| 00:01.28 | BugKhaM | why is asterisk running on tcp port 2000? |
| 00:01.43 | BugKhaM | netstat shows that it's a "sieve" port |
| 00:02.05 | mvanbaak | BugKhaM: you have chan_skinny loaded ? |
| 00:02.07 | *** join/#asterisk angryuser (i=nononon@df01t2-213-44-84-3.d4.club-internet.fr) |
| 00:02.31 | robl^ | port 2000 is skinny / sccp |
| 00:03.44 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 00:03.50 | BugKhaM | mvanbaak: yes, it's loaded |
| 00:04.07 | mvanbaak | BugKhaM: there it is. chan_skinny binds to port 2000 |
| 00:04.20 | BugKhaM | mvanbaak: ok |
| 00:04.24 | mvanbaak | <--- off to bed |
| 00:04.26 | mvanbaak | latero all |
| 00:04.34 | robl^ | <-- off to dinner |
| 00:05.42 | DrkShadow | Given this config (http://pastebin.ca/910475), could someone tell me how to make the incoming context work? I'm not sure what it is.. but it registers, calls come in, I just can't make it go to the context I want it to start in. |
| 00:05.50 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 00:06.02 | DrkShadow | if you're going to tell me to put an extension in the default context, please don't bother saying anything. |
| 00:06.29 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 00:10.54 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 00:10.54 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 00:14.51 | *** join/#asterisk nvrpunk (n=root@81.90.21.227) |
| 00:15.19 | nvrpunk | so, there is a 10 second delay when dialing other numbers, anyone familiar with this? |
| 00:15.41 | DrkShadow | "other" numbers? |
| 00:15.46 | DrkShadow | as in, non-local extensions? |
| 00:15.56 | DrkShadow | what does your log say/ |
| 00:15.57 | nvrpunk | yes, through pstn or local extensions |
| 00:15.59 | nvrpunk | same |
| 00:16.09 | DrkShadow | local extensions have the problem?... |
| 00:16.24 | DrkShadow | I suppose I've seen the problem.. for me, it's an Aastra 57i |
| 00:16.49 | nvrpunk | which log? |
| 00:16.57 | nvrpunk | /var/log/asterisk/blah |
| 00:17.05 | DrkShadow | full |
| 00:17.46 | nvrpunk | messages? |
| 00:17.50 | nvrpunk | i see no full |
| 00:17.50 | DrkShadow | Full |
| 00:17.52 | DrkShadow | .... |
| 00:17.54 | DrkShadow | hmm |
| 00:18.11 | nvrpunk | lemme slocate |
| 00:18.22 | DrkShadow | messages doesn't show enough |
| 00:18.37 | nvrpunk | hmm |
| 00:18.40 | nvrpunk | no full |
| 00:18.41 | DrkShadow | do "asterisk -r" and then type "set core verbose 9" and place a call.. walk through the lines and see exactly what it's doing. |
| 00:19.29 | nvrpunk | ok |
| 00:21.02 | nvrpunk | Verbosity was 0 and is now 9 |
| 00:21.03 | nvrpunk | <PROTECTED> |
| 00:21.03 | nvrpunk | <PROTECTED> |
| 00:21.03 | nvrpunk | <PROTECTED> |
| 00:21.03 | nvrpunk | <PROTECTED> |
| 00:21.03 | nvrpunk | <PROTECTED> |
| 00:21.05 | nvrpunk | <PROTECTED> |
| 00:21.07 | nvrpunk | jackalvoip*CLI> |
| 00:21.09 | nvrpunk | :( |
| 00:21.11 | DrkShadow | pasting is bad. |
| 00:21.21 | nvrpunk | didnt realize it was that long |
| 00:21.30 | nvrpunk | looked like 3 lines in putty |
| 00:21.35 | DrkShadow | there're no errors or issues as far as asterisk is concerned. |
| 00:21.45 | nvrpunk | but it takes 10 seconds |
| 00:21.57 | nvrpunk | before ringing starts |
| 00:22.01 | DrkShadow | it's either latency over the internet, something with your phone in particular or... well, I dunno what else would cause a 10 sec delay. |
| 00:22.39 | DrkShadow | I would honestly think it's either your phone, your NAT settings, or you have an Aastra 57i |
| 00:23.05 | nvrpunk | we have a NAT |
| 00:23.12 | nvrpunk | anything we can do in that dept |
| 00:23.14 | nvrpunk | to help |
| 00:23.22 | DrkShadow | is the asterisk inside or outside? how about the phone? |
| 00:23.32 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 00:23.34 | nvrpunk | inside |
| 00:23.37 | nvrpunk | both inside |
| 00:23.39 | DrkShadow | both? |
| 00:23.41 | DrkShadow | ok.. |
| 00:23.47 | DrkShadow | that won't affect much then.. |
| 00:23.50 | siya | Anyone with experience of UK callerid (SIP only) |
| 00:23.55 | phix | I found the problem! |
| 00:23.56 | siya | Worked fine in NL |
| 00:23.57 | *** join/#asterisk Lucky7 (i=Lucky7@207.200.28.175) |
| 00:24.05 | DrkShadow | the trunk, though, needs to be qualify=yes, canreinvite=no in sip.conf, I believe. |
| 00:24.07 | phix | with TDM400p, one of the RJ11 cables were faulty lol |
| 00:24.15 | siya | erm, works fine for NL mobile |
| 00:24.16 | Qwell | phix: d'oh! |
| 00:24.17 | Lucky7 | Gotta love retarded PRI's that dont want to work. |
| 00:24.23 | siya | calling UK mobile doesn't work |
| 00:24.26 | DrkShadow | also nat=yes on the trunk |
| 00:24.43 | phix | Qwell: :) |
| 00:25.08 | siya | Qwell: how are you? |
| 00:25.10 | phix | it works great now! |
| 00:25.13 | JT | trunk |
| 00:25.15 | JT | trunk |
| 00:25.18 | phix | yes Qwell, how are you this fine morning? |
| 00:25.47 | Qwell | great. would be better if t-mobile would hurry up and ship my phone :D |
| 00:26.12 | siya | lol, I'm working at T-mobile now |
| 00:26.16 | Qwell | oh? |
| 00:26.19 | Qwell | check on my phone! |
| 00:26.23 | Qwell | they won't give me any status :( |
| 00:26.24 | siya | note the at (for is only indirectly) |
| 00:26.49 | siya | doing a Network project there, no idea about phones etc |
| 00:26.51 | siya | sry |
| 00:26.56 | Qwell | oh? |
| 00:27.17 | Qwell | when am I gonna have fios speeds on my phone? |
| 00:27.32 | siya | inet access? |
| 00:27.41 | siya | well depends on your type of connectivity |
| 00:27.43 | nvrpunk | ok, so here's where the delay occurs |
| 00:28.01 | siya | but I'm about to test upgrade of that part of the network |
| 00:28.17 | nvrpunk | we have 1 extenstions context and then a group of 5 phones that can dial out, those all connect to other phones instantly |
| 00:28.17 | Qwell | siya: tower network, and such? |
| 00:28.23 | Qwell | or more backend? |
| 00:28.25 | nvrpunk | but when the 5 that can't call out dial |
| 00:28.30 | nvrpunk | its a 10 second delay |
| 00:28.32 | *** join/#asterisk mosty (n=mostyn@ppp191-34.static.internode.on.net) |
| 00:28.41 | siya | Qwell: Routing, LB and FW... |
| 00:28.43 | phix | I would feel better if my ASUS laptop was in stock :( |
| 00:28.48 | phix | been 2 weeks now |
| 00:29.00 | phix | I NEED IT! :( |
| 00:29.18 | siya | phix: slightly more than 10 sec delay for the lappy then ;) |
| 00:30.12 | phix | no :( |
| 00:30.13 | phix | I wish |
| 00:30.15 | coppice | 2 weeks is the production life of some models :-) |
| 00:30.26 | phix | I know! I better be getting a discount |
| 00:30.45 | phix | ASUS x55sv! |
| 00:30.52 | DrkShadow | "but when the 5 that can't call out dial" <-- clarify, pls |
| 00:32.02 | ManxPower | nvrpunk: any delay in dialing is an issue for the actual SIP phone to deal with |
| 00:32.25 | ManxPower | DrkShadow: he has two groups of users, one group is allowed to dial outside the system, the other group is not allowed |
| 00:32.42 | phix | any way, asterisk is working great at this new site I set up, it is still not working great at home though :( linksys sipuara ATA is giving me feedback. |
| 00:33.05 | phix | feedback / echo |
| 00:33.23 | phix | any linksys ATA specialists here? :) |
| 00:33.30 | drmessano | Why yes |
| 00:33.36 | drmessano | I wrote the firmware for the PAP1 |
| 00:33.39 | phix | yay |
| 00:33.43 | phix | help me :P |
| 00:33.46 | phix | hehe |
| 00:33.53 | *** join/#asterisk PepOSX (n=angeldav@190.72.131.172) |
| 00:34.12 | drmessano | Ok |
| 00:34.17 | drmessano | Well |
| 00:34.33 | drmessano | Lose 20 lbs, cut back on the coffee, and stop downloading all that portn |
| 00:34.37 | drmessano | porn* |
| 00:34.59 | coppice | 2 out of 3 sound reasonable |
| 00:35.02 | Qwell | He's a Dr. You should take his advice. |
| 00:37.54 | *** join/#asterisk RoyK (n=roy@ip-57-12-149-91.dialup.ice.no) |
| 00:40.59 | hmm-home | bah why is asterisk sending back a "proxy auth required" when I have insecure=port,invite |
| 00:42.06 | phix | drmessano: how heavr is 20 lbs? I am metric over here |
| 00:42.12 | phix | heavr = heavy |
| 00:42.38 | phix | I hardly drink coffee, I have had 2 cups this week, about 14 cups of tea though :S |
| 00:43.13 | phix | hmm-home: cause you set a password? |
| 00:44.05 | *** join/#asterisk nvrpunk (n=root@81.90.21.227) |
| 00:44.10 | nvrpunk | sorry, back |
| 00:51.43 | sweeper | 22lbs = 10 kg |
| 00:53.26 | lmadsen | http://contactsheet.org/articles/2003/04/23/how_to_make_friends_by_telephone.html |
| 00:54.02 | *** join/#asterisk Lucky7 (i=Lucky7@207.200.28.175) |
| 00:54.04 | Lucky7 | Hey |
| 00:54.08 | Lucky7 | if i've got a PRI |
| 00:54.25 | Lucky7 | PRI t1's - Should I still see its status via Zap show status |
| 00:54.26 | DrkShadow | sigh.. even for an incoming call.. Ringing("SIP/<ipaddr-id>") .. I just can't figure out how to make it work with a defined peer def |
| 00:54.30 | Lucky7 | or does that change iwth a PRI |
| 00:55.25 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:56.48 | drmessano | Bah |
| 00:56.57 | robl^ | humbug? |
| 00:57.21 | drmessano | Phone has been ringing off the hook |
| 00:57.26 | drmessano | Apparently I am popular |
| 00:58.02 | drmessano | I turned in my resignation today at work.. and it's been crazy |
| 00:58.15 | drmessano | First they shut off my analog cell phone, and now this |
| 00:58.40 | *** join/#asterisk MaliutaWrk (n=nikolai@203.201.152.211) |
| 01:04.06 | drmessano | phix |
| 01:07.05 | *** join/#asterisk mamacito62 (n=XD@201.208.84.21) |
| 01:07.58 | Lucky7 | lol |
| 01:08.34 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 01:08.47 | *** join/#asterisk mike-ekim (n=mike@24.238.28.248) |
| 01:09.21 | mike-ekim | Im having a problem with a polycom phone, im behind a cisco router that has NAT i narrowed it down toa network problem in my situation though because i took the phone on another network that has NAT, connecting to a remote asterisk, and worked fine. |
| 01:10.39 | *** join/#asterisk emrah (n=doke@unaffiliated/emrah) |
| 01:12.20 | Lucky7 | http://rafb.net/p/Ok6pks36.html |
| 01:12.27 | Lucky7 | so i've configured my Pri |
| 01:12.35 | Lucky7 | when i try call out |
| 01:13.02 | Lucky7 | i've turned on pri debug span 1 |
| 01:13.12 | Lucky7 | and thats the messages I get every time it tries to touch a channel |
| 01:13.58 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 01:14.28 | Lucky7 | i've never setup PRI before so i'm a little out of place, but from what i can tell, it gets pissy at line 38 |
| 01:14.33 | Lucky7 | with the reserved chansel |
| 01:15.42 | mosty | mike-ekim, it might help if you actually mention what the problem is |
| 01:17.09 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-b20fbc004e17910e) |
| 01:17.54 | *** join/#asterisk osiris (n=osiris@c-71-205-29-230.hsd1.mi.comcast.net) |
| 01:18.43 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-69-154.lns10.syd6.internode.on.net) |
| 01:20.50 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 01:28.00 | mike-ekim | mosty: the communication is not establishing |
| 01:28.23 | mike-ekim | cannot receive calls, but can place calls, once the other end picks up the line, you hear no audiuo |
| 01:32.09 | mosty | mike-ekim, is it registered? |
| 01:32.35 | *** join/#asterisk CrashSys (n=kumba@62-209.187-72.tampabay.res.rr.com) |
| 01:32.43 | mike-ekim | yes it registers |
| 01:32.53 | hmmhesays | anyone using chan_gtalk? |
| 01:33.12 | CrashSys | Anyone used NFAS Groups (Shared D-Channel) before? The trunkgroup directive I understand but the spanmap isn't quite clear to me... |
| 01:33.24 | hmmhesays | no gtalk capable clients to talk to. |
| 01:33.32 | hmmhesays | I get that when I try and dial any gtalk clients |
| 01:33.56 | mosty | mike-ekim, do you have nat=yes on the asterisk side? |
| 01:35.07 | CrashSys | Spanmap => <zapspan>,<trunkgroup>,<logicalspan>... zapspan = T1 port right? trunkgroup means my trunkgroup definition I apply it to, and logicalspan is the span that I refer to it as in asterisk? |
| 01:36.28 | mike-ekim | yep |
| 01:36.33 | mike-ekim | i knw polycoms have problems with nat tho |
| 01:36.58 | mike-ekim | my setup is cable modem --> ciscorouter(LAN) --> lanswitch --> IPPhone |
| 01:37.18 | CrashSys | or is logical span what the other side of the switch refers to in it's channelization? |
| 01:37.54 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 01:39.28 | mike-ekim | a linksys phone works perfectly for some reason |
| 01:39.32 | mike-ekim | even behind two routers |
| 01:39.47 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 01:39.47 | *** mode/#asterisk [+o anthm] by ChanServ |
| 01:40.15 | JT | i don't have issues with polycoms behind nat |
| 01:40.24 | JT | not more than another brand |
| 01:40.33 | mike-ekim | no? |
| 01:41.39 | mike-ekim | im using a soundpoint ip 330 |
| 01:54.35 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.ITB.ac.id) |
| 01:55.56 | *** join/#asterisk uribes (n=Administ@189.186.20.100) |
| 01:58.01 | uribes | hi, is it posible to use a analog modem (used to get a internet conection) with asterisk? |
| 01:59.27 | drmessano | no |
| 02:04.18 | uribes | ok, so i need to buy an especial card?! which one do you suggest? |
| 02:05.43 | mosty | uribes, what do you want to do? |
| 02:07.13 | *** join/#asterisk seanbright (n=chatzill@c-69-251-175-43.hsd1.md.comcast.net) |
| 02:07.57 | uribes | i have installed an asterisk in home.. and i want to use my analog line connected to the asterisk |
| 02:08.56 | [TK]D-Fender | uribes, Linksys SPA-3102 |
| 02:11.29 | sweeper | actually, several modem cards can be used with chan_zap |
| 02:11.37 | mosty | uribes, what [TK]D-Fender said, or a sangoma a200d with an fxo module |
| 02:11.40 | sweeper | they're just really unreliable |
| 02:12.15 | sweeper | http://www.voip-info.org/tiki-index.php?page=X100P+clone |
| 02:12.18 | sweeper | there is the info |
| 02:12.26 | sweeper | if your modem is one of those, you'd be ok |
| 02:12.37 | uribes | ok, let me check |
| 02:13.24 | sweeper | people here just like to sell hardware ;) |
| 02:14.16 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 02:17.50 | uribes | i have the intel 537PG, thank i gonna check how to config it!! |
| 02:18.49 | *** join/#asterisk rfernandez (n=rfernand@201.137.54.18) |
| 02:18.53 | rfernandez | hi!! |
| 02:19.32 | rfernandez | a openvox pci card (mod a1200p) can work together with a digium te122p without troubles of irqs and crash of the box? |
| 02:21.52 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 02:32.28 | *** join/#asterisk Maxous (n=Maxous@76.97.3.24) |
| 02:32.44 | *** part/#asterisk Maxous (n=Maxous@76.97.3.24) |
| 02:37.45 | *** join/#asterisk jaxxan (n=Jaxxan@202.70.125.67) |
| 02:37.49 | jaxxan | hey guys |
| 02:38.07 | bkw_ | RUN its jaxxan |
| 02:38.16 | jaxxan | (= |
| 02:38.59 | jaxxan | i wanna take an incoming call, run it through an IVR and spit out some variables to a script |
| 02:39.28 | jaxxan | AGI and perl the way to go for that ? |
| 02:41.05 | jaxxan | oh, and there's snmp support now in 1.4 yeah ? |
| 02:41.26 | mosty | agi would work |
| 02:42.35 | jaxxan | i live mostly by example. got any ? |
| 02:42.47 | mosty | mr google does |
| 02:43.58 | jaxxan | yeah |
| 02:44.13 | jaxxan | i'm that's where i've been |
| 02:44.20 | jaxxan | -i'm |
| 02:45.50 | *** join/#asterisk sbingner (n=john@pdpc/supporter/sustaining/sbingner) |
| 02:46.13 | jaxxan | oh there's one |
| 02:46.33 | *** join/#asterisk seanbright (n=chatzill@c-69-251-175-43.hsd1.md.comcast.net) |
| 02:47.42 | jaxxan | is asterisk-perl 0.10 still the latest ? |
| 02:48.09 | jaxxan | works with latest trunk i hope |
| 02:52.34 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ppp-70-248-182-215.dsl.hstntx.swbell.net) |
| 02:54.35 | *** join/#asterisk galeras (n=Martin@200.118.198.26) |
| 02:56.32 | *** part/#asterisk uribes (n=Administ@189.186.20.100) |
| 02:56.36 | galeras | Dear Sirs, Please suggest me a setup to connect 120 fxo, 4 fxs and 2 E1/pri to an asterisk server. I'm not sure among rhino, xorcom, mediatrix... |
| 02:57.11 | jaxxan | 120 fxo's ? |
| 02:57.49 | jaxxan | you can get a TE410P for your PRI's |
| 02:58.11 | jaxxan | the TDM cards i'm aware of only do 4 fxo's |
| 02:58.45 | mosty | 120 FXO, is that a typo? |
| 02:58.49 | jaxxan | i dunno what to tell you about 120 fxo's |
| 02:59.01 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 02:59.03 | galeras | for 120 analog extensions |
| 02:59.06 | hmodes | channel banks or a shitton of atas ;p |
| 02:59.07 | jaxxan | i can't think of a server with enough expansion slots to hold that many cards |
| 02:59.14 | JT | galeras: fxs |
| 02:59.23 | galeras | srry, fxs |
| 02:59.33 | JT | jaxxan: you'd use a channel bank, not a multiport fxo card |
| 02:59.39 | jaxxan | okies |
| 02:59.43 | JT | or sip gateways |
| 02:59.48 | galeras | which channelbank do you recommend? |
| 03:00.01 | galeras | xorcom, rhino? |
| 03:00.35 | *** join/#asterisk Maxous (n=Maxous@76.97.3.24) |
| 03:00.58 | *** part/#asterisk Maxous (n=Maxous@76.97.3.24) |
| 03:01.28 | galeras | i'm thinking to install 4 channelbank (32 ports each one), i ear rhino is a good option |
| 03:02.12 | *** join/#asterisk seanbright (n=chatzill@c-69-251-175-43.hsd1.md.comcast.net) |
| 03:02.16 | *** join/#asterisk ecto (n=ectospas@c-71-207-229-248.hsd1.al.comcast.net) |
| 03:04.01 | hmodes | imo if you're spending that much on hardware, ask for a demo box from a couple vendors and try them out first |
| 03:04.10 | JT | Adtran or CAC |
| 03:04.37 | hmodes | never really had any exposure to e1 tho, so i dunno |
| 03:04.51 | J4k3 | eh1 |
| 03:06.26 | JT | you'd use T1 channelised robbed bit signalling |
| 03:06.31 | JT | never heard of channelised E1 |
| 03:07.34 | J4k3 | I'm in ur trunks, robbin ur bits |
| 03:08.51 | JT | galeras: other options, possibly better options, are SIP to FXS gateways |
| 03:09.18 | JT | Mediatrix, Audiocodes, Vegastream (?), and Patton among others make them |
| 03:10.02 | [TK]D-Fender | galeras, Avoid channel banks. Mediatrix 1124's or AudioCodes MP-124's |
| 03:11.01 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 03:12.21 | *** join/#asterisk AndyGraybeal (n=andy@node150.34.251.72.1dial.com) |
| 03:13.10 | *** join/#asterisk Maxous (n=Maxous@76.97.3.24) |
| 03:14.32 | JT | [TK]D-Fender: some other companies make 32 port units, and possibly higher |
| 03:15.11 | galeras | [TK]D-Fender: what about sangoma card with 48 fxs ports? |
| 03:15.27 | JT | galeras: what a pain ;) |
| 03:16.02 | [TK]D-Fender | galeras, beyond pain. Complete waste. |
| 03:16.23 | [TK]D-Fender | galeras, All PCI FXS is to be avoided IMO |
| 03:16.40 | galeras | got it, thanks |
| 03:17.17 | LiNeTuX | [TK]D-Fender: Why do you say that? |
| 03:17.19 | Maxous | Anyone know a good networking channel where I can ask a networking question?? |
| 03:18.06 | JT | LiNeTuX: imagine terminating 100 pairs to a breakout cable or lots of RJ ports on an fxs card, stuff that ;) |
| 03:18.28 | JT | LiNeTuX: if there's a pc failure, lots of work to move machine |
| 03:18.34 | LiNeTuX | JT: Oh, I get that - but for just a few (8 or less) it's not so bad. |
| 03:18.45 | *** join/#asterisk supjigator (n=shanebur@152.53.16.10) |
| 03:18.47 | JT | yeah |
| 03:18.51 | [TK]D-Fender | LiNeTuX, no, even then |
| 03:18.53 | supjigator | Hello. |
| 03:19.12 | JT | i think cards are still the best if you're trying to do something funny |
| 03:19.19 | supjigator | I just upgraded from 1.2 to 1.4 and now my PRI span just keeps telling me the D channel is up but won't take any calls. |
| 03:19.33 | [TK]D-Fender | LiNeTuX, All analog phone are best left going through SIP gateways. Far simpler to setup far less load and risk of issues, more flexible wiring and redundant capable |
| 03:20.03 | supjigator | <PROTECTED> |
| 03:20.08 | supjigator | Over and over and over |
| 03:20.13 | supjigator | Any ideas? |
| 03:20.50 | galeras | [TK]D-Fender: which brand of sip gateway do you recommed? |
| 03:21.50 | galeras | [TK]D-Fender: do you think, 5 mediatrix or audiocodes can do the job? |
| 03:22.02 | [TK]D-Fender | galeras, I like the MediaTrix personally for 24+ density, but AudioCodes is a bit better respected |
| 03:22.09 | *** join/#asterisk logyati (n=paulo@201.29.65.142) |
| 03:22.11 | logyati | hello! |
| 03:22.32 | galeras | [TK]D-Fender: and better priced too |
| 03:22.39 | logyati | i was reading about asterisk and digium cards, and then i had a doubt |
| 03:23.05 | logyati | can i use a fax modem with asterisk? i mean, to receive and make calls |
| 03:23.20 | logyati | i have an old us robotics 56k full duplex |
| 03:23.27 | logyati | pci |
| 03:23.41 | [TK]D-Fender | logyati, No. |
| 03:24.18 | logyati | [TK]D-Fender, isnt there any model that i could use with asterisk? |
| 03:24.44 | [TK]D-Fender | logyati, go look up the X100P on the WIKI, and may God have mercy on your soul. |
| 03:24.52 | logyati | lol |
| 03:25.19 | logyati | this is the cheapest one right? |
| 03:25.20 | logyati | hehe |
| 03:25.27 | [TK]D-Fender | ~cheap |
| 03:25.28 | jbot | it has been said that cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 03:25.32 | [TK]D-Fender | logyati, ... yup |
| 03:25.37 | logyati | ~wiki |
| 03:25.39 | [TK]D-Fender | ~ygwypf |
| 03:25.40 | jbot | well, ygwypf is You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 03:25.45 | [TK]D-Fender | ~wikis |
| 03:25.46 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 03:26.58 | logyati | [TK]D-Fender, i wanna buy the cheaper one just for home use |
| 03:27.01 | logyati | testing purposes |
| 03:27.03 | logyati | hehe |
| 03:27.37 | *** join/#asterisk harryv (n=harry@0x55508034.adsl.cybercity.dk) |
| 03:27.43 | [TK]D-Fender | logyati, what is "testing purposes"? |
| 03:27.54 | angryuser | i had one to learn how * work, a year ago |
| 03:27.59 | logyati | [TK]D-Fender, i ment learning purposes |
| 03:28.00 | logyati | hehe |
| 03:28.21 | logyati | i wont buy a tdm400p just to learn hehe |
| 03:28.22 | harryv | hey. do any of you have any idea why i am able to register to my voip-provider from asterisk on my mac, but using the exact same sip.conf on a fresh debian install on the same LAN i get a time out all the time. |
| 03:28.26 | [TK]D-Fender | logyati, Nothing to learn from one of those expecpt pain. if you jsut want to use your line, get an SPA-3102 |
| 03:28.27 | logyati | too expensive here |
| 03:28.38 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:28.38 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 03:28.51 | logyati | spa-3102? |
| 03:29.02 | logyati | is it a card or something else? sry my question hehe |
| 03:29.03 | [TK]D-Fender | logyati, yes |
| 03:29.34 | *** join/#asterisk ecto (n=ectospas@c-71-207-229-248.hsd1.al.comcast.net) |
| 03:29.43 | [TK]D-Fender | harryv, bad settings in sip.conf or firewall settings |
| 03:30.07 | harryv | no firewall, and exactly the same sip.conf as on my mac.. |
| 03:30.09 | logyati | spa-3102 is an ATA |
| 03:31.14 | [TK]D-Fender | harryv, if those machines are behind NAt then the same SIP.CONF can't work as each would have its own IP and thus the externIP wouldn't be set right. Then there is the question of forwarding from your outside router, etc. |
| 03:31.54 | harryv | no forwarding should be needed. |
| 03:32.12 | harryv | and theres no externip-setting. tried to set it. no difference. |
| 03:33.25 | supjigator | no errors just D channel up over and over. Anyone have any ideas? |
| 03:35.43 | logyati | ok im gonna buy a x100p :P |
| 03:36.34 | logyati | well i gotta go sleep, its 00:30 here |
| 03:37.03 | logyati | cya tomorrow! |
| 03:50.11 | LiNeTuX | [TK]D-Fender: any opinion about Audiocodes? Like the MP-114-FXO? I need another 4 FXO's. |
| 03:52.36 | *** join/#asterisk ionix (n=ionix@p3104-ipbfp402motosinmat.mie.ocn.ne.jp) |
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| 04:34.53 | mocker | When I'm turning up a new channel on a Sangoma PRI card, do I just need to add it in zaptel/zapata? |
| 04:35.01 | mocker | Or do I need to modify the wanpipe stuff? |
| 04:39.55 | *** part/#asterisk galeras (n=Martin@200.118.198.26) |
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| 04:43.59 | JT | mocker: new channel or new span? |
| 04:44.07 | mocker | JT: New span |
| 04:44.17 | mocker | adding a PRI into the mix. |
| 04:44.40 | mocker | Just want to make sure there is no sangoma specific stuff to it, and it's just normal zaptel/zapata configs. |
| 04:49.58 | mosty | you need to setup wanpipe too |
| 04:52.38 | JT | indeed |
| 04:54.21 | mocker | Ahh, so /etc/wanpipe/wanpipeX.conf |
| 04:54.27 | mocker | Where X is a span on the card? |
| 04:55.41 | *** join/#asterisk ecto (n=ectospas@c-71-207-229-248.hsd1.al.comcast.net) |
| 04:55.55 | JT | mocker: just use the config generator. |
| 04:57.22 | *** join/#asterisk jameswf-home (n=james@ip72-204-221-181.ph.ph.cox.net) |
| 04:57.26 | mocker | JT: did that. |
| 04:57.30 | mocker | Thanks. |
| 04:57.38 | mocker | But am I right that they correspond like that? |
| 05:00.19 | jameswf-home | pong |
| 05:02.57 | mocker | Trying to configure 4 trunks as NFAS, should be a blast. |
| 05:03.00 | mocker | </sarcasm> |
| 05:03.01 | mocker | :) |
| 05:03.29 | JT | mocker: this in the US? |
| 05:03.43 | mocker | JT: Yup. |
| 05:03.56 | mocker | Doing a bridge from an Avaya system to the Asterisk system. |
| 05:04.03 | mocker | But want the trunk to appear as one channel. |
| 05:04.14 | mocker | Wow, that last line made no sense. |
| 05:04.26 | mocker | But want the trunks to to appear as one. :) |
| 05:05.10 | jameswf-home | ~book |
| 05:05.11 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 05:05.39 | harryv | uh, why do i get this error while `make install` zaptel: |
| 05:05.39 | harryv | <PROTECTED> |
| 05:05.40 | harryv | <PROTECTED> |
| 05:05.51 | harryv | i tried to google it. can't really find out anything substantial |
| 05:07.24 | JT | mocker: if you put them in the same zap group, same effect ;) |
| 05:07.27 | mocker | That looks like a warning, not an error. |
| 05:07.34 | mocker | JT: Not on the Avaya side. :( |
| 05:07.43 | vk4akp | I solved the whole gentoo thing |
| 05:07.55 | vk4akp | gentoo /asterisk /meetme |
| 05:07.56 | mocker | vk4akp: Good, those Gentoo users were getting on my nerves. |
| 05:08.03 | mocker | Oh, nevermind. |
| 05:08.09 | vk4akp | so if anyone needs help, send em my way |
| 05:08.19 | mocker | :P |
| 05:08.24 | vk4akp | ;) |
| 05:08.29 | jameswf-home | ~avaya |
| 05:08.29 | jbot | i guess avaya is some big company that equals Micro$oft in phone systems |
| 05:08.47 | mocker | jameswf-home: Migrating a large company is a slow and painful process. |
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| 05:20.34 | b11d | . |
| 05:20.52 | jameswf-home | , |
| 05:21.12 | mocker | b11d: Remember that problem I was haivng w/ the IP330s? |
| 05:22.06 | mocker | Turns out that the problem was someone deciding to us Ghost on our network in multicast mode, all of the ports on our network starting getting 50+ meg of traffic. |
| 05:22.15 | mocker | That's when the phones started locking or rebooting. |
| 05:22.49 | Mavvie | wasn't the idea that with multicast you subscribe to something and that the switch doesn't send it before you subscribe to it? |
| 05:23.05 | sweeper | depends on the kind of multicast |
| 05:23.13 | sweeper | iirc, anyways |
| 05:23.20 | mocker | Mavvie: I don't think it was *designed* to work like this. :) |
| 05:23.30 | hmodes | hrmm, is there an accepted tftpd that sources everything from port 69? |
| 05:24.32 | sweeper | I'm pretty sure port should be configurable :P |
| 05:24.34 | Mavvie | hmodes: it uses port 69 to send the first packet to, and gets a higher port number back from the server. |
| 05:25.10 | hmodes | yeah, nats don't like switching ports |
| 05:25.37 | sweeper | obviously, the easy solution is to change everything to ipv6 |
| 05:25.44 | hmodes | i've been using the same tftpd binary that doesn't ever switch ports for years, just wondering if there's one out there that has that built in |
| 05:26.06 | *** join/#asterisk MaliutaWrk (i=nikolai@119.11.102.38) |
| 05:26.42 | hmodes | seems like it should be a mainstream feature by now... |
| 05:29.21 | Mavvie | Is there, just like the test pattern for the TV, a test pattern for the facsimile system? |
| 05:31.24 | jameswf-home | Mavvie: http://www.pcstats.com/articleview.cfm?articleid=1339&page=4 |
| 05:33.14 | Mavvie | jameswf-home: I've found the IEEE test page (http://ltt-www.lcs.mit.edu/ltt-www/Public/Images/IEEE_Std_167A-1987.75x8f400x8.gif), now looking for the high resolution one. |
| 05:35.20 | hmodes | that looks much nicer then a fax is capable of reproducing to me... |
| 05:36.28 | hmodes | contrast is a bit screwed up tho' |
| 05:36.43 | Mavvie | hmodes: of course. It is to show of how much better your $faxproduct is compared with the one from the compitition. |
| 05:38.09 | hmodes | well, what i meant was with a bit of cleanup that could be used as a test page pretty easily |
| 05:38.20 | hmodes | unless you really need the original for some reason |
| 05:49.50 | hmodes | okay, i take that back |
| 05:50.09 | hmodes | it prints shitty, and i just noticed the 'printed by kodak' bit |
| 05:50.44 | hmodes | it's a shame if that's a lost image |
| 05:57.19 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 06:31.22 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 06:31.51 | pkunkra | hello |
| 06:38.04 | *** join/#asterisk jerlique (n=jerlique@lnk2.adl.adsl.esc.net.au) |
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| 06:41.10 | *** join/#asterisk nvrpunk (n=root@80.78.22.134) |
| 06:41.27 | nvrpunk | is there a command to stop and start voicemail? |
| 06:41.30 | nvrpunk | under asterisk |
| 06:42.03 | HaMYaI | is there a way to use a cron job to execute an agi script to call defined numbers let them listen to a voice prompt and hangup? |
| 06:43.39 | HaMYaI | I normally execute agi script from extensions.conf, don't know it's gonna work by executing from crons |
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| 06:45.08 | HaMYaI | will try with manager api anyway |
| 06:45.17 | Mavvie | HaMYaI: call-files. |
| 06:47.08 | HaMYaI | Mavvie: yeah, but it's hard to control the calls that way |
| 06:49.31 | HaMYaI | Mavvie: with agi, I think we can do more to the calls |
| 06:50.26 | Mavvie | HaMYaI: agi with callback scripts doesn't work anyway. |
| 06:50.35 | Mavvie | See http://bugs.digium.com/view.php?id=11917 |
| 06:51.37 | supjigator | Anyone running a sangoma with 1.4? |
| 06:51.52 | HaMYaI | Mavvie: what about with 1.2.x? |
| 06:52.06 | Mavvie | HaMYaI: it worked in 1.2 |
| 06:52.55 | HaMYaI | Mavvie: ok |
| 06:53.24 | nvrpunk | is there a command to turn off voicemail |
| 06:53.27 | nvrpunk | in asterisk |
| 06:53.46 | nvrpunk | or is there a place in asterisk.conf where the messages get left? |
| 06:53.49 | nvrpunk | ie written |
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| 07:37.33 | mocker | Woo, failing miserably at an NFAS trunk! |
| 07:38.59 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 07:39.14 | joelsolanki | Hi Room |
| 07:39.38 | *** join/#asterisk marc7 (n=marc@64.46.14.64.novuscom.net) |
| 07:39.45 | joelsolanki | i have sangoma 104D card. I want to use ss7 signalliing in asterisk |
| 07:40.01 | joelsolanki | would anyone recommend the working ss7 implementation with sangoma cards. |
| 07:40.07 | joelsolanki | i m looking for opensource solutions |
| 07:46.40 | nvrpunk | how do I make it so that asterisk accepts g729 for voicemail? |
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| 08:00.54 | Shaun2222 | has the issues with dial()+Macro's and wait()/waitextn() been resolved? |
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| 08:12.53 | Viras | Hi @ all |
| 08:12.56 | Viras | Anyone alive? |
| 08:13.36 | Viras | I got a question which may sound stupid but I will ask it anyway :) |
| 08:14.51 | Viras | I have the following setup: a single phone line coming in. I want to connect a fax to it using an extension, and a normal phone which should ring if no extension is dialed. All devices are "normal" phone devices. Is that possible with asterisk? |
| 08:14.52 | *** join/#asterisk lftsy (n=lftsy@120.194.210.62.te-dns.org) |
| 08:16.44 | creativx | fax over voip |
| 08:16.45 | creativx | ... |
| 08:16.48 | Viras | In addition, if I understand it correctly I need three jacks for phones on my PC then, I guess I need a special ISDN card then or? |
| 08:16.52 | Viras | no - normal fax |
| 08:17.03 | Viras | so my fax device is a normal phone device |
| 08:17.18 | Viras | I just would need asterisk to create the extension and pass it to the fax if it gets dialed |
| 08:18.31 | sweeper | Viras: I would actually suggest a fax stick |
| 08:18.39 | sweeper | and then networked phones |
| 08:19.04 | sweeper | either with ATA or an actual voip handset |
| 08:20.31 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 08:21.06 | Viras | I see - well the problem is all the hardware is in place - I'm just trying to decide if I should setup an asterisk server or buy a simple telephone switchboard (or however it is called in english :)) |
| 08:21.50 | sweeper | well, you're gonna have to buy hardware anyways :) |
| 08:22.16 | ionix | don't go with asterisk for that... Fax over IP is not very robust and requires tweaking |
| 08:22.45 | sweeper | if all you need is one phone, and one fax, and no voicemail or menus or anything, just get a fax stick and don't use asterisk at all |
| 08:25.05 | Viras | does a fax stick allow extensions? |
| 08:25.54 | Viras | (in fact I wanted to use asterisk just to get into it and finally start understanding it) |
| 08:26.22 | joelsolanki | i have sangoma 104D card. I want to use ss7 signalliing in asterisk |
| 08:26.29 | joelsolanki | would anyone recommend the working ss7 implementation with sangoma cards. |
| 08:26.35 | joelsolanki | i m looking for opensource solutions |
| 08:30.16 | joelsolanki | any ideas plz |
| 08:34.09 | agx | Viras, if you have 1 analog line and wanna share it with phone and fax i usually suggest a 30€ samsung fax machine with voice maessages too :) |
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| 08:40.18 | *** join/#asterisk Anthony_RR06 (n=Anthony@LAubervilliers-153-52-32-150.w217-128.abo.wanadoo.fr) |
| 08:40.34 | Anthony_RR06 | hello all. |
| 08:43.06 | Anthony_RR06 | I use sipx with Asterisk. Asterisk is a "proxy" (my SBC). It's works fine for incoming and outgoing calls. When I press the on Hold on my phone for internal call, it's ok. But when I execute the on old with external call, the music on old playing on sipx and also in asterisk. How can I disable the musiconhold in asterisk? Or how can I do to undetect the musiconhold in Asterisk? Thanks (I work with Asterisk 1.2. |
| 08:43.15 | Anthony_RR06 | thanks for your help |
| 08:53.30 | Anthony_RR06 | In log I have this: Music class default requested. How can I disabled this detection? |
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| 09:15.03 | imgrey | morning #asterisk |
| 09:15.15 | mort_gib | Hi, how do I prevent extensions.ael to be loaded?? |
| 09:15.24 | imgrey | s it exists default test number in asterisk ? |
| 09:19.24 | HaMYaI | can we actually use cmd Dial on CLI console? |
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| 09:28.53 | nebojsajsimic | little quick help need how to set file for MOH |
| 09:29.25 | nebojsajsimic | in extension.conf |
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| 09:32.19 | KostasA | HaMYal: No, I don't think you can use any command on the CLI. |
| 09:36.57 | nebojsajsimic | any help |
| 09:40.13 | KostasA | mort_gib: Inside modules.conf: noload => pbx_ael.so |
| 09:40.20 | KostasA | This will disable AEL support. |
| 09:40.41 | nebojsajsimic | does any know how to set files for MOH??? |
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| 10:06.42 | vk4akp | I'm gettign a lot of SIp calls staying on the system after people hang up and it chops up the audio. Any idea's? |
| 10:07.37 | defswork | vk4akp: I get that |
| 10:15.54 | *** join/#asterisk tsubasafr (n=sylvain@ABordeaux-156-1-48-134.w86-217.abo.wanadoo.fr) |
| 10:16.07 | tsubasafr | hello ! |
| 10:16.19 | tsubasafr | I have a problem with asterisk and mpg123 |
| 10:16.30 | tsubasafr | When I make linux mpg123 it fail without error |
| 10:16.35 | tsubasafr | http://rafb.net/p/IAFeTF65.html |
| 10:17.09 | defswork | vk4akp: I've not found the answer yet |
| 10:17.30 | defswork | vk4akp: when it happens (at the one site only) I clear the call in the cli |
| 10:24.20 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 10:30.45 | vk4akp | OK. Interesting. |
| 10:31.04 | vk4akp | I'm trying only registering that outbound sip client with Asterisk. |
| 10:31.14 | vk4akp | Previously it was also registered with my ATA as a fall back. |
| 10:31.17 | vk4akp | See if that helps. |
| 10:31.32 | *** join/#asterisk sx|lappy (n=sxpert@lgit-1225.obs.ujf-grenoble.fr) |
| 10:33.50 | *** join/#asterisk shinao1 (n=shinao1@41.221.165.57) |
| 10:35.40 | *** join/#asterisk shinao1 (n=shinao1@41.221.165.57) |
| 10:36.41 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 10:37.35 | festr_ | how is possible to get DTMF on SIP channel if there is no hearable dtmf and no rfc or info msgs? (i'v sniffed it). asterisk[11971]: DTMF[15261]: DTMF end '5' received on SIP/802cz-082ea7d8, duration 800 ms |
| 10:37.47 | festr_ | this issue is terrorising me a few days :) |
| 10:37.50 | festr_ | asterisk 1.4 |
| 10:37.53 | festr_ | any idea? |
| 10:38.24 | *** join/#asterisk SteveTotaro (n=root@pool-71-166-102-100.bltmmd.east.verizon.net) |
| 10:42.21 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 10:43.36 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 10:45.43 | patrick-- | Hey, ive got 1 BN4S0 HFC Card in my asterisk Machine. Port 1 is used as TE for Outgoing connections, while Ports 2-4 are used for ISDN phones within the company. I successful wrote a dialplan so that everyone is able to call out/intern etc. How can i insert callerID's so that the called person can see who is calling him (by name) |
| 10:47.10 | patrick-- | i know that there is a callerid function, but im not sure how to insert it into my dialplan |
| 10:47.15 | defswork | festr_: you've not got relaxed dtmf in sip.conf have you (added in 1.4) |
| 10:48.06 | defswork | festr_: set dtmf to SIP info or something like that iirc so its not audible tone triggered |
| 10:48.32 | festr_ | defswork: dsp is not involved, i've made brakepoints to dsp.c to proove it |
| 10:49.06 | festr_ | defswork: both sip channels are alaw with dtmf=sipinfo which prevents inband detection. |
| 10:51.13 | defswork | festr_: is that a sip to sip call ? |
| 11:02.32 | HaMYaI | anyone knows if we can use "Action: Command \r\nCommand: Dial(...)" with manager api? |
| 11:05.54 | *** join/#asterisk Datax (n=Datax@glou.nurvnet.org) |
| 11:05.58 | Datax | Quick question, what do I need to be able to dial/receive calls via a PSTN line ? I want PSTN on one side and SIP on the other |
| 11:06.02 | Datax | it's for a home usage |
| 11:06.12 | Datax | and I'm new to asterisk so no flames plz :) |
| 11:06.24 | Mavvie | ~beginner |
| 11:06.35 | Mavvie | no such thing.... |
| 11:06.40 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:07.39 | Datax | what ? doesn't such a "box" exist ? I can't easily use a PCI card because I'm currently running asterisk on a virtual server and intend moving it to my openwrt router |
| 11:08.41 | defswork | Datax: spa3102 |
| 11:08.51 | Datax | thxs I'll look it up |
| 11:09.35 | *** join/#asterisk guillote_GNU (n=guillote@host157.201-253-55.telecom.net.ar) |
| 11:12.34 | *** join/#asterisk nighty^ (n=nighty@p3047-adsau16honb13-acca.tokyo.ocn.ne.jp) |
| 11:24.11 | *** join/#asterisk quigon (n=matias@32.59.64.130) |
| 11:25.26 | vk4akp | Let us know how you go with that. I have 3x WRT54GS Rev 1.0's here with the large RAm. I am interested in putting Asterisk on some of them. (another 3 on the way as well!) :) |
| 11:28.00 | Datax | vk4akp: have you configured some kind of failover between them ? |
| 11:28.00 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:28.43 | *** part/#asterisk HaMYaI (n=LAMER@ppp-58-8-12-74.revip2.asianet.co.th) |
| 11:29.17 | Datax | vk4akp: because that's one of the things that worries me with the idea of running my own sip server |
| 11:29.28 | Datax | if it goes down ... baddddd things will happen ;) |
| 11:33.36 | vk4akp | I bought them to start a mesh in our area. |
| 11:33.57 | vk4akp | I also have an AMateur Radio TV and Voice repeater on a hilltop that I want to add a net feed to. |
| 11:34.05 | vk4akp | For APP_RPTR and local mesh. |
| 11:35.16 | *** join/#asterisk MaliutaWrk (i=nikolai@119.11.108.181) |
| 11:37.26 | J4k3_ | vk4akp: do yourself a favor and get some decent radios |
| 11:37.55 | J4k3_ | those, while cute from a software aspect, have some of the worst RF performance of any 802.11g gear |
| 11:39.30 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
| 11:45.24 | festr_ | defswork: yes it is |
| 11:46.05 | nebojsajsimic | hi all moh start and stop imidiotly can somone help |
| 11:46.06 | nebojsajsimic | ??? |
| 11:48.03 | nebojsajsimic | <PROTECTED> |
| 11:48.03 | nebojsajsimic | <PROTECTED> |
| 11:48.21 | vk4akp | J4K3 (As in Deaf?) |
| 11:48.33 | vk4akp | I've got Amps on them all to increase the RX sensativity. |
| 11:48.39 | J4k3 | hahahaha |
| 11:48.47 | J4k3 | wifi doesn't quite work that way |
| 11:49.00 | vk4akp | It's a radio like any other radio. |
| 11:49.06 | vk4akp | Wide as hell. |
| 11:49.20 | J4k3 | you're increasing the RX noise floor, it *appears* to give you more signal when in reality you're just giving yourself more headaches. |
| 11:49.27 | vk4akp | I've played with TV broadcasing. So I'm aware of how the wider signals are even more prone to signal to noise isues. |
| 11:49.43 | vk4akp | Yes, but you get that with any amp. |
| 11:49.48 | J4k3 | ntsc can survive a lot more than 3% evm. |
| 11:49.56 | vk4akp | Idealy U want to filter b4 you amplify. |
| 11:50.17 | vk4akp | BUt then teh filter has insertion loss. So doesn't matter how U look at it ther are issues. |
| 11:50.30 | vk4akp | I did tests anyhow, and the amps definatly improved the receive. |
| 11:50.51 | vk4akp | But anyhow, yes. Embedded PC's and Mini PCI cards would most likely be better. BUt they are way too expensive for me. |
| 11:51.06 | vk4akp | The Linksys's I got for 10GBP each + freight. |
| 11:51.06 | J4k3 | but cheezy radios and amps aren't? |
| 11:51.10 | J4k3 | you're why god made guns. |
| 11:51.22 | J4k3 | :P |
| 11:51.34 | J4k3 | (so you can shut down noisy noob-operated gear) |
| 11:52.16 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 11:53.08 | J4k3 | you've got a few thousand 802.11-based wisps out in the world before you.... there are two kinds of 802.11-based wisp. |
| 11:53.59 | J4k3 | wisps that buy shit gear then amplify their noise and can ping in every bathroom in the whole damned town, but can't break 50kB/sec real world throughput on their radios due to their own noise, and are trying to sell and run as fast as possible |
| 11:54.10 | J4k3 | and theres people who do a little more network planning... |
| 11:56.54 | vk4akp | The whole world isn't rich like you. |
| 11:56.57 | vk4akp | They use what they can get. |
| 11:57.06 | J4k3 | amplifiers are free? |
| 11:57.23 | J4k3 | I mean, if they were priced for their actual value they would be near that, but... |
| 11:57.25 | vk4akp | I could take it one step further and tell U your geat is shit compared to commercial Cisco gear. |
| 11:57.54 | vk4akp | LIke I said. I picked these wrt's up for 10GBP each. About US$20. |
| 11:58.01 | J4k3 | yeah, and...? |
| 11:58.04 | vk4akp | well worth the value. |
| 11:58.16 | vk4akp | Gets 6 nodes up on teh air that can be upgraded later if money allows. |
| 11:58.22 | J4k3 | yeah...for putting on your shelf and using as a home router |
| 11:58.58 | vk4akp | They work, we're happy. U can winge if U like, I'd rather do more productive things. |
| 11:58.58 | J4k3 | are you one of those dorks that buys a bigass amplifier and couple cheap chink HTs and pretends they're running a repeater? |
| 11:59.23 | J4k3 | well, my network would break yours... so you'd be sitting around pretty pissed off :) |
| 11:59.39 | J4k3 | omgwhyismynoisefloorsohigh |
| 12:00.45 | J4k3 | in fact, I have a customer, who, right now, as we speak, is connected to me with the exact set you're talking, except its the slightly superior v2 radio. |
| 12:00.56 | J4k3 | 250mW teletronics amp |
| 12:01.07 | J4k3 | and a 25 dbi parabolic pointing back at me, clear LOS, ~2 miles |
| 12:01.41 | J4k3 | I can hear his dinky ass CPE from 2360 to 2475, when associated to an AP on ch1 |
| 12:02.07 | *** join/#asterisk SteveTotaro (n=root@pool-71-166-102-100.bltmmd.east.verizon.net) |
| 12:02.14 | Wayhigh | wassup steve |
| 12:02.20 | J4k3 | my gear, while typically 150mW higher peak txpower (if selected) at the card itself, never bloom wider than 25 mhz. |
| 12:02.24 | J4k3 | nevereverever. |
| 12:05.25 | J4k3 | I hate it when people ghettothink rf related projects and/or act like they're personally allowed to degrade the very few publically-accessable radio bands available. |
| 12:05.27 | J4k3 | stick to making 2 meter pathetic. |
| 12:06.48 | vk4akp | horses 4 courses. |
| 12:06.54 | vk4akp | not everyone is rich |
| 12:07.35 | vk4akp | 99% of hobiest wifi mesh's in this country run from linksys or Minitar etc AP's |
| 12:08.02 | J4k3 | if you're poor, you don't need to play trashmesh |
| 12:08.07 | J4k3 | buy some post toasties for your kids |
| 12:08.09 | J4k3 | and some milk |
| 12:08.20 | J4k3 | and, quite simply, as I said already |
| 12:08.22 | vk4akp | I have no kids. |
| 12:08.24 | J4k3 | amplifiers are free? |
| 12:09.35 | J4k3 | you can buy really nice minipci radios capable of -95ish dBm RX and +26 dBm TX for $39 USD each |
| 12:09.45 | J4k3 | I belive they're even cheaper in europe |
| 12:09.50 | J4k3 | er believe |
| 12:10.13 | vk4akp | The cards may be reasonably priced, but U still ned very expensive embedded PC's to run them. |
| 12:10.36 | J4k3 | routerboard 411, $59 USD. |
| 12:10.49 | J4k3 | a product of europe, maybe cheaper there also |
| 12:10.56 | vk4akp | IN time we *mey* expand teh mesh to use more expensive equipment. But for now it's a hobby, done at a hobbiest level with hobbiest pricing. |
| 12:10.56 | J4k3 | 411 also has considerably more cpu |
| 12:11.20 | jerlique | Is there a way to display on polycom phones the status of other internal extensions (ie if they are on the phone etc)? |
| 12:11.40 | J4k3 | and you sir, are why, I refuse to get a ham license. |
| 12:13.00 | mosty | jerlique, yeah, look for the string "buddy" in the pdf manual |
| 12:13.05 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 12:13.27 | jerlique | thanks |
| 12:13.33 | J4k3 | every time I talk to a ham about data, they immediately act like they know everything... and they insist on doing it completely trashy |
| 12:13.49 | J4k3 | yet you walk into their shack and they'll have a $2500 hf rig sitting on their bench |
| 12:14.11 | vk4akp | U R so funny. |
| 12:14.21 | J4k3 | and its like... whats the fucking difference. you could be working HF with a hacked up heathkit basket case bought off ebay for $50, but you're not |
| 12:14.27 | vk4akp | Obviously just an attitude toward Amateur Radio operators. |
| 12:14.42 | J4k3 | yeah, they're know-it-all amateurs |
| 12:14.42 | vk4akp | Whats wrong? Couldn't U pass the test? LOL |
| 12:14.45 | J4k3 | professional assholes. |
| 12:14.55 | J4k3 | the test is a joke |
| 12:14.57 | J4k3 | heh |
| 12:15.08 | J4k3 | hence why 2 meter is jammed up full of cb-style chatter |
| 12:15.12 | J4k3 | even out in the middle of nowhere in the USA |
| 12:15.20 | J4k3 | 70cm catching right up |
| 12:15.43 | J4k3 | 6m is quiet because the antennas are too unsightly on their wife's SUV. |
| 12:16.00 | J4k3 | 10 meter and down is where the old people argue. |
| 12:16.15 | Dovid | good morning EV1 |
| 12:17.08 | vk4akp | LOL |
| 12:17.55 | vk4akp | As for my HF ring it's an old FT-890AT that I have had since B4 I was licenced. So no I don't spend a fortune on my HF radio instead of buying unnecessarily expensive WiFi gear. |
| 12:18.36 | vk4akp | And our bands are nothing like your US chook band's. V/UHF is very respectable here. |
| 12:19.00 | vk4akp | And your attitude to low end users is a waste of time. |
| 12:19.32 | vk4akp | Fon are spreading the low end $39 AP's around the world promoting free WiFi meshing. I'd like to see U take on them and tell them they can't do it either. |
| 12:19.38 | *** join/#asterisk simbol76ss (n=simbol@host65-209-dynamic.3-87-r.retail.telecomitalia.it) |
| 12:19.43 | J4k3 | well, I still don't see the value in purchasing *amplifiers* to hang on junk gear to attempt to make it appear sightly less craptastic on the surface |
| 12:19.47 | J4k3 | the economy doesn't work out |
| 12:20.07 | vk4akp | Lets face it . 802.11 was never ment as an elitist linking system. It's a public park for all to use from all walks of life. |
| 12:20.16 | J4k3 | the cheapest bidirectional amplifier I know of with onboard filters is $100 USD for the loaded pcb alone. |
| 12:20.30 | vk4akp | And I'm sure there are many many more lower quality appliences on the market and in use then what I'll ever set up. |
| 12:20.40 | J4k3 | well the problem for you is 802.11 coexists with a lot of other technology which doesn't peacefully co-exist |
| 12:21.06 | J4k3 | and when you're working with a csma/ca-based network protocol, noise is your enemy |
| 12:21.50 | vk4akp | Our Amplifiers are 500mWatt and only cost us $50 each new. |
| 12:21.50 | vk4akp | They are mainly used for their RX value as we are well aware of the deafness of the older equipment. |
| 12:21.50 | J4k3 | have you actually tried those outdoors on real antennas? |
| 12:21.56 | vk4akp | And yes I am aware it amplifies the noise floor as well. But it still works and increases teh RX ability. |
| 12:22.12 | vk4akp | Yes, I am running one right now here as we speak. |
| 12:22.16 | J4k3 | if theres *anything* else near 2.4, you're gonna lock the radios down with csma/ca |
| 12:22.40 | vk4akp | I have many other AP's in teh street. Still works fine. |
| 12:22.47 | vk4akp | Even with Co-Channel stuff. |
| 12:23.16 | J4k3 | *shrug* I'm just going to have to assume your version of fine and mine are different. |
| 12:23.29 | vk4akp | No not at all. |
| 12:23.32 | J4k3 | I get pissed when my links jitter more than 10ms. |
| 12:23.41 | vk4akp | I have put together a TV station at my Ham repeater site. |
| 12:23.43 | J4k3 | and any packet loss is pretty much unacceptable. |
| 12:23.47 | vk4akp | I am well aware of what U R saying. |
| 12:24.04 | vk4akp | In an ideal world, one strives for the very best of quality and setup with everything. |
| 12:24.04 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 12:24.28 | vk4akp | But I and others do not live in an ideal world where we can use $100 notes to light our Cigar's with. |
| 12:24.41 | J4k3 | well |
| 12:24.50 | J4k3 | when you've got legitimate need |
| 12:24.52 | vk4akp | Well, if U have the money. Fine good on U. Run the best. |
| 12:24.53 | J4k3 | you need reliability |
| 12:25.04 | J4k3 | and when somebody's holding lit $100s under your ass |
| 12:25.12 | vk4akp | I'd like a commercial exciter for the TV station also. But thats never goign to happen. |
| 12:25.15 | J4k3 | you tend to do whatever is required to keep it from happening too often. |
| 12:25.27 | vk4akp | So instead we just use a good (yet still home brewed) VSB filter and live with it. |
| 12:25.30 | J4k3 | a commercial exciter? like a vcr modulator? :D |
| 12:25.34 | *** join/#asterisk lftsy (n=lftsy@120.194.210.62.te-dns.org) |
| 12:26.20 | J4k3 | I'm wondering why VHF channels 3 and 4 (what our standard VCR modulators are) haven't turned into TV pirate radio heaven. |
| 12:26.26 | vk4akp | Some of our older exciters have been VSB TV modulators, that are then re-filtered at the exciter level, and then again later after amplification. |
| 12:26.30 | vk4akp | Cheap, and works well. |
| 12:26.37 | vk4akp | U just have the wrong attitude to things. |
| 12:26.46 | vk4akp | U think the mighty $'s rules everything. |
| 12:26.51 | J4k3 | eh, I just know how bad those radios are |
| 12:26.56 | J4k3 | those are *bad* radios |
| 12:27.05 | J4k3 | on the grand scheme of crappy consumer wifi gear |
| 12:27.08 | J4k3 | those things suck |
| 12:27.34 | J4k3 | not quite as bad as a zyxel adsl wifi router (which is practically a RF-based denial of service attack) |
| 12:27.37 | J4k3 | but close ;) |
| 12:27.54 | J4k3 | the self-installed customer I have using that setup makes a ton of noise |
| 12:27.59 | vk4akp | Oh well, they'll do to start us out. |
| 12:28.16 | vk4akp | It's only a matter of unplugging them and pluggin gsomethign else in when it becomes available and within our reach. |
| 12:28.27 | J4k3 | and actually lowers the performance of my AP site. I need to threaten to unlink him til he removes the amp |
| 12:28.46 | *** join/#asterisk lftsy (n=lftsy@120.194.210.62.te-dns.org) |
| 12:29.00 | vk4akp | If you want business level microwave links. Pay for your own dedicated spectrum. |
| 12:29.10 | J4k3 | why? |
| 12:29.20 | vk4akp | After all, money does not seem to be an issue with you. So wy are you bothering with 2.4Ghz? |
| 12:29.28 | J4k3 | when people aren't assholes about using it, the free spectrum works fine |
| 12:29.47 | vk4akp | Exactly. |
| 12:29.54 | vk4akp | So stop being an asshole about it! ;;) |
| 12:30.05 | J4k3 | you have neither the god given or the legal right to fuck up somebody else's radio experience with your crappy gear. |
| 12:30.20 | vk4akp | Err, wrong. |
| 12:30.25 | vk4akp | Public spectrum. |
| 12:30.30 | vk4akp | Public appliences. |
| 12:30.31 | J4k3 | there are laws |
| 12:30.39 | vk4akp | That meet local laws and requirements. |
| 12:30.42 | J4k3 | quite strict ones in the eu. |
| 12:30.45 | vk4akp | yep, every right! |
| 12:30.52 | J4k3 | nope, not every right |
| 12:30.54 | vk4akp | I'm not in the EU! |
| 12:31.05 | J4k3 | you're in the UK, close enough |
| 12:31.13 | vk4akp | NO. |
| 12:31.18 | vk4akp | See U can't even get that right! |
| 12:31.43 | J4k3 | heh, .au has laws too |
| 12:32.18 | J4k3 | see, thats the thing |
| 12:32.20 | vk4akp | Yep, and we abide by them fully! |
| 12:32.30 | J4k3 | hams here... they think they can take a pile of shit and throw it on the air without any testing and its legal |
| 12:32.50 | vk4akp | OK. Here is an example that I'm sure U will appreciate. |
| 12:32.56 | vk4akp | Ready? |
| 12:33.02 | J4k3 | and when they're spewing nasty harmonics from voice to light, 'unknowingly', its just a little oopsie. |
| 12:33.15 | simbol76ss | |
| 12:33.15 | simbol76ss | Hello everyone. |
| 12:33.15 | simbol76ss | Someone tell knows what form should I use 1.4 asterisk for the codec g729 on a Xeon processor? |
| 12:33.33 | *** join/#asterisk lftsy (n=lftsy@120.194.210.62.te-dns.org) |
| 12:33.52 | vk4akp | 500mWatt amp, switched back to 100mWatt, with teh WRT wound right down to drive it. This drives teh amplifiers at a better class level and gives cleaner signal!. |
| 12:34.08 | vk4akp | We are aware of these issues. |
| 12:34.14 | vk4akp | TV is just as bad. |
| 12:34.25 | J4k3 | do you own a spectrum analyzer that can do any sort of decent detection in the 2.4ghz range? |
| 12:34.30 | vk4akp | Drive the amps out of their linear range and the picture is terrible. |
| 12:34.42 | vk4akp | They are too expensive. |
| 12:34.46 | J4k3 | a real spectrum analyzer, not some crappy fhss card or a hacked up proprietary wireless mouse dongle |
| 12:34.47 | vk4akp | No we don't own them. |
| 12:34.51 | *** join/#asterisk MaliutaWrk (n=nikolai@119.11.108.181) |
| 12:35.05 | vk4akp | But we do get eh opportunity to use them when military members etc bring them from time to time. |
| 12:35.09 | J4k3 | so, you're throwing gear together with no knowledge of what noise you're creating outside your allowed license bands |
| 12:35.18 | vk4akp | So yes we are well aware of what settigns to use to minimise distortion. |
| 12:35.27 | vk4akp | Distortion helps no one. |
| 12:35.37 | vk4akp | No, read what I am typing. |
| 12:35.50 | J4k3 | in theory what you're saying is right |
| 12:35.54 | vk4akp | Put mouth in neutral for a while an duse eyes. instead. |
| 12:36.03 | J4k3 | in practice those crappy linkydink radios still suck profusely at any setting |
| 12:36.07 | coppice | nah. use constant amplitude modulation, and don't worry :-) |
| 12:37.38 | vk4akp | OUr TV amps are home brew as well. (MRF648 transistors). They are capable of 60Watts. We run them at 18Watts to get the best linear results. |
| 12:38.18 | J4k3 | well |
| 12:38.22 | J4k3 | this wifi crap is kinda different |
| 12:38.31 | J4k3 | for example |
| 12:38.42 | J4k3 | are your amplifiers fixed tx gain? |
| 12:38.45 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.239) |
| 12:38.47 | J4k3 | or do they feature any sort of AGC? |
| 12:38.56 | vk4akp | Fixed. |
| 12:39.06 | vk4akp | What ever U drive them with dictates teh outcome. |
| 12:39.08 | simbol76ss | I have the server Dell with Xeon processor, I'll want install the g729 codec,which module for asterisk 1.4 for Xeon??? |
| 12:39.17 | vk4akp | They aren't just auto AGC running flat out like some. |
| 12:39.20 | J4k3 | agc = evilevilevil. So you dial your gear down to 28 mW, and the amplifier punches itself up to match... and makes 10 db more noise doing it |
| 12:39.24 | J4k3 | yeah |
| 12:39.27 | J4k3 | agc = evil |
| 12:39.38 | vk4akp | Not necessarily . |
| 12:39.45 | J4k3 | well, in this case it is |
| 12:39.51 | vk4akp | If the AGC level is set correct so teh output is linear, then thats fine. |
| 12:40.04 | J4k3 | you can also forget getting high rate 11g through an agc amp. it rounds off the power peaks from what I can tell |
| 12:40.32 | J4k3 | ofdm is strange |
| 12:40.40 | vk4akp | We are not after `G' links anyhow. `B' is all one would expect over 10+K's. As specially with all teh timeing issues etc. |
| 12:40.51 | J4k3 | eh |
| 12:41.16 | J4k3 | I've got 25 mile (~40 km?) links that run 24mbit steady @ 2.4 |
| 12:41.28 | vk4akp | Different geat. |
| 12:41.28 | J4k3 | aproximately 12-13mbit actual tcp throughput |
| 12:41.45 | J4k3 | only real difference is atheros chipset instead of broadcom |
| 12:41.48 | vk4akp | Also most likely better line of sight and less path loss then what we have. |
| 12:41.49 | defswork | simbol76ss: doesnt g729 require licences ? |
| 12:42.01 | vk4akp | Our links are low tech. 1-2Mbit is fine for us. |
| 12:42.01 | simbol76ss | i have the license |
| 12:42.18 | J4k3 | eh, we do have towers |
| 12:42.19 | simbol76ss | but the processor for myserver is Xeon |
| 12:42.28 | J4k3 | our current repeater density is 1 tower per every 5 miles or so. |
| 12:42.29 | defswork | x686 then |
| 12:42.36 | vk4akp | Anyhow, best to let the ASterisk guys chat Asterisk here . There are people with questions. |
| 12:43.07 | mosty | simbol76ss, are you runing a 32bit or 64bit kernel? |
| 12:43.17 | simbol76ss | 32bit kernel |
| 12:43.24 | mosty | simbol76ss, pick the 32bit nocona module |
| 12:43.35 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:43.49 | mosty | oh wait, that's the intel 64bit one |
| 12:43.57 | mosty | just use i686 or pentium-m |
| 12:44.53 | simbol76ss | Ok now then try the nocona |
| 12:45.50 | mosty | nocona will only work if you're running in 64bit mode |
| 12:45.58 | simbol76ss | ok |
| 12:46.09 | simbol76ss | Ok now then try the i686 |
| 12:46.11 | simbol76ss | :) |
| 12:46.29 | simbol76ss | thanks!!! mosty :) |
| 12:56.14 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
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| 12:58.38 | *** part/#asterisk ming_zym (n=ming_zym@123.103.29.239) |
| 13:00.30 | festr_ | guys, i'm in the end. i've upgraded to the latest 1.4 svn and "dtmf from nowhere" is still here. if anyone have some time to debug this, i'm here :) (sorry for crosspoting from dev) |
| 13:01.29 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 13:05.12 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.239) |
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| 13:14.35 | patrick-- | Hey, ive got 1 BN4S0 HFC Card in my asterisk Machine. Port 1 is used as TE for Outgoing connections, while Ports 2-4 are used for ISDN phones within the company. I successful wrote a dialplan so that everyone is able to call out/intern etc. How can i insert callerID's so that the called person can see who is calling him (by name) |
| 13:17.29 | simbol76ss | Set=({callerid(name)}=Jonh) before Dial |
| 13:17.40 | simbol76ss | sorry |
| 13:17.54 | patrick-- | hm? |
| 13:17.57 | simbol76ss | exten => s,1,Set({callerid(name)}=Jonh) |
| 13:18.06 | patrick-- | i know that there is a callerid function, but im not sure how to insert it into my dialplan |
| 13:18.20 | patrick-- | simbol76ss: may i show you my dialplan? |
| 13:18.36 | simbol76ss | or into zapata.conf |
| 13:18.37 | simbol76ss | yes |
| 13:18.58 | simbol76ss | if you use digium card |
| 13:19.39 | patrick-- | beroNet |
| 13:19.43 | patrick-- | BN4S0 |
| 13:19.53 | simbol76ss | ok |
| 13:21.15 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 13:23.06 | *** join/#asterisk GBR_ (n=gbr@200.103.96.98) |
| 13:27.58 | hmmhesays | I'm getting no gtalk capable clients to talk to. when using chan_gtalk |
| 13:30.18 | nixguy | simbol76ss: i want to do the same but different :) |
| 13:30.32 | nixguy | i want outoing calls to the pstn to show the callerid of the one making the original call |
| 13:32.13 | simbol76ss | What hardware uses for outgoingcalls? |
| 13:32.33 | Wayhigh | is g-talk what rappers use? |
| 13:33.41 | hmmhesays | da dum ching |
| 13:34.02 | nebojsajsimic | little help with moh if any can |
| 13:34.06 | simbol76ss | nixguy I do not know if the carrier allows this |
| 13:34.49 | nixguy | im pretty sure they do |
| 13:35.05 | nixguy | now its just the its just one number |
| 13:35.10 | nixguy | no matter what phone i dial out with |
| 13:35.12 | nixguy | simbol76ss: |
| 13:35.28 | nixguy | im using a PRI card to connect to the pstn |
| 13:35.48 | simbol76ss | if you using zapata.conf |
| 13:35.54 | simbol76ss | useincomingcalleridonzaptransfer: If you set this option (Use Incoming Caller ID On Zap Transfer) to 'yes', then when you transfer a call to another phone, the original caller's Caller ID will get forwarded on too. Default: no. |
| 13:35.54 | nixguy | yes?! |
| 13:36.07 | simbol76ss | on transfer |
| 13:36.08 | nixguy | ah thnx ill have a look there |
| 13:37.14 | simbol76ss | or exten => s,1,Set(CALLERID(num)=123456) before Dial |
| 13:37.15 | the_5th_wheel | exten => s, 1, SetVar(MONITOR_FILENAME=/var/spool/asterisk/queuecalls/QSUPPORT-${UNIQUEID}) <-- i have this set in my configs, and it seems to run something called astmon. can anyone direct me to where i can configure this astmon? |
| 13:37.46 | simbol76ss | test this |
| 13:38.19 | *** join/#asterisk timeshell (n=timeshel@gw.lusi.on.ca) |
| 13:39.32 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
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| 13:39.51 | *** mode/#asterisk [+o denon] by ChanServ |
| 13:40.54 | mosty | the_5th_wheel, it doesn't run anything, it just sets a variable. and it should be Set, not SetVar in recent versions of asterisk |
| 13:41.02 | nixguy | dont forget =nooneknows |
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| 13:47.24 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
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| 13:51.02 | ZaVoid | hey guys |
| 13:51.10 | ZaVoid | what would cause this to happen in my console after an asterisk upgrade? |
| 13:51.11 | ZaVoid | <PROTECTED> |
| 13:51.12 | ZaVoid | <PROTECTED> |
| 13:51.15 | ZaVoid | i see that constantly now |
| 13:51.31 | *** join/#asterisk MaliutaWrk (i=nikolai@119.11.108.181) |
| 13:51.40 | mosty | something is logging in to the asterisk console, and then logging out |
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| 13:52.37 | *** join/#asterisk seanbright (n=elixer@65.207.74.18) |
| 13:53.22 | [TK]D-Fender | ZaVoid: * CLI, or AMI. You've got something out there polling your * server. |
| 13:53.51 | ZaVoid | yeah but it didn't do this in previous asterisk build i had running |
| 13:53.59 | ZaVoid | thats why i'm confused why its happening now |
| 13:54.06 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:54.06 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:54.07 | ZaVoid | or it did.. but was like once evetry 5 mins |
| 13:54.11 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:55.13 | tzafrir | patrick--, generally set callerid either in zapata.conf (callerid= ) or in the dialplan using the CALLERID function. |
| 13:55.32 | tzafrir | However be aware that your provider may not allow you to set the caller ID value |
| 13:55.50 | tzafrir | Or maybe it will allow you to set it only within a certian range |
| 13:56.09 | *** join/#asterisk xezz (n=phob@athedsl-238842.home.otenet.gr) |
| 13:57.52 | *** join/#asterisk Sajjad_Ali_Musht (n=Sajjad_A@octroi.enst-bretagne.fr) |
| 13:58.00 | patrick-- | tzafrir: i only want it for internal calls |
| 13:58.03 | patrick-- | the thing is |
| 13:58.09 | patrick-- | let me show you my dialplan |
| 13:58.12 | ZaVoid | so how do i get rid of that message fender? |
| 13:58.52 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 13:58.57 | dandre | hello, |
| 13:59.27 | dandre | How can I know, from CLI interface, in which groups a zap channel belongs? |
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| 14:07.42 | ik_5 | hello, I have a working Asterisk 1.2.24 (that was installed over a year ago) that recently started crashing on me, each time in different location, however it usually seems to happen when using the "free" command. i can't stop the work on that server, so I'm looking for ideas what to look for that will give me a clue to what going wrong |
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| 14:11.10 | *** join/#asterisk oej (n=olle@cust-IP-10.data.tre.se) |
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| 14:17.03 | tsubasafr | ik_5: you can start it in debug mode |
| 14:17.11 | tsubasafr | and play with you bugging command |
| 14:17.15 | tsubasafr | asterisk -c -vvv |
| 14:18.14 | xezz | hello, i have 3 isdn netmods with one "heading" number each but only one of them is known to the outside world. So when the first line rings the secretary's phone its ok, the second one rings ok, but there is no third line as the netmod supports only 2 voice lines..so the other 2 netmods (4 lines) are idling |
| 14:18.16 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 14:18.24 | xezz | is there a way to combine those 6 lines (3 netmods) ? |
| 14:19.47 | ik_5 | xezz, have you place all of them under the same group ? |
| 14:20.07 | xezz | i have them in different groups |
| 14:20.20 | xezz | like group=1 channel=1-2 |
| 14:20.31 | xezz | group=2 channel=4-5 |
| 14:20.44 | xezz | & group=3 channels=7-8 |
| 14:21.22 | xezz | should i put them in one group ? |
| 14:22.21 | ik_5 | xezz, as i understand that the second that channel is busy, then you can not go outside or get incomming calls with the theird ? |
| 14:22.44 | ik_5 | xezz, is it PRI or analog ?! |
| 14:23.29 | ik_5 | xezz, if it's digital you should have d channels as well for signaling imho |
| 14:24.14 | xezz | its BRI |
| 14:24.19 | xezz | i use 3 isdn bri cards |
| 14:24.46 | xezz | i want to be able to use the 6 lines that the 3 netmods provide me |
| 14:24.53 | xezz | i have mane incoming calls |
| 14:25.28 | xezz | and suppose i have the heading number wich is well known like: 2106500000 |
| 14:25.44 | xezz | but when 2 incoming calls reach asterisk at this number |
| 14:26.17 | xezz | the whole center blocks, it gets no incoming calls since the 2 lines of the first isdn netmod is full |
| 14:26.35 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:26.44 | xezz | but i have 2 more netmods so 4 lines... can i do something like a..merge or something ? |
| 14:27.10 | *** join/#asterisk stansmith (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 14:27.22 | tzafrir | xezz, trunkgroup? |
| 14:27.28 | stansmith | hello all |
| 14:27.36 | tzafrir | the [trunkgroups] section? |
| 14:27.48 | tzafrir | I have no idea if it makes sense for BRI |
| 14:28.06 | xezz | hmmm, never heard about trunkgroups |
| 14:28.16 | stansmith | I have set up the app_swift module, is there anyone else here that has worked with that module? I would like to check a couple of things |
| 14:29.00 | xezz | tzafrir should i put the three cards in a trunkgroup ? |
| 14:29.29 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 14:30.02 | agx | anyone has weird UDP SIP/RTP problem when using a Fortinet Fortifgate firewall between asterisk and a remote sip phone? |
| 14:30.50 | tzafrir | xezz, I think. I never tried using it for either pri or bri |
| 14:31.04 | Wayhigh | agx: I may be able to see what my firewall guys say about that if you like |
| 14:32.02 | Wayhigh | we use cisco gear for voip but I know that * has been at least tested before.. |
| 14:32.08 | xezz | ok tzafrir, but as an idea it suppose to work ? i mean, asterisk will be able to use the 6 lines for incoming calls ? |
| 14:32.20 | Wayhigh | sup tzafrir.. |
| 14:32.49 | tzafrir | xezz, in theory - yes, but as long as your ISP has the same config |
| 14:33.46 | xezz | ye, it sounds pretty logic |
| 14:34.04 | xezz | thanks i'll try it and give you feedback |
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| 14:35.11 | stansmith | Does anyone use the Cepstral TTS engine? |
| 14:35.15 | stansmith | here* |
| 14:39.39 | *** join/#asterisk angryuser (n=nononon@df01t2-213-44-84-3.d4.club-internet.fr) |
| 14:41.26 | *** join/#asterisk mkl1525 (n=qwertz@89.246.171.88) |
| 14:42.12 | mkl1525 | Hi, is there a way to set some vars in * that are in all extensions available (something like SERVER_NAME)? |
| 14:42.36 | stansmith | mkl1525: you mean a global var? |
| 14:42.57 | mkl1525 | stansmith yes |
| 14:43.14 | stansmith | i believe you can set those variables under the [global] directive in extensions.conf |
| 14:43.30 | stansmith | in mine, for example, i have "TIMEOUT = 10" |
| 14:43.33 | mkl1525 | stansmith thanks will try it |
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| 14:56.08 | Wayhigh | yep |
| 14:57.11 | *** join/#asterisk Uatec (n=uatec@adsl.ntsols.com) |
| 14:57.16 | Uatec | hiya |
| 14:57.37 | stansmith | I have set up the app_swift module, is there anyone else here that has worked with that module? I would like to check a couple of things |
| 14:58.28 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 14:58.46 | tzanger | wow there's a name I haven't seen in like f'rever |
| 14:58.56 | tzanger | good morning, Katty |
| 14:59.15 | Katty | hai |
| 15:00.05 | [TK]D-Fender | Katty: Mew. |
| 15:00.11 | Katty | g'morning mister fender (= |
| 15:00.12 | Katty | k |
| 15:00.34 | Katty | tzanger: i've got a server with a pri going to it! i'm moving up in the world! |
| 15:01.40 | stansmith | Does anyone use the Cepstral TTS engine? |
| 15:01.45 | stansmith | here* |
| 15:02.03 | Katty | stansmith: i haven't even heard of that before :/ |
| 15:02.19 | Katty | stansmith: but that's okay. my knowledge does not runnith over. |
| 15:02.27 | Katty | yet. |
| 15:02.35 | stansmith | wahhh |
| 15:02.43 | hmm-home | Holy crap its Katty |
| 15:03.15 | stansmith | does it usually take Digium a while to send you your username and password confirmation for the forums? |
| 15:03.21 | hmm-home | no |
| 15:03.21 | Katty | hmm-home: holy crap don't give it away! |
| 15:03.31 | hmm-home | lol |
| 15:04.33 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:04.37 | Katty | hello james. |
| 15:04.47 | jameswf-home | greetings |
| 15:05.04 | *** join/#asterisk quigon (n=matias@32.59.64.130) |
| 15:05.09 | Katty | hmm-home: i'm stuck on nat king cole this morning. it's keeping me sane against the onslaught of Stupid. |
| 15:05.19 | hmm-home | nice |
| 15:05.29 | jameswf | http://www.mhspot.com/mhspot/sippyskype.htm << looks like skype on asterisk may be possible without hacking the crap out of your asterisk box... |
| 15:06.05 | hmm-home | java...ew |
| 15:06.29 | *** join/#asterisk wmaulik (n=wmaulik@158.59.192.218) |
| 15:06.49 | jameswf | looks better than chan_skpe and its free, dont use skype but sadly quite a few people do |
| 15:07.21 | hmm-home | I've used chan_skype with pretty good success |
| 15:07.46 | jameswf | Since I dont use skyoe I am just assuming it works :) looks better in theory |
| 15:07.56 | hmm-home | psgw used to be good |
| 15:08.02 | *** join/#asterisk flint (n=flint@68-114-60-151.dhcp.davl.vt.charter.com) |
| 15:08.12 | wmaulik | hey flint |
| 15:08.13 | flint | hi tesh! |
| 15:08.42 | flint | lotta lurkers here... good place to ask questions... |
| 15:08.56 | jameswf | ~ask |
| 15:08.56 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 15:09.13 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 15:09.42 | flint | jameswf, indeed. --- tesh, back to the bat cave.... |
| 15:10.03 | flint | jameswf, we will come back with excellent stupid questions... forgive the intrusion. |
| 15:10.08 | flint | ~ask |
| 15:10.09 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 15:10.19 | tzanger | Katty: awesome :-) |
| 15:10.25 | tzanger | got echo on the PRI yet? |
| 15:11.08 | *** join/#asterisk wmaulik (n=wmaulik@158.59.192.218) |
| 15:11.39 | *** join/#asterisk YHSAsterisk (n=asterisk@158.59.192.218) |
| 15:12.30 | hmm-home | uncle toms cabin by warrant is just a cool song |
| 15:13.26 | *** join/#asterisk ahilb (n=noone@104.93.187.81.in-addr.arpa) |
| 15:15.18 | Katty | tzanger: i haven't even tested it yet >.< |
| 15:15.21 | [TK]D-Fender | hmm-home: Wow, Skid Row, Warrant.... you've gone hair-metal! |
| 15:15.26 | Katty | tzanger: i just got wanpipe configured last night.. |
| 15:15.32 | tzanger | ahh |
| 15:15.34 | hmm-home | well i've been working on the more technical stuff lately |
| 15:15.37 | Katty | [TK]D-Fender: you have no idea. |
| 15:15.44 | [TK]D-Fender | hmm-home: Which Warrant piece(s)? |
| 15:15.58 | [TK]D-Fender | Katty: 80's glam-rock was more mmy schtick :) |
| 15:16.14 | hmm-home | my head is a weapon |
| 15:16.41 | [TK]D-Fender | hmm-home: Er Skid Row that is.. |
| 15:16.42 | robl^ | I was more 80s Bubble-Gum Pop |
| 15:16.47 | [TK]D-Fender | robl^: EW. |
| 15:17.20 | *** join/#asterisk tobias (n=tobias@cpe-069-134-226-227.nc.res.rr.com) |
| 15:17.31 | coppice | [TK]D-Fender: did you attend the Slade School of Fine Art? :-) |
| 15:17.33 | Katty | i would not be just a nuffin' my head of full of stuffin' my heart all full of pain.... |
| 15:17.55 | hmm-home | 80's rock and 90's punk has been on my mp3 player lately... oh along with my chemical romance |
| 15:18.06 | Katty | in the 80s, i was just sitting around dumping spaghetti on my head and chasing the cats. |
| 15:18.17 | [TK]D-Fender | coppice: If there's a joke buried in there then you've gotten one completely past me :) |
| 15:18.25 | tzanger | Katty: so... what has changed in the last 18 years? |
| 15:18.26 | hmm-home | in the 80's I was wooing the ladies |
| 15:18.50 | Katty | tzanger: i graduated high school. |
| 15:18.52 | tzanger | in the 80s I was playing with transformers and lego |
| 15:18.57 | Katty | tzanger: and then i graduated university. |
| 15:19.07 | Katty | tzanger: and then i got various letters attached to the end of my name. |
| 15:19.16 | tzanger | Katty: yes, but have you stopped dumping spaghetti on your head and chasing cats? |
| 15:19.25 | Katty | the spaghetti part. ^_^ |
| 15:19.29 | coppice | [TK]D-Fender: The Slade School of Fine Art is a part of the University of London |
| 15:19.49 | [TK]D-Fender | coppice: Yes, Google told me that... anything else implied in what you asked? |
| 15:19.59 | Katty | tzanger: the 'cats' are now 'ferrets' and there is still chasing. |
| 15:20.14 | [TK]D-Fender | Katty: Cat for a rat and still in the race! |
| 15:20.21 | Katty | hehehehe |
| 15:22.11 | coppice | [TK]D-Fender: maybe Slade was too English a Glam-Rock group to have made it your way |
| 15:22.52 | Katty | [TK]D-Fender: you know of a nice lil php thingy for asterisk/sql stuffs |
| 15:23.06 | Katty | [TK]D-Fender: or is it called a php book? |
| 15:23.13 | [TK]D-Fender | coppice: Guess they didn't quite cross the atlantic... |
| 15:23.36 | [TK]D-Fender | Katty: You looking for a book, or a library for some common * stuff? |
| 15:23.48 | Katty | [TK]D-Fender: no. |
| 15:23.54 | Katty | [TK]D-Fender: just a php query thingy for sql call logs. |
| 15:23.55 | [TK]D-Fender | Katty: Neither? |
| 15:24.06 | Katty | [TK]D-Fender: but everyone seems to be telling me Go Read. |
| 15:24.18 | [TK]D-Fender | Katty: well PHP's SQL functiosn are very well documented. Nothing to see really. |
| 15:24.33 | Katty | but i don't wanna write my own )= |
| 15:24.40 | [TK]D-Fender | Katty: 1 line to connect, 1 to issue the query, 1 to pull records. |
| 15:24.50 | [TK]D-Fender | Katty: For everything else... ther's consulting ;) |
| 15:24.59 | Katty | [TK]D-Fender: yes but you know what happens when a company discovers you can Customize Things |
| 15:25.09 | Katty | [TK]D-Fender: they change their mind a BILLION times. that makes me all sad inside. |
| 15:25.16 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 15:25.32 | Katty | [TK]D-Fender: i think i might steal the one from freepbx that can generate neat little graphs. |
| 15:25.37 | [TK]D-Fender | Katty: Indecisiveness = Katty's long-term job security :) |
| 15:25.46 | hmm-home | Katty: thats asterisk-stat-v2 |
| 15:25.57 | javar | hi, somebody knows, which version of Asterisk was able to handle NSCR2 signalling? |
| 15:26.05 | [TK]D-Fender | Katty: CRD stuff is already out there, and there are other free * Queue ones, though not so great I'm sure |
| 15:26.06 | Wayhigh | let me know when you want to meet there |
| 15:26.23 | Katty | [TK]D-Fender: i should probably just stop whining and write some php pages. |
| 15:26.33 | hmm-home | you looking for a cdr interface? |
| 15:26.36 | Katty | [TK]D-Fender: and then give people choices, so they can't customize too much. |
| 15:26.39 | [TK]D-Fender | Katty: Typically the solution should be scaled to the need |
| 15:26.46 | Katty | hmm-home: yeah, a pretty wrap around. |
| 15:26.54 | Katty | hmm-home: we're still digging through csv. |
| 15:27.05 | hmm-home | yeah just grab areski's asterisk stat |
| 15:27.09 | Katty | hmm-home: you would not believe how much of a pain it is to teach someone how to use excel or access. *shudder* |
| 15:27.11 | hmm-home | quick, easy, pretty |
| 15:27.25 | Katty | hmm-home: and that's the one freepbx uses? |
| 15:27.28 | hmm-home | yep |
| 15:27.33 | hmm-home | www.areski.net I believe |
| 15:27.36 | Katty | cheers. |
| 15:27.56 | hmm-home | works with postgres and mysql |
| 15:28.21 | hmm-home | [TK]D-Fender: you can add late 90's punk metal to the list too |
| 15:28.37 | [TK]D-Fender | hmm-home: Who counts in that class exactly? |
| 15:28.42 | Katty | oh wow, that's it. for sure. i recognize the screenshots. |
| 15:28.44 | hmm-home | sum 41 |
| 15:28.49 | Katty | hmm-home: PURE GENIOUS! |
| 15:28.51 | [TK]D-Fender | hmm-home: Having trouble identifying punk-metal, and 80's. |
| 15:29.12 | [TK]D-Fender | hmm-home: err.. 90's. Sum 41 = 2000+ |
| 15:29.13 | Katty | you're going to help me with postgres later right? |
| 15:29.16 | Katty | :P |
| 15:29.23 | Katty | cause i've never made * dump to database. |
| 15:29.23 | hmm-home | sum 41 was around in the 90's |
| 15:29.29 | hmm-home | Sure |
| 15:29.33 | Katty | neat. |
| 15:29.37 | hmm-home | or mysql either works fine |
| 15:30.03 | stansmith | postgres > mysql |
| 15:30.08 | hmm-home | their metal influences really come out on their live album |
| 15:30.35 | Katty | at this point i'm just going: yay, database! |
| 15:30.57 | hmm-home | stansmith: for storing cdr's on a medium sized pbx it doesn't matter |
| 15:31.16 | wmaulik | hi i have a question, we are currently working with ubuntu clients and we would like to know if there is a windows client someone recommends, possibly one that is free |
| 15:31.24 | hmm-home | xlite |
| 15:32.07 | hmm-home | I use "pain for pleasure" as a pick hand warm up exercise |
| 15:32.14 | stansmith | would anyone here be able to chat about the "app_swift" module? I have it working, but just want to compare some things |
| 15:32.37 | flint | hmm-home, ¡ɥɔnɯ ʎɹÇÊŒ noÊŽ ÊžuÉɥʇ |
| 15:32.50 | hmm-home | hmm? |
| 15:32.56 | stansmith | howd you do that |
| 15:33.04 | Qwell | stansmith: magic |
| 15:33.05 | codefreeze | flint: yeah, how you do that? |
| 15:33.10 | stansmith | ou!!! magic! |
| 15:33.39 | codefreeze | Oh masters, reveal thy secrets! |
| 15:34.27 | flint | hmm-home, email me flint@flint.com |
| 15:34.37 | stansmith | you own the flint domain name? |
| 15:34.49 | hmm-home | for what purpose? |
| 15:34.56 | Katty | for what porpoise. |
| 15:35.35 | codefreeze | flipper? |
| 15:35.45 | *** join/#asterisk granviller (n=granvill@72x18x39x109.static.newroadstelecom.net) |
| 15:35.49 | granviller | hello |
| 15:36.01 | granviller | I'm having problems dialing in on a sip trunk |
| 15:36.25 | granviller | keep getting 484 address is not complete. Did i get the sip trunk setup wrong? |
| 15:36.30 | flint | stansmith hmm-home , because I am flint. |
| 15:36.53 | hmm-home | um |
| 15:36.55 | stansmith | ..word... |
| 15:36.57 | hmm-home | going to take more than that |
| 15:37.11 | hmm-home | find me a copy of sake bombs and happy endings |
| 15:37.50 | Katty | file: wherearechu |
| 15:38.20 | [TK]D-Fender | Katty: Gesundheit :) |
| 15:38.33 | Katty | [TK]D-Fender: you made me giggle. |
| 15:38.35 | Katty | [TK]D-Fender: how dare you. |
| 15:38.40 | [TK]D-Fender | Katty: :O |
| 15:38.45 | Katty | [TK]D-Fender: now the other people in this office are looking at me. |
| 15:39.31 | *** join/#asterisk Chumbaa (n=chatzill@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 15:39.39 | [TK]D-Fender | Katty: Bask in it! |
| 15:39.45 | granviller | anyone able to help me with 484 address not complete? |
| 15:40.17 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 15:40.18 | *** join/#asterisk jbigbee (n=jbigbee@216.207.245.1) |
| 15:40.28 | *** join/#asterisk simbol76ss (n=simbol@ip-212-18.sn1.eutelia.it) |
| 15:41.46 | [TK]D-Fender | granviller: pastebin the output of your failed call attemp with SIP debug enabled at verbose 10. |
| 15:41.48 | [TK]D-Fender | ~pb |
| 15:41.49 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:41.49 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 15:42.01 | [TK]D-Fender | granviller: And include your dialplan. |
| 15:42.14 | stansmith | and a pic |
| 15:42.28 | Katty | wow. |
| 15:42.34 | Katty | what's the biggest sangoma card they make? |
| 15:42.36 | Katty | in terms of pris. |
| 15:42.40 | granviller | pic? |
| 15:42.43 | [TK]D-Fender | Katty: A108d IIRC |
| 15:42.49 | Katty | [TK]D-Fender: how many pris does that handle? |
| 15:42.51 | Katty | [TK]D-Fender: 8? |
| 15:42.54 | [TK]D-Fender | Katty: 8 |
| 15:43.10 | Katty | how many of those cards can you safely put into a single beast of a server. |
| 15:43.20 | Katty | or is more than one per server just... sucidal. |
| 15:43.31 | [TK]D-Fender | Katty: Hrm, typically I'd say 2 depending on what that server will have to do. |
| 15:43.36 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 15:43.42 | Katty | kk |
| 15:44.03 | Katty | i'll call them later and see what they recommend in terms of specs if only half of it was used at any single time. |
| 15:44.07 | [TK]D-Fender | Katty: Without transcoding, etc PRI and G.711 conversion alone isn't so bad, and they are nice as far as PCI devices are concerned |
| 15:49.51 | Katty | [TK]D-Fender: in all honesty, i'm more worried about 341 phones and getting them all on the same ip range. |
| 15:50.06 | Katty | [TK]D-Fender: i've never set up a Big network. |
| 15:50.13 | [TK]D-Fender | Katty: /8 :) |
| 15:50.26 | Katty | [TK]D-Fender: that doesn't parse. |
| 15:50.43 | [TK]D-Fender | Katty: 10.0.0.0/8 |
| 15:50.50 | [TK]D-Fender | Katty: just ... bigger :) |
| 15:51.03 | Katty | i'm just going to smile and nod. |
| 15:51.16 | Katty | since i don't have any experience with setting up a big 10.0.0.etc network, i'm not sure how it works. |
| 15:51.51 | [TK]D-Fender | Katty: you just set your IP rang and DHCP and you're done. Nothing to it. |
| 15:52.04 | Katty | that makes my stress level go down. |
| 15:52.18 | [TK]D-Fender | Katty: and 5 seconds work different than your typical 192.168.XXX.XXX network. |
| 15:52.56 | Chumbaa | Anyone have experience in getting Aastra 53i connected throuhg a SonicWall Firewall? I connect just fine, and when the phones on the local lan behind the firewall, they work fine. But outside the firewall, they stop responding to inbound calls from other extensions. Sofphone, Polycom and other phones work fine. |
| 15:53.42 | Chumbaa | usually takes 2-4 calls. resetting the phone works. Coudl it be the firewall resetting the IP numbering that throws off the AAstra 53i? |
| 15:54.34 | Chumbaa | (just making sure I'm not losing my mind, because its th eonly Phone I have seen that does this with our system) |
| 15:54.49 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 15:55.04 | *** join/#asterisk pithen (n=pithen@mail.graphlogic.com) |
| 15:55.20 | *** join/#asterisk simbol76ss (n=simbol@ip-212-18.sn1.eutelia.it) |
| 15:55.50 | pithen | Hey all, is it possible to run an application from the console? I'm trying out a few things and can't figure out how to simply output the value of a variable without creating an extension in the dialplan that prints it |
| 15:56.34 | stansmith | pithen: is the variable in the dialplan or the application you are running from the console? |
| 15:56.36 | [TK]D-Fender | pithen: like what kind of variable? Output on the * server, but not in * CLI? |
| 15:57.03 | *** join/#asterisk joshaidan (n=joshaida@S0106001c1023e838.tb.shawcable.net) |
| 15:57.08 | pithen | i.e. can i output the value of ${ENUMLOOKUP(somenum,1,e164.org)} |
| 15:57.29 | [TK]D-Fender | pithen: Nope, that'll require dialplan. |
| 15:57.52 | pithen | meh :( |
| 15:59.10 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:59.10 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:59.50 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 16:00.25 | *** join/#asterisk CmdrChalupa (n=vincas@209.9.232.204) |
| 16:00.53 | CmdrChalupa | I just got the following error from asterisk: terminate called after throwing an instance of 'Wobbly' |
| 16:00.57 | CmdrChalupa | starting it |
| 16:01.03 | CmdrChalupa | where does this "Wobbly" come in |
| 16:01.26 | stansmith | CmdrChalupa: are you running some kind of custom code? |
| 16:02.02 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 16:02.46 | CmdrChalupa | stansmith: It's the debian release |
| 16:03.01 | ManxPower | CmdrChalupa: I've never heard of that error with Asterisk. |
| 16:03.16 | *** join/#asterisk monkeytype (n=Bill@216.207.245.1) |
| 16:03.18 | stansmith | did you compile asterisk or apt-get the deb? |
| 16:03.22 | ManxPower | I'll bet the packager did something strange. Try building Asterisk from scratch. |
| 16:03.28 | stansmith | ^ |
| 16:05.54 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.232.128) |
| 16:09.30 | CmdrChalupa | stansmith: I've submitted a bug report to the debian packager. I just figured I'd check to see if this was an asterisk problem or a debian one. Thanks |
| 16:09.40 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:09.43 | stansmith | np |
| 16:10.18 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 16:11.07 | *** join/#asterisk Chumbaa (n=chatzill@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 16:11.23 | hmmhesays | I'm still getting no compatible clients using chan gtalk |
| 16:11.35 | *** join/#asterisk XnOSX (i=d491af1a@gateway/web/ajax/mibbit.com/x-3204f24208802292) |
| 16:12.11 | *** join/#asterisk angryuser (i=nononon@df01t2-212-194-99-7.d4.club-internet.fr) |
| 16:12.31 | ManxPower | hmmhesays: no compatible clients or codecs? |
| 16:12.39 | hmmhesays | clients |
| 16:13.35 | hmmhesays | oops... no gtalk capable clients to talk to. |
| 16:13.42 | hmmhesays | capable, not compatible |
| 16:13.56 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:14.01 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 16:14.03 | dandre | Hello, |
| 16:14.19 | dandre | I have echo cancel off for my card: |
| 16:14.21 | dandre | Echo Cancellation: 128 taps, currently OFF |
| 16:14.30 | dandre | How can I set it on? |
| 16:14.34 | ManxPower | dandre: is there an active call? |
| 16:14.36 | hmmhesays | google isn't revealing much to me |
| 16:15.04 | *** part/#asterisk sjirsten (i=sjirsten@137.120.156.79) |
| 16:15.05 | ManxPower | EC will only be on for active channels |
| 16:15.38 | dandre | ok thanks |
| 16:15.51 | dandre | I find the trainig quite long |
| 16:16.04 | dandre | is there any way to reduce it? |
| 16:16.13 | *** join/#asterisk Docfxit (n=none@cpe-72-129-118-135.socal.res.rr.com) |
| 16:16.53 | ManxPower | echotraining=lowernumberthanyouhavenow |
| 16:17.29 | dandre | ok I'll try |
| 16:17.32 | dandre | bye |
| 16:17.35 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:22.29 | hmm-home | and chan gtalk gives absolutely no debugging output |
| 16:24.00 | *** join/#asterisk mchou (n=mchou@c-71-198-127-234.hsd1.ca.comcast.net) |
| 16:24.49 | *** join/#asterisk thudd (n=wolthuis@mimezine.com) |
| 16:24.55 | mchou | anyone here run asterisk on openwrt/wrt54g? |
| 16:26.46 | hmmhesays | used to |
| 16:27.28 | mchou | hmmhesays: is there enough juice in HW to support 2 simultaneous calls? |
| 16:27.43 | hmmhesays | yep |
| 16:27.49 | hmmhesays | you can even transcode them if you really want |
| 16:28.08 | mchou | wow. |
| 16:28.14 | *** join/#asterisk mitio (n=wireman@78.83.96.4) |
| 16:28.42 | mchou | hmmhesays: can it still be used as a router at the same time? :) |
| 16:28.52 | hmmhesays | yeah, I never had any problems |
| 16:29.34 | mchou | so for home use would you have any reservations using asterisk on openwrt? |
| 16:29.41 | hmmhesays | nope |
| 16:29.56 | mchou | ok, that good enough for me |
| 16:30.08 | hmmhesays | i had a wrtsl54gs running for a long time |
| 16:30.43 | mchou | so why did you switch? |
| 16:30.54 | hmmhesays | i wanted to play with it |
| 16:31.10 | af_ | asterisk can work with xlite and its instant messaging? |
| 16:31.16 | hmmhesays | af_ nope |
| 16:31.27 | hmmhesays | you need to have ser or openser in front of it |
| 16:31.30 | af_ | also the 1.4 branch? |
| 16:31.45 | af_ | it's something not supported in the sip protocol? |
| 16:31.46 | hmmhesays | asterisks broken sip stack doesn't lend itself to messaging |
| 16:31.50 | af_ | I see |
| 16:31.54 | russellb | errr |
| 16:31.59 | af_ | thanks hmm-home |
| 16:31.59 | *** join/#asterisk watchy (n=watchy@h200.176.255.206.cable.cmdn.cablelynx.com) |
| 16:32.01 | af_ | thanks hmmhesays |
| 16:32.23 | russellb | SIP messaging is supported within a call |
| 16:32.31 | af_ | hmmhesays, but also presence is supported with ser in front of it? |
| 16:32.31 | watchy | anyone here deal with polycoms + showing busy in * 1.4/ |
| 16:32.44 | hmmhesays | af_ if you do it right |
| 16:32.48 | russellb | it's not a broken SIP stack ... |
| 16:32.53 | [TK]D-Fender | watchy: Presence works fine |
| 16:32.56 | hmmhesays | its just missing some things |
| 16:32.59 | af_ | I should configure a common auth database for sip user I guess (ser + *)? |
| 16:33.07 | watchy | i upgraded to 1.4 and presence stoped workin tk |
| 16:33.23 | watchy | do i need to add limitonpeers=yes and call-limit=something to each device |
| 16:33.40 | [TK]D-Fender | watchy: [yourphone] "type=peer" "call-limit=99" <- do these. |
| 16:34.01 | watchy | oh |
| 16:34.29 | hmmhesays | chan_gtalk is really p1ss1ng me off right now |
| 16:34.52 | ManxPower | I always thought call-limit being required for presence was totally stupid. |
| 16:35.16 | hmmhesays | i've found call-limit to be completely useless |
| 16:35.49 | ManxPower | hmmhesays: well that's a given |
| 16:36.10 | hmmhesays | my jabber user show connected |
| 16:36.30 | hmmhesays | anyone actually have chan_gtalk working properly? |
| 16:39.13 | *** join/#asterisk lonebobwhite (n=rleblanc@74.231.171.198) |
| 16:39.26 | watchy | any phone rebooting required tk? |
| 16:41.03 | *** join/#asterisk granviller (n=granvill@72x18x39x109.static.newroadstelecom.net) |
| 16:41.14 | *** join/#asterisk SomethingISOdd (n=TestMast@S010600a0d1757bfb.cg.shawcable.net) |
| 16:41.22 | SomethingISOdd | hello all quick question is this correct $agi->set_veriable("clientrate",$row2['clientrate']); |
| 16:41.29 | granviller | http://pastebin.ca/911237 included the entire extenstion.conf and a portion of the sip.conf |
| 16:41.34 | seanbright | no |
| 16:41.41 | seanbright | you spelled variable wrong |
| 16:41.55 | SomethingISOdd | haha that would explain it thanks seanbright |
| 16:42.01 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:42.11 | granviller | I'm having a problem dialing in on a sip trunk. I can dial out on theat sip trunk |
| 16:43.25 | granviller | the extensions.conf file was created by hand byt a guy who is no longer here. We don't have a GUI on the machine that is running asterisk |
| 16:44.10 | stansmith | granviller: nano /etc/asterisk/extenstions.conf |
| 16:44.16 | granviller | ok |
| 16:44.32 | stansmith | i mispelled it on accident |
| 16:44.38 | stansmith | *extensions |
| 16:44.44 | *** join/#asterisk phocus (n=phocus@www.healthtech.net) |
| 16:44.47 | granviller | I figured, I do it too. |
| 16:45.42 | granviller | anyways, what do you think needs to be changed? |
| 16:45.50 | phocus | hey guys, i am trying to write a plugin for media center, i know you can telnet to asterisk on port 980 and log in and it will show you the log, but i cant fiigure out the syntax to login, or the account to log in with. does anybody have some documentation i can read to figure this stuff out? |
| 16:46.00 | ManxPower | ~trunk |
| 16:46.02 | jbot | rumour has it, trunk is is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 16:46.32 | granviller | OK |
| 16:46.39 | stansmith | ive never used asterisk for outgoing calls, ive strictly been using it for IVR stuff, which is incoming...but i think you need to add something to the extensions.conf, im not sure |
| 16:47.30 | hmmhesays | do I have to specify somewhere that my client i'm calling is jingle capable? |
| 16:48.06 | phocus | add that extension to your "ring list" |
| 16:50.18 | [TK]D-Fender | granviller: none of your extension patterns linked in [keysystem] are actually patterns because you did not put the "_" in front of them. |
| 16:51.20 | [TK]D-Fender | granviller: Hence nothing to match |
| 16:51.20 | granviller | the patterns are in context [kirkham |
| 16:52.19 | [TK]D-Fender | granviller: Nothing in there matches either |
| 16:53.02 | [TK]D-Fender | granviller: And you have opened your system to risk by merging a context with outbound capabilities to your INBOUND context. |
| 16:53.03 | hmmhesays | how does chan_gtalk decide if an endpoint is jingle capable |
| 16:53.06 | granviller | ok, what needs to match then? I guess part of it is that we're adding the sip provider while we have an IAX and Zap connections that work currelty |
| 16:53.11 | [TK]D-Fender | granviller: That is a big no-no |
| 16:53.31 | [TK]D-Fender | granviller: Fix your patterns by adding the "_" in front like you're supposed to. |
| 16:56.02 | granviller | which lines would be refering to? the ones that start with 479? |
| 16:56.15 | *** join/#asterisk didz_ (n=hoje@201.19.232.128) |
| 16:56.34 | granviller | Ok I got it to dial in and answer. but I got one way audio. |
| 16:58.48 | [TK]D-Fender | granviller: NEXT!!@!@ (c) BKW |
| 16:58.58 | [TK]D-Fender | granviller: now read up : |
| 16:58.59 | [TK]D-Fender | ~sipnat |
| 16:59.00 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 16:59.00 | stansmith | Anyone here use Cepstral text-to-speech engine? |
| 16:59.03 | [TK]D-Fender | ^^ |
| 16:59.20 | stansmith | ~cepstral |
| 16:59.34 | *** join/#asterisk clickonce (n=get@unaffiliated/clickonce) |
| 16:59.44 | clickonce | Hello |
| 16:59.50 | granviller | I got the ports forwarded 10001 o 20000 and rtp.conf is set with the same info |
| 16:59.55 | stansmith | clickonce: do you use the Cepstral TTS engine? |
| 17:00.06 | granviller | damn nat=no, didn't see that |
| 17:00.25 | granviller | but the sip connection is set to nat=yes |
| 17:00.39 | clickonce | Can you recommend any free SIP/VoIP software which I can use with Asterisk for the P990i phone? I'm considering buying OptiMobile UniPhone but I'd like to test the system first. (So, quailty isn't of the essence at first) |
| 17:01.07 | Qwell | clickonce: there are some very good free softphones. I wonder if the bot has recommendations...let's see |
| 17:01.09 | Qwell | ~softphones |
| 17:01.15 | Qwell | ~softphone |
| 17:01.15 | jbot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam |
| 17:01.40 | Qwell | never used bria, but the others are good, and the first three are free |
| 17:02.18 | clickonce | Are they for Windows PCs or WLAN equipped phones? |
| 17:02.31 | phocus | does anyone know what account to use to telnet to port 5038 to watch the caller id info? |
| 17:02.40 | phocus | and the syntax to login |
| 17:02.46 | ManxPower | philippel: manager.conf |
| 17:02.56 | Qwell | zoiper works on linux, windows, osx, xlite works on linux and windows (maybe osx too..), and twinkle works on at least linux |
| 17:03.03 | ManxPower | you REALLY want to read up on AMI the Asterisk Manager Interface |
| 17:03.10 | phocus | o fpimd tjos. action: Login |
| 17:03.10 | phocus | username: {user} |
| 17:03.10 | phocus | secret: {pass} |
| 17:03.17 | phocus | i found that, but cant get it to work |
| 17:03.23 | phocus | it says bad password/username |
| 17:03.32 | clickonce | Qwell: SEMC P990i runs Symbian / UIQ3 :) |
| 17:03.43 | Qwell | oh, you want something on your phone... |
| 17:03.59 | Qwell | yeah, no idea... I assume there is one though |
| 17:04.04 | clickonce | Yes, the P990i :) |
| 17:04.58 | *** join/#asterisk |omni| (n=rob@c-67-185-144-206.hsd1.wa.comcast.net) |
| 17:05.38 | granviller | Fender: sorry for frusting you. I know this is a kludge, i didn't create it, I'm just trying to get a new sip provider we have to work with it. I found the dial plans they were uin the [fireball] context, where they shouldn't have been, I moved them to the [kirkham] context. I also added the did number for the sip channel to the [day] and [night] contexts |
| 17:07.19 | clickonce | I found something called FRing that has SIP support but it requires some shitty FRING account which is utterly useless. |
| 17:09.10 | stansmith | they are trying to coin the phrase "fring me later!" |
| 17:09.19 | mchou | in theory gizmo works on some cell phones..... |
| 17:09.35 | clickonce | stansmith: What a piece of crap.. |
| 17:09.40 | mchou | but I've never tried it myself |
| 17:09.41 | clickonce | I'll just get the UniPhone solution. |
| 17:11.13 | clickonce | Fring test call worked fine though. :) |
| 17:11.19 | stansmith | lol |
| 17:11.34 | stansmith | i just made that catch phrase, if it makes you feel better |
| 17:12.53 | clickonce | stansmith: Damn. |
| 17:13.04 | clickonce | stansmith: You just made my reason to hate them void. |
| 17:13.12 | stansmith | haha sorry |
| 17:13.46 | Qwell | clickonce: does sjphone work on symbian? I know it runs on a few phone platforms.. |
| 17:14.09 | clickonce | Qwell: Let me check. :) |
| 17:20.29 | clickonce | Hmm, I think this crapring thing might work for testing purposes. But it's really nothing I'd recommend to poor children in afghanistan. |
| 17:20.54 | webar7 | I have a sipura SPA200 with an old phone hooked into it ... it worked great under 1.2 but now I can get dialtone but can't make a call ... is this likely to be firewall realted? |
| 17:21.06 | webar7 | sip.conf is super simple |
| 17:21.14 | clickonce | SPA200? Sounds extremly similar to Linksys SPA series ;) |
| 17:21.30 | Qwell | linksys bought sipura |
| 17:21.36 | Qwell | so, it is :) |
| 17:21.37 | clickonce | ah |
| 17:22.02 | clickonce | I see :) I've been looking at 962 along with a 932 :) Quite a bit pricey though.. |
| 17:22.08 | hmm-home | well my gtalk calls only work to gtalk clients on the same localnet |
| 17:22.18 | Qwell | clickonce: polycom > linksys |
| 17:22.29 | hmm-home | depends on the model |
| 17:22.45 | hmm-home | i'll take an spa-942 over an ip-320 |
| 17:23.12 | clickonce | Qwell: Checking. |
| 17:25.33 | clickonce | I wonder if the SPA-962 has video support... |
| 17:26.01 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 17:27.58 | Qwell | clickonce: if it's > $500, then yes, otherwise...probably not |
| 17:29.21 | [TK]D-Fender | bbiab |
| 17:29.41 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ppp-70-248-182-215.dsl.hstntx.swbell.net) |
| 17:30.33 | webar7 | SPA2000 actually |
| 17:30.56 | *** part/#asterisk FlatFoot (n=bigflatf@80.88.192.83) |
| 17:31.05 | SwK | anyone have tcap/cnam documentation? |
| 17:31.13 | webar7 | I weas thinking some SIP default changed 1.2 ->1.4 so now it doesn't work "out of the box" |
| 17:31.20 | SwK | holt shit twisted is on eye are sea |
| 17:31.26 | twisted | i'm always on irc |
| 17:31.28 | twisted | just usually idle |
| 17:31.36 | SwK | twisted, sup dude |
| 17:31.50 | twisted | meh, putting out fires |
| 17:31.55 | twisted | you know how it goes |
| 17:32.13 | SwK | yeah |
| 17:32.16 | webar7 | yow ... |
| 17:32.27 | webar7 | just saw a console message |
| 17:32.32 | stansmith | would anyone here be able to chat about the "app_swift" module? I have it working, but just want to compare some things |
| 17:32.36 | webar7 | 0 active SIP channels |
| 17:32.37 | webar7 | <PROTECTED> |
| 17:33.00 | twisted | uhm, webar7, congrats? |
| 17:33.05 | webar7 | that looks hopeful |
| 17:33.17 | webar7 | but still no dialout |
| 17:33.22 | webar7 | only dial tone |
| 17:33.30 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
| 17:33.36 | twisted | dialtone provided by your phone. you probably want to enable sip debug to determine why you can't dial out |
| 17:35.19 | webar7 | hmm |
| 17:35.48 | webar7 | prefered codec or something? |
| 17:36.35 | webar7 | not sure what might have changed since the SPA2000 config is the same |
| 17:39.15 | webar7 | gaak lots of sip debug messages |
| 17:42.47 | *** join/#asterisk supjigator (n=shanebur@152.53.16.10) |
| 17:43.54 | supjigator | Any ideas how to fix this: asterisk: symbol lookup error: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: pri_set_error |
| 17:44.31 | stansmith | supjigator: do 'ls -l /usr/lib/asterisk/modules/chan_zap.so' |
| 17:44.41 | stansmith | and see if its a broken symbolic link |
| 17:45.15 | supjigator | Its a file |
| 17:45.35 | stansmith | is your zaptel service running? |
| 17:45.46 | supjigator | Yep. |
| 17:45.58 | supjigator | ztcfg works with no errors |
| 17:46.09 | stansmith | /etc/zaptel.conf and /etc/asterisk/zapata.conf .. |
| 17:46.09 | stansmith | nm |
| 17:46.16 | stansmith | what about modules.conf? |
| 17:46.36 | tzafrir | supjigator, you built asterisk with support for libpri and now you run it without libpri? |
| 17:46.51 | tzafrir | supjigator, or with a different version of libpri? |
| 17:47.00 | supjigator | I have libpri |
| 17:47.11 | tzafrir | supjigator, ldd /usr/lib/asterisk/modules/chan_zap.so | grep libpri |
| 17:47.15 | supjigator | I don't think there is a way to run without it is there? |
| 17:47.31 | stansmith | you can run asterisk without libpri |
| 17:47.37 | tzafrir | there is. But this is a build-time decision |
| 17:47.44 | stansmith | yea |
| 17:47.50 | supjigator | nada |
| 17:48.25 | supjigator | So it built without it then? |
| 17:48.33 | supjigator | ldd returned no matches. |
| 17:49.04 | stansmith | supjigator: did you compile asterisk after first installing libpri? |
| 17:49.29 | supjigator | I did libpri, then zaptel then asterisk |
| 17:49.46 | supjigator | I'll try setting the config option. |
| 17:49.54 | supjigator | I see it now but I've never had to set it in the past. |
| 17:50.22 | stansmith | when i type that ldd command, i get this |
| 17:50.47 | stansmith | "libpri.so.1.0 => /usr/lib/libpri.so.1.0 (some hex i think)" |
| 17:50.58 | stansmith | err..location in memory perhaps |
| 17:52.14 | supjigator | Wonder why it built without it? |
| 17:53.01 | stansmith | did you make sure to get all the prerequisites? |
| 17:53.11 | *** part/#asterisk granviller (n=granvill@72x18x39x109.static.newroadstelecom.net) |
| 17:53.12 | stansmith | i believe you need the kernel-headers |
| 17:53.31 | supjigator | Yep |
| 17:53.41 | tzafrir | the error is not with the kernel. it is with libpri |
| 17:53.55 | tzafrir | pri_set_error |
| 17:54.10 | stansmith | did you make install libpri? |
| 17:54.11 | stansmith | or just make? |
| 17:54.52 | supjigator | just make |
| 17:54.54 | stansmith | ah |
| 17:54.59 | stansmith | yea you need to make install after make |
| 17:55.16 | supjigator | oh yea make install in the libpri |
| 17:55.18 | supjigator | I did that |
| 17:55.25 | supjigator | the includes libs etc are there and correct. |
| 17:56.11 | *** join/#asterisk granviller (n=granvill@72x18x39x109.static.newroadstelecom.net) |
| 17:57.35 | supjigator | Not that didn't fix it. |
| 17:57.46 | supjigator | and configure script looks good. It found libpri |
| 17:58.08 | webar7 | what is the correct syntax for replacing SetAccount(6135551212) (which lets our ITSP bill us ...) we have Set(CDR(6135551212)) |
| 17:58.14 | webar7 | is that wrong? |
| 17:58.29 | *** part/#asterisk nitrus^ (n=nitrus@cpe-76-166-248-27.socal.res.rr.com) |
| 17:58.35 | _ShrikE | webar7: yes, thats wrong |
| 17:58.35 | supjigator | http://www.pastebin.ca/911386 |
| 17:58.46 | supjigator | is the area of configure it seems to have found things |
| 17:58.59 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 17:59.22 | webar7 | _ShrikE, :-) |
| 18:00.42 | webar7 | _ShrikE, should it be |
| 18:00.59 | webar7 | Set(CDR(6135551212)=secretbillingnumber) |
| 18:01.03 | webar7 | or something? |
| 18:01.04 | stansmith | supjigator: i posted my install script |
| 18:01.39 | stansmith | take out the "-j3" off of make if you only 1 core on your CPU |
| 18:01.45 | supjigator | stansmith: Where? |
| 18:01.51 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 18:01.52 | stansmith | oops |
| 18:01.58 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:01.58 | stansmith | 1 sec |
| 18:01.58 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 18:02.07 | stansmith | http://www.pastebin.ca/911389 |
| 18:02.15 | _ShrikE | webar7: If you are were using SetAccount you should be using Set(CDR(accountcode)=foo) now. |
| 18:02.41 | *** join/#asterisk rcahilig (n=test@203.115.187.98) |
| 18:02.59 | stansmith | actually, what i posted is kinda sloppy, you might want to take out some of the comments such as "#make menuselect" and change it to "make menuselect" |
| 18:04.11 | stansmith | im going to go grab lunch, ill be back in a minute |
| 18:04.30 | ZPertee | for my outgoing context I don't want to use pattern matching and I don't want to use an exact digit can I do something like exten => s,1,Dial(Zap/1/${EXTEN})? |
| 18:04.33 | webar7 | _ShrikE, what's the "foo" any value? |
| 18:04.51 | _ShrikE | Its whatever you want to set your accountcode to |
| 18:05.46 | [TK]D-Fender | ZPertee: You want the account you're pointing to that context to only be able to dial 1 thing regardless of what they enter? |
| 18:07.11 | ZPertee | I dont want to have to dial 9 to reach an outside line or anything else. I just want my sip account to be able to pick up the phone and dial right out |
| 18:07.22 | Katty | hmm-home: you around? |
| 18:07.24 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 18:07.39 | hmmhesays | No but hmmhesays is |
| 18:07.43 | Katty | hmmhesays: hai! |
| 18:07.51 | hmmhesays | hey there |
| 18:07.59 | Katty | hmmhesays: have time to help me with cdr sql stuffs? |
| 18:08.16 | hmmhesays | yeah I can give you some pointers |
| 18:08.18 | Katty | kk |
| 18:08.20 | *** join/#asterisk Anthony76 (n=Anthony@LAubervilliers-153-52-32-150.w217-128.abo.wanadoo.fr) |
| 18:08.59 | supjigator | stansmith: I just blew away my asterisk modules dir and did make distclean. And now it works. |
| 18:10.06 | [TK]D-Fender | ZPertee: well what you were showing was a thought of only dialing 1 thing. Well for not needing a prefix, just don't USE one. You will need a pattern of course for this,. |
| 18:10.45 | webar7 | _ShrikE, oh so my ITSP doesn't care about it |
| 18:11.02 | _ShrikE | I dont even know who your I |
| 18:11.02 | *** join/#asterisk Datax (n=Datax@glou.nurvnet.org) |
| 18:11.05 | Datax | Hi, |
| 18:11.06 | Anthony76 | heloo all. I use asterisk as a "gateway for sipx". When I try to transfert an internal call to external call it don't work. I have this on my asterisk log: |
| 18:11.07 | Anthony76 | Feb 20 19:00:54 NOTICE[15374]: chan_sip.c:6932 get_refer_info: Supervised transfer requested, but unable to find callid '3c268e857a12-daspf0mgsqr3@snom360-000413236351'. Both legs must reside on Asterisk box to transfer at this time. |
| 18:11.08 | *** join/#asterisk bartpbx (n=bartpbx@217.24.210.201) |
| 18:11.08 | _ShrikE | I dont even know who your ITSP is :) |
| 18:11.27 | Anthony76 | How can I fix this problem. I think that is a problem about reinvite no? |
| 18:11.28 | Datax | I'm playing with mpg123 and can't get it to work. Does the server running asterisk need a sound card in it ? |
| 18:11.33 | Datax | because it doesn't have one |
| 18:11.51 | Anthony76 | thanks for your help |
| 18:11.58 | [TK]D-Fender | Datax: No. |
| 18:12.11 | bartpbx | hello, i have a problem with multible queues and the answer squence |
| 18:12.29 | Datax | I've tried running the command directly in the console and it give me a "error: No supported rate found!" message |
| 18:12.38 | Datax | any idea why ? |
| 18:12.43 | webar7 | _ShrikE, unlimitel |
| 18:12.56 | [TK]D-Fender | Datax: how are you calling it, what version are you using, and what are the specs of your mp3? |
| 18:13.28 | Datax | 0.61 |
| 18:13.29 | _ShrikE | webar7: I still dont know what your ITSP want you to do. You should verify that with them. |
| 18:13.31 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:13.38 | _ShrikE | s/want/wants/ |
| 18:13.42 | Datax | I'm trying to stream a webradio |
| 18:13.47 | Datax | off a relay on my lan |
| 18:13.55 | webar7 | I guess it should be --> Set(CDR(accountcode)=6135551212) |
| 18:14.00 | Datax | I'm guessing that I need to convert the stream after what you have just asked me :p |
| 18:14.00 | webar7 | or something |
| 18:14.32 | webar7 | since it used to be ---- exten => _NXXNXXXXXX,1,SetAccount(6135551212) |
| 18:14.37 | [TK]D-Fender | Accountcode is foryour benefit (for CDR tracking) and should have no impact on the call being placed |
| 18:14.45 | _ShrikE | ^^^^^ |
| 18:14.51 | webar7 | hmm |
| 18:14.57 | webar7 | ok |
| 18:15.27 | *** join/#asterisk Skarmeth (n=Skarmeth@201009024030.user.veloxzone.com.br) |
| 18:15.43 | [TK]D-Fender | Datax: Yes, you clearly need to convert the stream. There are samples on the WIKI, go look them up |
| 18:16.02 | Datax | thanks I'll test them and if they work I'll know what my problem is :) |
| 18:16.23 | [TK]D-Fender | Datax: 8khz mono <- |
| 18:16.25 | webar7 | I think the files are from there default setup http://www.unlimitel.ca/iaxsetup.html |
| 18:16.45 | webar7 | which is for 1.2 perhaps I don't have to convert anything |
| 18:16.46 | [TK]D-Fender | webar7: Very dated, thats 1.0 syntax |
| 18:16.51 | granviller | http://pastebin.ca/index.php I've setup the sip provider to dial my extension directly and when it does, I get two-way voice, when I have it set to dial using the s set of command it is one way audio. context [day] |
| 18:16.56 | webar7 | yeah |
| 18:17.01 | granviller | oops |
| 18:17.02 | granviller | http://pastebin.ca/index.php |
| 18:17.07 | robl^ | webar7: the account code only places the "code" in one of asterisk's CDR log files.. it doesn't get used by your provider |
| 18:17.07 | granviller | http://pastebin.ca/911421 |
| 18:17.22 | webar7 | [TK]D-Fender, found that out here last nite :-) |
| 18:17.26 | webar7 | robl^, good |
| 18:17.33 | webar7 | then I will leave it out for now |
| 18:19.34 | robl^ | webar7: for example -- a law firm may use it to bill calls back to a client. they will sort the CDR logs by account code -- where 10001 may equal "Super Client, Inc." |
| 18:20.03 | Idle | whats the name of a good IAX softphone? |
| 18:21.23 | Anthony76 | Hello, How can I fix this problem??? -> Feb 20 19:00:54 NOTICE[15374]: chan_sip.c:6932 get_refer_info: Supervised transfer requested, but unable to find callid '3c268e857a12-daspf0mgsqr3@snom360-000413236351'. Both legs must reside on Asterisk box to transfer at this time. |
| 18:22.05 | *** join/#asterisk simbol76ss (n=simbol@87.3.209.65) |
| 18:22.13 | jameswf | mozphone does iax |
| 18:22.47 | SomethingISOdd | hello quick question i am using php $time = $agi->get_variable("ANSWEREDTIME"); when i echo time all i get is Array |
| 18:23.09 | SomethingISOdd | anyone know how i can get the actual value? |
| 18:24.17 | webar7 | ok good |
| 18:25.20 | webar7 | so the bit in Set(CDR() can be whatever I want in my records |
| 18:25.35 | SomethingISOdd | seanbright are you still here sit? |
| 18:25.37 | SomethingISOdd | sir |
| 18:27.03 | Juggie | SomethingISOdd, do a print_r($time) |
| 18:27.06 | Juggie | to print the array |
| 18:27.39 | SomethingISOdd | ok |
| 18:28.29 | SomethingISOdd | Juggie but thats going to only print array isnt it. or is it going to actually print the value? |
| 18:32.03 | Katty | hi Juggie! |
| 18:32.43 | stansmith | would anyone here be able to chat about the "app_swift" module? I have it working, but just want to compare some things |
| 18:32.47 | drmessano | I wonder if ABC is going to show Armageddon during the satellite shoot down |
| 18:33.07 | granviller | can anyone help me figure out why when I have the incoming route set to dial an extension I get two -way audio, but when I have it ring a group it gets one way audio? |
| 18:33.11 | robl^ | twisted: you scare me! did you forget to take the tablets again? |
| 18:33.25 | granviller | drmessano: lol that would be awesome |
| 18:35.27 | drmessano | $100 bucks says some small island nation of under 200 people declares war on the US for launching a missile at the satellite |
| 18:35.37 | drmessano | I bet that crap happens all the time and we never hear about it |
| 18:36.22 | jameswf | we are actually shooting a missle at iran, the satalite deal is a cover |
| 18:36.50 | jameswf | then we will say oops dont know why it hit iran |
| 18:37.05 | drmessano | SHHH... not so loud Ahlajamakahamed will hear you and believe you |
| 18:37.07 | stansmith | "a zero must of been off..hmm.." |
| 18:37.12 | granviller | ~sipnat |
| 18:37.13 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:37.13 | robl^ | I thought it was canada due to socialized healthcare? |
| 18:37.22 | jameswf | damn floating point argh |
| 18:37.47 | drmessano | Its amazing how 33 degrees becomes 23 degrees.. damn buttons so close together |
| 18:37.54 | drmessano | "Korean Keyboard" |
| 18:38.07 | [TK]D-Fender | jameswf: Yes... will likely be launched from a floating point... somewhere not too far off the coase no doubt :p |
| 18:38.09 | granviller | no no, it's that russian keyboard |
| 18:38.10 | robl^ | drmessano: metric conversion error |
| 18:38.11 | jameswf | its like the worm on office spaceve mis placed a decimal... you said it would take years .. I must ha |
| 18:38.26 | clickonce | This fring shit really sucks. I can't get it to contact my Asterisk server. |
| 18:38.35 | jameswf | wow that was fd up |
| 18:38.48 | drmessano | "Sorry guys, we calculated the coordinates in Celsius, and the rocket only understood metres" |
| 18:38.48 | JenniferAkemi- | i have a question about templates (!). Are there any variables like the kind in macros I can use to access for instance the context i am in or something? |
| 18:39.19 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-85-11.vif.net) |
| 18:39.22 | flush | yo.. |
| 18:39.37 | stansmith | would anyone here be able to chat about the "app_swift" module? I have it working, but just want to compare some things |
| 18:39.39 | flush | i received my TDM400P card today but the guy shipped it with 3x FX0 and 1x FXS |
| 18:39.43 | flush | i wanna use one line, with 3 phones |
| 18:39.51 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 18:39.51 | *** mode/#asterisk [+o russellb] by ChanServ |
| 18:39.57 | flush | can it work anyway or i gotta send him the modules back to get them changed for FXS |
| 18:40.02 | flush | *changed for FXO |
| 18:40.29 | flush | shit thats the inverse, i have the card with 3x FXS and 1x FX0 -- can i use one line with 3 phones ? |
| 18:40.31 | granviller | what kind of phones are they? |
| 18:40.54 | jameswf | fxo and fxs are NOT interchangible |
| 18:40.57 | flush | normal phones.. analog |
| 18:41.02 | granviller | FXS is used to connect to analog phones. |
| 18:41.05 | Nivex | somebody remind me again how to download Asterisk and Zaptel without a GUI browser |
| 18:41.07 | jameswf | ~fxx |
| 18:41.11 | jameswf | ~fxs |
| 18:41.12 | jbot | i heard fxs is foreign exchange station - or the type of port you need to connect a analog device (phone, fax machine) to a pbx |
| 18:41.13 | *** part/#asterisk bartpbx (n=bartpbx@217.24.210.201) |
| 18:41.16 | stansmith | Nivex: wget |
| 18:41.27 | flush | oh |
| 18:41.28 | jameswf | ~wget |
| 18:41.28 | jbot | extra, extra, read all about it, wget is an excellent file retrieval utility (HTTP and FTP) with many options, or the command `apt-get install wget wget-ssl`, or virtual bacon |
| 18:41.31 | flush | and FXO is for what then ? |
| 18:41.36 | Nivex | stansmith: doesn't get you past the javascript redirect thingie |
| 18:41.39 | Nivex | I'm using links |
| 18:41.42 | robl^ | ~fxo |
| 18:41.43 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 18:41.43 | stansmith | 1 sec |
| 18:41.50 | granviller | to put it simply, FXo connects to the wall jack |
| 18:41.54 | flush | i thought FXS was for phone lines connected to the pbx and FXO was to connect phones to the pbx |
| 18:41.54 | drmessano | go to downloads.digium.com |
| 18:41.57 | stansmith | wget http://downloads.digium.com/pub/asterisk |
| 18:42.00 | [TK]D-Fender | flush: You should ahve 3 Green, 1 Red. Is that the case? |
| 18:42.02 | *** join/#asterisk qdk (n=qdk@195.242.194.42) |
| 18:42.04 | stansmith | that is the base asterisk directory Nivex |
| 18:42.08 | flush | [TK]D-Fender affirmative |
| 18:42.19 | [TK]D-Fender | flush: tHEN YOU HAV WHAT YOU NEED |
| 18:42.22 | [TK]D-Fender | darn caps |
| 18:42.28 | flush | cool |
| 18:42.37 | granviller | Fender, would you mind looking at my file again? |
| 18:42.45 | Katty | what's happened to file? |
| 18:42.50 | JenniferAkemi- | the book talks about this template thing (!) which sounds great |
| 18:42.52 | Katty | did he fall off the face of the planet? |
| 18:42.59 | twisted | smoking in the boys room |
| 18:43.18 | Nivex | a'ight, manual typage required but I got it. |
| 18:43.23 | flush | so with that card with 3x FXS and 1x FXO modules i can rock a asterisk box ? |
| 18:43.40 | stansmith | flush: depends on what u want |
| 18:43.41 | flush | have the ultimate control of my phone line ? |
| 18:43.44 | granviller | yes, and all three phones will share that one line. |
| 18:43.45 | [TK]D-Fender | JenniferAkemi-: its almost never mentioned here, in fact I last brought it up about a month ago in that I thought it'd be a great idea to implement, only to find that it HAD. |
| 18:43.46 | *** join/#asterisk quigon (n=matias@32.59.64.130) |
| 18:43.54 | jameswf | Step 1 cut a hole in the (asterisk) box |
| 18:44.12 | flush | just want to be able to set the conversation record, phonebox.. options like "press 1 or press 2" and music in background |
| 18:44.14 | JenniferAkemi- | [TK]D-Fender: do you know if there's a way to use variables in it? |
| 18:44.20 | Katty | file: horay! |
| 18:44.21 | file | Katty: what does Miss KittyKatty require? |
| 18:44.23 | [TK]D-Fender | granviller: They do not inherently share anything. What you do is up to your dialplan. |
| 18:44.24 | Katty | twisted: herroes (= |
| 18:44.27 | Katty | file: oh. |
| 18:44.30 | Katty | file: i dunno. |
| 18:44.32 | twisted | hi Katty |
| 18:44.34 | Katty | file: oj? |
| 18:44.40 | Katty | twisted: did you know more storms are coming? |
| 18:44.41 | file | Katty: I don't have any :( how about apple juice? |
| 18:44.43 | drmessano | simpson? |
| 18:44.46 | Katty | file: k |
| 18:44.46 | drmessano | Hes in jail |
| 18:44.51 | flush | can i do this with the TDM400P ? |
| 18:45.13 | [TK]D-Fender | flush: Do waht? |
| 18:45.16 | drmessano | I've heard that once women get a little OJ in them, it's fatal |
| 18:45.23 | [TK]D-Fender | flush: Oh, all that... yes |
| 18:45.29 | flush | omg nice |
| 18:45.34 | Katty | i don't even want to think about that, drmessano |
| 18:45.36 | [TK]D-Fender | flush: Get reading : |
| 18:45.37 | [TK]D-Fender | ~book |
| 18:45.38 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 18:45.40 | [TK]D-Fender | ^^^^^ |
| 18:45.48 | flush | aight ill read for sure |
| 18:45.51 | robl^ | granviller: Asterisk is NOT a phone system. It's like a PBX kit. You get all the bits and you assemble them to create about any phonesystem you can imagine. "Some assembly required." Out of the box isdoes nothing |
| 18:45.56 | drmessano | Or buy the book |
| 18:46.04 | drmessano | Its a big heap of bathroom reading |
| 18:46.27 | Katty | robl^: oh it does something. |
| 18:46.33 | Katty | robl^: it makes a loverly paper weight. |
| 18:46.44 | *** join/#asterisk joelsolanki (i=joelsola@220.224.122.65) |
| 18:46.45 | Katty | robl^: holds doors open nicely too. |
| 18:46.56 | *** join/#asterisk bronson (n=bronson@adsl-76-199-198-19.dsl.pltn13.sbcglobal.net) |
| 18:47.00 | joelsolanki | Hi room |
| 18:47.04 | [TK]D-Fender | Katty: Doubly ironic thats is a large AMOUNT of paper itself. |
| 18:47.06 | Katty | <room> hai. |
| 18:47.08 | robl^ | Katty: hahah! true. ;-) But if you spend a couple hours reading and learning, it can do marvelous things |
| 18:47.28 | Katty | robl^: yeah, like redirect all your calls to me to china. |
| 18:47.47 | twisted | Katty: i'm quite familiar with the weather. no more storms coming |
| 18:47.55 | twisted | at least, not here. |
| 18:48.01 | Katty | twisted: oh? we're expecting another 8 inches of ice :/ |
| 18:48.09 | twisted | hahahaha |
| 18:48.09 | Katty | twisted: can i live in your closet for awhile? |
| 18:48.09 | drmessano | The book is also just the right size to knock a n00b unconcious without causing permanent damage |
| 18:48.14 | twisted | ice is fun. |
| 18:48.15 | joelsolanki | buyed sangoma 104D. need to have ss7 signalling with 104D sangoma. |
| 18:48.34 | joelsolanki | sangoma's ss7 solution seems costly to us. |
| 18:48.44 | joelsolanki | therefore we want to use opensource |
| 18:49.02 | stansmith | u buyed? |
| 18:49.09 | Katty | twisted: i drove in the dark, for 2 hours, on black ice. |
| 18:49.16 | joelsolanki | yes sangoma 104D card. |
| 18:49.19 | RypPn | I'm looking at zaptel-1.4.9 atmo, and i'm wondering how I set the echo canceller, zconfig.h seems to have gone |
| 18:49.19 | twisted | Katty: yeah, done it many times before. |
| 18:49.21 | Katty | twisted: because some moron of a manager thought i had to go to a place an hour away that day. |
| 18:50.13 | Katty | i actually have an asterisk book around here somewhere |
| 18:50.24 | Katty | not really sure where. |
| 18:50.47 | angryuser | and why dont we OCR the Book SE? |
| 18:51.02 | Katty | ocr is nasty. |
| 18:51.08 | Katty | i don't even wanna think about proof reading it. |
| 18:51.18 | drmessano | OCR? |
| 18:51.21 | drmessano | Why? |
| 18:51.26 | clickonce | Wie! It works! Let's go get UniPhone! :D |
| 18:51.31 | mchou | it's available free as pdf |
| 18:51.43 | mchou | no need to ocr |
| 18:51.43 | Katty | there you go, problem solved. |
| 18:51.43 | joelsolanki | founded that sangoma 104D works with chan_ss7 |
| 18:51.48 | clickonce | I truly hope UniPhone is easier to get working that the shitty Fring. |
| 18:51.54 | stansmith | what is OCR and Book SE? |
| 18:51.57 | clickonce | Anyone got UniPhone/Asterisk experience? |
| 18:51.59 | angryuser | well the ~book give us first edition, when second is out |
| 18:52.04 | drmessano | No |
| 18:52.07 | drmessano | ~book |
| 18:52.07 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 18:52.13 | drmessano | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) |
| 18:52.15 | stansmith | i got that book |
| 18:52.18 | joelsolanki | can chan_ss7 + sangoma 104D used in production server ? |
| 18:52.19 | drmessano | 2nd means SECOND |
| 18:52.36 | Katty | ocr is taking a printed document, scanning it in, and turning it into text via an automated process. |
| 18:52.38 | angryuser | ah is see changed then ;) |
| 18:52.40 | stansmith | o yea, i was thinking of something else |
| 18:52.42 | Katty | which is not very good yet. |
| 18:52.46 | mchou | drmessano: no need to belabor the obvious |
| 18:52.48 | angryuser | very good |
| 18:52.48 | Katty | in fact, it mostly still sucks. |
| 18:52.48 | drmessano | angryuser: Months and months ago |
| 18:53.29 | stansmith | the books alright in my opinion |
| 18:53.35 | angryuser | <Katty> finereader works fine with printed documents |
| 18:53.54 | drmessano | OCR is a waste |
| 18:54.01 | stansmith | ~OCR |
| 18:54.01 | jbot | methinks ocr is (Optical Character Recognition) This is the technology that allows computers to "read" the text from physical objects. It requires a graphical representation of text to interpret. This usually comes from a scanned image. |
| 18:54.07 | mchou | stansmith: only if you need a tutorial on configuration |
| 18:54.14 | stansmith | mchou: thats what i needed |
| 18:54.24 | mchou | stansmith: rest of the book could have used more depth |
| 18:54.29 | stansmith | agreed |
| 18:54.30 | Katty | angryuser: i don't believe you. |
| 18:54.30 | angryuser | <drmessano> you prefer to retype it all ? |
| 18:54.50 | angryuser | <Katty> as you wish ;) |
| 18:55.19 | Katty | *hee* |
| 18:55.23 | mchou | stansmith: plus that book took way too long to get to the point |
| 18:55.36 | drmessano | Most documents are created electronically now.. So no need to OCR something from a typewriter in 2008.. Most scans of documents now are for archive purposes, so a PDF is more than sufficient |
| 18:55.40 | drmessano | OCR has no niche anymore |
| 18:55.52 | Katty | angryuser: i wanna watch that movie tonight now. |
| 18:56.10 | mchou | drmessano: maybe in banks, who knows |
| 18:56.42 | angryuser | <drmessano> yes, so when you buy a book/magasine , you assume that you get a copy of that document ? |
| 18:56.43 | mchou | maybe even law firms |
| 18:56.44 | thudd | OCR is nice for search indexing paper archives and mail and whatnot, but is still pretty rotten for actual readable conversion |
| 18:57.29 | drmessano | angryuser: Still havent convinced me why you need to OCR it when you can PDF it just the same.. Unless you're copying the text out to use as your own, which is lame |
| 18:57.42 | stansmith | lol@infringement |
| 18:58.40 | angryuser | <drmessano> it is usefull when you need to extract some data from many documents and work with them |
| 18:59.22 | drmessano | Yep, infringement from magazines is cool |
| 18:59.55 | angryuser | <Katty> movie ? |
| 19:00.05 | angryuser | <Katty> try las vegas parano |
| 19:00.45 | seanbright | angryuser: the princess bride, man. |
| 19:02.40 | SomethingISOdd | hello quick question i am using php $time = $agi->get_variable("ANSWEREDTIME"); when i echo time all i get is Array, can anyone recommend a way of getting the actual value |
| 19:02.45 | angryuser | ah ok , as you wish thing |
| 19:03.21 | seanbright | SomethingISOdd: http://us2.php.net/array |
| 19:03.31 | seanbright | people write documentation for a reason, man. |
| 19:03.40 | stansmith | pwn |
| 19:06.07 | Anthony76 | I use 2 ipbx (sipx and asterisk). asterisk is my b2bua for sipx. How can I configure asterisk so it's as a gateway patton for example. I don't wan't that astrisk detect musiconhold, transfert etc...How can I do? |
| 19:06.35 | *** part/#asterisk joelsolanki (i=joelsola@220.224.122.65) |
| 19:09.26 | phocus | What account do you use to login when you telnet to asterisk |
| 19:10.56 | *** join/#asterisk kyron (n=kyron@211-217-static-ppp.3menatwork.com) |
| 19:11.35 | phocus | ? |
| 19:11.50 | *** join/#asterisk guillote_GNU (n=guillote@host157.201-253-55.telecom.net.ar) |
| 19:12.16 | [TK]D-Fender | phocus: First you don't use Telnet to log into * for anything other than AMI, and even then... you don't normally use this yourself. What are you trying to do>? |
| 19:12.35 | phocus | i am ging to write a plugin for media center |
| 19:12.44 | phocus | that will wait and watch for an incomming call |
| 19:12.49 | phocus | and display caller id on the tv |
| 19:13.10 | [TK]D-Fender | phocus: Go read up on AMI : |
| 19:13.12 | [TK]D-Fender | ~book |
| 19:13.13 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 19:13.14 | [TK]D-Fender | ^^^^^^^ |
| 19:13.35 | phocus | those are ami books? |
| 19:13.42 | [TK]D-Fender | phocus: Or you could jsut do a "System" call in your dialplan. |
| 19:13.52 | [TK]D-Fender | ~ami |
| 19:13.53 | jbot | ami is, like, the Asterisk Manager Interface, a way to control an Asterisk server via a TCP/IP socket. See http://voip-info.org/wiki/view/Asterisk+manager+API |
| 19:13.53 | Yourname`` | Hi the caller ID format is "Name" <4195550555> right? |
| 19:13.56 | [TK]D-Fender | ]^^^ |
| 19:14.41 | phocus | you mean have the server side sit on the media center, and have asterisk send a message to it |
| 19:14.44 | phocus | that is not a bad idea |
| 19:14.44 | [TK]D-Fender | Yourname` : for sip.conf etc, yes |
| 19:15.35 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 19:16.48 | phocus | ok, this is good |
| 19:16.53 | phocus | this is exactly what i need |
| 19:17.04 | phocus | the only problem i am having is, i dont know what user name and password to give it |
| 19:17.10 | phocus | i am giving it the maint login |
| 19:17.13 | phocus | but it wont work |
| 19:17.15 | Katty | [TK]D-Fender: i want M&Ms :/ |
| 19:17.26 | Katty | [TK]D-Fender: do you have M&Ms? |
| 19:18.42 | [TK]D-Fender | Katty: Nope, only W's here.... |
| 19:18.47 | ZPertee | any idea what would happen if I plugged a message waiting indicator tones generator device and a regular dial tone into a single phone line. on the phone would the mwi come through or the regular dial tone? |
| 19:18.51 | Katty | [TK]D-Fender: :< |
| 19:19.21 | [TK]D-Fender | ZPertee: Both supply voltage so when the line rings, etc, you'll likely fry something. |
| 19:20.29 | ZPertee | [TK]D-Fender: ok bad idea then. thought it was worth asking instead of running extra wires |
| 19:21.10 | *** join/#asterisk Netgeeks (n=chris@204.11.231.198.static.etheric.net) |
| 19:21.33 | Katty | Netgeeks: hai. |
| 19:21.44 | Katty | Netgeeks: did you bring me M&Ms? |
| 19:22.06 | Yourname`` | [TK]D-Fender: What about callfile? callerid: "User" <4195555555> |
| 19:22.19 | Netgeeks | Katty: Hi! No, I brought a nasty virus. If I give you M&M's they would have nasty virus too |
| 19:22.54 | Katty | Netgeeks: AVAST! |
| 19:23.05 | [TK]D-Fender | Yourname` :sounds right |
| 19:23.28 | [TK]D-Fender | Netgeeks: Nothing a little penicillin won't cure.... |
| 19:23.57 | Katty | Netgeeks: i have some moldy bread you can have. |
| 19:24.07 | Netgeeks | [TK]D-Fender And the codiene the doc gave me helps the ride out alot too! |
| 19:24.14 | Katty | gosh. |
| 19:24.22 | Katty | must be some virus. |
| 19:24.48 | Yourname`` | [TK]D-Fender: Thanks.. let me try |
| 19:24.54 | Netgeeks | Katty: Hrm, I think I'd need Mr. Salk to help me with that mold |
| 19:25.11 | [TK]D-Fender | Netgeeks: For the "ride", Astroglide! |
| 19:25.52 | Netgeeks | Katty: so how goes it with you? haven't chatted with you since LAST year! |
| 19:25.53 | *** join/#asterisk granviller (n=granvill@72x18x39x109.static.newroadstelecom.net) |
| 19:26.04 | Katty | Netgeeks: gosh, it was that long ago? |
| 19:26.18 | Katty | Netgeeks: hmmhesays is helping me with my first postgres/cdr thingy |
| 19:26.22 | granviller | what would be the reason I can get one-way audio on one extension but not on other extensions? |
| 19:26.26 | [TK]D-Fender | Katty: only implies <2 months :) |
| 19:26.43 | [TK]D-Fender | granviller: its not EXTENSIONS. is the config of your sip entries. |
| 19:26.45 | Netgeeks | Katty: I do think so! |
| 19:26.45 | Katty | [TK]D-Fender: oh, right. |
| 19:26.52 | Katty | Netgeeks: mad. |
| 19:26.57 | Katty | Netgeeks: shame on me. |
| 19:27.08 | Netgeeks | Katty: storing cdr in db is fun! well, relatively fun |
| 19:27.31 | Katty | Netgeeks: it's going to be more fun that digging through csv! |
| 19:27.47 | Katty | Netgeeks: hmmhesays also pointed me in the direction of asterisk-stat which freepbx uses to access db |
| 19:27.53 | Katty | Netgeeks: i shall have pretty graphs!! |
| 19:27.56 | Netgeeks | Katty: SQL is your friend! |
| 19:28.23 | Katty | Netgeeks: yeah, and i'm already worried about how to backup the database :/ |
| 19:28.38 | Katty | Netgeeks: but one thing at a time. my database doesn't even have tables yet. *hee* |
| 19:29.01 | granviller | [TK]D-Fender: this much I understand, but why would it allow for 2-way audio for one particular sip but not another when all the sip devices are setup the same, albeit, with different firmware on them. |
| 19:29.11 | Katty | Netgeeks: what's new with you? |
| 19:29.45 | [TK]D-Fender | granviller: PASTEBIN is your friend. |
| 19:29.58 | granviller | what would you like to see? |
| 19:30.04 | drmessano | SQL is a WASTE |
| 19:30.05 | Netgeeks | Katty: not much new here. I'm battling trying to figure out how to get lumenvox to work when I'm limited to using a version of asterisk that they don't support directly |
| 19:30.09 | Katty | [TK]D-Fender: for some reason, whenever is see that word, i think of 5 year olds in elementary eating glue. |
| 19:30.12 | drmessano | XML ! |
| 19:30.22 | Katty | Netgeeks: horrors :< |
| 19:30.26 | JerJer | select * from users where clue = 1 |
| 19:30.32 | Katty | Netgeeks: you can borrow my magic wand, if you want. |
| 19:30.44 | Katty | Netgeeks: but i cannot guarantee when it'll show up in the mail. |
| 19:31.01 | Netgeeks | Katty: so I got on here to ask russell about a patch he made for 1.4.13 release that might allow me to run a version after 1.4.13... but he's not answering in asterisk-dev yet. |
| 19:31.10 | drmessano | I tried to send Magic Missile via UPS |
| 19:31.19 | Netgeeks | Katty: oooooh, the magic wand! do send please! |
| 19:31.20 | drmessano | Driver hit a bump and it sent his truck to Saturn :( |
| 19:31.25 | granviller | [TK]D-Fender what would you like to see? |
| 19:31.48 | Katty | Netgeeks: kk, one magic wand coming up! |
| 19:31.59 | Katty | Netgeeks: pls read disclaimer before opening package. |
| 19:32.19 | [TK]D-Fender | granviller: sip.conf |
| 19:33.11 | drmessano | Don't ever send CLOUDSONG via UPS |
| 19:33.19 | drmessano | UPS drivers will STEAL your CLOUDSONG |
| 19:33.20 | stansmith | hello.conf |
| 19:33.45 | Qwell | CLOUDSONG? |
| 19:33.52 | Netgeeks | Katty: Thanks!!! talking to russell in other channel - will be spaced out for a bit |
| 19:33.59 | Katty | Netgeeks: kk |
| 19:34.01 | drmessano | http://www.youtube.com/watch?v=hN08jEthoB0 <-- NSFW |
| 19:34.11 | drmessano | Thats CLOUDSONG |
| 19:35.02 | drmessano | Gamers gone wrong |
| 19:35.11 | drmessano | Do NOT steal that dudes Cloudsong |
| 19:35.15 | Katty | I'm Epic. |
| 19:35.20 | stansmith | hey man |
| 19:35.31 | stansmith | my level 30 charizard would so kill ur level 36 pikachu |
| 19:35.40 | Katty | you do not parse. |
| 19:36.06 | drmessano | Alrighty guys |
| 19:36.08 | drmessano | Lets do this |
| 19:36.13 | drmessano | LEEEROY JENKINSSSS |
| 19:36.18 | Katty | sorry. i don't feel like running kara right now. |
| 19:36.30 | drmessano | Leeroy Jenkins is my hero |
| 19:36.38 | drmessano | "At least I got chicken" |
| 19:36.53 | Katty | a mechanical chicken. |
| 19:36.54 | [TK]D-Fender | .... |
| 19:37.05 | granviller | [TK]D-Fender http://pastebin.ca/911520 my entire sip.conf |
| 19:37.14 | stansmith | i am really a human cyborg |
| 19:37.26 | drmessano | Ok dude, seriously |
| 19:37.31 | drmessano | I want my cloudsong back |
| 19:37.58 | [TK]D-Fender | granviller: You did not prevent reinvites all around like you're supposed to have. |
| 19:38.44 | stansmith | granviller: are you located in Ohio? |
| 19:38.56 | granviller | no |
| 19:42.06 | granviller | [TK]D-Fender: ok, but if 105 has canreinvite=yes, and so do all the other sip deivices, how does changing that allow for 2-way audio? |
| 19:43.26 | [TK]D-Fender | that is BAD! |
| 19:43.34 | [TK]D-Fender | granviller: you have to DISABLE it all around |
| 19:44.43 | vk4akp | Say I've got an ATA on the same LAN as my Asterisk server. How do I set up an internal SIP server on Asterisk so I can register my ATA with teh Asterisk box over the LAN? |
| 19:44.51 | granviller | [TK]D-Fender: I'm doing it, I just don't understand how that's gonna gix my 1 way audio, which is what I'm trying to get fixed at the momment. I'm frustrating you agian aren't I? |
| 19:45.25 | granviller | gix=fix |
| 19:45.50 | *** join/#asterisk Stingy1 (n=tbaa@80.120.42.206) |
| 19:45.55 | Stingy1 | hi |
| 19:46.00 | granviller | hello |
| 19:46.05 | [TK]D-Fender | granviller: Your reinvites are killing your audio because your phone cannot successfully take over the call from your provider |
| 19:46.06 | stansmith | ~hi | Stingy1 |
| 19:46.07 | jbot | Many greetings, | Stingy1, most strange traveller, to this IRCdom of plenty. |
| 19:46.28 | stansmith | ~hi stansmith |
| 19:46.29 | jbot | Many greetings, stansmith, most strange traveller, to this IRCdom of plenty. |
| 19:46.33 | stansmith | oops |
| 19:46.37 | [TK]D-Fender | granviller: Just set "canreinvite=no" in every entry and under [general] |
| 19:46.42 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:46.42 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 19:47.01 | Stingy1 | i installed * on etch with apt |
| 19:47.12 | stansmith | Stingy1: o yea? |
| 19:47.13 | Stingy1 | and try to use show dialplan |
| 19:47.22 | Stingy1 | No such command 'show' (type 'help' for help) |
| 19:47.45 | [TK]D-Fender | Stingy1: "dialplan show" |
| 19:47.51 | Stingy1 | show dialplan Show dialplan |
| 19:48.01 | Stingy1 | No such command 'dialplan' (type 'help' for help) |
| 19:48.04 | stansmith | yea its dialplan show |
| 19:48.09 | stansmith | dialplan reload, dialplan show |
| 19:48.32 | [TK]D-Fender | Stingy1: Go check your modules.conf and your moudles folder. Odds are you have no configs so no modules loaded |
| 19:49.27 | vk4akp | Say I've got an ATA on the same LAN as my Asterisk server. How do I set up an internal SIP server on Asterisk so I can register my ATA with the Asterisk box over the LAN? |
| 19:49.31 | Katty | i has banana! |
| 19:50.04 | `Sauron | Mew |
| 19:50.17 | lmadsen | Katty: interesting... when did you go in for the surgery? |
| 19:50.19 | `Sauron | I has a banana earllier |
| 19:50.22 | `Sauron | *earlier |
| 19:50.25 | *** join/#asterisk Wayhigh (i=noid@www.kevinlynn.com) |
| 19:50.51 | Katty | lmadsen: surgery? |
| 19:50.59 | Katty | lmadsen: baroo? |
| 19:51.05 | lmadsen | <Katty> i has banana! |
| 19:51.07 | Katty | oh |
| 19:51.08 | Katty | geeze. |
| 19:51.09 | lmadsen | :) |
| 19:51.11 | Katty | you perv. |
| 19:51.13 | lmadsen | ya... bad joke |
| 19:51.24 | lmadsen | Katty: ya... apparently you haven't met me |
| 19:51.26 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:51.37 | Katty | no, i don't believe so. |
| 19:51.49 | drmessano | lmadsen: My days of being a TFOT freeloader are over.. I got my hardcopy in the mail yesterday |
| 19:51.57 | drmessano | You're welcome :) |
| 19:51.59 | lmadsen | drmessano: w00t! thx! |
| 19:52.00 | Katty | lmadsen: in fact, most of the people i've met at * conference don't talk to me. |
| 19:52.07 | Katty | lmadsen: i think they're all shy. |
| 19:52.11 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 19:52.19 | lmadsen | Katty: really? interesting |
| 19:52.34 | lmadsen | I'd hope that I'd be conversive with you |
| 19:52.56 | Katty | lmadsen: you'd be surprising how depressing it is when no one wants to talk to you )= |
| 19:53.15 | lmadsen | Katty: I bet... and you're even cute... so I don't understand |
| 19:53.51 | Katty | ^_- |
| 19:54.03 | Katty | how do you know. |
| 19:54.14 | lmadsen | Katty: I saw a picture of you from cluecon onece |
| 19:54.15 | lmadsen | once* |
| 19:54.16 | Katty | i could be a horribly ugly wow geek. |
| 19:54.19 | Katty | oh, right. |
| 19:54.20 | Katty | k |
| 19:54.21 | lmadsen | heh |
| 19:54.24 | lmadsen | sorry, no foolin' me |
| 19:54.26 | `Sauron | Katty is all brains |
| 19:54.27 | Katty | :< |
| 19:54.34 | lmadsen | I'm smrt |
| 19:54.39 | Katty | and bananas, apparently. |
| 19:54.54 | djs | Uh oh, watch out for monkeys... |
| 19:54.55 | granviller | [TK]D- Fender: ok, I got that done, reloaded the sip and I still get 1 way audio. Before I was getting 2 way audio on a singel extension 105, now I don't |
| 19:55.03 | lmadsen | Katty: well, if we ever end up at a conference together, I'll make a point of at least saying hi, heh |
| 19:55.40 | granviller | [TK]D-Fender: the CLI now says it's attemping native bridinge between the incoming sip and the sip device |
| 19:55.44 | Katty | how do i switch users in postgres? |
| 19:55.51 | Katty | it's not \c |
| 19:56.02 | Katty | lmadsen: horay! |
| 19:56.16 | granviller | bridging* |
| 19:56.42 | lmadsen | Katty: ya... not sure if that is something you're looking forward to, but at least ONE person would talk to you, haha |
| 19:56.50 | Corydon76-dig | Katty: \h |
| 19:56.51 | *** join/#asterisk pithen (n=pithen@mail.graphlogic.com) |
| 19:57.00 | Katty | Corydon76-dig: i read help. |
| 19:57.06 | Katty | Corydon76-dig: i'm not sure what term i'm even looking for her. |
| 19:57.08 | Katty | here. |
| 19:57.11 | Corydon76-dig | Katty: could be no way |
| 19:57.28 | Katty | there must be a way! |
| 19:57.35 | Katty | else i shall asplode! |
| 19:57.58 | stansmith | hello? |
| 19:58.11 | Katty | elo. |
| 19:58.43 | Corydon76-dig | Katty: \set USER foo |
| 19:58.49 | *** join/#asterisk steliosk (n=Stelios@athedsl-290446.home.otenet.gr) |
| 20:00.00 | clickonce | Can you recommend any Asterisk howto that describes how to do certain tasks rather than explaining the entire Asterisk system? I.e. "how to intercept dial-outs to 911 and redirect them to a menu", etc. (Or any other number for that matter) |
| 20:00.31 | stansmith | me? |
| 20:00.45 | Katty | Corydon76-dig: neat. |
| 20:00.54 | Katty | Corydon76-dig: do you think it'd tell me if it did it or not? |
| 20:01.01 | Katty | Corydon76-dig: or just silently return me to prompt |
| 20:01.09 | Corydon76-dig | Katty: \set |
| 20:01.31 | Katty | you must think i know something about postgres. |
| 20:01.38 | Katty | you, sir, are horribly mistaken. |
| 20:01.53 | stansmith | postgres > mysql |
| 20:02.10 | Nugget | postgresql > * > sticking a pencil in your ear > mysql |
| 20:02.12 | Corydon76-dig | Katty: just type: \set |
| 20:03.05 | drmessano | hahaha |
| 20:03.19 | Corydon76-dig | with no arguments, it tells you what it knows |
| 20:03.37 | Katty | oh |
| 20:03.39 | Katty | neat |
| 20:04.38 | Corydon76-dig | and you think people don't pay attention to you... |
| 20:04.44 | [TK]D-Fender | clickonce: exten => 911,1,Goto(hell,s,1) |
| 20:05.03 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 20:05.06 | Katty | s,1,Background(lionelrichie) |
| 20:05.18 | clickonce | [TK]D-Fender: Thanks alot! |
| 20:05.36 | Katty | Corydon76-dig: i don't even pay attention to me. |
| 20:05.58 | Katty | and my pear is overly ripe. what a crappy afternoon. |
| 20:07.58 | J4k3 | this reminds me of a couple lines out of fear and loathing in las vegas |
| 20:08.41 | Katty | why would anyone want screaming monsters to take care of? |
| 20:08.43 | Katty | go get a dog. |
| 20:09.08 | JenniferAkemi- | woot :) I successfully made a call from my to my sip phone and from my sip phone out my pri to my landline |
| 20:09.27 | BBHoss | JenniferAkemi-, nice |
| 20:09.31 | JenniferAkemi- | :D |
| 20:09.57 | Katty | Corydon76-dig: http://www.lifestylepets.com/dogs.html problem solved. |
| 20:10.06 | Corydon76-dig | JenniferAkemi-: wait until you inadvertently send your internal 3-digit callerid across the country |
| 20:10.16 | JenniferAkemi- | only problem is that from pri -> sip sounds ok (it goes ulaw) but sip -> pri sounds awful (g726) now i'm going to install the g729 licenses i bought and see how that sounds |
| 20:10.37 | lirakis | probably not some thing terribly unique... but i just put together a script that will email new voice mail messages out automatically.. its 'generic' and should work for any number of vm boxes |
| 20:10.42 | Corydon76-dig | Katty: I don't like dogs enough to spend $15k on one |
| 20:10.45 | [TK]D-Fender | JenniferAkemi-: Where is your phone relative to your * server? |
| 20:10.54 | Katty | Corydon76-dig: kids cost far more than that. |
| 20:11.06 | lirakis | here is a temporary posting of the code for any one who wants it |
| 20:11.10 | Corydon76-dig | Katty: I like kids a whole lot more. :-P |
| 20:11.19 | Katty | Corydon76-dig: you, sir, are insane. |
| 20:11.21 | JenniferAkemi- | Corydon76-dig: heh. I can totally see that happening, but it won't since the pri is attached to my legacy switch and we don't send call display (MF FGD network) |
| 20:11.23 | J4k3 | kids will take care of you when you get old (ie - shove you in a nursing home) |
| 20:11.26 | lirakis | http://pastebin.ca/911563 |
| 20:11.31 | Katty | insanity pickup on aisle 5 peoples! |
| 20:11.43 | Katty | get locked and loaded and get him outta here! |
| 20:11.46 | JenniferAkemi- | [TK]D-Fender: my phone is in Newmarket, and the * server is in Quebec city |
| 20:12.01 | JenniferAkemi- | [TK]D-Fender: i have a vpn over residential dsl to the intranet in quebec city |
| 20:12.23 | [TK]D-Fender | JenniferAkemi-: low BW then? |
| 20:12.32 | JenniferAkemi- | [TK]D-Fender: this is the test setup of course. eventually the * server will be in montreal, and I guess we will be buying bw |
| 20:12.45 | [TK]D-Fender | JenniferAkemi-: if you can, go ulaw |
| 20:12.57 | JenniferAkemi- | [TK]D-Fender: no, that's not the problem, because when i set it to ulaw it sounds great. |
| 20:13.20 | BBHoss | how are you routing it to use g726 when incoming from the pri? |
| 20:13.24 | JenniferAkemi- | the application can't stand a ton of ulaw calls though, since BW is a concern. |
| 20:13.27 | clickonce | "_X.,1,Dial(SIP/${EXTEN}@" that is what I have no to dial out. Would it be possible to require the user to dial 0... to get to dial out? If so, what would _X. be replaced with and, what would EXTEN be replaced with to remove the prefixed 0? |
| 20:13.35 | pithen | Do you guys have a pasteboard I could paste to? |
| 20:13.41 | clickonce | I'm not sure what to look for in the documentation. |
| 20:14.10 | Katty | pithen: pastebin.ca |
| 20:14.14 | pithen | ty |
| 20:14.23 | JenniferAkemi- | BBHoss: in my sip.conf i have disallow=all, allow=g726, allow=g720,allow=ulaw |
| 20:14.36 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:14.46 | BBHoss | clickonce, : exten=> _0X.,1,Dial(SIP/${EXTEN:1}) |
| 20:14.54 | clickonce | BBHoss: Yay :) |
| 20:15.14 | BBHoss | JenniferAkemi-, why don't you use gsm or ulaw there? |
| 20:15.29 | BBHoss | if the problem is sound quality, not packet loss etc |
| 20:15.57 | JenniferAkemi- | BBHoss: eventually we hope to have enough users that ulaw will be too much bw to be practical |
| 20:16.18 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 20:16.26 | BBHoss | JenniferAkemi-, well what about gsm then |
| 20:16.38 | JenniferAkemi- | BBHoss: i thought that g729 was the "best" |
| 20:16.39 | BBHoss | or ilbc, speex etc |
| 20:16.48 | [hC] | Katty: PRI install go well? |
| 20:16.48 | JenniferAkemi- | BBHoss: we bought some licenses to do testing with, i jsut haven't installed them yet |
| 20:16.50 | BBHoss | JenniferAkemi-, if you want to pay big bux yeah |
| 20:16.54 | Katty | [hC]: i think so! |
| 20:16.59 | Katty | [hC]: fighting with postgres right now. |
| 20:17.00 | JenniferAkemi- | BBHoss: i think we can do $10 per channel |
| 20:17.06 | [hC] | Katty: excellent! (about the pri) |
| 20:17.13 | Katty | [hC]: thanks (= |
| 20:17.16 | [hC] | I have spent my fair share of time fighting with postgres. |
| 20:17.19 | JenniferAkemi- | BBHoss: probably get one of those cards |
| 20:17.20 | BBHoss | JenniferAkemi-, how many channels are you planning on using with g729? |
| 20:17.47 | Katty | so does a mcdonalds burger in their mouth. |
| 20:17.51 | J4k3 | but at least they don't sound like they're being put through a cheese grater, ilbc or gsm style |
| 20:17.53 | JenniferAkemi- | BBHoss: I don't really know. 3 t1's per box? |
| 20:18.17 | BBHoss | yeah youll need a card |
| 20:18.18 | J4k3 | could also be the junk I use that I call a phone |
| 20:18.22 | pithen | Could somebody take a look at http://pastebin.ca/911579? For some reason extensions dialed from outside don't work, and voicemail doesn't work on no answer (but does on busy) |
| 20:18.27 | BBHoss | J4k3, probably |
| 20:18.35 | Katty | and to think i'm going to put 16 t1s in a box. |
| 20:18.40 | Katty | there's something wrong with me. terribly wrong. |
| 20:18.46 | [TK]D-Fender | clickonce: Time to sit down and read the book. You seem to quite clearly have no understanding about dialplan patterns. This is pretty much the most important thing to learn & master in all of *. |
| 20:18.47 | [TK]D-Fender | ~book |
| 20:18.48 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 20:18.50 | [TK]D-Fender | ^^^^^^^^^^ |
| 20:18.56 | J4k3 | Katty: your personal cray? |
| 20:18.56 | J4k3 | :D |
| 20:19.03 | Katty | J4k3: i don't know what cray means. |
| 20:19.11 | Katty | J4k3: but it's not personal. |
| 20:19.49 | J4k3 | http://en.wikipedia.org/wiki/Cray_supercomputer#The_Cray_Research_years |
| 20:19.56 | J4k3 | computers for people with a real budget |
| 20:20.04 | J4k3 | 10-20 years ago |
| 20:20.59 | *** part/#asterisk granviller (n=granvill@72x18x39x109.static.newroadstelecom.net) |
| 20:21.49 | *** join/#asterisk tobias (n=tobias@cpe-069-134-226-227.nc.res.rr.com) |
| 20:24.02 | BBHoss | pithen, is your dial-string in macro-remote context? |
| 20:25.21 | BBHoss | if it is, you need to set a timeout and also actually send it to voicemail |
| 20:25.55 | pithen | BBHoss, that macro is called from the extensions context, which passes the phone # as an arg. Then the remote macro calls dialenum which builds the full string.. in any case thats not being used presently, the oneline macro is the one for internal ext's |
| 20:26.23 | BBHoss | ahh ok |
| 20:27.04 | *** join/#asterisk Skarmeth (n=Skarmeth@201009024030.user.veloxzone.com.br) |
| 20:29.07 | BBHoss | pithen, in my extensions.conf voicemail looks like this: exten => _X.,n(unavail),Voicemail(6000@default,u) |
| 20:29.20 | clickonce | "exten => 5,1,Dial(SIP/01234567@" what is so wrong with that? Do I need to espace the number somehow? In all the other calls to Dial in the config file they have not quoted the number. The error I get is that it can't register extension _X., priority 3, already in use. |
| 20:29.48 | SomethingISOdd | i hope someone can tell me i am trying to get a variable from one script to another now i have used set_variable and on the second script i used get_variable but no matter what i do all i keep getting is Array instead of the value |
| 20:29.58 | BBHoss | clickonce, post your entire extensions.conf |
| 20:30.57 | pithen | BBHoss, right.. voip info show both formats as being valid, although i could try switching it to be similar to yours (im running 1.2.5 btw) |
| 20:30.58 | [TK]D-Fender | clickonce: _X. is clearly NOT "5,1". They have nothing to dow with that error |
| 20:31.16 | clickonce | Well, if I comment that line I don't get the error :) |
| 20:31.28 | [TK]D-Fender | pithen: You should pastebin a failed attempt for us to see. |
| 20:31.30 | BBHoss | pithen, ahh, for 1.2 i can't help you, i'm on 1.4 |
| 20:31.57 | pithen | BBHoss, heh..someday I'll get around to updating.. |
| 20:32.21 | BBHoss | you have a really simple extensions.conf, it wouldnt be that big of a deal |
| 20:33.07 | pithen | BBHoss, I think it might carry over without any modifications (once I get it working right anyway).. I just rewrote it so I tried to not use anything depreciated |
| 20:33.26 | *** join/#asterisk mitio (n=wireman@78.83.96.4) |
| 20:33.53 | BBHoss | pithen, Also, take EVERYTHING from voip-info with a grain of salt |
| 20:34.17 | Katty | how do i add a password for a user in postgres, so i can login as them? |
| 20:34.25 | pithen | BBHoss, hah, i've taken it as bible.. do you have a recommendation instead? |
| 20:34.25 | Uni | 6 |
| 20:34.29 | Katty | some \set user password thingy? |
| 20:34.38 | BBHoss | pithen, not really |
| 20:35.02 | BBHoss | pithen, you can usually rely on the book to be correct |
| 20:35.21 | SomethingISOdd | anyone please |
| 20:35.58 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 20:36.44 | vk4akp | Anyone got a link on how to set up Asterisk as a SIP server? |
| 20:36.54 | BBHoss | ~book |
| 20:36.55 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 20:37.00 | vk4akp | Got the book. |
| 20:37.02 | vk4akp | Can't find it. |
| 20:37.11 | BBHoss | what do you mean by SIP server then? |
| 20:37.29 | vk4akp | OK. I want to register my ATA on the same LAN with the Asterisk server using SIP. |
| 20:37.40 | [hC] | thats pretty much step 1 in the book |
| 20:37.43 | BBHoss | yeah |
| 20:37.53 | [TK]D-Fender | vk4akp: Just like setting up about any other device out there. Go read. Check this as well for a quick sample : |
| 20:37.55 | [TK]D-Fender | ~jerjerguide |
| 20:37.56 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 20:38.15 | vk4akp | Basically so I don't have to loop back through the WAN to get a dial in. (Efectivly doubling the data used). |
| 20:38.28 | pithen | BBHoss, I do have a copy in pdf, ill have to double check what ive done against it |
| 20:38.41 | vk4akp | I've set up an Outbound SIP account. But now I want to set up the opposite. |
| 20:38.54 | [hC] | vk4akp: read. you were probably in too much of a rush to see it |
| 20:39.02 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.232.128) |
| 20:39.07 | BBHoss | vk4akp, are you using a control panel or just asterisk? |
| 20:39.09 | [hC] | vk4akp: its all right there, its a pretty simple process. read jeremy's post |
| 20:39.26 | vk4akp | Looking through your links now. |
| 20:39.27 | *** join/#asterisk cornelinux (n=corny@dslb-088-068-177-111.pools.arcor-ip.net) |
| 20:39.40 | *** join/#asterisk zobia (n=laurashr@222.212.75.120) |
| 20:39.45 | vk4akp | I am using plain Asterisk 1.4.11 emerged using Gentoo Linux. |
| 20:39.48 | zobia | hello everyone |
| 20:40.01 | zobia | i got a 500 internal error when using sip trunk. |
| 20:40.04 | vk4akp | No Control Panel. |
| 20:40.13 | zobia | but i don't know how to see more detail of the 500 error |
| 20:40.19 | zobia | anyone can help me out? |
| 20:41.13 | cornelinux | Hi, i am using external misdn. how can I change the callerid for outgoing calls? |
| 20:41.27 | vk4akp | No, that link had nothing relevant in it. :( |
| 20:41.37 | stansmith | hey man you got some food on your chin |
| 20:41.43 | bronson | Is there any talk about making Asterisk do better than 8MHz? |
| 20:41.48 | BBHoss | vk4akp, you are blind then |
| 20:41.55 | J4k3 | stansmith: nuts? :D |
| 20:41.59 | stansmith | lol |
| 20:42.19 | BBHoss | vk4akp, you configure the sip "server" in sip.conf |
| 20:42.31 | BBHoss | you setup users, peers, friends, etc |
| 20:42.34 | J4k3 | vk4akp: plz don't tell me you're trying to voip over what we discussed last night :) |
| 20:42.48 | clickonce | YEAH! This works wonders, now I just have to get a good ol machine to run this damn thing on and a brand new SPA-962 :) |
| 20:43.17 | vk4akp | Nope, went through it again. Definatly nothing relevant to what I asked. :( |
| 20:44.06 | BBHoss | vk4akp, then you aren't trying to setup a sip "server" you aren't explaining yourself |
| 20:44.19 | BBHoss | look all you need is this |
| 20:44.23 | BBHoss | [6000] |
| 20:44.28 | [TK]D-Fender | vk4akp: see the sip.conf section for [100] ? Well thats to set up an account for the sip device yuo want to be able to connect to *. From there processing is the job of your dialplan |
| 20:44.43 | vk4akp | There would have to be somewhere where I give a SIP server name for the Asterisk box. And create an account name and password, and codecs alowd etc. |
| 20:44.49 | mchou | clickonce: you get a good deal on the spa-962? |
| 20:45.06 | *** join/#asterisk Datax (n=Datax@glou.nurvnet.org) |
| 20:45.15 | BBHoss | vk4akp, you create accounts in sip.conf |
| 20:45.30 | [TK]D-Fender | vk4akp: read the book, and that guide. They show a complete picture for a sample config including SIP "phones" and even an ITSP, and the relevant dialplan to have them function together. |
| 20:45.30 | clickonce | mchou: ~2400 SEK (~$380) |
| 20:45.32 | SomethingISOdd | i hope someone can tell me i am trying to get a variable from one script to another now i have used set_variable and on the second script i used get_variable but no matter what i do all i keep getting is Array instead of the value can anyone help me fix this? |
| 20:45.33 | BBHoss | then you setup your ata to use them (usually with a web browser) |
| 20:45.39 | vk4akp | I will Explane again. I want to register my ATA on teh same LAN as my ASterisk Server, so ti does not have to dial out to an external SIP provider on the WAN to then get back to my ASterisk Server via the WAN into my LAN. |
| 20:45.49 | mchou | clickonce: damn...too rich for my blood |
| 20:46.05 | [TK]D-Fender | vk4akp: Its all in that guide. Get reading |
| 20:46.11 | clickonce | mchou: No, I'll have to ditch everything else :P |
| 20:46.13 | vk4akp | Read it. No help. |
| 20:46.18 | BBHoss | or you can pay me and ill do it for you :) |
| 20:46.26 | clickonce | mchou: Either living or the SPA962... I choose the 962 :P |
| 20:46.30 | [TK]D-Fender | vk4akp: What part don't you understand? |
| 20:46.31 | vk4akp | OK. Normal RTFM attitude. Fine. I will ask somewhere else. |
| 20:46.40 | BBHoss | vk4akp, no |
| 20:46.51 | *** join/#asterisk granviller (n=granvill@72x18x39x109.static.newroadstelecom.net) |
| 20:46.56 | BBHoss | vk4akp, you just want someone to do it for you, if we did that you'd never learn |
| 20:46.58 | granviller | hello again. |
| 20:47.31 | vk4akp | No I don't. I want a very basic understanding of what I have to do. Then I will quiet happily do all the hard work. I have not ffound anywhere in teh book what I am after. |
| 20:47.46 | mchou | huh??? |
| 20:47.54 | BBHoss | you setup a sip account in sip.conf, then configure your device to use it |
| 20:47.57 | [TK]D-Fender | vk4akp: What point have you gotten yourself to in configuring your system? |
| 20:48.04 | mchou | vk4akp: the book gives exactly what you just stated |
| 20:48.14 | vk4akp | It's installed. You can dial in, and I have even set up a conference. |
| 20:48.34 | vk4akp | Thats how far I have got it configured. |
| 20:48.46 | vk4akp | So I have outbound SIP registered and working. |
| 20:48.46 | [TK]D-Fender | vk4akp: Show us your configs (minus passwords) so we can see where you have gottent hings to. (pastebin) |
| 20:48.53 | [TK]D-Fender | ~pb |
| 20:48.53 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 20:48.55 | [TK]D-Fender | ^^^^^^^^^ |
| 20:48.57 | granviller | it seems that my asterisk box is not using the rtpend in the dtp.conf file, I saw it trying to use a port in the 40k range. where should it be reading in the rtp.conf file? |
| 20:49.08 | BBHoss | vk4akp, here is my sip.conf http://pastebin.com/m269e8531 |
| 20:49.13 | vk4akp | Yea , I have Wgetpaste. |
| 20:49.25 | vk4akp | OK. I'll look at yours. |
| 20:49.30 | granviller | err rtp.conf |
| 20:49.35 | vk4akp | Have you done what I am talking about within your config? |
| 20:49.43 | [TK]D-Fender | vk4akp: show us what YOU'VE done so we can see whats "missing" |
| 20:50.05 | vk4akp | yes, reading now. Give me time. |
| 20:50.10 | mchou | lol |
| 20:50.15 | stansmith | ha |
| 20:50.19 | mchou | what a BS artist |
| 20:51.02 | vk4akp | No. Thats not what I'm talking about. |
| 20:51.08 | vk4akp | U R missing the point. |
| 20:51.13 | clickonce | Would it be possible to send a different text string to the SIP client as "caller id"? E.g. "Support" or "Financial" depending on what number the user dialed to get in? I.e. extension 01 for support, 02 for financial, etc. |
| 20:51.35 | SomethingISOdd | clickonce yes its under carrier id name |
| 20:51.46 | outtolunc | the point was, that was a tasty burger i got to eat while watching this chatter with amusement <G> |
| 20:51.51 | [TK]D-Fender | vk4akp: Can you please show us what you've done so we can help you... |
| 20:52.25 | J4k3 | big kahuna burger? |
| 20:52.36 | J4k3 | Sprite! (tm) |
| 20:52.43 | stansmith | openwrt r0x! |
| 20:52.45 | mchou | [TK]D-Fender: come on man, the only thing this dude is good at is social engineering |
| 20:52.57 | [TK]D-Fender | clickonce: "core show function CALLERID" |
| 20:53.08 | BBHoss | social deengineering you mean |
| 20:53.09 | [TK]D-Fender | mchou: We shall see. |
| 20:53.10 | vk4akp | An ATA requires user name, password, and a SIP proxy. For example au.voxalot.com now to register my ATA locally with Asterisk. Where would I for example put au.sabayon.com in the Asterisk conf files to make Asterisk allow registration via sip to it's own internal server called au.sabayon.com |
| 20:53.17 | SomethingISOdd | [TK]D-Fender do you happen to know anything about phpagi scripting? |
| 20:53.21 | mchou | stansmith: can you make openwrt work with my wrtp54g? |
| 20:53.34 | [TK]D-Fender | SomethingISOdd: Only a little bit. |
| 20:53.36 | BBHoss | vk4akp, we're not helping till you post sip.conf |
| 20:53.52 | stansmith | mchou: i actually use dd-wrt at my house |
| 20:54.02 | SomethingISOdd | [TK]D-Fender i think any agi script would be the same i am getting stuck on extracting the value out of a get_variable that was set in another script |
| 20:54.08 | vk4akp | And D-Fender. I can't send U any files. I don't even know which one (or more) I am supposed to chaneg yet. What do U want the whole /etc/asterisk/*.conf ??? :( |
| 20:54.14 | [TK]D-Fender | SomethingISOdd: Pastebin <- |
| 20:54.25 | BBHoss | vk4akp, pastebin sip.conf |
| 20:54.31 | SomethingISOdd | [TK]D-Fender ok second did you want me to paste bin the set varable and the get varable? |
| 20:54.46 | vk4akp | Amazing. :( |
| 20:54.48 | [TK]D-Fender | vk4akp: I asked you to show us what you've done so far in terms of configuring your system. This would involve extensions.conf and sip.conf. |
| 20:55.38 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 20:55.38 | *** mode/#asterisk [+o anthm] by ChanServ |
| 20:56.21 | [hC] | So, any of you guys done any load testing with SIPp? |
| 20:56.43 | *** join/#asterisk granviller (n=granvill@72x18x39x109.static.newroadstelecom.net) |
| 20:56.50 | clickonce | SomethingISOdd: Care to give me a URL? I've searched asteriskguru and the other sites without finding any information about how to send a CID to the client on an incoming call. |
| 20:57.42 | SomethingISOdd | <[TK]D-Fender> clickonce: "core show function CALLERID" |
| 20:57.43 | clickonce | Ah, could it be as simple as Set(CALLERID... let me try |
| 20:57.53 | clickonce | Ah, that I missed! :) Thanks! |
| 20:57.54 | stansmith | lmadsen: your literature is literal, right? |
| 20:58.01 | SomethingISOdd | clickonce anytime |
| 20:58.41 | zobia | hi guys. i got chan_sip.c:14106 handle_request_invite: Unable to create/find SIP channel for this INVITE |
| 20:58.50 | lmadsen | stansmith: I make no guarantees to the literal interpretation of the literature in question.... literally. |
| 20:59.04 | zobia | anyone knows what can do with that error? |
| 20:59.10 | stansmith | lmadsen: touchette |
| 20:59.25 | supjigator | Anyone know how to get zaptel to do 56k d channels? |
| 20:59.27 | SomethingISOdd | [TK]D-Fender http://www.pastebin.ca/911645 lines 38,39,40 is were i set the variable just a moment while i put up where i try to get the variable |
| 21:00.04 | tzafrir | supjigator, 56k d channel? what for? |
| 21:00.16 | supjigator | ss7 link |
| 21:00.41 | supjigator | It appears it defaults to 64k and I need 56k |
| 21:01.15 | tzafrir | supjigator, I'm probably ignorant. But so far I didn't see any documentation regarding changes at the zaptel level |
| 21:01.21 | SomethingISOdd | [TK]D-Fender here is were i try to get the variable http://www.pastebin.ca/911649 two different scripts. its lines 20-30 |
| 21:01.33 | tzafrir | What exact ss7 channel do you use? |
| 21:01.45 | supjigator | I have it on 24 on span 1 |
| 21:01.56 | [TK]D-Fender | SomethingISOdd: enable AGI debug, and show a call attempt so we can see it in action |
| 21:02.09 | SomethingISOdd | ok second dialing now |
| 21:02.30 | cornelinux | Hi, how can I change the callerid for outgoing calls on chan_misdn? |
| 21:06.24 | SomethingISOdd | [TK]D-Fender http://www.pastebin.ca/911656 as you will see it shows the variables its just not out putting them |
| 21:11.23 | SomethingISOdd | any other information you need [TK]D-Fender? |
| 21:12.07 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 21:12.34 | Katty | anyone familiar with postgres? |
| 21:12.56 | SomethingISOdd | sorry Katty not I |
| 21:13.04 | seanbright | SomethingISOdd: this line: $agi->exec_set("TIMEOUT(absolute)='60'"); |
| 21:13.08 | seanbright | its wrong |
| 21:13.15 | seanbright | there is no exec_set that i am aware of |
| 21:13.19 | *** join/#asterisk jbigbee (n=jbigbee@216.207.245.1) |
| 21:13.24 | stansmith | Katty: somewhat, what are you trying to do |
| 21:13.26 | x86 | is it possible to MixMonitor right to an mp3 file? |
| 21:13.35 | seanbright | $agi->set_variable("TIMEOUT(absolute)", "60"); |
| 21:13.38 | SomethingISOdd | seanbright i have the time out working thats not the problem. the problem is getting the variables over to the next script |
| 21:14.08 | seanbright | SomethingISOdd: line 77 of this $agi->exec_set("TIMEOUT(absolute)='60'"); |
| 21:14.10 | seanbright | err |
| 21:14.18 | seanbright | SomethingISOdd: line 77 of http://www.pastebin.ca/911645 |
| 21:14.22 | Katty | stansmith: i think there's something wrong with my ph_hba.conf |
| 21:14.25 | seanbright | that will cause your script to fail |
| 21:14.27 | Katty | stansmith: where i can login as my user. |
| 21:14.33 | Katty | stansmith: know anything about that? |
| 21:14.35 | vk4akp | Just for your info, I got the answer from someone that actually wanted to help. |
| 21:14.37 | Katty | stansmith: erm, can't. |
| 21:14.51 | stansmith | 1 sec |
| 21:15.02 | webar7 | gaak noiw the SIP signal is getting through but no audio signal .. .sigh |
| 21:15.09 | webar7 | so my phone rings |
| 21:15.25 | vk4akp | The answer was , I don't need to put an invented SIP proxy server name in the configs anywhere. I just point the ATA at the internal IP address. |
| 21:15.31 | SomethingISOdd | seanbright on sorry i should have set i am not accessing that part right now. i have my caller id set i figure i will get the first part working as i need and then finish off the second part |
| 21:15.40 | SomethingISOdd | set=send |
| 21:15.48 | webar7 | but when I pick up asterisk doesn't notice and doesn't set up the media channel |
| 21:16.07 | webar7 | plus the phone's DTMF doesn't work ... sigh |
| 21:16.17 | webar7 | why why why ?! :( |
| 21:16.19 | webar7 | :) |
| 21:16.34 | [TK]D-Fender | SomethingISOdd: the Get's seemed to work... |
| 21:16.36 | seanbright | SomethingISOdd: http://www.pastebin.ca/911649 is the script you are having problems with? |
| 21:16.47 | stansmith | Katty: no im sorry, im not familiar with that |
| 21:16.53 | SomethingISOdd | [TK]D-Fender but not into the script itself. |
| 21:17.02 | stansmith | pgadmin3 might be able to help you though |
| 21:17.13 | Katty | stansmith: k |
| 21:17.17 | x86 | [TK]D-Fender: do you know if it's possible to MixMonitor right to an mp3? |
| 21:17.19 | SomethingISOdd | seanbright no that was the first part the set_variable second let me get that link for the get_variable |
| 21:17.30 | [TK]D-Fender | vk4akp: Your diescription was not very meaningful and you showed us nothing. Not much for us to go on with there.... |
| 21:17.38 | x86 | sucks having to convert it from wav to mp3 each time, and I certainly dont want to put a bunch of system calls in my dialplan |
| 21:17.41 | [TK]D-Fender | x86: No, * cannot ENCODE to MP3 |
| 21:17.46 | x86 | weak |
| 21:18.22 | SomethingISOdd | seanbright http://www.pastebin.ca/911649 |
| 21:18.34 | [TK]D-Fender | SomethingISOdd: 1139-1140 looks fine to me. |
| 21:18.41 | seanbright | SomethingISOdd: yes, that is the exact same link i just sent you |
| 21:18.42 | x86 | [TK]D-Fender: do you know if the ${TIMESTAMP} channel variable follows from the original context to a called Macro? |
| 21:18.46 | *** join/#asterisk Tigerplug (n=chatzill@89.100.141.160) |
| 21:18.50 | SomethingISOdd | oh sorry seanbright |
| 21:19.07 | Corydon76-dig | x86: TIMESTAMP is a builtin, not a channel variable |
| 21:19.13 | [TK]D-Fender | x86: Deprecated var, and it (was) global and read-only |
| 21:19.21 | x86 | Corydon76-dig: seems empty when I try to use it |
| 21:19.28 | Tigerplug | Having problems with my Asterisk Server. I just build it on CentOS 5. --> Details of problem and config files here: http://boards.ie/vbulletin/showthread.php?p=55192201#post55192201 ---> anyone care to take a look? |
| 21:19.28 | SomethingISOdd | [TK]D-Fender it should work according to everything i have read but when i do an echo to the channel on the different variables they are not set |
| 21:19.29 | Corydon76-dig | in 1.6? |
| 21:19.30 | [TK]D-Fender | x86: see "deprecated" |
| 21:19.31 | x86 | [TK]D-Fender: dont want to write to it by any means... |
| 21:19.46 | x86 | [TK]D-Fender: well i'm trying to use a unique value for my recorded MixMonitor files |
| 21:19.57 | [TK]D-Fender | x86: then use one that WORKS :p |
| 21:20.00 | x86 | [TK]D-Fender: and it seems ${TIMESTAMP} is empty, so I was looking for options ;) |
| 21:20.06 | x86 | [TK]D-Fender: recommendations? :) |
| 21:20.14 | [TK]D-Fender | x86: "core show functions" |
| 21:20.14 | Tigerplug | any help? |
| 21:20.15 | x86 | date/time stamp would be ideal |
| 21:20.16 | Corydon76-dig | Already removed in 1.4. See STRFTIME() |
| 21:20.24 | x86 | ah cool |
| 21:20.37 | seanbright | SomethingISOdd: stop doing an "echo" and do an $agi->verbose($my_variable_here); |
| 21:20.39 | x86 | can I use *'s strftime just like I use C's strftime? |
| 21:20.52 | Corydon76-dig | x86: similar, yes |
| 21:20.53 | SomethingISOdd | seanbright ok let me try that |
| 21:21.04 | Tigerplug | my xlite phones wont register with new asterisk server :( |
| 21:21.07 | x86 | Corydon76-dig: that's just plain genious, imho :) |
| 21:21.16 | [TK]D-Fender | SomethingISOdd: everything you showed me seems fine |
| 21:21.29 | seanbright | SomethingISOdd: try it and post a new AGI debug output to a pastebin |
| 21:21.36 | *** join/#asterisk Datax (n=Datax@glou.nurvnet.org) |
| 21:21.37 | Corydon76-dig | "genius" |
| 21:21.45 | Tigerplug | http://boards.ie/vbulletin/showthread.php?p=55192201#post55192201 <--- can someone look at this |
| 21:21.47 | seanbright | "irony" |
| 21:21.54 | Tigerplug | it would be appreciated very much! |
| 21:21.54 | SomethingISOdd | seanbright ok |
| 21:22.08 | x86 | Corydon76-dig: that was the joke ;) |
| 21:22.25 | x86 | Corydon76-dig: just trying to keep you on your toes :P |
| 21:22.30 | Corydon76-dig | x86: I hate being called a genius, anyway |
| 21:22.40 | Tigerplug | lol |
| 21:22.44 | stansmith | lol |
| 21:23.04 | x86 | hehehe |
| 21:23.14 | stansmith | u broke protocol with that "hehehe" |
| 21:23.15 | x86 | Corydon76-dig: tis why i called you a genious instead ;) |
| 21:23.25 | Corydon76-dig | ANYway |
| 21:23.44 | x86 | ah cool, ${EPOCH} would work even ;) |
| 21:24.00 | Tigerplug | guys - Im fairly new to asterisk but I think I have my config files right (mainly sip.conf and extensions.conf) but I can't register my xlite (Im using for testing) with the server... I get a "408 request timeout" :( |
| 21:24.24 | [TK]D-Fender | Tigerplug: Where is your soft-phone located relative to you server? |
| 21:24.33 | Tigerplug | same subnet |
| 21:24.54 | SomethingISOdd | seanbright http://www.pastebin.ca/911687 |
| 21:25.27 | Tigerplug | [TK]D-Fender ---> its on the same subnet, its just a home network |
| 21:25.29 | [TK]D-Fender | Tigerplug: check your firewall on the server |
| 21:25.37 | [TK]D-Fender | Tigerplug: Betting its blocking it |
| 21:25.45 | Tigerplug | will check now |
| 21:26.08 | SomethingISOdd | Tigerplug and if you are using centos or fedora try disabling selinux |
| 21:26.15 | seanbright | SomethingISOdd: post the latest version of your script |
| 21:26.23 | SomethingISOdd | ok second |
| 21:26.27 | Tigerplug | SomethingISOdd --> will try that |
| 21:27.10 | *** part/#asterisk granviller (n=granvill@72x18x39x109.static.newroadstelecom.net) |
| 21:27.10 | Anthony76 | hello, what does it mean: chan_sip.c:10214 handle_response: Failed to authenticate on BYE to |
| 21:27.32 | Tigerplug | brb |
| 21:28.51 | [TK]D-Fender | Tigerplug: oki, checking out, BBL |
| 21:28.57 | SomethingISOdd | seanbright http://www.pastebin.ca/911692 |
| 21:29.04 | JenniferAkemi- | sweet. my g729 sounds WAY better to ulaw than the g726 did. actually totally useable. (although only with the ata. the grandstream sucks SO bad.) |
| 21:29.37 | SomethingISOdd | JenniferAkemi- GS has sucked since i can remember |
| 21:30.05 | JenniferAkemi- | SomethingISOdd: yeah, unfortunately my boss ordered a bunch of hardware then told me to learn about voip after it all got delivered. |
| 21:30.15 | JenniferAkemi- | now i know he should have been buying polycom :) |
| 21:30.43 | SomethingISOdd | JenniferAkemi- i have never used polycom so i cant comment, my boss ordered some no name phones out of japan. that are great phones |
| 21:30.52 | SomethingISOdd | that==they |
| 21:31.10 | seanbright | SomethingISOdd: give this a try http://www.pastebin.ca/911698 |
| 21:31.16 | SomethingISOdd | seanbright ok |
| 21:31.33 | seanbright | SomethingISOdd: notice the change from $g['g'] to $g['data'] |
| 21:31.34 | *** part/#asterisk simbol76ss (n=simbol@87.3.209.65) |
| 21:31.39 | seanbright | etc |
| 21:31.47 | Tigerplug | [TK]D-Fender ---> I've just googled the firewall issue in CentOS with Asterisk. Im pretty much a newbie with CentOS. I can't find any info relevant (or dont know what is relevant). Does anyone know if it uses IPtables? |
| 21:31.48 | seanbright | SomethingISOdd: and not to be rude, but you should probably learn PHP |
| 21:31.59 | SomethingISOdd | i tried $g['data'] in the first place.. |
| 21:32.43 | seanbright | SomethingISOdd: well the documentation here: http://phpagi.sourceforge.net/phpagi2/docs/ says that there is a 'result' and a 'data' element in each array returned by get_variable |
| 21:33.09 | SomethingISOdd | agreed thats the first way i tried it.. |
| 21:33.21 | seanbright | SomethingISOdd: well give it another whirl |
| 21:33.34 | SomethingISOdd | trying now |
| 21:35.03 | SomethingISOdd | what what the heck thank seanbright it worked for some reason this time |
| 21:35.16 | seanbright | SomethingISOdd: i'm a genious. |
| 21:35.34 | Tigerplug | guys , what ports do I need to open in IPtables for asterisk? |
| 21:35.35 | SomethingISOdd | but that ticks me off because i swear i tried that lastnight.. |
| 21:35.43 | *** join/#asterisk DarkRift (n=dark@bas4-montrealak-1128595447.dsl.bell.ca) |
| 21:35.59 | vk4akp | D-Fender, well I cut and pasted the same questions elsewhere and got an answer straight away. |
| 21:36.12 | vk4akp | NP, It is solved now |
| 21:46.46 | Tigerplug | D-Fender ---> Problem solved with the firewall (I have temporarily disabled it to test) phones can now register. However: I have 2 extensions 2000 and 2001 here is my sip.conf :http://pastie.caboo.se/155023 and here is my extensions.conf: http://pastie.caboo.se/155025 |
| 21:47.28 | Tigerplug | when I call 2000 from 2001 I get a "the person you are calling is unavailable" :( |
| 21:50.00 | Datax | Hi again, how can I use music on hold when Asterisk can't find /dev/dsp because I don't have a sound card in the machine ? |
| 21:50.17 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 21:50.22 | Datax | the error message i'm getting is : [Feb 20 22:34:07] WARNING[18821]: chan_oss.c:682 setformat: Unable to re-open DSP device /dev/dsp: No such file or directorynable to re-open DSP device /dev/dsp: |
| 21:50.34 | stansmith | Datax: are you, by any chance, using a HP ML350 server? |
| 21:50.58 | *** join/#asterisk kombi (n=kombi@port-213-160-14-18.static.km-it.de) |
| 21:51.01 | Datax | no i'm running asterisk in a vmware, I could put a virtual sound card on it but from what I have read I don't need to |
| 21:51.22 | Datax | I want to avoid adding "bits" to it |
| 21:51.24 | x86 | Datax: i dont have a sound card and I use MoH just fine |
| 21:51.28 | kombi | what kind of hardware do you need for ss7? |
| 21:51.45 | x86 | kombi: are you a CLEC, LEC, or ILEC? |
| 21:51.57 | Datax | x86: what distribution did you install Asterisk on ? |
| 21:52.02 | Datax | I'm running a debian etch |
| 21:52.08 | x86 | Datax: Gentoo, but that doesn't matter one bit |
| 21:52.29 | kombi | x86: I'm afraid I don't know what those three acronyms mean.. |
| 21:52.39 | Datax | did you install alsa and install some kind of virtual sound card ? |
| 21:52.41 | x86 | kombi: then you'll never have to worry about SS7 ;) |
| 21:53.05 | stansmith | Datax: my understanding is that you need no sound card for asterisk |
| 21:53.11 | kombi | x86: actually I do, I have to connect a box via ss7 soon |
| 21:53.29 | kombi | so mayby I am a clec, lec or ilec..;) |
| 21:53.42 | Datax | stansmith: yeah that's what I've understood too yet when I put a call on hold I get an error message in the asterisk console and no music ;) |
| 21:54.15 | Anthony76 | hello, what does it mean: chan_sip.c:10214 handle_response: Failed to authenticate on BYE ??? thanks |
| 21:55.14 | Datax | x86: any idea what my problem could be ? I don't have a /dev/dsp file or /dev/audio |
| 21:55.33 | Datax | and that is obviously what Asterisk is looking for |
| 21:56.22 | *** join/#asterisk worgil (n=worgil@88.252.190.40) |
| 21:56.39 | Shaun2222 | anybody know how i can acheive waht i'm trying to do? http://lists.digium.com/pipermail/asterisk-users/2008-February/206103.html |
| 21:57.18 | kombi | x86: what kind of card does one? |
| 21:57.25 | kombi | ..use |
| 21:57.30 | kombi | (sorry) |
| 21:59.07 | worgil | can i use instant message with asterisk ? |
| 21:59.22 | Nugget | all depends on what you mean by "use" and "with" I suspect. :) |
| 21:59.47 | Tigerplug | guys im using a VOIP provider called blueface ... but I cant make out going / incoming calls. Can anyone tell me what config file this is configured in |
| 22:00.04 | kombi | just looked it up, I am a CLEC! (competitve local exchange carrier), wow! x86, you can talk to me now..;) |
| 22:00.37 | denon | you didn't know you were a clec? |
| 22:00.54 | kombi | denon: no, just found out! |
| 22:01.14 | denon | you didn't get it from ShadyDiplomas.com did you? |
| 22:01.24 | denon | $19.95 and they made you a clec |
| 22:01.33 | kombi | denon: actually, we are looking into the possibility,.. oh really? |
| 22:02.04 | denon | kidding .. :) |
| 22:03.16 | kombi | denon: thing is: we route calls over voip now, but client requires more features that only being a clec allows. Our carrier offered a ss7 to us and I am researching the hardware needs |
| 22:03.16 | worgil | can anyone help me about instant message with asterisk ? |
| 22:03.57 | kombi | worgil: sorry, never done it, just good old voice.. |
| 22:04.32 | kombi | denon: do have same insight of the matter? |
| 22:04.34 | worgil | kombi: thanks |
| 22:04.49 | kombi | s/do/do you/ |
| 22:04.54 | kombi | oops... |
| 22:05.07 | kombi | jbot does regex.. |
| 22:05.25 | denon | my insight would be ... odds are you don't need to do it for whatever it is you think you do . . |
| 22:05.36 | denon | and it's a lot more involved than you think |
| 22:05.53 | kombi | denon: how do you know the former / the latter? |
| 22:06.01 | kombi | ;) |
| 22:06.10 | denon | that doesn't really matter :) |
| 22:06.48 | kombi | denon: do you do not, right? (all three things, that is) |
| 22:07.02 | denon | English, please.. |
| 22:07.16 | kombi | s/do you/so you/ |
| 22:07.27 | kombi | thanks, jbot.. |
| 22:07.34 | denon | I do not what? |
| 22:07.51 | denon | this conversation is taking far too long |
| 22:08.04 | denon | what I'm saying is, if you have a single client that needs some funky signalling .. |
| 22:08.10 | denon | and you think you need to become a clec to meet that one need |
| 22:08.21 | denon | you will put far too much effort into it |
| 22:09.33 | kombi | denon: well, I understand you, but that client might be big and it might be important funky signalling. But to the point, do have experience in the field or are you just trying to talk me out of it based on presumption? |
| 22:09.34 | denon | I've come to this conclusion after helping a few companies overcome some provider hurdles, where they thought they needed to be a clec to turn on the lights in the noc |
| 22:09.53 | kombi | ok.. |
| 22:10.05 | denon | "oh noes, the light bulb went out .. where's the clec forms?!" |
| 22:10.13 | kombi | denon: lol.. |
| 22:10.27 | denon | you think I'm kidding .. |
| 22:10.28 | *** join/#asterisk MaliutaWrk (i=nikolai@119.11.100.249) |
| 22:10.29 | denon | :) |
| 22:10.45 | denon | I am, of course .. but yeah, it's true |
| 22:11.09 | kombi | denon: I like that..,) This client here is actually a tv station having to fence of anonymous callers |
| 22:11.38 | denon | fence of? |
| 22:11.52 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 22:11.56 | denon | if you mean fend off.. |
| 22:12.03 | denon | why can't you do that with standard PRI signalling? |
| 22:12.05 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 22:12.06 | kombi | denon: I mean, identify them when they do nasty calls repeatedly |
| 22:12.14 | denon | oh, pull ANI then |
| 22:12.34 | kombi | denon: how do you mean? |
| 22:12.42 | denon | are you familiar with ANI? |
| 22:12.52 | kombi | denon: maybe not.. |
| 22:13.09 | kombi | denon: what does it do? |
| 22:13.21 | denon | perhaps it's time to curl up on the couch with a couple'a google searches |
| 22:13.30 | denon | let me start you in the right direction |
| 22:13.52 | kombi | denon: please do, but bear in mind the carrier does not forward the cid |
| 22:13.56 | denon | http://www.voip-info.org/wiki-ANI |
| 22:14.00 | denon | ANI != CID |
| 22:14.09 | kombi | hmm.. let me check.. |
| 22:14.10 | JerJer | CLEC baby - woot woot! |
| 22:14.26 | kombi | JerJer...;) |
| 22:14.35 | kombi | (wootwoot) |
| 22:15.27 | variable_office | what do you all keep track of call quality with for your users? |
| 22:15.37 | denon | if your phone doesn't ring .. life is good |
| 22:15.43 | denon | if there's a problem, they yell |
| 22:15.44 | denon | :) |
| 22:16.11 | variable_office | humm, i think i need something a little more scientific |
| 22:16.21 | denon | hehe |
| 22:16.33 | kombi | denon: hmm, is ANI allways available? |
| 22:16.36 | *** join/#asterisk Chumbaa (n=chatzill@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 22:16.49 | denon | kombi: well, I don't know what your carrier is doing to your calls .. |
| 22:17.27 | denon | but if you're getting real pri signalling, jacked into a dms-100 or something . no weird configs .. |
| 22:17.29 | denon | probably |
| 22:17.39 | kombi | denon: groovy! |
| 22:17.40 | denon | if not, call your telco, and ask em why not |
| 22:18.49 | denon | *now* I'm groovy, a few minutes ago I was incompetent because I wanted to hook up a callerID box instead of buying 51% of SBC |
| 22:19.13 | kombi | denon: I didn't say that!..;) |
| 22:19.21 | denon | suuure |
| 22:19.23 | denon | hehe |
| 22:19.32 | denon | anyway, yeah, that's the scoop. |
| 22:20.22 | kombi | denon: sounds good to me.. so is there even a standard variable in the dialplan that I didn't know of? |
| 22:20.30 | kombi | $ANI? |
| 22:21.50 | *** join/#asterisk AndyGraybeal (n=andy@node248.34.251.72.1dial.com) |
| 22:22.13 | *** join/#asterisk SteveTotaro (n=Elizabet@pool-71-166-102-100.bltmmd.east.verizon.net) |
| 22:23.25 | SteveTotaro | WayHigh: any luck with the tdm400p? |
| 22:23.41 | denon | kombi: you'll want to look at ${CALLERID(ani)} |
| 22:24.07 | kombi | denon: I'm psyched here! thanks! |
| 22:24.13 | denon | sure |
| 22:24.17 | denon | here's my paypal address |
| 22:24.27 | kombi | sorry.. gotta go...;) |
| 22:24.45 | *** join/#asterisk nvrpunk (n=root@81.90.21.227) |
| 22:24.50 | kombi | actually, off to the pub to sink some pints now |
| 22:25.07 | denon | btw, look at ANI2 as well |
| 22:25.09 | denon | or not |
| 22:27.26 | SteveTotaro | ANI2 is kewl |
| 22:27.32 | *** join/#asterisk Docfxit (n=none@cpe-72-129-118-135.socal.res.rr.com) |
| 22:27.50 | denon | but he'll never know |
| 22:27.56 | denon | that's what he gets for mocking my paypal comment! |
| 22:28.39 | SteveTotaro | i was lucky in inherit a few PRIs with ANI and ANI2 |
| 22:29.51 | SteveTotaro | then we swtiched to another carrier (GXing) and I had to edumicate the sales reps that callerid and ani are different beasts and they were absolutely cluess about ANI2 |
| 22:30.19 | SteveTotaro | that went on for months |
| 22:32.14 | nvrpunk | Hello |
| 22:32.29 | nvrpunk | anyone have a quick minute to help me figure out some contexts? |
| 22:32.37 | nvrpunk | I am having a problem setting up DIDs |
| 22:32.40 | SteveTotaro | i have a long minute |
| 22:32.50 | nvrpunk | http://www.pastebin.ca/911777 |
| 22:32.51 | nvrpunk | http://www.pastebin.ca/911782 |
| 22:33.02 | nvrpunk | there's my extension.conf and iax.conf |
| 22:33.16 | nvrpunk | what I am wondering is how to make it so that I have multiple accounts on junction networks |
| 22:33.22 | nvrpunk | with DIDs on each account |
| 22:33.29 | nvrpunk | all across the same box |
| 22:34.13 | SteveTotaro | can't you just add DIDs under the same account |
| 22:34.13 | SteveTotaro | ? |
| 22:34.19 | nvrpunk | when i try just making the extension NXXNXXXXXX it won't route |
| 22:34.23 | nvrpunk | SteveTotaro, I wish |
| 22:34.34 | nvrpunk | SteveTotaro, in practical application, yes |
| 22:34.54 | Tigerplug | guys --> I have a username and login info for a demo account with a VOIP provider. Their website mentions how to use it with Asterisk. Can anyone help me out at all? I have SSH on public IP ---> if someone could create a sample config for me I would appreciate it. At least then I could play about with it and learn more. Any takers? |
| 22:34.56 | nvrpunk | but the deed has been done, now im trying to work around the inconvenience |
| 22:35.10 | SteveTotaro | what does the asterisk cli show with verbose turned up? |
| 22:35.26 | nvrpunk | Tigerplug, have you looked at http://www.asteriskguru.com ? |
| 22:35.33 | nvrpunk | theres example configs there |
| 22:35.52 | nvrpunk | SteveTotaro, well, it looks as though the calls are just routing internally |
| 22:35.54 | nvrpunk | sec |
| 22:35.58 | nvrpunk | lemme generate output |
| 22:37.05 | Tigerplug | nvrpunk - thanks for that but I have attempted this several times using resources from there and "clear instructions" from my VOIP provider here:http://forums.blueface.ie/showthread.php?t=339 |
| 22:37.10 | nvrpunk | -- Executing [14237090274@pstn-access:1] Answer("SIP/sip2-081e9f20", "") in new stack |
| 22:37.18 | nvrpunk | -- Executing [14237090274@pstn-access:2] Hangup("SIP/sip2-081e9f20", "") in new stack |
| 22:37.34 | nvrpunk | then non zero exite accross ptsn-access |
| 22:37.50 | Tigerplug | NVRpunk I can give you ssh access if you think you can help. Its just for learning --> its not a production box |
| 22:37.52 | SteveTotaro | so this is an inbound or outbound call? |
| 22:37.53 | nvrpunk | Tigerplug, who is the provider, ill help |
| 22:38.13 | nvrpunk | SteveTotaro, both, i want the call to route out external then come back into our box |
| 22:38.15 | nvrpunk | for testing |
| 22:42.54 | Tigerplug | nvrpunk ---> I sent you private |
| 22:42.57 | Tigerplug | message |
| 22:45.15 | Tigerplug | nvrpunk ---> you there? |
| 22:45.17 | tzanger | oh nice |
| 22:45.24 | tzanger | I just found rogers' TAP gateways |
| 22:47.32 | [hC] | do tell |
| 22:50.31 | RypPn | I'm looking at zaptel-1.4.9 atmo, and i'm wondering how I set the echo canceller, zconfig.h seems to have gone. Anyone got any ideas? |
| 22:50.43 | tzanger | http://www.txt.ca/otherlinks.htm#aggregator |
| 22:50.44 | tzanger | aww yeah |
| 22:50.49 | tzanger | [hC]: me? |
| 22:52.00 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 22:52.21 | [hC] | tzanger: yes you! |
| 22:52.28 | *** join/#asterisk puzzled (n=patrick@53533DDB.cable.casema.nl) |
| 22:52.31 | tzanger | where the hell did I get that link from |
| 22:52.50 | tzanger | http://www.downloads.rogers.com/wireless/Network_Specifications_GPRS_EDGE.pdf |
| 22:52.53 | tzanger | TONS of info on that |
| 22:53.05 | tzanger | APNs, login credentials, technical detaisl of their gateways, etc |
| 22:53.11 | [hC] | holy crap |
| 22:54.41 | [hC] | something tells me thats not ideal to be public knowledge |
| 22:54.46 | tzanger | why not? |
| 22:54.54 | tzanger | it's meant for mobile application developers |
| 22:54.58 | tzanger | perfectly legit |
| 22:55.26 | [hC] | oh, devs ... that makes more sense |
| 22:57.36 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 23:00.29 | *** join/#asterisk webar7 (n=webart@CPE0080c8f208a5-CM001371173cf8.cpe.net.cable.rogers.com) |
| 23:00.41 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 23:01.18 | *** join/#asterisk yenno (n=yenno@92.104.83.122) |
| 23:01.22 | *** join/#asterisk RangerElf (n=gustavo@CableLink222-147.telefonia.InterCable.net) |
| 23:01.28 | *** join/#asterisk webar7 (n=webart@CPE0080c8f208a5-CM001371173cf8.cpe.net.cable.rogers.com) |
| 23:01.43 | grandpapadot | When doing a show channel on an outbound sip call, there's a "native format" codec and a "write/read format" codec setting. If "native format" is gsm (2) and write/read is ulaw (64), what's my effective codec for bandwidth purposes? Is write/read transcoded to native? |
| 23:01.52 | yenno | hi, is it easy to brute force a sip password when having a trace? |
| 23:01.54 | jameswf | we got one of those 199 walmart pc's put a dual span t1 and looped it the system crapped out in 30 seconds cooked the hard drive, I would have expected to toast the processor first |
| 23:02.30 | RangerElf | Howdy people, I've a question. Is it required, necessary, or even important to run a traffic-shaping script on a linux firewall, if it's the same server which runs the Asterisk? |
| 23:02.54 | RangerElf | (hmhph. That should be "required, important, or even necessary...") :P |
| 23:04.32 | tzanger | RangerElf: I do it, I find it helps |
| 23:04.41 | tzanger | it's not perfect but I haven't given it my all in trying to configure it yet |
| 23:04.48 | tzanger | www.mixdown.ca/~andrew/dump/rc.tc is what I use |
| 23:04.51 | tzanger | I think that's the link anyway |
| 23:05.13 | tzanger | jameswf: sounds coincidental to me |
| 23:05.30 | tzanger | I ran a quadspan T1 going back-to-back on a little P3 for testing |
| 23:05.53 | RangerElf | argh... gotta get a usable script... brb... |
| 23:06.11 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:06.11 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 23:06.14 | yenno | nobody knows? |
| 23:08.38 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:09.25 | *** join/#asterisk Tigerplug (n=chatzill@89.100.141.160) |
| 23:09.31 | Tigerplug | connection problems :( |
| 23:11.12 | jameswf | the $199 pc had a segate drive so actualy we expected it to crap.... |
| 23:12.06 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 23:12.30 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 23:15.00 | siya | yenno: all you need is the sip account logging in as the password is not encrypted in sip |
| 23:15.32 | siya | not sure how it's done once the registration has succeeded but I recon it's not majorly secure |
| 23:16.27 | grandpapadot | When doing a show channel on an outbound sip call, there's a "native format" codec and a "write/read format" codec setting. If "native format" is gsm (2) and write/read is ulaw (64), what's my effective codec for bandwidth purposes? Is write/read transcoded to native? |
| 23:18.06 | grandpapadot | Rather, how do I tell which codec is actually being used between two peers? |
| 23:18.46 | nvrpunk | core set verbose 9 |
| 23:18.49 | nvrpunk | in asterisk -r |
| 23:18.51 | nvrpunk | make a call |
| 23:18.52 | nvrpunk | watch |
| 23:19.08 | grandpapadot | asterisk 1.2.26 |
| 23:19.23 | nvrpunk | hmm |
| 23:19.25 | Shaun2222 | anybody know how i can acheive waht i'm trying to do? http://lists.digium.com/pipermail/asterisk-users/2008-February/206103.html |
| 23:19.36 | nvrpunk | up the verbosity |
| 23:19.38 | nvrpunk | and watch |
| 23:19.47 | nvrpunk | should be a similar comman -core |
| 23:20.15 | mvanbaak | in 1.2 it's: set verbose ( |
| 23:20.19 | mvanbaak | in 1.2 it's: set verbose (9 |
| 23:20.22 | mvanbaak | ah wel |
| 23:20.26 | grandpapadot | nvrpunk: Is that to me? The codec isn't displayed when a call comes in. |
| 23:20.26 | mvanbaak | you get the idea |
| 23:20.39 | nvrpunk | grandpapadot, yeah it was |
| 23:20.50 | grandpapadot | Is GSM a good compromise when ulaw is just to much and nothing else is available? |
| 23:21.04 | nvrpunk | what type of phone is it |
| 23:21.07 | nvrpunk | thats being called |
| 23:21.10 | mvanbaak | grandpapadot: yes it is |
| 23:21.12 | grandpapadot | This is asterisk to asterisk |
| 23:21.27 | grandpapadot | I'm just trying to reduce bandwidth which is going to reduce latency and get some of my delay out of calls. |
| 23:21.28 | nvrpunk | use g729 |
| 23:21.41 | nvrpunk | g729 with trunking |
| 23:21.44 | grandpapadot | Our phones use g729 |
| 23:21.44 | nvrpunk | lower the pps |
| 23:21.48 | nvrpunk | and overhead |
| 23:22.03 | nvrpunk | then asterisk can go to asterisk g729 |
| 23:22.07 | nvrpunk | in passthrough |
| 23:22.10 | nvrpunk | no licensing |
| 23:22.15 | grandpapadot | It's asterisk-to-asterisk. I can use g729, but I was hoping GSM would be a bit better qualtiy, i.e., MoH and such woudln't sound so bad. |
| 23:22.35 | mvanbaak | grandpapadot: g729 is better audio quality then gsm |
| 23:22.40 | nvrpunk | yeah MoH suck on g729 |
| 23:22.52 | nvrpunk | mvanbaak, than not then |
| 23:23.07 | mvanbaak | convert your moh to g729 |
| 23:23.17 | grandpapadot | mvanbaak: I realize its definitely better speech quality, but what about MoH and such. |
| 23:23.30 | *** join/#asterisk RangerElf (n=chatzill@cablelink222-147.telefonia.intercable.net) |
| 23:23.45 | RangerElf | OK, back. |
| 23:23.58 | *** join/#asterisk kombi (n=kombi@port-213-160-14-18.static.km-it.de) |
| 23:32.11 | grandpapadot | Given the choice between GSM and g729a, which has better call quality across the board, including music on hold? |
| 23:32.34 | nvrpunk | g729 |
| 23:35.00 | Tigerplug | nvrpunk ---> I lost my connection |
| 23:35.34 | mvanbaak | G729 |
| 23:36.03 | grandpapadot | Does vonage use g729 across the board? |
| 23:36.05 | *** join/#asterisk benburkert (n=ben@209.189.228.58) |
| 23:36.23 | nvrpunk | grandpapadot, prolly not |
| 23:36.24 | mvanbaak | no idea |
| 23:36.25 | nvrpunk | but for most |
| 23:36.27 | nvrpunk | yes |
| 23:36.30 | nvrpunk | or g723 |
| 23:36.34 | nvrpunk | no one can really know |
| 23:36.38 | nvrpunk | for certain |
| 23:37.22 | hmodes | vonage defaults to 711 |
| 23:37.54 | *** join/#asterisk fanx (n=fanx@1807ds2-soeb.0.fullrate.dk) |
| 23:38.19 | yenno | ok thanks siya :) |
| 23:38.57 | fanx | g'evening :) |
| 23:39.02 | hmodes | i think packet8 uses 723 don't they? |
| 23:39.41 | Tigerplug | nvrpunk --> I fixed that. Just give me a shout when your free. Thanks! |
| 23:40.12 | *** join/#asterisk rjsjr (n=rjs@dsl081-071-002.sfo1.dsl.speakeasy.net) |
| 23:40.17 | fanx | I was wondering, do you think it is possible to use asterisk for at replacement for teamspeak/ventrilo? :) |
| 23:41.07 | *** join/#asterisk AndyGraybeal (n=andy@node248.34.251.72.1dial.com) |
| 23:41.10 | hmodes | the hardest part of an asterisk ventrilo-work-alike is the client, imo |
| 23:41.43 | fanx | hmodes: to get a client that "works-like-used-to" ? |
| 23:41.52 | hmodes | yupper |
| 23:41.55 | Tigerplug | hmodes ---> tell me more! or if you have a link thats great. I need an alternative to vent |
| 23:42.01 | hmodes | there's also the issue of selective mute |
| 23:43.28 | fanx | hmodes: any reason why I couldn't throw 8-10 friends in a conference call (I haven't tried setting up asterisk before, please forgive me if I am using the wrong terms)? |
| 23:43.56 | hmodes | not really |
| 23:44.03 | fanx | hmodes: thats fantastic :) |
| 23:44.11 | hmodes | they'd just need push to talk capable clients, in the simplest form |
| 23:44.30 | hmodes | but there are a lot of features in vent/teamspeak that are not addressed natively in asterisk, afaik |
| 23:44.58 | fanx | hmodes: but they don't use push to talk - that just .. ahrem .. chat - voice detection in some way I think. |
| 23:45.44 | fanx | hmodes: well - I don't think my friends care about anything but "clear speak" - so asterisk might be the way to go :) |
| 23:46.04 | hmodes | yeah, if you just want a giant partyline, meetme does that just fine with any sip client ;p |
| 23:46.38 | fanx | isn't meetme some asterisk plugin thingy? |
| 23:47.53 | fanx | but anyways - it sounds like the way to go (when vent/ts isn't an option) - so I will read up on it and give it a try. Thanks :) |
| 23:49.17 | *** join/#asterisk Peaceful (n=peaceful@70.102.57.178) |
| 23:49.36 | hmodes | there is something vageuly ironic about the fact that 90s style push to talk voip is now something that's considered 'desireable' ;p |
| 23:49.59 | Peaceful | Has anyone ran into a case where a Polycom phone (550 in my case) won't dial any 3-digit numbers? Like 911? It gives me "Enter more digits" |
| 23:50.00 | fanx | heh ;) |
| 23:52.07 | Peaceful | It doesn't seem to be the dialplan digitmap setting. It's already set to match 911 with "[2-9]11", and when I added "3xx" to try to match an extension 301, it gave me the same "Enter more digits" error on the 550 |
| 23:52.18 | Peaceful | It just won't send the request to asterisk. |
| 23:52.52 | Peaceful | 911 is sorta an important number... |
| 23:56.56 | *** join/#asterisk RoyK (n=roy@ip-202-17-149-91.dialup.ice.no) |
| 23:57.33 | *** part/#asterisk fanx (n=fanx@1807ds2-soeb.0.fullrate.dk) |
| 23:57.53 | *** join/#asterisk seanbright (n=seanbrig@c-69-251-175-43.hsd1.md.comcast.net) |
| 23:58.19 | grandpapadot | Peaceful: Are you provisioning through FTP/TFP/HTTP or just using the built-in web server to do it? |
| 23:59.46 | drmessano | "U.S. Shoots down ailing Satellite; France Surrenders" |
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