may be better now. But I had a horrible experience with them last year... never again.. support was ok on the things they knew about (SIP was not one of them) and just really unprofessional attitude |
| 04:12.27 | habakuk | but thats just me.. like I said maybe they've gotten better since then |
| 04:13.09 | denon | habakuk: yeah - they're doing a lot more SIP now .. initially they preferred IAX |
| 04:13.09 | fmenard_ | chan_sip.c:7681 handle_request: Registration |
| 04:13.12 | denon | but things have changed |
| 04:13.18 | fmenard_ | where would this be logged^ |
| 04:13.24 | denon | less idealistic, more flexible :) |
| 04:14.13 | habakuk | I have emails detailing exactly what was wrong with his sip implementation. Referencing exactly where in the rfc 3261 he was wrong. I kept getting a generic " Its your configuration problem" |
| 04:14.25 | *** part/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
| 04:14.31 | habakuk | it was ridiculous |
| 04:15.32 | NTJOCK | hey gang |
| 04:15.51 | NTJOCK | if I have extensions that start with 5, what is the best way to do a "dial extension" from the IVR? |
| 04:15.53 | denon | habakuk: i'd give it a shot again .. with a little tinkering, I doubt you'd need to hassle support |
| 04:16.11 | NTJOCK | can I do a pat = 5XX => dial(sip/$ext) type thing? |
| 04:16.46 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 04:18.22 | habakuk | denon, how much do they charge for 800 #s? |
| 04:18.26 | denon | free |
| 04:18.32 | denon | you just have to pay the minutes |
| 04:18.42 | habakuk | thats not bad. How much? |
| 04:19.03 | denon | dunno, hit nufone.net |
| 04:19.05 | denon | think its 2c |
| 04:19.14 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
| 04:19.40 | habakuk | yeah thats a little step for my volume |
| 04:19.56 | shido6 | what kind of volume? |
| 04:20.10 | denon | everyone will get cheaper based on volume .. |
| 04:21.04 | NTJOCK | 2c seems pretty cheap |
| 04:21.10 | *** join/#asterisk BurnedOutGeek (nobody@res-66-191-172-161.spa.sc.charter.com) |
| 04:21.11 | NTJOCK | guess it depends on how much you do |
| 04:21.12 | NTJOCK | ;) |
| 04:21.30 | NTJOCK | I'm trying to get my inbound callers to be able to dial an extension and it's not behaving.... |
| 04:21.52 | NTJOCK | I was thinking it should be exten=> _5xx,1,Dial(sip/${exten}) |
| 04:21.59 | NTJOCK | what am I not doing right here? |
| 04:22.08 | NTJOCK | my inbound callers have no privelages to speak of. |
| 04:22.41 | km- | denon: sup dude! |
| 04:23.23 | denon | hey km, how goes? |
| 04:23.41 | km- | eh, its goin. Tryin to put some extra debug in zaptel.c so I can figure out why my T1 isn't winking |
| 04:24.07 | km- | kram tells me I should call digium tech support -- have you had any experience with them in regards to weird T1 problems? |
| 04:24.33 | km- | I fear I'm going to get a "uh, what's prewink?" and that's going to be about it. I get a mental image of someone telling me I should just buy 24 iaxys and get a channel bank or something |
| 04:24.48 | NTJOCK | they seem better then that |
| 04:24.48 | NTJOCK | :) |
| 04:24.54 | denon | haha |
| 04:24.59 | km- | hehe, I hope so, it's probably going to come down to calling them soon |
| 04:25.01 | denon | I'd call tech before I spent too much time on it if I were you |
| 04:25.23 | km- | it sucks because the phone company winks to me fine, the NEC pbx winks to asterisk fine, but asterisk is not winking to either side |
| 04:25.23 | *** join/#asterisk Mike (~mike@201.135.48.217) |
| 04:25.40 | km- | heh, tzanger and I already spent mounds of time trying to figure it out, it's about the damned weirdest thing |
| 04:28.58 | *** join/#asterisk IOscanner (~IOscanner@c-67-162-251-133.client.comcast.net) |
| 04:31.55 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 04:31.59 | *** join/#asterisk atmel (~vlad@ip68-4-101-199.oc.oc.cox.net) |
| 04:33.25 | Jones | anyone here use amp to config .. I have some questions on how the config files it makes processes calls |
| 04:33.45 | *** join/#asterisk Jer13261 (~Jer@rdu57-251-152.nc.rr.com) |
| 04:34.51 | Jones | Ask it .. who knows .. mabey even I could help |
| 04:34.51 | Jer13261 | using a SPA-2000 device is there anyway to place a call on hold without parking it? |
| 04:35.18 | Jones | I dont use thoes .. so I cant say |
| 04:35.49 | Jer13261 | oh well....:) thanks wanyway |
| 04:37.09 | shido6 | flash hook |
| 04:37.13 | shido6 | should work |
| 04:37.23 | shido6 | but |
| 04:37.27 | shido6 | thats 3 way calling |
| 04:37.28 | shido6 | n/m |
| 04:37.34 | Jer13261 | right..it does |
| 04:37.52 | Jer13261 | how come 3 way calling does an atxfer and not a old fashion 3 way? |
| 04:38.03 | *** join/#asterisk lilneon (~tj_r3@cuscon12826.tstt.net.tt) |
| 04:38.05 | Jer13261 | if i 3 way a call and hang up ... the other 2 parties stay connected together |
| 04:38.10 | lilneon | hi everyone |
| 04:39.11 | herag | sometime last week my broadvoice service made some changes to their settings, I had to add an authname parameter to my sip.conf, that fixed the general dial out issue, but now call forwarding won't work, in the sense that if I have an incoming call from broadvoice, I can no longer forward that call out through broadvoice to another number, it will dial, but then when the call connects, I get a bunch of SIP 400 errors, any ide |
| 04:39.11 | herag | as? http://pastebin.ca/7388 |
| 04:39.12 | lilneon | i'm in search of a good ISP or broadband provider? the only one here. sucks big time.. any recommendations? |
| 04:39.15 | atmel | elf |
| 04:39.44 | mikegrb | herag: this is not #broadvoice support |
| 04:39.47 | mikegrb | herag: check google |
| 04:41.51 | herag | mikegrb: thank you for your help...but I have checked google, and been in contact with broadvoice tech support, this is however a very _recent_ issue, and nothing of relevance seems to be up...I was hoping someone here might have an idea or workaround |
| 04:42.20 | mikegrb | herag: I am not a bv user and I knkow for a fact google has the answer |
| 04:43.18 | *** part/#asterisk lilneon (~tj_r3@cuscon12826.tstt.net.tt) |
| 04:43.57 | herag | mikegrb: well, if you would be so kind, could you point me in the right direction? I have spent hours on this problem already, but to no avail |
| 04:44.18 | mikegrb | add site:lists.digium.com to the end of your google query |
| 04:44.20 | mikegrb | or just go there |
| 04:45.00 | km- | denon: do you know much about T1's? |
| 04:45.56 | km- | Asterisk is definitely not winking either side when it originates a call |
| 04:46.08 | km- | I have two T1's, and it's not winking on either one of em |
| 04:46.20 | km- | either side will wink it just fine, but asterisk itself will not initiate winking |
| 04:48.22 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 04:48.22 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 04:48.38 | km- | brian! |
| 04:48.47 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || 1.0.7 RC - bug #3746 || Dev Conf 1PM CST MARCH 17th -> IAX2/guest@66.250.68.194/996 || ClueCon Dev Conf Aug 3rd - 5th |
| 04:48.52 | km- | bkw: why wont asterisk wink at me? I'm an astmaster for chrissakes! :P |
| 04:49.15 | bkw_ | no clue |
| 04:49.20 | bkw_ | telco crack |
| 04:49.25 | Jones | that wink issue sounds more like something in the config for the card |
| 04:49.26 | km- | nah man, thats the weird thing |
| 04:49.33 | bkw_ | ok ok |
| 04:49.33 | Jones | not a * thing |
| 04:49.40 | bkw_ | we got HDTV today |
| 04:49.42 | km- | jones: I'd be willing to bet that it's a problem with my config |
| 04:49.49 | bkw_ | just waiting on voom to get installed next week |
| 04:50.00 | km- | jones: problem is, I've had some pretty technical guys look at it and they can't seem to figure out whats wrong with it |
| 04:50.03 | Jones | yea .. but its in the config for the t-1 card .. |
| 04:50.13 | km- | jones: I'm not above believing that I screwed up :) |
| 04:50.28 | Jones | well .. its YOUR t-1 ?? |
| 04:50.36 | Jones | you have control of both sides ? |
| 04:50.40 | Jones | you have a t-bird ? |
| 04:50.42 | km- | it goes like this |
| 04:50.48 | km- | CO ---> Asterisk -----> Legacy PBX |
| 04:50.54 | km- | the CO calling Asterisk produces a wink |
| 04:50.59 | km- | the Legacy calling PBX produces a wink |
| 04:51.01 | Jones | OH .. so its a did t |
| 04:51.08 | km- | Asterisk calling either the CO or the PBX is no winkie |
| 04:51.32 | Jones | is the call moving thru * proper |
| 04:51.40 | km- | for the co-> ast, it goes zt_start, zt_prewink, zt_wink, zt_debounce |
| 04:51.55 | km- | well, the calls that are coming from the CO are routed properly because the CO is winking right |
| 04:51.58 | Jones | you watching this where ?? on the span with a bird ? |
| 04:52.09 | km- | I've got like 300 lines of debug in zaptel.c |
| 04:52.51 | km- | The asterisk box, when it's calling the CO or the legacy system, goes zt_start,zt_txstate_start,zt_txstate_afterstart and just hangs there |
| 04:52.57 | Jones | if you have a bird .. Personaly I would use it .. that way you dont have to worry about the card being misconfiged |
| 04:52.59 | km- | and the line always stays in "Ringing" state |
| 04:53.00 | bkw_ | the legacy PBX could be fucked |
| 04:53.04 | km- | nope |
| 04:53.10 | km- | the problem isn't the legacy pbx |
| 04:53.20 | km- | the problem is definitely isolated to the Asterisk system |
| 04:53.32 | km- | because the Legacy box will wink asterisk fine |
| 04:53.36 | km- | the CO will wink asterisk fine |
| 04:53.40 | km- | but asterisk will wink neither side |
| 04:53.43 | *** join/#asterisk amsterdam (~ak@p5087BDBB.dip.t-dialin.net) |
| 04:53.53 | Qwell | Are they buying asterisk a drink first? |
| 04:53.58 | km- | jones: a "bird"? |
| 04:54.00 | Qwell | I wouldn't wink without a drink either. |
| 04:54.02 | Jones | t-bird |
| 04:54.06 | km- | qwell: hahaha, if only it was that simple |
| 04:54.08 | Jones | test box |
| 04:54.20 | km- | jones: heh, I dont have the budget to buy a signal debugger |
| 04:54.31 | Jones | most folks dont .. |
| 04:54.35 | km- | if the signalling was wrong |
| 04:54.38 | km- | I wouldnt be getting phone calls |
| 04:54.39 | km- | besides |
| 04:54.44 | Jones | but it helps SO much on something like this |
| 04:54.46 | km- | I confirmed with the phone company that it is esf,b8zs |
| 04:54.56 | km- | so the problem at this point is with e&m wink |
| 04:55.04 | Jones | thats low lvl .. |
| 04:55.18 | Jones | where is this ?? usa ?? |
| 04:55.21 | km- | yeah, usa |
| 04:55.36 | Jones | the reason your not using pri is ? |
| 04:55.55 | km- | the package we have is integrated access |
| 04:56.01 | km- | 16 vox, 8 data lines |
| 04:56.06 | km- | the PRI would have been more expensive, I guess |
| 04:56.23 | Jones | only sometimes .. but pri is easier to debug |
| 04:56.39 | Jones | can you watch the abcd bits ? |
| 04:56.42 | km- | to change to PRI would violate two seperate items of this rollout |
| 04:56.49 | km- | oh sure, I know what the states are |
| 04:57.11 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
| 04:57.26 | km- | just a sec |
| 04:57.44 | Jones | a wink is just a playing with thoes bits |
| 04:57.58 | km- | Mar 13 23:37:56 pbx01 kernel: Setting bits to 15 for channel TE4/0/2/12 state 2 in 64 signalling |
| 04:57.58 | km- | Mar 13 23:37:57 pbx01 kernel: Setting bits to 15 for channel TE4/0/2/12 state 1 in 64 signalling |
| 04:58.08 | km- | that's from the ast->legacy side |
| 04:58.52 | km- | Mar 13 23:41:34 pbx01 kernel: Setting bits to 15 for channel TE4/0/1/12 state 1 in 64 signalling |
| 04:58.52 | km- | Mar 13 23:41:34 pbx01 kernel: Setting bits to 0 for channel TE4/0/1/12 state 0 in 64 signalling |
| 04:58.52 | km- | Mar 13 23:41:35 pbx01 kernel: Setting bits to 15 for channel TE4/0/1/12 state 1 in 64 signalling |
| 04:59.00 | km- | that's when the CO is calling the asterisk system |
| 04:59.14 | Qwell | ~pastebin |
| 04:59.15 | jbot | methinks pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 04:59.44 | km- | apologies qwell, it was only that much worth of data, so it seemed within the realm of chatroom pasteability |
| 05:00.32 | Jones | and your SURE that * is trying to send the call to the span ? |
| 05:00.45 | km- | yes, because, the call does connect |
| 05:00.53 | km- | but asterisk never sees the call status change from Ringing |
| 05:01.03 | km- | from the beginning of the call till the time the person hangs up the call state remains "Ringing" |
| 05:01.18 | km- | it never changes to "Up" |
| 05:01.19 | *** join/#asterisk hmodes (hmodes@pcp0010853935pcs.potshe01.pa.comcast.net) |
| 05:01.21 | Jones | a ring is just a toggle of the a and b bits |
| 05:01.22 | km- | hunter! |
| 05:01.30 | hmodes | whoah |
| 05:01.33 | hmodes | howdy thar km |
| 05:01.36 | km- | hehe |
| 05:01.42 | km- | what's cookin |
| 05:01.45 | km- | I bought a house in pottstown |
| 05:01.49 | hmodes | oh my |
| 05:01.51 | hmodes | 'grats |
| 05:01.54 | km- | settlement is march 31 |
| 05:02.26 | km- | hehe, you're gonna have to move in, the vp of voip services at comcast told me he was going to "accidentally" drop a 6 count to my house |
| 05:02.34 | Jones | its been a LONG time science ive played with Robed Bit Signaling .. I remember that theer was a whole bunch of this that needed to be set correct |
| 05:02.42 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 05:02.42 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:02.52 | km- | mmmm, triple OC-192 rings |
| 05:02.54 | hmodes | yup, count me there ;p |
| 05:03.00 | km- | more bandwidth than I could ever possibly use at once |
| 05:03.05 | hmodes | lemme know when you take delivery :) |
| 05:03.09 | km- | hahahaha :) |
| 05:03.19 | km- | my basement is going to be a datacenter to rival vonage |
| 05:03.33 | km- | hmodes: you play with t1 cards much with asterisk? |
| 05:04.24 | hmodes | not as much as i'd like to |
| 05:04.47 | km- | jones: I'd be really interested in switching to PRI, but the problem is that I have to have the setups coexistant, i.e., the NEC Legacy system has to be setup for PRI as well, and that'll cost a site visit by the telco jocks |
| 05:05.13 | km- | hmodes: yeah, this old e&m wink shit is atrocious |
| 05:05.29 | hmodes | yeeeah, telco is for the birds |
| 05:05.33 | Jones | really it shoudlnt |
| 05:05.52 | km- | jones: no, I mean, the people who are our "Authorized NEC systems dealer" |
| 05:06.02 | hmodes | i'm much happier having my calls delivered in a format that isn't massively retarted ;p |
| 05:06.03 | km- | i.e., $400 charge for them to come out and make a single MAC |
| 05:06.09 | km- | hmodes: hahaha |
| 05:06.12 | km- | hmodes: word |
| 05:06.21 | Jones | I dont see why a site visit .. or even a change to the pbx |
| 05:06.22 | km- | I'm also wondering why I'm not getting caller ID on my CO |
| 05:06.29 | Jones | your just changeing the one leg |
| 05:06.29 | *** join/#asterisk chetan (freetibet@24-193-188-21.nyc.rr.com) |
| 05:06.42 | Jones | but anyway |
| 05:06.53 | hmodes | oh, yeah, so the reason I popped in |
| 05:06.55 | km- | jones: because if the asterisk box dies or otherwise fails (or I get hit by a bus and noone knows what to do) the people at the office have been advised to remove the two t1 wires and put a bypass between the biscuits |
| 05:06.57 | hmodes | has dundi taken off yet? |
| 05:07.15 | Jones | Ok understood |
| 05:07.24 | hmodes | I was sortof hoping jerjer has a means for advertising routes or something |
| 05:07.32 | km- | jones: so, if I change to PRI, I'd have to change the nec system as well so that that plan could still work |
| 05:07.44 | Jones | Uh huh |
| 05:07.44 | km- | hmodes: I haven't played with dundi yet |
| 05:08.04 | km- | still trying to get these damn T1's to behave before I jump into the cooler stuff |
| 05:08.11 | hmodes | yeah, after reading the new aol tos from /. I immediately switched to skype for IM |
| 05:08.12 | km- | kinda like eating your veggies before the good stuff |
| 05:08.35 | hmodes | i'm sortof hoping dundi+iax can replace it |
| 05:08.46 | km- | yeah, or someone make a crossover |
| 05:08.54 | km- | so skype users can call asterisk users |
| 05:08.58 | hmodes | yeah, that would be ideal |
| 05:09.14 | hmodes | i mostly just want my skype calls delivered to *, and a skype im plugin for gaim ;p |
| 05:09.16 | km- | jones: do you know much about caller ID with e&m wink spans? |
| 05:09.19 | hmodes | i'd be perfectly happy with that |
| 05:09.25 | km- | hmodes: hehe, word |
| 05:09.37 | hmodes | but i think dundi could do the same thing with less reliance on an evil corporation |
| 05:10.56 | Jones | only that its 1200 baud transmited between teh rings |
| 05:11.20 | km- | hehe |
| 05:11.22 | hmodes | mmm, shady hacks |
| 05:12.57 | km- | what's the variable for CLID? ${CLID} or ${DNIS} or something |
| 05:13.18 | Jones | dnis is exactly teh reverse of cid |
| 05:13.29 | Jones | DNID is the dialed digits |
| 05:13.34 | Jones | DNIS even |
| 05:13.40 | km- | hmm |
| 05:13.48 | km- | ${CLID} is blank from the calls from the T1 |
| 05:14.18 | Jones | could be that they arnt sending it |
| 05:14.32 | km- | they can disable that? |
| 05:14.36 | km- | I thought that all T1's had to have ANI? |
| 05:14.37 | Jones | Im betting its something in the config for the span .. |
| 05:14.47 | Jones | no .. |
| 05:14.50 | bkw_ | since when did we have ${CLID}? |
| 05:14.51 | Jones | not a requirment |
| 05:14.59 | *** join/#asterisk topping (~topping@dhcp024-210-082-196.columbus.rr.com) |
| 05:14.59 | bkw_ | someone needs to check README.variables |
| 05:15.02 | bkw_ | :P |
| 05:15.20 | bkw_ | ${CALLERIDNUM} |
| 05:15.30 | km- | thanks bkw! I was wondering where that doc was |
| 05:15.37 | Jones | and caller id is totaly diffrent from ani |
| 05:15.45 | bkw_ | not in some cases |
| 05:16.00 | bkw_ | callerid and ANI are the same thing usually on a tollfree circuit and PRI in may cases |
| 05:16.06 | bkw_ | er many |
| 05:16.27 | Jones | it COULD be the same .. but its carried in two seprate places thru thr ss7 network .. so it doesnt HAVE to be the same |
| 05:16.40 | km- | hmm |
| 05:16.45 | bkw_ | no i'm not going to argue about it... i'm just saying what I see |
| 05:16.58 | fmenard_ | quit |
| 05:17.08 | Jones | Yea .. most of the time it is .. but its not required to be by spec |
| 05:17.11 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 05:17.12 | km- | does asterisk handle the decoding of callerid or does the zaptel driver? |
| 05:17.42 | bkw_ | km-, depends |
| 05:17.44 | Jones | caller id is a bell 212 singal sent over the audio path |
| 05:17.51 | bkw_ | no |
| 05:17.54 | bkw_ | it depends |
| 05:18.05 | bkw_ | its sent over the d channel on PRI |
| 05:18.06 | Jones | hes using e&m |
| 05:18.16 | km- | yeah, it's not a PRI span, unfortunately |
| 05:18.19 | km- | it's good old crap |
| 05:18.20 | km- | :) |
| 05:18.21 | bkw_ | isn't DTMF isn't it |
| 05:18.24 | bkw_ | on those |
| 05:18.39 | km- | someone did say something to me about that once |
| 05:18.41 | bkw_ | 212 is for POTS lines or what ever |
| 05:19.00 | km- | that the e&m wink signal was something like 7115551212*123*1442445 |
| 05:19.02 | km- | or something like that |
| 05:19.10 | Jones | it COULD be .. |
| 05:19.19 | km- | where the first part was caller id, second part was the exten being reached, and the third part was some sort of check message |
| 05:19.25 | km- | or something |
| 05:19.26 | km- | heh |
| 05:19.26 | bkw_ | the problem here is km- needs to bitch smack the telco around |
| 05:19.30 | bkw_ | or what ever |
| 05:19.31 | bkw_ | oh wait |
| 05:19.33 | bkw_ | this is a PBX |
| 05:19.44 | km- | well, in this case |
| 05:19.44 | bkw_ | bet its not to spec |
| 05:19.52 | km- | I bet I have to slap XO around a bit |
| 05:19.55 | bkw_ | try standard FXS signalling |
| 05:20.07 | bkw_ | or FXO.. or what ever |
| 05:20.17 | Jones | heheh |
| 05:20.18 | km- | I'm trying to get the clid from the CO E&M wink line to asterisk |
| 05:20.23 | Jones | I mix that up myself all the time also |
| 05:20.25 | km- | the phone system wont support caller ID even if I pushed it |
| 05:20.29 | bkw_ | Jones, me too |
| 05:20.34 | Essobi | UNF |
| 05:20.40 | bkw_ | I get all kinds of shit backwards.. dyslexic from hell here |
| 05:20.42 | km- | we tried switching my e&m wink line to FXO and it didnt work |
| 05:20.59 | Jones | S = sub .. O = office .. but thats from the point of view of the port itself |
| 05:21.01 | *** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc) |
| 05:21.05 | bkw_ | S=station |
| 05:21.07 | bkw_ | not SUB |
| 05:21.07 | *** join/#asterisk guugmember (~nachoramo@168.234.226.39) |
| 05:21.13 | Essobi | UNF UNF UNF |
| 05:21.25 | Jones | station .. sub .. whatever .. it means PHONE |
| 05:21.30 | bkw_ | hehe |
| 05:21.49 | angler | bkw_, whats up |
| 05:22.00 | Essobi | This is beutiful.. I'm updating a sql table from Manager, non-blocking.. It's fricking SWEET. |
| 05:22.33 | Jones | anyone here use AMP to do configs .. im havign a hard time figuring out some of these scriptys |
| 05:22.55 | km- | essobi: pretty nifty, what's Manager? |
| 05:23.13 | Essobi | The.. |
| 05:23.16 | Essobi | manager... port. |
| 05:23.22 | km- | oh |
| 05:23.22 | *** join/#asterisk Mikrozaur (~mikrozaur@213.206.33.132) |
| 05:23.23 | Essobi | You know.. events. |
| 05:23.26 | Essobi | :) |
| 05:23.29 | km- | thought it was a particular program that did manager stuff |
| 05:23.38 | km- | been looking for a good manager app |
| 05:23.45 | Essobi | I'm working this thing up to track state in SQL.. |
| 05:23.57 | Essobi | SQL memory tables to be exact. |
| 05:24.09 | guugmember | km- amp seems the better cost-free solution |
| 05:24.11 | km- | neato |
| 05:24.11 | Essobi | So I can write PHP code to do nifty things. :) |
| 05:24.20 | km- | guugmember: isn't amp just a configuration program? |
| 05:24.30 | Essobi | AMP is weak if you're trying to support multiple companies on a single box. |
| 05:24.31 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
| 05:24.39 | Essobi | and it's command set is pretty limited.. |
| 05:24.43 | bkw_ | angler yo yo |
| 05:24.45 | guugmember | km- what do you want your manager to to? |
| 05:24.58 | km- | I thought amp was only a configuration file generator |
| 05:25.03 | Essobi | I dled amp.. didn't like a few things.. fixed them, then deleted it. :) |
| 05:25.12 | km- | as opposed to a live manager program where you can zapbarge and hangup spans and such |
| 05:25.19 | Jones | yea .. its a config maker thingy .. |
| 05:25.33 | guugmember | Essobi, that is what I meant, amp is a good start, then you can make your own solution |
| 05:25.35 | hmodes | speaking of barge, why oh why is there still no iax/sip barge? |
| 05:25.48 | Essobi | I can transfer calls, originate calls, hangup calls, and monitor calls now from PHP.. So I get everything in SQL and I'm banging like fat tuesday. |
| 05:25.56 | Jones | im just trying to use it to learn how things work .. and its not clicking .. |
| 05:26.15 | Essobi | guugmember Frankly.. I'm a database engineer.. and what they wrote, won't scale for shit. |
| 05:26.24 | km- | hmodes: the problem is that sipbarge/iaxbarge will only work if you dont hand the call off |
| 05:26.34 | km- | hmodes: sip and IAX both support handing the call off, almost like initiating a DCC chat |
| 05:26.37 | guugmember | Essobi, are you monitoring calls in a non exclusive * environment? |
| 05:26.44 | hmodes | right, but nat |
| 05:26.50 | hmodes | i never hand calls off, evar ;p |
| 05:26.56 | Essobi | guugmember I got 3 different soft switches we rung. |
| 05:26.56 | km- | hahaha |
| 05:26.59 | Essobi | run |
| 05:27.09 | hmodes | i don't understand why i can't barge channels that are very obviously going through my box |
| 05:27.30 | hmodes | other then people being as apathetic about it as me that is |
| 05:27.52 | km- | it probably just hasnt been written because not many people keep their calls through the box? |
| 05:28.08 | topping | http://adm.hamnett.org/ |
| 05:28.35 | Essobi | guugmember No wait.. 4 softswitches.. asterisk is the 5th. :) |
| 05:28.44 | hmodes | yeah, i suppose that makes sense |
| 05:28.53 | *** join/#asterisk mazirian (~mazirian@cpe-69-203-176-65.maine.res.rr.com) |
| 05:28.55 | topping | not really a manager app tho |
| 05:29.22 | guugmember | so you are monitoring non-asterisk extensions? |
| 05:29.25 | Essobi | km- It's selective call-recording.. if someone wants to record a call, they prepend a route to the number which bridges it to *. |
| 05:29.47 | Essobi | guugmember that answer your question? |
| 05:30.11 | guugmember | thought you were joking, because of the :) |
| 05:30.20 | Essobi | Anope. |
| 05:30.41 | *** join/#asterisk clive- (~pirch@rndf-146-35-07.telkomadsl.co.za) |
| 05:30.57 | guugmember | I am working in a recording program too, but we need some functions of the other pbx´s to record them |
| 05:31.05 | guugmember | like the observing function in Avaya products |
| 05:31.44 | guugmember | we have built a WaitforSound() function from the WaitforSilence() one |
| 05:31.44 | Jones | OH recording functions .. |
| 05:31.46 | Jones | PLEASE |
| 05:31.57 | Jones | Then I could throw out this oasis system |
| 05:33.03 | Essobi | Ehh.. All of our PSTNs are H323 or SIP routers, so it's easy to inject a call leg like that. |
| 05:34.07 | guugmember | Essobi, what is a call leg? |
| 05:35.29 | *** part/#asterisk elric (~kavit@ppp114-10.static.internode.on.net) |
| 05:35.45 | clive- | its what one uses to kick with |
| 05:39.34 | Mikrozaur | Hi pipl/ I'm serch one small bug. I'm have "asterisk"+"soft iax2 client(freefly)". Clent have nuber 4444 and if he call to it's number 4444. Asterisk is crush. Any body already see it ? |
| 05:41.09 | Mikrozaur | Last message from asterisk. http://pastebin.ca/raw/7236 |
| 05:41.24 | *** join/#asterisk Tarox (someone@pD9E7B884.dip.t-dialin.net) |
| 05:43.37 | *** part/#asterisk mazirian (~mazirian@cpe-69-203-176-65.maine.res.rr.com) |
| 05:44.37 | km- | bleh |
| 05:44.40 | km- | longest e-mail ever |
| 05:45.02 | *** join/#asterisk nassy (~mark@24-193-228-118.nyc.rr.com) |
| 05:45.02 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 05:49.07 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:51.15 | *** join/#asterisk jmhunter (~jmhunter@wire3-148.razzolink.com) |
| 05:51.15 | *** mode/#asterisk [+o jmhunter] by ChanServ |
| 05:52.35 | jmhunter | canadian beer rocks |
| 05:53.01 | jmhunter | goes down like water, and gets you drunk like malt liquor |
| 05:57.15 | jmhunter | fucking feel like i drank 3 40's o OLDE ENGLISH 800 |
| 05:57.26 | km- | haha |
| 05:57.29 | km- | nice |
| 05:57.51 | jmhunter | that was meant for #moosepenis |
| 05:59.55 | *** join/#asterisk Inv_arp (junya@adsl-3-237-170.mia.bellsouth.net) |
| 05:59.58 | *** join/#asterisk tuxinator_linuxM (~tuxinator@ip68-109-146-168.ph.ph.cox.net) |
| 06:02.02 | km- | bleh |
| 06:02.11 | hmodes | bleh! |
| 06:02.20 | km- | hmodes: we got a new president at our company |
| 06:02.24 | km- | hmodes: the dude is a fucking nazi |
| 06:02.50 | km- | hmodes: the previous people in charge at the company were like "sure, whatever, come in at 10:30 every day, as long as you're doing 40 hours a week we're cool with it" |
| 06:03.19 | km- | this new dude is being a jerk and pretty much telling me that I have to be in 8:30-5:30 on top of the countless evenings I've spent there |
| 06:03.30 | hmodes | ouch |
| 06:03.39 | hmodes | that's evil |
| 06:03.46 | km- | plus I cant wear jeans anymore, gotta be dress slacks and at least polo, if not shirt and tie |
| 06:03.52 | km- | I'm thinking this guy has never been in IS before |
| 06:03.55 | hmodes | eeek |
| 06:03.57 | *** join/#asterisk Vco (~rbowler@S0106080020aa7650.wp.shawcable.net) |
| 06:03.58 | km- | probably not, he's a sales guy |
| 06:04.04 | hmodes | pwned by corporate bullshit |
| 06:04.10 | km- | next time I have to crawl under a desk I'm going to have to get him to get under the desk instead |
| 06:04.15 | km- | and see how he likes ripped dress khakis |
| 06:04.20 | hmodes | hehe |
| 06:04.25 | *** join/#asterisk yxa (~void@203.118.40.42) |
| 06:04.28 | km- | gotta wear dress shoes and shit |
| 06:04.29 | km- | and get this |
| 06:04.37 | km- | everyone in the organization *MUST* change their VM greeting every day |
| 06:04.49 | hmodes | hahahaha |
| 06:04.51 | opus___ | heh.. |
| 06:04.53 | opus___ | write a script |
| 06:04.56 | hmodes | i worked somewhere that required that once |
| 06:05.13 | *** join/#asterisk IQ (~IQ@70-59-164-164.omah.qwest.net) |
| 06:05.14 | km- | "Hi, this is peter grace, today is monday, march 14th. Nothing has changed since yesterday, so, leave a message and we'll see what happens.." |
| 06:05.23 | Inv_arp | lol |
| 06:05.24 | km- | hahaha I'll get allison smith to do my vm's every morning |
| 06:05.42 | guugmember | what is a vm gretting? |
| 06:05.43 | km- | hello.. this.. is... march.. 14... leave.. a ... message |
| 06:05.51 | hmodes | i changed my voicemail to 'i can't answer your phone call today because i might use phrases such as 'can not', please leave a message, and if I can determine your issue will not elicit such negative sentiments, i'll be sure to call you back!' |
| 06:05.53 | guugmember | ahh, ok |
| 06:06.00 | hmodes | ironically, they immediately went out of business |
| 06:06.03 | opus___ | this .. is ... . march ... one ... four.. |
| 06:06.14 | km- | guugmember: you know when someone leaves you a voicemail, they get your greeting that says "hi, whoever I am, I cannot get to the pohne, leave a message, yadda yadda" |
| 06:06.26 | km- | hmodes: rofl! |
| 06:06.39 | km- | hmodes: I miss your zapateller-like voicemail back when you were in north wales |
| 06:06.47 | jmhunter | ~seen sleep |
| 06:06.49 | jbot | sleep <~dustin@c-24-16-13-109.client.comcast.net> was last seen on IRC in channel #tacobeam, 194d 23h 29m 29s ago, saying: 'haha'. |
| 06:06.52 | hmodes | hehe |
| 06:06.59 | jmhunter | ~seen slepp |
| 06:07.00 | jbot | slepp <~slepp@S01060040f48412ad.ed.shawcable.net> was last seen on IRC in channel #asterisk, 1d 22h 57m 48s ago, saying: 'okay, i'm done. time for beer'n'TV'. |
| 06:07.02 | hmodes | yeah, that was a good one |
| 06:07.10 | jmhunter | ~seen cuban |
| 06:07.11 | jbot | cuban <~djimenez@69-150-136-26.ded.swbell.net> was last seen on IRC in channel #asterisk, 16d 5h 47m 10s ago, saying: 'brb'. |
| 06:07.20 | hmodes | i should make that my ivr ;p |
| 06:07.23 | *** join/#asterisk nassy (~mark@24-193-228-118.nyc.rr.com) |
| 06:07.26 | km- | hmodes: haha |
| 06:07.37 | km- | hmodes: I used to have zapateller when the phone got answered |
| 06:07.38 | hmodes | if you know the right extension, you'll dial it |
| 06:07.42 | km- | got too many people confused |
| 06:07.42 | hmodes | otherwise you'll be confused |
| 06:07.48 | km- | kristy's grandmom especially got really confused |
| 06:07.51 | km- | even when we told her to dial 1 |
| 06:07.51 | hmodes | hehe |
| 06:08.20 | `Sauron | All my IVR says, is "Dial 1 for <my name>" |
| 06:08.30 | `Sauron | Got rid of lots of spam.. :) |
| 06:08.33 | km- | hahaha |
| 06:08.35 | km- | nice |
| 06:08.36 | `Sauron | s/spam/telemarketers |
| 06:08.42 | km- | same thing |
| 06:08.42 | km- | hehe |
| 06:08.45 | km- | ok well |
| 06:08.53 | km- | since I have to get up with my wife tomorrow at 4:45 |
| 06:08.59 | km- | I guess I'd better get to sleep |
| 06:09.09 | km- | if my wife doesnt wake me up, I'll never wake up till like 9 |
| 06:09.14 | km- | and she leaves for work at 5 |
| 06:09.15 | hmodes | yezz |
| 06:09.16 | km- | so, meh |
| 06:09.19 | hmodes | need to sleeep |
| 06:09.29 | Inv_arp | hmm if i create seperate context that just says ; [context] include=>A include => B can contexts A see B? |
| 06:09.41 | hmodes | good to see ya' km |
| 06:09.50 | km- | hmodes: oh, I'm around a bit more often now |
| 06:09.58 | km- | hmodes: we'll have to hook up for some chinese buffet one of these days |
| 06:10.07 | km- | hmodes: specially if I'm up in pottstown and you're still in pburg |
| 06:10.10 | hmodes | hells yezz |
| 06:10.36 | km- | word |
| 06:10.37 | km- | hehe |
| 06:10.40 | km- | aight, sleep time |
| 06:10.43 | km- | later |
| 06:10.46 | *** part/#asterisk km- (~pgrace@aeneas.fierymoon.com) |
| 06:10.50 | *** join/#asterisk terrapen_ (~cjs@cpe-66-25-86-139.satx.res.rr.com) |
| 06:12.17 | tuxinator_linuxM | ~seen file |
| 06:12.19 | jbot | file <~jcolp@mctn1-0841.nb.aliant.net> was last seen on IRC in channel #tacobeam, 5h 34m 50s ago, saying: 'yes'. |
| 06:12.26 | *** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc) |
| 06:12.55 | guugmember | ~seen jsolares |
| 06:12.56 | jbot | jsolares <~jsolares@200.12.33.64> was last seen on IRC in channel #asterisk, 12h 21m 25s ago, saying: 'hehe'. |
| 06:13.29 | guugmember | ~seen shido |
| 06:13.30 | jbot | shido <~greg@d57-87-253.home.cgocable.net> was last seen on IRC in channel #asterisk, 7d 22h 8m 20s ago, saying: 'bbl'. |
| 06:13.51 | Inv_arp | shido6 i think |
| 06:14.43 | guugmember | ~seen shido6 |
| 06:14.44 | jbot | shido6 is currently on #asterisk (7h 3m 43s). Has said a total of 34 messages. Is idling for 1h 37m 16s |
| 06:15.15 | guugmember | have never used this ~seen function |
| 06:15.27 | guugmember | where can i see other functions of irc? |
| 06:16.07 | Inv_arp | guugmember: its a bot |
| 06:16.53 | opus___ | ~seen jbot |
| 06:16.54 | jbot | jbot <ibot@apt.bot.TimRiker.active.supporter.pdpc> was last seen on IRC in channel #dub, 3d 15h 12m 15s ago, saying: 'somebody said leave was what lazarus should do now'. |
| 06:17.13 | opus___ | really |
| 06:19.30 | Inv_arp | opus___: i msg earlier some1 who fixed their outgoing BV issue |
| 06:20.30 | jmhunter | BV HAS BEENa problem even since i had them last summer |
| 06:21.24 | jmhunter | BV is a hobby, not a provider |
| 06:21.27 | terrapen_ | ahhh |
| 06:21.39 | terrapen_ | i have a plate of hot, fresh-out-of-the-oven orange sweet rolls |
| 06:21.49 | terrapen_ | topped with peices of texas pecan |
| 06:21.54 | terrapen_ | and a tall glass of ice cold milk |
| 06:24.22 | terrapen_ | i am going to be sick if i eat all of this |
| 06:30.38 | terrapen_ | i found a nice new VoIP provider |
| 06:30.46 | puppet | terrapen_: free? ;P |
| 06:30.47 | terrapen_ | so far, their service has been awesome |
| 06:30.48 | terrapen_ | nope |
| 06:31.05 | puppet | best serivce ive sene is swedish rixtelecom |
| 06:31.10 | puppet | seen* |
| 06:31.11 | terrapen_ | IAX.cc is who i use now |
| 06:31.19 | terrapen_ | but i am in the US |
| 06:31.37 | puppet | im using fwd |
| 06:32.09 | terrapen_ | i actually use IAX.cc and NuFone |
| 06:32.15 | terrapen_ | each has their advantages |
| 06:32.19 | puppet | i just want so many DIDs i can get for free |
| 06:32.25 | puppet | got uk/us/german now |
| 06:32.34 | puppet | anyone know anymore sip forwarders? |
| 06:32.43 | puppet | or not forwareds just free sip account |
| 06:32.45 | puppet | or iax |
| 06:32.47 | terrapen_ | i wonder how much a German DID would run me |
| 06:32.59 | puppet | terrapen_: 0 |
| 06:33.08 | terrapen_ | and is there any DID I could get that my friends in Aachen could call me for free? |
| 06:33.19 | terrapen_ | puppet ? |
| 06:33.24 | Jer13261 | d3 offers that :) but it anit free |
| 06:33.35 | puppet | www.sipgate.de |
| 06:33.45 | terrapen_ | are intra-Germany calls free for German telephone users? |
| 06:35.36 | terrapen_ | its so cheap for me to call my German friends |
| 06:35.49 | terrapen_ | i should set up some kind of callback system for them |
| 06:36.42 | terrapen_ | it sucks that international Caller*ID doesnt really work |
| 06:36.46 | terrapen_ | sometimes it seems to work |
| 06:36.50 | terrapen_ | most of the time it doesn't |
| 06:37.02 | puppet | it does work if you dont use forwarding |
| 06:37.10 | puppet | f.ex my uk number goes to my us number |
| 06:37.14 | puppet | that goes iax ot me |
| 06:37.25 | puppet | if someone call uk number i see the uk number, my number |
| 06:37.43 | *** join/#asterisk shuric (alexander@80.70.96.195) |
| 06:37.43 | puppet | but thats more the config |
| 06:37.47 | *** join/#asterisk Nukemizer (~Nuke@66.237.85.58.ptr.us.xo.net) |
| 06:38.00 | Jer13261 | i can get cid from australia and from spain :) |
| 06:38.25 | puppet | anyone know any free sip providers with a own DiD more then uk/us/germany? |
| 06:41.43 | jmhunter | ~seen damin |
| 06:41.45 | jbot | damin <~damin@nucleus.nacs.net> was last seen on IRC in channel #asterisk, 4h 7m 10s ago, saying: 'lately: It's not Asterisk's job reall, but it can set TOS bits..'. |
| 06:42.11 | jmhunter | ~seen pfn |
| 06:42.12 | jbot | pfn <500@adsl-69-107-210-254.dsl.pltn13.pacbell.net> was last seen on IRC in channel #asterisk, 20d 4h 48m 35s ago, saying: 'only with iax'. |
| 06:42.55 | terrapen_ | i guess there probably aren't many pay-as-you-go providers in germany, eh? |
| 06:45.24 | puppet | sipgate.de |
| 06:45.27 | puppet | i think have it to |
| 06:46.37 | *** join/#asterisk santiago (~santiago@63.245.86.111) |
| 06:47.29 | nassy | where would be a good place to go on the web to learn about what a DID is |
| 06:49.08 | shido6 | Direct inward Dialxvavm |
| 06:49.16 | shido6 | local number for your ppl to call you on |
| 06:49.23 | *** part/#asterisk santiago (~santiago@63.245.86.111) |
| 06:49.25 | shido6 | and have your ovip provider ring your ip phone |
| 06:49.45 | puppet | i want a russia did for fun :/ |
| 06:50.48 | opus___ | direct inward dial |
| 06:51.13 | nassy | ah ok, so basically a service like iax.cc would allow me to connect via the internet to their asterisk servers and then they would provide me with a did phone number. i would not need a pots line? |
| 06:51.27 | opus___ | DID is the number to your pots line |
| 06:51.39 | opus___ | or what would appear to be POTS |
| 06:51.44 | opus___ | for everyone else on POTS |
| 06:51.49 | nassy | ok |
| 06:52.08 | opus___ | like, you could choice the DID of 555-555-1212 for your line |
| 06:52.38 | puppet | 360-812-2374 |
| 06:52.40 | puppet | my did :) |
| 06:52.43 | puppet | us |
| 06:56.19 | nassy | we have an awful phone company at work with a 2 year stangle hold on our PBX. we have some POTS lines and some DID's off of the T1 (I think PRI or something like that). is it possible to take my extension that comes from the PBX and hook it up to asterisk. this way i would not have to beg them to do anything. dont want to replace the company pbx |
| 06:59.41 | herag | what does a 400 bad-request-uri error signify? |
| 07:00.12 | herag | or, more importantly...how do I fix it |
| 07:06.11 | *** join/#asterisk Mikrozaur (~mikrozaur@213.206.33.132) |
| 07:07.46 | *** join/#asterisk _mountie (~mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com) |
| 07:10.16 | *** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au) |
| 07:10.26 | jmhunter | wassup bitches |
| 07:10.30 | jmhunter | :) |
| 07:10.34 | jmhunter | lalala |
| 07:12.02 | puppet | jmhunter: ;P |
| 07:16.58 | *** join/#asterisk TheEmperor (~mattn@202.187.40.2) |
| 07:17.45 | clive- | can anyone help me , I am feeling silly today, what is the easiest way for one extension to call onother extension ? |
| 07:19.01 | Inv_arp | clive-: dial it? |
| 07:19.33 | clive- | inv, somethings not working 100%, ..:( |
| 07:19.49 | clive- | I have Dial(IAX2/${EXTEN}) |
| 07:20.03 | Inv_arp | ~paste |
| 07:20.06 | jbot | paste is probably see http://paste.husk.org |
| 07:20.30 | Inv_arp | clive-: what r u tryin to do? |
| 07:20.40 | clive- | inv dial one extension to another |
| 07:21.18 | Inv_arp | clive-: and the extension are IAX phones? |
| 07:21.25 | clive- | it seems to add in the context to the dial string as I get this error: 9871234@iaxphone |
| 07:21.38 | clive- | does not exsit |
| 07:21.58 | clive- | yes |
| 07:23.55 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
| 07:25.46 | Inv_arp | clive-: eg.... [internal] exten => 100,1,Dial(IAX2/${EXTEN},20,tr) exten => 200,1,Dial(IAX2/${EXTEN},20,tr) |
| 07:26.02 | Inv_arp | creates to extensions 100 and 200 |
| 07:26.09 | Inv_arp | err two |
| 07:27.37 | clive- | maybe I should add in the ",20,tr" |
| 07:27.46 | Inv_arp | clive-: http://www.voip-info.org/wiki-Asterisk+IAX+channels |
| 07:28.12 | Beirdo | sigh |
| 07:28.56 | clive- | I think the phone I am trying to call is diconnected..:(..lol |
| 07:29.27 | Beirdo | anyone here use OpenBSD as a firewall, NAT'ing SIP? |
| 07:29.43 | Inv_arp | Beirdo: lemme gues no audio |
| 07:29.51 | Beirdo | heh |
| 07:29.57 | Beirdo | haven't got that far yet |
| 07:30.12 | Inv_arp | whats the prob |
| 07:30.20 | Beirdo | I can't get the damn return traffic to go through, and I'm sure I'm missing something obvious |
| 07:31.30 | Beirdo | 02:31:01.301327 64.26.176.126.54522 > 216.127.66.119.sip: udp 376 (DF) |
| 07:31.43 | Beirdo | 02:31:01.390963 216.127.66.119.sip > 64.26.176.126.sip: udp 625 (DF) |
| 07:31.45 | Inv_arp | forward ports 5060 and rtp ports in rtp.conf to your box |
| 07:31.52 | Beirdo | I have that |
| 07:31.58 | Beirdo | it works fine for IAX |
| 07:32.00 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 07:32.19 | Inv_arp | IAX always works :) |
| 07:32.21 | Beirdo | but for some reason, the redirect for incoming 5060/udp isn't doing squat |
| 07:32.41 | Inv_arp | Beirdo: did redir rtp ports also (udp) |
| 07:32.55 | Beirdo | on the internal IF, I see the outbound packet, but not the return |
| 07:33.29 | Inv_arp | Beirdo: maybe bcause you box is stopping it |
| 07:33.30 | Beirdo | not yet, but that's not the issue here |
| 07:34.24 | clive- | beirdo make sure your phone is sending to port 5060 |
| 07:34.35 | Beirdo | pass in proto udp from any to any port { 5060 5061 } keep state queue medium |
| 07:34.51 | clive- | on myphones, if I dont specify 5060, it sends signalling on any port |
| 07:34.59 | Beirdo | clive-: it is. this is a registration from asterisk -> SER |
| 07:35.15 | Beirdo | and yes, the source port seen by SER will not be 5060 |
| 07:35.37 | Beirdo | and SER is not replying to the port that it was sent from (stupid-ass protocol) |
| 07:36.27 | Beirdo | but the fact remains, somehow the firewall is blocking it, and I don't see how it should be |
| 07:45.41 | *** part/#asterisk jmhunter (~jmhunter@wire3-148.razzolink.com) |
| 07:47.04 | *** join/#asterisk amsterdam (~ak@xdsl-195-14-204-21.netcologne.de) |
| 07:47.57 | *** join/#asterisk RoyK (~roy@143.80-202-166.nextgentel.com) |
| 07:58.08 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
| 07:59.21 | *** join/#asterisk Astinus_ (~abba@213.167.111.138) |
| 07:59.31 | *** part/#asterisk Astinus_ (~abba@213.167.111.138) |
| 08:01.39 | *** join/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 08:04.22 | *** join/#asterisk Alexis (~alexis@www.trim.it) |
| 08:04.28 | Alexis | hello |
| 08:05.28 | MuppetMaster | Hello |
| 08:06.02 | Beirdo | http://lists.digium.com/pipermail/asterisk-users/2004-July/057007.html |
| 08:06.04 | Beirdo | wow. |
| 08:06.12 | Beirdo | precisely what I'm seeing |
| 08:07.25 | *** part/#asterisk guugmember (~nachoramo@168.234.226.39) |
| 08:07.42 | MuppetMaster | Beirdo: Have you set the sip.conf [general] options to provide the NAT details? |
| 08:07.52 | Beirdo | yes |
| 08:08.04 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 08:08.05 | Beirdo | the packets are making it back to the firewall |
| 08:08.18 | MuppetMaster | Beirdo: Then definately an openBSD firewall issue not directly related to Asterisk. Would recommend having a look at the openBSD forums, where ever they may exist. |
| 08:08.47 | MuppetMaster | Beirdo: I use a Linksys WRT54G and it works great... |
| 08:08.48 | Beirdo | uh huh |
| 08:13.29 | *** join/#asterisk RGi_- (~rgi@gw-a.adcom.stord.as) |
| 08:15.59 | Beirdo | SIP just sucks ass |
| 08:16.43 | Beirdo | I might just have to give up and run SER on the firewall |
| 08:16.47 | Beirdo | which is lame |
| 08:17.42 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 08:20.57 | shido6 | nooo |
| 08:21.16 | shido6 | might as well run asterisk on that public ip |
| 08:21.17 | shido6 | box |
| 08:21.21 | shido6 | and call it a day |
| 08:22.06 | rvhi | when i dial out to pstn, it always timeout after 6-7 rings |
| 08:22.44 | rvhi | i use Dial(Zap/r1/5551212,120) |
| 08:22.53 | rvhi | should it ring for 120sec? |
| 08:22.53 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 08:23.43 | Beirdo | I don't want to run asterisk on my OpenBSD firewall, that's just asking for compile issues |
| 08:24.06 | Beirdo | I could run it on my colo box with public IP and trunk to it with IAX |
| 08:24.15 | Beirdo | but that's a lot of extra work |
| 08:25.39 | Beirdo | of course, setting up SER as a proxy will be a lot of work too |
| 08:25.53 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 08:27.13 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 08:27.37 | *** join/#asterisk amer (~aaa@203.99.60.27) |
| 08:29.25 | *** join/#asterisk GDRA (1054@209.51.68.88) |
| 08:29.35 | GDRA | Anyone alive for a quick question? |
| 08:31.11 | amer | sure |
| 08:31.26 | amer | I might be able to help |
| 08:31.37 | GDRA | Can I specify the codec I want to use for a specific call in the call file? |
| 08:33.16 | Zeeek | GDRA I can think of an off the wall way to do this using different contexts and peers |
| 08:33.45 | shido6 | ZzzZZzz |
| 08:33.47 | GDRA | Thats what I thought to |
| 08:34.08 | GDRA | But I've been having a lot of trouble getting multiple peers for a single IP to work |
| 08:34.22 | shido6 | its not difficult |
| 08:34.27 | GDRA | It will either identify the LAST peer I define or always say INVITE AUTH failed |
| 08:34.34 | Zeeek | you mean with different codec prefs for each? |
| 08:34.37 | GDRA | Yeah |
| 08:34.39 | shido6 | GDRA what? |
| 08:34.47 | shido6 | what are you doing what are you doing what are you doing? |
| 08:34.55 | Zeeek | the Dial refers the desired peer/codec? |
| 08:35.10 | GDRA | It's like this, if I make the peer definitions "insecure" it will use the LAST peer I define for an IP |
| 08:35.20 | GDRA | If I leave them secure it never auths correctly... |
| 08:35.25 | shido6 | u dont need "insecure" in the peer statement in iax.conf |
| 08:35.30 | GDRA | I'm using SIP |
| 08:35.35 | shido6 | even wiff sip |
| 08:35.40 | GDRA | Sorry |
| 08:35.44 | GDRA | Heres what I want to do basically |
| 08:35.46 | shido6 | on the peer u can create diff users |
| 08:35.50 | shido6 | and set whatever you want for diff users |
| 08:35.54 | shido6 | even on the same peer |
| 08:35.59 | shido6 | u know what |
| 08:36.01 | shido6 | im going to bed :) |
| 08:36.02 | GDRA | Have A call B, and let me pick what codec to use when it happens |
| 08:36.10 | shido6 | well |
| 08:36.12 | shido6 | B the peer |
| 08:36.14 | shido6 | can pick |
| 08:36.18 | shido6 | or A the user can pick |
| 08:36.37 | shido6 | they both should match |
| 08:36.41 | shido6 | or agree |
| 08:36.42 | GDRA | Right |
| 08:36.49 | shido6 | so on the one |
| 08:36.54 | shido6 | u can specify all codecs |
| 08:36.56 | shido6 | and on the other |
| 08:36.58 | shido6 | specify one |
| 08:37.02 | trelane | why? language is a complete barrier to human communication why not use codecs too? |
| 08:37.08 | shido6 | so all codecs are set in [general] |
| 08:37.14 | *** join/#asterisk GhostXz (~GhostXz@CPE000d88a9ac16-CM001225419a6c.cpe.net.cable.rogers.com) |
| 08:37.16 | GDRA | I understand that part, the problem I'm having it getting the sperate definitions to sync hup |
| 08:37.18 | shido6 | or |
| 08:37.21 | shido6 | set it in the peer |
| 08:37.26 | shido6 | grr |
| 08:37.28 | GhostXz | what dose PBX stand for? |
| 08:37.35 | shido6 | Private Branch eXchange |
| 08:37.45 | shido6 | fancy smancy |
| 08:37.55 | GhostXz | Right,, ill pretend i know what that means |
| 08:37.55 | Beirdo | well, I'm gonna go to bed and deal with this mess tomorrow (remembering to tell asterisk to stop the failing SIP register first) |
| 08:37.58 | GDRA | When A calls B, it goes out on a USER and when B picks up it looks for PEERS, correct? |
| 08:38.11 | shido6 | well |
| 08:38.17 | shido6 | what are you using to get to A ? |
| 08:38.32 | GDRA | A -> B is just asterisk to asterisk |
| 08:38.33 | GDRA | Internal network |
| 08:38.36 | shido6 | no |
| 08:38.41 | shido6 | regular phone? |
| 08:38.43 | shido6 | ip phone? |
| 08:38.45 | *** part/#asterisk GhostXz (~GhostXz@CPE000d88a9ac16-CM001225419a6c.cpe.net.cable.rogers.com) |
| 08:38.46 | GDRA | No phones |
| 08:38.46 | shido6 | softphone? |
| 08:38.50 | shido6 | err k |
| 08:38.53 | GDRA | Just wav files being transmitted |
| 08:38.55 | GDRA | For testing |
| 08:39.10 | shido6 | ? |
| 08:39.29 | GDRA | It's just one linux box talking to another, no phones at all |
| 08:39.36 | shido6 | whats the point? |
| 08:39.45 | GDRA | To test the bandwidth and whatnot |
| 08:39.55 | shido6 | why not just ftp |
| 08:39.58 | GDRA | It's a project |
| 08:39.58 | shido6 | yeesh :) |
| 08:40.08 | GDRA | The bandwidth changes with different codecs and protocols |
| 08:40.18 | shido6 | yes.... |
| 08:40.33 | shido6 | so in your project you will never use any real phones? |
| 08:40.38 | GDRA | Nope, never |
| 08:40.49 | shido6 | so how are you executing the call? |
| 08:40.59 | GDRA | Random lengths |
| 08:41.21 | shido6 | I dont understand... I cant say "Random Length" and a call gets executed on your box |
| 08:41.26 | shido6 | how do you initiate the call |
| 08:41.26 | GDRA | Oh |
| 08:41.26 | shido6 | ? |
| 08:41.43 | GDRA | A program generates call files for me and inside the context an AGI plays a wav a random number of times |
| 08:41.47 | GDRA | Then just hangs up |
| 08:41.54 | shido6 | in the dialplan or agi |
| 08:42.00 | shido6 | what are ou using to call the other box |
| 08:42.04 | GDRA | Call files |
| 08:42.05 | shido6 | whats the exten line look like |
| 08:42.20 | GDRA | I make call files in /var/spool/asterisk/outgoing/ |
| 08:42.55 | shido6 | are you not using the dialplan when u make the call? |
| 08:43.12 | GDRA | I dont think so, no |
| 08:43.17 | shido6 | ok |
| 08:43.18 | GDRA | It goes to a context that executes the AGI |
| 08:43.29 | shido6 | "it" ? |
| 08:43.37 | GDRA | The call file has a Context line |
| 08:43.46 | shido6 | pastebin.ca |
| 08:43.47 | GDRA | When the call picks up asterisk executes that context for me |
| 08:44.19 | shido6 | you are dialing a user and a peer |
| 08:44.26 | shido6 | whats the user and peer |
| 08:44.33 | GDRA | My understanding is this |
| 08:45.00 | GDRA | When having A call B, A looks for a USER account for B, and when B picks up it looks for a PEER with A's host in it |
| 08:45.23 | shido6 | doesnt have to have A's host |
| 08:45.33 | shido6 | when dialing out to a peer |
| 08:45.39 | shido6 | that peer doesnt need to know the users host |
| 08:46.05 | shido6 | but |
| 08:46.11 | shido6 | you can set a codec on that peer |
| 08:46.19 | GDRA | I think par of the problem may be it seems to think it's going through a proxy but it isnt |
| 08:46.24 | GDRA | part* |
| 08:46.43 | shido6 | u know what |
| 08:46.46 | shido6 | screw it |
| 08:46.49 | shido6 | stick a disallow=all |
| 08:46.54 | shido6 | allow=pickyourcodechere |
| 08:46.56 | shido6 | on both ends |
| 08:46.57 | shido6 | the user |
| 08:46.59 | shido6 | and the peer |
| 08:47.01 | shido6 | and call it a day |
| 08:47.06 | shido6 | im going to bed now :) |
| 08:47.10 | GDRA | Goodnight |
| 08:55.47 | *** join/#asterisk j_vianna (vianna@node-40247a6a.ewr.onnet.us.uu.net) |
| 08:57.16 | j_vianna | When I receive a call from a DID (inbound) I have this error message: ast_best_codec: Don't know any of 0xf800 formats... Anyone know what is mean ? |
| 08:59.03 | feral_kid | I have having a real issue... I running A@H, and I have a Sipura 2000 and a few softphones (X-Lite). Both the Sipura and X-Lite register, but when I attempt to dial a number, not only does it not work, nothing shows up on the console... Am I missing something? |
| 09:00.06 | Zeeek | feral_kid set verbose 9999 ? |
| 09:00.35 | feral_kid | Zeeek: How do I set that, within the console? |
| 09:02.02 | *** join/#asterisk markak (~twist@ndn-165-140-201.telkomadsl.co.za) |
| 09:02.27 | markak | hi all thanks for all the help over the past few weeks server is running stable. |
| 09:02.27 | Inv_arp | feral_kid: quit the asterisk -rvvvvvvvvvvvvv |
| 09:02.43 | Inv_arp | feral_kid: err quit then asterisk -rvvvvvvvvvvvvv |
| 09:03.20 | Inv_arp | markak: oh yea what set up ya got |
| 09:04.22 | markak | redhat 9. p4 2.8 512M 80gig asterisk latest CVS Digium 4port fxo card 8 softphones using iax and 1 hardphone using sip |
| 09:04.42 | markak | after a few problems its working like a dream now |
| 09:04.55 | *** join/#asterisk Fraeggl (~ke@rkom.r-kom.de) |
| 09:05.23 | *** part/#asterisk Fraeggl (~ke@rkom.r-kom.de) |
| 09:06.22 | markak | zeeek: i read through the t and T options, can one use them in the same line. ie exten => _9.,1,Dial(Zap/g1/${EXTEN:1},20,Ttm) |
| 09:06.23 | Inv_arp | markak: hah no match for my amd 350mhz , 128M 3gig HT486 and ZAP fxo :) |
| 09:07.13 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 09:08.05 | Zeeek | markak yest use both on the same line |
| 09:08.07 | *** join/#asterisk mbranca (~matteo@80.152.73.227) |
| 09:08.17 | markak | thanks |
| 09:10.09 | feral_kid | Inv_arp: Ok, I have put that in, but I am still not having any response from the console... Neither from any calls from the X-Lite or the Sipura... But like I said, * shows that both clients show themselves being peers... |
| 09:10.56 | *** part/#asterisk markak (~twist@ndn-165-140-201.telkomadsl.co.za) |
| 09:12.22 | *** join/#asterisk tuxinator_linux (~tuxinator@ip68-109-146-168.ph.ph.cox.net) |
| 09:13.17 | *** join/#asterisk dubphil (~dubphil@80.124.137.201) |
| 09:15.17 | dubphil | Hello ! Anyone can point me in order to interface an alcatel 4400 PBX with Asterisk in order to have outgoing call on IP ? |
| 09:15.39 | *** join/#asterisk smurfix (~smurf@smurfix.developer.debian) |
| 09:15.42 | *** join/#asterisk mousex (~mousex@adsl-69-209-103-179.dsl.klmzmi.ameritech.net) |
| 09:15.51 | mousex | hi, there! :) |
| 09:15.52 | feral_kid | So can I * have a client registered, but get no data from it... |
| 09:16.34 | mousex | I'm getting started with asterisk. Is there an easy way to configure it (especially SIP accounts)? A GUI maybe? |
| 09:16.52 | Inv_arp | feral_kid: what happens when sipura dials xlite extension? |
| 09:17.12 | Inv_arp | ~wiki |
| 09:17.21 | *** join/#asterisk r3d5un (~r3d5un@81.223.189.114) |
| 09:17.34 | Inv_arp | ~wiki gui |
| 09:17.43 | feral_kid | Fast busy, and vice versa... |
| 09:17.59 | Inv_arp | feral_kid: no errors on cli? |
| 09:18.15 | feral_kid | Inv_arp: Nothing at all... |
| 09:18.43 | Inv_arp | feral_kid: are you sure verbosity is set when u started *? with vvvvvvvvvvvv |
| 09:18.49 | r3d5un | hi, is there a newer library than the iaxlib 0.2.2 (which seems pretty old). Because if i try to register at my asterisk server with my program i get "Undecodable frame"? |
| 09:19.03 | tuxinator_linux | Inv_arp: there is a max of 3 verbosity |
| 09:19.10 | feral_kid | Inv_arp: Definitely... |
| 09:19.21 | *** part/#asterisk mousex (~mousex@adsl-69-209-103-179.dsl.klmzmi.ameritech.net) |
| 09:19.32 | Inv_arp | heh i just continue on forevever vvvvvvvvvvvvvvv |
| 09:20.02 | feral_kid | Verbosity was 4 and is now 20 |
| 09:22.00 | *** join/#asterisk matjing (~Miranda@viper.wananchi.com) |
| 09:22.37 | r3d5un | no answer? |
| 09:23.05 | r3d5un | does that mean there is no library or does that mean "hey r3d5un, shut up"? |
| 09:24.31 | tuxinator_linux | Inv_arp: looking at the source code, there is only a verbosity of 0,1,2 used |
| 09:24.49 | mog_home | well up to 255 of them |
| 09:25.14 | tuxinator_linux | mog_home: ? |
| 09:27.51 | dubphil | r3d5un: you ha ve to case: 1 your question is probably a wellknown issue, very easy to fix and too often ask in this channel - 2 Your issue as never be found by anyone else here and nobody can help you |
| 09:27.55 | mog_home | amount of verbosity that is recognized in asterisk is 0-255 |
| 09:28.03 | mog_home | but i thought 0-3 was user |
| 09:28.10 | mog_home | s/user/used |
| 09:28.29 | tuxinator_linux | mog_home: where did you find that info? |
| 09:28.50 | r3d5un | dubphil: ok thanks, i will try on my own |
| 09:29.21 | feral_kid | Hell, I wonder should I just reinstall Asterisk@Home one more time... |
| 09:31.55 | feral_kid | Nice interface, but somewhat of a pain in the ass... |
| 09:32.51 | tuxinator_linux | *@home is suppose to be the easy way |
| 09:34.04 | mog_home | the source.... |
| 09:34.06 | mog_home | ^_^ |
| 09:35.06 | feral_kid | tuxinator_linux: Yeah, I know... I was just being lazy... |
| 09:41.32 | tuxinator_linux | mog_home: asterisk*CLI> set verbose 2147483627 |
| 09:41.32 | tuxinator_linux | mog_home: Verbosity was 0 and is now 2147483627 |
| 09:41.46 | tuxinator_linux | mog_home: is what it will accept |
| 09:41.57 | tuxinator_linux | but I don't think it changes anything after 2 |
| 09:46.43 | *** join/#asterisk HuangDi (~mattn@203.121.47.100) |
| 09:46.48 | *** join/#asterisk sysdef (~sysdef@sysdef.admin.debiancenter) |
| 09:49.33 | mog_home | do a grep |
| 09:49.56 | mog_home | you will see VERBOSE_PREFIX_3 |
| 09:50.06 | mog_home | and stuff at 4 |
| 09:50.37 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 09:50.37 | mog_home | but that is the last of stuff defined |
| 09:51.19 | *** join/#asterisk mesi (~player@dsl-082-083-052-192.arcor-ip.net) |
| 09:51.30 | HuangDi | hi guys, having a little problem here with NAT on my softphone |
| 09:51.41 | HuangDi | it says Firewall: Port Restricted Cone NAT |
| 09:51.44 | HuangDi | any ideas? |
| 09:51.53 | HuangDi | I've turned off all firewalls on my * server |
| 09:52.33 | HuangDi | and even modified the /etc/services to include port 5060 |
| 09:52.35 | HuangDi | for sip |
| 09:52.56 | dubphil | asking again for newcommers :) Anyone can point me in order to interface an alcatel 4400 PBX with Asterisk in order to have outgoing call on IP ? |
| 09:53.40 | tuxinator_linux | HuangDi: http://www.ietf.org/rfc/rfc3489.txt |
| 09:53.47 | tuxinator_linux | http://voip-info.org/tiki-index.php?page=NAT%20survey |
| 09:53.53 | dubphil | there is nothing on google or http://voip-info.org :-( |
| 09:53.58 | tuxinator_linux | http://voip-info.org/wiki-NAT+and+VOIP |
| 09:53.59 | mesi | Huang: What about a firewall on the machine running your softphone? |
| 09:54.28 | tuxinator_linux | using SIP or IAX? |
| 09:54.36 | tuxinator_linux | use IAX if possible |
| 09:54.52 | tuxinator_linux | http://voip-info.org/wiki-Asterisk+SIP+NAT+solutions |
| 09:54.53 | mesi | I've got a ques, too. With Flash() I can flash a Zap trunk. How can I flash a Modem trunk (BRI-card)? |
| 09:55.13 | tuxinator_linux | http://voip-info.org/wiki-STUN |
| 09:55.32 | tuxinator_linux | mesi: sorry, can't help you there |
| 09:56.04 | mesi | Tux: Yes, seems nobody is using BRI-cards ;-) Perhaps I should try the mailinglist. Thanks. |
| 09:56.43 | tuxinator_linux | I think there is not much support for BRI cards |
| 09:57.07 | tuxinator_linux | PRI cards there is a lot of support for |
| 09:57.07 | HuangDi | tuxinator_linux: using a sip soft phone.. |
| 09:57.24 | Mw3 | there's good support for active bri cards |
| 09:57.37 | tuxinator_linux | HuangDi: Do you know how SIP works? |
| 09:57.54 | mesi | Mw3: No, it's passive of course :-\ |
| 09:57.56 | HuangDi | tuxinator_linux: trouble is it was working fine until recently.. |
| 09:57.57 | tuxinator_linux | mesi: Talk to Mw3 |
| 09:58.06 | mesi | Tux: Done ;-) |
| 09:58.18 | tuxinator_linux | HuangDi: did you install service pack 2 on XP? |
| 09:58.20 | Mw3 | mesi: then you can mess with i4l linux and modem channel driver |
| 09:58.28 | Mw3 | mesi: which is not the best ;) |
| 09:58.39 | HuangDi | tuxinator_linux: yeah..but it was working fine until today? |
| 09:58.40 | mesi | Mw3: I do. But I don't find any possibility to Flash() it. |
| 09:59.11 | mesi | Mw3: Perhaps one can directly send AT-commands to the "modem" and perhaps there is an AT-command to flash a line. |
| 09:59.38 | tuxinator_linux | HuangDi: nothing changed? |
| 09:59.49 | Mw3 | i dunno :) i also have a passive bri card, but i'm gonna change to active with capi support ... |
| 10:00.11 | HuangDi | tuxinator_linux: no.. |
| 10:00.55 | tuxinator_linux | HuangDi: I'm out of ideas right now |
| 10:01.02 | mesi | Mw3: Actually my passive card can run on linux capi, too. But I couldn't manage installing it. The source-code is quite old for it is an old card. Anyway, capi drivers are proprietary. :-\ |
| 10:01.13 | HuangDi | tuxinator_linux: yeah, it's so weird.. |
| 10:01.30 | tuxinator_linux | HuangDi: Still questioning SP2 though. It could change a firewall setting on you, I suppose. |
| 10:01.58 | HuangDi | tuxinator_linux: thing is, i was messing with the * box today, not my pc.. |
| 10:02.52 | tuxinator_linux | mess with SIP on *? |
| 10:03.01 | HuangDi | tuxinator_linux: nope, didn't touch that |
| 10:03.11 | tuxinator_linux | what did you touch? |
| 10:03.27 | HuangDi | tuxinator_linux: i tried to change /etc/services to include 5060 to be allowed and turned off all firewalls |
| 10:03.38 | HuangDi | tuxinator_linux: maybe it's my network? |
| 10:04.40 | HuangDi | tuxinator_linux: thanks anyway, but I have to run now :) |
| 10:05.08 | mesi | Mw3: Ah! Found one! There is a generic telephony driver in asterisk, file chan_phone.c. Seems like dial an f will flash *any* line ;-) |
| 10:06.12 | Mw3 | mesi: nice catch :) |
| 10:06.44 | mesi | Mw3: I hope it works. Haven't tested it yet, but it looks great! :-D |
| 10:18.14 | *** join/#asterisk ckruetze (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com) |
| 10:21.32 | *** join/#asterisk RoyK (~roy@80.239.107.121) |
| 10:21.47 | RoyK | ~seen coppice |
| 10:21.50 | jbot | coppice <~chatzilla@111.196.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 17h 58m 43s ago, saying: 'yep'. |
| 10:22.29 | Zeeek | ~lart RoyK |
| 10:22.34 | RoyK | ~kill Zeeek |
| 10:22.36 | jbot | ACTION shoots a photon gun at Zeeek |
| 10:23.06 | Zeeek | yes please |
| 10:29.51 | *** join/#asterisk shadebob (~shadebob@rnis-162-206-192-81.marocconnect.com) |
| 10:30.59 | *** join/#asterisk welby (~welby@i.live.in.a.wheely-bin.co.uk) |
| 10:32.04 | *** join/#asterisk Mother_ (~m@53.Red-217-126-93.pooles.rima-tde.net) |
| 10:32.07 | Mother_ | hi all |
| 10:34.01 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 10:38.22 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 10:41.55 | tuxinator_linux | Hi Mother_ |
| 10:45.12 | Mother_ | hiya :) |
| 10:45.44 | Mother_ | hrrrmmm I cannot compile all of wavetools, I get some missing vga.h |
| 10:46.01 | tuxinator_linux | hmm, haven't play with that yet |
| 10:46.35 | Mother_ | I wanted to do call recording, but some macro examples I've found use wmix to put the two resulting files into one |
| 10:47.51 | tuxinator_linux | ya, I don't have experence there yet, sorry |
| 10:57.06 | *** join/#asterisk jhiver (~jhiver@AStDenis-103-1-2-27.w81-249.abo.wanadoo.fr) |
| 10:57.11 | jhiver | hi everybody |
| 10:57.40 | jhiver | is there a way for asterisk to sort of magically fix the volume if it's too low for VoIP calls? |
| 11:01.27 | Mavvie | jhiver: one persons low is another persons high. |
| 11:01.40 | *** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk) |
| 11:07.35 | shadebob | Hi, how i can show a popup with caller id on the screen when a phone ring? For example, I have a CISCO 7940G (192.168.1.240) and a PC (192.168.1.241). When cisco ring I want display a popup on the PC... |
| 11:07.43 | *** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk) |
| 11:08.17 | Zeeek | shadebob there is something on this on the wiki... where I don't know |
| 11:08.43 | *** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au) |
| 11:08.47 | Zeeek | I use a win freeware called Yac and the nc command on the asterisk box |
| 11:14.08 | cypromis | you can also use jabber for it ;) |
| 11:14.52 | shadebob | cypromis : I see this on last sineapps news.... |
| 11:14.57 | shadebob | zeeek : thanks |
| 11:15.19 | Zeeek | shadebob the Yac think works well with windows |
| 11:15.39 | Zeeek | I think source is avail |
| 11:15.59 | shadebob | zeeek : yes source is avail |
| 11:17.45 | Mother_ | question: when I use monitor to record to a file, in what format is it recorded? I know it's wav, but what bitrate etc? |
| 11:18.07 | Mother_ | hi Zeeeeeeeeek |
| 11:18.11 | Mother_ | ;) |
| 11:18.14 | tuxinator_linux | Mother_: http://voip-info.org/wiki-Asterisk+sound+files |
| 11:18.24 | Zeeek | hi mutha |
| 11:18.35 | Mother_ | ergh...I checked that but maybe I missed the obvious...will re-read ty |
| 11:19.39 | Zeeek | exit |
| 11:20.41 | Mother_ | not there |
| 11:23.06 | Mother_ | and trying the usual 16bit mono 8kHz results in distorted audio |
| 11:23.07 | tuxinator_linux | 8k bitrate |
| 11:24.00 | tuxinator_linux | 16bit, 8K, looks right |
| 11:24.35 | Mother_ | but it sounds like crushing a piece of paper :) |
| 11:24.43 | Mother_ | repeatedly |
| 11:24.54 | tuxinator_linux | need a better mic? |
| 11:25.50 | *** join/#asterisk cjk (~cjk@80.92.64.103) |
| 11:26.18 | cjk | hi, is there a way to originate a call to an extension,priority2 using the channel LOCAL? |
| 11:29.29 | cjk | other question: i want to "interconnect" with a mobile provider, what product to you suggest? |
| 11:29.43 | tuxinator_linux | defince internconenct |
| 11:30.22 | Mw3 | do you plan to use a gsm modem, or you will have for example e1 to their system ? |
| 11:32.02 | cjk | no |
| 11:32.06 | cjk | gsm modem or similar |
| 11:32.13 | cjk | i thought of a sip-gsm hardware |
| 11:32.17 | cjk | or something similar |
| 11:32.32 | cjk | more of an inofficial connection |
| 11:32.41 | cjk | but i need at least 8 channels |
| 11:33.41 | Mother_ | tuxinator_linux: this is recording from a Zap to SIP call |
| 11:33.50 | Mother_ | tuxinator_linux: so it's not a mic problem :) |
| 11:34.24 | *** join/#asterisk jhiver (~jhiver@ABesancon-102-1-2-188.w80-11.abo.wanadoo.fr) |
| 11:34.29 | Mother_ | cjk: SIP to GSM gateways are VERY expensive right now, try a Nokia M-32 (I believe it's M) |
| 11:34.52 | Mother_ | it gives you an FXS interface that you can use as a normal line |
| 11:34.56 | tuxinator_linux | Mother_: Don't know how to help you, good luck |
| 11:35.32 | Mother_ | http://www.nokia.com/nokia/0,8764,38621,00.html |
| 11:35.36 | Mother_ | ok |
| 11:36.51 | cjk | Mother_, what do you mean with very expensive? |
| 11:36.52 | Mother_ | you can get 8 of those into a channel bank or whatever you want to use |
| 11:36.59 | Mother_ | hold on a sec |
| 11:37.29 | cjk | k |
| 11:39.35 | jhiver | Well I have a Quescom AS400 here |
| 11:39.55 | jhiver | 12 SIP channels, costs around 7.5k euros |
| 11:40.05 | jhiver | so it's pretty expensive |
| 11:40.23 | cjk | thats pretty expensive |
| 11:42.07 | jhiver | but then again the nokia 32 is around 500+ bucks |
| 11:42.29 | jhiver | so 500x12 = 6k, plus the asterisk box and the FXO cards... |
| 11:42.31 | Mother_ | 500+???? |
| 11:42.35 | heison | zeeek: are you using YAC? |
| 11:42.41 | Mother_ | that's way over the top |
| 11:42.42 | jhiver | http://froogle.google.com/froogle?q=%22nokia+32%22&btnG=Search+Froogle |
| 11:42.57 | Mother_ | I can get nokia equivalents for around 170 euros |
| 11:43.16 | jhiver | Where? show me i'm interested :) |
| 11:44.22 | Mother_ | http://www.grupohilmon.com/ |
| 11:44.27 | cjk | what about http://www.2n.cz/products/gsm_gateways/voip/voiceblue.html |
| 11:44.33 | Mother_ | they live about 10 minutes from me |
| 11:44.50 | Mother_ | cjk: that's the expensive one I was refering to, last I looked it went for like 2000 euro |
| 11:45.19 | jhiver | darn their site is all flash I can't view it |
| 11:45.37 | jhiver | fucking stupid flash, when will ppl realize it's complete shite? |
| 11:45.41 | cjk | Mother_, 2000 euro well for 4 channels this is ok, no? a 4 port bri costs quite alot also, including the serverhardware |
| 11:46.17 | Mother_ | I think the 2000 was for 1 or 2 channels, the 4 channel cost more |
| 11:46.44 | cjk | iegh |
| 11:47.11 | jhiver | yeah around 5k |
| 11:47.33 | jhiver | the trouble with voiceblue is that it's just a sip client, not really a full featured gateway |
| 11:47.45 | jhiver | should be enough for use with * though, although never tested |
| 11:47.57 | Mother_ | really? I never really looked much into it once I knew the price :) |
| 11:49.59 | jhiver | well these things *are* expensive |
| 11:50.13 | jhiver | about this site you gave me |
| 11:50.19 | jhiver | has it got HTML pages at all? |
| 11:50.25 | jhiver | I can't get through the first flash page |
| 11:50.26 | Mother_ | no, it's a PITA |
| 11:50.33 | Mother_ | hold on a sec |
| 11:51.05 | *** join/#asterisk Weezey (Weezey@lan6.LO.iasl.com) |
| 11:51.16 | Mother_ | they live about 10 minutes from them |
| 11:51.20 | Mother_ | from me |
| 11:51.44 | jhiver | cause @ 170 a pop you could make a 4 channel gateway for quite cheap... |
| 11:51.50 | Mother_ | yep |
| 11:52.16 | Mother_ | I'm buying GSM OEM modems for about 60 euros now, add a few passive components and you're done |
| 11:52.44 | Mother_ | maybe a cheap microprocessor to handle dialing etc. and a DTMF detector, no more than 30 euros extra in parts |
| 11:52.45 | cjk | Mother_, ok tell me more about those modems |
| 11:52.51 | cjk | how do you connect them to * |
| 11:53.00 | jhiver | GSM modems? Can you use them for voice with *? Seems like serious hackstuff to me... |
| 11:53.18 | Mother_ | well these are OEM, i.e. a PCB with 120-pin small-pitch connector that you have to integrate into a PCB |
| 11:53.57 | Mother_ | all you are doing is adding the analog side of the telephone line to simulate a POTS jack, and a DTMF detector to pass the digits to the GSM modem via a microprocessor |
| 11:54.13 | cjk | Mother_, ok thats hack stuff |
| 11:54.16 | jhiver | it would be nice to have cheap-ish asterisk supported GSM boards... |
| 11:54.19 | Mother_ | it's not really *that* hard to, I just don't have the time and rather buy them :) |
| 11:55.03 | Mother_ | the hardest part would be to implement a full SIP or IAX stack that goes against the modem - that would mean digitising sound, compression, all the network stuff, etc. etc. |
| 11:55.49 | jhiver | All currently out of my technical knowledge :à |
| 11:56.24 | jhiver | So I guess I'm stuck with paying meunay... :) |
| 11:57.16 | jhiver | Or what would be real nice would be some GSM modules for the TDM cards |
| 11:57.23 | jhiver | that'd be really cool |
| 11:57.35 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 11:59.21 | tuxinator_linux | Night guys |
| 12:01.37 | Mother_ | jhiver: you can connect these "line simulators" to analog ports on, for example, a TDM400P |
| 12:01.58 | Mother_ | all you need is the FXO board(s) |
| 12:09.15 | jhiver | yeah I know but these Nokia 32 'line simulators' aren't cheap... cheapest I could find is 400 euro per unit |
| 12:11.21 | *** join/#asterisk RoyK (~roy@80.239.107.121) |
| 12:11.35 | Mother_ | whoa |
| 12:11.58 | Mother_ | as usual, with Nokia you pay for the image |
| 12:13.40 | jhiver | well I don't know... have you actually *tested* cheaper ones that *work*? |
| 12:13.58 | *** join/#asterisk Jas_Williams (~Jason@host81-155-66-178.range81-155.btcentralplus.com) |
| 12:14.38 | *** join/#asterisk navy31 (~navy@maket3.crec.mipt.ru) |
| 12:14.41 | navy31 | hi ppl |
| 12:14.45 | Mother_ | well, they are a very simple concept, I haven't tried them with * specifically, but they are basically an analog phone line |
| 12:14.57 | Mother_ | they give you line voltage, signalling, etc. |
| 12:15.34 | Mother_ | in the couple of places I've installed * they were given Nokia Premicells by the operator for free :) |
| 12:15.44 | Weezey | I saw some guy built a rotary phone with one of those. |
| 12:16.31 | jhiver | what is a 'rotary phone' ? |
| 12:17.03 | *** join/#asterisk Fraeggl (~Fraeggl@rkom.r-kom.de) |
| 12:17.40 | Mother_ | bwaaaahahahaha |
| 12:18.04 | Mother_ | are you so young you don't remember those phones with rotating dials? |
| 12:18.31 | Chuji | ~rotary |
| 12:19.05 | jhiver | oh yeah of course |
| 12:19.12 | jhiver | it's just the terminology |
| 12:19.24 | jhiver | you know that thing about english not being your own language... |
| 12:19.52 | jhiver | it's called pulse dialing anyway god damn it |
| 12:20.13 | Mother_ | lol yeah |
| 12:20.27 | jhiver | and I'm like 27 so I'm like old for a geek man :) |
| 12:21.11 | jhiver | that being said I was very young when we had one of those... 7 or 8 maybe |
| 12:21.19 | jhiver | I used to love dialing '9' :) |
| 12:21.37 | jhiver | even better, '0' - joy! |
| 12:21.50 | Mother_ | lol |
| 12:21.52 | Chuji | that I remember |
| 12:22.01 | Mother_ | I had a few of those made of bakelite at home |
| 12:22.16 | jhiver | what's bakelite? |
| 12:22.30 | RoyK | ~seen coppice |
| 12:22.32 | jbot | coppice <~chatzilla@111.196.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 19h 59m 25s ago, saying: 'yep'. |
| 12:22.32 | Mother_ | that black plastic stuff they made old phones out of |
| 12:22.40 | jhiver | ah ok |
| 12:22.53 | jhiver | oohh that's cool this command |
| 12:22.58 | jhiver | ~seen p0lar |
| 12:22.59 | jbot | p0lar <~p0lar@64.254.225.62> was last seen on IRC in channel #asterisk, 2d 18h 53m 44s ago, saying: 'closer'. |
| 12:23.19 | *** join/#asterisk ennuyeux72 (~ennuyeux7@83.146.53.34) |
| 12:23.22 | jhiver | kewl :) |
| 12:25.08 | jhiver | ok I have installed stupid flash so I can go and check that stupid website |
| 12:25.12 | jhiver | god I hate flash |
| 12:25.36 | jhiver | and this blinking 'cargando' is just doing my head in |
| 12:25.53 | Mother_ | it's a really shitty site, I must admit |
| 12:26.05 | Mother_ | they should at least offer a non-flash site |
| 12:26.32 | jhiver | and there's no prices on it |
| 12:26.40 | jhiver | and it's in god damn spanish :) |
| 12:26.49 | jhiver | fortunate i understand it a little |
| 12:27.29 | Mother_ | OK if you need help holler, I live some 10 minutes away from them |
| 12:27.33 | Mother_ | gotta go now |
| 12:32.27 | *** join/#asterisk dreamcode (~je@81.181.199.39) |
| 12:32.43 | Eight | ~seen eight |
| 12:33.42 | jbot | eight <~blake@12-205-155-39.client.mchsi.com> was last seen on IRC in channel #asterisk, 59s ago, saying: '~seen eight'. |
| 12:33.42 | Eight | oh well. |
| 12:35.01 | Weezey | anyone know if the channel admins use meetme() or conference() ? |
| 12:46.12 | shadebob | I have an E1 and 2 TE110P cards. How can i create 2 groupe (channel g1=1-12 g2=14-30) and work with it in my dialplan. g1 must be sent to IP phones and g2 tranfer on a PBX by the second TE11OP without be answering... |
| 12:50.47 | nope | If I got 2 X100P cards in my Asterisk server, is it possible to get all numbers that begins on 4 & 9 to be used on one specific card? |
| 12:50.58 | nope | (outgoing calls) |
| 12:51.58 | Jas_Williams | nope: yes |
| 12:52.06 | *** join/#asterisk Dobaj (~root@avonstreet.plus.com) |
| 12:52.10 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 12:52.24 | nope | sweet |
| 12:52.45 | nope | just crossing my fingers that the X100P card works here in Norway then |
| 12:52.47 | Jas_Williams | shadebob: It all depends how the calls are deliverd to you, in E1 pri calls are normailly delivered to all channells |
| 12:53.07 | *** join/#asterisk Dobaj (~root@avonstreet.plus.com) |
| 12:53.53 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 12:54.33 | Dobaj | ne1 know if Asterisk is now stable and fully working on Fedora Core 3? |
| 12:54.57 | Chuji | ? |
| 12:55.04 | Chuji | Was it not? |
| 12:55.28 | Dobaj | last time I tested it there was a problem due to the kernel being a newer version |
| 12:55.58 | Chuji | hmm, wasn't aware of that. However, yes, no problems with fc3 |
| 12:56.23 | Dobaj | Cool... it was about 5 months ago when I last used FC3 |
| 12:57.35 | shadebob | jas_williams : and how ca I separate call? By PRI channel or SDA number or anything else? |
| 13:01.06 | feral_kid | Ok, quick question... Does anyone have a voicemail tutorial or caommand sheet for VM2? I need to present some type of command sheet to go with this system I am passing to a client? |
| 13:02.29 | ariel_ | shadebob, most of the time calls over an pri lines you can route them via your inbound did number. |
| 13:03.02 | Zeeek | feral_kid there is a page about this somewhere |
| 13:03.14 | Jas_Williams | shadebob: The can be separated by dnis (incoming digits from pri) you would be able to split out different numbers and do different things in your extensions.conf |
| 13:03.17 | Zeeek | linked from wiki so look at all the wiki vm stuff and I bet you'll find it |
| 13:03.40 | *** join/#asterisk mesi (~player@dsl-082-083-052-192.arcor-ip.net) |
| 13:03.42 | *** join/#asterisk Darwin[laptop] (~Darwin@c-24-3-241-22.client.comcast.net) |
| 13:04.37 | feral_kid | Zeeek: I am looking over voip-info and haven't found it... If you know of some place, I would be greatly appreciative... |
| 13:04.44 | *** join/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 13:04.51 | Zeeek | I know there is one but I don't know where |
| 13:04.54 | MuppetMaster | Hello everyone. |
| 13:05.28 | Darwin[laptop] | use the search function and key words |
| 13:05.42 | MuppetMaster | Anyone here get the Asterisk CVS HEAD compiled on OSX per this?: http://www.voip-info.org/tiki-index.php?page=Asterisk%20MacOSX%20Support |
| 13:06.12 | MuppetMaster | I have the Bison upgraded, but during the compilation it appears that Asterisk is looking for the Linux Kernel source, but of course it is not present. |
| 13:06.49 | MuppetMaster | The minis are beautiful...I love my new Powerbook 17" though.... |
| 13:06.54 | Zeeek | feral_kid http://www.universaltime.org/~brad/vmail/ and http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf |
| 13:07.01 | shadebob | ariel_ , Jas_williams : someone have an extension.conf, zaptel and zapata.conf for a TE11OP? |
| 13:07.23 | *** join/#asterisk X-Gen (~x-gen@rrba-146-124-131.telkomadsl.co.za) |
| 13:07.42 | Darwin[laptop] | voice mail is easy to setup |
| 13:07.59 | Darwin[laptop] | just put the exten nmbr and passwd |
| 13:08.14 | feral_kid | Darwin[laptop]: Was that directed towards me? |
| 13:08.19 | Darwin[laptop] | the voicemail.cof explains it all |
| 13:08.38 | Jas_Williams | ~documentation |
| 13:08.51 | jbot | i heard documentation is not the issue for why the new stuff isn't completed |
| 13:08.58 | Jas_Williams | ~docs |
| 13:08.59 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 13:09.00 | *** join/#asterisk afe ([V9F41D+X2@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 13:09.53 | Jas_Williams | shadebob: Look at the links above lots of samples |
| 13:10.06 | shadebob | jax_williams... ok |
| 13:11.01 | feral_kid | Darwin[laptop]: I know how to setup vmail... What I need for a client of mine is a command sheet, so that he can look at this and see that one presses 4 to hear new messages, 5 to blah, etc. |
| 13:11.36 | Darwin[laptop] | it tell you that when you call it |
| 13:12.00 | shadebob | jas_williams : ok. maybe "exten => s/9184238080..." is my solution |
| 13:12.02 | X-Gen | Darwin[laptop]: but that is work, he wants u to type out the doc for him |
| 13:12.19 | feral_kid | Zeeek: Thanks for the pointer... That is good... |
| 13:12.19 | Jas_Williams | shadebob: No that will not work |
| 13:12.22 | Darwin[laptop] | if they dont listen and fallow it then that thier problem imho |
| 13:12.42 | ariel_ | feral_kid, there is no one sheet but you can make one out of this settings at the on the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20VoiceMailMain |
| 13:13.06 | MuppetMaster | So no one has experience on OSX? |
| 13:13.15 | shadebob | jas_williams : why? |
| 13:13.26 | mesi | iaxcomm says: |
| 13:13.33 | mesi | Ups, sorry. |
| 13:13.34 | mesi | ;-) |
| 13:13.41 | Darwin[laptop] | is the not a asterisk-osx channel ? |
| 13:13.51 | Darwin[laptop] | like the asterisk-bsd channel |
| 13:14.34 | Darwin[laptop] | mac/fbsd are not officaly supported |
| 13:14.47 | Darwin[laptop] | but we do help where we can |
| 13:14.55 | nope | Asterix@Home should work for a small company aswell, right? (maximum 5 phones internal, 2 lines external) |
| 13:14.55 | Darwin[laptop] | I do my half with fbsd |
| 13:15.01 | feral_kid | Darwin[laptop]: That is the difference in * being a hobby, and make it a viable revenue generator... Sometimes work is required... |
| 13:15.18 | Jas_Williams | shadebob: s/9184238080.... will only match an incoming CLI of 9184238080 |
| 13:15.26 | Darwin[laptop] | I put my tim ein on it |
| 13:15.33 | Darwin[laptop] | I keep it working on fbsd |
| 13:15.44 | MuppetMaster | On OSX, it is actually Zaptel that is having the problem. But since OSX does not support PCI cards (don't need them anyway) I don't actually need Zaptel. Compiling Asterisk itself now and that appears to be fine. |
| 13:15.46 | sambal | how big cpu / mem is required for a asterisk box doing only 1 sip to sip or iax connection? |
| 13:15.47 | *** part/#asterisk X-Gen (~x-gen@rrba-146-124-131.telkomadsl.co.za) |
| 13:16.16 | *** join/#asterisk aggelos (~aggelos@egate.eleven.de) |
| 13:16.19 | Darwin[laptop] | you need zaptel/ztdummy for timing for meetme |
| 13:16.21 | Zeeek | hehblithe blanket statement: 256M |
| 13:16.31 | feral_kid | ariel_: Yes, that is perfect... |
| 13:17.13 | MuppetMaster | Darwin: That is my concern... |
| 13:17.21 | shadebob | jas_williams : i known... my problem is when 022484810 or 022484811 or 022484812 have an incoming call then I send to ip phones... all over calls are tranfered to an another PBX by E1 link |
| 13:17.38 | shadebob | jas_williams : so s/number can work in my case? |
| 13:18.01 | Jas_Williams | shadebob: no s only matches no number |
| 13:18.22 | Jas_Williams | feral_kid: look at http://www.wheely-bin.co.uk/62pbx/ for voicemail key comands |
| 13:19.24 | MuppetMaster | Darwin: Well, my main Asterisk server runs on Suse v9.2, and the one on OSX will really be a local instance as my sandbox and interconnected to the 'master' version with DUNDi. |
| 13:19.32 | MuppetMaster | Darwin: So maybe I can get by without Zaptel drivers? |
| 13:20.23 | *** join/#asterisk Mother_ (~m@53.Red-217-126-93.pooles.rima-tde.net) |
| 13:20.25 | Mother_ | hi again |
| 13:20.38 | Jas_Williams | shadebob: Match the extensions you wish to send to sip, first then include a context called catch-all in [catch-all] have exten => _X.,1,Dial(ZAP/g2... where g2 is the e1 link to the PBX |
| 13:20.42 | Mother_ | I cannot figure out why these recordings come out like crap when played in windoze |
| 13:20.50 | *** join/#asterisk lidl (~little@213-140-6-96.fastres.net) |
| 13:21.12 | sambal | are there any ATA boxes which can be configured with a dialplan to different SIP ISP's? |
| 13:21.24 | Mother_ | I use Monitor to record a call, join the sides with wmix and all that is recorded sounds completely distorted |
| 13:21.43 | lidl | do you recommend to create a basic * configuration with the gui tools displayed on voip-info.org, and then adapt it by hand? |
| 13:21.54 | lidl | or is it better to start from scratch? |
| 13:22.16 | shadebob | jas_williams : ok. I will try. thanks |
| 13:22.43 | MuppetMaster | Well, the latest CVS HEAD compiles and runs fine on OSX, just w/o Zaptel... |
| 13:22.46 | Zeeek | lidl maybe start from published examples? |
| 13:24.08 | lidl | Zeeek, for example AMP seems a good tool |
| 13:24.24 | *** join/#asterisk MacDeath (david@196.22.239.13) |
| 13:24.48 | MacDeath | hi |
| 13:24.59 | Zeeek | I don't use any of those so I can't recommend them but you mearn more without those crutches |
| 13:25.07 | MacDeath | a while ago someone showed me how to debug a segmentation fault |
| 13:25.08 | Mother_ | the riff structure seems OK but the contents are crap |
| 13:25.25 | MuppetMaster | MacDeath: What are you looking to do? |
| 13:25.26 | MacDeath | im getting a seg fault every 30 seonds now |
| 13:25.39 | MacDeath | not trying anything specuial |
| 13:25.42 | MacDeath | 2 SIP phones |
| 13:25.46 | MuppetMaster | MacDeath: You need GNU Debugger installed and then run Asterisk in debug mode |
| 13:25.49 | MacDeath | no zaptel cards |
| 13:26.07 | MacDeath | i think i havea debugger installed |
| 13:26.12 | MacDeath | cause i did it once before |
| 13:26.30 | MacDeath | but i have no idea how |
| 13:26.59 | Jas_Williams | MacDeath: run "gdb /usr/sbin/asterisk <corefile>", hit |
| 13:26.59 | Jas_Williams | Enter a few times and run a backtrace using "bt" |
| 13:27.25 | MuppetMaster | do a bt and a bt full. |
| 13:28.40 | *** join/#asterisk Dirtysole (~Frank@cpe.atm0-0-0-1201091.0x503fc556.bynxx2.customer.tele.dk) |
| 13:28.52 | MacDeath | what will the corefile be |
| 13:29.05 | MuppetMaster | core.a_number |
| 13:29.14 | MuppetMaster | And will be in the directory where you ran Asterisk from. |
| 13:29.28 | MuppetMaster | In debug mode of course. |
| 13:30.34 | *** join/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
| 13:31.59 | *** join/#asterisk sysdef (~sysdef@sysdef.admin.debiancenter) |
| 13:33.09 | ctooley | Can someone give me a good hint on parsing Asterisk config files in Perl? Is there a Config::Asterisk module that will correctly parse them? |
| 13:33.43 | MuppetMaster | ctooley: Not that I am aware of, have not seen anything like that on the Wiki. |
| 13:34.29 | ctooley | MuppetMaster, So it's just read in and stop at the semicolon? |
| 13:34.39 | *** join/#asterisk mesi (~player@dsl-082-083-181-138.arcor-ip.net) |
| 13:35.05 | MuppetMaster | ctooley: There are no semicolons in the config files, unless you put them yourself at the end of every line. |
| 13:35.17 | ctooley | bah, that needs resolved then, that makes 2 things I have to write today. |
| 13:35.24 | ctooley | MuppetMaster, semi-colons are used for comments |
| 13:35.27 | MuppetMaster | ctooley: As those demarcate comments. |
| 13:35.49 | mesi | I use iaxcomm-lin-1.0rc2.tar. I can call it and talk in both directions, but outgoing calls are mute. What can I do? |
| 13:35.52 | ctooley | MuppetMaster, right, so I don't need things after semi-colons, they aren't a relevant part of the configuration. |
| 13:36.29 | ctooley | split (/;/, $config_line) |
| 13:36.31 | MuppetMaster | ctooley: Correct. And then you would just want to author your config files so that you put a semicolon at the end of each line. |
| 13:36.51 | ctooley | MuppetMaster, doesn't matter, if there's no ; it's still going to find $0 |
| 13:37.05 | ctooley | or whatever |
| 13:37.20 | MuppetMaster | ctooley: Correct. |
| 13:37.25 | Eight | feh, perl. |
| 13:37.35 | Eight | That's all I have to contribute =) |
| 13:37.40 | *** join/#asterisk NTJOCK (~brian@txshirts.com) |
| 13:37.43 | NTJOCK | good morning |
| 13:38.06 | Zeeek | Eight, change you pseudo to MrPython :) |
| 13:38.07 | NTJOCK | Anyone here have a Polcycom 500? |
| 13:38.23 | ctooley | Eight, If I write in Perl and someone gets upset enough to re-write it in C. I won't be upset, but I don't know C, so Perl it is. |
| 13:38.33 | Zeeek | now you done it |
| 13:38.35 | NTJOCK | I've got an issue where the sound on it is too low for me, but when I crank up the volume the speaker becomes too loud.... is there a adjustment somewhere on those things? |
| 13:39.04 | Eight | ctooley: lol. I have no problem with people using what they're familiar with. |
| 13:39.12 | *** join/#asterisk claint (~claint@195.174.24.202) |
| 13:39.15 | MuppetMaster | For what I do PHP is enough... |
| 13:39.21 | ctooley | And since app_directory has been completely useless for me (reading back the name one letter at a time? Really?) I'm writing my own. |
| 13:39.22 | Eight | ctooley: but when you ask Perl questions... I don't feel like helping =) |
| 13:39.58 | ctooley | Eight, Got the answer I was expecting, will solve it myself. |
| 13:40.25 | Eight | ctooley: So I saw =) |
| 13:40.53 | ctooley | Eight, :) It's the open source way isn't it? |
| 13:40.54 | *** part/#asterisk claint (~claint@195.174.24.202) |
| 13:46.57 | *** join/#asterisk claint (~claint@195.174.24.202) |
| 13:48.13 | *** join/#asterisk DEVILoper (~x@202.5.145.50) |
| 13:48.22 | DEVILoper | HI All |
| 13:48.25 | DEVILoper | undefined symbol: ast_unregister_modem_driver |
| 13:48.28 | *** part/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 13:48.38 | DEVILoper | any help y i am having this error ? |
| 13:48.51 | Dobaj | ne1 got an X100P compiled and working on FC3? |
| 13:48.51 | ctooley | DEVILoper, make clean && make && make install |
| 13:48.52 | shido6 | mornin |
| 13:48.57 | shido6 | dobaj |
| 13:49.01 | shido6 | yesh |
| 13:49.07 | shido6 | 2.6 |
| 13:49.10 | ctooley | running make clean before make solved that problem for me. |
| 13:49.23 | DEVILoper | hmm ok lemme try |
| 13:49.50 | claint | squeaky clean |
| 13:53.57 | lidl | are there know issues for running asterisk on amd64 for software only pbx? |
| 13:54.56 | DEVILoper | Ctooley it doesnt work in this way |
| 13:55.11 | DEVILoper | hmm seems some version conflict of drivers |
| 13:57.50 | ctooley | DEVILoper, dunno thne |
| 13:58.40 | *** join/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com) |
| 14:00.35 | *** join/#asterisk clive- (~pirch@rndf-146-28-208.telkomadsl.co.za) |
| 14:02.51 | *** part/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 14:02.57 | DEVILoper | hmm i've got the solution . You have to load chan_modem.so from /etc/asterisk/modules.conf |
| 14:03.27 | *** join/#asterisk zoa (~zoa@ip-212-239-162-26.dsl.scarlet.be) |
| 14:06.30 | lidl | if i trying to call * from k |
| 14:06.33 | lidl | erhm |
| 14:07.29 | lidl | if I call * from kphone on the same pc, is it normal i will hear garbage instead of demo-* gsm files? |
| 14:07.57 | lidl | i have disabled * alsa and oss channels |
| 14:08.11 | lidl | so the soundcard shouldn't be busy |
| 14:08.40 | lidl | the echo test works well (extension 600 from fresh * install) |
| 14:08.56 | lidl | so i guess there's a problem with the gsm codec |
| 14:09.16 | shido6 | ? |
| 14:09.33 | shido6 | start with ulaw |
| 14:09.35 | shido6 | and work out from there |
| 14:09.49 | *** join/#asterisk sudhir492 (~sudhir@wbar1.wdc2-4-8-141-004.wdc2.dsl-verizon.net) |
| 14:10.39 | wildgoose | how can I debug whether callerid is being received to the * box from my sipgate peer? I am not seeing the correct caller id on incoming to my budgetone and I'm not sure whether it's the phone or *...? |
| 14:10.43 | *** part/#asterisk sysdef (~sysdef@sysdef.admin.debiancenter) |
| 14:11.37 | lidl | i cannot still understand anything of the demo file (i set the codec to ulaw) |
| 14:12.22 | *** join/#asterisk elric (~kavit@ppp114-10.static.internode.on.net) |
| 14:13.42 | wildgoose | ctooley: The perl config modules for ini file style syntax should parse the asterisk config files |
| 14:14.39 | *** join/#asterisk pascals (~248d34d6@ip503c8584.speed.planet.nl) |
| 14:14.47 | *** join/#asterisk santiago (~santiago@201.245.167.104) |
| 14:16.24 | pascals | Can I use a single beronet BN4S0 quadbri card with the junghanns bristuff drivers to have two NT ports and two TE ports? |
| 14:16.59 | elriah | Anyone know of a faq that outlines the general terminology of asterisk? What is FXO, etc? |
| 14:17.39 | pascals | Go to http://voip-info.org/ |
| 14:17.47 | elriah | Thanks - |
| 14:19.16 | Eight | ~docs |
| 14:19.17 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 14:19.48 | elriah | Thanks, Eight. |
| 14:19.58 | ctooley | wildgoose, the problem is that there a significant amount of variance in the config files |
| 14:20.28 | dubphil | asking again for newcommers :) Anyone can point me in order to interface an alcatel 4400 PBX with Asterisk in order to have outgoing call on IP ? |
| 14:21.22 | *** join/#asterisk adjacent (~scott@office.bftwave.com) |
| 14:22.40 | shido6 | dubphil what kind of interfaces do you have on that PBX? |
| 14:23.10 | *** join/#asterisk stevekstevek (~stevekste@slim-eth0.horizonlive.net) |
| 14:23.24 | elric | asterisk documentation has a long way to go though. I might not be lazy one day and contribute. |
| 14:24.24 | Essobi | PSsh. |
| 14:24.33 | Essobi | Lack of documentation gives me job security. ;) |
| 14:24.42 | Dandan | job security? |
| 14:25.00 | Dandan | VoIP shaman? :D |
| 14:25.03 | Essobi | VOOODOOOO RUNNING FROM MY MAGIC! |
| 14:25.11 | Dandan | lol |
| 14:25.19 | Katty | hmm |
| 14:25.25 | Dandan | ? |
| 14:25.30 | elric | Essobi, haha true but if we are to spread we need better documentation. people will always need support |
| 14:25.34 | feral_kid | Anyone connected to FWD right now for a test call? |
| 14:25.41 | Essobi | dub Check www.voip-info.org... most stuff can be there. |
| 14:25.56 | Dandan | Essobi: most, but u gotta look really hard |
| 14:26.03 | Essobi | elric That was sarcasm. ;) |
| 14:26.13 | Dandan | and I have the only book about asterisk: VoIP telephony with Asterisk |
| 14:26.15 | Dandan | by Mahler |
| 14:26.16 | Essobi | Dandan Tain't hard to use that little google box on the side. :) |
| 14:26.29 | Dandan | and it is iffy and wattery at best |
| 14:26.29 | elric | Essobi, haha oops sorry too late to be up |
| 14:26.45 | Essobi | Outdated before it even hit the presses Ehh? |
| 14:26.54 | feral_kid | Dandan: I was thinking about that book, I take it isn't that good, huh? |
| 14:26.56 | Dandan | btw. anyone running * with IPv6? |
| 14:27.03 | Essobi | I get aggravated when "features" don't come to light until one RTFS. |
| 14:27.23 | Dandan | feral_kid: well, he is not 'to the point' - u gotta read the book to find all the pieces of the puzzle you want |
| 14:27.25 | Essobi | Dan I don't believe there's any IPV6 support yer. |
| 14:27.26 | Essobi | yet |
| 14:27.36 | Dandan | Essobi: just wondering :) |
| 14:27.39 | Essobi | unless you're tunneling 4 across 6 |
| 14:27.43 | Dandan | one day will start porting it to ipv6... |
| 14:27.54 | Dandan | no, i am talking native 6 |
| 14:28.00 | Essobi | :) |
| 14:28.02 | elric | anyway what codec do you guys reckon I should use for VoIP over the internet? bandwidth is not an issue, neither is licencing. |
| 14:28.03 | *** join/#asterisk eKo1 (~bernd@63.245.57.70) |
| 14:28.08 | Essobi | You one of those 6 nuts? |
| 14:28.25 | Dandan | Essobi: not nuts, but I like to be ahead of the times :) |
| 14:28.37 | Essobi | elric 711 is in my opinion the best, and always will be if bw is not a concern. |
| 14:28.44 | Weezey | elric: g726-16 seems to work pretty well. |
| 14:29.06 | elric | hrm I will play around a bit with those. |
| 14:29.26 | Essobi | Pssh. |
| 14:29.28 | Essobi | G711! |
| 14:29.32 | elric | i really like getting rid of the panasonic PBX we have at work right now. |
| 14:29.35 | pascals | Do I need to do something to the qozap driver to make it work correctly with a beronet card? |
| 14:30.00 | Essobi | The what what in the what what? |
| 14:30.03 | Weezey | essobi: 711 will always be the best, but if you want some compression... |
| 14:30.17 | pascals | The beronet driver just doesn't work for me :( |
| 14:30.28 | Essobi | Weezey Read his above statement. :) Bandwidth and money is not an issue. |
| 14:31.06 | Essobi | 711 can support faxes, full DTMF, and a wider audio range.. So why not under those conditions? |
| 14:31.15 | sivana | ~seen JerJer |
| 14:31.17 | jbot | jerjer <~JerJer@dsl-106-170.che.centurytel.net> was last seen on IRC in channel #asterisk, 14d 22h 12m 22s ago, saying: 'mrgoby: sure'. |
| 14:31.20 | *** join/#asterisk MikeJ_ (~icechat5@65.170.43.34) |
| 14:31.53 | elric | I am also surprised at the state of Zaptel driver support for FreeBSD. Asterisk runs like a dream on it. |
| 14:32.04 | Essobi | I need to thank Jer for all the H323 work he's done since I last checked.. |
| 14:32.14 | tzanger | don't do that |
| 14:32.15 | Essobi | elric State of? I havn't ran it but I heard it's there. |
| 14:32.18 | tzanger | it'll go straight to his head |
| 14:32.40 | Essobi | tz: Heh.. I know but I gave him a hard time last time I tried to do some stuff. ;) |
| 14:32.44 | elric | Essobi, yes, beta status and TE4XXP support is dodgy at best. |
| 14:32.55 | elric | i am using a TDM400P right now. |
| 14:33.09 | Essobi | elric: Really? The sangoma drivers are out for * and freebsd.. maybe they're better. |
| 14:33.11 | elric | might have to go with Sangoma |
| 14:33.14 | Essobi | :) |
| 14:33.25 | elric | yes Essobi but I wanted to support Digium |
| 14:33.26 | feral_kid | I wonder about the book from Cole, if that book is any good... |
| 14:33.36 | Essobi | I havn't heard of anyone running them together yet thou.. |
| 14:33.45 | feral_kid | Dandan: Know anything about that book? |
| 14:33.55 | Dandan | feral_kid: what? |
| 14:34.12 | Dandan | i read it |
| 14:34.13 | Essobi | tz: Yea.. last time I ran H323 it crashed my core router, and a 5000 customer softswitch. :) I had some choice words for jer that day. |
| 14:34.14 | Dandan | cover to cover |
| 14:34.45 | Dandan | and i definetely prefer oreilly :) |
| 14:34.52 | Dandan | i would grade it C+ |
| 14:34.57 | Dandan | for this book |
| 14:34.58 | feral_kid | Dandan: I take it you weren't moved by that one either... :) |
| 14:35.08 | Dandan | feral_kid: no, not really |
| 14:36.20 | Essobi | UNF. |
| 14:36.22 | feral_kid | Dandan: I doubt that there will ever be a timely book to keep up with the changes to *... Maybe need to put more time into working on the Asterisk Documentation development... |
| 14:36.47 | Essobi | My daemon is coming along nicely for only having 2 hours on it. :) |
| 14:36.49 | *** join/#asterisk bonez41 (~aint@c-67-166-77-14.client.comcast.net) |
| 14:37.05 | Essobi | feral_kid Books are about $$. |
| 14:37.18 | Dandan | feral_kid: true :) |
| 14:37.34 | Dandan | Essobi: I do not hesitate to spend $$$ on a good book |
| 14:37.39 | feral_kid | Dandan: Although, there is some fluff about an "advanced" book for Asterisk from Cole... Pricing it at $59.00 whenever it get's released... |
| 14:37.46 | Dandan | (Hacks series by e'reilly for example) |
| 14:38.07 | Dandan | feral_kid: we'll see :) prolly go out and buy it - company sponsoring |
| 14:38.07 | feral_kid | Dandan: I agree wholeheatedly... I am never slow in purchasing a good book... |
| 14:38.14 | zoa | afaik 2 new books will be published soon |
| 14:38.35 | Dandan | heh, any of them by o'reilly? :D |
| 14:38.37 | feral_kid | zoa: Oh, what is that? |
| 14:38.40 | Essobi | I've got a multi-threaded app that listens to the management port, and updates a SQL table. I only got a few events in it thus far.. I need to hack the rest in. :) |
| 14:39.02 | jontow | i'd like one as a documentation reference, a study of the source and how to proceed on ripping it apart mentally, with someone's crazy notes added :) |
| 14:39.04 | zoa | all 2 made by people very familiar with asterisk |
| 14:39.24 | *** join/#asterisk ws9455 (~ws9455@adsl-68-94-61-239.dsl.rcsntx.swbell.net) |
| 14:39.33 | Essobi | jontow Use the source, luke. It'll come to you. |
| 14:39.35 | zoa | cant tell anything more |
| 14:39.35 | elric | Essobi, what is it written in? |
| 14:39.41 | jontow | :) have been where possible |
| 14:39.57 | Essobi | jontow I was really surprised when I started looking at code, how easy it was to get your head around. |
| 14:40.12 | Essobi | Elric It's a sekret. :) |
| 14:40.47 | elric | heh Essobi i am meaning to write something similiar soon a few pointers would help |
| 14:40.51 | elric | :P |
| 14:41.00 | Essobi | :) |
| 14:41.01 | *** join/#asterisk cbachman (~chatzilla@129.105.7.250) |
| 14:41.27 | Essobi | I have to sign NDAs for most of my work.. if I told you, i'd have to kill you. |
| 14:41.31 | elric | ah |
| 14:41.35 | elric | ok you can try and kill me |
| 14:41.37 | `Sauron | Blah blah whatever, Essobi |
| 14:41.39 | Essobi | ;) |
| 14:41.39 | elric | :) |
| 14:41.45 | Essobi | `Sauron :P |
| 14:42.03 | elric | ok Essobi let me rephrase, if you had to write a manager app |
| 14:42.10 | `Sauron | Have you even ever signed an NDA? |
| 14:42.11 | `Sauron | :p |
| 14:42.11 | elric | what would your language of choice be |
| 14:42.18 | *** join/#asterisk MattB2 (~mattb@pcp01068561pcs.andrsn01.tn.comcast.net) |
| 14:42.22 | Essobi | `Sauron Yup. |
| 14:42.23 | MattB2 | hi all :) |
| 14:42.36 | `Sauron | Hum. I guess my #php assumption can't hold true here. |
| 14:42.46 | `Sauron | most of the kids in there, are <20 |
| 14:42.49 | Essobi | elric I'd probably start with a proof-of-concept in poe perl. |
| 14:42.51 | `Sauron | I figure that's not true here. |
| 14:43.00 | Essobi | Sauron Anewp. |
| 14:43.05 | Essobi | We're all old fogeys. |
| 14:43.06 | MattB2 | got a q about a proposed asterisk setup - are there any problems with running 4 ISDN voice lines into 4 PCI Fritz ISDN cards with asterisk? |
| 14:43.08 | `Sauron | poc in perl or python |
| 14:43.23 | Essobi | POE? |
| 14:43.26 | eKo1 | ruby... |
| 14:43.27 | Essobi | Oh POC.. |
| 14:43.29 | `Sauron | proof of concept |
| 14:43.30 | Essobi | EWWWW |
| 14:43.39 | Essobi | I meant POE when I said POE. |
| 14:43.41 | Essobi | :) |
| 14:43.52 | `Sauron | poe |
| 14:43.54 | `Sauron | poo |
| 14:43.55 | Essobi | I thought you typoed.. POE in python is called twisted |
| 14:44.00 | Dandan | POE(w): Piece Of Excellen (Wrk) ? :D |
| 14:44.06 | Dandan | *(Work) |
| 14:44.21 | Essobi | It's a multithreading library for perl |
| 14:44.29 | `Sauron | I figured that out. |
| 14:44.29 | Dandan | i know :) |
| 14:44.49 | Essobi | Prototyping with callbacks and threads is stupid easy then. |
| 14:45.28 | Essobi | So aaaaany ways. |
| 14:45.44 | elric | heh my boss wants a fancy GUI too. |
| 14:45.51 | jontow | eww, i just heard that we're gonna be doing a lot with MGCP |
| 14:45.55 | Dandan | elriah: mine too :) |
| 14:45.55 | `Sauron | tell him he can't have it |
| 14:45.56 | Essobi | I knew a guy from kentucky that used to go by Sauron. |
| 14:46.01 | jontow | arr, etc |
| 14:46.02 | Dandan | elric: mine too |
| 14:46.06 | `Sauron | not him |
| 14:46.19 | Essobi | Yea.. He wasn't a prick. ;) |
| 14:46.30 | elric | i could use php but I hate it |
| 14:46.43 | Essobi | He was ubergeeked however. So it was a possibility. ;) |
| 14:47.11 | Dandan | RoyK: I am running windows atm, u dummy! |
| 14:47.14 | elric | RoyK, fancy GUI will soon be E17 |
| 14:47.17 | `Sauron | Hum. It might've been me, but I never was from Kentucky |
| 14:47.25 | elric | it looks hot. |
| 14:47.29 | `Sauron | I used to be a real prick, then I got a life. |
| 14:47.42 | Dandan | jbot: :P |
| 14:47.43 | jbot | somebody said p was pee |
| 14:48.18 | Essobi | jbot ~seen Katty's ass, but I heard it's fine. |
| 14:48.35 | Essobi | Awww who broke jbot? :P |
| 14:48.41 | elric | if the manager application i write is half decent I will put it on sourceforge. |
| 14:48.44 | Essobi | ~seen Katty's ass, but I heard it's fine. |
| 14:48.45 | jbot | Essobi: i haven't seen 'katty's ass, but i heard it's fine.' |
| 14:48.49 | Essobi | Maha. |
| 14:48.49 | elric | and call is Obelix |
| 14:48.51 | `Sauron | ~seen Nugget |
| 14:48.52 | jbot | nugget is currently on #asterisk (3d 24m 39s). Has said a total of 97 messages. Is idling for 1d 7h 41m 40s |
| 14:48.58 | elric | s/is/it/ |
| 14:49.09 | `Sauron | Pfft. he must've had too much fun at SXSW yesterday |
| 14:49.17 | Essobi | Elric I'm working up one in php.. |
| 14:49.18 | elric | or dogmatix |
| 14:49.21 | Katty | Essobi: no one has seen my ass, thank you very much |
| 14:49.26 | Essobi | Katty WEEEE you're awake! |
| 14:49.27 | `Sauron | rofl |
| 14:49.33 | Nugget | Moo |
| 14:49.36 | Essobi | Jbot is a gurl, FYI. :) |
| 14:49.50 | Katty | moo, Nugget |
| 14:49.58 | `Sauron | wow |
| 14:50.01 | `Sauron | everybody's waking up |
| 14:50.03 | elric | Essobi, i cant stand php, i just finished making a 3 month project in it and it has given me so many bad programming habitd |
| 14:50.07 | elric | habits even |
| 14:50.14 | Nugget | php is to programming what mysql is to databases. |
| 14:50.24 | eKo1 | ha ha |
| 14:50.32 | eKo1 | Good analogy. |
| 14:50.53 | Nugget | give people php and you end up with stuff like <a href="http://slacker.com/photos/strange/redhat1"><img src="http://gallery.slacker.com/albums/strange/redhat1.thumb.jpg"></a> |
| 14:50.56 | Nugget | erp |
| 14:51.00 | Nugget | stuff like http://www.thephpwtf.com/node/47 |
| 14:51.09 | Essobi | I have no problems with LAMP.. PHP however has it's place.. albeit a very small niche in web programming. |
| 14:51.13 | Nugget | although, technicall, the first url works too. :) |
| 14:51.31 | elric | i agree Essobi |
| 14:51.47 | elric | its good for spitting out websites fast |
| 14:51.52 | `Sauron | Nugget: That's ignorance, not php's fault. |
| 14:51.54 | Essobi | Socket programming.. daemons, threads.. No freeeking way in php. |
| 14:51.55 | elric | without too much effort |
| 14:51.58 | Nugget | you mean [LFON]A[MP][PR]? |
| 14:52.10 | *** join/#asterisk TheBear (~brif8@lazyjtrainingcenter.com) |
| 14:52.21 | `Sauron | Essobi: Sadly, I've seen it done. |
| 14:52.25 | elric | Essobi, althought www.m0n0.ch have made an init system in php |
| 14:52.26 | `Sauron | Well, sans the threads part. |
| 14:52.31 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 14:52.32 | elric | heh sheer bravery |
| 14:53.18 | Nugget | `Sauron: of course it's ignorance, but you get that when you provide inexperienced developers with a language which has inconsistent and poorly-thought-out design |
| 14:53.41 | *** join/#asterisk jmacz (~jmacz@63.245.86.116) |
| 14:53.44 | Nugget | in pretty much the exact same way that mysql has led to a legion of ignorant database developers |
| 14:53.56 | *** join/#asterisk maruz (~maumar@adsl-123-3.38-151.net24.it) |
| 14:54.06 | *** join/#asterisk wizhippo (~wizhippo@Quebec-HSE-ppp233869.qc.sympatico.ca) |
| 14:54.10 | eKo1 | and php + mysql has lead to the myriad of slow shitty websites. |
| 14:54.11 | Nugget | "mysql is great for quickly churning out websites! what's a nested select?" |
| 14:54.24 | `Sauron | hehe |
| 14:54.25 | elric | or a foreign key |
| 14:54.28 | maruz | # host cvs.digium.com |
| 14:54.28 | maruz | ;; connection timed out; no servers could be reached |
| 14:54.31 | maruz | wtf?? |
| 14:54.40 | *** join/#asterisk acek (~acek@p50870B86.dip0.t-ipconnect.de) |
| 14:54.41 | Nugget | "php is great for quickly churning out websites! what's this about safe variables?" |
| 14:54.47 | `Sauron | the only reason I still use php, is because I had a hard time migrating off the old version of aolserver I was running |
| 14:54.55 | `Sauron | learning tcl was the easy part |
| 14:55.11 | eKo1 | So if not php, what would you use? |
| 14:55.15 | elric | i am going to look at python and ruby |
| 14:55.18 | pascals | Tcl :) |
| 14:55.30 | eKo1 | I heard Ruby on Rails is pretty good. |
| 14:55.32 | `Sauron | That, and I really don't want to re-write my blog yet again... :) |
| 14:55.36 | pascals | What migration problems? |
| 14:55.37 | Nugget | ruby on rails is slick |
| 14:56.06 | `Sauron | pascals: I was going to use aolserver+openacs - but the new versions of openacs have... issues. |
| 14:56.14 | pascals | Indeed. |
| 14:56.29 | eKo1 | Anybody tried using Apple's WebObjects? |
| 14:56.40 | TheBear | is pastebin googled ? |
| 14:56.47 | pascals | But just using the database api from Openacs 3.x and Aolserver 4 works great for me |
| 14:56.47 | `Sauron | But I really like the speed of aolserver |
| 14:56.50 | `Sauron | Mmmm. |
| 14:57.04 | `Sauron | Maybe I'll end up doing that. |
| 14:57.09 | *** join/#asterisk spackle (~spackle@209.234.83.19) |
| 14:57.18 | `Sauron | I was also bummed that you couldn't do OO stuff |
| 14:57.25 | pascals | Just rip out all modules, and you only need a few small fixes to get it running on Postgres 7.4/8.0 |
| 14:57.27 | MattB2 | anyone using multiple ISDN lines with asterisk? does it work well or is it best to use multiple analog lines? |
| 14:57.38 | pascals | OO stuff in Aolserver 4 is simple. |
| 14:57.44 | spackle | 'Sauron, how is the gumstix going? |
| 14:57.55 | pascals | MattB2: fighting with it at the moment. |
| 14:57.59 | `Sauron | spackle: Great - once I find out what I did to my CF wifi card. |
| 14:58.11 | `Sauron | pascals: OO stuff in tcl, or what? |
| 14:58.24 | pascals | Actually - it really works, the trouble I have is connecting the old ISDN pbx. |
| 14:58.27 | MattB2 | pascals: what are the problems? tryin to decide whether to risk 4 isdn lines or stick with 4 analog |
| 14:58.28 | spackle | 'Sauron, accidental firmware "upgrade" |
| 14:58.38 | `Sauron | I have a bunch of OO-ified template parsers, etc - that I'd want to port over |
| 14:58.39 | santiago | hi, does anyone knows which is the default value for "jitterbuffer" in iax.conf ? |
| 14:58.44 | pascals | `Sauron: OO is simple in Aolserver 4 - just package require Itcl |
| 14:58.56 | `Sauron | ooooh |
| 14:58.58 | `Sauron | Mmm. |
| 14:58.59 | *** join/#asterisk mogorman (~mogorman@dhcp-162.digium.com) |
| 14:59.09 | pascals | Make sure it happens *really* early. |
| 14:59.41 | `Sauron | spackle: Dunno. Read a few threads talking about how it doesn't like promisc mode |
| 14:59.46 | *** join/#asterisk Shido (~greg@d57-87-253.home.cgocable.net) |
| 14:59.52 | `Sauron | You have to do some fiddling around in winblows to get it working again |
| 14:59.54 | `Sauron | Grr. |
| 15:00.09 | pascals | `Sauron: to be sure - ask Dossy or someone else over on #aolserver |
| 15:00.37 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 15:00.37 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 15:00.49 | Dirtysole | Has anyone ever tried setting Asterisk voicemail up to use an external mailserver instead of local sendmail??? I have to do that with our setup here and I really can't figure out how to do it :-( |
| 15:01.00 | pascals | MattB2: My problems are probably related to the sucky PBX |
| 15:01.13 | santiago | Dirtysole, yes, it works |
| 15:01.25 | `Sauron | Oh how fun |
| 15:01.32 | Shido | Dirtysole |
| 15:01.36 | Shido | its not that hard |
| 15:01.38 | `Sauron | I love how you have to download all the ns_ stuff seperately and compile |
| 15:01.40 | `Sauron | dum di dum |
| 15:01.46 | ^Fenris | how do I get the latest broadvoice patch? |
| 15:01.53 | wizhippo | anyone have problems with voicemail messages being cut short? |
| 15:02.13 | pascals | MattB2: Voip, sip and iax, to and from isdn is working like a charm. Buy the Junghanns.net card and use the bristuff driver |
| 15:02.17 | `Sauron | Fenris: grab cvs-head? :) |
| 15:02.26 | Dirtysole | Santiago: Okay, howtodo? Is there a fake sendmail client or something I can use? Or is it an undocumentet future? |
| 15:02.57 | ^Fenris | `Sauron: only thing I've done with CVS before is make update |
| 15:04.03 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
| 15:04.03 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:04.09 | Dirtysole | Shido: Please share. Don't keep me in suspenders ;) |
| 15:04.10 | pascals | `Sauron: again - abuse the openacs supplied aolserver distribution ;-) |
| 15:04.30 | `Sauron | I will. |
| 15:04.43 | `Sauron | Fenris: Maybe it's time you try... :) |
| 15:05.08 | `Sauron | 4.0.10 is tiny |
| 15:05.11 | `Sauron | I thought it was bigger |
| 15:05.13 | `Sauron | hum di dum |
| 15:05.18 | *** part/#asterisk wizhippo (~wizhippo@Quebec-HSE-ppp233869.qc.sympatico.ca) |
| 15:05.29 | spackle | dirtysole: do you have to do other email on the asterisk box, or is voicemail/email all you are doing? |
| 15:06.05 | pascals | Aolserver 4.0 links against the local Tcl version, they do not supply it anymore. |
| 15:06.22 | pascals | `Sauron: make sure you have 8.4.5+, ISTR |
| 15:06.29 | `Sauron | I just found that out |
| 15:06.31 | `Sauron | Ugh. |
| 15:06.59 | pascals | `Sauron: better go to the #aolserver channel or the #tcl channel... |
| 15:07.04 | `Sauron | nod |
| 15:07.33 | Dirtysole | Spackle: I have to do other email. But the actual problem is that my Asterisk has to run inside a V(irtual) server where there simply is no sendmail. I can only access one by network. |
| 15:07.56 | *** join/#asterisk angler_ (~angler@suid.digium.com) |
| 15:08.09 | *** join/#asterisk jhiver (~jhiver@ABesancon-102-1-2-188.w80-11.abo.wanadoo.fr) |
| 15:08.12 | *** part/#asterisk maruz (~maumar@adsl-123-3.38-151.net24.it) |
| 15:08.29 | jhiver | hey guys can somebody remind me how to get help on a given app. from the CLI? |
| 15:08.34 | spackle | Dirtysole: so can you send all mail from asterisk to a smarthost? |
| 15:09.13 | afe | jhiver: show application monitor (for example) |
| 15:09.27 | jhiver | cheers! |
| 15:11.13 | *** join/#asterisk heison (~heison@ns.somanetworks.com) |
| 15:11.17 | Dirtysole | Spackle: I have to, but I don't have access to sendmail to do it. So I have to find a way to make voicemail deliver voicemail-emails directly to a network smtp port. |
| 15:11.58 | tzafrir | Dirtysole, so install ssmtp or whatever as /usr/sbin/sendmail |
| 15:12.23 | *** join/#asterisk didz_ (didz_@200.218.192.52) |
| 15:12.24 | tzafrir | ssmtp, nullmailer, esmtp, mstmtp, and there are a couple more. |
| 15:13.01 | tzafrir | s/mstsmtp/msmtp/ |
| 15:13.20 | Dirtysole | tzafrir: Thanks, I think you have just said the words I was looking for there. ;) |
| 15:14.21 | *** join/#asterisk adjacent_ (~scott@office.bftwave.com) |
| 15:16.50 | jhiver | I wonder if there's a way do tell * to do something every X seconds upon connection of a call, it would be useful for pre-pay type situations |
| 15:17.41 | *** join/#asterisk Hmmhesays (negative3k@66.173.103.108) |
| 15:17.43 | pascals | jhiver: just start an application which does that for you |
| 15:17.57 | jhiver | I'm not sure to follow you... |
| 15:18.08 | tzafrir | pascals, but you're inside Dial, right? |
| 15:19.58 | Katty | hmm |
| 15:20.33 | Katty | ooooh |
| 15:20.35 | pascals | tzafrir: do it right before the Dial command |
| 15:20.38 | Katty | :>>>> |
| 15:20.52 | pascals | Just like Background() in a voice menu |
| 15:20.57 | Zeeek | Katty what is your interest in asterisk (if any?) |
| 15:21.29 | Katty | Zeeek: i set it up for the office...we'll probably resell it too other offices |
| 15:21.30 | jhiver | pascals: how would you run an app in the background doing this? can u show a generic example? |
| 15:21.34 | pascals | tzafrir: haven't tried it - yet. I need something similar here. |
| 15:22.04 | pascals | jhiver: sorry, not yet |
| 15:22.11 | Zeeek | Katty so, fun & profit |
| 15:22.25 | `Sauron | grr |
| 15:22.27 | Katty | Zeeek: fun or me...profit for the company |
| 15:22.32 | jhiver | I've seen this option for Dial: M(x) which looks kinda interesting |
| 15:22.40 | pascals | But I your best guess is looking at the code for Background |
| 15:22.45 | `Sauron | pascals: Now I remember why I quit doing the aolserver upgrade dance |
| 15:22.45 | Zeeek | jhiver you want to check and see if the caller still has credit? |
| 15:23.06 | pascals | `Sauron: what platform/os are you on? |
| 15:23.07 | jhiver | Yeah something like that... |
| 15:23.07 | `Sauron | when they quit including it, the distribution-installed versions never were compiled correctly |
| 15:23.11 | Katty | Zeeek: does it really matter? |
| 15:23.19 | Zeeek | jhiver someone must have done this in prepaid stuff |
| 15:23.30 | pascals | `Sauron: you need tcl compiled with threads enabled. |
| 15:23.37 | `Sauron | bingo |
| 15:23.41 | Zeeek | Katty - on the cosmic scale, nothing matters. Ever read any Tom Robbins? |
| 15:23.45 | `Sauron | the version installed isn't threaded |
| 15:24.00 | Katty | Zeeek: hmm, no..doesn't sound framilier. i prefer piers anthongy |
| 15:24.02 | jhiver | yeah I guess I should go and sneak on other's code, but I thought maybe somebody had the answer here :) |
| 15:24.03 | `Sauron | but I don't feel like recompiling it just to get aolserver working |
| 15:24.04 | `Sauron | blah |
| 15:24.06 | Katty | anthony |
| 15:24.06 | zoa | hey on von europe |
| 15:24.13 | Katty | xanth++ |
| 15:24.15 | zoa | the director of Asterisk = Olle |
| 15:24.16 | zoa | :) |
| 15:24.21 | zoa | as stated in their brochure |
| 15:24.26 | Zeeek | Katty did you read Procto plus about the dentist that ios kidnapped by aliens? |
| 15:24.38 | Katty | Zeeek: nope |
| 15:24.41 | DEVILoper | would any one plz tell me wat will be the dial string if using Aopen as a modem |
| 15:24.44 | Zeeek | Piers Anthony |
| 15:25.23 | Zeeek | "Being a Green Mother" |
| 15:25.48 | tzanger | you're a mother something, alright Zeeek |
| 15:25.59 | Katty | Zeeek: i like the xanth novels |
| 15:26.05 | Zeeek | the peanut gallery has awakened.... :) |
| 15:26.06 | tzanger | piers anthony? eek |
| 15:26.10 | Katty | Zeeek: 'procto plus' does not sound like a xanth novel |
| 15:26.20 | tzanger | Katty: no, it sounds like a rectal cream |
| 15:26.25 | Zeeek | no it's a comedy series of short stories |
| 15:26.31 | Hmmhesays | it sounds like an ointment for your hind end |
| 15:27.03 | Katty | Hmmhesays (= |
| 15:27.14 | Hmmhesays | haha tzanger I didn't even read that before I typed mine out |
| 15:27.31 | Zeeek | errr, yes it was "Prostho Plus", sorry for the confusion |
| 15:27.47 | tessier | Innuendo: The new Italian suppository |
| 15:27.52 | Zeeek | The Procto is the guy that "looks up all hist old friends" |
| 15:27.58 | epoch | you guys talking about anal prosthetics or something? |
| 15:27.58 | Katty | mmm, innuendo |
| 15:28.12 | Hmmhesays | :) |
| 15:28.17 | *** join/#asterisk TheBear (~brif8@lazyjtrainingcenter.com) |
| 15:28.20 | tessier | anal prosthetics? For war vets who got their asses shot off? |
| 15:28.34 | epoch | well sure |
| 15:28.38 | epoch | everybody needs to shit! |
| 15:29.16 | Hmmhesays | lol.. this conversation is going down hill in a hurry |
| 15:29.19 | Weezey | I have a regular phone line, and when someone calls in and then hangs up, during voicemail, no problem. I have another line which comes in via a Norstar ATA and when users hang up, it just keeps on recording. It's like the SPA-3000 doesn't recognize that the call's been hung up. What setting should I adjust to get the call to recognize that it's been hung up? |
| 15:29.35 | `Sauron | Hmm, so he said. |
| 15:29.36 | `Sauron | ;) |
| 15:29.44 | Katty | Hmmhesays: everything goes downhill in a hurry |
| 15:29.56 | lidl | ~docs |
| 15:29.58 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 15:30.04 | `Sauron | Depending on the hill, your acceleration is 9.81m/s/s |
| 15:30.10 | TheBear | I have my incoming line connected to my X100p I have a std. phone connected to TDM10B. On the std phone there is a bad echo, like you are talking underwater. the same in on the SIP (snom200) phones but not as noticable, where could my mistake be ? |
| 15:30.27 | Hmmhesays | Katty: most of the time yeah <chuckle> |
| 15:31.10 | Katty | speaking of downhill, this sure is a crappy monday |
| 15:31.18 | MattB2 | pascals: sorry, just been on a call... so you think that running 4 ISDN lines into PCI Fritz cards is a viable setup? |
| 15:31.30 | `Sauron | .. what Katty said. |
| 15:31.38 | Hmmhesays | well besides monday's being inherently crappy its not so bad |
| 15:31.44 | Katty | for you |
| 15:31.45 | Zeeek | Katty if you only knew! |
| 15:31.55 | Katty | you're not the one who's running a fever and sick of being in bed :P |
| 15:32.13 | Katty | to quote my mother's bird: OUTTTTT!!!! |
| 15:32.14 | Zeeek | I have flu symptoms but am not in bed |
| 15:32.15 | Hmmhesays | no, but I got problems of my own... being in bed sounds nice right about now |
| 15:32.18 | TheBear | http://pastebin.ca/7411 holds my zapata.conf file, which has echocanel = yes ? |
| 15:32.24 | *** join/#asterisk Grooby (~Grooby@66.160.105.186) |
| 15:32.25 | zoa | Zeeek, did i give that to you ? |
| 15:32.32 | Zeeek | prolly |
| 15:32.36 | Katty | Hmmhesays: :< |
| 15:32.40 | `Sauron | anything to do that doesn't involve typing, would be nice |
| 15:32.42 | Katty | Hmmhesays: comenap |
| 15:32.42 | Zeeek | that and the substance I shot up last night |
| 15:33.01 | Hmmhesays | Hmm it's a long drive... i might make it some time middle day tomorrow |
| 15:33.06 | lordcian | anyone clued into the cause of this error with asterisk and AMP? |
| 15:33.07 | lordcian | Warning: main(DB.php): failed to open stream: No such file or directory in /var/www/html/admin/common/db_connect.php on line 14 |
| 15:33.07 | lordcian | <PROTECTED> |
| 15:33.07 | lordcian | <PROTECTED> |
| 15:33.22 | Hmmhesays | lordcian: that one is pretty self explanatory |
| 15:33.24 | Katty | Hmmhesays: k |
| 15:33.31 | *** join/#asterisk Corydon-w (white@vcchgate.vcch01.springfield.tn.us.vcch.net) |
| 15:33.39 | lordcian | well, yes, but not sure how db.php gets generated? |
| 15:33.52 | `Sauron | DB.php doesnt exist where /var/www/html/admin/common/db_connect.php thinks it is |
| 15:33.54 | Hmmhesays | you need to have pear installed with the DB module.. or whatever they call it |
| 15:34.17 | lordcian | ah, what, exactly is PEAR. Therin lies my problem? |
| 15:34.24 | Hmmhesays | google |
| 15:34.32 | lordcian | ok, yes, thianks |
| 15:35.06 | Hmmhesays | pear is alright if you want to connect to a lot of different databases without changing your code |
| 15:35.11 | lordcian | rpm is installed already |
| 15:35.35 | Hmmhesays | google you're problem |
| 15:35.38 | *** part/#asterisk DEVILoper (~x@202.5.145.50) |
| 15:35.41 | Hmmhesays | or go to the appropriate irc channel |
| 15:35.54 | Hmmhesays | #php is active |
| 15:36.03 | lordcian | thanks, i'll stop bugging.... |
| 15:36.15 | *** join/#asterisk JerJer[mobile] (~jj@feth100-fw.fament.net) |
| 15:36.19 | Hmmhesays | haha that came off kind of short |
| 15:36.21 | Zeeek | this is more an issue you should be able to find in AMP FAQ or forum if such things exist |
| 15:36.40 | lordcian | yeah, forum is quiet this early....... |
| 15:36.46 | Zeeek | one guess, you were supposed to furnish the DB.php file and didn't |
| 15:36.56 | lordcian | yeah, its not in install faq |
| 15:36.59 | Hmmhesays | DB.php comes with the db part of the pear install |
| 15:37.02 | feral_kid | Has anyone played with Flash Operator Panel? |
| 15:37.06 | Zeeek | aha |
| 15:37.09 | Hmmhesays | feral_kid: yes |
| 15:37.16 | Zeeek | the FOP ? |
| 15:37.21 | Hmmhesays | but not in about 2 weeks |
| 15:37.28 | feral_kid | Zeeek: That is one... :) |
| 15:38.01 | Zeeek | what is AMP, a gui addon? |
| 15:38.39 | feral_kid | Zeeek: Yes a GUI manager... |
| 15:38.51 | Hmmhesays | gui is a fun word |
| 15:39.21 | TheBear | I have my incoming line connected to my X100p I have a std. phone connected to TDM10B. On the std phone there is a bad echo, like you are talking underwater. the same in on the SIP (snom200) phones but not as noticable, where could my mistake be ? http://pastebin.ca/7412 holds my zapata.conf file, which has echocanel = yes ? |
| 15:39.30 | tessier | gooey gooey |
| 15:40.09 | Zeeek | nothing to do with your problem TheBear but you have a typo on callerid after the first channel decl |
| 15:40.29 | TheBear | Zeeek: thanks I'll change it anyway |
| 15:40.35 | Zeeek | line 25 usercalledid |
| 15:40.50 | Zeeek | unless you meant Hot DID! in Spanish? |
| 15:43.19 | Eight | huh, does anyone know if Speakeasy's VoIP plays nice with Asterisk? |
| 15:43.47 | *** join/#asterisk CosmicRay (~jgoerzen@2002:4463:7269:1:20e:a6ff:fe66:c5a3) |
| 15:43.56 | *** join/#asterisk n3tar (~geno@201.254.96.181) |
| 15:43.59 | n3tar | hi |
| 15:44.14 | TheBear | Zeeek: no I meant callerid, I will change it, any ideas on the echo ? |
| 15:44.52 | Zeeek | only one: tons of pages have been written on the subject in the mailing list. That's where I'd start |
| 15:45.25 | TheBear | Zeeek: I have started searching the mailing list. I can't post because my email address isn't recognized. |
| 15:45.32 | Eight | ah, it's only for their DSL lines. |
| 15:45.34 | Zeeek | don't post, just search |
| 15:45.58 | TheBear | yeah, but I would also like to be able to post. |
| 15:46.00 | Zeeek | TheBear did you look at this? http://www.voip-info.org/wiki-Asterisk+Echo+Cancellation |
| 15:46.43 | Zeeek | One thing that creates echo a lot os really loud phones |
| 15:46.52 | Zeeek | (or voices) |
| 15:47.29 | Eight | lol. |
| 15:47.36 | Eight | the only thing Speakeasy offers to me is a T1. |
| 15:47.48 | TheBear | thanks it is 2-wire to the phones |
| 15:48.27 | *** join/#asterisk Skysky (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
| 15:48.57 | ctooley | Anyone else using Realtime configuration that _isn't_ MySQL? |
| 15:49.13 | Skysky | hi, I wanna ask for SetGroup command. I would like to limit the # of channels allow for both incoming and outgoing calls. is it still possible to do that? |
| 15:49.13 | Zeeek | If you post on the mailing list asking about echo, you will get "this has been discussed many times, blah, blah" |
| 15:50.09 | Nugget | http://www.livejournal.com/users/fantasygoat/44088.html <-- apple pi(e) |
| 15:50.54 | Hmmhesays | which is why there are mailing list archives |
| 15:51.03 | Zeeek | nice, Nugget: http://www.syslog.com/~jwilson/hotdogs/HotDog83.jpg |
| 15:51.33 | Hmmhesays | that's a stomach ache in a bun |
| 15:51.49 | tzanger | Zeeek: ha |
| 15:51.54 | *** join/#asterisk Moc____ (~mochouina@h66-201-214-109.gtconnect.net) |
| 15:51.56 | tzanger | I prefer the pic of the goat going "w00t!" |
| 15:51.59 | Essobi | Baaah. |
| 15:52.05 | Essobi | Damn frame bugs. |
| 15:52.28 | tessier | Why doesn't asterisk match longest string first when looking for an extension in any given context? |
| 15:52.41 | tessier | That is quite annoying and makes it harder to work out a dialplan to do some basic LCR. |
| 15:52.57 | Hmmhesays | harder but not impossible |
| 15:53.02 | Essobi | tessier It matches in the order of the context.. Move it. |
| 15:53.03 | tessier | I'll be coding all of this up in SER soon anyhow so I guess it doesn't really matter but it sure would be nice for those who want a simple phone system with basic LCR |
| 15:53.13 | tessier | Essobi: Yeah, I know. But that's a nasty kludge. |
| 15:53.18 | Hmmhesays | actually it's not even hard |
| 15:54.04 | tessier | That means I have to put _01184. in a separate context all by itself and include it before _011. just because asterisk is missing this feature. |
| 15:54.42 | john867530 | I am working on some new meetme code if anyone has any insight on how I can mute users just by setting a confflag let me know. |
| 15:54.46 | Hmmhesays | or you could just put it in a *sql table and write a query |
| 15:55.16 | Hmmhesays | or use mysql in the dialplan and use a query to find the longest match |
| 15:55.18 | tessier | Man, I just saw the biggest mosquito...I better hunt him down and smash him before I go to bed. |
| 15:55.28 | tzanger | tessier: why you might lose 30lbs overnight! |
| 15:55.52 | tessier | That wouldn't be so bad actually but the bites are annoying. |
| 15:55.56 | Hmmhesays | gotta love those deadly virii.... great for your figure |
| 15:55.59 | tessier | They've got some big ones here in Saigon. |
| 15:56.21 | TheBear | anyone on the list.digium.com and/or gnophone.com admin team/staff. I have send submission requests and got no emails back on approval :( ? |
| 15:56.29 | Hmmhesays | Saigon HA try north dakota buddy, it's our state bird |
| 15:56.32 | john867530 | I use an SQL table now for example if I want to kick a user I use -2 in the conflags on in the table. and I call user->adminflags=ADMINFLAG_KICKME; but Mute/unmute is not the same |
| 15:57.05 | tessier | Hmmhesays: Our mosquitos here feed year round since it is always earm and wet. North Dakota only has them during a short part of the year. |
| 15:57.30 | Hmmhesays | short part, where they for some reason or another get huge |
| 15:57.32 | Essobi | Anyone running an H323 slowstart router/softswitch with *? I found a bug I think is related to slow start.. |
| 15:57.39 | tzanger | Hmmhesays: that's nothing up here in northern ontario they can carry off small children |
| 15:58.00 | Hmmhesays | lol, it's like there is some radiation mutating them |
| 15:58.02 | Necko | Hi ,i've installed Asterisk from binary files(using apt-get in debian-just for overviewing the system),does it have a web configuration interface? |
| 15:58.03 | *** join/#asterisk n3tar (~geno@201.254.92.137) |
| 15:58.13 | Hmmhesays | ~docs |
| 15:58.14 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 15:58.19 | Essobi | Necko nope. |
| 15:58.34 | Essobi | Go read as jbot suggests if you wish to pursue it. |
| 15:58.45 | Necko | hmm weird..because i so something called AMP.. |
| 15:58.52 | Hmmhesays | ~docs |
| 15:58.53 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 15:58.57 | Necko | i thought its binded into the binary packages... |
| 15:59.02 | Hmmhesays | ok i'll stop now |
| 15:59.03 | john867530 | does anyone have any ideas about my Mute/Unmute issue or some flag I can set? |
| 15:59.23 | Necko | take a look at this: http://techdatapros.com/asterisk/extensions.jpg |
| 15:59.57 | tessier | Essobi: I have always wondered: What does slow/fast start mean |
| 15:59.58 | tessier | ? |
| 16:00.44 | *** join/#asterisk Frantic (~ab@TechnologicPartners35.dsl.concentric.net) |
| 16:00.57 | Essobi | tessier Umm. well the media pairs and codec are not defined until after/just-in-time the call is past the call proceeding state if I remember correctly |
| 16:01.15 | Essobi | negotiation occurs after the phone starts ringing.. |
| 16:01.34 | Essobi | ~seen jerjer[mobile] |
| 16:01.36 | jbot | jerjer[mobile] is currently on #asterisk (25m 21s) |
| 16:01.53 | Essobi | Hey JerJer[mobile] ... Want to lookie lookie? |
| 16:02.06 | Necko | can asterisk perform as an h323 GK? |
| 16:02.27 | Essobi | It only occurs with a queue.. DTMF options out of the queue.. |
| 16:02.29 | Essobi | necho No. |
| 16:02.37 | Essobi | only a gateway |
| 16:02.40 | Necko | ohh i see... |
| 16:02.40 | Essobi | it's not a GK |
| 16:02.46 | Necko | yeah it's a pbx |
| 16:02.52 | Necko | but it has some voip features.. |
| 16:02.58 | Necko | so i thought that maybe.. |
| 16:02.59 | Essobi | lol |
| 16:03.10 | Essobi | I'd say it's a voip with some PBX features myself. |
| 16:03.21 | Necko | but it can perform as a voice gw right? |
| 16:03.43 | Essobi | Yup |
| 16:03.54 | Essobi | http://www.voip-info.org |
| 16:03.56 | Necko | i mean...calling from the internet to the Asterisk box and the asterisk route it thorugh the PSTN righ? |
| 16:04.11 | Essobi | Dude.. Go read up. |
| 16:04.12 | tzanger | Necko: that's possible |
| 16:04.19 | Essobi | :) |
| 16:04.23 | Necko | another little silly question.....does h323 has to have a GK? |
| 16:04.31 | Essobi | No. |
| 16:04.46 | Necko | ok:) |
| 16:04.47 | Hmmhesays | negative |
| 16:05.04 | Hmmhesays | www.protocols.com |
| 16:05.10 | Essobi | GKs are used for centralized call distribution to other gateways where the numbers on the gateways rehgister with the GK. |
| 16:05.13 | Hmmhesays | go there... bath in the knowledge |
| 16:05.20 | tzanger | haha |
| 16:05.21 | Necko | so...who controls the calls? |
| 16:05.37 | Necko | i mean ,who decides if the call can be made or not? |
| 16:05.48 | Hmmhesays | in a gateway setup? |
| 16:05.50 | Necko | i mean,can pass thorugh to the wan link ...for example.. |
| 16:05.51 | Necko | yeah.. |
| 16:05.51 | Hmmhesays | your gateways |
| 16:06.02 | Necko | i mean what protocol ...or what componenet.. |
| 16:06.08 | Necko | component* |
| 16:06.14 | Hmmhesays | most gateways have rudimentary routing capabilities |
| 16:06.32 | Essobi | Via a dialplan |
| 16:06.43 | Necko | ok...right..:0 dial plan |
| 16:06.48 | Essobi | 555-555-12XX lands on 192.168.0.1 |
| 16:06.57 | Essobi | 555-555-13XX lands on 192.168.0.2 |
| 16:07.01 | Essobi | etc etc.. |
| 16:07.02 | MikeJ_ | can you use a global variable in the extension portion of a exten => statement in extensions.conf?... |
| 16:07.15 | Essobi | A GK replaces the need to build that dialplan as it's dynamic. |
| 16:07.16 | Hmmhesays | or all calls go to another gateway which performs routing |
| 16:07.22 | *** join/#asterisk imrankamal (~as@202.88.44.24) |
| 16:07.26 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 16:07.28 | Hmmhesays | such as asterisk |
| 16:07.29 | Essobi | MikeJ Yup. |
| 16:07.33 | imrankamal | hi ppls |
| 16:07.55 | TheBear | Is there anyone who terminates VoIP -> pstn in TX, and/or AR, if so what rates ? |
| 16:07.58 | MikeJ_ | e.g. exten => _XXX,1,Congestion(); can _XXX be replaced by a global var like exten => ${VARNAME},104,Congestion(); |
| 16:08.12 | Weezey | besides D-Link who makes an MGCP User Agent? |
| 16:08.30 | MikeJ_ | O look at that... question already answered... thanks essobi. |
| 16:08.47 | Hmmhesays | i really wish acrobat reader wasn't so damn bloated |
| 16:09.12 | imrankamal | asterisk newbie here... requesting some help with config files |
| 16:09.26 | Hmmhesays | helps to ask a question |
| 16:09.41 | feral_kid | Anyone connected to FWD right now? |
| 16:09.57 | Zeeek | feral_kid no, why you need to know if it's up? |
| 16:10.33 | feral_kid | Zeeek: No, I need a test call into my server, to see if it is responding the way I would like... |
| 16:10.43 | ariel_ | feral_kid, give me you number |
| 16:11.03 | imrankamal | i installed FC 3 and asterisk 1.0.5 .. installation went smooth but whn i start * the system freezes |
| 16:11.13 | imrankamal | after 5 mins |
| 16:11.56 | eKo1 | Does it core dump? |
| 16:12.06 | eKo1 | Did you start it with the -g option? |
| 16:12.26 | imrankamal | no -vvvc |
| 16:13.15 | Skysky | can anyone tell me how to use festival's text2wav features? |
| 16:13.35 | Nugget | all speech synthesis sounds like a speak and spell. |
| 16:14.25 | Zeeek | especially if you feed it the wrong language! |
| 16:16.20 | eKo1 | Skysky: eh, type swift --help and rtfclo. |
| 16:17.37 | imrankamal | i bought 1 AudioCodes MP108 SIP gateways, never worked with SIP nor asterisk, can any1 pls tell me wht files i need to modify to place calls, our carrier is in US and i need to route calls to thr gatekeeper.. |
| 16:18.13 | *** join/#asterisk UBiQUiTY (~mike@68.160.103.76) |
| 16:18.21 | Skysky | eKo1: I got command not found |
| 16:18.37 | eKo1 | Skysky: Well then you don't have festival. |
| 16:18.55 | Skysky | eKo1: oh. i think i just compiled festival without installing |
| 16:19.19 | *** join/#asterisk seong (~seong@218.208.206.55) |
| 16:19.21 | UBiQUiTY | can anybody recommend the best filesystem to use for asterisk to store the voice files.... I'm expecting dictations of 10 minutes or less.... I'm debating between reiserfs or xfs, any ideas? |
| 16:19.32 | eKo1 | imrankamal: sip gateways...gatekeeper <---- You want to route sip to h.323? good luck. |
| 16:19.42 | MikeJ_ | is there any special syntax for including a second global section in an included file from extensions.conf... or can you just put a gloabs in each and they add |
| 16:19.51 | feral_kid | I am behind a nat... In sip.conf do I still need to specify nat=yes if I am going through iax2 for the connection? |
| 16:19.54 | eKo1 | UBiQUiTY: reiserfs is always a safe bet. |
| 16:19.59 | UBiQUiTY | ok cool thanX |
| 16:20.15 | eKo1 | MikeJ_: Try it and find out. |
| 16:20.42 | eKo1 | feral_kid: No. |
| 16:21.08 | tzanger | reiser? yikes |
| 16:21.10 | tzanger | xfs man, all the way |
| 16:21.22 | *** join/#asterisk wbzz (~3ed4e70d@grosha.majordomo.ru) |
| 16:21.28 | imrankamal | no eko1 :) sip to sip .. |
| 16:21.29 | tzanger | if you're looking for compatibiliy, use vfat on a CF card or something |
| 16:21.40 | tzanger | really though voicemail is *not* a high performance demanding kind of app |
| 16:22.20 | eKo1 | imrankamal: 'i need to route calls to thr gatekeeper..' <---- Gatekeepers are H.323 entities. |
| 16:22.21 | gr8nash | im reading about paging on lists.digium.com it says to create and extension that links to your sound card for overhead paging.. does anyone have an example i could borrow? =) |
| 16:22.42 | feral_kid | Hmmm... I tried to use FWD as a trunk, and when I had nat=yes, I could dial out to FWD (via iax2), but calling in caused congestion... I removed the nat=yes, and now I just get to hear Allison's voice telling me that circuits are busy... |
| 16:23.04 | imrankamal | eko1 sorry didn`t knew tht .. new to VoIP |
| 16:25.12 | eKo1 | imrankamal: You want to chat, do it here. |
| 16:25.42 | johnnyb | Did I hear that 1.07 is coming out soon? |
| 16:27.16 | Frantic | anyone knows if the atxfr (pbx attendent transfer) will be in 1.0.7? |
| 16:28.00 | tzanger | Frantic: my understanding is *NO NEW FEATURES* in 1.0.x, bugfixes only |
| 16:28.21 | johnnyb | It's in CVS, I know. I'm using it right now. |
| 16:29.11 | tzanger | yes I have used it too |
| 16:29.15 | tzanger | but CVS HEAD != 1.0.x |
| 16:29.38 | bjohnson | gr8nash: i think the sample extensions.conf has one |
| 16:29.49 | bjohnson | gr8nash: look at the console lines |
| 16:29.50 | *** join/#asterisk hemant (hemant@220.226.46.135) |
| 16:30.12 | *** join/#asterisk olivier_ (~olivier_@82.127.99.32) |
| 16:30.15 | *** join/#asterisk topping (~topping@dhcp024-210-082-196.columbus.rr.com) |
| 16:30.23 | gr8nash | bjohnson, thanks ill look now =) |
| 16:32.20 | *** join/#asterisk file (~jcolp@mctn1-142166192018.nb.aliant.net) |
| 16:35.34 | imrankamal | thanks for everything ...bye |
| 16:36.16 | *** join/#asterisk simonides (simon@byte.unitycode.org) |
| 16:36.28 | wbzz | hi |
| 16:36.34 | wbzz | i have a questio |
| 16:36.35 | wbzz | n |
| 16:37.29 | Frantic | tzanger: there's a bug with the current CVS |
| 16:37.51 | wbzz | is is possible to connect avaya gateway with asterisk ? |
| 16:38.00 | Frantic | i don't know what is it except i have all my sip extensions disappearing |
| 16:38.03 | wbzz | i found acticle on "voip-info" |
| 16:38.12 | *** join/#asterisk Hmmhesays (negative3k@66.173.103.108) |
| 16:38.16 | wbzz | though E1 |
| 16:38.18 | wbzz | ? |
| 16:38.38 | wbzz | does anybody did this ? |
| 16:38.39 | heison | wbzz: define connect... |
| 16:38.52 | heison | what are u trying to do? |
| 16:39.29 | wbzz | heison a have avaya gateway wich have 1200 extensions |
| 16:40.16 | wbzz | and it's need to call that extensions and play some voice message , from my application |
| 16:40.33 | *** join/#asterisk dsfr (~dsfr@216.207.244.183) |
| 16:40.45 | tzanger | Frantic: fun, see if it's already logged in mantis |
| 16:40.49 | wbzz | and i also have digium E1 board |
| 16:40.58 | *** join/#asterisk julianjm (~julianjm@250.Red-80-59-67.pooles.rima-tde.net) |
| 16:41.02 | wbzz | and want to connect my avaya station with asterisk |
| 16:41.06 | Weezey | anyone know who I can contact about the mgcp development in asterisk? |
| 16:41.18 | wbzz | heison is it possible ? |
| 16:41.20 | julianjm | http://www.voip-info.org/tiki-index.php?page=Asterisk+CDR+Areski+GUI |
| 16:41.29 | julianjm | sorry |
| 16:41.38 | *** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au) |
| 16:41.50 | *** join/#asterisk jeffik (~jeffik@m466f36d0.tmodns.net) |
| 16:41.53 | Faithful | hi guys, can we run speex? |
| 16:42.17 | Hmmhesays | if you stand on one foot whistle dixie and spin around |
| 16:42.19 | Faithful | or better would we want to run speex |
| 16:42.40 | darkskiez | Faithful, if your are bridging two * over the internet, maybe. |
| 16:42.54 | *** join/#asterisk SexyKen (~sexyken@c-67-161-5-149.client.comcast.net) |
| 16:43.07 | SexyKen | Hey guys I paid someone to setup asterisk and I can't get a hold of them and now my phones are offline |
| 16:43.09 | Faithful | hmmm what about sip clients? |
| 16:43.11 | SexyKen | how do I start asterisk properly |
| 16:43.20 | Frantic | tzanger> it's not- and I don't know how to describe the problem |
| 16:43.25 | Frantic | tzanger> or how to post it |
| 16:43.26 | Hmmhesays | asterisk -vvvvc should do it |
| 16:43.27 | *** join/#asterisk algorithmn (~na@ool-18bce89c.dyn.optonline.net) |
| 16:45.12 | SexyKen | Ahhh |
| 16:45.16 | SexyKen | Asterisk says 'returning 0' in clie |
| 16:45.18 | SexyKen | a lot |
| 16:47.24 | *** join/#asterisk jdg (~jdg@CA03F80A.adsl.mana.pf) |
| 16:49.55 | *** join/#asterisk xyznyc (~xyznyc@151.204.100.228) |
| 16:51.23 | Ron-Na | Does anybody has experience with realtime? |
| 16:51.43 | xyznyc | ? |
| 16:52.37 | johnnyb | Out of curiosity, how much to T1 and E1 channel banks cost, and who has the best ones? |
| 16:52.42 | Ron-Na | sip show peer 621 shows Codeces: 0x0 (nothing) when loaded from the database |
| 16:52.56 | johnnyb | Also, do you need as CSU/DSU with those? I've never worked with telephones before, just networks. |
| 16:54.02 | *** join/#asterisk deodax (~whoareyou@203.115.185.216) |
| 16:54.05 | deodax | hello people |
| 16:54.16 | deodax | i just moved my asterisk configuration to my new server |
| 16:54.44 | deodax | when someone from the outside calls me up he cant hear a ring... whats wrong with that? |
| 16:55.37 | algorithmn | Ron-Na: realtime? |
| 16:56.16 | *** join/#asterisk zapa (zapa@200.77.109.193) |
| 16:57.11 | pascals | I am having trouble connecting a PBX to my HFC-4S card. |
| 16:57.38 | pascals | I can dial in and out using a sip phone, but the pbx connected to the other two channels doesn't work |
| 16:58.42 | pascals | Anyone have a similar setup? |
| 16:59.07 | *** join/#asterisk habakuk (~chatzilla@24-117-8-113.cpe.cableone.net) |
| 17:00.28 | *** join/#asterisk yaaar (~chatzilla@lifebook.tranquility.net) |
| 17:00.33 | yaaar | word |
| 17:00.55 | TheBear | in dialing out from a std. phone via TDM -> X100p (within *) when "Attempting native bridge of Zap/2-1 and Zap/1-1" appears you can hear yourself in the handset of the std. phone. thus causing and echo. I now have echocancel = 16 echocancelwhenbriged = no and echotraining = 800, what else can I try to get rid of the echo |
| 17:01.25 | TheBear | same works on SIP phone |
| 17:03.50 | deodax | whats a good SIP fone for the pocket PC? |
| 17:04.04 | MikeJ_ | ~seen xkev |
| 17:04.08 | jbot | xkev <kevin@orbit.xmission.com> was last seen on IRC in channel #asterisk, 2d 8h 11m 56s ago, saying: 'lousy mysql licensing'. |
| 17:04.08 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 17:04.09 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 17:04.34 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 17:05.07 | *** join/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 17:06.11 | zapa | thebear i have the same problem with an E1 and i can't resolve it |
| 17:07.37 | nestAr | TheBear: echos on pots lines are hard to clear up |
| 17:07.42 | TheBear | zapa: :( well at least I'm not the only one :( |
| 17:07.45 | Eight | deodax: the only ones I know of are SJphone and xten's pocketPC phone. |
| 17:07.47 | zapa | thebear try to play with txgain rxgain |
| 17:07.51 | nestAr | yeah |
| 17:07.58 | nestAr | turn down the gains |
| 17:08.01 | nestAr | see if that helps |
| 17:08.05 | Eight | deodax: haven't tried either. |
| 17:08.08 | TheBear | zapa: ok will try that |
| 17:08.32 | TheBear | that would be rxgain=-0.5 right ? |
| 17:08.40 | zapa | try |
| 17:08.42 | nestAr | that's a good starting point |
| 17:08.43 | zapa | ztmonitor |
| 17:08.50 | nestAr | you need to restart asterisk |
| 17:08.53 | nestAr | a reload won't do it |
| 17:08.59 | nestAr | each time you edit the gains |
| 17:10.21 | *** part/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 17:10.47 | nestAr | good to know "show agents" does nothing on the CLI |
| 17:11.53 | *** part/#asterisk dubphil (~dubphil@80.124.137.201) |
| 17:13.05 | jontow | well.. hmm, got kphone to receive audio; but i can't get the mic to work |
| 17:14.07 | TheBear | ok besides echo what else is rxgain and txgain going to effect? that is helping |
| 17:14.11 | *** join/#asterisk Darwin35 (~Darin@c-24-3-241-22.client.comcast.net) |
| 17:14.43 | johnnyb | How do you determine what codec a call is using? |
| 17:14.45 | *** join/#asterisk charles___ (~charles@64.35.172.37) |
| 17:14.50 | charles___ | Hi |
| 17:14.55 | Darwin35 | *= 1 giant bug |
| 17:15.02 | charles___ | Does anyone use MTA-102 boxes ? |
| 17:15.06 | zapa | thebear the volumen in the Zap call |
| 17:15.07 | Darwin35 | -/= |
| 17:15.54 | Darwin35 | good afternoon all |
| 17:16.01 | charles___ | bkw_ have you had any experience with icablesystem / mta-v102 ? |
| 17:19.12 | lidl | anyone available for for iax2 test call? |
| 17:20.19 | *** join/#asterisk kram (~mark@kram.digium.sponsor.pdpc) |
| 17:20.19 | *** mode/#asterisk [+o kram] by ChanServ |
| 17:25.03 | TheBear | zapa: ok besides echo what else is rxgain and txgain going to effect? that is helping much. also what is ztmonitor ? |
| 17:25.52 | *** join/#asterisk Tall-guy (tall-guy@hssxrg207-195-103-110.sasknet.sk.ca) |
| 17:25.59 | *** part/#asterisk Dirtysole (~Frank@cpe.atm0-0-0-1201091.0x503fc556.bynxx2.customer.tele.dk) |
| 17:26.00 | yaaar | hey guys. anybody know why microphone recording in linux would play back at double speed? it's not really an asterisk issue, as I see the behavior in gnome sound recorder...but it certainly is giving me a hard time making phone calls from the box |
| 17:26.12 | Weezey | can someone dial this to make sure it's working: |
| 17:26.14 | Weezey | IAX2/guest@txrx.ca/2525 |
| 17:28.26 | Dandan | yaaar: it is a sampling issue |
| 17:28.31 | Dandan | i have the same thing |
| 17:28.49 | Dandan | yaaar: if you sample at 8 and play at 16 kHZ that's what happens |
| 17:29.58 | shido6 | i hear nothing |
| 17:30.00 | Tall-guy | Anyone using "Eyebeam" (video) and Asterisk? |
| 17:31.11 | *** join/#asterisk Flash_ (~Flash_@pc-80-235-133-235-na.blueyonder.co.uk) |
| 17:31.19 | Flash_ | Afternoon all |
| 17:31.31 | Flash_ | Has anyone used these phones: http://www.sipura.com/products/spa841.htm ? |
| 17:34.03 | Eight | Flash_: not personally, but people seem happy with them. |
| 17:34.10 | *** join/#asterisk km- (~pgrace@brdgw1.rttx.com) |
| 17:34.17 | dreamcode | how do i install MeetMee |
| 17:34.24 | shido6 | got a timing source? |
| 17:34.27 | shido6 | usb or digium card |
| 17:34.32 | shido6 | fxo fxs card... |
| 17:34.35 | shido6 | t1 or e1 card |
| 17:34.35 | km- | hehe |
| 17:34.37 | km- | "Got Timing?" |
| 17:34.44 | Eight | shido6: or with kernel 2.6, you don't need either of those. |
| 17:35.48 | dreamcode | :( i have kernel 2.4 |
| 17:35.56 | dreamcode | and no timing source |
| 17:36.13 | Eight | dreamcode: ZTDummy will use USB hardware to do it. |
| 17:36.17 | Eight | dreamcode: so there's still hope. |
| 17:36.24 | Eight | dreamcode: but not from me (never tried it) |
| 17:36.26 | dreamcode | .. i'm tring to create ca conference |
| 17:37.55 | JerJer[mobile] | ztdummy will blow chunks after more than just a few users |
| 17:38.15 | km- | heh |
| 17:38.21 | johnnyb | Is there a reason that asterisk excludes iLBC by default if you have no codecs specified in sip.conf? |
| 17:38.23 | Eight | JerJer[mobile]: what are the symptoms? |
| 17:38.36 | km- | jerjer: hey, what's GTS's website? I tried www.gtsolutions.biz but it wasnt the right one |
| 17:38.56 | *** join/#asterisk mad_router (~ornate@84.246.176.66) |
| 17:39.20 | km- | doesnt seem to pop up with a google search, guess I could try to troll ebay for one of their posts |
| 17:40.17 | *** join/#asterisk srosebush (~steve@initrd.com) |
| 17:40.27 | Delvar | anyone remember the setting to split the CDR's, so for each call you get 2 CDR's (one from sip user, and another to pstn gateway)? |
| 17:41.31 | srosebush | Hey I got a small problem -- Using * 1.0.6.. When I call from one SIP device to the other it shows the called id as 'asterisk'?? I've tried specifying 'callerid=' in the sip.conf for each of the sip accounts but that doesn't seem to work.. any ideas? |
| 17:41.51 | *** join/#asterisk gr8nash (~andy@mamabear.si-forest.com) |
| 17:42.12 | *** join/#asterisk skiold (~skiold@84-121-68-176.onocable.ono.com) |
| 17:43.48 | *** join/#asterisk santiago (~santiago@201.245.167.104) |
| 17:46.09 | *** join/#asterisk sssk (~Administr@152.25.static.dsl.luna.net) |
| 17:46.14 | sssk | hi all |
| 17:46.35 | sssk | I have a question about SIP |
| 17:46.41 | sssk | Can I? |
| 17:47.16 | Flash_ | sssk: go for it |
| 17:47.18 | santiago | sssk, don't ask if you can ask, just ask |
| 17:47.23 | sssk | Is it possible to call from Grandstream BudgeTone to asterisk without register? |
| 17:47.25 | *** join/#asterisk caesar2 (caesar@pD9E4E037.dip.t-dialin.net) |
| 17:47.29 | Flash_ | doesn't that really get your goat ? |
| 17:47.38 | outtolunc | bic! |
| 17:47.41 | outtolunc | oops |
| 17:48.43 | caesar2 | hi, is it possible so send a "pease re-register yourselfe" command to a phone (nat prob. after 24h dieconnect) |
| 17:49.05 | sssk | caesar2: What phone? |
| 17:49.07 | km- | most phones have a configurable registration interval |
| 17:49.18 | caesar2 | allnet, grandstream |
| 17:49.43 | Zeeek | some grandstream firmware drops out of register cycle |
| 17:49.46 | sssk | caesar2: You can reboot it... |
| 17:49.49 | Zeeek | get 5.18 |
| 17:49.55 | Zeeek | or later |
| 17:50.08 | caesar2 | prob. is that after my 24 disconnect and reconnect the phone keeps sending "0" udp packets to my asterisk but asterisk does not refresh the phones ip so it seem to be offline.. |
| 17:50.26 | lidl | did anyone get mpg123 compiled for amd64? |
| 17:50.42 | hardwire | whoop whoooop whoooooop whoop |
| 17:52.38 | caesar2 | i hate nat ! |
| 17:52.42 | km- | word |
| 17:52.45 | hardwire | heh! |
| 17:52.47 | sssk | _Flash: What do you mean with that goat stuff? |
| 17:52.52 | jpayne | anyone experiencing clock drift on 7960s hooked up to asterisk? |
| 17:53.03 | jpayne | s/anyone/anyone else/ |
| 17:53.37 | km- | he spelled it wrong, it's goad, I believe |
| 17:53.40 | *** join/#asterisk amsterdam (~ak@xdsl-81-173-129-48.netcologne.de) |
| 17:54.03 | km- | <PROTECTED> |
| 17:54.17 | km- | when someone or something "gets your goad" it means they push your buttons, piss you off, etc. |
| 17:54.31 | caesar2 | strange prob : i have an extern nat phone.. ping from office1 to office2 is 60ms.. whenn i start to speak ping gets 3000ms an rising... connection speed is enough ! so why ? any idear ? |
| 17:54.51 | Flash_ | it's now 6pm |
| 17:55.33 | sssk | ah great |
| 17:55.33 | sssk | Thanks flash |
| 17:55.53 | *** part/#asterisk srosebush (~steve@initrd.com) |
| 17:56.51 | caesar2 | anyone knows what to use best for music on old on an 133 mhz soekris mini pc ? |
| 17:57.17 | hardwire | on old? |
| 17:57.20 | hardwire | oh |
| 17:57.22 | hardwire | on hold |
| 17:57.26 | caesar2 | hold sorry |
| 17:57.27 | hardwire | 133mhz? |
| 17:57.27 | sssk | caesar2: Probably some uncompressed music |
| 17:57.33 | hardwire | yeh.. convert everything to gsm |
| 17:57.33 | *** join/#asterisk lordcian (~kvirc@209.194.32.60) |
| 17:57.51 | *** join/#asterisk file[mac] (~file[mac]@mctn1-142166192018.nb.aliant.net) |
| 17:57.51 | caesar2 | and no mpg123 ? |
| 17:57.52 | *** join/#asterisk IQ (~IQ@204-26-74-243.omah.qwest.net) |
| 17:57.52 | caesar2 | ok |
| 17:58.04 | hardwire | file[mac]: what kind o sicko are you? |
| 17:58.21 | caesar2 | asterisk runns great at 120mhz 40 ram... !! |
| 17:58.31 | file[mac] | hrm |
| 17:58.41 | file[mac] | the kind who has a Mac |
| 17:58.59 | file[mac] | this irc client isn't behaving however |
| 17:59.01 | lordcian | has anyone managed to get Zap working after installing AMP? |
| 17:59.08 | caesar2 | p120 -> 50 mb ! 8 sim. calls.. 50% cpu |
| 17:59.31 | caesar2 | sorry 40 mb |
| 17:59.53 | IQ | Hi, Anyone using WiFi SIP phone ? |
| 18:00.03 | hardwire | caesar2: I use 233 mhz geodes as well |
| 18:00.16 | hardwire | and can pass 20~ gsm lines |
| 18:00.21 | hardwire | a hell of a lot more probably.. using SER |
| 18:00.46 | hardwire | IQ: I love the Hitachi Cable phones |
| 18:00.51 | hardwire | they work wonderfully |
| 18:00.55 | hardwire | but I haven't range tested them yet |
| 18:01.14 | hardwire | if you want range grab some FXS lines and use the Engenius phones |
| 18:01.17 | *** join/#asterisk visik7 (~ciao@visik7.user) |
| 18:01.21 | *** part/#asterisk visik7 (~ciao@visik7.user) |
| 18:01.26 | caesar2 | http://www.soekris.com/net4501.htm -> just trying to run on that... |
| 18:02.12 | jontow | caesar2; that model doesn't offer a lot of horsepower regarding the CPU ... i wondered how * would run |
| 18:02.21 | IQ | hardwire: thanks for the input :) |
| 18:02.34 | hardwire | IQ: I output a lot. |
| 18:02.42 | hardwire | or is that I Putout a lot. |
| 18:02.43 | caesar2 | it does! perfect... but still have probs with music on hold |
| 18:02.44 | hardwire | I forget. |
| 18:02.53 | jontow | it has enough though, for a couple people in the Voicemail app, a handful of calls, and some transcoding? |
| 18:03.18 | IQ | hardwire: since we have WiFi everywhere we go. Like at work and at home... I was thinking of getting one of those WiFi SIP phones they'll be kinda cell phones to me :) |
| 18:03.37 | lordcian | has anyone successfully configured an extension with a inbound zap channel? |
| 18:03.39 | jontow | and what OS are you using? linux? |
| 18:03.46 | *** join/#asterisk zagaya972 (~d2s-compa@APointe-a-Pitre-102-1-17-26.w81-248.abo.wanadoo.fr) |
| 18:03.48 | IQ | of course - they wont work while I'm in the car - for that I've my cell phone too :P |
| 18:03.53 | hardwire | IQ: I wouldn't suggest it |
| 18:03.53 | lordcian | and are willing to answer a couple of questions? |
| 18:03.53 | hardwire | heh |
| 18:03.59 | caesar2 | linux26 init=/start.sh -> starts asterisk... nothing more... |
| 18:04.01 | TheBear | I can dial 2500 to get to Voicemailmain() enter mailbox and password. How do I get to record your greeting, I've tried 3 all it has i * for main menu ? |
| 18:04.06 | hardwire | but the Hitachi Cables have multi lsip provider support |
| 18:04.08 | hardwire | and wireless profiles |
| 18:04.12 | jontow | hah |
| 18:04.23 | jontow | nice, but what about remote management? anything? (ssh?) |
| 18:04.31 | hardwire | serial |
| 18:04.31 | IQ | hardwire: hmmm... let me check HItachi Cable phone... any model number or url? |
| 18:04.31 | jontow | even ftp to upload new configs, and an extension to reload * ? |
| 18:04.35 | caesar2 | web interface.. and serial |
| 18:04.38 | hardwire | www.abptech.com |
| 18:04.47 | hardwire | IQ: I am a reseller to so send me moneys |
| 18:05.18 | IQ | hardwire: looool... sure why not... someone will make money out of me so why not a member of Asterisk :) |
| 18:05.25 | lordcian | can anyone help me configure a zap channel? |
| 18:06.05 | JunK-Y | lordcian: www.voip-info.org, loop for zaptel.conf |
| 18:06.05 | *** join/#asterisk mystik (nobody@ip68-14-64-188.ri.ri.cox.net) |
| 18:06.10 | JunK-Y | and zapata.conf |
| 18:06.16 | lordcian | got them working. |
| 18:06.23 | lordcian | its the extension i can't figure... |
| 18:06.29 | zapa | thebear: sorry ztmonitor is tu visualize the gain |
| 18:06.35 | JunK-Y | then search for extensions.conf |
| 18:07.10 | Dandan | <PROTECTED> |
| 18:07.21 | Dandan | anyone else getting it? |
| 18:07.26 | Dandan | is it my misconfiguration? |
| 18:08.16 | IQ | hardwire: these things are expensive :) |
| 18:09.13 | deodax | when someone from the outside calls me up he cant hear a ring... whats wrong with that? |
| 18:09.38 | Flash_ | question: much difference between 1.0.2 and current version ? |
| 18:10.45 | *** join/#asterisk file[laptop] (~file@mctn1-142166192018.nb.aliant.net) |
| 18:11.05 | spackle | file, are you home now? |
| 18:11.08 | Delvar | mn found it http://www.voip-info.org/wiki-Asterisk+cmd+ForkCDR |
| 18:11.28 | spackle | file: or are you writing from the bottom of the ocean? |
| 18:14.58 | *** join/#asterisk mitcheloc (~mitchel@69-169-28-46.anhmca.adelphia.net) |
| 18:15.43 | mitcheloc | does anyone know agillis? |
| 18:16.13 | spackle | outtolunc: I think perhaps file was eaten by a bear and a moose stepped on his laptop. |
| 18:17.00 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
| 18:17.00 | mitcheloc | bkw_: that asterisk distro you were working on, who was it wyou were working on it with? |
| 18:17.02 | *** join/#asterisk feral_kid (~not@209.205.207.130) |
| 18:17.07 | *** part/#asterisk lidl (~little@213-140-6-96.fastres.net) |
| 18:21.06 | mitcheloc | slow day or is no one talking? |
| 18:22.15 | Hmmhesays | it is monday |
| 18:22.27 | Hmmhesays | a majority of people dislike monday's |
| 18:23.02 | Zeeek | which is why they should be venting here |
| 18:23.21 | Zeeek | I'll start: today sucked, but it's almost over |
| 18:23.36 | km- | haha |
| 18:23.40 | km- | almost over? |
| 18:23.42 | UBiQUiTY | slow day eh? ive got a mystery for us |
| 18:23.45 | km- | it's not even 1:30! |
| 18:23.53 | km- | I'm waiting for digium to get back to me |
| 18:23.56 | Zeeek | 19:30 in thi part of da woild |
| 18:24.21 | km- | I can hear the headscratching in alabama all the way from PA |
| 18:24.37 | km- | my report on my T1 problems was like 4 pages long |
| 18:24.37 | km- | hehe |
| 18:24.38 | UBiQUiTY | i'm using the T405P, and i'm getting all this bogus activity on all of my channels.... does this mean my dchannel is wrong? or signalling? |
| 18:24.38 | Zeeek | lice |
| 18:25.05 | km- | ubiquity: that'd be my guess. Sometimes when I first activate the 405p I get a whole bunch of 1-second calls, but they all disconnect after a second or two |
| 18:25.22 | UBiQUiTY | yes thats exactly what im getting |
| 18:25.35 | UBiQUiTY | except they are copnstantly coming in, then get dropped over and over again |
| 18:25.39 | km- | for me though, it settles down after a second or two |
| 18:25.47 | UBiQUiTY | hmm |
| 18:25.47 | km- | hmm |
| 18:25.50 | km- | hehe |
| 18:25.53 | UBiQUiTY | :-) |
| 18:25.59 | km- | possible that you're on the wrong channels? |
| 18:26.17 | km- | if it's an integrated access t1, you sure you're not provisioning the channels the data portion is using for voice? |
| 18:26.23 | UBiQUiTY | well, i have it set in zaptel.conf for e&m=1-23, and dchan=24 |
| 18:26.32 | km- | e&m doesnt have a dchannel |
| 18:26.39 | km- | only PRI has a d-channel |
| 18:26.47 | UBiQUiTY | damn that must be it then |
| 18:26.58 | km- | yeah, e&m=1-24 should set you to rights |
| 18:27.05 | UBiQUiTY | i was told by the phone company that the dchannel is on 24 tho |
| 18:27.06 | km- | then setup channels 1-24 as em or em_w |
| 18:27.07 | tzanger | I think GR303 has a D channel too |
| 18:27.14 | UBiQUiTY | ok i will try that now |
| 18:27.18 | km- | ubiquity: you sure you dont have PRI? |
| 18:27.25 | UBiQUiTY | i do have a PRI |
| 18:27.27 | km- | tzanger: hey, dude, I did some added debugging last night on my t1 problem |
| 18:27.34 | km- | ubiquity: then you dont want channels 1-23 on e&m I dont believe |
| 18:27.43 | JerJer[mobile] | then why are you doing E&M? |
| 18:27.43 | UBiQUiTY | but the guy from the phone company says the PRI speaks e&m wink |
| 18:27.48 | km- | ubiquity: I think you want it something like bchan=1-23, dchan=24 |
| 18:27.51 | km- | WHAT? |
| 18:27.51 | tzanger | JerJer[mobile]: are you still travelling? jeeebus |
| 18:27.52 | JerJer[mobile] | he's full of shit then |
| 18:27.55 | UBiQUiTY | hahahaah |
| 18:27.59 | km- | What phone company you use |
| 18:28.01 | km- | so I can avoid them |
| 18:28.02 | JerJer[mobile] | PRI talks PRI |
| 18:28.09 | *** join/#asterisk afe ([vRz+ErN9D@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 18:28.14 | UBiQUiTY | VERIZON |
| 18:28.16 | *** join/#asterisk file[laptop] (~file@mctnnbsah25-142166091160.nb.aliant.net) |
| 18:28.17 | km- | "PRI talking e&m wink" is like saying "TCP/IP talks DECNET" |
| 18:28.21 | km- | hahahaha |
| 18:28.25 | km- | make progress every day... riiight. |
| 18:28.25 | UBiQUiTY | lol |
| 18:28.29 | Katty | wocka |
| 18:28.31 | *** join/#asterisk RickTick (RickTick@c-66-176-54-161.se.client2.attbi.com) |
| 18:28.33 | tzanger | km-: what'd you find? |
| 18:28.33 | UBiQUiTY | its not actually a verizon guy |
| 18:28.35 | tzanger | wocka? |
| 18:28.36 | UBiQUiTY | its a mitel guy |
| 18:28.38 | JerJer[mobile] | progress backwards |
| 18:28.40 | Katty | like moo |
| 18:28.44 | km- | tzanger: lemme grab the e-mail, just a sec |
| 18:28.44 | tzanger | did you get a soybean stuck in your throat or something, katty? |
| 18:28.56 | Katty | no ;) |
| 18:29.01 | UBiQUiTY | we have the pri's coming into a mitel switch, then the mitel puts out a PRI directly into my asterisk box |
| 18:29.08 | JerJer[mobile] | that's not a soybean |
| 18:29.14 | JerJer[mobile] | more like tube steak |
| 18:29.15 | tzanger | JerJer[mobile]: *coughs* |
| 18:29.19 | km- | tzanger: going to pastebin and give you relevant line numbers |
| 18:29.19 | Katty | JerJer[mobile]: umm. |
| 18:29.20 | tzanger | dammit |
| 18:29.29 | tzanger | km-: ok |
| 18:29.51 | JerJer[mobile] | UBiQUiTY: then the mitel isn't delivering you a PRI |
| 18:30.04 | km- | tzanger: http://pastebin.ca/7416 |
| 18:30.08 | JerJer[mobile] | if the switch monkey tells you E&M wink |
| 18:30.20 | km- | tzanger: lines 49-88 are the relevant bug stuff |
| 18:30.31 | km- | tzanger: I added debug before every zt_rbs_sethook to say where the functions were being called from |
| 18:30.43 | km- | tzanger: on the good calls from the CO, the winking occurs correctly |
| 18:30.58 | UBiQUiTY | JerJer: what can mitel be giving me BESIDES a pri? |
| 18:31.15 | UBiQUiTY | oh, i see what u mean |
| 18:31.17 | UBiQUiTY | hmmmm |
| 18:31.22 | JerJer[mobile] | km-: why do you have to use E&M ? |
| 18:31.33 | km- | tzanger:my zapata.conf and zaptel.conf are there as well |
| 18:31.33 | tzanger | UBiQUiTY: CAS T1? |
| 18:31.37 | UBiQUiTY | i'm trying pri_cpe, bchan=1-23, dchan=24 right now |
| 18:31.39 | tzanger | km-: I see that, reading |
| 18:31.58 | tzanger | JerJer[mobile]: he is just "injecting" asterisk into an existing configuration |
| 18:32.08 | *** join/#asterisk Mother_ (~mother@93.Red-80-32-127.pooles.rima-tde.net) |
| 18:32.11 | Mother_ | hi all |
| 18:32.12 | JerJer[mobile] | upgrade |
| 18:32.16 | km- | jerjer: if I get the phne company to switch to PRI, I'll have to have the NEC guys come out and reset the NEC PBX to PRI as well, because if the asterisk box blows up or I get hit by a bus, the marching orders are to unplug asterisk and bypass the biscuits together so that the NEC system becomes the only system at that point |
| 18:32.18 | RickTick | hello all: When I bridge an incoming calling to cellular , the cellular only rings once before going to voicemail. I have extended the rings of the cellular to 30 secs. Any way to make the cellular ...ring longer? |
| 18:32.20 | tzanger | he could probably get PRI but this *should* be working |
| 18:32.48 | Mother_ | should raw wav/riff files recorded using Monitor() be playable in Windows by copying the file directly? |
| 18:32.57 | km- | yeah I mean, if I get PRI it invalidates the proof of concept that Asterisk can do this simple kind of magic |
| 18:33.08 | Mother_ | I just get a bunch of crap out the speakers, even though the header is OK (16-bit 8kHz mono) |
| 18:33.12 | *** join/#asterisk fjoe (~fjoe@samodelkin.net) |
| 18:33.14 | fjoe | re |
| 18:33.26 | fjoe | anyone using quadBRI cards with *? |
| 18:33.33 | km- | Mother: there are two types of wave files, your particular format may not be the one windows can use |
| 18:33.40 | km- | mother: theres wav49 then there's WAV or something |
| 18:33.55 | tzanger | km-: when asterisk signals the CO it does not wink. This is normal. Only the receiving side winks to let the originating side know it's ready |
| 18:33.59 | Mother_ | km-: argh, I suspected that - thanks, will look into it |
| 18:34.35 | km- | tzanger: ok, so, Asterisk doesnt have to wink either end |
| 18:34.38 | MooingLemur | RickTick: I called my cell provider and had them completely disable voicemail. That way I just give out my main number and asterisk does the call forwarding |
| 18:34.43 | tzanger | no no when it accepts calls from the CO it should wink |
| 18:34.47 | tzanger | and it looks like it is |
| 18:34.50 | km- | right |
| 18:35.04 | tzanger | and when it accepts calls from the NEC it will wink. The terminating side winks to let the originating side know it's ready |
| 18:35.07 | km- | but if it's originating the call, either side (the CO or the PBX) should be winking back, right? |
| 18:35.15 | tzanger | now having said that, it should be WAITING for a wink when originating |
| 18:35.28 | Mother_ | hehe just googling wav49 file format brings up loads of asterisk stuff |
| 18:35.36 | tzanger | you didn't add any printks to the receiving side so I can't see if it's waiting ofr the wink or not |
| 18:35.46 | km- | tzanger: I can always throw more in |
| 18:35.50 | km- | tzanger: know what function I should tap? |
| 18:36.03 | tzanger | yeah I did... let me look |
| 18:36.32 | Mother_ | so any way to convert wav49 to "normal" wav? |
| 18:36.47 | JerJer[mobile] | sox? |
| 18:36.57 | km- | yeah, or you could change your monitor settings to use the other codec, right? |
| 18:37.07 | JerJer[mobile] | yeppers |
| 18:37.30 | Mother_ | km: just add :<format> right? |
| 18:37.34 | km- | I believe so |
| 18:37.34 | afe | I think if you set monitor to use WAV it's actually wav49 (doesn't work well in windows) and wav will generate regular wavs |
| 18:37.52 | Mother_ | OK, thanks a bunch |
| 18:38.08 | km- | my pleasure to assist you |
| 18:38.13 | Mother_ | ;) |
| 18:38.25 | Mother_ | now comes the line "that'll be $10000 please" |
| 18:38.26 | Mother_ | hehe |
| 18:38.51 | km- | haha no |
| 18:38.55 | km- | I'm through charging for asterisk support |
| 18:39.04 | tzanger | km-: I'm gonna guess zt_q_sig() |
| 18:39.08 | RickTick | MooingLemur: But I wanted the cellular to answer ..unanswered calls. |
| 18:39.22 | km- | tzanger: ok, lemme throw some debugs in there and see if I can clear the system for a restart |
| 18:39.50 | tzanger | not sure if htat's it though |
| 18:39.55 | tzanger | it looks at incoming signaling |
| 18:40.11 | km- | tzanger: mark calls to zt_q_sig, or log specifics from the zt_q_sig operation? |
| 18:40.20 | tzanger | hang on |
| 18:40.23 | tzanger | it doesn't look right |
| 18:40.30 | tzanger | E&M is done in asterisk not zaptel |
| 18:40.45 | km- | ok |
| 18:40.48 | Juggie | E&M is done in libpri |
| 18:41.01 | RickTick | MooingLemur: When I dial the cell directly from my ATA... the cellular ring fine ... enough rings to pick up |
| 18:41.32 | tzanger | chan_zap.c chan_call() looks at signaling, I'm still digging |
| 18:41.34 | km- | huh? e&M in libpri? |
| 18:41.35 | km- | hah |
| 18:41.42 | UBiQUiTY | i got it!!! you're right... my mitel guy doesnt know what he's saying |
| 18:41.48 | km- | hahaha |
| 18:41.54 | UBiQUiTY | i have it as PRI_NET bchan=1-23 dchan=24 |
| 18:41.55 | km- | wouldnt be the first time that a telco jock got something wrong |
| 18:41.57 | UBiQUiTY | lol |
| 18:42.07 | UBiQUiTY | damn why did i listen to him all these weeks??? |
| 18:42.07 | UBiQUiTY | hahah |
| 18:43.20 | km- | someone should write a zttool that's on crack or somethin |
| 18:43.32 | km- | like, one that'll drop a log of everything |
| 18:43.36 | km- | as opposed to seeing on the screen |
| 18:43.40 | UBiQUiTY | and you're right, for about 2 sec, all the channels were going haywire, then settled down to idle |
| 18:43.42 | *** join/#asterisk dooder (~nateputna@66.241.89.77) |
| 18:43.45 | km- | so that we can capture and watch ABCD |
| 18:44.05 | km- | ubi: yeah, not sure why it does that, but it seems pretty inoccuous |
| 18:44.11 | *** join/#asterisk TedC (~ted@gray.impulse.net) |
| 18:44.50 | TedC | Anybody know the local CO test line number in Verizon CA territory? |
| 18:46.21 | JerJer[mobile] | TedC: try a local npa-nxx-9999 |
| 18:46.31 | deodax | hey people |
| 18:46.38 | km- | tzanger: is it about time that I rent a T1 test set for a couple days? :) |
| 18:46.39 | JerJer[mobile] | no people here, just bots |
| 18:46.39 | UBiQUiTY | by innoccuous, do u mean harmless? |
| 18:46.51 | TedC | JerJer[mobile]: good call, but no luck. I've tried 987-1102, etc. |
| 18:46.54 | km- | ubiquity: yeah |
| 18:47.00 | UBiQUiTY | ok cool |
| 18:47.11 | tzanger | km-: no |
| 18:47.27 | tzanger | I'm just looking at the code in zt_call |
| 18:47.38 | tzanger | it really does appear that asterisk doesn't wait for any wink whatsoever |
| 18:48.09 | km- | thats weird |
| 18:48.15 | km- | how is this working for other people |
| 18:48.30 | km- | is it possible I'm the only person with em wink who notices this problem? |
| 18:48.40 | tzanger | km-: all FXS signaling, all E&M signaling, (basically everything but FXO signaling) uses the same code |
| 18:48.52 | km- | It's totally screwing my SIP rollout, because the sip phones stay in call progress mode because the system never knows the phone is picked up |
| 18:49.30 | km- | totally odd crap |
| 18:49.31 | km- | heh |
| 18:49.51 | tzanger | wow |
| 18:49.57 | tzanger | zt_dial looks really nasty |
| 18:50.02 | tzanger | it builds a dialstring like what you'd send to a modme |
| 18:50.20 | km- | well, with the FXO's, they are modems |
| 18:50.26 | km- | so it doesnt surprise me |
| 18:50.41 | Mother_ | this is weird, I was already specifying wav in the Monitor command |
| 18:50.48 | Mother_ | maybe it takes wav == wav49 |
| 18:50.50 | tzanger | ZT_DIAL_OP_REPLACE |
| 18:50.56 | tzanger | so it comes back into Zaptel as that |
| 18:52.04 | tzanger | god damn is this messy |
| 18:52.14 | RickTick | Anybody: My Problem: When I bridge an incoming call to cellulr, the cell only rings once and goes to voicemail. I want the cell to ring at least 4 times and to answer ...unanswered calls with its voicemail. Extending the ring time ... on the cellular ...does not work. HELP |
| 18:52.48 | tzanger | ioctl(p->subs[SUB_REAL].zfd, ZT_DIAL, &p->dop) |
| 18:52.51 | tzanger | that's what you hit in zaptel |
| 18:52.54 | tzanger | now back to zaptel.c |
| 18:52.57 | Essobi | JerJer[mobile] Hey.. I got the 2-way DTMF working I was talking to you about the other day.. that was on inbound to the h323 device.. now outbound is being a bitch. I'm geting an error on MediaTransport.. Isn't q931 cause 7, call completed? |
| 18:53.16 | km- | makes me wish I could put a debugger on the module |
| 18:53.33 | tzanger | km-: it just jumps to __do_dtmf() |
| 18:53.46 | tzanger | so I am willing to put real good money on zaptel's NOT waiting for a wink |
| 18:53.55 | tzanger | because nothing in zaptel looks like it waits for wink |
| 18:54.00 | tzanger | and nothing in chan_zap.c looks that way either |
| 18:54.22 | tzanger | km-: what exactly happens when you try to dial out to telco again? |
| 18:54.27 | tzanger | missed leading digits? |
| 18:54.43 | km- | no, that problem was fixed with emdigitwait= |
| 18:54.57 | km- | the problem is that when I originate the call from the asterisk system to the CO, the line never leaves "Ringing" state |
| 18:54.59 | km- | because of this |
| 18:55.17 | km- | SIP phones are always stuck in "Session Progress" mode, as opposed to "Connected" or "Complete" |
| 18:55.24 | Essobi | cause it's not winking at you to change it's state? or * isn't seeing the wink? |
| 18:55.34 | tzanger | Essobi: * does not appear to ever LOOK for wink |
| 18:55.38 | km- | asterisk apparently isn't seeing the wink, but it may not even be looking for it. |
| 18:55.40 | Essobi | Ahh, nice. |
| 18:55.42 | tzanger | Essobi: * certainly EMITS wink |
| 18:56.02 | km- | Asterisk is not even checking out my winking, regardless of how many drinks I buy it |
| 18:56.07 | tzanger | haha |
| 18:56.10 | Essobi | Right right.. where's this line at? |
| 18:56.17 | km- | philadelphia suburbs |
| 18:56.26 | Essobi | km- I get the same response.. That and 302 moved.. Damnit. |
| 18:56.36 | km- | esf/b8zs channelized T1, 1/3 of it is data, the other 12 or so channels are e&m wink |
| 18:56.45 | tzanger | km-: I'm going to see if I can't make it ignore wink on originate |
| 18:56.49 | Essobi | Ahh, e&m. |
| 18:56.49 | tzanger | just to see if it cleans up the problem |
| 18:57.08 | km- | tzanger: force it to up as soon as the call is originated? That sounds like a good workaround |
| 18:57.21 | Essobi | H323 isn't co-operating with me right now either.. |
| 18:57.23 | tzanger | km-: basically |
| 18:57.28 | km- | H323 has never cooperated with me |
| 18:57.30 | tzanger | km-: as I said all E&M functions are done in userland |
| 18:57.37 | km- | tzanger: gotcha.. |
| 18:57.44 | km- | I dont see how that'd be a problem |
| 18:57.57 | km- | I mean, I dont seem to have problems originating the call, its just the asterisk never sees that it connects |
| 18:58.04 | Essobi | unless you want to track other endcall states in cdrs? |
| 18:58.07 | km- | only other thing is that it doesn't detect the hangup either |
| 18:58.08 | tzanger | km-: ok |
| 18:58.10 | tzanger | chan_zap.c |
| 18:58.18 | tzanger | zt_handle_event() |
| 18:58.28 | RickTick | Anyone ... call forwarding inbound calls to a cellular? |
| 18:58.31 | tzanger | around like 3116 in my CVS pull |
| 18:58.58 | km- | I'm running cvs as of like, end of february |
| 18:59.00 | Essobi | What's the standard mediatransporterror in H323 mean? |
| 18:59.15 | km- | CVS-HEAD-02/23/05-18:20:10 |
| 18:59.21 | tzanger | km-: you'll see (around line 3205) a HUGEASS set if p->sig == BLAH || p->sig == BLAH2 ... |
| 18:59.23 | km- | so just give me a few lines so I can see where you're putting it |
| 18:59.30 | tzanger | } else if (p->confirmanswer || (!p->dialednone && ((p->sig == SIG_EM) || (p->sig == SIG_EM_E1) || (p->sig == SIG_EMWINK) || (p->sig == SIG_FEATD) || (p->sig == SIG_FEATDMF) || (p->sig == SIG_E911) || (p->sig == SIG_FEATB) || (p->sig == SIG_SF) || (p->sig == SIG_SFWINK) || (p->sig == SIG_SF_FEATD) || (p->sig == SIG_SF_FEATDMF) || (p->sig == SIG_SF_FEATB)))) { |
| 18:59.35 | tzanger | that hugeass line |
| 18:59.41 | Dandan | anyone here with BV experience willing to help me troubleshoot? |
| 18:59.44 | tzanger | just remove "(p->sig == SIG_EMWINK) || " |
| 19:00.09 | tzanger | this is certainly not the correct way to fix it |
| 19:00.13 | tzanger | but let's see what happens :-) |
| 19:00.16 | Essobi | mehe |
| 19:00.28 | `Sauron | ess oh bee |
| 19:00.30 | Essobi | He broxored jooo |
| 19:00.41 | Essobi | Wha-a-at. :) |
| 19:00.41 | km- | we'll check this shit out |
| 19:00.45 | `Sauron | nuttin |
| 19:00.51 | `Sauron | how's the weather there? |
| 19:00.55 | Essobi | You just figure out what my name meant? :) |
| 19:01.14 | km- | of course there are like 6 lines active |
| 19:01.18 | Essobi | Ehh.. It's cold.. and I'm going to go kick a dog. |
| 19:01.20 | tzanger | km-: it might not be right htough |
| 19:01.21 | km- | bastards always gotta be on the phone when I want to fix shit |
| 19:01.26 | tzanger | you see it's also checking if it can call progress |
| 19:01.27 | Essobi | lol |
| 19:01.28 | km- | tzanger: well, we'll pray to god it is |
| 19:01.30 | tzanger | and E&MWINK may set it already |
| 19:01.32 | *** join/#asterisk TheBear (~brif8@lazyjtrainingcenter.com) |
| 19:01.38 | tzanger | so it may do EERIE THINGS to your calls |
| 19:01.44 | km- | hahaha |
| 19:01.45 | km- | wonderful |
| 19:01.53 | km- | so maybe I should be doing this as an evening maintenance event |
| 19:01.57 | tzanger | km-: you might want to say fuck it and change the next line instead |
| 19:02.01 | `Sauron | hum |
| 19:02.07 | km- | too late, I'm already compiling |
| 19:02.08 | `Sauron | weather here's awesome |
| 19:02.10 | Essobi | maha |
| 19:02.13 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 19:02.13 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 19:02.16 | `Sauron | ~weather KATT |
| 19:02.17 | *** join/#asterisk darius_ (darius@c-67-171-44-101.client.comcast.net) |
| 19:02.18 | tzanger | instead of ast_setstate(ast, AST_STATE_RINGING) you might want to say AST_STATE_UP |
| 19:02.25 | Essobi | Nice. |
| 19:02.26 | darius_ | alright, I thought someone said Teliax had good customer service? |
| 19:02.32 | km- | brr |
| 19:02.35 | km- | ~weather KNXX |
| 19:02.45 | `Sauron | Brr |
| 19:02.53 | `Sauron | poor km |
| 19:02.57 | km- | tzanger: that sounds like a better solution |
| 19:03.00 | TheBear | what is the best termination provider for TX and/or AR ? I'm wanting to terminate my long distance calls via * and VoIP. most of my long distance calls are dest for TX or AR. |
| 19:03.12 | `Sauron | thebear: bv |
| 19:03.19 | `Sauron | shrug |
| 19:03.31 | `Sauron | $9.99 and you get free nationwide calling |
| 19:03.33 | Qwell | ~weather KONT |
| 19:03.37 | `Sauron | or something like that |
| 19:03.44 | Qwell | mmm, sunny southern CA |
| 19:03.49 | TheBear | `Sauron: got a website or something ? |
| 19:03.55 | `Sauron | www.broadvoice.com |
| 19:04.03 | `Sauron | their van was at VON ;) |
| 19:04.07 | km- | ITS 61 IN ONTARIO?!?!?! |
| 19:04.11 | km- | and it's 37 here in philly?! |
| 19:04.12 | Qwell | km-: California |
| 19:04.12 | km- | bah |
| 19:04.16 | km- | Oh. |
| 19:04.18 | km- | hahahahaa. |
| 19:04.19 | Qwell | heh |
| 19:04.24 | tzanger | km-: your billing will likely go to shit though |
| 19:04.38 | km- | tzanger: why do you say that? |
| 19:04.39 | tzanger | since asterisk will no longer have any way to know the difference between an unanswered and answered call |
| 19:04.50 | tzanger | since AST_STATE_RINGING will no longer exist |
| 19:04.58 | Qwell | nice 25mph winds here too |
| 19:05.05 | *** join/#asterisk emitrax (~emitrax@host140-51.pool80181.interbusiness.it) |
| 19:05.12 | jontow | ~weather KMSS |
| 19:05.36 | Dandan | ~weather kbdl |
| 19:05.38 | km- | tzanger: hmm. Who do you think is the right person to try and fix this problem? I tried getting kram to look at it but he's intent on me using digium tech support |
| 19:05.44 | jontow | big difference.. that is 2hrs from the dorval airport in Montreal :) |
| 19:06.12 | `Sauron | I <3 ONT |
| 19:06.17 | km- | I simply cant imagine that I'm the only person in asterisk world who has an e&m wink setup |
| 19:06.22 | km- | this sounds like a global problem to me |
| 19:06.23 | `Sauron | makes getting to joshua tree wasier |
| 19:06.26 | TheBear | `Sauron: do you use them ? |
| 19:06.32 | tzanger | km-: well use that channel but it is clear (to me anyway) that asterisk's E&M Wink support is only one-way at this point in time |
| 19:06.33 | `Sauron | TheBear: yes |
| 19:06.45 | *** join/#asterisk Aviator (~ask@ip-65-111-77-10.customer.accelacom.net) |
| 19:06.47 | Essobi | ~weather ksdf |
| 19:06.48 | Qwell | `Sauron: Yeah, LAX is much too far away for that |
| 19:06.54 | tzanger | jontow: perhaps but that's EASTERN ontario... <shudder> :-) |
| 19:07.02 | Qwell | joshua tree...heh, might as well land in Vegas. :p |
| 19:07.02 | `Sauron | Qwell: We're talking 3 hours difference in driving time |
| 19:07.04 | `Sauron | if not more |
| 19:07.07 | Qwell | yeah, easily |
| 19:07.17 | TheBear | `Sauron: what's the quality and service like ? |
| 19:07.18 | `Sauron | I drove lax -> josh once |
| 19:07.19 | Qwell | Damn the 91 |
| 19:07.21 | km- | tzanger: how could someone implement one side and claim it's done? That's pretty crappy shit! |
| 19:07.23 | `Sauron | 5 hours |
| 19:07.30 | `Sauron | then I drove josh -> long beach |
| 19:07.32 | km- | heh |
| 19:07.33 | `Sauron | 2 hours 10 minutes |
| 19:07.36 | Qwell | yeah |
| 19:07.41 | Qwell | the 91 east is retarded |
| 19:07.43 | `Sauron | I was following someone who knew the roads :) |
| 19:07.43 | km- | asterisk is stable, as long as you dont use any of its addons! |
| 19:07.53 | Essobi | lol |
| 19:07.56 | `Sauron | TheBear: I haven't had any problems with them yet |
| 19:08.05 | Essobi | E&M support is incomplete? |
| 19:08.07 | km- | essobi: heh, pbx.c is hella stable! |
| 19:08.07 | Essobi | D'oh. |
| 19:08.12 | km- | essobi: thats what it appears to be |
| 19:08.14 | `Sauron | It's worth noting that I was passed by CHP while I was going 85 |
| 19:08.17 | Essobi | Nice.. |
| 19:08.22 | Qwell | yeah, that happens |
| 19:08.26 | km- | essobi: what's weird though is, we cant be the only two people in the world trying to do e&m wink on asterisk |
| 19:08.32 | Essobi | Well.. don't feel bad.. I can only get half of H323 working too. :) |
| 19:08.41 | `Sauron | wink wink |
| 19:08.44 | `Sauron | I hate winkstart |
| 19:08.52 | Essobi | BUY ME A DRINK |
| 19:08.55 | Essobi | heh.. |
| 19:08.56 | km- | haha |
| 19:08.58 | jontow | "Please accept my signalling.. *wink wink*" |
| 19:09.03 | Essobi | 302 MOVED |
| 19:09.12 | Qwell | `Sauron: I don't know if the speedometer was broken, or what...but on the way to work one morning, the flow of traffic was 100 |
| 19:09.14 | `Sauron | if you ever come to aus, I might |
| 19:09.16 | km- | TE405p is about to be RTS'd |
| 19:09.37 | Essobi | pssh |
| 19:09.42 | Essobi | mail it to me |
| 19:09.44 | km- | haha |
| 19:09.52 | `Sauron | Qwell: That would be nice |
| 19:10.00 | `Sauron | I bet your speedo was broken, though :) |
| 19:10.02 | km- | I can't switch the signalling on the NEC system without getting a site visit |
| 19:10.13 | tuxinator_linux | speedo? |
| 19:10.15 | km- | and besides, PRI will incur an extra $100 a month |
| 19:10.16 | Essobi | It's a NEC pbx? |
| 19:10.20 | km- | essob: yeah |
| 19:10.24 | Essobi | ouch |
| 19:10.36 | Qwell | `Sauron: dunno, it always worked before and after that day |
| 19:10.38 | km- | the damn thing works fine |
| 19:10.50 | km- | I'm actually surprised at how much better it's taking this situation than asterisk is.. |
| 19:10.59 | tzanger | km-: I agree with you |
| 19:11.01 | herag | for the asterisk-users list at lists.digium.com, does the moderator have to accept your subscription? or is it just email confirmation |
| 19:11.15 | `Sauron | too bad a friend of mine's male.. I'd marry him on the spot |
| 19:11.21 | Eight | herag: the latter. |
| 19:11.22 | `Sauron | he made me this killer chocolate pie |
| 19:11.24 | Essobi | Man.. H323 is pissing me off.. from * to my softswitch works fine.. two way DTMF and all.. from my soft to my *.. I get a media transport error.. GRR |
| 19:11.28 | `Sauron | as thanks for fixing some stuff |
| 19:11.30 | `Sauron | Mmmmmmmmmmm |
| 19:11.32 | tzanger | receiving E&M wink though I bet is not trivial, but yes I agree |
| 19:11.41 | jlewis | anyone ever been flooded with "chan_zap.c:4182 zt_write: Unable to set linear mode on channel 10" or know how to stop it other than perhaps killing * and unloading/reloading the zap modules? |
| 19:11.41 | km- | tzanger: yeah, I'm rather miffed... Ah well, c'est la vie |
| 19:11.44 | tzanger | km-: does this "fix" it though? |
| 19:11.48 | tzanger | maybe I'm enetirely wrong |
| 19:11.51 | km- | I wont know till I restart asterisk |
| 19:12.03 | km- | still one caller |
| 19:12.10 | *** join/#asterisk Juxt (~Juxt@adsl-068-213-216-087.sip.bct.bellsouth.net) |
| 19:12.15 | Juxt | good afternoon |
| 19:12.32 | Essobi | zapbarge and tell him to get off the damn phone, he's burning up your dime. ;) |
| 19:12.33 | herag | Eight: is there any reason that I wouldn't have received my email confirmation yet? I tried to subscripbe last night |
| 19:12.34 | `Sauron | juxt-re-position |
| 19:12.40 | tzanger | km-: I mean looking at that it goes into RINGING state but then there's no code to take the answer with wink |
| 19:12.42 | Essobi | booooo booo |
| 19:13.04 | agave-txlink | <km-> essobi: what's weird though is, we cant be the only two people in the world trying to do e&m wink on asterisk |
| 19:13.08 | km- | tzanger: yeah, I'll try and get a short window this afternoon to give it a shot |
| 19:13.10 | agave-txlink | I have a customer who is doing E/M wink and it works. |
| 19:13.13 | Juxt | has anyone done a hosted PBX with asterisk? |
| 19:13.17 | km- | agave: both sides? |
| 19:13.19 | Juxt | i am trying to wrap my head around this |
| 19:13.20 | MikeJ_ | is anyone else experiencing a proble with arithmatic expressions in recent head.... SetVar(i=$[${i} + 1]); is not working for me... |
| 19:13.21 | Essobi | agave-txlink winking on both sides? |
| 19:13.21 | agave-txlink | km: yes |
| 19:13.26 | agave-txlink | harris 20-20 switch to asterisk |
| 19:13.33 | km- | heh, riddle me this |
| 19:13.33 | agave-txlink | actually I take it back i think they have it set up as FGD |
| 19:13.38 | MikeJ_ | it is returning [1 + 1] instead of 2 |
| 19:13.40 | km- | how does it wink if there's no code to do so |
| 19:13.41 | km- | heh |
| 19:13.47 | Essobi | ;) |
| 19:13.53 | Essobi | Pun intended. |
| 19:13.56 | tzanger | km-: please do yourself a favour |
| 19:13.57 | km- | ROFL |
| 19:14.11 | agave-txlink | just get a PRI |
| 19:14.13 | agave-txlink | :) |
| 19:14.15 | km- | tzanger: buy life insurance? See a psychiatrist? :) |
| 19:14.21 | tzanger | km-: keep your old chan_zap.so around |
| 19:14.28 | tzanger | km-: and just overwrite it with the new one |
| 19:14.28 | km- | PRI is not viable unfortunately |
| 19:14.30 | TheBear | how well does vonage and * work together |
| 19:14.31 | km- | tzanger: very good ida |
| 19:14.32 | Essobi | I need a smoke.. |
| 19:14.39 | tzanger | then restart *... if it works, great... if not, copy the old one back over and restart quickly |
| 19:14.45 | tzanger | you don't need to do a full reinstall for htis |
| 19:14.47 | tzanger | just change the .so |
| 19:14.52 | Eight | TheBear: doesn't, as far as I know. |
| 19:15.00 | tzanger | hell you could reload chan_zap.so without restarting * even |
| 19:15.08 | Juxt | is there a possibility of binding various contexts to various IPs? |
| 19:15.18 | Eight | TheBear: might be able to hack something together doing some network sniffing. |
| 19:15.21 | km- | not with open channels |
| 19:15.22 | km- | can I? |
| 19:15.29 | MikeJ_ | anyone? |
| 19:15.36 | km- | eh, I wont even give that a shot |
| 19:15.40 | TheBear | Eight: which termination provider do you use (if any) ? |
| 19:15.43 | km- | last thing I need is bitching people |
| 19:15.46 | Eight | TheBear: Broadvoice. |
| 19:15.48 | *** join/#asterisk juanjoc (~jcomellas@200.73.189.82) |
| 19:15.54 | tzanger | km-: noooooooooooooo |
| 19:15.58 | tzanger | km-: not with open channels |
| 19:16.05 | tzanger | but when it goes quiet |
| 19:16.12 | Eight | TheBear: you'll want 1.0.6+ and to check the BV settings page on the wiki. |
| 19:16.20 | tzanger | copy, reload chan_zap.so, test and be ready to copy and reload again :-) |
| 19:16.22 | Eight | TheBear: With that, should work flawlessly |
| 19:16.43 | juanjoc | Hi, does anyone here have any knowledge of the IAXclient code? |
| 19:16.50 | TheBear | Eight: what about the CVS version ? |
| 19:16.55 | tzanger | juanjoc: very very little, but stevekstevek wrote the damn thing :-) |
| 19:17.11 | Juxt | libiax |
| 19:17.29 | juanjoc | Thanks, I'll see if he's available. |
| 19:17.30 | Eight | TheBear: No idea. |
| 19:17.55 | *** part/#asterisk fjoe (~fjoe@samodelkin.net) |
| 19:18.21 | tzanger | juanjoc: he was to be in meetings all afternoon but you might get lucky |
| 19:19.15 | *** join/#asterisk topping (~topping@dhcp024-210-082-196.columbus.rr.com) |
| 19:19.21 | juanjoc | No, apparently he's not available. |
| 19:19.54 | TheBear | will BV give me Little Rock Number (I see they service Little Rock). |
| 19:20.11 | *** part/#asterisk Juxt (~Juxt@adsl-068-213-216-087.sip.bct.bellsouth.net) |
| 19:20.23 | Qwell | TheBear: sales@broadvoice.com |
| 19:20.30 | *** join/#asterisk spackle (~spackle@209.234.83.19) |
| 19:20.39 | juanjoc | It seems that the app_conference application in the iaxclient site only works with Linear PCM, GSM and u-Law codecs. |
| 19:20.49 | *** join/#asterisk DJ-Pyro_ (~DJ-Pyro@firewall.brevient.net) |
| 19:21.04 | DJ-Pyro_ | anyone mess around with the digitmap settings for the polycom 500/600 phones? |
| 19:21.30 | *** join/#asterisk widowlicker (~Naturalbl@62.77.178.121) |
| 19:21.47 | widowlicker | hi |
| 19:22.18 | widowlicker | i was wondering if anyone could give us a hand with configuring a sip connection using amp |
| 19:22.26 | widowlicker | sorry if this is the wrong channel |
| 19:22.39 | yaaar | widowlicker: well, i've done it once or twice.....what's the trouble |
| 19:22.55 | tzanger | ... widowlicker? what kind of nick is that?! |
| 19:23.10 | widowlicker | i can get it to work wirh calls going out but coming in i just get to the mailbox on my sip proxy |
| 19:23.38 | widowlicker | i tried for windowlicker (you kow aphextwin) but it was already used |
| 19:24.10 | DJ-Pyro_ | nm, think I got it |
| 19:24.34 | km- | .aphex twin! |
| 19:24.36 | DJ-Pyro_ | man, these Hitachi WIP-5000's are so cool |
| 19:24.46 | widowlicker | yep km |
| 19:24.50 | km- | I wasn't all that enthused by the windowlicker cd |
| 19:25.01 | *** join/#asterisk zno (~chatzilla@ip-160-79-174-98.autorev.intellispace.net) |
| 19:25.17 | widowlicker | i thought it was ok |
| 19:25.20 | widowlicker | anyway |
| 19:25.22 | *** part/#asterisk lindi- (~lindi@kulho150.adsl.netsonic.fi) |
| 19:25.26 | widowlicker | here is my setup |
| 19:26.12 | widowlicker | asterisk server (asterisk@home) and a client (xlite). these are both behind a firewall. |
| 19:26.44 | widowlicker | i have set up a sip trunk in amp trunks |
| 19:26.48 | JerJer[mobile] | why not learn asterisk the right way? |
| 19:27.00 | JerJer[mobile] | a gui is going to taint you |
| 19:27.06 | widowlicker | going out emails are fine but nothing happening coming in |
| 19:27.26 | widowlicker | i dont have time at the moment to learn it from config files |
| 19:27.49 | widowlicker | i am trying to show my bodss that if it works rigth we can replace the pbx we have here |
| 19:28.06 | Essobi | Hey jer.. You mind looking at a h323 debug and tell me what's going stupid? |
| 19:29.00 | *** join/#asterisk Derkommissar (~alberto@66.64.215.7.nw.nuvox.net) |
| 19:29.34 | widowlicker | do you want me to run h323 debug |
| 19:30.18 | Dandan | BV: - Got SIP response 404 "Not Found" back from 147.135.0.128 |
| 19:30.22 | Dandan | Anyone can help?!? |
| 19:30.48 | tzanger | widowlicker sounds midly pornographic |
| 19:30.50 | Derkommissar | Does the TE110P do R2 signaling with e1's ? |
| 19:30.54 | Essobi | JerJer[mobile] I see a clearcall cause code of 7 and mediaTransportFailure |
| 19:31.12 | widowlicker | i know it does but it was funny |
| 19:31.20 | Essobi | tzanger could be worse.. widowtickler |
| 19:31.26 | Essobi | windowlicker |
| 19:31.26 | Qwell | windowlicker is an Aphex Twin song, or something |
| 19:31.32 | widowlicker | yep |
| 19:31.36 | Essobi | Heh |
| 19:31.38 | Qwell | disturbing video |
| 19:31.39 | km- | it's an album |
| 19:31.48 | widowlicker | and a song name |
| 19:31.55 | tzanger | km-: have you tested yet? |
| 19:31.57 | km- | I enjoyed I care because you do |
| 19:31.59 | widowlicker | video is very funny |
| 19:32.06 | km- | tzanger: still open lines |
| 19:32.16 | tzanger | tell them to get off the fucking phone, you're trying to work! :-) |
| 19:32.18 | km- | tzanger: I'm going to be waiting for a while here |
| 19:32.20 | km- | hahaha |
| 19:32.36 | Dandan | anyone able to help me with that BV problem?!? |
| 19:32.53 | Essobi | zapbarge and conference a 1-900- in.. that'll get'm off the phone. |
| 19:32.59 | *** join/#asterisk loick (~loick@APuteaux-151-1-20-96.w82-124.abo.wanadoo.fr) |
| 19:33.01 | km- | hahahhaha |
| 19:33.04 | tzanger | km-: yeah the E&M thing looks bad |
| 19:33.16 | Derkommissar | :-/ |
| 19:33.20 | tzanger | km-: I mean chan_zap sends the ZT_DIAL ioctl without waiting for a wink |
| 19:33.24 | tzanger | so it's a chan_zap bug IMO |
| 19:33.33 | tzanger | chan_zap should wait for the wink beore dialing |
| 19:33.51 | km- | if I put it in mantis |
| 19:33.57 | km- | will you bugnote a corroboration? |
| 19:34.05 | rvhi | in voicemail, if user presses a # to stop recording, the # key is recorded in the wav file |
| 19:34.12 | rvhi | is there a way to get rid of it? |
| 19:34.21 | Essobi | yea.. don't press # |
| 19:34.47 | tzanger | km-: yes |
| 19:35.02 | rvhi | haha, the intro says press # when finish |
| 19:35.03 | tzanger | just basically enter a bug saying that chan_zap does not wait for wink when in E&M Wink signaling before dialing |
| 19:35.09 | bjohnson | damn sixtel still hasn't got me my Kitchener DID .. been a month |
| 19:35.18 | tzanger | bjohnson: that sucks, I got mine relatively fast |
| 19:35.31 | tzanger | they gave me a London one first though, but within a week I had that all corrected |
| 19:35.35 | bjohnson | I got my London one through them in about a week |
| 19:35.45 | bjohnson | maybe I got yours |
| 19:35.48 | Derkommissar | :-/ |
| 19:36.29 | tzanger | haha |
| 19:36.54 | km- | tzanger: I gotta download full head first and give it a shot |
| 19:37.05 | km- | tzanger: just in case some other weird oddities are causing the problem |
| 19:37.26 | *** join/#asterisk parf3 (~parf@paytone-dvp.patientkeeper.com) |
| 19:37.37 | tzanger | ok |
| 19:37.55 | km- | later tonight I'll test it from home |
| 19:38.08 | km- | tzanger: I appreciate all the help, btw |
| 19:38.15 | tzanger | km-: no worries, this is crazy |
| 19:38.17 | parf3 | Hi, can someone send me a link with good explaination how DID assignent work? |
| 19:38.31 | tzanger | parf3: what do you mean, there is no magic |
| 19:38.36 | tzanger | the telco will route DIDs to your PRI |
| 19:38.43 | tzanger | they appear in your dialplan |
| 19:38.46 | tzanger | pretty straightforward |
| 19:39.06 | parf3 | How did they do route? What is a PRI? |
| 19:39.17 | parf3 | I am somewhat begginer |
| 19:39.20 | tzanger | parf3: ahh |
| 19:39.29 | xkev | has anyone figured how to get the 1.4.1 polycoms to actually do the check-cfg & reboot? |
| 19:39.46 | xkev | they changed stuff and didn't document it sufficiently (surprise surprise) |
| 19:40.23 | xkev | with voipProt.SIP.specialEvent.checkSync.alwaysReboot="1" it checks more files, but never <mac>.cfg and never reloads |
| 19:40.33 | Skysky | excuse me, if i don want any other zaptel modules other than ztdummy, can i just get rid of the rest in the Makefile? |
| 19:40.34 | tzanger | parf3: well think of a PRI as 23 phone lines on a single cable |
| 19:40.35 | xkev | ..without that it checks only the *.ld images |
| 19:40.48 | tzanger | parf3: except you can have hundreds and even thousands of phone numbers appear on it |
| 19:41.09 | tzanger | there are 23 voice "circuits" and a call can appear on any of them |
| 19:41.38 | tzanger | unlike a regular telephone line where there is only one number assigned to it |
| 19:42.23 | zno | variable ${EXTEN:1} gets rid of the first digit, is there a way to get rid of the last, for example a pound symbol? |
| 19:42.36 | tzanger | zno: yes but I don't remember how |
| 19:42.39 | parf3 | How do PRI related with internet. as i understand i can just buy some DIDs |
| 19:42.52 | tzanger | parf3: you buy DIDs from VOIP providers |
| 19:43.06 | tzanger | the DID is actually on their PRIs and they will route the call to you over the internet |
| 19:43.09 | zno | tzanger: if you don't remember, I assume it's some kind of manual string handling, instead of using the global variable |
| 19:43.55 | tzanger | zno: could be, I've never had to do it |
| 19:44.03 | tzanger | in the old days you'd use Cut() I think |
| 19:44.26 | parf3 | Are there any difference if i buy bunch of DID from Level3 or from LiveVoip? |
| 19:44.33 | *** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com) |
| 19:45.26 | *** join/#asterisk ACiDV (~joel@iteckGW.infoteck.qc.ca) |
| 19:45.38 | MikeJ_ | hey tzanger |
| 19:46.21 | tzanger | MikeJ_: |
| 19:46.29 | MikeJ_ | http://pastebin.ca/7420 |
| 19:46.34 | RickTick | where does someone ..report an * bug? Digium? |
| 19:46.40 | tzanger | parf3: not really -- I wasn't aware, though, that Level3 was a VOIP provider |
| 19:46.41 | ACiDV | Hi... if I want to use only port 2 of a TE405, I set zaptel.conf like this: unused = 1-24 ... span=2,1,0,esf,b8zs ? or span=1 ? does span=x where x = the physical port ? |
| 19:46.44 | MikeJ_ | shouldn't the 2nd one be 2? |
| 19:46.51 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 19:47.05 | *** join/#asterisk Darwin_35 (~Darin@c-24-3-226-147.client.comcast.net) |
| 19:47.17 | tzanger | wow |
| 19:47.20 | tuxinator_linux | Congrats |
| 19:47.21 | tzanger | congrats shmaltz ! |
| 19:47.25 | shmaltz | thanks guys |
| 19:47.25 | Essobi | congratz |
| 19:47.30 | tzanger | what are the particulars? |
| 19:47.32 | shmaltz | It's a girl |
| 19:47.35 | tzanger | MikeJ_: oh jeez, I have no idea |
| 19:47.35 | Essobi | boy or a girl? |
| 19:47.41 | tuxinator_linux | 4 or 5 toes? |
| 19:47.42 | Essobi | Heh.. |
| 19:47.42 | shmaltz | 6 lbs and 8 oz |
| 19:47.45 | tzanger | that is awesome shmaltz :-) |
| 19:47.46 | aminorex | congratulations on not having a boy!!! |
| 19:47.51 | MikeJ_ | well.. damn |
| 19:47.54 | aminorex | girls rock |
| 19:47.56 | tzanger | oh she's a little one... my daughter was 7lb 7oz |
| 19:47.59 | Essobi | aminorex It's hell either way. |
| 19:48.01 | shmaltz | tuxinator_linux, if you count it with Fedora Core then 4 toes |
| 19:48.03 | shmaltz | :) |
| 19:48.18 | tzanger | Essobi: I disagree... my baby girl is FAR more work ot raise than my two sons combined |
| 19:48.19 | tuxinator_linux | funny guy |
| 19:48.25 | tzanger | Essobi: I wouldn't trade her for anything in the world though |
| 19:48.35 | shmaltz | I told the Doc I was very happy it's a girl b/c that way it will not go thru 9 months that I went thru (i'm male :) |
| 19:48.46 | Essobi | Nah.. they're different in both their own ways. |
| 19:48.53 | tzanger | shmaltz: huh? |
| 19:48.58 | parf3 | tzanger, for example i bought 100 DIDs for my company, they are somehow connected to some switch, if later i want to change switch, how i can do it? Do i need to contact once more DIDs issuer? |
| 19:49.00 | Essobi | Just hell on wheels at different ages. :) |
| 19:49.22 | shmaltz | I still didn't sleep from 3 days ago. tzanger |
| 19:49.29 | shmaltz | tzanger you female? |
| 19:49.43 | tzanger | parf3: they're not connected to a switch, they're on a PRI somewhere |
| 19:49.50 | tzanger | shmaltz: nope I've got an outie |
| 19:49.56 | Essobi | hah |
| 19:50.04 | shmaltz | so why u asking huh? |
| 19:50.17 | ACiDV | If I set unused for port 1, 3 & 4 and I set span=2, then I have a RED alarm on port 1 and port 2 is OK but no cable connected :| |
| 19:50.29 | tzanger | shmaltz: I don't understand your statement "you're happy you've got a daughter because that way she won't go through 9 months that you went through" |
| 19:51.00 | tzanger | parf3: explain what you have *right now* and I'll try to understand :-) |
| 19:51.01 | shmaltz | when she is pregnant |
| 19:51.02 | km- | huh? |
| 19:51.18 | tzanger | ... I still odn't quite get it |
| 19:51.19 | shmaltz | cmon guys, if you don't get it forget it |
| 19:51.29 | tzanger | you're happy you have a daughter so that she won't have to go through being an expectant father? |
| 19:51.36 | shmaltz | nope |
| 19:51.39 | km- | tzanger: Yeah, I think it's this newfangled humor, us old farts wouldnt get it? |
| 19:51.47 | tzanger | km-: hahaha could be |
| 19:52.00 | tzanger | I love being a father... hardest 30 seconds of my life... :-) |
| 19:52.02 | km- | haha, when I was your age we didn't have 3d ultrasounds! |
| 19:52.05 | km- | ROFL |
| 19:52.10 | km- | gave it your all there, huh |
| 19:52.14 | tzanger | hahaha |
| 19:52.24 | tzanger | actually I've got three kids |
| 19:52.28 | tzanger | two boys and a girl |
| 19:52.32 | km- | heh, my wife and I are waiting till she's 30 |
| 19:52.35 | MikeJ_ | poor kids :) |
| 19:52.37 | *** part/#asterisk chetan (freetibet@24-193-188-21.nyc.rr.com) |
| 19:52.37 | tzanger | MikeJ_: :-) |
| 19:52.38 | shmaltz | well, even when my mother was pregnant with me they already had us |
| 19:52.39 | km- | we're just buying our first house now |
| 19:52.40 | km- | rofl |
| 19:52.53 | tzanger | shmaltz: 3D ultrasound, not traditional ultrasound |
| 19:52.56 | tzanger | km-: congrats |
| 19:53.01 | tzanger | that's a bundle of laughs |
| 19:53.02 | km- | thanks, we're pumped |
| 19:53.10 | km- | yeah, the mortgage group is especially cheery |
| 19:53.15 | km- | "lets take your money and screw you on the side!" |
| 19:53.16 | tzanger | everything you learn about AFTERward |
| 19:53.26 | shmaltz | tzanger , exactly " love being a father... hardest 30 seconds of my life... :-)" |
| 19:53.28 | km- | yeah exactly, I just want the damn house |
| 19:53.38 | km- | http://www.fierymoon.com/~pgrace/gallery |
| 19:53.41 | km- | there's house pics there |
| 19:53.46 | km- | nice 4bdroom, $150 |
| 19:54.00 | parf3 | tzanger, i want to create vonage like company, we already have some customer base and termination partners, some company offer their switch services, we want to start with this company, but later buy our own equipment, we are worring about trasferring our customers from someones switch to ours in future |
| 19:54.20 | tzanger | parf3: you need to put down the keyboard and run, not walk, to the employment center |
| 19:54.29 | tzanger | you need to hire a telco guy and a VOIP guy |
| 19:54.42 | tzanger | km-: nice |
| 19:54.45 | eKo1 | so they can kill each other |
| 19:54.47 | parf3 | Anyone in Boston area? |
| 19:54.52 | shmaltz | tzanger, very funny that answer to parf3 |
| 19:55.03 | shmaltz | km-, nice house |
| 19:55.03 | tzanger | www.mixdown.ca/~andrew/photos/House |
| 19:55.06 | tzanger | that's my place |
| 19:55.27 | km- | tzanger: you have a barn? That's quality! |
| 19:55.28 | shmaltz | tzanger, where is that? what state? |
| 19:55.32 | tzanger | haha |
| 19:55.34 | tzanger | Ontario, Canada |
| 19:55.47 | shmaltz | nice, tzanger |
| 19:56.11 | km- | I love it... cool place |
| 19:56.32 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 19:56.49 | km- | haha, you better get working on that basement |
| 19:56.54 | km- | cant leave that thing half finished! |
| 19:57.04 | *** part/#asterisk Alexis (~alexis@www.trim.it) |
| 19:57.07 | tzanger | km-: yeah... it's all time and money |
| 19:57.39 | Eight | I don't get it... where's the asterisk server? =) |
| 19:57.43 | *** part/#asterisk darius_ (darius@c-67-171-44-101.client.comcast.net) |
| 19:57.45 | km- | haha |
| 19:57.50 | tzanger | Eight: that's not in the pic |
| 19:57.53 | km- | tzanger: how much money have you spent on it so far? |
| 19:57.55 | tzanger | the old part of the basement isn't shown |
| 19:58.00 | tzanger | km-: oh lord |
| 19:58.01 | Eight | tzanger: well it should be =) |
| 19:58.17 | km- | km-: I'm curious because I'm going to be finishing my basement |
| 19:58.22 | km- | whoops |
| 19:58.28 | km- | tzanger: right, what I just said to myself, was meant for you |
| 19:58.30 | km- | :) |
| 19:59.45 | *** part/#asterisk parf3 (~parf@paytone-dvp.patientkeeper.com) |
| 19:59.48 | tzanger | let's see... 2 months after I moved in the insurance comany did an inspection and cancelled my policy becuase of the oil tank... tank guy couldn't fix the tank without doing a furnace inspection whihc revealed a cracked heat exchanger |
| 20:00.12 | tzanger | a 1/8" crack that nobody would weld because they didn't want to be held liable cost me $5000 for a furnace + oil tank + chimney liner |
| 20:01.01 | tzanger | I need to repair the roof too now, but I knew that when I bought it... they repaired **HALF** of it... wtf?? |
| 20:01.07 | tzanger | so probably another couple grand for that |
| 20:01.29 | tzanger | the basement was that way when I bought it (minus all the junk) |
| 20:01.43 | km- | tzanger: shit! |
| 20:01.51 | km- | ah' |
| 20:01.57 | tzanger | so yeah it was an expensive first year |
| 20:02.04 | km- | yeah, our aquaplate is corroding on the oil heater |
| 20:02.14 | tzanger | aquaplate? |
| 20:02.15 | km- | it needs to be replaced and the injectors cleaned because it's burning dirty |
| 20:02.21 | km- | the plate where the water lines come into the burner |
| 20:02.23 | km- | is corroding |
| 20:02.31 | tzanger | there's a water plate in your oil furnace?! |
| 20:02.38 | km- | I gave it the term "aquaplate" for a lack of nomenclature skills |
| 20:02.38 | MikeJ_ | tzanger... I fixed my issue, |
| 20:02.43 | tzanger | oh good, what was it? |
| 20:02.45 | MikeJ_ | I wasn't crazy, it was broken |
| 20:02.59 | km- | tzanger: yeah, the hot water goes through the pipes, where the return line comes in, the joint is corroded |
| 20:03.08 | tzanger | interesting |
| 20:03.16 | tzanger | so it's hot water heating with an oil burner? |
| 20:03.17 | km- | the home inspector says it's probably only $200 all told to fix it |
| 20:03.20 | km- | right |
| 20:03.26 | *** part/#asterisk Moc____ (~mochouina@h66-201-214-109.gtconnect.net) |
| 20:03.37 | km- | and we've got a tankless water heater which is cool |
| 20:04.52 | shmaltz | gtg guys and catch some sleep |
| 20:04.55 | tzanger | ahhhhh okay |
| 20:05.01 | tzanger | shmaltz: I hear ya there... tkae care of your girls |
| 20:05.19 | shmaltz | thanks, tzanger |
| 20:05.21 | tzanger | km-: yeah those things look cool alright, I am curious as to just how efficient they are... they would suck some major power |
| 20:05.58 | km- | ah, this goes through our oil boiler |
| 20:06.02 | km- | so it just costs the gas |
| 20:06.06 | km- | or rather the oil |
| 20:06.16 | tzanger | km-: ahh okay |
| 20:06.45 | *** join/#asterisk Juxt (~Juxt@adsl-068-213-216-087.sip.bct.bellsouth.net) |
| 20:07.04 | Juxt | is there an easy way to record calls that go via asterisk |
| 20:07.27 | modulus_ | cdr |
| 20:07.41 | spackle | I think he means monitor |
| 20:07.45 | ACiDV | record or monitoring ? |
| 20:09.59 | tzanger | km-: I want to play with some solar heating just to augment the oil furnace |
| 20:10.08 | tzanger | and maybe heat recovery on the drains too |
| 20:11.50 | Juxt | i want to do both |
| 20:11.53 | km- | tzanger: just snuck the upstate mod into asterisk |
| 20:11.56 | km- | tzanger: appears to work |
| 20:12.05 | tzanger | km-: yeah but dial a busy number |
| 20:12.11 | km- | hmm |
| 20:12.30 | km- | hmm, how do I make a number busy |
| 20:12.53 | Flash_ | km-: pick up the phone ;-) |
| 20:12.55 | km- | congestion or busy? |
| 20:12.58 | Essobi | FFS |
| 20:12.59 | km- | flash: easy for you to say, home user! |
| 20:13.03 | Essobi | I hate h323 |
| 20:13.11 | heison | tzanger: where in Ontario r u from? |
| 20:13.11 | km- | those of us with T1's and such have a harder time of it! |
| 20:13.18 | km- | tzanger: is congestion fine? |
| 20:14.07 | km- | actually |
| 20:14.11 | km- | thats interesting |
| 20:14.14 | Flash_ | km-: sorry, was just trying to be funny.... |
| 20:14.31 | tzanger | heison: midwestern ontario, just outside the smell town of Listowel |
| 20:14.41 | tzanger | km-: busy |
| 20:14.43 | km- | flash: haha I know, I was jokin with ya too |
| 20:14.54 | tzanger | km-: any call you make asterisk will think has been answered with that hack |
| 20:14.55 | Flash_ | ;-) |
| 20:14.55 | km- | tzanger: interestingly enough, busy detection seems to behave now too |
| 20:15.05 | Essobi | nice |
| 20:15.06 | tzanger | km-: are you using busydetect and callprogress??? |
| 20:15.07 | Juxt | so how do i monitor with asterisk? |
| 20:15.12 | km- | tzanger: so, the worst that can happen is someone sits and listens to a busy signal |
| 20:15.30 | km- | tzanger: negative, I heard they were both wrong to use for em_w |
| 20:15.35 | tzanger | km-: correct |
| 20:16.10 | km- | hmmm |
| 20:16.12 | Essobi | too much damn winking going on here |
| 20:16.13 | km- | first curiosity |
| 20:16.29 | km- | zt_handle_event: Ring/Off-hook in strange state 6 on channel 36 |
| 20:16.58 | tzanger | km-: heh |
| 20:16.59 | tzanger | lovely |
| 20:17.11 | km- | it doesnt seem to be wrong though |
| 20:17.22 | Essobi | not wrong, just strange |
| 20:17.32 | tzanger | haha |
| 20:17.32 | Essobi | heh |
| 20:19.18 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 20:19.29 | *** join/#asterisk Frantic (~ab@TechnologicPartners35.dsl.concentric.net) |
| 20:20.11 | Essobi | I'm really hating H323 softswitches right now |
| 20:20.49 | Hmmhesays | I hate voip in general |
| 20:21.07 | hardwire | suppose maybe you are in the wrong channel then |
| 20:21.15 | *** join/#asterisk _THEEND_ (~DrEaM@80.18.184.226) |
| 20:21.18 | Hmmhesays | well no |
| 20:21.18 | spackle | Hmmhesays, I can top that, I hate phones. lousy user interface. |
| 20:21.38 | epoch | I hate talking to people, does that count? |
| 20:21.40 | Juxt | i hate internet :-) |
| 20:21.41 | Hmmhesays | well i'm pretty much at the burn out stage |
| 20:21.47 | Hmmhesays | time to move on |
| 20:21.51 | tzanger | Hmmhesays: burnout for what |
| 20:22.00 | epoch | gonna be a farmer Hmmhesays? |
| 20:22.01 | Hmmhesays | tech support burnout |
| 20:22.05 | tzanger | Hmmhesays: ahh |
| 20:22.09 | tzanger | I'd love to be a farmer |
| 20:22.14 | _THEEND_ | someone could tell me how to work cisco 7912 with sip? |
| 20:22.15 | tzanger | don't underestimate the amount of work involved in that though |
| 20:22.20 | Hmmhesays | there might be ranching in my future |
| 20:22.21 | epoch | heh, I'm starting a new career in april :) |
| 20:22.26 | epoch | no more tech support! |
| 20:22.34 | tzanger | epoch: oh? giggilo? |
| 20:22.35 | _THEEND_ | why? |
| 20:22.46 | spackle | tzanger: so you could put SIP phones all over your farm? |
| 20:22.48 | Skysky | hi, i wonder if there's a way to dial an extension, then the console will play a sound file and then execute more commands... |
| 20:22.50 | epoch | tzanger: well no ;P |
| 20:22.56 | tzanger | spackle: no, but I do have fiber between the barn and house! |
| 20:22.58 | epoch | tzanger: BI software |
| 20:23.17 | Hmmhesays | Skysky cmd playback |
| 20:23.27 | spackle | tzanger, good, you avoid the grounding difference then. Lightening can be nasty in those situations. |
| 20:23.35 | Skysky | Hmmhesays: playback only plays sound file to caller right? |
| 20:23.35 | tzanger | spackle: yup |
| 20:23.42 | Hmmhesays | who else would it play to ? |
| 20:23.47 | tzanger | spackle: unfortunately power is shared |
| 20:24.05 | spackle | tzanger, did you burn or blow anything up figuring it out? |
| 20:24.08 | *** join/#asterisk Uther_P (~uther_p@66.180.120.83) |
| 20:24.16 | tzanger | spackle: hahah no not yet |
| 20:24.16 | Hmmhesays | unfortunately chan_telepath has not been finished yet |
| 20:24.22 | tzanger | I have all power to my barn shut down at the moment |
| 20:24.28 | Dandan | just got off the phone with BV |
| 20:24.29 | tzanger | completely redoing the electrical there |
| 20:24.39 | Dandan | they changed slightly their authorization scheme |
| 20:24.46 | Skysky | right now i can barely do it by "Dial (CONSOLE/dsp,20,A(soundFile)S(1))" this would call console and make it play a soundFile then hangup, but i want it to continue on .. any suggestion of doing so? |
| 20:24.53 | Juxt | any idea why when i use ilbc sometimes the person on the other end can't hear me but with gsm it always works? |
| 20:25.15 | spackle | Juxt, how do you have reinvite set? |
| 20:25.34 | Juxt | hmm i don't even have reinvite set |
| 20:25.39 | Juxt | what should it be set to? |
| 20:26.48 | spackle | if you set reinvite to no it keeps asterisk in the loop and might negotiate the difference in codecs. |
| 20:26.56 | *** part/#asterisk Grooby (~Grooby@66.160.105.186) |
| 20:27.13 | Juxt | let me try |
| 20:27.30 | spackle | tzanger: more power? |
| 20:28.01 | Juxt | so is ilibc generally better than gsm? |
| 20:28.10 | *** join/#asterisk juco_hd (~vickeryj@216-166-233-146.clec.commercial.madisonriver.net) |
| 20:28.13 | Katty | hmm |
| 20:28.14 | bjohnson | ot question .. I can bash script to look for filenames inside html files with grep .. but how can I move the file to a subdir if the filename is NOT found in the files being grepped? |
| 20:28.24 | spackle | I think people prefer gsm. |
| 20:28.37 | Juxt | so gsm is the de-facto? |
| 20:28.49 | bjohnson | Juxt: ulaw is the defacto |
| 20:29.08 | spackle | It works well for lower-bandwidth applications over the internet. on a LAN, ulaw. |
| 20:29.44 | spackle | Juxt, are you using softphones? |
| 20:29.53 | Juxt | yes, firefly |
| 20:30.04 | Juxt | i am just a beginner thus lame questions |
| 20:30.30 | ACiDV | What are the biggest difference between a vanilla kernel and a fedora kernel ? fedora include, backport, etc some patch that are not normally there ? I'm not sure if switch from FC3 kernel to a vanilla (2.6.11.3) will solve my TE405 problem :( |
| 20:31.05 | spackle | Juxt: I asked the other day, apparently there aren't a lot of hardphones that support gsm, or even ilbc. you will almost always find ulaw, alaw, and g7xx something. what 23 or 29? |
| 20:31.54 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 20:31.54 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 20:31.58 | zoa | heya brian |
| 20:32.05 | zoa | will get you the patches |
| 20:32.06 | zoa | lemme check |
| 20:32.13 | Juxt | gotcha |
| 20:32.21 | bkw_ | zoa ya please ;) |
| 20:32.23 | Juxt | i just enabled ulaw on my system |
| 20:33.31 | spackle | Juxt, if you set canreinvite=no in sip.conf it will make asterisk stay in the loop and if necessary do conversions for the codec. |
| 20:34.00 | Juxt | actually i am using iax phones |
| 20:35.11 | *** join/#asterisk jmhunter (~jmhunter@wire3-148.razzolink.com) |
| 20:35.11 | *** mode/#asterisk [+o jmhunter] by ChanServ |
| 20:35.37 | eKo1 | Hmm...I did a 'sip show channels' and thirty of them have Format = unknow. |
| 20:35.47 | km- | heh |
| 20:35.49 | eKo1 | Why is that? |
| 20:36.07 | km- | tzanger: I'll let this mod stay and see what happens. Looks like it has somewhat rigged the problem |
| 20:36.40 | eKo1 | The the User/ANR appears to be a wrong tel. number. |
| 20:37.13 | tzanger | km-: yeha |
| 20:37.28 | *** join/#asterisk MatsK (~NNSCRIPT@107.80-202-57.nextgentel.com) |
| 20:37.37 | spackle | eKo1: you have stealth codec. |
| 20:37.55 | hardwire | its a spooky one? |
| 20:38.04 | jontow | hmm.. anyone know of a phone that supports ADSI? |
| 20:38.12 | hardwire | www.digium.com |
| 20:38.23 | eKo1 | I think it's due to failed calls. |
| 20:38.50 | eKo1 | Specifically, calls where the caller dialed an incomplete number. |
| 20:38.52 | hardwire | http://www.digium.com/index.php?menu=adsi |
| 20:40.11 | eKo1 | Maybe it's a bug in chan_sip... |
| 20:40.26 | eKo1 | Hmm... |
| 20:40.34 | *** join/#asterisk SPoon_TSX (~SPoon_TSX@24.83.96.211) |
| 20:40.42 | jontow | nice, thank you :) |
| 20:40.48 | deodax | when someone from the outside calls me up he cant hear a ring... whats wrong with that? |
| 20:41.46 | jontow | set the Ringing state :) |
| 20:41.52 | jontow | exten => DID...,1,Ringing |
| 20:41.56 | ACiDV | deodax I have the same problem if I have an exten => xxx,s,Answer |
| 20:42.00 | jontow | exten => DID...,2,Wait(2) |
| 20:42.02 | jontow | ... |
| 20:42.08 | deodax | oic.. |
| 20:42.09 | deodax | really.. |
| 20:42.21 | jontow | yeah, if you're doing the signalling.. you are doing the work ;) |
| 20:42.22 | deodax | because i have 2 extensions in the inside |
| 20:42.29 | deodax | and 1 provider |
| 20:42.38 | SPoon_TSX | Hi there, just wondering if anyone heard about anything on Sayson IP Phone? |
| 20:42.46 | jontow | spoon_tsx; i was just asking about that ;) |
| 20:43.02 | ACiDV | Sayson ? Aastra 480i ? work fine here |
| 20:43.10 | SPoon_TSX | jontow: o, any comments? |
| 20:43.18 | ACiDV | can setup all 18 soft keys but still no support for blf :( |
| 20:43.25 | jontow | none so far.. i haven't tried 'em, was just curious about a phone with ADSI capabilities |
| 20:44.05 | eKo1 | What is ADSI anyways. I never understood it. |
| 20:44.19 | jontow | on-screen menus |
| 20:44.24 | jontow | terminal capabilities, more or less :) |
| 20:44.30 | SPoon_TSX | eKo1: It is the way how the PBX talk to your Phone to display some information or features. |
| 20:44.43 | opus___ | whats the advantages of ADSI |
| 20:44.44 | *** join/#asterisk vinding (~vinding@c-24-98-150-21.hsd1.ga.comcast.net) |
| 20:45.04 | jontow | i was thinking about using it in a call-center environment.. ie. multiple agents who support multiple clients, have the branding information ring through on an ADSI menu |
| 20:45.08 | SPoon_TSX | opus___: It can make your phone set feature button usable with Asterisk. |
| 20:45.10 | jontow | i suppose it could be done in caller-id |
| 20:45.19 | eKo1 | So ADSI is to PBX what SS7 is to PSTN? |
| 20:45.25 | vinding | Hi all! new user, love it. Just paying respects :p |
| 20:45.34 | denon | eKo1: no .. |
| 20:45.39 | denon | ADSI is a scripting language for phones |
| 20:45.50 | SPoon_TSX | Agree with denon! |
| 20:45.55 | eKo1 | I though it was a protocol. |
| 20:46.03 | SPoon_TSX | eKo1: nope. |
| 20:46.20 | denon | its a language, really .. though the way it's transmitted to the phone, I suppose, could be considered .. kind of a protocol |
| 20:47.13 | SPoon_TSX | Anyone here from Vancouver?? |
| 20:47.17 | *** part/#asterisk juco_hd (~vickeryj@216-166-233-146.clec.commercial.madisonriver.net) |
| 20:47.31 | cypromis | a protocol is a language |
| 20:47.31 | *** join/#asterisk Matt-E- (~Matt-E-@66-224-125-137.atgi.net) |
| 20:47.39 | vinding | I am trying to compile h323 support in @home 0.6 - i'm not super hardcore and I am not sure how to get the right version of pwlib that is required - advice? |
| 20:47.48 | eKo1 | cypromis: eh, no. |
| 20:48.08 | eKo1 | Man, my 'sip show channels' list is just getting bigger and bigger. |
| 20:48.10 | SPoon_TSX | So, no one really try the Sayson 480i yet?? |
| 20:48.11 | Corydon-w | Advice is don't use H.323 |
| 20:48.14 | denon | cypromis: Computer Science. A standard procedure for regulating data transmission between computers. |
| 20:48.20 | cypromis | a protocol is a semantic definition of an exchange of information |
| 20:48.34 | cypromis | so where does that differe from a language ? |
| 20:48.39 | vinding | heh.. i have 4 h323 phones I'd love to use around here... i like my sip phones better too, but i figured i'd attack the 323 beast.. |
| 20:48.59 | Uther_P | cypromis: write a program in a protocol |
| 20:49.16 | eKo1 | cypromis: That's a grammar. |
| 20:49.17 | jontow | oh.. oh wow, 'externnotify' in voicemail.conf is gonna be pretty handy for me :) |
| 20:49.18 | cypromis | write a program in swahili |
| 20:49.41 | Uther_P | cypromis: everyone here is talking about computers but you |
| 20:49.44 | Uther_P | heheh |
| 20:49.44 | cypromis | Uther_P: that just states that it has a limited syntax |
| 20:50.07 | ACiDV | Spacebar yes, I have it at home and work |
| 20:50.10 | denon | we all know there are lots of definitions of "protocol". The one we're current discussing is a communication |
| 20:50.14 | ACiDV | oops not Spacebar byt SPoon_TSX |
| 20:50.17 | eKo1 | And anyways, protocols aren't necessarily used to exchange information. |
| 20:51.06 | SPoon_TSX | ACiDV: How does it work? How is the quality?? Are they reliable? Does it need any reboot periodically?? |
| 20:51.34 | jontow | i love the technical people around here.. one just walked by and looked at my desk and made a comment about a junk sale |
| 20:51.35 | jontow | bah. |
| 20:52.26 | *** join/#asterisk firestrm (F1r3570rm@S010600047577bccd.gv.shawcable.net) |
| 20:52.36 | ACiDV | I have to reboot it 1-2 times a week because all freeze, very easy tftp config files... the problem is that if you want to create a 3-conference, you must set 2 lines |
| 20:52.43 | eKo1 | jontow: I think that person was just bored. |
| 20:53.13 | ACiDV | Does exist other phone that snom that support BLF ? |
| 20:53.16 | jontow | no.. he's a redneck :) |
| 20:53.18 | firestrm | questio... is the use of txgain and rxgain valid within sip.conf connection definitions? |
| 20:53.27 | jontow | he also made a comment about cars in the back yard :) |
| 20:53.37 | eKo1 | firestrm: no. |
| 20:53.59 | firestrm | eKo1, so it only works in zapata.conf? |
| 20:54.25 | eKo1 | I've only seen it in zapata.conf. |
| 20:54.39 | zoa | new jitter buffer patches and astertest load tester on www.astertest.com now |
| 20:54.46 | tzanger | nice |
| 20:54.47 | zoa | [/end newsflash] |
| 20:54.47 | firestrm | hmm.. bummer.. i need to set gain in sip channels |
| 20:54.51 | eKo1 | You can grep all the *.conf.sample files and find out for yourself though. |
| 20:54.52 | tzanger | i've been running hte new jb stuff for about a month |
| 20:55.02 | zoa | thats the iax2 code |
| 20:55.06 | zoa | now its also there for sip |
| 20:55.22 | zoa | oh tzanger in the sip code there is a patch for a memleak in the iax2 code |
| 20:55.34 | zoa | we sent it to stevek too but he didnt fix it yet i think |
| 20:55.35 | tzanger | zoa: mem leak? |
| 20:55.38 | Essobi | Hmm.. I tracked the h323 code that is failing.. now I have to figure out WTF it means.. :| |
| 20:55.39 | tzanger | I've been running for... |
| 20:55.46 | zoa | there is a memory leak in there |
| 20:55.58 | tzanger | *CLI> show uptime |
| 20:55.58 | tzanger | System uptime: 2 weeks, 16 hours, 30 minutes, 3 seconds |
| 20:55.59 | tzanger | Last reload: 1 week, 8 hours, 5 minutes, 7 seconds |
| 20:56.12 | tzanger | I read about it in the bugtracker but haven't hit it yet :-) |
| 20:56.19 | zoa | http://astertest.com/downloads/sip%20jitter%20buffer/jitterbuf.patch3-memleak-patch1.txt -> this is the fix |
| 20:57.22 | tzanger | zoa: is the fix in the latest set of patches? |
| 20:58.27 | zoa | no its not |
| 20:58.28 | zoa | afaik |
| 20:58.33 | zoa | i just posted it on mantis |
| 20:58.38 | zoa | so that stevek sees it there too |
| 20:59.17 | eKo1 | I'm not doing any more 'soft hangup's any more. Screw it. I'll just wait til it crashes. Stupid channels. |
| 21:00.58 | *** join/#asterisk chaoscon (~ph33r@chaoscon.user) |
| 21:03.36 | greg_work | is it possible to change the amount of time someone has to enter an extension number when pressing # ? |
| 21:03.44 | bjohnson | yes |
| 21:04.35 | bjohnson | DigitTimeout and ResponseTimeout |
| 21:05.09 | *** join/#asterisk jelque (~jelque@adsl-68-94-116-225.dsl.ltrkar.swbell.net) |
| 21:05.22 | *** join/#asterisk Uther_P (~uther_p@66.180.120.83) |
| 21:05.59 | *** part/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com) |
| 21:06.45 | *** join/#asterisk sangee (ravi@209.250.129.135) |
| 21:08.42 | sangee | How to setup Xlite for non static ip to work with Asterisk (it does not work for local ip) |
| 21:09.00 | *** join/#asterisk kimc (~kimc@pcp04039944pcs.wbrmfd01.mi.comcast.net) |
| 21:09.46 | kimc | hello asterisk people.. anyone around? |
| 21:09.52 | jontow | hmm, of course |
| 21:09.53 | Uther_P | sangee: with xlite you have to specify the asterisk server as both the server and the proxy... its gay |
| 21:10.01 | *** join/#asterisk montoya (~darklab@200.214.193.122) |
| 21:10.06 | kimc | great |
| 21:10.14 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 21:10.46 | kimc | I've been trying to get a small system running with 1 fxo and 1 fxs |
| 21:11.24 | *** join/#asterisk hajekd (~hajekd@21.208.65.212.contactel.net) |
| 21:11.24 | kimc | Anyone know an example I could adapt to get things running ? |
| 21:11.33 | jontow | does anyone have a sample entry of an SMDI output capture? |
| 21:12.15 | sangee | I did, but it works for static ip but not work for local ip |
| 21:12.56 | *** join/#asterisk ^Fenris (~chatzilla@d3-99.rb.ot.centurytel.net) |
| 21:13.19 | xkev | can anyone explain the difference between 'restart gracefully' and 'restart when convenient' ? |
| 21:13.26 | hajekd | How will be the voice quality over network with 700 ms latency? There is no packet loss... |
| 21:13.44 | johnnyb | Ouch. |
| 21:13.45 | xkev | hajekd jitter and loss are your enemy. you'll just have delays |
| 21:13.54 | xkev | might have issues with echo cancellation |
| 21:13.54 | ^Fenris | Mar 14 16:13:04 NOTICE[24041]: chan_sip.c:4309 sip_reg_timeout: -- Registration for '614xxxxxxx@sip.broadvoice.com@sip.broadvoice.com' timed out, trying again -- Got SIP response 404 "Not Found" back from 147.135.0.128 |
| 21:14.02 | johnnyb | I thought that 120ms is when you start having noticeable delay |
| 21:14.11 | ^Fenris | just updated to CVS-head |
| 21:14.58 | TedC | How do I get * to accept anonymous inbound SIP calls? |
| 21:15.03 | Uther_P | xkev: I think when concenient still accepts new calls, and exits when there are no calls, where gracefully doesn't accept any new calls |
| 21:15.04 | TedC | I get a 404 whenever I try it. |
| 21:15.54 | *** part/#asterisk kimc (~kimc@pcp04039944pcs.wbrmfd01.mi.comcast.net) |
| 21:16.27 | xkev | uther, that seems to explain what I ran into last week then. thx |
| 21:17.16 | Uther_P | xkev: heh, what.. did I never run out of calls and never stop? |
| 21:17.26 | Uther_P | s/I/it/ |
| 21:19.30 | *** join/#asterisk tessier (~treed@210.245.97.149) |
| 21:19.35 | Corydon-w | TedC: perhaps allowguest=true ? |
| 21:19.59 | *** part/#asterisk km- (~pgrace@brdgw1.rttx.com) |
| 21:21.28 | TedC | Corydon-w: It defaults to yes, but I tried it set that way and the same thing happens. |
| 21:21.49 | TedC | I'm just Dialing SIP/<hostname> on my other * box. |
| 21:22.27 | Corydon-w | Perhaps "username=" |
| 21:22.37 | TedC | where? |
| 21:22.45 | Corydon-w | In the user entry |
| 21:22.55 | TedC | I don't have a user entry on either box. |
| 21:23.14 | Corydon-w | You might need one |
| 21:23.27 | Corydon-w | and a peer entry on the box you're sending from |
| 21:23.46 | Corydon-w | Or make type=friend on both |
| 21:24.21 | TedC | tried a type=user and username= on the recieving box, and still the same thin. |
| 21:24.42 | TedC | BTW: Is there a way to get better debugging on CVS? I miss the verbose 3 from 1.0 |
| 21:25.28 | TedC | I don't want to have to put something in sip.conf on the calling box if I don't have to, and it seems to be correctly contacting the server with the SIP/<domainname> syntax. |
| 21:25.29 | *** join/#asterisk zapa (~hector@200.77.109.193) |
| 21:25.41 | *** join/#asterisk Uther_P (~uther_p@66.180.120.83) |
| 21:26.48 | *** join/#asterisk sneak (~sneak@64.220.234.21.ptr.us.xo.net) |
| 21:29.17 | Corydon-w | TedC: what's the SIP error? |
| 21:29.40 | Corydon-w | Try doing a 'sip debug' on the calling box |
| 21:30.52 | *** join/#asterisk DJ-Pyro_ (~DJ-Pyro@firewall.brevient.net) |
| 21:31.32 | DJ-Pyro_ | question, is it possible to have 2 sip phones share an extention? if I have 2 phones registered to the same extention only the last one to register will ring |
| 21:32.10 | DJ-Pyro_ | is that something we need a sip proxy for? |
| 21:32.22 | herag | DJ-Pyro_: you should set them up seperately, but then put both of them in the Dial command: Dial(SIP/phone1&SIP/phone2) |
| 21:33.24 | DJ-Pyro_ | yeah, I figured that would be how to do it, we were trying to add some of the executive lines to the receptionists phone so she could answer for them |
| 21:34.50 | *** join/#asterisk mad_router (~ornate@84.246.176.66) |
| 21:35.22 | *** join/#asterisk KalD|Work (~KalD@proxy.corp.telesym.com) |
| 21:36.02 | KalD|Work | hi all - I'm getting Mar 14 13:32:02 NOTICE[30765]: channel.c:1352 ast_read: Dropping incompatible voice frame on IAX2/4256912043/22 of format gsm since our native format has changed to ulaw when I dial from an outside line into my iaxy but when I dial from my iaxy out it works fine... what am I missing? |
| 21:37.27 | jontow | SMDI is ugly :( |
| 21:41.45 | hardwire | bleh |
| 21:41.51 | hardwire | I can't handle IRC nick colors |
| 21:42.06 | hardwire | I tried.. |
| 21:42.09 | Dandan | lol |
| 21:42.12 | Dandan | change your client |
| 21:42.17 | Dandan | BitchX awaits |
| 21:44.26 | hardwire | I used to use it |
| 21:44.43 | *** join/#asterisk darby_t (~tom@dml229.neoplus.adsl.tpnet.pl) |
| 21:44.44 | hardwire | primarily for screening ability |
| 21:44.50 | hardwire | but now I use dircproxy for that |
| 21:44.54 | hardwire | so it doesn't matter what client I use |
| 21:45.03 | hardwire | and I am gnomey.. so xchat is nice |
| 21:45.14 | Uther_P | I wrote a unix irc client a long time ago that used ncurses windows |
| 21:45.15 | hardwire | and I can support xchat w/ other workers easily |
| 21:45.25 | Uther_P | hadn't seen one like that before |
| 21:45.39 | hardwire | cool stuff |
| 21:45.46 | hardwire | but I like my logging disconnected IRC proxy |
| 21:45.54 | hardwire | it makes me happy in places I shouldn't talk about. |
| 21:45.54 | Uther_P | but I foolishly wrote it from scratch before I even thought to use an irc lib, heh |
| 21:45.57 | Uther_P | hahah |
| 21:46.20 | hardwire | tricky stuff. |
| 21:46.41 | Uther_P | the irc protocol is so simple though |
| 21:46.45 | hardwire | yeh |
| 21:46.46 | hardwire | it really is |
| 21:46.57 | hardwire | II turned off flooding on an irc server |
| 21:46.58 | Uther_P | I don't know how you would screw it up unless you just didn't even bother reading the rfcs on it |
| 21:47.09 | hardwire | to test using it for a media streaming system |
| 21:47.12 | hardwire | for text based streams |
| 21:47.14 | hardwire | mainly stock quotes |
| 21:47.20 | hardwire | it was incredibly effecient |
| 21:47.29 | hardwire | w/ wonderfull server expandibility over compressed trunks |
| 21:47.39 | hardwire | but it could have been more network effecient |
| 21:47.45 | hardwire | by becoming much more binary |
| 21:47.49 | hardwire | and much less human readable |
| 21:48.14 | hardwire | and adding support for binary lookups to the server. |
| 21:48.21 | hardwire | so that it can become human readable again |
| 21:49.14 | Uther_P | does anyone know if there is a way to initiate calls in asterisk from an external applicaiton? i.e. a library that an application can use to make and bridge calls and such? |
| 21:49.38 | hardwire | write a sip client to do that :) |
| 21:50.04 | hardwire | I don't think there is anything to initiate a callback like that |
| 21:50.12 | Uther_P | hrm... yea, that would work... seems like a bit more work than nessesarry though |
| 21:50.27 | Uther_P | damn... maybe I'll write one then |
| 21:50.28 | hardwire | Uther_P: but asterisk is made to interface that way :) |
| 21:50.43 | hardwire | that way you can deal with your own timers to initiate calls. |
| 21:51.07 | Uther_P | I want to write a client application for extension control... cid, initiate calls, call routes, voicemail, ect... |
| 21:51.50 | *** join/#asterisk ckruetze (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com) |
| 21:52.59 | *** join/#asterisk atuc (~atuc@dsl-213-023-252-203.arcor-ip.net) |
| 21:55.18 | Qwell | kram: Any idea if the gaim guys are working on an IAX plugin? heh |
| 21:57.31 | ^Fenris | back to "Failed to Authenticate on INVITE to '"asterisk" |
| 21:57.31 | ^Fenris | just updated to cvs-head |
| 21:57.31 | ^Fenris | used to work though...any ideas anyone? |
| 21:57.51 | zapa | hi all where i can find the differences in zaptel drivers from verison 1.0.5 to 1.0.6 |
| 21:58.08 | *** join/#asterisk TheBear (~brif8@lazyjtrainingcenter.com) |
| 21:58.23 | opus___ | i just built cvs-head, and now asterisk won't start complaining that there is no zaptel.conf, which I don't use.. |
| 21:58.24 | TheBear | can you terminate now 1700, 18xx calls via iaxtel as well ? |
| 21:58.28 | opus___ | Is this a new problem? |
| 21:58.44 | Qwell | TheBear: huh? |
| 21:58.48 | TheBear | nwo -> non |
| 21:58.56 | TheBear | can you terminate non 1700, 18xx calls via iaxtel as well ? |
| 21:59.30 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 22:00.08 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 22:00.52 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 22:01.08 | *** join/#asterisk ta[i]nted (~ta_i_nted@65-60-70-243-cust.telepacific.net) |
| 22:01.34 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 22:02.20 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 22:03.06 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 22:05.36 | *** join/#asterisk buddah (~hnic@208.179.86.5) |
| 22:06.48 | *** join/#asterisk SPoon_TSX (~SPoon_TSX@24.83.96.211) |
| 22:07.17 | SPoon_TSX | Hi there, just wondering anyone here using Sayson 480i on their Asterisk environment? |
| 22:07.59 | gambolputty | VoIP to Fuel Plague of 'Dialing for Dollars'/Spam http://it.slashdot.org/it/05/03/14/2044255.shtml?tid=215&tid=111 |
| 22:08.16 | herag | does the email sent to cofirm digium list subscription arrive immediately after signing up? |
| 22:08.58 | opus___ | has anybody got a spam call before? I thought it was a pretty good joke |
| 22:09.29 | opus___ | they're going to nmap the entire internet for iax ports to call people, or something |
| 22:10.05 | *** join/#asterisk outsidefactor (~blah@203-173-32-225.dyn.iinet.net.au) |
| 22:14.23 | tuxinator_linux | Does anyone have experince using Digium T1 cards using PRI lines with BGP? |
| 22:14.28 | *** part/#asterisk Juxt (~Juxt@adsl-068-213-216-087.sip.bct.bellsouth.net) |
| 22:17.47 | mitcheloc | opus___: they can't do that, they would piss off all the anal net admins, and get tons of nasty letters |
| 22:19.54 | *** join/#asterisk Mother_ (~m@53.Red-217-126-93.pooles.rima-tde.net) |
| 22:19.57 | Mother_ | hi all |
| 22:20.05 | Mother_ | this wav49 crap is driving me nuts |
| 22:20.23 | nestAr | whee.. |
| 22:20.44 | Mother_ | if it's actually a form of GSM 6.10 then why does it say riff and wave in the header?!?!?! |
| 22:21.05 | nestAr | wav49 works for me.. |
| 22:21.06 | nestAr | dunno |
| 22:21.14 | Damin | Hmmm... |
| 22:21.17 | Mother_ | but how do you play it in windoze |
| 22:21.22 | RickTick | anybody ...using Broadvoice on their box ...here? |
| 22:21.24 | Damin | DTMF Inband just stopped working on my Polycom.. |
| 22:21.32 | nestAr | i play mine in media player or winamp |
| 22:21.34 | Mother_ | I can do playback() during a call just fine, audio is fine |
| 22:21.50 | Mother_ | well mine sound like crushing a sheet of paper |
| 22:21.58 | *** join/#asterisk brettnem (~mive29@user-0ccsr2l.cable.mindspring.com) |
| 22:21.58 | Mother_ | no matter what I play them with |
| 22:22.11 | opus___ | hmm, still having problems calling out to broadvoice: Mar 14 15:29:32 NOTICE[25882]: chan_sip.c:6814 handle_response: Failed to authenticate on INVITE to '"D" |
| 22:22.17 | opus___ | failed to authenticate on invite |
| 22:22.29 | Mother_ | nestAr: what format tag/flag do you use to record? |
| 22:22.53 | nestAr | i was talking more about Voicemail sent out |
| 22:22.55 | nestAr | in email |
| 22:23.16 | *** join/#asterisk Darwin_35 (~Darin@c-24-3-226-147.client.comcast.net) |
| 22:23.18 | Mother_ | nestAr: yes, I'm recording calls, but for the purpose of the exercise it's the same thing |
| 22:23.28 | nestAr | Monitor? |
| 22:23.30 | Mother_ | yeah |
| 22:23.42 | Mother_ | I use Monitor(wav,filename) |
| 22:24.09 | RickTick | opus: when you bridge an incoming call ...to a cellular(using BV) ...does the call jump straight into the cellular voicemail? |
| 22:24.40 | nestAr | i've done that before.. |
| 22:24.44 | nestAr | and it worked ok.. |
| 22:25.06 | Mother_ | with the wav flag? really? |
| 22:25.06 | Mother_ | hrrrmfffpfff |
| 22:25.06 | nestAr | I was using it on outgoing calls for the purpose of stealing the menus off our old PBX |
| 22:25.33 | Mother_ | yes, the recorded audio plays back just fine with *, it's just that I cannot play the recordings in windows |
| 22:25.35 | nestAr | i was doing exten => _9NXXXXXX,2,Monitor(wav,/tmp/monitor) |
| 22:25.48 | nestAr | i played them back in windows |
| 22:25.52 | nestAr | i had to crop them |
| 22:25.57 | Mother_ | ah! |
| 22:26.03 | nestAr | hold, i'll test it |
| 22:26.10 | Mother_ | thanks a lot :) |
| 22:26.52 | *** join/#asterisk sd-tux (sd@2001:6f8:1372:0:0:0:0:2) |
| 22:29.11 | Mother_ | ergh gotta run, will be back in 30-40 minutes |
| 22:29.22 | *** join/#asterisk bah (048830696@ACA11D4D.ipt.aol.com) |
| 22:29.22 | nestAr | http://wewt.net/temp/my-out.wav |
| 22:29.23 | Mother_ | ty for the help |
| 22:29.25 | nestAr | works for me |
| 22:29.27 | Mother_ | ah OK thanks |
| 22:29.30 | Mother_ | lemme grab it |
| 22:29.30 | nestAr | see if it plays for you |
| 22:30.13 | Mother_ | it plays just fine |
| 22:30.19 | opus___ | ricktick -- yes |
| 22:30.27 | Mother_ | thanks a lot, now I have something to compare against |
| 22:30.28 | nestAr | exten => _9NXXXXXX,2,Monitor(wav,/tmp/my) |
| 22:30.36 | nestAr | that's the exact line i'm using |
| 22:30.40 | nestAr | for the sake of noting it |
| 22:30.40 | opus___ | can somebody show me their working broadvoice config? |
| 22:30.49 | Mother_ | OK will try that too, thanks again! |
| 22:30.52 | nestAr | np |
| 22:30.53 | nestAr | later |
| 22:30.55 | Mother_ | gotta run or risk wife beating :D |
| 22:31.07 | nestAr | haha |
| 22:31.08 | Mother_ | i.e. wife beating me, not otherwise |
| 22:31.10 | Mother_ | hehe |
| 22:31.17 | nestAr | yeah |
| 22:31.19 | nestAr | i gottchaq |
| 22:31.22 | nestAr | -q |
| 22:33.14 | *** join/#asterisk darius_ (darius@c-67-171-44-101.client.comcast.net) |
| 22:35.54 | *** part/#asterisk darius_ (darius@c-67-171-44-101.client.comcast.net) |
| 22:36.12 | *** join/#asterisk jaxxan (~jaxxan@202.70.125.109) |
| 22:36.18 | jaxxan | hey ya'll |
| 22:36.48 | jaxxan | anyone know any good open source VOIP Troubleshooting Apps ? |
| 22:37.04 | mitcheloc | ethereal |
| 22:38.05 | jaxxan | voip using G723 sounds kinda digitized and feels like lost packets over +551ms satellite latency |
| 22:38.20 | jaxxan | not sure how ethereal would help me |
| 22:38.38 | jaxxan | anything else besides ethereal ? |
| 22:38.48 | jaxxan | i do use ethereal though... (= |
| 22:39.02 | hardwire | jaxxan: you going over sat? |
| 22:39.05 | jaxxan | yes |
| 22:39.09 | hardwire | so am I |
| 22:39.11 | jaxxan | i live on a small island |
| 22:39.18 | hardwire | so do I :) |
| 22:39.18 | mitcheloc | from what i understand ethereal can captures your calls and convert them into sound files so you can play them back later =) |
| 22:39.20 | hardwire | to a degree |
| 22:39.29 | hardwire | I adminster a small islands ISPness |
| 22:39.39 | denon | what island? |
| 22:39.39 | hardwire | can I give you a suggestion? |
| 22:39.45 | hardwire | st paul island, AK |
| 22:39.45 | jaxxan | american samoa |
| 22:39.48 | jaxxan | hardwire: sure |
| 22:39.50 | denon | AK? |
| 22:39.55 | hardwire | jaxxan: using SIP or IAX? |
| 22:40.00 | jaxxan | SIP |
| 22:40.00 | hardwire | denon: alaska |
| 22:40.02 | mitcheloc | hardwire: need some help? |
| 22:40.03 | denon | oh, US.. |
| 22:40.06 | *** part/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net) |
| 22:40.15 | denon | I figured you were on some remote island in the pacific with no name .. ;) |
| 22:40.21 | hardwire | jaxxan: try getting an IAX provider that supports jitter buffer |
| 22:40.25 | hardwire | and trunk w/ them |
| 22:40.40 | hardwire | depending on what your jitter is |
| 22:40.44 | hardwire | you can measure that by using smokeping |
| 22:40.45 | jaxxan | who do you use? |
| 22:40.53 | jaxxan | we got smokeping setup |
| 22:40.58 | hardwire | me myself and I for the most part |
| 22:41.07 | hardwire | iax trunking offices |
| 22:41.15 | hardwire | starband? |
| 22:41.18 | hardwire | or direcpc? |
| 22:41.20 | hardwire | or vsat? |
| 22:42.14 | *** join/#asterisk scottish23 (~new@cpc1-renf3-5-0-cust52.renf.cable.ntl.com) |
| 22:42.39 | hardwire | jaxxan: if you want you can bounce it off of me and see how well it works |
| 22:42.48 | hardwire | I can set up a test account w/ echo abilities |
| 22:43.21 | jaxxan | we're in the middle of troubleshooting some things right now with one of our voip providers |
| 22:43.28 | hardwire | ko |
| 22:43.30 | hardwire | er |
| 22:43.31 | hardwire | ko |
| 22:43.32 | jaxxan | we could do it later though if you're available |
| 22:43.32 | hardwire | err |
| 22:43.34 | hardwire | o k |
| 22:43.35 | scottish23 | Hi all, I'm having problems setting up call parking. Do I need to create the parking extns ? |
| 22:43.39 | hardwire | jaxxan: just privmsg me |
| 22:43.45 | jaxxan | will do |
| 22:43.54 | hardwire | what IP provider ? |
| 22:44.02 | hardwire | for the sat. |
| 22:44.47 | hardwire | they drive my dog crazy with their cookouts |
| 22:45.12 | hardwire | we have to feed him meat before he chills out |
| 22:45.17 | jaxxan | intelsat |
| 22:45.23 | julianjm | Hello... anyone in Spain has app_sms working? |
| 22:45.37 | hardwire | jaxxan: whats the bandwidth like? |
| 22:45.44 | hardwire | and how many lines are you expecting to push? |
| 22:46.09 | *** join/#asterisk _GiGi_ (gigi@jabber.szczecin.pl) |
| 22:46.22 | hardwire | jaxxan: about 100% of the issues you are going to have over SAT is your router. |
| 22:46.32 | hardwire | packet prioritization is key to your survival!!!! |
| 22:46.36 | _GiGi_ | hello. |
| 22:46.39 | jaxxan | we have a packetshaper |
| 22:46.52 | jaxxan | we're looking to push 48 b channels |
| 22:46.56 | _GiGi_ | Can i authorize IAX2/SIP clinets from mysql ? |
| 22:47.06 | jaxxan | 24in 24out |
| 22:47.45 | *** join/#asterisk Darwin35 (~Darin@c-24-3-226-147.client.comcast.net) |
| 22:47.59 | Darwin35 | Darwin_35 give up my nick |
| 22:48.06 | Darwin35 | I am the only darwin 35 |
| 22:48.12 | Darwin35 | grr |
| 22:48.46 | eKo1 | Fighting over nicks is pathetic. |
| 22:49.01 | file[laptop] | considering it's you... :p |
| 22:49.35 | hardwire | jaxxan: neat. |
| 22:49.39 | *** join/#asterisk MatsK (~NNSCRIPT@107.80-202-57.nextgentel.com) |
| 22:49.39 | _GiGi_ | hm ? |
| 22:49.45 | hardwire | but using TDMoIP? thats what it sounded like |
| 22:49.53 | Darwin35 | i hate when people have to use a nick thats already in use |
| 22:49.57 | Darwin35 | but hey that sme |
| 22:50.09 | hardwire | atleast its not like AOL |
| 22:50.15 | Darwin35 | no |
| 22:50.20 | Darwin35 | ? |
| 22:50.21 | hardwire | we dould know that youa re the 35th lamo to use Darwin on this server |
| 22:50.25 | hardwire | :) |
| 22:50.26 | Qwell | Darwin35: ...you should really whois Darwin_35, unless you're trying to be funny. |
| 22:50.38 | opus___ | in BV do I need to put a "1" in front of my area code when dialing my own area code? |
| 22:50.41 | eKo1 | ~nickometer Darwin35 |
| 22:50.51 | Darwin35 | his name is darin |
| 22:50.52 | eKo1 | Well, there you have it. |
| 22:51.01 | denon | ~nickometer denon |
| 22:51.45 | Darwin35 | ok so I am in a mood tonight and want my name to my self |
| 22:51.48 | Darwin35 | I am sorry |
| 22:52.15 | Darwin35 | and its my nick +my age |
| 22:52.15 | Qwell | Darwin35: You DO realize, that both of the nicks are from...you? |
| 22:52.28 | opus___ | I get dead air after calling out from BV |
| 22:52.29 | opus___ | <PROTECTED> |
| 22:52.46 | Darwin35 | hmm ok shoot myself |
| 22:53.03 | Darwin35 | it autolaunched a 2nd ver of xchat |
| 22:53.06 | Darwin35 | oops |
| 22:53.16 | opus___ | ~nickometer opus___ |
| 22:53.24 | opus___ | sounds about right |
| 22:53.58 | Darwin35 | and having issue with asterisk loading but not working like it shold |
| 22:54.23 | Darwin35 | not getting dialtone on the sip |
| 22:56.06 | *** part/#asterisk IOscanner (~IOscanner@c-67-162-251-133.client.comcast.net) |
| 22:56.51 | jaxxan | hardwire: you're using Asterisk with IAX? or do you happen to be going through a nortel switch too ? |
| 22:57.21 | hardwire | jaxxan: no nortel |
| 22:57.30 | *** join/#asterisk cr4z3d (~cr4z3d@ool-4350ba69.dyn.optonline.net) |
| 22:57.31 | hardwire | I assumed you weren't using nortel sip hardware |
| 22:57.36 | hardwire | since you are in (*) |
| 22:58.04 | jaxxan | my scenario: Asterisk --> PRI --> Cisco 5350 Media Gateway SIP Conversion --> Internet --> Media Gateway --> Sip provider |
| 22:58.10 | cr4z3d | i've heard you can use * to caller id spoof is this true? |
| 22:58.16 | eKo1 | Yes. |
| 22:58.18 | jaxxan | my scenario: Asterisk --> PRI --> --> Nortel DMS100 --> Cisco 5350 Media Gateway SIP Conversion --> Internet --> Media Gateway --> Sip provider |
| 22:58.24 | jaxxan | 2nd one... forgot DMS100 (= |
| 22:58.40 | jaxxan | in a nutshell |
| 22:59.01 | cr4z3d | well if you have a VoIP service do you need to have one of those phone line out things? |
| 22:59.33 | eKo1 | cr4z3d: Yes, you need a phone line out thing. |
| 22:59.59 | cr4z3d | do they make any for laptops? |
| 23:00.24 | *** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
| 23:00.39 | Ron-Na | regex question: ASTCC - Regex: How to "Country" but "special City" |
| 23:00.41 | jaxxan | see, i think that Asterisk by itself is great. it's just the DMS100 and all the conversions and the satellite latency that's causing the issues |
| 23:00.59 | eKo1 | Ron-Na: What? |
| 23:01.34 | eKo1 | jaxxan: VoIP over satellite will always suffer from latency unless you have a dedicated link. |
| 23:01.48 | Ron-Na | I try to setup a city with a price and the country with another price, ... but how to do this overlapping, like China is 86, but Beijing is 861 |
| 23:02.27 | eKo1 | Ron-Na: I don't use astcc. Sorry. |
| 23:02.48 | Ron-Na | eKo1 does not matter, it is a regex, ... |
| 23:03.38 | Ron-Na | If I say ^861 and ^86 it will (in my thinking) use ^86 and never ^861 |
| 23:03.40 | _GiGi_ | mhm |
| 23:05.14 | _GiGi_ | no one have mysql auth for IAX2/SIP clients ? :) |
| 23:05.38 | eKo1 | Ron-Na: Well, so you want to match a number with the most amount of digits as possible? |
| 23:05.50 | *** part/#asterisk mad_router (~ornate@84.246.176.66) |
| 23:05.57 | Ron-Na | eKo1, yes! |
| 23:06.27 | Ron-Na | first match as most and than fall back to less exactly, ... |
| 23:06.49 | jaxxan | eKo1: dedicted link? we have two e1's |
| 23:06.55 | *** join/#asterisk mentat_ (Mentat@pcp01260498pcs.nhaven01.ct.comcast.net) |
| 23:07.09 | eKo1 | Usually I test each entry and then to an order by desc. |
| 23:07.16 | mentat_ | Hello |
| 23:07.25 | eKo1 | jaxxan: eh, I though you had a sat. link. |
| 23:07.37 | jaxxan | eKo1: have two e1's over a sat link |
| 23:08.38 | mentat_ | quick (perhaps) question: is there a way to let an incoming line ring and answer it only if a dialled softphone answers? |
| 23:08.56 | jaxxan | i've always routed my LD calls on Asterisk thru PRI to DMS100. I think with all the problems that i'm encountering, perhaps i should route my LD calls thru ethernet to a voip provider in the states. |
| 23:09.04 | jaxxan | for testing purposes to start |
| 23:09.18 | jaxxan | what would be a good provider to start with ? |
| 23:09.37 | jaxxan | that can terminate my land line destined calls ? |
| 23:11.14 | eKo1 | The all stink I'm afraid. |
| 23:11.20 | eKo1 | *They |
| 23:11.51 | *** join/#asterisk TheDingy (~notknown@adsl-69-153-158-188.dsl.hstntx.swbell.net) |
| 23:12.05 | TheDingy | hey gus... who all has had problems with nufone? |
| 23:12.09 | TheDingy | it seems like they suck bad |
| 23:12.21 | Qwell | Whats wrong with NuFone? |
| 23:13.00 | TheDingy | qwell... they take more than a week to respond to support, tell you after you ordersomething, it is going to be 10 days to get a did and then |
| 23:13.10 | TheDingy | tell you "we have your money, so we don't care" |
| 23:13.28 | jaxxan | i need a provider that doesn't require me to use additional hardware |
| 23:13.35 | jaxxan | vonage would be out of the question i think |
| 23:13.42 | mentat_ | what about using FWDOUT? |
| 23:13.52 | jaxxan | that's that ? |
| 23:13.55 | jaxxan | what's that i mean |
| 23:13.55 | hardwire | jaxxan: I like nufone and teliax |
| 23:14.13 | jaxxan | hardwire: they relatively inexpensive ? |
| 23:14.15 | mentat_ | http://www.fwdout.net/web/ |
| 23:14.20 | hardwire | jaxxan: naw.. |
| 23:14.27 | hardwire | most are around 0.02/min |
| 23:14.39 | hardwire | if you have 48 lines I would assume you could get lower rates. |
| 23:14.46 | shido6 | TheDingy we do care. |
| 23:14.51 | mentat_ | For testing FWDOUT terminates for free with about a billion points of entry in CC 1 |
| 23:15.28 | jaxxan | k, i'll check those out |
| 23:15.40 | hardwire | TheDingy: I didn't have that response |
| 23:15.46 | *** join/#asterisk pr0ph (~blah@c-24-1-0-47.client.comcast.net) |
| 23:15.53 | hardwire | infact Iwas set up and using nufone.. with support.. in less than a day. |
| 23:15.58 | TheDingy | lol |
| 23:16.00 | hardwire | and support after that came quickly |
| 23:16.09 | TheDingy | well.. it seems that you are the exception, look in the wiki |
| 23:17.00 | jaxxan | i guess the nice thing is that i can keep all my local calls thru the PRI and keep the call center up while testing the LD VOIP portion |
| 23:17.26 | hardwire | jaxxan: I dropped our voice over sat system |
| 23:17.30 | hardwire | which was 4 E&M lines |
| 23:17.35 | hardwire | and rolled it all into data |
| 23:19.07 | jaxxan | hardwire: that's exactly what we're trying to do |
| 23:19.14 | hardwire | gotcha |
| 23:19.22 | hardwire | over say I suggest using IAX2 trunking and g.729 |
| 23:19.24 | Ron-Na | my $sth = $dbh->prepare("SELECT * FROM routes WHERE " . $dbh->quote($num |
| 23:19.24 | Ron-Na | ber) . " RLIKE pattern ORDER BY LENGTH(pattern) DESC"); |
| 23:19.28 | jaxxan | we're not having much luck with quality though |
| 23:19.42 | jaxxan | iax2 trunking and g729 |
| 23:19.44 | hardwire | or SER and some cool silence supression |
| 23:19.52 | jaxxan | we have some ser boxes |
| 23:20.06 | hardwire | yeh |
| 23:20.07 | hardwire | oh well |
| 23:20.10 | Ron-Na | Sorry, does pattern ORDER BY LENGTH(pattern) DESC mean that it first compares the longer numbers??? |
| 23:20.15 | hardwire | I need to get these SPA-3000's configged |
| 23:20.16 | hardwire | bbl |
| 23:21.06 | pr0ph | First of all, I appologize for being a total asterisk newb. Is is possible to have askterisk do outbound calling? I am looking for a good way to automate patient appointment reminder calls for a doctors office. |
| 23:21.12 | shido6 | bbl |
| 23:21.12 | *** join/#asterisk SPoon_TSX (~SPoon_TSX@24.83.96.211) |
| 23:21.28 | jaxxan | pr0ph: in a nutshell yes |
| 23:21.34 | SPoon_TSX | Just wondering how is the quality of using X-Lite compare to CISCO 7960? |
| 23:21.52 | pr0ph | cool... anywhere I can read about setting it up to do it? |
| 23:21.53 | jaxxan | SPoon_TSX: well ones software and ones hardware |
| 23:22.04 | jaxxan | SPoon_TSX: they both sound good |
| 23:22.09 | jaxxan | SPoon_TSX: i'd prefer the phone though |
| 23:22.58 | SPoon_TSX | Hi Jaxxan: Regarding the Echo Cancellation. How will that setup? Since I tried X-Lite, and I do hear some echo on my! But it didn't happen like that via the Phone. |
| 23:23.28 | jaxxan | SPoon_TSX: i only uses x-lite when i first started using asterisk and i use all 7960 |
| 23:23.30 | jaxxan | now |
| 23:23.32 | AgiNamu | Anyone here know about "VoiceScape"? |
| 23:23.45 | jaxxan | you prolly just need to play with your software configuration in x-lite |
| 23:23.46 | AgiNamu | They've licensed something from Digium |
| 23:24.16 | SPoon_TSX | oic. But you like the Cisco more? Do you mind to share with us why you like the Cisco phone more then the X-Lite? |
| 23:25.05 | AgiNamu | The Cisco is a phone and X-Lite is just a bit of software? |
| 23:28.07 | eric_ | anyone know of any DID services in Canada? (besides the ones listed on voip-wiki.org) |
| 23:29.31 | SPoon_TSX | I am jsut very concern about the sound quality. If they sound about the same then I will go toward to X-Lite instead. |
| 23:30.04 | *** join/#asterisk Darwin35 (~Darin@c-24-3-226-147.client.comcast.net) |
| 23:30.50 | pr0ph | Is there any desktop software available that works with asterisk to popup information about an inbound caller based on their caller ID? Or maybe have it run a script of some sort (and send the caller ID as a parameter)? Basically, I want to pull up a veterinary patients record autmatically if the inbound caller's phone number matches the one in our database. |
| 23:30.50 | *** join/#asterisk darkskiez (~mhb@host-84-9-68-172.bulldogdsl.com) |
| 23:31.59 | hardwire | heya |
| 23:32.05 | hardwire | anybody using 7960 cisco phones w/ (*)? |
| 23:32.47 | *** join/#asterisk zagaya972 (moi@APointe-a-Pitre-102-1-12-167.w81-248.abo.wanadoo.fr) |
| 23:33.04 | Darwin35 | what is the sox command to convert a ,wav to gsm |
| 23:33.10 | Darwin35 | I use to know |
| 23:33.21 | Darwin35 | having a bad da with this |
| 23:35.08 | Darwin35 | Pro you will have to sript it or make a agi to do it |
| 23:35.20 | Darwin35 | I have not seen anything like what you want yet |
| 23:35.28 | opus___ | <PROTECTED> |
| 23:35.29 | opus___ | hmmm |
| 23:35.33 | opus___ | What does that mean? |
| 23:36.51 | *** join/#asterisk signal15 (~austad@c-24-118-220-176.mn.client2.attbi.com) |
| 23:36.52 | opus___ | I have broadvoice apparently dialing out, however, my cellphone never rings |
| 23:37.04 | opus___ | The last line it mentions is -- SIP/sip.broadvoice.com-2d37 answered SIP/david-e785 |
| 23:37.04 | opus___ | <PROTECTED> |
| 23:37.15 | opus___ | My BV account shows that I have used zero outgoing minutes |
| 23:37.24 | signal15 | does anyone have a link to any whitepapers comparing the costs of going with an Asterisk solution versus Avaya and/or Cisco? |
| 23:38.02 | buddah | anyone know the default l/p for mitel phones' web configuration? |
| 23:38.06 | jaxxan | Darwin35: man sox |
| 23:38.08 | buddah | cant figure out what it is |
| 23:38.09 | opus___ | signal - no, but i can tell you that it'd probably be 40% more to go with avaya |
| 23:38.16 | buddah | and not in the manual i have |
| 23:38.40 | signal15 | yeah, I kind of figured it out with Cisco, and depending on the phones I use with Asterisk, it's anywhere from 30-50% of the cost |
| 23:39.03 | signal15 | but I need some kind of whitepaper, preferably showing a real world implementation |
| 23:39.30 | opus___ | signal - there isn't really any call manager software for asterisk that is comparable to cisco's call manager software |
| 23:39.54 | signal15 | What features does the CallManager have that asterisk doesn't? |
| 23:40.12 | pr0ph | darwin35: yeah, i figured that was the case. It doesn't seem like it would be all that hard to pass the inbound caller ID to a seperate script. I just wish it was already part of asterisk. Maybe I'll submit it as a feature request. |
| 23:40.14 | signal15 | besides decent Skinny support. :) |
| 23:40.19 | Syncros | frequent crashes |
| 23:40.22 | signal15 | haha |
| 23:40.24 | signal15 | true |
| 23:40.50 | opus___ | signal - from what I can tell, asterisk doesn't have a production quality call manager software client |
| 23:41.08 | opus___ | so cisco has all features , asterisk none |
| 23:41.25 | Syncros | 8| |
| 23:41.33 | Darwin35 | asterisk is loaded with features |
| 23:41.38 | Darwin35 | read learn |
| 23:41.45 | opus___ | darwin : re: call manager |
| 23:42.06 | signal15 | I'm looking at this from a functionality standpoint.... conf rooms, transfer capability, VM, etc... |
| 23:42.08 | Darwin35 | ahh |
| 23:42.12 | signal15 | call center stuff |
| 23:42.17 | signal15 | etc |
| 23:42.20 | Darwin35 | it has it |
| 23:42.37 | opus___ | signal - asterisk has everything built in, but no proven production front end to have joe dumbshit user figure it out |
| 23:42.39 | Darwin35 | I have run a call center with asterisk |
| 23:43.02 | opus___ | darwin - how do agents login? via software or extensions? |
| 23:43.02 | signal15 | honestly, I don't really care about ease of use. I know the config of asterisk sucks, I've set it up a few times. but, I charge hourly for it, so the more hours, the better. |
| 23:43.14 | Darwin35 | I have a sales queue and a tech queue and a billing queue |
| 23:43.31 | opus___ | signal -- the asterisk config is easier then other commercial providers, i find. |
| 23:43.55 | signal15 | basically, I'm just looking for some info to present to clients on how much money they would save going with *, instead of cisco or some other commercial solution |
| 23:43.56 | opus___ | asterisk. for admins by admins |
| 23:44.01 | Darwin35 | I am writing a extensions.conf that in time will have most functions in it |
| 23:44.31 | opus___ | signal - get a quote from avaya, cisco, and shoretel |
| 23:44.46 | signal15 | yeah, I can do that, I work for a cisco reseller |
| 23:44.47 | opus___ | then, divide it in half, and thats the asterisk cost |
| 23:44.51 | signal15 | haha |
| 23:44.56 | Darwin35 | thats easy asterisk =free software/opensrc vs. othe pbx system where vm is a addon and so are many other features |
| 23:45.24 | opus___ | the biggest cost in asterisk will be the fucking phones $240/pop for a polycom |
| 23:45.33 | signal15 | so, if you were going to sell a company on asterisk that was previously considering cisco... what phones would you suggest? Cisco phones? Polycom? |
| 23:45.33 | *** join/#asterisk coolacid-p (~jason@216.99.98.39) |
| 23:45.35 | signal15 | SNOM? |
| 23:45.38 | opus___ | at like 80 phones your talking 20k |
| 23:46.02 | opus___ | signal -- good question, i use polycom phones but I have to RMA about 1 a week ( 50 user install) |
| 23:46.07 | signal15 | I'm talking a company with 500-1000 users |
| 23:46.13 | *** join/#asterisk T` (~total@MY.RES.cmu.edu) |
| 23:46.21 | opus___ | polycom is definitly getting their fab process down better |
| 23:46.41 | signal15 | do you buy directly from Polycom, or off the net? |
| 23:46.47 | *** join/#asterisk n3tar (~geno@201.254.92.137) |
| 23:46.50 | opus___ | i did however beta test cisco phones 5 years ago and they were pretty cool. Crashed all the time:) but that was 2000 |
| 23:46.51 | n3tar | hi |
| 23:46.56 | opus___ | signal - i dunno if polycom sells directly |
| 23:47.10 | signal15 | I know a polycom rep, I should give her a ring |
| 23:47.19 | opus___ | you need to be a solutions provider... but since your a consultant you could probably do that |
| 23:47.20 | opus___ | yeah |
| 23:47.27 | opus___ | signal -- ask for the IP600 |
| 23:47.36 | signal15 | how much are they online? |
| 23:47.42 | opus___ | signal -- the IP500 claims it has PoE, but it actually doesn't. they are lying |
| 23:47.46 | opus___ | $240 |
| 23:48.19 | signal15 | nice, any suggestions for POE patch panels? |
| 23:48.26 | opus___ | you can buy a broken ip500 (most likely broken headset port) on ebay from anywhere from $40 to $120 |
| 23:48.39 | opus___ | signal - 3com 4400pwr |
| 23:49.02 | Darwin35 | what is the sox command to convert a ,wav to gsm |
| 23:49.06 | Darwin35 | pppls |
| 23:49.18 | signal15 | that's a whole switch isn't it? |
| 23:50.11 | signal15 | no way I'm going to get 3com switches into most of these companies I deal with, they are all cisco shops. Some already have POE switches, but others don't want to have to upgrade. |
| 23:50.40 | signal15 | I've seen boxes that have 48 ports on them, one for the ethernet from the switch going in, and the other out to the jack, but I forgot who made them |
| 23:50.43 | SPoon_TSX | I just wondering how come people using Cisco phone instead of Xten Software phone? Any dis-advantage on Software phone? |
| 23:50.43 | slePP | Essobi: woohoo :> |
| 23:50.46 | slePP | now fix it ;> |
| 23:51.04 | opus___ | signal - Cisco has great technology but there upper management /sales staff and executive team can lick my boots |
| 23:51.08 | *** join/#asterisk nemith (~brbennet@nemith.sustaining.supporter.pdpc) |
| 23:51.10 | signal15 | The cisco phone is better quality sound in my experience |
| 23:51.27 | opus___ | signal - foundry |
| 23:51.36 | signal15 | eek! :) |
| 23:51.56 | SPoon_TSX | signal15: So you mean Xten software phone sound really bad? |
| 23:51.59 | slePP | SPoon_TSX: software phones tend to be choppy |
| 23:52.03 | slePP | whenever the system comes under load |
| 23:52.06 | signal15 | I like their load balancers, they are cheap and work well. Other than that, I've had serious crashing problems with BigIrons |
| 23:52.10 | slePP | no to mention you have to have a computer to use it |
| 23:52.20 | slePP | and if you are using the computer, you have to often make sure your soundcard doesn't do strange audio-remixing |
| 23:52.25 | *** join/#asterisk goatmilk (~goatmilk@130-127-45-50.chouse.resnet.clemson.edu) |
| 23:52.31 | slePP | or the person you're talking to is going to hear all the bings/boops of your computer |
| 23:52.35 | signal15 | Plus, with a softphone, there's nothing to slam down when you get pissed off. :) |
| 23:52.41 | slePP | signal15: exactly |
| 23:52.42 | slePP | :> |
| 23:52.44 | coolacid-p | lol |
| 23:53.03 | SPoon_TSX | lolz: good point signal15! |
| 23:53.08 | coolacid-p | W/ softphone, just transfer to infinate loop menu! |
| 23:53.22 | slePP | coolacid-p: that doesn't help with physically hitting something |
| 23:53.31 | slePP | wtf is 'lolz' anyway... |
| 23:53.33 | slePP | like lol wasn't bad enough :P |
| 23:53.41 | antifuchs | hit the monitor |
| 23:53.46 | JerJer[mobile] | so who is giving odds there will be someone bitching about NuFone on one of the asterisk lists very soon? |
| 23:53.47 | coolacid-p | Touch screen |
| 23:53.49 | slePP | antifuchs: could through the headset |
| 23:53.53 | antifuchs | alternatively, slam down the mouse (: |
| 23:53.55 | signal15 | I had a boss who tore the screen off 3 different laptops. Maybe he was preparing for a softphone. |
| 23:54.00 | Qwell | JerJer[mobile]: I'm in for $20 |
| 23:54.09 | antifuchs | hmm. that could work |
| 23:54.13 | slePP | JerJer[mobile]: 10:1 against |
| 23:54.21 | SPoon_TSX | slePP: "lolz" is more then "lol"... hahaaa... |
| 23:54.36 | slePP | SPoon_TSX: i gathered that much. but a 'z'? |
| 23:54.36 | antifuchs | "laughing out louds" |
| 23:54.44 | signal15 | I thought maybe you lol'd, and then fell asleep |
| 23:54.48 | signal15 | zzz |
| 23:54.50 | slePP | that's like lolizzle m'shizzle! |
| 23:54.57 | coolacid-p | Oh god |
| 23:54.58 | antifuchs | place cursor over hangup button. hold the mouse in your fist, with the buttons looking down. hit the table. |
| 23:54.58 | Qwell | the z is for zergush! |
| 23:54.58 | SPoon_TSX | singal15: BINGO! |
| 23:54.59 | goatmilk | JerJer[mobile]: is nufone down? |
| 23:55.05 | antifuchs | better than a receiver (: |
| 23:55.22 | signal15 | is anyone even using bellster lately? I don't see any calls coming through it |
| 23:55.32 | JerJer[mobile] | slePP: someone just demanded a full refund because our toll-free numbers take 7 to 10 days to provision... then he proceeded to send an obviously edited screen shot of our website making it look like we never state how long takes |
| 23:55.40 | coolacid-p | Make a transfer of a sound of a phone slamming? |
| 23:55.50 | slePP | JerJer[mobile]: ohoh! put up the screenshot :> |
| 23:55.52 | Qwell | edited screenshot? |
| 23:55.55 | Qwell | yes, please do |
| 23:55.57 | SPoon_TSX | Damn, but the hardware phone is pretty expensive (To get a good one.) that way I cannot convince my manager to use VoIP. |
| 23:56.00 | coolacid-p | Mmmm.. |
| 23:56.02 | JerJer[mobile] | then he responds "Get ready for this one" |
| 23:56.12 | slePP | NuFone Sucks! I Know It Does! Oh wait. Slashdot said that. Bastards. |
| 23:56.40 | signal15 | is there an asterisk users group in minneapolis? |
| 23:56.43 | slePP | why do lil' shitheads always feel that they're the commanders of the universe and because they didn't get their way, they figure they'll take down the entire company that annoyed them (albeit at their own fault).. |
| 23:57.00 | JerJer[mobile] | goatmilk: no |
| 23:57.10 | goatmilk | i'm followin' |
| 23:57.10 | Qwell | wtf |
| 23:57.28 | slePP | http://netmonks.ca/billgatesrunsapple.mp3 |
| 23:57.32 | slePP | like that guy ^^ |
| 23:57.44 | Qwell | Verizon just sent me a bill for $47.84...I don't even have long distance. |
| 23:57.53 | coolacid-p | Nice |
| 23:58.22 | Qwell | So, now that I'm looking at this bill...where's this alleged DTMF fee? |
| 23:58.24 | *** join/#asterisk Dawson64 (Dawson64@24.247.102.233.gha.mi.chartermi.net) |
| 23:58.43 | PatrickDK | dtmf? |
| 23:58.54 | Qwell | dunno, somebody said that last night |
| 23:58.55 | coolacid-p | I love how they feel the need to charge you for DTMF nowadays.. |
| 23:59.26 | goatmilk | slePP: i love how they always threaten something at the end. |
| 23:59.38 | slePP | goatmilk: yup |
| 23:59.44 | slePP | 'I'm going to take you down! You are at my whim!' |
