| 00:00.11 | SpaceBass | fuck this thing with SIP |
| 00:00.19 | SpaceBass | arrugg... its configured right and just will not register |
| 00:01.19 | *** join/#asterisk PTG1234 (PTG123@66.213.239.122) |
| 00:01.23 | AgiNamu | and ethereal says? |
| 00:01.44 | SpaceBass | haven't tried it yet... gotta set up my switch and all |
| 00:01.54 | SpaceBass | or posion the arp cache :) |
| 00:01.55 | AgiNamu | jsut run it on your asterisk machine |
| 00:02.01 | AgiNamu | tethereal |
| 00:02.02 | AgiNamu | or tcpdump |
| 00:02.18 | SpaceBass | dont have tetheral |
| 00:02.25 | AgiNamu | are you using STUN |
| 00:02.28 | SpaceBass | i can try tcpdump, but dont know it |
| 00:02.30 | AgiNamu | you might need a STUN server or something. |
| 00:02.38 | AgiNamu | tcpdump udp port 5060 should do it? |
| 00:02.41 | SpaceBass | not using stun that i know of |
| 00:02.51 | AgiNamu | or tcpdump udp host <your phone ip> |
| 00:03.10 | Supaplex | stun? |
| 00:03.23 | AgiNamu | simple traversal of udp thru nat |
| 00:03.25 | AgiNamu | or some shit like that. |
| 00:03.27 | Supaplex | ahh |
| 00:03.35 | AgiNamu | maybe it'll help. |
| 00:03.41 | AgiNamu | again, sip sucks |
| 00:03.52 | AgiNamu | ok, well, im out |
| 00:03.54 | Supaplex | correction. it is the suck. |
| 00:04.01 | sbarrius | later AgiNamu! |
| 00:04.04 | AgiNamu | teh sux0rs. |
| 00:04.14 | Supaplex | yeh :) |
| 00:04.21 | Los415 | hey guys anyone experience asterisk crashing randomly when using the asterisk-addons mainly the mysql addon |
| 00:04.22 | AgiNamu | g'luck SpaceBass. |
| 00:04.40 | Supaplex | Los415: define crash |
| 00:05.00 | SpaceBass | trying tcpdump now |
| 00:05.35 | *** join/#asterisk soundguy (~soundguy@zeus.blendtek.com.au) |
| 00:05.53 | SpaceBass | and called away to make dinner |
| 00:06.08 | Los415 | suppalex asterisk just stops |
| 00:06.22 | Los415 | there is no core dump |
| 00:06.27 | Los415 | nothing in the logs |
| 00:06.34 | Los415 | and it's random |
| 00:13.45 | sneak | hey - anyone know why my 7960 has a lower version than the file specified in OS79XX.TXT and it's not downloading the new one? |
| 00:14.09 | jesster | Im troubleshooting a phone hanging off a FXS port of a channel bank. The the phone is in use, and another call comes in, asterisk cli shows "Zap/106-busy-678259056 is busy" but the caller hears ring, then eventually goes dead. I've pasted my relevant zapata.conf here http://pastebin.ca/8080 |
| 00:14.09 | *** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net) |
| 00:14.18 | jontow | sneak; you also have to edit that in SIPmacaddr.cnf |
| 00:14.37 | jontow | oh, my bad |
| 00:14.41 | jontow | sneak; i meant SIPDefault.cnf |
| 00:14.56 | jontow | example: |
| 00:14.57 | jontow | # SIP Default Generic Configuration File |
| 00:14.57 | jontow | <PROTECTED> |
| 00:14.57 | jontow | # Image Version |
| 00:14.57 | jontow | image_version: P0S3-07-3-00 |
| 00:14.58 | jontow | ... |
| 00:15.14 | jontow | synchronize those two files, reboot the phone.. and you'll be upgrading :) |
| 00:15.15 | sneak | thanks |
| 00:15.28 | jontow | np :) |
| 00:15.35 | dmccollum | Good evening everyone. I just installed a X100P card in my asterisk box and when I try to dial-out using a xlite client I get the all cicuits are busy message from asterisk. Anyone know what might be wrong? |
| 00:16.30 | sneak | is there a non-pain-in-the-ass way of getting like 100 phones running 3.1 callmanager firmware upgraded to 7.x SIP firmware? |
| 00:16.35 | bkw_ | ya you failed to read all the docs? |
| 00:16.46 | bkw_ | can you provide us the dial line from your extensions.conf |
| 00:16.53 | bkw_ | otherwise we are just guessing why you can't dialout |
| 00:17.49 | BrianR___ | sneak: Yes. About a half dozen files on the tftp server :) |
| 00:18.00 | BrianR___ | sneak: but the phones will all upgrade unattended.. |
| 00:18.45 | sneak | ok |
| 00:19.35 | kram | dmccollum: is your x100p from digium? |
| 00:19.52 | kram | dmccollum: if so you can get free support with it of course, and i can make sure someone stays around to help |
| 00:20.12 | bkw_ | if he will provide the .conf stuffs ;) |
| 00:20.14 | dmccollum | No, I bought it from ebay. |
| 00:20.15 | bkw_ | or what the CLI says |
| 00:20.16 | sneak | ok yet another dumb question - i've put the image_version: line in my SIPDefault.cnf and it's also the same file basename as is in the OS79XX.TXT, and the phone, upon booting, tftps the new 5.3 SIP image down - but it doesn't say "upgrading software" and it doesn't reboot - it just says "phone unprovisioned" |
| 00:20.25 | kram | i see |
| 00:20.28 | bkw_ | dmccollum, you're sol maybe.. but paste what the CLI says. |
| 00:20.35 | dmccollum | [macro-dialout] |
| 00:20.35 | dmccollum | exten => s,1,GotoIf($[foo${ECID${CALLERIDNUM}} = foo]?4) ;check for CID override for exten |
| 00:20.35 | dmccollum | exten => s,2,SetCallerID(${ECID${CALLERIDNUM}}) |
| 00:20.35 | dmccollum | exten => s,3,Goto(6) |
| 00:20.35 | dmccollum | exten => s,4,GotoIf($[foo${OUTCID_${ARG1}} = foo]?6) ;check for CID override for trunk |
| 00:20.36 | dmccollum | exten => s,5,SetCallerID(${OUTCID_${ARG1}}) |
| 00:20.38 | dmccollum | exten => s,6,SetVar(length=${LEN(${DIAL_OUT_${ARG1}})}) |
| 00:20.39 | bkw_ | SCK |
| 00:20.40 | dmccollum | exten => s,7,Dial(${OUT_${ARG1}}/${ARG2:${length}}) |
| 00:20.42 | dmccollum | exten => s,8,Congestion |
| 00:20.43 | bkw_ | NO NO NOT IN HERE |
| 00:20.44 | dmccollum | exten => s,108,Macro(outisbusy) |
| 00:20.50 | dwmw2_gone | has anyone looked hard at chan_bluetooth.c? |
| 00:20.52 | dmccollum | Is that the dial-out lines you're looking for? |
| 00:20.53 | dwmw2_gone | (and remained sane) |
| 00:21.05 | bkw_ | no I need to know what calls that macro |
| 00:21.23 | bkw_ | you'll have something exten => blah,1,Macro(dialout) |
| 00:21.25 | bkw_ | something like that |
| 00:22.03 | bkw_ | watching the CLI while you dial.. and paste that into pastebin.ca |
| 00:22.11 | bkw_ | would be most helpful |
| 00:22.48 | dmccollum | Executing Macro("SIP/201-b73a", "dialout|3|96785208657") in new stack |
| 00:23.12 | bkw_ | just do this |
| 00:23.31 | bkw_ | exten => _X.,1,Dial(Zap/1/${EXTEN}) |
| 00:23.36 | bkw_ | make it simeple |
| 00:23.53 | bkw_ | then when you understand what you're doing them you can jump into more complex things |
| 00:24.22 | bkw_ | i'm all for helping people but we aren't going to school you on ever aspect of what to do unless you atleast try. |
| 00:24.23 | dmccollum | k, I'm using the asterisk@home install. So a lot of that stuff is put in there by them. Trying to figure it all out. |
| 00:24.35 | bkw_ | ACK |
| 00:24.44 | bkw_ | someone needs to smack those people |
| 00:24.47 | bkw_ | and make them do something simple |
| 00:24.50 | bkw_ | but NOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOO |
| 00:24.57 | bkw_ | they gotta make it all complex |
| 00:25.09 | fgravato | haha |
| 00:25.10 | bkw_ | after the executing macro line |
| 00:25.12 | bkw_ | what do you see? |
| 00:25.23 | fgravato | so very try bry |
| 00:25.37 | dmccollum | there's like 20 lines of stuff. i can paste it here if you want. |
| 00:25.37 | bkw_ | asterisk is hard enuf to understand as it is |
| 00:25.43 | bkw_ | pastebin.c |
| 00:25.44 | bkw_ | er |
| 00:25.46 | bkw_ | pastebin.ca |
| 00:25.47 | bkw_ | go use that |
| 00:25.50 | bkw_ | www.pastebin.ca |
| 00:25.53 | bkw_ | paste me the URL |
| 00:25.54 | bkw_ | in here |
| 00:26.18 | bkw_ | kram my man |
| 00:26.22 | bkw_ | no app_chanspy.c? |
| 00:26.36 | bkw_ | cvs add baby? |
| 00:26.38 | bkw_ | hehe |
| 00:27.26 | kram | we're working on chan_spy |
| 00:27.27 | buddah | should have pastebin'd that |
| 00:27.30 | kram | i'm working on it |
| 00:27.33 | kram | we got the first part |
| 00:27.42 | bkw_ | ah kewl |
| 00:27.44 | *** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com) |
| 00:27.53 | bkw_ | ;) |
| 00:28.24 | sivana | kram: is rumors true about a TE411P? :) |
| 00:28.25 | sbarrius | OK, I know I have bitching about broadvoice... but I just talked to a very nice and help tech...they are redeemed in my eyse |
| 00:28.28 | bkw_ | sivana, yes |
| 00:28.37 | jesster | Im troubleshooting a phone hanging off a FXS port of a channel bank. The the phone is in use, and another call comes in, asterisk cli shows "Zap/106-busy-678259056 is busy" but the caller hears ring, then eventually goes dead. I've pasted my relevant zapata.conf here http://pastebin.ca/8080 |
| 00:28.48 | tainted- | sbarrius what kind of problems where u having |
| 00:28.48 | G0shen | I like the visual enhancements to openoffice 2.0beta |
| 00:28.49 | bkw_ | jesster, dude you ghave asked that like 3 times already right? |
| 00:28.50 | Wonka | "cvs commit suicide" |
| 00:29.01 | sbarrius | turns out that after I added my 800 number the password on there side changed... |
| 00:29.04 | jesster | i've asked it a few times since this morning |
| 00:29.06 | dmccollum | bkw: here's the link. http://pastebin.ca/8092 |
| 00:29.31 | bkw_ | output of ztcfg -vvv please |
| 00:29.32 | tainted- | sbarrius i've had problems with my 800 # lately too |
| 00:29.33 | bkw_ | in the pastebin |
| 00:29.56 | bkw_ | dmccollum, I suspect you're about to learn why you don't buy X101's from ebay |
| 00:30.12 | sivana | bah.. another one |
| 00:30.53 | *** join/#asterisk Geo- (~no@h-66-134-200-254.snvacaid.covad.net) |
| 00:31.21 | bkw_ | jesster, you have this hooked to a channelbank? |
| 00:31.32 | bkw_ | paste the output of ztcfg -vvv |
| 00:31.38 | bkw_ | and cat /proc/interrupts |
| 00:31.45 | bkw_ | along with lsmod |
| 00:31.50 | bkw_ | and show version |
| 00:31.57 | jesster | bkw_: yes |
| 00:31.58 | bkw_ | same for you dmccollum |
| 00:32.07 | sbarrius | tainted: did they change your password on you too? |
| 00:33.12 | bkw_ | oh watch this.. brb |
| 00:34.41 | dmccollum | bkw: here's the new link. http://pastebin.ca/8094 |
| 00:35.55 | terrapen | Right about now... |
| 00:36.08 | terrapen | You are about to be posessed by the sounds of... |
| 00:36.10 | bkw_ | show me zaptel.conf and /etc/zapata.conf |
| 00:36.19 | terrapen | MC Rob Base and DJ EZ Rock |
| 00:36.28 | Exstatica | anyone ever seen this? Unable to find key '1231231234' in family 'SIP/Registry' |
| 00:38.10 | buddah | rob base and ez rock |
| 00:38.11 | buddah | nice |
| 00:38.19 | *** join/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net) |
| 00:38.23 | buddah | heh |
| 00:38.23 | jesster | bkw im gonna paste output of ztcfg -vvv to pastebin has it's lengthy |
| 00:38.25 | bkw_ | Exstatica, yes you're trying to do a dbget |
| 00:38.30 | buddah | was base and rock too much for you terrapen? |
| 00:38.46 | terrapen | yeah |
| 00:38.49 | buddah | lol |
| 00:38.50 | terrapen | IT TAKES TWO |
| 00:38.54 | buddah | they are quite a good act to see live |
| 00:38.58 | terrapen | uhhh |
| 00:39.00 | Exstatica | bkw_, yeah i'm trying to use realtime |
| 00:39.05 | terrapen | i saw them on spring break at south padre island |
| 00:39.07 | buddah | rob base's ring he wore nye in 01 was worth more than my house |
| 00:39.07 | terrapen | in about 1996 |
| 00:39.13 | terrapen | and they were lip-syncing |
| 00:39.15 | buddah | wow |
| 00:39.16 | terrapen | (no shit) |
| 00:39.18 | terrapen | it was so depressing |
| 00:39.18 | bkw_ | ok lost interest brb |
| 00:39.19 | buddah | that sucks |
| 00:39.23 | buddah | i hate that |
| 00:39.32 | buddah | its ok for stupid mtv people and like pop chick singers |
| 00:39.37 | buddah | but good acts, dissapointing |
| 00:39.37 | jedaustin | On Broadvoice, other than being $10 more is there any difference between Unlimited World, and Unlimted Business? |
| 00:39.46 | terrapen | it didn't even look like Rob Base |
| 00:39.47 | jesster | bkw_: ztcfg -vvv is now avil on http://pastebin.ca/8095 |
| 00:39.52 | buddah | haha |
| 00:39.54 | buddah | imposter |
| 00:39.55 | terrapen | and i don't remember EZ Rock being with him |
| 00:40.04 | terrapen | it was almost surreal though |
| 00:40.12 | terrapen | i was getting shithoused at this bar with my friend |
| 00:40.18 | terrapen | a giant huge beach bar |
| 00:40.24 | terrapen | it held like 4,000 people or something |
| 00:40.28 | KalD|WORK | is there any way to change the order in which Zap dials out? i.e. Dial out high channels vs low channels? |
| 00:40.36 | terrapen | and the MC introduced him |
| 00:40.40 | terrapen | it was pretty funny |
| 00:40.50 | terrapen | but, i swear, they played the track of the CD |
| 00:40.58 | terrapen | and he lipped along |
| 00:41.47 | *** join/#asterisk sysdef (~s-y-s-d-e@sysdef.admin.debiancenter) |
| 00:42.01 | dmccollum | bkw: here's the updated post with config files. http://pastebin.ca/8097 |
| 00:42.06 | jesster | bkw_: do you want interrupts pasted to private msg or on pastebin |
| 00:43.07 | jesster | bkw_: show version output: Asterisk CVS-Nv1-0-7-03/20/05-16:52:03 |
| 00:43.18 | tclark | KalD|WORK: ya use g or G or are i think for round robin |
| 00:44.15 | harryvv | <PROTECTED> |
| 00:44.34 | harryvv | Dont know if thats high or low based on other carriers. |
| 00:44.40 | jedaustin | harryvv: is that including the local loop? |
| 00:44.51 | harryvv | That was not discussed. |
| 00:45.02 | harryvv | you mean the instalation of the physical plant? |
| 00:45.07 | harryvv | laying the cable? |
| 00:45.44 | jedaustin | In AZ they charge you once for the physical line, an again for the isp connection (T1 might be different) |
| 00:45.58 | *** part/#asterisk sysdef (~s-y-s-d-e@sysdef.admin.debiancenter) |
| 00:46.59 | terrapen | i wonder if this Asterisk consulting gig will ever pan out |
| 00:47.08 | terrapen | supposedly it is August time frame |
| 00:47.11 | terrapen | which really blows |
| 00:47.19 | terrapen | thats like.... six fucking months almost |
| 00:47.25 | harryvv | Get it in writting |
| 00:47.31 | terrapen | this lady is probably just jerking us off |
| 00:47.36 | jedaustin | terrapen: dont wait for it.. take on other work |
| 00:47.51 | terrapen | the friend that brings these things to me...about 1/3 of the project actually pan out |
| 00:47.54 | harryvv | Why August? |
| 00:48.00 | terrapen | jed: oh, hell no, of course not :) |
| 00:48.06 | harryvv | yea was going to say the same thing. |
| 00:48.16 | terrapen | harry: who knows...supposedly this company is moving into an office in august and needs the phone system |
| 00:48.27 | terrapen | i hate paying commission to my friend |
| 00:48.29 | terrapen | but i will have to |
| 00:48.34 | harryvv | hate it? |
| 00:48.43 | terrapen | anyone have suggestions for developing your own business leads? |
| 00:48.44 | *** join/#asterisk bitbucket (~user@H203.C18.B96.tor.eicat.ca) |
| 00:48.44 | harryvv | hecj he is brining you the work |
| 00:48.45 | harryvv | :) |
| 00:48.48 | terrapen | i know :) |
| 00:48.54 | terrapen | but im greedy |
| 00:48.58 | terrapen | i will gladly pay him |
| 00:48.59 | jedaustin | terrapen: too iffy.. move on but keep it on the back burner |
| 00:48.59 | bitbucket | hello |
| 00:49.05 | bitbucket | have question re: dtmf |
| 00:49.06 | terrapen | but i want to start bringing my own things in |
| 00:49.11 | jedaustin | terrapen: word of mouth works for me |
| 00:49.16 | harryvv | you really cant do this terracon. Look at the big picture. |
| 00:49.17 | terrapen | anyone ever advertised phone stuff in the Business Journal? |
| 00:49.24 | bitbucket | is there someone who can help? |
| 00:49.37 | harryvv | Honesty in biz really pays off. |
| 00:49.42 | terrapen | hary: cant do what? |
| 00:49.45 | jedaustin | bitbucket: just ask your question if someone knows they'll speak up |
| 00:50.14 | bitbucket | i have a sip provider, using rfc2833 for dtmf |
| 00:50.25 | bitbucket | but sometimes digits are doubled or dropped |
| 00:50.28 | jedaustin | bitbucket: heres are good lists to join http://lists.digium.com/mailman/listinfo/ |
| 00:50.29 | bitbucket | inconsistently... |
| 00:50.47 | Shido6 | rfc2833 shouldnt do that |
| 00:50.51 | Shido6 | u sure its not inband? |
| 00:50.56 | bitbucket | certain. |
| 00:51.04 | jedaustin | bitbucket: which provider? |
| 00:51.19 | *** join/#asterisk JerJer[mobile] (~jj@65.173.197.174) |
| 00:51.22 | bitbucket | for example, level3 |
| 00:52.07 | roamer323 | bitbucket - which codec is used? and for incoming or outgoing calls or both? |
| 00:52.18 | bitbucket | only outgoing calls, and g729 |
| 00:52.32 | jedaustin | Hmm.. does anyone know the difference between Broadvoice's Unlimited World and Unlimited Business accounts? |
| 00:52.37 | jedaustin | Other than $10 :) |
| 00:52.52 | roamer323 | bitbucket -well - g.729 , you're definitely rfc2833 - inband would've never worked |
| 00:53.16 | *** part/#asterisk oo (~marko@marko.horde) |
| 00:53.32 | roamer323 | bitbucket - are you on 1.0.7? |
| 00:53.42 | *** join/#asterisk ManxPower (~eric@ip-209-16-83-10.i-55.com) |
| 00:54.08 | bitbucket | no, 1.0.5 |
| 00:54.21 | bitbucket | has something changed about dtmf handling between those releases? |
| 00:54.48 | roamer323 | bitbucket - no but some g.729 bugs - are you on digum's for-paid codec? |
| 00:54.53 | bitbucket | yes |
| 00:54.54 | *** join/#asterisk testing234235453 (~testing23@cc2-24.217.112.154.charter-stl.com) |
| 00:55.13 | *** join/#asterisk jeffik (~jeffik@69.158.37.207) |
| 00:55.23 | roamer323 | bitbucket - sorry, no further ideas - hope someone else does |
| 00:55.35 | bitbucket | hmmm... |
| 00:55.36 | bitbucket | thanks |
| 00:55.52 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 00:56.04 | testing234235453 | Got a question about flashing FXO |
| 00:56.21 | jedaustin | bitbucket, sure it's not your phone? |
| 00:56.35 | FuriousGeorge | there isnt any problem with using a softphone on the server to test my initial setup, is there? does that create any sort of conflictsetup |
| 00:56.49 | *** join/#asterisk epoch (epoch@octane.breakbeats.org) |
| 00:56.51 | jedaustin | FuriousGeorge: nope, that's what I did |
| 00:57.06 | FuriousGeorge | one last ?: where can i find x-lite for linux. its like hidden |
| 00:57.45 | jedaustin | Hmm.. good question |
| 00:58.08 | testing234235453 | I bought a TDM11B so I could do call waiting, need some help |
| 00:58.23 | bitbucket | jedaustin -- i doubt it. i've tried with several softphones and a cisco 7905, and the same behaviour with each |
| 00:58.35 | testing234235453 | I configured a macro to flash the line then transfer back but no luck |
| 01:00.11 | jedaustin | FuriousGeorge: I can't vouch for it, but theres a project on source forge called 'shtoom' that claims to be a sip framework/phone written in python |
| 01:00.25 | jedaustin | Siphon http://sourceforge.net/projects/siphon/ |
| 01:00.46 | jedaustin | Oops.. that one is still in progress |
| 01:03.06 | Geo- | hmm can i get my outgoing caller ID to a DID i own that isnt hooked into the PRI im using to call out from? the DID forwards to the DIDs on the PRI.. |
| 01:05.03 | *** join/#asterisk MikeJ[Jayden] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
| 01:05.33 | *** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc) |
| 01:05.54 | *** join/#asterisk zhier (~nick@61.144.20.242) |
| 01:06.43 | *** join/#asterisk pigpigpig (~pig@165.21.246.202) |
| 01:07.05 | ariel_ | question for anyone with T1/Pri use. Is there any program or utility for asterisk or for the linux server that will monitor and email or page if the t1 line goes down? |
| 01:07.23 | zhier | can i configure my pc as a sever with Asterisk PBX? |
| 01:07.52 | MikeJ[Jayden] | ~docs |
| 01:07.54 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 01:08.05 | MikeJ[Jayden] | zhier, ^^^^ |
| 01:08.50 | zhier | MikeJ[Jayden] :what |
| 01:09.53 | *** join/#asterisk Max629 (Max629@h-64-105-18-75.chcgilgm.covad.net) |
| 01:10.03 | *** part/#asterisk Max629 (Max629@h-64-105-18-75.chcgilgm.covad.net) |
| 01:11.20 | MikeJ[Jayden] | read about * a little. |
| 01:11.37 | MikeJ[Jayden] | you had an EXTREMELY broad question... |
| 01:11.44 | drumkilla | the answer is yes :) |
| 01:12.14 | che | hehehe |
| 01:12.16 | MikeJ[Jayden] | the answer is asterisk is a realtime app, and generally shuold run on a dedicated box |
| 01:12.19 | che | but a short answer ;) |
| 01:12.43 | drumkilla | MikeJ[Jayden]: well, if you're just doing your personal stuff in your house, it's not that big of a deal |
| 01:12.43 | MikeJ[Jayden] | hehe |
| 01:12.57 | zhier | and i know the documentation says "yes", but i can't do this. i have tried for several days! |
| 01:12.59 | drumkilla | if you're just starting out and playing |
| 01:13.11 | drumkilla | zhier: well maybe you should specify your problem :) |
| 01:13.19 | niZon | I run asterisk in vmware :P |
| 01:13.25 | MikeJ[Jayden] | ummmm, I suppose, untill you are runing it on a machine with X or windows, and any time you do somthing cpu intensive your audio goes to hell |
| 01:13.29 | niZon | It sucks horribly though :( |
| 01:13.37 | *** join/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 01:13.46 | zhier | i want configure my pc as a sever. |
| 01:13.56 | MikeJ[Jayden] | os? |
| 01:14.04 | drumkilla | (i'm going to guess windows) |
| 01:14.46 | testing234235453 | I just got a TDM11B, can call outside numbers from FXS but not internal. I know it's probably dialplan but I don't see error on console |
| 01:15.18 | MikeJ[Jayden] | running it on windows would be fine if you are running asteriskwin32, BUT, that is only a 1.0.5 release (2 minor releases behind) and you dont have sourcecode (as he will not be releasing the patches till next week) |
| 01:15.37 | *** join/#asterisk IQ (~IQ@70-59-165-54.omah.qwest.net) |
| 01:15.40 | drumkilla | i wonder if his patches are commit-worthy ... |
| 01:15.43 | MikeJ[Jayden] | and by fine I mean, if nothing else is running on the box, especially on windows |
| 01:16.20 | MikeJ[Jayden] | drumkilla, he said he would have them out on monday, I also have some stuff another guy worked on, but in written, not patch form |
| 01:16.29 | drumkilla | cool stuff |
| 01:16.39 | drumkilla | is it written in a way that it could be merged with the tree? |
| 01:16.57 | MikeJ[Jayden] | the reason I cautioned so much about other stuff on the same box with windows is because the cygwin support for realtime priority is messed up |
| 01:17.12 | MikeJ[Jayden] | the patches I have are pretty minor, |
| 01:17.14 | MikeJ[Jayden] | BUT |
| 01:17.34 | drumkilla | I'm curious to see what had to be changed |
| 01:17.54 | MikeJ[Jayden] | there are still issues w/ the patches I have...but I don't have several things working yet |
| 01:18.09 | MikeJ[Jayden] | one sec... |
| 01:18.43 | dwmw2_gone | wheee. chan_bluetooth actually works and does full duplex |
| 01:18.53 | MikeJ[Jayden] | I have seen one set, and about to look on monday at another set... problem is, if they are not disclaimed, I may not be able to do the patches.... |
| 01:19.01 | MikeJ[Jayden] | do you know the line on that? |
| 01:19.08 | MikeJ[Jayden] | can I point you in the right direction? |
| 01:19.52 | MikeJ[Jayden] | or is it fine, as long as I re do it my way, as the stuff I have is pretty rough, mostly... This thread priority stuff dosn't work in cygwin so ifdef it out. |
| 01:19.53 | drumkilla | hm ... why wouldn't the authors be willing to disclaim it if it was minor? |
| 01:20.11 | drumkilla | MikeJ[Jayden]: I think it's fine if you redo it your own way |
| 01:20.23 | MikeJ[Jayden] | well... the asteriskwin32 guy is trying to sell customized versions |
| 01:20.27 | IQ | Hi, what is "MOH Server" ? |
| 01:20.33 | MikeJ[Jayden] | so kram and him probably need to chat |
| 01:20.39 | drumkilla | I think the knowhow for porting that stuff is public knowlege |
| 01:20.51 | MikeJ[Jayden] | it is |
| 01:20.54 | testing234235453 | I just got a TDM11B, can call outside numbers from FXS but not internal. I know it's probably dialplan but I don't see error on console |
| 01:20.55 | drumkilla | well, he's releasing the code |
| 01:21.11 | MikeJ[Jayden] | releasing gpl and disclaiming are 2 diff things |
| 01:21.15 | drumkilla | he's selling the service of customization, not the code ... so I don't see why it would matter |
| 01:21.20 | drumkilla | yeah, i know ... |
| 01:21.29 | MikeJ[Jayden] | but I will do the leg work on the patches if need be |
| 01:21.56 | drumkilla | well I think that would be pretty lame if he wouldn't be willing to disclaim it |
| 01:21.59 | MikeJ[Jayden] | I kinda want to do some more ont he windows side than he did, having * run as a service and such |
| 01:22.09 | drumkilla | that's hardcore |
| 01:22.24 | MikeJ[Jayden] | service stuff is really easy w/ cygwin... |
| 01:22.25 | SpaceBass | anyone know about a pa168 ata? |
| 01:22.33 | drumkilla | I know *nothing* about windows programming |
| 01:23.22 | MikeJ[Jayden] | in cygwin there is a couple system calls, basically, you need a mode w/ no io to stdin or stdout... |
| 01:23.24 | drumkilla | MikeJ[Jayden]: you better put on your flame-retardant gear when you get to work on that :) |
| 01:23.46 | MikeJ[Jayden] | I have had it running on m,y laptop for a couple months.... |
| 01:24.04 | drumkilla | cool |
| 01:25.00 | MikeJ[Jayden] | every once in a while, when people get going off abnout linux, I paste in my console stuff that says asterisk 1.0.x on cygwin and get people going, mark was receptive when I mentioned I had patches, so when I have a better idea of what this guy did, I will call him to discuss before I post it |
| 01:25.24 | drumkilla | well I think that's great |
| 01:25.31 | drumkilla | anything that will help adoption |
| 01:25.42 | drumkilla | and having the option to try it in windows will help get people to try it out |
| 01:25.49 | MikeJ[Jayden] | I just need to do a little reorg of cli vs daemon so that I can have it run smooth as a service, with either a diff executable, or somthing for the cli |
| 01:25.57 | SpaceBass | how do I set the RTP port in sip.conf? |
| 01:26.05 | MikeJ[Jayden] | exactly, but no zap |
| 01:26.07 | drumkilla | SpaceBass: rtp.conf |
| 01:26.13 | MikeJ[Jayden] | now, that would be fun to get working... |
| 01:26.15 | drumkilla | MikeJ[Jayden]: sure, as with any other OS != linux |
| 01:26.17 | SpaceBass | drumkilla i have a start and an end range |
| 01:26.27 | drumkilla | SpaceBass: right |
| 01:26.37 | SpaceBass | but I have a sip ATA that needs a specific port |
| 01:27.04 | MikeJ[Jayden] | drumkilla, are you on break? I have seen you on a lot more than usual? |
| 01:27.09 | drumkilla | MikeJ[Jayden]: yeah :) |
| 01:27.20 | *** join/#asterisk topping (~topping@dhcp024-210-082-196.columbus.rr.com) |
| 01:27.24 | MikeJ[Jayden] | nice, I miss being in school |
| 01:27.37 | SpaceBass | so trying to figure out exactly which port it needs |
| 01:28.12 | dwmw2_gone | are there restrictions on when you can call ast_verbose() ? |
| 01:28.33 | FuriousGeorge | does anyone know where to get x-lite (or any decent softphone) for linux |
| 01:28.49 | SpaceBass | did you try the digium one? gnophone or what ever? |
| 01:29.36 | *** part/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 01:29.48 | SpaceBass | ok, finally got this sip ata to register |
| 01:29.48 | dwmw2_gone | thread 12 in http://david.woodhou.se/backtrace.txt seems to have deadlocked in ast_verbose() |
| 01:29.58 | SpaceBass | but it still not working... has to be an rtp issue |
| 01:30.57 | FuriousGeorge | i am emrging gnophone |
| 01:31.07 | *** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net) |
| 01:31.22 | FuriousGeorge | spacebass: did you say it wasnt working? |
| 01:31.34 | SpaceBass | FuriousGeorge my sip ata isnt working |
| 01:31.43 | FuriousGeorge | ahh |
| 01:31.52 | *** join/#asterisk mickm (~mickm@220-245-98-72-qld-pppoe.tpgi.com.au) |
| 01:32.13 | SpaceBass | does anyone know, is there a way to define a specific rtp port for a sip client? |
| 01:32.17 | testing234235453 | I just got a TDM11B, can call outside numbers from FXS but not internal. I know it's probably dialplan but I don't see error on console |
| 01:32.35 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || 1.0.7 Released || http://www.theregister.co.uk/2005/03/16/asterisk_open_source_pbx/ || Bravo.. ChanSpy is in CVS... Everyone send loads of cash to anthm ;) |
| 01:32.39 | SpaceBass | testing234235453 did you change the manage.conf ? |
| 01:32.52 | SpaceBass | i only ask b/c I had a similar problem and that was the cause |
| 01:32.57 | testing234235453 | manage.conf? |
| 01:33.09 | testing234235453 | hmmm, what did you change? |
| 01:33.14 | testing234235453 | I'm looking now |
| 01:33.27 | SpaceBass | or manager.conf i guess |
| 01:34.15 | testing234235453 | I did a while ago, but it just has access info for the manager interface |
| 01:34.28 | *** join/#asterisk SexyKen (~sexyken@c-67-161-5-149.client.comcast.net) |
| 01:34.33 | mickm | hi all, i am new to asterisk & have a hardware enquiry |
| 01:35.20 | SpaceBass | drumkilla if I have a range of ports in rtp.conf how do i know which one a client is using? for instance if I was opening ports on a firewall |
| 01:36.13 | SpaceBass | mickm just ask |
| 01:36.22 | mickm | is the 4-line Dialogic D/41D supported |
| 01:36.31 | *** join/#asterisk xeet2 (~joe@es.jsci.net) |
| 01:36.53 | testing234235453 | I don't think the manager.conf would help me |
| 01:37.20 | SpaceBass | testing234235453 i didn't see why it affected me at all, but it did... just thought I'd throw it out there |
| 01:37.27 | *** join/#asterisk riksta (~rick@81-178-199-213.dsl.pipex.com) |
| 01:37.42 | testing234235453 | ok, do you remember what you changed? |
| 01:37.48 | SpaceBass | yeah, the admin password |
| 01:38.02 | *** join/#asterisk [0xBoTNeTBusTr] (~bitbuster@67-23-242-78.atlaga.adelphia.net) |
| 01:38.07 | SpaceBass | I made it the correct password which fixed the error i was getting in the CLI, but broke sip to sip calling |
| 01:38.10 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 01:38.11 | SpaceBass | broke internal calls |
| 01:38.18 | SpaceBass | fuck this SIP ata |
| 01:38.21 | SpaceBass | arrruuugggg |
| 01:38.28 | *** join/#asterisk jojoba (~aaronzhon@220.248.36.42) |
| 01:39.09 | testing234235453 | what pwd did you associate with the manager pwd |
| 01:39.23 | SpaceBass | what I made my manager password :) |
| 01:39.49 | SpaceBass | as long as the password is wrong, everything works fine |
| 01:39.58 | stustu | Can someone tell me if it's possible for Asterisk to send Caller Id after polarity reversal, before the first ring, on an fxs channel? |
| 01:40.00 | SpaceBass | i'm using asterisk@home which might have something to do with it |
| 01:40.44 | testing234235453 | ahhh, I had an X100p in this system with an ATA186 working just fine |
| 01:41.08 | testing234235453 | everything works with the TDM11B except internal calling |
| 01:41.21 | testing234235453 | the numbers are in the same context |
| 01:41.30 | SpaceBass | i have 2 x100ps and I could call out and in from them just fine, it was just internal calling... all in the same context |
| 01:41.45 | SpaceBass | i reinstalled 2x before I realized that is what I was changing that affected it |
| 01:42.22 | Geo- | hm |
| 01:43.21 | testing234235453 | I get an immediate reorder |
| 01:43.35 | *** join/#asterisk meanphil (~pmurray@222-152-246-166.jetstream.xtra.co.nz) |
| 01:43.41 | mickm | anyone know if the 4-line Dialogic D/41D works ok with asterisk |
| 01:43.59 | *** join/#asterisk che (~che@che.user) |
| 01:44.23 | Qwell | Note to self: Do not ask a friend to leave you a test voicemail. |
| 01:44.30 | SpaceBass | lol |
| 01:44.55 | bonez41 | Qwell, get a funny test voicemail? |
| 01:45.03 | Qwell | bonez41: something like that |
| 01:45.03 | testing234235453 | I get an immediate hangup and reorder |
| 01:48.09 | *** join/#asterisk ronn (~zakforeve@84.45.132.117) |
| 02:02.32 | SexyKen | Hey guys...anyone work with faxing & asterisk much? |
| 02:03.01 | SpaceBass | anyone ever use a pa168 or 1t-168 ata? |
| 02:03.32 | SexyKen | Asterisk prints this: http://pastebin.ca/8103 -- but the Fax I'm sending from says 'Fax did not answer' |
| 02:03.36 | SexyKen | Anyone know why this would happen? |
| 02:03.49 | Katty | hmm |
| 02:04.28 | *** join/#asterisk Rob- (~robbie@haylott.plus.com) |
| 02:04.50 | SexyKen | Anyone know how to make the call use a certain codec? |
| 02:05.11 | Katty | call collect! </sarcasm> |
| 02:06.11 | testing234235453 | what can I debug to see a phone off FXS fail to call a SIP device? Console is not showing error |
| 02:07.20 | mikegrb | set verbose 10 |
| 02:07.26 | mikegrb | set debug 10 |
| 02:07.55 | tainted- | sbarrius you there? |
| 02:08.25 | SexyKen | Hey -- > actual format = gsm, -- is there anyway to force G711 on an incoming call? (only for a certain did) |
| 02:09.13 | testing234235453 | did that ... all I get is Hungup 'Zap/1-1' |
| 02:10.59 | *** join/#asterisk omarc55 (~omarc55@adsl-2-211-118.mia.bellsouth.net) |
| 02:12.09 | testing234235453 | set debug 10 and tried it. Got same but also said 'urgent handler' ???? |
| 02:13.09 | Katty | s/????/? |
| 02:13.11 | tainted- | how do i turn off sip debug |
| 02:13.25 | Qwell | Katty: s/??/?/g :) |
| 02:13.36 | Katty | Qwell: (= |
| 02:14.19 | Sedorox | hmmm |
| 02:14.44 | tzanger | oh guys listen up |
| 02:14.45 | tzanger | http://audio.cdbaby.com/rmj/laliberte-07.rm |
| 02:14.48 | tzanger | I want more music like that |
| 02:14.53 | tzanger | wtf kind of music is that |
| 02:15.07 | TomL | ManxPower: you around? |
| 02:15.17 | Qwell | Whats with all these realplayer links lately? |
| 02:15.22 | Katty | tzanger: tsk tsk, always demanding |
| 02:15.42 | tzanger | Katty: I am a very demanding person. |
| 02:15.50 | Katty | tzanger: i see |
| 02:15.50 | tzanger | very demanding of myself and of others |
| 02:16.15 | Katty | k |
| 02:16.30 | tzanger | I think I just discovered that I like latin guitar |
| 02:16.37 | tzanger | http://kevinlaliberte.com/recording.html |
| 02:16.45 | tzanger | I just practised for an hour and I sound NOTHING like that |
| 02:16.50 | TomL | dant: you here? |
| 02:17.22 | TomL | maybe someone else knows a bit about about cisco qos... |
| 02:17.36 | testing234235453 | try me, on qos |
| 02:17.39 | TomL | does "match dscp ef" match SIP packets automatically or do they need to be mangled before they reach the cisco? |
| 02:17.46 | tzanger | I'm a QoS guru, just not on Cisco :-) |
| 02:18.12 | testing234235453 | ef doesn't = sip |
| 02:18.27 | TomL | what is it? |
| 02:18.37 | jojoba | Anyone knows how to modify asterisk source code to detect answer on FXO channels outside US? |
| 02:18.37 | testing234235453 | they hav e to be marked first by something |
| 02:18.50 | testing234235453 | phone, or application |
| 02:18.59 | TomL | the phone itself might do it? |
| 02:19.01 | TomL | k |
| 02:19.22 | testing234235453 | cisco's set ip precedence 5 by default |
| 02:19.37 | TomL | oh no shit |
| 02:19.48 | TomL | thats kinda... gay :) |
| 02:20.00 | TomL | oh goddamnit, I don't remember the password for my sipura |
| 02:20.12 | SexyKen | Hey -- > actual format = gsm, -- is there anyway to force G711 on an incoming call? (only for a certain did) |
| 02:21.04 | testing234235453 | Voice is king when it comes to Cisco QoS |
| 02:21.05 | tzanger | SexyKen: if you can have them come in to a different type=user, of course |
| 02:21.30 | tzanger | SexyKen: otherwise there was some talk of per-call codec negotiation on the list a little while back but I don't think it really got anywhere |
| 02:22.12 | SexyKen | This is for incoming |
| 02:22.58 | tzanger | SexyKen: I know |
| 02:23.04 | dwmw2_gone | I suppose it would be more userfriendly if you didn't have to recompile chan_bluetooth if you want it to dial a different number? |
| 02:23.41 | *** join/#asterisk FxMulder (~da_wally@firewall.goldenwesttechnologies.com) |
| 02:24.07 | FxMulder | anyone know why I'd be getting "Unable to specify channel 1: No such device or address" in http://dev1.jiffe.com/asterisk |
| 02:24.37 | *** join/#asterisk chaoscon (~ph33r@chaoscon.user) |
| 02:24.41 | fgravato | tzanger you trying to be Ottmar Liebert |
| 02:24.43 | fgravato | :-) |
| 02:25.11 | tzanger | who's that? |
| 02:25.18 | tzanger | Katty: http://www.cdbaby.com/cd/laliberte |
| 02:25.24 | fgravato | another latin guitar |
| 02:25.26 | fgravato | player |
| 02:25.26 | tzanger | especially track 2 and 7... I had no idea |
| 02:25.30 | tzanger | I liked this stuff |
| 02:25.35 | Katty | beep |
| 02:25.38 | Qwell | FxMulder: did you mean to put Zap/1? |
| 02:25.42 | Katty | tzanger: k |
| 02:25.49 | tzanger | Katty: my birthday's coming up, hint, hint |
| 02:25.59 | tzanger | actually no hint, I'll have bought it by then |
| 02:26.00 | *** join/#asterisk zhier (~nick@219.137.40.17) |
| 02:26.11 | tzanger | anyway bedtime |
| 02:26.11 | Katty | tzanger: silly rabbit, i don't waste money |
| 02:26.17 | Katty | and irish |
| 02:26.20 | tzanger | Katty: it's not wasting... that's good music |
| 02:26.26 | tzanger | worth buying |
| 02:26.27 | Katty | tzanger: k |
| 02:26.42 | *** join/#asterisk MikeJ[Laptop] (~icechat5@pcp02795302pcs.roylok01.mi.comcast.net) |
| 02:26.48 | Katty | tzanger: nini |
| 02:28.27 | omarc55 | Hi all, I am getting chan_zap.c:6939 zt_request: Unknown option '}' in '1}' when I dial another extension, I've tried to find out what this is but no luck. anybody know what this might be? |
| 02:28.50 | SexyKen | tzanger -- so how would I fix it up to force G711 on the call? |
| 02:30.47 | fgravato | tzanger i have some Ottmar Liebert |
| 02:30.49 | fgravato | on cd |
| 02:30.53 | fgravato | if you want ? |
| 02:30.55 | FxMulder | its referring to zapata.conf |
| 02:31.10 | FxMulder | channel=>1 |
| 02:31.28 | Qwell | omarc55: Whats your dial line look like? |
| 02:31.46 | Qwell | FxMulder: oh |
| 02:32.24 | omarc55 | exten => 640,1,Dial(ZAP/1}) |
| 02:32.26 | Qwell | FxMulder: the zt tools show everything is working ok? |
| 02:32.31 | Qwell | omarc55: There's obviously a typo there |
| 02:32.34 | Qwell | remove the } |
| 02:32.35 | SexyKen | How can I force G711 on an incoming call? |
| 02:32.45 | bkw_ | only allow it |
| 02:32.47 | bkw_ | or allow it first |
| 02:33.09 | Hmmhesays | heh |
| 02:33.31 | FxMulder | I've only used those once, sec |
| 02:33.40 | omarc55 | well, that was easy. thanks! |
| 02:33.54 | *** join/#asterisk guugmember (~Casa@200.6.219.149) |
| 02:34.05 | SexyKen | bkw_ -> Can I set this 'allow' for only a certian DID? |
| 02:34.17 | bkw_ | omg |
| 02:34.21 | bkw_ | you're kidding me right? |
| 02:34.27 | bkw_ | does it come from X or Y provider |
| 02:34.33 | bkw_ | and is that the only did from said provider? |
| 02:34.43 | SexyKen | I have 10 DID's froma single provider. |
| 02:34.52 | bkw_ | and you just want this one to be ulaw right? |
| 02:34.53 | SexyKen | Only 1 of the DID's needs to be G711 only |
| 02:34.57 | omarc55 | I am having problems with call parking too, its not working. I dial #700 and I hear the numbers being pressed on the other phone but its not doing anything. features.conf says the parkext is 700. and I've included the parkedcalls in my context. do you know what could be wrong? |
| 02:34.59 | bkw_ | let me guess faxing? |
| 02:35.01 | SexyKen | Yes. |
| 02:35.04 | bkw_ | EVIL |
| 02:35.05 | bkw_ | good luck |
| 02:35.06 | bkw_ | NEXT!!! |
| 02:35.09 | SexyKen | :? |
| 02:35.15 | bkw_ | faxing over g711 is spotty at best |
| 02:35.17 | fgravato | hahaha |
| 02:35.17 | guugmember | how can I make a call to a 911 through internet, with the location issue |
| 02:35.30 | bkw_ | guugmember, go read on google |
| 02:35.42 | FxMulder | "/usr/src/zaptel/ztcfg" fails on ZT_CHANCONFIG failed on channel 1: Invalid argument (22) |
| 02:35.50 | bkw_ | I smell no hardware |
| 02:35.54 | bkw_ | or a misconfigure |
| 02:35.57 | guugmember | bkw_, that is why im here |
| 02:36.01 | Qwell | missing driver? |
| 02:36.05 | bkw_ | we aren't 911 experts |
| 02:36.22 | FxMulder | it works, thats whats weird.. it works and then I notice the server stalls, I restart and I get configuration errors |
| 02:36.27 | bkw_ | that is something that is done 100 different ways |
| 02:36.29 | *** part/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 02:36.36 | bkw_ | some people link 911 to the local police dept |
| 02:36.38 | JerJer[mobile] | don't you just push 911 out your PRIs? :P |
| 02:36.44 | bkw_ | ya really |
| 02:36.44 | Qwell | JerJer[mobile]: y0 |
| 02:36.46 | guugmember | bkw_, that was the answer i was expecting |
| 02:36.48 | SexyKen | bkw_, http://pastebin.ca/8103 - that's the result that asterisk shows, but the fax machine I'm sending from says 'no answer from fax' |
| 02:36.52 | JerJer[mobile] | let the telco deal wth it :) |
| 02:37.15 | fgravato | yeah let nufone deal with it - wink wink :-p |
| 02:37.15 | Qwell | JerJer[mobile]: Did Greg already tell you about my funky provisioned tollfree DID? :p |
| 02:37.15 | brc_ | bkw_, your box crashed |
| 02:37.28 | JerJer[mobile] | Qwell: yep... try getting another one |
| 02:37.32 | JerJer[mobile] | if we have any more |
| 02:37.34 | Qwell | he already did |
| 02:37.35 | bkw_ | nope |
| 02:37.50 | Qwell | JerJer[mobile]: Any idea what happened there? Kinda curious |
| 02:37.52 | SexyKen | bkw_, Does that mean it's using gsm and not g711? |
| 02:38.03 | JerJer[mobile] | that number doesn't land on our switch, so the RespOrg screwed up |
| 02:38.26 | Qwell | JerJer[mobile]: weird. when I called later last night, I got a comedian mailbox |
| 02:38.50 | FxMulder | dunno, I'll have to play with it when I get home |
| 02:39.03 | JerJer[mobile] | there are more than just one asterisk user |
| 02:39.13 | Qwell | JerJer[mobile]: I know, heh |
| 02:40.17 | Beirdo | JerJer[mobile]: thanks for the quick turn around on the DID :) |
| 02:40.50 | guugmember | who has played with the varion cards here? |
| 02:41.06 | JerJer[mobile] | guugmember: I don't think many are that stupid |
| 02:41.25 | guugmember | JerJer[mobile], why? not good cards? |
| 02:41.28 | SexyKen | Can I chang ethis: exten => 2122034845,1,Macro(faxreceive) to force G711 or no |
| 02:41.39 | *** join/#asterisk voyage (Savek@192.109.89.3) |
| 02:41.43 | voyage | hi all |
| 02:42.20 | JerJer[mobile] | guugmember: Digium has dramatically improved the zapata design |
| 02:42.36 | zhier | and i don't know how to construct a sever with my own pc. |
| 02:42.38 | voyage | is it possible to use asterisk to just play SIP-Proxy for a couple of clients? |
| 02:42.42 | TomL | testing234235453: would you happen to know if the Sipura-1001 does that by default? |
| 02:42.50 | bkw_ | SexyKen, since when has exten => had anything to do with receiving a call? |
| 02:42.52 | bkw_ | the call is already up |
| 02:42.56 | bkw_ | by the time it gets there |
| 02:42.58 | guugmember | JerJer[mobile], http://lists.digium.com/pipermail/asterisk-users/2005-January/086450.html |
| 02:43.12 | guugmember | JerJer[mobile], but the price difference is considerable |
| 02:43.19 | SexyKen | bkw_, I see. I'm so confused man...I just want to force G711 for a single did. |
| 02:43.35 | JerJer[mobile] | guugmember: Caveat Emptor |
| 02:44.06 | JerJer[mobile] | they produce their own boards in-house |
| 02:44.15 | JerJer[mobile] | and they are using the old dallas T-1 framer |
| 02:44.22 | JerJer[mobile] | buy a Digium board |
| 02:44.25 | *** join/#asterisk maruz (~maumar@adsl-123-3.38-151.net24.it) |
| 02:44.28 | mog_home3 | bkw_ can you explain the f option to me for app dial? |
| 02:44.37 | guugmember | JerJer[mobile], digium board? or digium card? |
| 02:44.45 | Qwell | board ~= card |
| 02:44.45 | docelmo | guugmember, same thing |
| 02:44.47 | JerJer[mobile] | what's the differenece? |
| 02:44.50 | guugmember | ok |
| 02:45.03 | guugmember | JerJer[mobile], what is Caveat Emptor? |
| 02:45.03 | SexyKen | bkw, I suppose if I was using a completely different provider for the fax did, I'd understand it but since I have 10 did's, 9 of them being voice did's, I dont know what to do. |
| 02:45.15 | bkw_ | mog_home3, so 1000 don't get leaked to the PSTN |
| 02:45.17 | bkw_ | as the cid |
| 02:45.28 | JerJer[mobile] | SexyKen: DNIS |
| 02:45.32 | bkw_ | he can't |
| 02:45.43 | JerJer[mobile] | then fire that lame provider |
| 02:45.48 | mog_home3 | but what does |
| 02:45.51 | mog_home3 | a did |
| 02:45.54 | bkw_ | it all hits the same user/pass |
| 02:46.01 | mog_home3 | and how does it know to get which did |
| 02:46.02 | JerJer[mobile] | so? |
| 02:46.10 | JerJer[mobile] | it still should come down with DNIS |
| 02:46.21 | drumkilla | bkw_: how come app_chanspy.c includes channel_pvt.h ? |
| 02:46.31 | drumkilla | that doesn't exist :) |
| 02:46.38 | JerJer[mobile] | not any more |
| 02:46.54 | JerJer[mobile] | the author needs to update his code |
| 02:46.56 | drumkilla | the one that just got committed tries to include it |
| 02:47.22 | blitzrage | yo |
| 02:47.35 | JerJer[mobile] | hoe |
| 02:47.39 | blitzrage | *gasp* |
| 02:47.40 | drumkilla | just wanted to get a sanity check before I committed the change |
| 02:48.30 | sudhir492 | For terminating calls on H323, which is preferred : h323 or oh323? |
| 02:48.46 | bkw_ | drumkilla, hold off chanspy is broken |
| 02:49.01 | mog_home3 | bkw_? |
| 02:49.07 | mog_home3 | the f thing? |
| 02:49.09 | drumkilla | well, it at least compiles with that line out |
| 02:49.17 | bkw_ | ya |
| 02:49.25 | bkw_ | mog_home3, go read the show application dial |
| 02:49.27 | bkw_ | it tells you boi |
| 02:49.29 | mog_home3 | i have |
| 02:49.30 | bkw_ | I call A to B |
| 02:49.31 | mog_home3 | boi |
| 02:49.31 | drumkilla | bkw_: so you tell me |
| 02:49.35 | bkw_ | b transfers to C on the PSTN |
| 02:49.42 | bkw_ | the callerid doesn't need to be A's cid |
| 02:50.15 | mog_home3 | but how does it know caller id of c |
| 02:50.29 | bkw_ | it doesn't... it gives you away to FORCE it |
| 02:50.33 | bkw_ | to be something other than it is |
| 02:50.34 | mog_home3 | whatever , if i dont need to plug vards i dont really care |
| 02:50.35 | bkw_ | ask mark |
| 02:50.37 | bkw_ | he wrote it |
| 02:50.43 | mog_home3 | i am just testing dial plan stuff for mark |
| 02:50.45 | bkw_ | hehe |
| 02:50.48 | mog_home3 | every app is going to be tested |
| 02:50.50 | bkw_ | its a hack |
| 02:51.00 | bkw_ | fixing the real problem would have been a bigger patch I suspect |
| 02:51.03 | mog_home3 | closest thing we will have to unit testing for a while |
| 02:51.36 | testing234235453 | having problems with FXS calling SIP ... they are in same context ... FXS can call out FXO (on TDM11B) |
| 02:52.00 | testing234235453 | I had a ATA186 and X100P working no problem |
| 02:52.25 | *** join/#asterisk pigpigpig (~pig@165.21.246.202) |
| 02:53.06 | testing234235453 | I get an immediate hangup and reorder |
| 02:54.33 | *** join/#asterisk pdracevich (~bob@smtp.aucklandtax.co.nz) |
| 02:55.32 | pdracevich | a good conferecing, program, that is not Meetme |
| 02:55.58 | Qwell | Was that a question? |
| 02:55.59 | JerJer[mobile] | Meetme |
| 02:56.07 | mikegrb | JerJer[mobile]: I will! |
| 02:56.12 | mikegrb | JerJer[mobile]: Will you marry me? |
| 02:56.17 | Qwell | maybe we're playing jeopardy |
| 02:56.22 | JerJer[mobile] | sorry i'm taken |
| 02:56.53 | pdracevich | do you know of a good conferecing, program, that is not Meetme? *blush* |
| 02:57.10 | JerJer[mobile] | Meetme |
| 02:58.04 | bjohnson | I think it's Yoda |
| 02:58.05 | *** join/#asterisk tessier (~treed@222.253.65.202) |
| 02:58.17 | JerJer[mobile] | Meetme I will |
| 02:58.33 | drumkilla | the only other one that is public is app_conference, i think |
| 02:59.19 | testing234235453 | I know it's something stupid, but 'set debug 10' is not showing me anything |
| 03:00.14 | bjohnson | try set verbose 5 |
| 03:00.16 | *** join/#asterisk Half_Dome (~jelway@mail.westmarkinc.com) |
| 03:00.26 | testing234235453 | I have set verbose 10 on |
| 03:00.28 | testing234235453 | also |
| 03:01.09 | bjohnson | the fxs is using sip? |
| 03:01.36 | bjohnson | what is new compared to what you had working? |
| 03:01.47 | bkw_ | NOT MEETME |
| 03:01.48 | bkw_ | haha |
| 03:01.52 | bkw_ | but you can't have it |
| 03:01.54 | bkw_ | na na ne boo boo |
| 03:02.00 | testing234235453 | no, I'm trying to call a SIP device from a analog off of the FXS |
| 03:02.17 | bjohnson | the fxs is on the pci card? |
| 03:02.18 | testing234235453 | what changed, ATA186/X100P to TDM11B |
| 03:02.40 | bjohnson | you turned 2 pieces of hardware into one? |
| 03:02.57 | SexyKen | exten => 2201,1,Macro(faxreceive) |
| 03:02.57 | SexyKen | exten => h,1,system(/usr/local/sbin/mailfax ${FAXFILE} "ksandell@successfulhosting.com" "${CALLERIDNUM} ${CALLERIDNAME}") |
| 03:03.09 | testing234235453 | ATA186/X100P everything wrked, just couldn't make call waiting work |
| 03:03.10 | SexyKen | Does it make any sense why Asterisk wouldn't execute the hangup extension ? |
| 03:03.30 | bjohnson | SexyKen: not from that |
| 03:03.34 | bkw_ | because h needs to be in the macro |
| 03:03.48 | bjohnson | testing234235453: you're gonna have to explain better |
| 03:03.49 | testing234235453 | bjohnson, yes 2 into 1 |
| 03:04.19 | Half_Dome | If I have an * box with a 4 port FXO card, do my SIP phones have to support 4 lines? |
| 03:04.30 | bjohnson | Half_Dome: no |
| 03:04.35 | Half_Dome | cool |
| 03:04.37 | Half_Dome | thanks |
| 03:04.55 | bjohnson | Half_Dome: they could be one line and you can config * to just pick the first one of the four that is available |
| 03:05.06 | testing234235453 | I have a TDM11b, analog off FXS and PSTN off of FXO. I want my analog to be able to call other SIP devices |
| 03:05.30 | bjohnson | testing234235453: of course you have an analog phone off a fxs .. where is the fxs? |
| 03:05.34 | testing234235453 | it can call out the FXO, and the SIP devices can call it...but no callin into sop |
| 03:05.41 | testing234235453 | TDM11B |
| 03:05.47 | bjohnson | and of course the pstn is off the fxo .. where is the fxo? |
| 03:05.52 | testing234235453 | TDM11B |
| 03:05.57 | *** join/#asterisk TheEmperor (~mattn@203.121.47.100) |
| 03:06.00 | bjohnson | finally |
| 03:06.29 | SexyKen | bkw_, I moved h to the macro and that doesn't execute either. |
| 03:07.09 | testing234235453 | bjohnson, did I explain a bit better? |
| 03:08.10 | bjohnson | did the sip device work before? |
| 03:08.35 | testing234235453 | yes |
| 03:08.44 | testing234235453 | sip to sip works fine |
| 03:08.54 | bjohnson | can the sip dial anything on the pci card? |
| 03:09.33 | bjohnson | ie can it use the fxo but not call the fxs? |
| 03:09.37 | testing234235453 | it can dial out fxo and can dial the analog |
| 03:09.57 | testing234235453 | SIP device that is |
| 03:10.30 | bjohnson | and can dial the analog? the sip device can dial the fxs on the pci AND the fxo on the pci? |
| 03:11.10 | bjohnson | but the fxs on the pci cannot dial the sip device? |
| 03:11.44 | testing234235453 | both are correct statements |
| 03:11.52 | JerJer[mobile] | Dial,SIP/bob |
| 03:12.21 | *** join/#asterisk NormAst (NormAst@toronto-HSE-ppp3959338.sympatico.ca) |
| 03:13.18 | bjohnson | SexyKen: try using a system command like echo "${DATETIME} - ${CALLERID} - ${CHANNEL}" >> /var/log/asterisk/calls to see if the hangup extension is getting invoked at all |
| 03:13.48 | bjohnson | SexyKen: it would be best to see the contents of that macro too .. don't paste it here |
| 03:14.22 | SexyKen | Okay -- sec. |
| 03:14.25 | SexyKen | I'll pastebin it |
| 03:14.27 | bjohnson | JerJer[mobile]: you're very cryptic |
| 03:15.09 | *** join/#asterisk jsharp (~jsharp@65.90.64.82) |
| 03:15.30 | jsharp | Anyone using AMP? Is it useful? |
| 03:15.35 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
| 03:15.35 | *** mode/#asterisk [+o anthm] by ChanServ |
| 03:15.48 | bjohnson | testing234235453: and the context= line in zapata.conf for the fxs is the name of the context that contains the sip extension definition? |
| 03:16.00 | bjohnson | testing234235453: and you've restarted asterisk? |
| 03:16.16 | bjohnson | jsharp: if it does exactly what you want .. it is useful |
| 03:16.18 | SexyKen | bjohnson, http://pastebin.ca/8109 -- that is the relevant data I believe. |
| 03:16.27 | *** join/#asterisk MikeJ_ (~icechat5@pcp02795302pcs.roylok01.mi.comcast.net) |
| 03:16.37 | bjohnson | jsharp: if it does not .. you're immediately into debugging hundreds of lines of scripts |
| 03:16.50 | testing234235453 | bjohnson, I have tried that , but will try again |
| 03:17.13 | blitzrage | http://100777.com/node/1227 |
| 03:18.23 | bjohnson | SexyKen: does it receive the fax correctly ..just doesn't send the email? |
| 03:18.51 | SexyKen | bjohnson, It appears to recieve corrctly |
| 03:18.54 | testing234235453 | bjohnson, no good ... yes did a restart |
| 03:22.32 | *** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.we.client2.attbi.com) |
| 03:23.03 | bjohnson | testing234235453: maybe it's time to call digium support |
| 03:23.35 | bjohnson | SexyKen: try a simpler system command |
| 03:24.48 | testing234235453 | bjohnson, that's for your help |
| 03:25.01 | testing234235453 | that would be "thanks" |
| 03:25.48 | Shido6 | ok |
| 03:25.54 | Shido6 | whats wrong, testing234235453? |
| 03:26.14 | Qwell | Shido6: Thanks for the help earlier. |
| 03:26.30 | Shido6 | Qwell, its what I do. |
| 03:26.43 | bjohnson | SexyKen: this http://www.voip-info.org/wiki-Asterisk+Fax+to+email makes it look like the variables should all be in their own set of quotes |
| 03:27.00 | Shido6 | if it werent for Digium, I would be out on the street with nothing shivering in the freezing canadian weather |
| 03:27.20 | testing234235453 | Shido6, can't make call from TDM11B ananlog to SIP device. EVERYTHING else works |
| 03:27.25 | jsharp | You and me both. Well, except for the canadian part. |
| 03:27.26 | bjohnson | Shido6: in short .. he has a fxs and a fxo on a TDM11b |
| 03:27.28 | Qwell | Shido6: used to work for Digium? |
| 03:27.37 | Shido6 | not to mention JerJer's help :) |
| 03:28.10 | bjohnson | Shido6: everything seems to work except for the fxs on the pci cannot dial a remote sip device |
| 03:28.35 | *** part/#asterisk [0xBoTNeTBusTr] (~bitbuster@67-23-242-78.atlaga.adelphia.net) |
| 03:28.36 | Sedorox | hey slePP you around? |
| 03:28.58 | Shido6 | well |
| 03:29.07 | bjohnson | Shido6: but the sip device CAN call the fxs (and fxo) on the pci card |
| 03:29.07 | Shido6 | Got pastebin? |
| 03:29.44 | bjohnson | no .. he says the zapata.conf context= line points the fxs at a extension that contains the sip exten => |
| 03:29.46 | Shido6 | zaptel.conf , zapata.conf and do a cat /proc/zaptel/* ( if u have a 1 , whats it say, if there's a 2 whats it say, if there's a 3.....) |
| 03:29.49 | Shido6 | oh |
| 03:29.52 | Shido6 | and sip.conf , too |
| 03:30.10 | Shido6 | and kernel version |
| 03:30.14 | NormAst | Any one got the firmware for the AudioCodes MP-108? |
| 03:30.16 | bjohnson | he says other sip devices can call each other .. so the sip system seems to be working |
| 03:30.28 | bjohnson | the pci card is the new piece in the working system |
| 03:30.43 | Shido6 | okie dokie, so what do the zaptel gear configs say |
| 03:30.50 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 03:30.55 | shmaltz | helo every1 |
| 03:31.02 | bjohnson | testing234235453: all of that is correct? |
| 03:31.29 | shmaltz | does' setgroup check group get set on the group when set until exit from extension? or only on active dials? |
| 03:32.14 | testing234235453 | yes, correct ... been a long time .. pastebin a website for pasting files? |
| 03:32.23 | Qwell | testing234235453: pastebin.ca |
| 03:32.26 | bjohnson | ~pastebin |
| 03:32.27 | jbot | [pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 03:32.49 | bjohnson | SexyKen: try moving the quote just before ${CALLERIDNUM} to in front of ${FAXFILE} |
| 03:33.25 | shmaltz | I'm trying to increase the value of a group in the same exten, is this possible? |
| 03:33.32 | *** join/#asterisk Vco (~Vco@S0106080020aa7650.wp.shawcable.net) |
| 03:34.21 | Shido6 | +1 |
| 03:34.42 | Shido6 | updatedb ; locate README.variables |
| 03:34.52 | bjohnson | shmaltz: wiki says SetGroup is same as SetVar(GROUP=group) so I assume you can increment it with setvar |
| 03:35.22 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) [NETSPLIT VICTIM] |
| 03:35.22 | *** join/#asterisk fenlander (~irc@82.152.81.57) [NETSPLIT VICTIM] |
| 03:35.22 | *** join/#asterisk elriah (~jfulcrum@adsl-068-209-198-242.sip.bhm.bellsouth.net) [NETSPLIT VICTIM] |
| 03:35.22 | *** join/#asterisk cftbl (hector@dipsy.tch.org) [NETSPLIT VICTIM] |
| 03:35.22 | *** join/#asterisk roamer323 (~sing@67.71.60.211) [NETSPLIT VICTIM] |
| 03:35.22 | *** join/#asterisk ArkyLady (ArkyLady@h248.76.255.206.cable.htsp.cablelynx.com) [NETSPLIT VICTIM] |
| 03:35.22 | *** join/#asterisk astlog (astlog@cpe-24-58-84-250.twcny.res.rr.com) [NETSPLIT VICTIM] |
| 03:35.22 | *** join/#asterisk _Vile (~vile@90.b160.bendtel.net) [NETSPLIT VICTIM] |
| 03:35.22 | *** join/#asterisk CoolAcid (~jk@216.99.98.39) [NETSPLIT VICTIM] |
| 03:35.28 | shmaltz | bjohnson, but show application says its per channel |
| 03:36.42 | bjohnson | shmaltz: bottom of this http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup makes it look like setgroup actually does the increment |
| 03:37.00 | bjohnson | so exten=>s,1,setgroup(this) |
| 03:37.06 | bjohnson | exten=>s,2,setgroup(this) |
| 03:37.11 | shmaltz | bjohnson, I agree, but maybe just one per channel |
| 03:37.13 | bjohnson | would actually increment it by 2 |
| 03:37.34 | shmaltz | meaning that the same group can never be incremented from the same channel |
| 03:37.38 | shmaltz | twice |
| 03:37.44 | bjohnson | give it a try |
| 03:37.50 | shmaltz | oh, lets see that one |
| 03:38.04 | *** join/#asterisk zhier (~nick@219.137.40.17) |
| 03:38.14 | bjohnson | that example increments 2 groups for the same call |
| 03:38.39 | bjohnson | so I don't know if you can just increment tham group twice for the same cll |
| 03:39.32 | *** join/#asterisk Half_Dome (~jelway@mail.westmarkinc.com) |
| 03:39.39 | shmaltz | bjohnson, but those examples set 2 different groups |
| 03:39.56 | shmaltz | exactly (sorrry didn't see your post) |
| 03:40.04 | shmaltz | :) |
| 03:40.51 | testing234235453 | Shido6, 8110 |
| 03:41.41 | bjohnson | shmaltz: try it and see |
| 03:42.27 | Shido6 | you already know what Im going to say |
| 03:42.34 | Shido6 | but lets start with zapata.conf, testing234235453 |
| 03:42.38 | shmaltz | oh I could try it using my cell and calling in |
| 03:42.38 | Shido6 | where is zapata.conf? :) |
| 03:42.43 | bjohnson | shmaltz: otherwise you maybe just need to manually increment your own variable |
| 03:42.56 | shmaltz | I know, that the obvious |
| 03:43.04 | bjohnson | and use gotif or something based on that value |
| 03:43.10 | bjohnson | but would be messy |
| 03:43.14 | shmaltz | it's just much cleaner to have the app jump for me |
| 03:43.23 | shmaltz | exactly |
| 03:43.24 | testing234235453 | standby |
| 03:43.32 | shmaltz | great minds think alike |
| 03:43.36 | shmaltz | ;) |
| 03:43.37 | kram | and so do yours |
| 03:43.50 | kram | :) |
| 03:44.02 | kram | i always use that line |
| 03:44.08 | kram | "great minds think alike, and so do ours" |
| 03:44.16 | *** join/#asterisk kulp (~kulp@generic-net216-173.mtc.net) |
| 03:44.56 | shmaltz | kram, not bad:) |
| 03:45.02 | testing234235453 | Shido6, 8111 |
| 03:45.15 | testing234235453 | zapata @ the bottom |
| 03:45.42 | brc_ | hey testing234235453 type /nick a_clever_nick_name_here |
| 03:45.51 | Shido6 | um |
| 03:46.20 | Shido6 | ok |
| 03:46.27 | Shido6 | now [default] |
| 03:46.30 | Shido6 | from extensions.conf |
| 03:50.13 | testing234235453 | SHido6, 8112 |
| 03:51.02 | elriah | Hey guys, how do I pass caller id when someone hits my voicemail extension so they don't get prompted for their mailbox? |
| 03:52.11 | Shido6 | ok |
| 03:52.13 | Shido6 | look |
| 03:52.14 | Shido6 | testing234235453 |
| 03:52.17 | Shido6 | honestly... |
| 03:52.24 | Shido6 | if you dont believe me |
| 03:52.31 | Shido6 | just change one of your sip extensions |
| 03:52.35 | Shido6 | from friend |
| 03:52.37 | Shido6 | to user and peer |
| 03:52.50 | Shido6 | your info is good, but take out the host for the user |
| 03:52.55 | Shido6 | and take out the context for the peer |
| 03:53.02 | *** join/#asterisk sandnigg0r (~crunkuser@66-55-197-254.gwi.net) |
| 03:53.07 | Shido6 | and refer to the table in /usr/src/asterisk/configs/sip.conf.sample |
| 03:53.11 | Shido6 | to find out what users can use |
| 03:53.15 | Shido6 | and what peers can use |
| 03:53.16 | Shido6 | then reload |
| 03:53.19 | Shido6 | and ........... BAM!!!!!!!!! |
| 03:53.25 | Shido6 | you're all set to go |
| 03:53.43 | Shido6 | elriah |
| 03:53.54 | Shido6 | like on a 7960 message_uri: ? |
| 03:54.49 | Shido6 | check out http://pastebin.ca/8113 |
| 03:54.57 | JerJer[mobile] | calm down |
| 03:55.04 | JerJer[mobile] | wholy crazy messages batman |
| 03:55.54 | elriah | I've seen the example before - just can't find it, it had $CALLERID or something similiar. So when I hit my voicemail button, which goes to extension 8500, the caller id passed automatically bypasses the prompt to enter the extension in * |
| 03:56.21 | shmaltz | bjohnson, http://pastebin.ca/8114 |
| 03:56.23 | Shido6 | checkout the pastebin, elriah... |
| 03:57.06 | Shido6 | messages_uri: 693000 where 3000 is my extension |
| 03:57.12 | *** join/#asterisk chaoscon (~ph33r@chaoscon.user) |
| 03:57.52 | elriah | Shido6: I see it - but I'm not sure this is what I was after... Let me google it and see if I can track it down |
| 03:58.00 | elriah | Thanks, btw, again!! |
| 03:58.04 | elriah | (Shido6 is very helpful) |
| 03:58.20 | Shido6 | i press my messages button |
| 03:58.24 | *** join/#asterisk tessier (~treed@222.253.65.202) |
| 03:58.27 | Shido6 | and she asks me for my password |
| 03:58.48 | *** part/#asterisk Half_Dome (~jelway@mail.westmarkinc.com) |
| 03:58.56 | Shido6 | I also have a mailbox=3000 in sip.conf for my phone(s) |
| 03:59.11 | Shido6 | for MWI |
| 03:59.26 | elriah | Yea - that's it, maybe two ways to get to the same thing? |
| 03:59.26 | elriah | Well, I'm still fighting the MWI issue. |
| 03:59.26 | elriah | On my polycom phones. |
| 04:02.13 | shmaltz | anybody interested in looking at this: |
| 04:02.14 | shmaltz | http://pastebin.ca/8114 |
| 04:03.16 | SexyKen | How do I delete all mails in a mailbox using 'mail' in the cli |
| 04:03.45 | elriah | For some reason, on my polycom ip500's, when I do sip show peers, the "Host" shows "(unspecified)". Shouldn't it show the IP? |
| 04:04.22 | Shido6 | if they are configured properly, yes |
| 04:04.23 | dwmw2_gone | this is definitely the wrong answer. We need AST_FORMAT_SLINEAR_WRONGENDIAN. |
| 04:04.31 | dwmw2_gone | or _LE and _BE |
| 04:04.44 | elriah | Hrm... |
| 04:05.12 | jsharp | Or we could just do the linux thing and say it only runs on i386. |
| 04:05.34 | dwmw2_gone | but that's no fun |
| 04:06.01 | dwmw2_gone | there's not a lot of real Linux software nowadays that doesn't run on PPC |
| 04:06.20 | dwmw2_gone | asterisk is mostly ok |
| 04:09.18 | *** join/#asterisk roamer323 (~sing@HSE-Toronto-ppp130667.sympatico.ca) |
| 04:10.06 | shmaltz | bjohnson, check this out |
| 04:10.16 | shmaltz | guess what it shows on each groupcount? |
| 04:10.19 | SexyKen | mailfax returns this error: Malformed UTF-8 character (unexpected non-continuation byte 0x10, immediately after start byte 0xc3) in transliteration (tr///) at /usr/bin/mime-construct line 198. |
| 04:10.19 | SexyKen | Malformed UTF-8 character (unexpected continuation byte 0x98, with no preceding start byte) in transliteration (tr///) at /usr/bin/m |
| 04:10.20 | shmaltz | http://pastebin.ca/8115 |
| 04:10.23 | SexyKen | Anyone know why this'd happen? |
| 04:10.35 | elriah | Shido6: In my sip.conf, do I need a username and secret to get the ip to register with asterisk? |
| 04:12.49 | elriah | Or anyone... |
| 04:13.14 | testing234235453 | Shido6, made the change 8116...reloaded ... nope Am I still missing something? |
| 04:13.52 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 04:18.49 | SexyKen | Malformed UTF-8 character (unexpected continuation byte 0x98, with no preceding start byte) in transliteration (tr///) at /usr/bin/m |
| 04:18.52 | SexyKen | Anyone know why this'd happen? |
| 04:18.55 | SexyKen | With mailfax |
| 04:19.22 | testing234235453 | MMMAAAANNNN, it didn't like me hitting # to skip the interdigit timeout |
| 04:19.36 | testing234235453 | it works without the SIP changes |
| 04:22.12 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 04:22.39 | asingh | Any one knows about Asterisk+dialogic howto ? |
| 04:23.50 | FuriousGeorge | hi all. i was looking at some example extension.conf files, and i got to wondering if it would be difficult to set up huntgroups? for instance, how difficult would it be to have all the phones ring if caller hits 0 |
| 04:24.14 | shmaltz | bjohnson the wiki is wrong |
| 04:24.31 | shmaltz | look at this: |
| 04:24.32 | shmaltz | http://bugs.digium.com/bug_view_page.php?bug_id=0003067 |
| 04:29.43 | *** join/#asterisk t3t (~t3t@galley.pangalacticgargleblaster.com) |
| 04:30.18 | Shido6 | err |
| 04:30.19 | Shido6 | what? |
| 04:30.21 | Shido6 | back |
| 04:30.46 | afrosheen | so about the meetme delay, is there a patch to fix that or do I just have to use the quiet switch |
| 04:31.36 | *** join/#asterisk KirkL (~me@c-24-22-57-111.client.comcast.net) |
| 04:34.20 | *** join/#asterisk chaoscon (~ph33r@chaoscon.user) |
| 04:50.12 | *** join/#asterisk w0w0 (~w0w0@80.26.162.27) |
| 04:51.28 | elriah | Is there any control over how the directory spells out users names? |
| 04:51.45 | Shido6 | "/etc/asterisk/voicemail.conf" |
| 04:52.23 | elriah | tnx - Shido6, any experience with these polycom ip500 phones? |
| 04:53.21 | *** join/#asterisk Rival (~rival@66.177.249.219) |
| 04:54.19 | *** join/#asterisk nix000 (~nix000@66.11.188.165) |
| 04:56.06 | Supaplex | yehou |
| 05:00.54 | elriah | Is there a way to have asterisk voicemail call you on say a mobile phone to let you know you have a new voicemail? |
| 05:01.30 | Sedorox | elriah: it can send sms |
| 05:01.48 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 05:01.48 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 05:02.06 | elriah | Yea, I got the email notifications working. But it would be cool if it just called my cell phone. Oh well, maybe v2. |
| 05:02.19 | Sedorox | hmmm |
| 05:02.34 | shmaltz | bjohnson, you still around? |
| 05:11.37 | elriah | HELL YES PRAISE D'LORD I GOT MY MWI WORKING!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! |
| 05:11.50 | elriah | All the faqs and wikis are wrong for the latest firmware. |
| 05:12.09 | IQ | elriah: what u got working :) ? |
| 05:12.21 | elriah | MWI on the polycom IP 500 |
| 05:12.33 | IQ | elriah: so when are you throwing the party ;) ? |
| 05:12.50 | Rival | can anyone tell me why i can make outgoing calls but not recieve them with my teliax account |
| 05:12.54 | Rival | it should be working |
| 05:13.00 | t3t | elriah: http://www.voip-info.org/wiki-Asterisk+tips+callback |
| 05:13.13 | t3t | elriah: congats on getting MWI working |
| 05:13.54 | elriah | Hey - tnx for the url.. I need to get some freakin' sleep, I've been up til 2'am the past three nights working on my * prototype |
| 05:13.58 | t3t | elriah: what's wrong with the faqs/wikis? |
| 05:14.19 | IQ | what is MWI? |
| 05:14.29 | t3t | message waiting indication |
| 05:14.57 | IQ | oh i c |
| 05:15.21 | Rival | anyone here use teliax? |
| 05:17.07 | shmaltz | elriah, you should be able to get callback working by setting the option in voicemail.conf |
| 05:17.18 | shmaltz | see the commentsin voicemail.conf |
| 05:17.39 | *** part/#asterisk xarg (~Administr@ool-4354c55c.dyn.optonline.net) |
| 05:19.05 | niZon | does MWI work on the cisco phones? |
| 05:19.40 | Juggie | yes |
| 05:19.46 | shmaltz | niZon, of course |
| 05:19.54 | IQ | Hi... anyone got a small IVR script for * ? |
| 05:19.55 | t3t | shmaltz: where is callback in voicemail.conf? Do you mean 'press x to call this person back'? |
| 05:20.18 | t3t | IQ: like what? |
| 05:20.25 | shmaltz | t3t, nope, there is another option, perhaps its called something else |
| 05:20.47 | nix000 | anyone used the mfc/r2 stuff in here ? |
| 05:20.57 | t3t | shmaltz: thanks. I didn't see it, but that doesn't mean much at this hour |
| 05:21.01 | IQ | t3t: Like play a prompt, get dtmf input and make a decesion, etc. ? |
| 05:21.16 | t3t | s,1,Answer() |
| 05:21.31 | t3t | s,2,Background(<promptFile>) |
| 05:21.53 | t3t | #,1,DoSomething |
| 05:21.55 | t3t | etc |
| 05:21.57 | t3t | real simple |
| 05:22.25 | IQ | thanks, good starting point :) ... maybe I can copy the 500 sample script that comes with * |
| 05:22.37 | t3t | IQ: after the s extension you want to put the digits that you want people to be able to press |
| 05:23.01 | shmaltz | t3t, elriah, my mistake |
| 05:23.02 | t3t | IQ: it's good to have all of your based covered though... like h,t,i,etc |
| 05:23.21 | t3t | shmaltz: no problem. that would have been a pleasant surprise though |
| 05:23.24 | shmaltz | confused dialout and callback as either one being that, actualy neither is |
| 05:23.37 | niZon | has anyone used chan_sccp? http://chan-sccp.sourceforge.net/ |
| 05:23.42 | IQ | t3t: okay, I'll do that --- thanks |
| 05:23.44 | t3t | shmaltz: i've done worse |
| 05:23.46 | shmaltz | however using .call files it's possible to do it |
| 05:23.50 | shmaltz | :) |
| 05:24.32 | t3t | niZon: never used it, from what i've read, it works but i wouldn't depend on it... you can't use sip? |
| 05:25.12 | niZon | Just checking out my options, I'm looking into getting a cisco 7940/60 |
| 05:25.18 | t3t | use sip |
| 05:25.31 | t3t | the firmware upgrade can be a pain, but it's probably worth it |
| 05:25.38 | SexyKen | Malformed UTF-8 character (unexpected continuation byte 0x98, with no preceding start byte) in transliteration (tr///) at /usr/bin/m |
| 05:25.40 | SexyKen | Anyone know why this'd happen? |
| 05:25.42 | niZon | it looks like it might be a pain to get the firmware as well |
| 05:26.06 | shmaltz | niZon, I love those Cisco phones |
| 05:26.14 | t3t | niZon: w/o a support contract it's nearly impossible to do it legitimately |
| 05:26.21 | niZon | shmaltz: They look nice :P |
| 05:26.36 | niZon | t3t: I'm told those are pricey... |
| 05:26.40 | t3t | niZon: get a polycom, they work nice and look OK |
| 05:26.47 | shmaltz | t3t, niZon, in most cases your reseller will be able to give you the image |
| 05:26.48 | niZon | hm |
| 05:26.49 | t3t | niZon: wouldn't know |
| 05:27.02 | niZon | I saw a few with SIP on ebay |
| 05:27.08 | niZon | as well as the SIP images |
| 05:27.10 | IQ | Any SMS service provider to use with * ? |
| 05:27.30 | shmaltz | I bought one Cisco phone from an official Cisco sales partner for $350+ only b/c I wanted that CCO account |
| 05:27.39 | t3t | niZon: make sure it's v6+ w/SIP... the older revisions had even more bugs than usual with cisco |
| 05:27.39 | shmaltz | IQ, which country? |
| 05:28.18 | IQ | shmaltz: international would be better. But to start with US or Europe will work |
| 05:28.30 | shmaltz | for the US you don't need one. |
| 05:28.40 | niZon | t3t: I'll keep that in mind |
| 05:28.55 | shmaltz | for Europe check the list the last 2 months there was lots of talk about it |
| 05:29.11 | IQ | shmaltz: I'll check the list - btw how do u do US ? |
| 05:29.16 | t3t | SexyKen: try the error text before the char code in google |
| 05:29.34 | shmaltz | in the US every provider has an email gateway to support email - > SMS |
| 05:29.50 | shmaltz | verizon = phonenumber@vtext.com |
| 05:30.07 | IQ | ya like number@messages.sprintpcs.com |
| 05:30.07 | shmaltz | sprint = phonenumber@messaging.sprintpcs.com |
| 05:30.16 | IQ | but what about SMS -> * |
| 05:30.29 | shmaltz | I don't know the address of the others, but I know they all have one |
| 05:30.51 | shmaltz | just use an email in you phone when using sms, yep it works on every sms capable phone on the US market |
| 05:30.52 | cobryce | IQ: IIRC, you need a GSM modem and connection with a cell phone network for starters |
| 05:31.04 | shmaltz | cobryce, not in the US |
| 05:31.18 | *** join/#asterisk trig_hm (~jb@home.monkeypr0n.org) |
| 05:31.23 | cobryce | shmaltz: He does if he doesn't want to do the e-mail thing |
| 05:31.26 | afrosheen | shmaltz: what do you need in the US? |
| 05:31.47 | shmaltz | just email |
| 05:31.53 | cobryce | In the US, and I suspect the world abroad, you can send email using the provider's gateway |
| 05:32.15 | cobryce | That just leaves delivering mail to Asterisk |
| 05:32.16 | shmaltz | the opposite send SMS using the providers gateway |
| 05:32.23 | IQ | yeah, sending part we can use email. but being able to receive is important |
| 05:32.26 | shmaltz | cobryce, nope it doesn't work abroad |
| 05:32.40 | shmaltz | IQ, in the US you can receive |
| 05:33.02 | shmaltz | abroad they don't let you send emails just from SMS, you need a data plan |
| 05:33.12 | cobryce | Interesting |
| 05:33.17 | shmaltz | and most SMS phones don't allow you |
| 05:33.33 | IQ | shmaltz: yeah, I got your point, like I can send emails from my sprint phone. but this won't be a real SMS, right? |
| 05:33.49 | shmaltz | IQ, it will |
| 05:33.52 | shmaltz | if you use SMS |
| 05:33.56 | cobryce | Not as Asterisk sees it. |
| 05:33.58 | shmaltz | I will repeat |
| 05:34.17 | shmaltz | every US phone that can send SMS can send SMS to an email |
| 05:34.20 | cobryce | You send SMS, e-mail address receives the message. |
| 05:34.26 | shmaltz | not email to email, but SMS to email |
| 05:34.34 | shmaltz | cobryce, exactly |
| 05:34.57 | cobryce | The issue now is the receipt and handling of e-mail by Asterisk |
| 05:34.57 | shmaltz | IQ, you in the US? |
| 05:35.09 | IQ | shmaltz: yes, I am |
| 05:35.19 | shmaltz | cobryce, for that one will need to setup MX records |
| 05:35.23 | IQ | shmaltz: there are two boxes - one for Phone Number and other for E-Mail ID |
| 05:35.29 | shmaltz | IQ, do you have a cell phone? |
| 05:35.56 | IQ | shmaltz: yes SprintPCS. and when I compose a text message I can either put a Phone Number or E-Mail ID to send text message |
| 05:36.00 | shmaltz | IQ, what do you mean one for phone number and the other for email? |
| 05:36.14 | shmaltz | IQ, so what don't you understand? |
| 05:36.27 | afrosheen | hmm my at&t phone asks for a mail server address |
| 05:36.33 | afrosheen | or number |
| 05:36.50 | shmaltz | afrosheen, mail server (as in SMTP)? or email address? |
| 05:37.04 | afrosheen | when I try sms email |
| 05:37.05 | shmaltz | gn |
| 05:37.06 | IQ | shmaltz: I can send SMS to my Yahoo account using Sprint Phone. But is this a real SMS? |
| 05:37.24 | shmaltz | IQ, of course |
| 05:37.45 | shmaltz | you can even send back, using the from address |
| 05:37.46 | afrosheen | it's asking for "email server number" and I can browse my contacts to enter it |
| 05:38.09 | *** join/#asterisk techie (gus@antibala.com) |
| 05:38.15 | afrosheen | weak |
| 05:38.15 | shmaltz | afrosheen, go ahead and put in your email address. you might have to create a contact? maybe? |
| 05:38.28 | afrosheen | I think it's asking for the sms gateway |
| 05:38.29 | IQ | shmaltz: Actually, what I am looking for is being able to register as SMS receiver. So pepole can SMS me using DID not email ID |
| 05:38.37 | afrosheen | I've never dealt with sms on my phone, only text messaging |
| 05:39.02 | shmaltz | IQ, and it will arrive where? |
| 05:39.12 | IQ | * |
| 05:39.17 | shmaltz | afrosheen, sms and text is the same |
| 05:39.32 | SexyKen | Is there anyway to set the 'from' field for Mailfax |
| 05:39.39 | shmaltz | IQ, where in *? how will the recipient read it? |
| 05:39.47 | niZon | bah just write an email to * gateway :P |
| 05:40.02 | IQ | Note: Some VoIP phones (like the SNOM products) support SMS. |
| 05:40.03 | shmaltz | niZon, don't jump he might not have to |
| 05:40.03 | afrosheen | but text asks for a destination contact or phone number ..fark it |
| 05:40.14 | IQ | http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Sms |
| 05:40.22 | shmaltz | afrosheen, where are you? in th US? |
| 05:40.25 | afrosheen | yeah |
| 05:41.21 | nix000 | anyone know what is the story behind asterisk and dialogic support ? |
| 05:41.53 | IQ | nix000: I think - might have to pay to Digium to buy per port software license |
| 05:42.42 | *** join/#asterisk B4 (~B4@202.69.48.245) |
| 05:42.45 | nix000 | IQ, do they support the latest cards .. specially the ss7 ones ? asterisk.org mentions some. i do not think support ss7. |
| 05:42.48 | *** join/#asterisk Los415 (~los415@c-24-126-63-233.we.client2.attbi.com) |
| 05:43.28 | B4 | ~seen zx81 |
| 05:43.30 | jbot | zx81 <matt@222-153-16-58.jetstream.xtra.co.nz> was last seen on IRC in channel #asterisk, 10d 5h 18m 30s ago, saying: 'nevermind'. |
| 05:43.31 | IQ | nix000: I dont have much information. you might have to call them in the morning :) |
| 05:44.44 | afrosheen | !seen bentley |
| 05:44.44 | chetan | Bentley!~Bentley@S01060080c8135e6a.cg.shawcable.net is currently on #asterisk |
| 05:44.49 | afrosheen | Bentley: hey |
| 05:48.06 | shmaltz | for starters |
| 05:48.08 | shmaltz | http://www.smsclient.org/providers.php |
| 05:48.22 | shmaltz | look at the list how different it is implemented in the US |
| 05:48.31 | shmaltz | use email it's much easier and cleaner |
| 05:48.44 | IQ | and free :) |
| 05:49.05 | shmaltz | now this guy has got a product that is obsolete in the US market |
| 05:49.06 | shmaltz | http://www.bayhamsystems.com/asterisk.html |
| 05:49.16 | shmaltz | unelss one want's international sms |
| 05:49.27 | shmaltz | reminds me of a story |
| 05:49.36 | *** join/#asterisk clive- (~pirch@rndf-146-44-91.telkomadsl.co.za) |
| 05:49.38 | shmaltz | 2 yrs ago at the Cbit Show in NY |
| 05:50.03 | afrosheen | here we go with the cebit story |
| 05:50.03 | shmaltz | I met this guy a rep from some Chek comapny |
| 05:50.13 | clive- | shmaltz , ..nice nick |
| 05:50.28 | shmaltz | clive-, who are you? |
| 05:50.38 | shmaltz | he sells GSM gateways |
| 05:50.48 | shmaltz | so I ask him whats this for? |
| 05:51.03 | shmaltz | so he tells me you can save tons of money on call to mobiles? |
| 05:51.09 | shmaltz | so I ask him how so |
| 05:51.35 | shmaltz | so he goes on to explain that I get an account with a gsm provider and blah blah blah |
| 05:51.55 | IQ | did you buy one :) ? |
| 05:52.00 | shmaltz | which is very usefull and popular abroad where you pay more to call a gsm network then to call china |
| 05:52.03 | clive- | schmaltz thats big business here in south africa |
| 05:52.04 | shmaltz | of couse nto |
| 05:52.22 | afrosheen | you said 'for what do I need to save money here' |
| 05:52.41 | shmaltz | so I explained to him that in the US one doesn't pay to call a cell phone, in fact it is very hard to say if one is calling a cell phone or landline |
| 05:53.02 | shmaltz | he looked at me like I'm off the moon, |
| 05:53.14 | IQ | unless you get the voice mail that says "You've reached Sprint PCS voice mail box of XXXXXX" |
| 05:53.18 | shmaltz | what it costs the same to call a cell phone then landline? |
| 05:53.19 | afrosheen | *blink blink* |
| 05:53.46 | afrosheen | yeah if your call accidentally sounds good you know it's a landlilne |
| 05:53.53 | shmaltz | thats what he asked me, so I told him, next time you make a trip to the US make sure you do your home work |
| 05:54.04 | Rival | anyone here use teliax? |
| 05:54.06 | shmaltz | afrosheen :) |
| 05:54.23 | IQ | yeah, I think it is only in US that you dont have to pay extra to call cell. in Europe, Asia and Mid-East you pay extra if you call a cell phone |
| 05:54.29 | afrosheen | yep |
| 05:54.30 | IQ | Rival: what is teliax? |
| 05:54.38 | shmaltz | yep |
| 05:54.41 | afrosheen | alot of iax trunk providers reflect that in their price lists |
| 05:54.47 | Rival | a voip termination provider |
| 05:54.52 | shmaltz | ok, to get back to topic |
| 05:55.08 | afrosheen | werd |
| 05:55.20 | shmaltz | SMS from within * will only work in coutries where they allow sms over analog |
| 05:55.40 | IQ | unless you have T1/ISDN ? |
| 05:55.47 | afrosheen | south korea: 7 cents a minute. north korea: 79 cents a minute. lesson? communism is expensive. |
| 05:56.03 | Supaplex | lol |
| 05:56.08 | shmaltz | afrosheen lol |
| 05:56.10 | IQ | afrosheen: check Afghanistan |
| 05:56.21 | Supaplex | 24.95/min ;) |
| 05:56.38 | afrosheen | yeah |
| 05:56.44 | shmaltz | IQ, afghanistan, I can still understand, they are lacking a decent infarstructure |
| 05:56.46 | afrosheen | their copper wiring is hard to come by |
| 05:56.53 | Moc | Damn some provider think that their client are stupid.. |
| 05:56.58 | afrosheen | most of it gets ripped up and used for remote tnt detonation |
| 05:57.07 | yxa | how does SMS over analog work? |
| 05:57.07 | IQ | shmaltz: yeah, thats correct. they're still suffering :( |
| 05:57.09 | Supaplex | Moc: some of them don't think. they know! |
| 05:57.10 | afrosheen | those boxes with the little handles you push down on |
| 05:57.13 | Moc | he trying to justify me a 4 hours Redhat Enterprise Minimal Install is NORMAL!!!! |
| 05:57.18 | afrosheen | jeeez |
| 05:57.28 | afrosheen | that's like a solaris 'vacation' install |
| 05:57.31 | shmaltz | yxa, look it up on the wiki |
| 05:57.37 | Moc | I mean, 10min and it should be running and UPDATED !!! |
| 05:57.40 | denon | he probably figures anyone who wants to install redhat will buy anything he says :) |
| 05:57.49 | afrosheen | RHEL and Solaris installs are consultants wet dreams |
| 05:57.59 | Moc | denon(it actually CentOS, but people understand more rhel ;) |
| 05:58.00 | afrosheen | $200 per hour OH YEAH |
| 05:58.17 | afrosheen | then the kool aid guy breaks through the wall |
| 05:58.24 | IQ | any cheap Service Provider for Mid-East ? |
| 05:58.26 | denon | Moc: hehe |
| 05:58.41 | shmaltz | http://www.voip-info.org/wiki-Asterisk+cmd+sms |
| 05:59.03 | shmaltz | IQ, i ws looking for the same, havn't found anybody |
| 05:59.09 | IQ | shmaltz: can we do it if we have T1, ISDN? |
| 05:59.10 | Moc | I mean if I were to install box everyday... I would have a local copy of centos updates !!! |
| 05:59.20 | Moc | he saying the update take sometime up to 6 hours |
| 05:59.26 | IQ | shmaltz: I need a cheap SP for Saudi Arabia |
| 05:59.56 | afrosheen | a 6 hour update huih |
| 06:00.05 | afrosheen | is that like 3 gigs of packages over dsl |
| 06:00.05 | shmaltz | IQ, you muslim? |
| 06:00.10 | shmaltz | I'm jewish |
| 06:00.15 | shmaltz | ;p |
| 06:00.16 | IQ | shmaltz: Yes :) |
| 06:00.22 | afrosheen | oh no |
| 06:00.25 | afrosheen | don't introduce all that |
| 06:00.30 | shmaltz | wow, a jew helping a muslim |
| 06:00.33 | shmaltz | :) |
| 06:00.34 | afrosheen | no no |
| 06:00.42 | IQ | shmaltz: I thought I was helping you :P |
| 06:01.06 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
| 06:01.09 | shmaltz | lol |
| 06:01.27 | IQ | I got lots of friends fro Palestine and Israeil - its hard to tell the difference |
| 06:01.32 | shmaltz | I'm looking for Israel |
| 06:01.58 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 06:02.09 | IQ | Beautiful Place |
| 06:02.20 | afrosheen | weird laws there |
| 06:02.38 | niZon | hey it's BoRiS |
| 06:02.39 | niZon | :P |
| 06:02.44 | SexyKen | Does anyone know how to send faxes with Asterisk/SpanDSP? |
| 06:02.45 | afrosheen | yeltsin? |
| 06:02.46 | elriah | Hey guys, back. I can't seem to dial out my Zap/1 interface from any of my hard phones, any ideas? |
| 06:02.56 | elriah | x100p, it receives calls fine. |
| 06:03.27 | *** join/#asterisk tessier (~treed@222.253.65.202) |
| 06:03.59 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 06:05.42 | *** join/#asterisk marc324 (~marc32344@69-90-241-15.dsl.teksavvy.com) |
| 06:06.52 | IQ | shmaltz: what u use to call israel? |
| 06:07.06 | shmaltz | right now my landline |
| 06:07.11 | shmaltz | it's the cheapest |
| 06:07.14 | afrosheen | shmaltz: have relatives there? |
| 06:07.20 | shmaltz | 4.5 cents a min |
| 06:07.27 | yxa | shmaltz i cant find sms over analog over at the wiki |
| 06:07.30 | shmaltz | yep, 4 siblings |
| 06:07.31 | IQ | shmaltz: are you in US ? |
| 06:07.42 | shmaltz | yep im in the US |
| 06:07.51 | shmaltz | my wife has one sis there |
| 06:07.58 | afrosheen | shmaltz: if they have broadband, send them an iAXY and make an extension for them |
| 06:08.11 | IQ | so it cost less to call Israel from US than calling US from US :) |
| 06:08.37 | shmaltz | afrosheen, but tehy don't have internet |
| 06:08.42 | shmaltz | we were thinking about it |
| 06:08.53 | shmaltz | IQ, why? |
| 06:08.58 | shmaltz | I pay 2.9 cents |
| 06:09.07 | shmaltz | in US Canada and UK |
| 06:09.23 | afrosheen | no internet... |
| 06:09.27 | afrosheen | how do they live |
| 06:09.29 | shmaltz | 3.9 Cents to Western Europe (but UK) and australia |
| 06:09.45 | IQ | not bad. Lingo is a good choise too. Free US, Canada and Europe |
| 06:09.51 | shmaltz | afrosheen, you know what ultra orthodox means? |
| 06:10.04 | afrosheen | ultra scary? |
| 06:10.21 | afrosheen | can't call them on sunday :) |
| 06:10.38 | shmaltz | no make that saturday |
| 06:10.47 | shmaltz | I"m one of those untra scary ones |
| 06:10.55 | shmaltz | so are others on this channel |
| 06:11.01 | IQ | Friday for Muslims :) |
| 06:11.15 | shmaltz | so we have a 3 day rest on this channel ;) |
| 06:11.21 | IQ | lool |
| 06:11.24 | afrosheen | that would explain the slowness |
| 06:11.49 | afrosheen | 'it's shabbatz! I'm not even supposed to be driving, Dude!' |
| 06:11.58 | afrosheen | big lebowski :) |
| 06:11.58 | shmaltz | exactly |
| 06:12.07 | shmaltz | tomorrow night is a big holiday |
| 06:12.20 | shmaltz | afrosheen, you lubavitch? |
| 06:12.50 | afrosheen | nope, not jewish in the least |
| 06:13.08 | jakepdev | anyone ever hear of DS1FD trunks? |
| 06:13.08 | afrosheen | maybe a little ashkenazi blood somehow but who knows |
| 06:13.13 | shmaltz | where you in the US? afrosheen |
| 06:13.13 | BoRiS | hi niZon! |
| 06:13.19 | afrosheen | texas |
| 06:13.23 | afrosheen | the really big state |
| 06:13.23 | shmaltz | ~ashkenazic |
| 06:13.24 | jbot | ashkenazic is probably the oposite of sepharad |
| 06:13.34 | shmaltz | texas? |
| 06:13.37 | shmaltz | or ca |
| 06:13.39 | shmaltz | ? |
| 06:13.45 | shmaltz | or maybe NY |
| 06:13.48 | afrosheen | read ^^ |
| 06:14.02 | shmaltz | whats that? |
| 06:14.02 | IQ | shmaltz: u in NY ? |
| 06:14.53 | IQ | New York |
| 06:14.57 | shmaltz | nope NJ |
| 06:15.05 | shmaltz | afrosheen where you ? |
| 06:15.16 | shmaltz | IQ, you ? |
| 06:15.22 | afrosheen | TEXAS |
| 06:15.26 | afrosheen | CAN YOU SEE THIS TEXT |
| 06:15.27 | IQ | Omaha, Nebraska |
| 06:15.50 | shmaltz | sorry |
| 06:15.58 | shmaltz | ~sphard |
| 06:16.04 | shmaltz | ~saphard |
| 06:16.12 | shmaltz | ~spharad |
| 06:16.13 | jbot | from memory, spharad is some locality unknown. The modern Jews think that Spain is meant, and hence they designate the Spanish Jews "Sephardim," as they do the German Jews by the name "Ashkenazim," because the rabbis call Germany Ashkenaz. Others identify it with Sardis, the capital of Lydia. |
| 06:17.07 | afrosheen | weird, so hitler got Nazi from a jewish word? |
| 06:17.14 | BoRiS | Whats up NiZon? |
| 06:17.56 | shmaltz | afrosheen, who knows |
| 06:18.13 | IQ | anyone remember those VoIP Phones with built in modems? |
| 06:18.55 | *** join/#asterisk SplasPood (jwb@paravolve.net) |
| 06:18.55 | clive- | IQ: Komodo phones |
| 06:19.49 | shmaltz | yxa, you got that page? about SMS? |
| 06:21.29 | IQ | clive-: can we still use them? |
| 06:23.56 | debaser | afrosheen: nazi is an abbreviation of 'national socialist german workers' party' |
| 06:24.30 | clive- | IQ:, the Komodo phones I have still work, only -problem is they only will talk to net2phone, who are really a pain to deal with if you live in south africa, they keeop telling you your creduit card is fraudulent |
| 06:25.38 | IQ | clive-: I was thinking of getting something like this for my mom. She does now know how to use PC and doesn't have broadband in her area |
| 06:26.53 | nix000 | anyone tried interfacing dialogic/intel card via sip ? |
| 06:27.29 | clive- | IQ thye have a verison called yapjack - (net2phone) |
| 06:28.58 | *** join/#asterisk Mazda-MX5 (~root@220-130-142-43.HINET-IP.hinet.net) |
| 06:29.08 | Mazda-MX5 | hi, all |
| 06:29.14 | dersteer | http://www.unleadedjokes.com/html/Double-Parked.html |
| 06:29.19 | IQ | clive-: thanks :) |
| 06:29.38 | afrosheen | debaser: yeah I just googled it myself |
| 06:30.21 | Mazda-MX5 | sorry, I have stupid question, who to add a accout in SIP server ? edit sip.conf ?? |
| 06:30.25 | IQ | this looks like a parking gurage. how do they do that :O |
| 06:30.51 | SexyKen | Doesn't anyone know anything about mime construct |
| 06:30.59 | IQ | Mazda-MX5: thats how I add SIP users |
| 06:31.21 | Mazda-MX5 | IQ , what mean ? |
| 06:31.44 | jsharp | I just unboxed a Sipura 841 and discovered that it was missing the entire main PC board. It was just a plastic box. |
| 06:31.45 | Mazda-MX5 | your mean is edit sip.conf to add a SIP user ? |
| 06:31.58 | IQ | Mazda-MX5: yes |
| 06:32.17 | IQ | jsharp: you must be one of those lucky people who receive a tested dell machine without RAM :) |
| 06:32.17 | Mazda-MX5 | IQ , thank you~~ |
| 06:32.50 | Mazda-MX5 | SIP user should is SIP:name@ip .. |
| 06:33.04 | IQ | jsharp: I got my SPA-3000 yesterday. and I'm really impressed with what it does. There is not much documentation but it is something |
| 06:33.07 | shmaltz | jsharp, who was the reseller? |
| 06:33.26 | IQ | Mazda-MX5: sorry, didn't get your question |
| 06:33.28 | shmaltz | IQ, in my opinion the SPA-3000 is the best ATA |
| 06:33.52 | Mazda-MX5 | SIP user format should is SIP:name@ip? |
| 06:33.54 | IQ | shmaltz: yes - its the best I've used so far. |
| 06:33.56 | jsharp | Does the SPA-3000 support T.38? |
| 06:34.12 | shmaltz | yep, jsharp |
| 06:34.47 | IQ | Mazda-MX5: dont you have the sample sip.conf? Just copy/paste one of the existing user and change it accordingly |
| 06:35.09 | Mazda-MX5 | ok , I will do it. IQ, thank you. |
| 06:35.58 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 06:36.08 | Mazda-MX5 | I buy the "VoIP teleph?? for asterisk" book form amazon yesterday. |
| 06:36.10 | IQ | it runs about $99, its so compact makes my D-Link look dumb |
| 06:36.57 | IQ | Mazda-MX5: thats great. Also read http://www.voip-info.org |
| 06:37.27 | IQ | shmaltz: used IAXy ? |
| 06:37.47 | Mazda-MX5 | thank you , I am reading "voip-info.org" |
| 06:38.02 | shmaltz | IQ, nope not yet |
| 06:40.06 | IQ | how long it takes to tall asleep after taking sleeping pills? |
| 06:40.15 | jsharp | 30-40 minutes. |
| 06:40.30 | shmaltz | ~sleep |
| 06:40.31 | jbot | it has been said that sleep is overrated, and a poor substitute for caffeine. |
| 06:40.48 | jsharp | Which aint far off at this rate. |
| 06:40.51 | Sedorox | kernel upgrade |
| 06:40.52 | Sedorox | brb |
| 06:41.08 | shmaltz | ~sleep |
| 06:41.09 | jbot | sleep is probably overrated, and a poor substitute for caffeine. |
| 06:41.24 | shmaltz | ~goto sleep |
| 06:41.30 | shmaltz | ~do sleep |
| 06:41.32 | jbot | ACTION does sleep. |
| 06:41.40 | shmaltz | ~gn |
| 06:41.41 | jbot | from memory, gn is Guinea. Good Night Bastards |
| 06:43.44 | shmaltz | ~ doe /exit |
| 06:43.53 | shmaltz | ~do /quit |
| 06:43.55 | jbot | ACTION does /quit. |
| 06:44.04 | shmaltz | ~ you still here? |
| 06:44.22 | afrosheen | Mazda-MX5: I got it too |
| 06:44.23 | shmaltz | ~helo |
| 06:44.24 | jbot | [helo] the first command issued during smtp |
| 06:44.26 | afrosheen | Mazda-MX5: the yellow one? |
| 06:44.33 | debaser | shmaltz: what are you trying to do to the bot? |
| 06:44.59 | shmaltz | debaser, trying to make it quit it's irc client? |
| 06:45.02 | shmaltz | :) |
| 06:45.50 | mikegrb | you will not be successful |
| 06:45.55 | *** part/#asterisk afrosheen (~afro@c-67-166-172-141.client.comcast.net) |
| 06:46.33 | IQ | Good Night |
| 06:47.05 | shmaltz | I know that, I'm just playing around |
| 06:47.20 | CoaxD | Okay, SOMEBODY here has to know what the FUCK is with this singer - "Wing" |
| 06:47.32 | CoaxD | She SUCKS! |
| 06:47.43 | CoaxD | But...she has a website! And apparently, SHE SELLS CD'S! |
| 06:47.48 | CoaxD | (Hell, they even had her music on Southpark!) |
| 06:47.58 | CoaxD | Can someone PLEASE tell me I have NOT gone NUTS? |
| 06:48.00 | shmaltz | CoaxD, you sure you on the right channel? |
| 06:48.08 | debaser | CoaxD: brittany spears sells CDs and has a website, too |
| 06:48.15 | CoaxD | shmaltz: Yes, I am indeed on the right channel |
| 06:48.20 | CoaxD | debaser: Um. Hmmm. Good point |
| 06:48.25 | shmaltz | CoaxD, what makes you think so? |
| 06:48.32 | CoaxD | debaser: This chick is..well...worse than britney spears ;/ |
| 06:48.43 | CoaxD | debaser: (As muhc as I gag to even think that it is possible) |
| 06:48.51 | CoaxD | shmaltz: Because. Because I own your a$$. |
| 06:48.51 | Mazda-MX5 | .. |
| 06:49.01 | CoaxD | Mazda: I *know* man. I'm so sorry :( |
| 06:49.23 | shmaltz | CoaxD, and you tellin me you on the right channel? |
| 06:49.34 | debaser | CoaxD: frightening, but not suprising. see this comic: http://www.catandgirl.com/view.php?loc=240 |
| 06:49.34 | shmaltz | I think you belong on #kids |
| 06:50.07 | Supaplex | heh |
| 06:50.09 | *** join/#asterisk kb1_kanobe (~krisbouti@h24-207-80-55.cst.dccnet.com) |
| 06:50.33 | kb1_kanobe | evening all. |
| 06:51.06 | CoaxD | shmaltz: I do believe you're a poor judge of character. |
| 06:51.15 | CoaxD | debaser: Heh. troo dat |
| 06:51.23 | shmaltz | CoaxD, how so? |
| 06:51.37 | shmaltz | how come you said that nonesense about owning blah |
| 06:51.53 | *** join/#asterisk zhier (~nick@219.136.12.205) |
| 06:52.03 | CoaxD | shmaltz: because I was entertaining myself. its 12:51am on a wednesday night. And i'm tired, and bored. Do i need additional reason, sir? |
| 06:52.21 | shmaltz | actualy its 1:52 am |
| 06:52.25 | shmaltz | ~time |
| 06:52.26 | jbot | well, time is 1 dimensional, or everlasting |
| 06:52.28 | shmaltz | ~date |
| 06:52.29 | jbot | Thu Mar 24 06:52:29 2005 |
| 06:52.37 | CoaxD | shmaltz: Yes, but onliy technically. |
| 06:52.45 | shmaltz | wow, it's my birthday |
| 06:52.46 | CoaxD | shmaltz: If you live in a frozen tundra, its 12:52am. |
| 06:52.51 | Supaplex | Thu Mar 24 06:52:54 UTC 2005 |
| 06:52.56 | CoaxD | shmaltz: Happy somethingorother. |
| 06:53.12 | shmaltz | thansk |
| 06:53.21 | debaser | shmaltz: i am from the moon. my time cannot be comprehended by your meager earth brains. |
| 06:53.45 | Mazda-MX5 | happy birthday , shmaltz |
| 06:54.01 | shmaltz | debaser, well since you on the moon, you should be falling off in about 17 days, when the moon is not visible ;) |
| 06:54.04 | shmaltz | thanks |
| 06:54.16 | debaser | now, bend over, its time for your spanking! |
| 06:54.20 | CoaxD | Oh! *I* get it! |
| 06:54.26 | CoaxD | She sings at rest homes and hospitals! |
| 06:54.29 | shmaltz | ~lart debaser |
| 06:54.37 | CoaxD | Okay, it makes more sense now |
| 06:54.43 | CoaxD | (She's not half bad for a rest home and hospital singer) |
| 06:54.50 | debaser | CoaxD: they can't hear her, so they love her! |
| 06:54.59 | shmaltz | CoaxD, can I know who you talking about? |
| 06:55.09 | CoaxD | debaser: Actually, her voice is phenomenal. She just doesn't know how to use it |
| 06:55.19 | CoaxD | shmaltz: http://www.wingmusic.co.nz |
| 06:55.33 | CoaxD | shmaltz: Trust me on this. listen to the sample music |
| 06:55.48 | shmaltz | CoaxD she looks awfull |
| 06:56.41 | debaser | CoaxD: i really don't want to register. can you stick the mp3s somewhere? =] |
| 06:56.59 | *** join/#asterisk Inv_arp (junya@adsl-3-251-111.mia.bellsouth.net) |
| 06:56.59 | debaser | er, nevermind. |
| 06:57.15 | shmaltz | ok, guys gtg |
| 06:57.17 | shmaltz | bye |
| 06:57.19 | shmaltz | c ya |
| 06:57.20 | CoaxD | debaser: You dont have to register |
| 06:57.21 | shmaltz | ~bye |
| 06:57.22 | jbot | cya |
| 06:57.28 | CoaxD | ~botsnack |
| 06:57.28 | jbot | thanks, CoaxD |
| 06:57.34 | CoaxD | welcome |
| 06:57.37 | debaser | (and its funny she has 'castle on a cloud' on the phantom of the opera cd, considering its from les mis) |
| 06:57.37 | CoaxD | :) |
| 06:57.57 | CoaxD | debaser: Yeah, and if you listen to her rendition of phantom of the opera, it aint even half bad |
| 06:58.56 | CoaxD | debaser: The music is poorly mixed, obviously, but hell, she DOES have a nice voice; just has to be trained on..you know...how to actually..USE IT |
| 06:59.17 | CoaxD | The reaosn I looked at this was because they made fun of her on southpark. haha. |
| 06:59.20 | Mazda-MX5 | thank all for your anser, bye, I wanna disconnect |
| 06:59.23 | debaser | woah, the concepts of enunciation and articulation are beyond her. |
| 06:59.42 | CoaxD | debaser: Absolutely |
| 06:59.44 | nix000 | my latest finding is cdrtool ! |
| 07:00.22 | CoaxD | debaser; As is sadly the case with folks who have their palletes molded to something completely different than english :) |
| 07:00.41 | debaser | CoaxD: it could probably be fixed with protools. |
| 07:00.49 | debaser | like most pop music |
| 07:01.00 | CoaxD | debaser: hehe |
| 07:01.42 | debaser | beerrun, brb |
| 07:03.35 | debaser | ahh, beer. proof that god loves us and wants us to be happy. |
| 07:04.05 | CoaxD | beer GOOD |
| 07:04.21 | Shido6 | hheh |
| 07:04.32 | *** join/#asterisk terrapen_ (~cjs@cpe-66-25-86-139.satx.res.rr.com) |
| 07:04.41 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 07:06.29 | nix000 | anyone of you guys can give me advise on the cheapest way of getting asterisk to talk to an ss7 network ? |
| 07:06.29 | debaser | this is some english bitter that isn't really bitter/hoppy at all. creamy and nutty ale, bizarrely good. |
| 07:07.08 | terrapen_ | sounds more like a brown ale, debaser |
| 07:07.30 | debaser | its not a brown ale, though. |
| 07:07.52 | kb1_kanobe | nix000: it comes up on the mailing list with increasing frequency... do some searching there. I beleive it's only available as a commercial derivative because of the certification requirements to interface ss7. |
| 07:08.17 | *** join/#asterisk peted20 (~chatzilla@d6-119.rb.gh.centurytel.net) |
| 07:10.18 | debaser | different grains/yeast. this is definately an english bitter yeast and it doesn't have the color of the roasted grains |
| 07:12.02 | terrapen_ | does anybody here do asterisk consulting? |
| 07:12.22 | terrapen_ | and when you do, do you set your clients up with a voip service provider? |
| 07:12.40 | terrapen_ | and if you do that, do you collect a commission/premium on their long distance? |
| 07:14.24 | habakuk | terrapen_, I do consulting. I've thought of setting up a way to get a commision off of long distance.. some providers give you a kick back for sending them customers though |
| 07:15.03 | terrapen_ | i really want to do this |
| 07:15.10 | terrapen_ | this could be a great revenue stream for consultants |
| 07:15.27 | terrapen_ | most consultants are going to set their clients up with someone |
| 07:15.34 | terrapen_ | and the clients rarely understand how it works |
| 07:15.35 | debaser | you'd need to push a fuckload of minutes to make residual commisions amount to anything |
| 07:15.44 | terrapen_ | and are just happy not to be paying monthly fees |
| 07:15.54 | habakuk | I'm almost finished with an optimized cdr_module that could be used for something like this |
| 07:16.04 | terrapen_ | well, if you commanded $0.01/min premium |
| 07:16.16 | terrapen_ | 100 minutes puts a dollar in your pocket |
| 07:16.29 | terrapen_ | a busy office probably does a hell of a lot more outbound every hour |
| 07:16.48 | habakuk | terrapen_, exactly |
| 07:16.52 | debaser | thats a huge markup, though. |
| 07:17.19 | terrapen_ | debaser, not really |
| 07:17.21 | habakuk | not really.. I f you get minutes for .01 per minute |
| 07:17.33 | terrapen_ | this customer is payinig SBC probably $40/month AT LEAST for business lines |
| 07:17.37 | habakuk | and mark it up to .02 thats a great deal for most businesses |
| 07:17.44 | terrapen_ | and at BEST they pay $0.03/min for LD |
| 07:18.16 | terrapen_ | so if I offer them $0.02 or even $0.035/min for LD with no monthly fees and virtually unlimited simultaneous calls... |
| 07:18.17 | habakuk | heck you could probably mark it to 2.5c :) |
| 07:18.21 | terrapen_ | its a sweet deal for both of us |
| 07:18.27 | habakuk | yep |
| 07:18.30 | terrapen_ | habakuk: exactly |
| 07:18.30 | kb1_kanobe | However, if you're a busy office you're certainly going to feel the difference between $0.01/min and $0.02/min... |
| 07:18.50 | terrapen_ | actually, i dont think that this is something that should be coded into asterisk |
| 07:18.59 | terrapen_ | it should be a part of the billing system on top |
| 07:19.09 | terrapen_ | for example, IAX.cc or NuFone let you prepay money |
| 07:20.02 | habakuk | my system that I'm almost done with uses a hash table. So you can store your rates in a mysql database. When the module loads it stores the rates in a hash table for fast access |
| 07:20.07 | terrapen_ | they should make it so that, for every $1.00 of LD used, I get $0.20 put in my account |
| 07:20.19 | terrapen_ | deducted from the deposit, of course |
| 07:20.39 | habakuk | nah.. you just build it yourself is my idea |
| 07:20.53 | habakuk | that way you get multiple routes yourself. |
| 07:20.57 | terrapen_ | or better yet, they make it transparent |
| 07:21.06 | habakuk | have them send calls to your server and your rate the call |
| 07:21.07 | terrapen_ | customer uses 1 minute of US long distance |
| 07:21.10 | terrapen_ | gets billed $0.03 |
| 07:21.18 | debaser | or just invoice your client directly every month |
| 07:21.24 | terrapen_ | and $0.019 goes into my account |
| 07:21.39 | terrapen_ | i don't want to have to run a server |
| 07:21.48 | terrapen_ | why add another server into the mix |
| 07:21.54 | terrapen_ | its terribly easy to program |
| 07:22.00 | terrapen_ | you just add a differential to your rate |
| 07:22.16 | terrapen_ | and that differential goes into the commission account |
| 07:22.17 | habakuk | well trying to convince a provider to do this maybe difficult |
| 07:22.25 | terrapen_ | well, there is benefit in it for them |
| 07:22.44 | debaser | cdrs are in csv format, right? |
| 07:22.50 | terrapen_ | they get more big customers because the consultants are sending them their way |
| 07:23.06 | debaser | just take it into excel, multiply the values by 1.1 or 1.2 or whatever your markup is, and print it out |
| 07:23.15 | terrapen_ | if they were really evil, they could charge the commission-earner $0.005/min for the service |
| 07:23.17 | habakuk | debaser, can be in any format you want :) if you write a cdr module |
| 07:23.19 | terrapen_ | but that's evil |
| 07:23.35 | terrapen_ | and chances are, somebody will offer the service for free, just to attract customers |
| 07:23.53 | terrapen_ | how is VoIP resale currently done? |
| 07:24.56 | terrapen_ | or is it done? |
| 07:24.57 | habakuk | debaser, thats what my module does, but in realtime |
| 07:25.48 | habakuk | terrapen_, sure its done all the time. Typically by running a server and controlling mutiple routes |
| 07:26.03 | habakuk | its best with a b2bua though ( if using sip) |
| 07:26.09 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
| 07:26.12 | terrapen_ | ???? |
| 07:26.16 | debaser | i'd rather just apply a flat rate 'administration fee' every month with the usage and billable time (if any) every month. consolidated billing is good. |
| 07:26.35 | terrapen_ | so would that mean that I would accept the call on my server and forward it on to somebody like NuFone? |
| 07:26.42 | terrapen_ | wouldn't that increase latency? |
| 07:26.53 | terrapen_ | like, I would be a proxy for my customer? |
| 07:26.57 | habakuk | yeah. thats why its best to use a b2bua |
| 07:27.07 | terrapen_ | b2bua??? |
| 07:27.15 | habakuk | back to back user agentr |
| 07:27.18 | habakuk | agent |
| 07:27.33 | habakuk | you would only handle the signalling |
| 07:27.42 | habakuk | media would go directly to the provider |
| 07:28.02 | habakuk | unfortunately there are no opensource b2bua's that I'm aware of |
| 07:28.11 | habakuk | that are any good that is |
| 07:28.22 | *** join/#asterisk WilliamK (~wkeller@c-24-0-130-177.client.comcast.net) |
| 07:28.32 | habakuk | thats whats cool about SIP that iax can't do |
| 07:28.44 | terrapen_ | this is something that providers should offer |
| 07:28.49 | terrapen_ | i should write the software |
| 07:29.07 | habakuk | theres a bunch of good software solutions outthere already |
| 07:29.14 | habakuk | one of the best is nextone |
| 07:29.15 | terrapen_ | but i would need to know the nitty-gritty of how a provider's billing system worked |
| 07:29.23 | habakuk | acme packet has something similar |
| 07:30.50 | habakuk | unfortunately these solutions aren't cheap though |
| 07:31.41 | clive- | is anyone else besides me using the pa168 iax phones? |
| 07:31.49 | terrapen_ | hrmmm... |
| 07:31.53 | terrapen_ | doesn't make much sense |
| 07:32.06 | *** join/#asterisk hanhoong (~hanhoong@218.111.48.116) |
| 07:32.14 | terrapen_ | too bad jerjer isn't around right now |
| 07:32.28 | terrapen_ | he would probably know why this isn't being done |
| 07:33.10 | hanhoong | hi ... |
| 07:33.46 | hanhoong | need help here... newbie here... looking for some guidance on asterisk |
| 07:33.59 | habakuk | terrapen_, do a search for A-Z provider in google. chances are this is exactly what they are doing. They are just pocketing the extra for themselves though :) |
| 07:34.09 | terrapen_ | habakuk, acme packets stuff doesn't make much sense... |
| 07:34.11 | terrapen_ | to me |
| 07:34.26 | hanhoong | just finished installed the redhat 9 with * v9 now i wanna know whats next |
| 07:34.43 | habakuk | terrapen_, what doesn't make sense? As is in what its for? |
| 07:34.59 | *** join/#asterisk Isme (some@218.111.10.168) |
| 07:35.05 | terrapen_ | habakuk...those guys doing hosted PBXes...are they using a VPN to get to their customers? |
| 07:35.10 | *** join/#asterisk Alexi1 (~alexis@www.trim.it) |
| 07:35.16 | terrapen_ | habakuk, i guess i don't understand how you route SIP |
| 07:35.18 | Alexi1 | hello all |
| 07:35.29 | terrapen_ | i guess i always thought SIP was a point-to-point thing |
| 07:35.35 | Isme | hi, can anyone recomend me a billing software under * ? |
| 07:35.39 | terrapen_ | you had the originator and the terminator |
| 07:35.54 | terrapen_ | im not understanding how you could have something in the middle |
| 07:35.59 | terrapen_ | unless it is a proxy of some sort |
| 07:36.02 | habakuk | terrapen_, the easiest way to think of sip is you have 2 parts media ( or rtp) and signalling (sip) |
| 07:36.07 | terrapen_ | are those things just so fancy proxy? |
| 07:36.16 | habakuk | the thing in the middle is called a B2BUA |
| 07:36.30 | habakuk | its basically two user agents |
| 07:36.54 | habakuk | so boxA ---> B2BUA <---- box b |
| 07:36.55 | clive- | the problem with sip is that the RTP seldom goes point to point because of NAT proxying etc. |
| 07:37.14 | terrapen_ | ~rtp |
| 07:37.15 | jbot | rtp is probably The Internet-standard protocol for the transport of real-time data, including audio and video. RTP is used in virtually all voice-over-IP architectures, for videoconferencing, media-on-demand, and other applications. A thin protocol, it supports content identification, timing reconstruction, and detection of lost packets. |
| 07:37.38 | terrapen_ | ~sip |
| 07:37.39 | jbot | X11 PPP dialer interface written in gtk+. URL: http://www.geocities.com/SiliconValley/Campus/3104/sip/ Session Initiation Protocol (see RFC 3261) |
| 07:37.39 | habakuk | in this case box A can not talk to box directly, only through b2bua |
| 07:37.54 | terrapen_ | uhh, thats wrong |
| 07:37.59 | terrapen_ | (jbot) |
| 07:38.09 | terrapen_ | ok |
| 07:39.28 | habakuk | clive-, true NAT proxying is a challenge, but there are on premise equipment providers that solve that problem |
| 07:39.46 | habakuk | I helped design one :) |
| 07:40.00 | terrapen_ | ~SER |
| 07:40.01 | jbot | ser is probably Sip Express Router - see http://www.iptel.org/ser/ |
| 07:40.30 | clive- | habakuk tell me more |
| 07:42.02 | habakuk | clive-, the idea is the box is an ALG, it has one leg on the private network, and the other leg on the public side. So all natted traffic talks to the private side. All incoming traffic talks to the public side |
| 07:42.05 | *** join/#asterisk langals (~icechat5@196.7.14.183) |
| 07:42.16 | habakuk | heck you could use an asterisk box for this same purpose |
| 07:43.19 | *** join/#asterisk zhier (~nick@218.19.66.127) |
| 07:43.26 | terrapen_ | doesn't SER do the same thing? |
| 07:43.40 | *** join/#asterisk zhier (~nick@218.19.66.127) |
| 07:44.54 | habakuk | terrapen_, no its not fully stateful which you need for b2bua functionality |
| 07:45.01 | *** join/#asterisk channan (~channan9@66.180.121.185) |
| 07:45.26 | habakuk | the closest thing to an opensource b2bua is the vovida code.. but that has issues |
| 07:46.19 | clive- | its bascially a fancy nat with sip-ability |
| 07:46.49 | terrapen_ | k |
| 07:46.55 | terrapen_ | well its bedtime for me |
| 07:46.58 | terrapen_ | thanks for the info guys |
| 07:47.19 | nix000 | kb1_kanobe, still there ? |
| 07:47.26 | terrapen_ | BTW, SER might not be a bad choice for my billing idea |
| 07:47.38 | clive- | terrapen, ser is great |
| 07:47.39 | terrapen_ | i would set up SER on my clients PBX server |
| 07:47.47 | clive- | but then you have tons of sip-nat issues |
| 07:47.51 | terrapen_ | and i would have it do the transaction logging |
| 07:48.02 | SexyKen | I need to setup an Asterisk Server for a company. Virtual setup. Multiple companies one one instance. (For instance, I need to have 3 different 200 extensions go to differe people..)....anyone do this for a living? |
| 07:48.10 | terrapen_ | and i would run this client through my account at NuFone/IAX.cc/wherever |
| 07:48.16 | terrapen_ | and just bill the client for the LD |
| 07:48.26 | terrapen_ | or better yet, set up a pre-pay system for them on my website |
| 07:48.31 | habakuk | terrapen_, yeah except for a prepaid solution would have problems |
| 07:48.41 | terrapen_ | and since i know how much they used, i could deduct that from their balance on my end |
| 07:48.50 | terrapen_ | how so? |
| 07:49.26 | terrapen_ | i would always let them make the calls |
| 07:49.32 | terrapen_ | but i would have to keep track of the CDRs |
| 07:49.40 | terrapen_ | or else i would lose money |
| 07:49.45 | nix000 | anyone can tell me an ss7 to sip gateway that will work with asterisk ? |
| 07:49.46 | habakuk | because if you have a dishonest customer has .50 in his account, and makes a call to some expensive place for 5 hours you can't stop the call |
| 07:49.58 | terrapen_ | their calls would go through the SER proxy and then directly to my provider |
| 07:50.02 | channan | 'morning everyone... Does anyone knows if you can put an asterisk PBX using Vonage service at home? Is it legal? I read the terms of service but not too sure? |
| 07:50.03 | terrapen_ | who I would pay |
| 07:50.09 | terrapen_ | and my client would pay me |
| 07:50.16 | terrapen_ | with paypal or however |
| 07:50.23 | terrapen_ | i just keep track of their usage |
| 07:50.35 | terrapen_ | mark it up by whatever commission i want to charge |
| 07:50.40 | terrapen_ | and deduct from their funds |
| 07:50.50 | habakuk | terrapen_, right, like I said it works fine, except for cases when you need to shut off a call in progress |
| 07:51.01 | terrapen_ | true |
| 07:51.07 | terrapen_ | well, NuFone would take care of that :) |
| 07:51.16 | terrapen_ | if the balance is drained, they will cut it off, IIRC |
| 07:51.29 | habakuk | terrapen_, you may have a point there. |
| 07:51.36 | terrapen_ | i would use a seperate NuFone account for each customer |
| 07:51.41 | habakuk | basically setup a new account for each customer |
| 07:51.54 | habakuk | yeah exactly :) |
| 07:51.56 | terrapen_ | JoeBobInc goes through my SER proxy to JoeBobInc@Nufone |
| 07:52.24 | terrapen_ | the SER proxy resides at my clients |
| 07:52.46 | terrapen_ | and once every 10min or so, it would synch its CDRs with my server in my co-lo |
| 07:52.50 | habakuk | but there is a problem :) If you charge 2cents a minte, and nufone charges you 1cent. and lets say you have 100 minutes in your nufone account |
| 07:52.55 | terrapen_ | and i would deduct from their balance |
| 07:53.18 | habakuk | you only want the customer to use 50 minutes |
| 07:53.25 | terrapen_ | ahhhh |
| 07:53.27 | terrapen_ | hmmmmmmm |
| 07:53.39 | terrapen_ | wait, there has to be a solution |
| 07:55.01 | terrapen_ | ok, the solution is simple |
| 07:55.10 | terrapen_ | they are allowed to go into a negative balance with me |
| 07:55.29 | terrapen_ | but i do not put more money into their nufone until they bring their account with me back into the black |
| 07:55.45 | terrapen_ | if they want to use up all their nufone minutes, big fucking deal |
| 07:55.53 | habakuk | ok as long as you trust them |
| 07:55.54 | terrapen_ | but i won't give them any more until they pay me what they owe me |
| 07:56.02 | debaser | terracon: companies don't want to prepay for anything. |
| 07:56.03 | terrapen_ | and they won't have phone service until they do |
| 07:56.15 | Isme | hi, can anyone recommend me a billing software for * ? |
| 07:56.23 | terrapen_ | well, we shall see |
| 07:56.36 | terrapen_ | i think i can convince them to pre-pay, given the low cost of the service |
| 07:56.41 | debaser | and not having phone service is not an option. just suck it up and give them net 30 |
| 07:57.05 | terrapen_ | hrmmm |
| 07:57.22 | terrapen_ | fuck, i don't want to be a phone company |
| 07:57.26 | terrapen_ | back to square one |
| 07:57.33 | terrapen_ | the VoIP termination provider needs to do this |
| 07:57.36 | *** part/#asterisk kb1_kanobe (~krisbouti@h24-207-80-55.cst.dccnet.com) |
| 07:57.42 | nix000 | can anyone recommend an ss7 gateway on the market that would speak sip on the voip side ? |
| 07:57.56 | _Vile | verisign |
| 07:57.59 | _Vile | sip-7 |
| 07:58.15 | debaser | terracon: if you/your company has a good credit rating, your provider will probably give you net30 |
| 07:58.25 | nix000 | _Vile, are you responding to my question ? |
| 07:58.32 | terrapen_ | http://voip-info.org/wiki-SS7+Protocol+Converters+and+Gateways |
| 07:58.38 | _Vile | of course not, but that's probably your answer |
| 07:58.56 | terrapen_ | debaser, but then i have to become a telco essentially, billing my customers |
| 07:59.10 | terrapen_ | i have to collect CDRs, print bills, wait for checks, etc |
| 07:59.18 | terrapen_ | no more vacation for me |
| 07:59.29 | _Vile | terra, post billing is a bish |
| 07:59.35 | terrapen_ | i would rather make a little less on the deal and be able to just watch the money come in |
| 07:59.35 | debaser | making it automagical would be simple. |
| 07:59.46 | _Vile | I created a billing system about a year ago to handle it |
| 07:59.51 | _Vile | using ms word as a template |
| 07:59.53 | terrapen_ | i envision this as being very similar to Amazon.com commissions |
| 08:00.12 | debaser | fuck, give me a linux machine with a postscript printer attached that gets the cdrs and i could hack something up in perl in an hour |
| 08:00.19 | nix000 | terracon, i ve been .. there .. i ve exhausted these already. |
| 08:00.23 | terrapen_ | oh, i could write some neat software to use LaTeX to print pretty bills for my customers based on CDRs from a PostgreSQL database |
| 08:00.33 | terrapen_ | but who the fuck wants to mail bills every month? |
| 08:00.55 | nix000 | _Vile, i actually hapen to need a gateway .. not just a service |
| 08:01.08 | terrapen_ | debaser, want to work with me on some opensource bill printing stuff? |
| 08:01.16 | terrapen_ | that would be a fun project |
| 08:01.35 | habakuk | _Vile, ms word billing engine? thats sound uh.. interesting |
| 08:01.51 | terrapen_ | i would use Perl and LaTeX |
| 08:01.52 | _Vile | verisign can provide the pstn gateway.. for LD you can contact LD providers |
| 08:02.30 | _Vile | habak, ms word can be controlled |
| 08:02.39 | _Vile | via com |
| 08:03.33 | _Vile | PDF, postscript etc can be an easier solution, but style and ease of use wise, easier to modify a word template than program postscript |
| 08:03.52 | nix000 | _Vile, where did you see that they will provide a gateway ? |
| 08:04.00 | _Vile | nix, call them |
| 08:04.10 | debaser | _Vile: a word template? |
| 08:04.11 | _Vile | they won't give you the LD etc |
| 08:04.23 | _Vile | debaser, I use a word document and not a word template |
| 08:04.40 | _Vile | with {BLAH} as my replacement variables |
| 08:04.49 | debaser | i want completely automagical, when the billing cycle is up, the printer prints out a cover sheet telling you whos bill it is, the actual bill formated for a window envelope, and a blank throwaway page at the end |
| 08:04.49 | _Vile | which I insert HTML intop |
| 08:04.53 | _Vile | s/intop/into |
| 08:04.56 | terrapen_ | hrmmm |
| 08:05.06 | _Vile | from my VB app |
| 08:05.10 | terrapen_ | in CDRs, is there any unique-per-call value included? |
| 08:05.16 | terrapen_ | like, a PRIMARY KEY |
| 08:05.19 | nix000 | _Vile, any idea who makes the gateway ? |
| 08:05.32 | _Vile | nix, you're in the * channel |
| 08:05.55 | _Vile | Max TNT can support g.729 |
| 08:05.56 | debaser | if you show up in the morning and see shit laying in the printer, you toss them into the letter folder, toss that in an envelope, run it through the postage meter, and stick it in the outbound mail pile. 30 seconds of work. |
| 08:06.49 | nix000 | _Vile, yes ... i need the gateway to setup a calling card service based on * ... |
| 08:07.16 | _Vile | look at a Max TNT |
| 08:07.26 | _Vile | w/ * |
| 08:07.29 | _Vile | use 729 |
| 08:07.40 | _Vile | a 4 port card to connect to your carrier via PRI |
| 08:07.46 | _Vile | 4 port T1 |
| 08:07.57 | _Vile | TE410OP or TE405P |
| 08:08.11 | _Vile | Max TNT will talk SIP to * |
| 08:08.23 | _Vile | the TE's will allow the inbound 800 service |
| 08:08.52 | _Vile | * will be the middleman for the 800 service and the TNT, controlling billing and CDR collection |
| 08:09.53 | nix000 | _Vile, actualy i dont need 800 services (for now) |
| 08:11.07 | nix000 | tow! .. maybe that was not for me ! |
| 08:12.16 | _Vile | you'll need an 800 # if you expect to compete in the calling card market |
| 08:12.47 | _Vile | but yes, these messages are not for you |
| 08:13.45 | nix000 | _Vile, ok .. i see what you meant now. actually the only reason i need ss7 is because the equipment will be colocated with the local telco ! so ppl dial speacl *77 to call in. |
| 08:14.39 | nix000 | does the max tnt speak ss7 ? |
| 08:17.13 | *** join/#asterisk Manipura (~chatzilla@dsl-ep-209-115-250-i114-cgy.nucleus.com) |
| 08:19.43 | *** join/#asterisk langals (~icechat5@196.7.14.183) |
| 08:21.03 | terrapen_ | 'id' being the primary key |
| 08:21.06 | terrapen_ | oops |
| 08:22.40 | langals | I am trying to set up conferencing and am gettimg some errors - wondering if someone could help me? |
| 08:23.04 | SexyKen | Anyone know where I can hire someone to setup a single asterisk server for 3 companies to be hosted on? |
| 08:23.28 | jontow | sexyken; im sure you can get a lot of offers on that |
| 08:23.35 | nix000 | anyone familiar with sysmaster gateways ? |
| 08:23.48 | SexyKen | jontow, Where do you suggest I start? |
| 08:26.04 | jontow | during business hours ;) |
| 08:26.09 | jontow | where're you located? |
| 08:26.53 | jontow | i'd do it if i had better bandwidth for you to use :) |
| 08:27.16 | SexyKen | I'm located in the Bay Area. |
| 08:27.41 | SexyKen | I already have the server. It's a dedicated server at a data center in New York, where the actual businesses are based. |
| 08:27.48 | jontow | yeah, business hours in here might not be a bad place to start :) |
| 08:27.57 | jontow | where in new york? (im in new york :)) |
| 08:28.01 | SexyKen | Brooklyn |
| 08:28.09 | jontow | ah, far from me, heheh |
| 08:28.31 | SexyKen | What's your experience with Asterisk? |
| 08:31.02 | jontow | i use it at home in a random testing environment, and at work i've implemented 5-7 of them doing various things, 2 attached to T1-type PRIs, the others over IAX.. using SIP phones, MGCP adapters (with minimal luck .oO{?}), SCCP/Skinny (cisco) phones, ATAs (IAXy, Sipura SPA-2000).. I use wcfxo clone cards at home and wildcard T100P cards at work.. im currently implementing a voicemail server for a small telco |
| 08:31.45 | jontow | i've done some basic work with AGI, some decent IVRs including recording, etc :) |
| 08:32.07 | jontow | and i've done a lot with IAX2 in the last few weeks |
| 08:32.51 | SexyKen | Well -- our Provider is TelIAX and we're using SIP phones. Currently we have a (barely) working system...but it's not at all efficient, and often doesn't work right. |
| 08:35.00 | jontow | :) |
| 08:35.04 | jontow | what's wrong with it? |
| 08:35.44 | SexyKen | Queues aren't working properly or the way we want them to. Transfers often dont work. And part of it is based on conf files and part is based on realtime. |
| 08:36.58 | jontow | using the regular queues in asterisk i've seen some strange things myself.. and whats wrong with transferring? (and what specifically do you need with the realtime stuff?) |
| 08:37.36 | SexyKen | See, I didn't setup the system. |
| 08:37.39 | SexyKen | Someone else did. |
| 08:37.44 | jontow | aha :) |
| 08:37.44 | SexyKen | So I dont know much about it. |
| 08:37.51 | jontow | what's your role in it? (and asterisk?) |
| 08:38.27 | SexyKen | I manage the project (implementing asterisk as the companies(3)) phone system. |
| 08:43.02 | crash3m_ | can someone here with a Polycom IP300 with firmware version 1.4.1.0040 verify that there is a bug in the firmware that doesnt allow you to properly enter the "Auth password" ? |
| 08:44.23 | jontow | hmm |
| 08:45.20 | *** join/#asterisk _THEEND_ (~DrEaM@host37-42.pool8248.interbusiness.it) |
| 08:46.47 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 08:47.39 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 08:49.06 | langals | Whenever I start asterisk and run in debug mode I get the following error, which is repeated about 12 times: WARNING[5404]: chan_oss.c:269 sound_thread: Read error on sound device: Resource temporarily unavailable. Does anyone know what the problem is - assistance would be much appreciated |
| 08:50.30 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 09:02.20 | _Vile | nix, no... what equipment? |
| 09:02.58 | *** join/#asterisk ckruetze_ (~nospam@i3ED660CE.versanet.de) |
| 09:07.05 | *** join/#asterisk Supaplex (supaplex@205.208.245.134) |
| 09:09.04 | *** join/#asterisk MacDeath (david@196.22.239.13) |
| 09:09.09 | MacDeath | Morning All |
| 09:09.51 | Supaplex | mmmm00 |
| 09:12.07 | wildcard0 | anyone have any information on the sip jitterbuffer in progress? im interested in helping |
| 09:12.19 | clive- | wildcard speak to zoa |
| 09:12.28 | wildcard0 | thanks |
| 09:14.06 | MacDeath | Im having a problem with my zap cards |
| 09:14.12 | MacDeath | I get this when trying to make a call |
| 09:14.14 | MacDeath | Unable to create channel of type 'Zap' |
| 09:14.42 | Isme | i want * to dial from 1 cell phone to another cell. do i need some kind of adaptor to plug in to FXO ? |
| 09:18.40 | *** join/#asterisk ckruetze (~nospam@i3ED660CE.versanet.de) |
| 09:23.42 | *** join/#asterisk W1thdraw (~Withdraw@ip70-181-96-254.oc.oc.cox.net) |
| 09:24.21 | *** join/#asterisk W1thdraw (~Withdraw@ip70-181-96-254.oc.oc.cox.net) |
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| 09:34.24 | cjk | hi its me again, anyone here who got some experience in grandstream firmware preconfiguration (cfg.txt) |
| 09:34.48 | cjk | another question: do you known any hardphone supporting ilbc which are not called grandstream? |
| 09:36.51 | *** join/#asterisk pascals (~248d34d6@ip503c8584.speed.planet.nl) |
| 09:37.51 | *** join/#asterisk fishboy1669 (proxyuser@62.69.81.129) |
| 09:38.50 | MacDeath | cjk : I cant help you on that |
| 09:38.57 | MacDeath | I only used grandstream |
| 09:39.02 | MacDeath | and use ilbc |
| 09:39.07 | MacDeath | but |
| 09:39.12 | MacDeath | since i switched to ilbc |
| 09:39.20 | MacDeath | I cant use my zaptel cards |
| 09:39.23 | clive- | cjk the pa168 phones say they will support ilbc shortly |
| 09:39.24 | MacDeath | I get this message |
| 09:39.25 | MacDeath | <PROTECTED> |
| 09:40.00 | *** join/#asterisk DrFrancky (~chaos@pirus.securax.be) |
| 09:40.13 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 09:46.13 | Supaplex | does it matter what sampling rate I use for oog/mp3s for MOH? |
| 09:46.20 | Supaplex | er, ogg/ |
| 09:54.32 | *** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it) |
| 09:56.29 | darkskiez | so, why does the TE4xx series cards have two specified voltage cards, and the TE110P card not specify the voltage at all. |
| 09:57.48 | *** join/#asterisk Fabe_ (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 10:01.16 | darkskiez | im getting soo many google server errors the last few days |
| 10:02.37 | *** part/#asterisk Isme (some@218.111.10.168) |
| 10:04.41 | *** join/#asterisk meppl (~mephisto@pD9542582.dip.t-dialin.net) |
| 10:04.45 | meppl | guten morgen |
| 10:16.38 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 10:20.31 | *** join/#asterisk mAsH` (~mAsH@ppp-217-133-150-46.cust-adsl.tiscali.it) |
| 10:21.18 | mAsH` | hi all |
| 10:27.00 | mAsH` | can i connect asterisk with skype? |
| 10:27.24 | Alexi1 | i don't think so |
| 10:27.42 | Alexi1 | because skype seems to be non standard |
| 10:27.47 | Alexi1 | but if you find .... |
| 10:27.53 | Alexi1 | tell me :p |
| 10:27.56 | mAsH` | :) |
| 10:29.58 | *** join/#asterisk blackadder (~sburley@163-177.adsl.totalweb.net.uk) |
| 10:30.22 | blackadder | hi anyone here good with ISDN errors |
| 10:30.56 | blackadder | keep getting this error : chan_capi.c:955 capi_write: error sending DATA_B3_REQ (error=0x1103, datalen=160) B3in=1 |
| 10:31.07 | blackadder | anyone know why? |
| 10:31.22 | Mavvie | debug pri, and Q.931 (98-05) ISDN user-network interface layer 3 specification for basic call control.pdf :-) |
| 10:31.50 | Mavvie | maybe it's a lower layer, but that's all I have. |
| 10:32.10 | *** join/#asterisk Abbas (Abbas@203.81.194.242) |
| 10:32.13 | Abbas | Hi |
| 10:32.26 | Abbas | any body knows SIP Proxy of vonage |
| 10:35.04 | blackadder | Mawie what did yuo mean? |
| 10:37.31 | *** join/#asterisk lsc (~chatzilla@ARennes-303-1-8-225.w80-14.abo.wanadoo.fr) |
| 10:37.34 | *** join/#asterisk _THEEND_ (~DrEaM@80.18.184.226) |
| 10:37.49 | lsc | hello |
| 10:37.53 | lsc | any french here ? |
| 10:37.55 | lsc | need help :) |
| 10:38.14 | blackadder | no der iz no french eer |
| 10:38.31 | blackadder | :) |
| 10:39.40 | lsc | okay :s |
| 10:40.51 | Abbas | ~ SIP PROXY of VONAGE |
| 10:40.58 | Abbas | ~SIP PROXY of VONAGE |
| 10:41.40 | pif | hi, my musiconhold only works when transfering with '#' not when putting the call on hold or using the phone's transfer, any idea? |
| 10:42.03 | langals | <PROTECTED> |
| 10:42.39 | pif | lsc: what do you need that is 'french' about * ? |
| 10:45.42 | Zeeek | lsc what ? |
| 10:45.43 | *** join/#asterisk deckel (~deckel@DSL01.212.114.233.229.NEFkom.net) |
| 10:51.11 | *** join/#asterisk sandnigg0r (~crunkuser@66-55-197-254.gwi.net) |
| 11:00.18 | lsc | i'm searching for a french asterisk documentation |
| 11:00.58 | *** join/#asterisk montag___ (~montag@lan.desys.it) |
| 11:03.12 | Makenshi | can someone give me a sip proxy that can terminate calls to +1-800? |
| 11:03.23 | Makenshi | tf.voipmich.com never seems to work |
| 11:03.43 | Makenshi | for free, that is |
| 11:03.46 | *** part/#asterisk deckel (~deckel@DSL01.212.114.233.229.NEFkom.net) |
| 11:07.59 | *** join/#asterisk Los415 (~los415@c-24-126-63-233.we.client2.attbi.com) |
| 11:11.25 | riksta | Makenshi: voipuser.org |
| 11:15.11 | Makenshi | riksta, 404 (SIP not HTTP) |
| 11:15.53 | riksta | you need a www. |
| 11:16.03 | Makenshi | riksta, i'm making a sip call :p |
| 11:16.16 | riksta | if you register for the sip account |
| 11:16.19 | riksta | you get free 800 calls |
| 11:17.26 | Aze` | anyone use hisax |
| 11:18.29 | Aze` | ? |
| 11:19.06 | Makenshi | hmp |
| 11:20.04 | Makenshi | (hmping at something else) |
| 11:20.39 | *** join/#asterisk briiiiiiiiii (~strace@ADSL-F49-S197-critical-coi.nortenet.pt) |
| 11:21.28 | Makenshi | no dns srv records |
| 11:22.31 | langals | Is there anyone out there who would be able to help me with Meetme conferencing - having a few issues |
| 11:24.10 | Makenshi | langals, what's the problem? |
| 11:26.40 | langals | Makenshi: I can log into a conference, and it plays the "first participant" message, but in the sip debug screen it comes up with the same error over and over again, and then boots me out after about 30 seconds.... |
| 11:27.03 | langals | If someone else logs into the conference then we cannot communicate |
| 11:27.10 | *** join/#asterisk arbrandes (~arbrandes@200.184.189.132) |
| 11:27.30 | Aze` | How i can compose dtmf after cmd DIAL() ? |
| 11:30.04 | langals | Makenshi: the error is WARNING [6206]: chan_sip.c: 1829 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 64/64) - this is repeated over and over again |
| 11:31.22 | arbrandes | Morning everybody. |
| 11:31.35 | arbrandes | So... does chanspy work? |
| 11:33.05 | n1gg4s | when use the command "/usr/sbin/safe_asterisk" it shows to the message |
| 11:33.05 | n1gg4s | "Asterisk ended with exit status 1 |
| 11:33.05 | n1gg4s | Asterisk died with code 1. |
| 11:33.05 | n1gg4s | Automatically restarting Asterisk. they anyone knows because? |
| 11:34.49 | langals | Makenshi, any ideas about the errors? |
| 11:35.02 | *** join/#asterisk brc-tux (~brc-tux@pD9E9A2ED.dip0.t-ipconnect.de) |
| 11:35.30 | *** part/#asterisk brc-tux (~brc-tux@pD9E9A2ED.dip0.t-ipconnect.de) |
| 11:40.31 | *** join/#asterisk robl^ (~robl@dsl093-025-118.hou1.dsl.speakeasy.net) |
| 11:40.37 | goatmilk | robl^: !!!! |
| 11:42.12 | *** join/#asterisk robl^ (~robl@dsl093-025-118.hou1.dsl.speakeasy.net) |
| 11:43.19 | *** join/#asterisk Dabba (~d@ipv6.mfnx.ip6net.net) |
| 11:43.55 | *** part/#asterisk Dabba (~d@ipv6.mfnx.ip6net.net) |
| 11:45.36 | langals | Could anyone out there point me to some documentation on how to configure codec options in asterisk. I seem to be having a problem of codec incompatibility between my client and asterisk |
| 11:48.47 | arbrandes | langals: did you try http://voip-info.org? |
| 11:51.56 | c00w | hello all |
| 11:51.59 | c00w | i have a question |
| 11:52.11 | c00w | heres the example |
| 11:52.29 | c00w | i have an exterenal extension say 12345 |
| 11:52.33 | Druken | wholly c00w |
| 11:52.33 | langals | arbrandes - yes, but does not deal with my specific problem |
| 11:52.42 | c00w | now theres 4 different ways to get to this extension |
| 11:52.59 | c00w | example being 4 different ip addresses |
| 11:53.19 | langals | arbrandes - I get this error - WARNING [6206]: chan_sip.c: 1829 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 64/64) - this is repeated over and over again |
| 11:53.28 | c00w | now if i get a responce back from the first route saying No one is available to answer at this time |
| 11:53.42 | c00w | how do i then tell it to go to the new route |
| 11:53.59 | c00w | and if it says ringing how do i then tell it to go to voicemail instead of the next route. |
| 11:54.03 | c00w | do you understand ?? |
| 11:55.32 | goatmilk | c00w: you want to look at the extention order |
| 11:55.41 | goatmilk | the priority |
| 11:55.46 | pascals | Is there a cli command to show which channels are connected to each other? |
| 11:55.49 | c00w | yeah i understand that |
| 11:55.59 | Druken | c00w: when doing a dial, use g, as well as have a timeout for the voicemail |
| 11:56.00 | c00w | then take this for example |
| 11:56.20 | c00w | if it rings on route 1 |
| 11:56.27 | c00w | i just want it to go to voicemail |
| 11:56.35 | c00w | instead of going the other routes |
| 11:56.54 | c00w | i was looking @ the gotoif but i don't think i have any way of returning the status of the call so like ringing hungup etc |
| 11:57.00 | c00w | i'll look @ this g flag |
| 11:57.04 | Druken | why do you have 4 routes to the same place? |
| 11:57.18 | c00w | yeah |
| 11:57.33 | c00w | its more for load then anything |
| 11:57.39 | c00w | 4x30 channel boxes |
| 11:57.45 | goatmilk | c00w: also look at the help in the CLI.. "show application dial" |
| 11:57.53 | c00w | so if you go over 30 you want to spill on to the next one |
| 11:58.18 | Druken | how are you dialing these channel boxes? |
| 11:58.23 | Druken | e1 channels? |
| 11:58.31 | c00w | will be h323 |
| 11:58.37 | Druken | oh.. |
| 11:58.41 | c00w | yeah i know |
| 11:58.51 | c00w | caus if it was e1 i could group |
| 11:59.00 | c00w | which would be a class idea if you could group sip or h323 |
| 11:59.06 | *** join/#asterisk christo (~chris@office.enovi.com) |
| 11:59.10 | christo | mmm dicky wiki |
| 11:59.20 | Druken | wiki dicky? :) |
| 11:59.32 | pascals | How do I ask asterisk which channels are currently connected to each other? |
| 11:59.38 | c00w | i've just found this variable called dialstatus |
| 11:59.40 | christo | aue |
| 11:59.44 | christo | aye |
| 11:59.47 | Druken | pascals: show channels |
| 11:59.48 | c00w | that may be able to do i |
| 11:59.49 | c00w | it |
| 12:00.31 | Druken | c00w: how are you useing these?, is it for inside or outside calls? |
| 12:00.38 | pascals | Ah - when Bridged, the data says where. T |
| 12:00.40 | pascals | Thanks |
| 12:00.44 | Druken | perhaps a queue would work |
| 12:01.12 | c00w | well it will be from outside to inside |
| 12:01.27 | c00w | its will be a kind of routing from pstn 0870 to internal route numbers that i have allocated |
| 12:01.46 | c00w | it basically does a looking on the ddi number that is being passed from pstn |
| 12:02.07 | Abbas | any body knows SIP Proxy of vonage |
| 12:02.30 | c00w | then it gets back route-number and device list (poss of 5 different paths) then the type of call zap/sip/h323 etc and an account number (used for voicemail later on) |
| 12:02.46 | c00w | it takes the route number along with the device and the call type |
| 12:02.52 | c00w | and dials accordingly |
| 12:03.13 | c00w | the route lookup is custom module writen bymyself |
| 12:03.31 | c00w | don't think people would be interested in it |
| 12:03.59 | Druken | it's a strange setup you have... not sure what your trying to accomplish |
| 12:04.09 | c00w | its hard to understand |
| 12:04.21 | c00w | basically people have 0870 geo numbers |
| 12:04.32 | c00w | they are presented to us by carrier |
| 12:04.43 | pascals | How do I make sure the external clid is still shown when I transfer a call from one phone to the next? |
| 12:04.50 | c00w | we then have a rather large voip based network (40000000 ext) there about |
| 12:05.00 | c00w | prob more |
| 12:05.26 | Druken | 40 million extensions eh? |
| 12:05.41 | goatmilk | pascals: show application setcallerid |
| 12:05.43 | c00w | and the ext pstn geo number needs translated to the internal route number (sitenumber + extention) |
| 12:05.50 | c00w | yeah prob =) |
| 12:06.13 | c00w | its split over a lot of sites |
| 12:06.20 | c00w | all students. |
| 12:06.22 | Druken | well, call me an asshole... but if you have 40 million extensions, you can afford to pay someone to fix your problem :) |
| 12:06.31 | c00w | lol |
| 12:06.37 | c00w | thats fair enough |
| 12:06.47 | c00w | i'm just doing some testing |
| 12:06.54 | c00w | thats fine if you feel that way =) |
| 12:06.58 | pascals | goatmilk: yes, I use that on incomming calls - I don't see how to set it back to that value when someone transfers the call to someone else |
| 12:07.10 | c00w | and trust me its not that many calls for the ammount of extensions |
| 12:07.35 | c00w | an example site with 7-8k extions |
| 12:07.54 | c00w | have average of 10 calls concurrent @ 10am |
| 12:08.08 | c00w | going to about 40 concurrent calls @ 6-7 @ nite |
| 12:11.42 | langals | Does anyone have an idea what the capacity of meetme is on a 2ghz machine? |
| 12:12.25 | arbrandes | langals: that's something I'd like to know, too. |
| 12:12.43 | Chuji | 46 people |
| 12:12.55 | arbrandes | heh |
| 12:13.49 | Chuji | langals : I've tried as many as 50 on my 2.8 and it handled it. Just don't let it do any transcoding or recording |
| 12:14.18 | pascals | I hope not all 50 speak at once... |
| 12:15.05 | tzanger | pascals: shouldn't matter |
| 12:15.12 | tzanger | pascals: asterisk does no VAD or CNG |
| 12:15.53 | tzanger | basically voice detection... asteirsk sends audio no matter if it's silence or not |
| 12:15.56 | robl^ | but asterisk has Allison saying "Moose penis!" :) |
| 12:15.58 | tzanger | so whether those 50 calls are speaking or not doesn't matter |
| 12:16.01 | tzanger | robl^: hahaha |
| 12:16.10 | pascals | But 50 people speaking will be confusing :) |
| 12:16.37 | tzanger | pascals: heh |
| 12:16.46 | langals | Chuhji - how does one prevent it from doing transcoding? - I am having a bit of a problem with codecs at the moment |
| 12:17.08 | langals | I am using a standard client (built on the MS RTC Core library) |
| 12:17.52 | *** join/#asterisk Luhiwu (~marsosa@200.63.89.243) |
| 12:19.14 | langals | What you might want to have is 10 people in a conference, but 5 conferences running concurrently |
| 12:19.24 | langals | that is the type of implementation I am looking for |
| 12:20.21 | langals | tzanger: does asterisk meetme carry on broadcasting even if there is nothing sent from clients? |
| 12:21.31 | langals | have you guys all gone? |
| 12:21.52 | *** join/#asterisk whmok (~acidBurn@219.94.82.55) |
| 12:25.06 | tzanger | langals: yes |
| 12:26.36 | *** join/#asterisk mickm (~mickm@220-245-98-72-qld-pppoe.tpgi.com.au) |
| 12:27.16 | langals | tzanger - and will meetme cut a client off if there is VAD on the client side? |
| 12:27.40 | langals | tzanger - are there any open source conference servers which support VAD and CNG/ |
| 12:27.50 | tzanger | langals: I am not sure. I think their voice will be "odd" but again I'm not sure |
| 12:27.52 | *** join/#asterisk afe ([lku+QXZQZ@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 12:28.00 | tzanger | langals: I don't know, you'll have to do your own research |
| 12:28.05 | *** join/#asterisk Darwin[laptop] (~darwin-la@c-24-3-226-147.client.comcast.net) |
| 12:30.03 | langals | Chuji:what have you found with VAD and CNG? |
| 12:30.07 | Darwin35 | <PROTECTED> |
| 12:31.00 | Darwin35 | I have a comapny asking for it and not finding anything in the wiki |
| 12:34.23 | Darwin35 | ~seen goshen |
| 12:36.00 | jbot | goshen <~Goshen@70-57-80-147.slkc.qwest.net> was last seen on IRC in channel #asterisk, 2d 11h 8m 51s ago, saying: 'thats an odd one... Mar 21 18:26:25 NOTICE[1131]: callerid.c:306 callerid_feed: Caller*ID failed checksum'. |
| 12:38.48 | blackadder | anyone know why i would get this : SIP/2.0 403 Forbidden |
| 12:39.22 | Darwin35 | need more info do sip debug |
| 12:40.02 | *** join/#asterisk psirac (~psirac@AStDenis-103-1-9-178.w81-248.abo.wanadoo.fr) |
| 12:40.34 | blackadder | what is the site i can cut and paste to to avoid flooding this chanel |
| 12:41.39 | blitzrage | pastebin.ca |
| 12:46.05 | Darwin35 | paste the url heree |
| 12:46.24 | tzanger | Darwin35: what's wrong with DISA? |
| 12:46.53 | tzanger | I imagine you could do something simialr with Read and GotoIf |
| 12:47.59 | blackadder | Darwin35 http://pastebin.ca/8136 |
| 12:49.14 | Darwin35 | why is it people give phones names insted of just exten numbers |
| 12:49.20 | Darwin35 | grrr |
| 12:49.30 | blackadder | sorry its not just for me |
| 12:49.40 | blackadder | i am trying t ohelp asterisk noob |
| 12:50.26 | Darwin35 | I dont see it failing |
| 12:51.03 | Darwin35 | is the phone behind nat |
| 12:51.23 | Darwin35 | is so nat=yes canreinvite=no |
| 12:51.58 | blackadder | ok |
| 12:53.31 | mrtwister | hi - is anyone have bluetooth package, compiled to .deb? :) i installed * at home with app-get and wish to try bluetooth. dont offer to compile, too much job with mepis linux :) |
| 12:54.15 | blackadder | so you would suggest using numbers instead of names |
| 12:54.26 | Darwin35 | I do |
| 12:54.35 | Darwin35 | I think its easier to control |
| 12:54.42 | Darwin35 | in the sip,conf |
| 12:55.01 | Darwin35 | and when working the extensions.coonf |
| 12:55.06 | Darwin35 | and when working the extensions.conf |
| 12:55.27 | blackadder | so what does 403 forbidden mean |
| 12:56.44 | Darwin35 | it did not load sip2.0 |
| 12:56.58 | Darwin35 | the phone does not support it . it seems |
| 12:57.16 | blackadder | so how do i stop it constantly retrying |
| 12:57.49 | Darwin35 | not sure I shhot my gs |
| 12:58.05 | Darwin35 | moved over to a sipra |
| 12:58.13 | Darwin35 | they work bette |
| 12:58.14 | Darwin35 | r |
| 12:58.30 | Darwin35 | the 849 I think it is |
| 12:59.38 | Darwin35 | let me look at some thing brb |
| 13:06.41 | *** join/#asterisk mesi (~player@dsl-082-083-153-228.arcor-ip.net) |
| 13:07.35 | mesi | Is there a Command like Gosub() in Asterisk which allows returning from a context to the calling context? |
| 13:09.52 | RaYmAn-Bx | depending on what you want to do you might want to look into macros |
| 13:12.52 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 13:15.10 | kcir2 | my phones are trying to subscribe to my voicemail extension for thier built-in message functions and MWI, but sip debuging shows that the asterisk server is replying 403 |
| 13:15.25 | langals | Is there anyone who would be able to help me sort out codec issues with Meetme |
| 13:15.31 | kcir2 | have i overlooked some aspect of voicemailmain() |
| 13:16.02 | mesi | RayMan: Hm... I'm using macros, but not for my menu-functionality. Perhaps it is really the right thing for me. |
| 13:16.13 | *** join/#asterisk crich1999 (~crich@pD95D0568.dip.t-dialin.net) |
| 13:16.19 | crich1999 | hi all |
| 13:16.24 | mesi | Hi Crich. |
| 13:16.46 | crich1999 | Has anyone a clue how I can write a non GPL Module for the asterisk ? |
| 13:17.33 | kcir2 | for what reason? |
| 13:17.41 | crich1999 | for selling it |
| 13:18.00 | crich1999 | like codec_g792 from digium |
| 13:18.11 | crich1999 | s/792/729 |
| 13:18.40 | Darwin35 | if its a gs turn off subscribe in the interface |
| 13:19.06 | Darwin35 | <PROTECTED> |
| 13:19.55 | Darwin35 | no one will buy very fe g729 have ben sold |
| 13:20.12 | *** join/#asterisk lespiggot (~les@217.206.141.131) |
| 13:20.12 | crich1999 | Darwin35, what do you mean with gs ? where can i turn off subscribe |
| 13:20.17 | Darwin35 | people use speex and ilbc and other codecs to take its place |
| 13:20.23 | Darwin35 | inthe phone |
| 13:21.50 | robl^ | I have 6 licenses for g729.. but that was out necessity for one application |
| 13:23.08 | *** join/#asterisk Mimmus (~viggiani@ext.pitagora.it) |
| 13:23.30 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 13:24.18 | Mimmus | hi, I installed Asterisk a few hour ago, I'm trying to setup a simple gateway from VOIP software phones to external public network |
| 13:24.43 | Mimmus | I created a context named 'from-sip' |
| 13:25.05 | Mimmus | How can I 'Dial' the desired number? |
| 13:25.52 | `Sauron | Darwin: AgiNamu has (had) 729 g.729 licenses. |
| 13:25.56 | `Sauron | s/had// |
| 13:27.20 | Darwin35 | People buy them I did not say tey dont |
| 13:27.33 | Darwin35 | but they tend to use free over purchasiiing |
| 13:28.29 | `Sauron | There's not many free g729 codecs |
| 13:28.42 | `Sauron | and, the irony of Aginamu's licenses escaped you. |
| 13:31.13 | `Sauron | gregory |
| 13:33.38 | mountie | crich1999: You have to get a licence from digium for the asterisk code that is not GPL |
| 13:34.56 | mountie | crich1999: Then you are not bound by the GPL, but by the terms of the licence you negotiate with Digium |
| 13:35.44 | bjohnson | `Sauron: and they weren't puchased |
| 13:36.10 | *** join/#asterisk zoa (~zoa@pirus.securax.be) |
| 13:36.27 | bjohnson | Mimmus: use the dial command .. the format of which will change depending on how you connect to the pstn |
| 13:36.47 | bjohnson | Mimmus: there are lots of exmples on the wiki and the sample conf files should get you started too |
| 13:36.55 | bjohnson | ask specific questions here if needed |
| 13:37.19 | bjohnson | Mimmus: but you will have to be specific about your hardware and configuration options |
| 13:39.46 | *** join/#asterisk Emore (~Yoda@ip-138-151.sn2.eutelia.it) |
| 13:39.53 | Emore | hi all |
| 13:41.03 | crich1999 | mountie: do you know how much such a license cost ? |
| 13:41.23 | mountie | crich1999: No - I haven't bothered to inquire. |
| 13:41.54 | Mimmus | bjohnson: I use CAPI |
| 13:42.09 | mountie | crich1999: Contact them and ask... They must do it, becaues the SS7 people have done it. |
| 13:42.14 | crich1999 | Mimmus: better use mISDN ; ) |
| 13:42.55 | crich1999 | mountie: sure, they must have done it also wth their own g729 codecs, i think there are also other hw channels which do so |
| 13:43.28 | Mimmus | crich1999: no no. CAPI works, I don't know how to use 'Dial' |
| 13:43.44 | MacDeath | ummmm |
| 13:43.52 | MacDeath | wonder if someone can help me please |
| 13:44.02 | MacDeath | I'm using a grandstram phone |
| 13:44.06 | MacDeath | and diginum cards |
| 13:44.16 | MacDeath | grandstream -> grandstream work fine |
| 13:44.25 | MacDeath | xlite - grandstream works |
| 13:44.37 | MacDeath | grandstream - pstn doesnt work |
| 13:44.44 | MacDeath | "No translator path exists for channel type Zap" |
| 13:45.33 | bjohnson | crich1999: I think the 729 licenses may be listed on the digium web site .. I can't remember how much off hand but they seem inexpensive |
| 13:46.13 | *** join/#asterisk Rick_Hunter (~rhunter@02-148.008.popsite.net) |
| 13:46.18 | bjohnson | MacDeath: make sure they all allow the same codec |
| 13:46.49 | bjohnson | MacDeath: and make sure your extensions.conf section that the gs comes in on has access to the extens that dial out the zap |
| 13:47.35 | Mimmus | bjohnson: what is the syntax of Dial command? |
| 13:47.47 | Mimmus | I use Dial(CAPI/98426849:b${EXTEN:1},30) |
| 13:48.23 | crich1999 | mountie: where can i get infos about the SS7 channel ? |
| 13:48.36 | *** join/#asterisk jakepdev (~jakepdev@pool-68-236-56-226.phil.east.verizon.net) |
| 13:48.44 | Darwin35 | Dial(protocol,exten, Function) |
| 13:49.01 | Darwin35 | transfer/tomeout/musiconhold |
| 13:49.20 | bjohnson | Mimmus: sorry .. I don't ue capi so don't know the specifics of that |
| 13:49.40 | bjohnson | Mimmus: but the 'b' looks out of plae |
| 13:49.42 | bjohnson | place |
| 13:49.43 | Darwin35 | cli > show application dial |
| 13:50.36 | Mimmus | bjohnson: ok, but what is the exact way to pass destination number to dial? |
| 13:50.38 | bjohnson | Darwin35: he's got the deneral idea .. it's the specifics that he needs an examples of |
| 13:50.39 | MacDeath | bjohnson : i have given it access to all codecs |
| 13:50.55 | MacDeath | and it does have access in extensions.conf |
| 13:50.58 | pascals | When we transfer calls among local phones, the caller id is set to the person transfering the call - how can I detect a transfer and set the caller id back to the correct one after the transfer completes? |
| 13:51.05 | MacDeath | cause it used to work till i changed the default protocol |
| 13:51.28 | jakepdev | anyone know about DS1FD? |
| 13:51.30 | bjohnson | Mimmus: I don't use capi .. but for sip it would be dial(SIP/${EXTEN:1}@sip_out,30) |
| 13:51.40 | pascals | setcallerid I know about, but I don't see where to use it in a transfer |
| 13:51.54 | bjohnson | and for iax2 .. dial(IAX2/iax2_out/${EXTEN:1},30) |
| 13:52.29 | bjohnson | MacDeath: changed the default protocol? how do you do that? I think you mean default codec |
| 13:52.49 | bjohnson | MacDeath: protocol = SIP, IAX2, etc |
| 13:53.30 | *** join/#asterisk Dabba (~d@ipv6.mfnx.ip6net.net) |
| 13:54.07 | bjohnson | pascals: good question .. I thought it did that by default. Check the wiki and mail archives to see if any info there |
| 13:54.16 | bjohnson | pascals: no idea what to search for though |
| 13:54.27 | pascals | bjohnson: exactly my problem |
| 13:54.34 | Mimmus | bjohnson: I tried Dial(CAPI/98426849:b${EXTEN:1},30) but var ${EXTEN:1} is not expanded |
| 13:55.10 | Manipura | I just found a g729 key I got a year ago, anyone know if these things expire? |
| 13:55.16 | Manipura | I bought it from digium |
| 13:55.17 | pascals | bjohnson: perhaps the phones are not configured correctly (polycom IP500) |
| 13:55.39 | bjohnson | Mimmus: do you have a goto in the extensions.conf that it is following? goto clears the EXTEN variable |
| 13:55.58 | bjohnson | Manipura: no idea .. ask them. Might be info on their site already |
| 13:56.03 | langals | Hi there... |
| 13:56.38 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 13:56.39 | bjohnson | pascals: I just have ATAs that feed a Nortel so completely different situation here since transfers are alll within the Nortel system |
| 13:56.58 | Zeeek | anyone using flash operator panel? |
| 13:57.01 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 13:57.07 | MacDeath | bjohnson : I mean codecs, i put the 1st codec of the grandstream onto ilbc |
| 13:57.14 | MacDeath | Zeeek : yeah, I am |
| 13:57.32 | Zeeek | MacDeath I can't get the Flash to talk to the server |
| 13:57.37 | Zeeek | same sub net |
| 13:57.40 | Dabba | given the new app_chanspy and cvs head from 12hours ago is this expected http://pastebin.ca/raw/8137 |
| 13:57.45 | bjohnson | MacDeath: make sure the zapata allows that code |
| 13:57.47 | bjohnson | codec |
| 13:57.54 | Zeeek | but Apache is running on a non standard port. Does that matter? |
| 13:58.03 | bjohnson | MacDeath: also make sure the grandstream supports it |
| 13:58.17 | Zeeek | ilbc is on GS |
| 13:58.25 | MacDeath | Zeeek : apache port doesnt make a difference |
| 13:58.31 | MacDeath | let me check my config |
| 13:58.32 | Zeeek | didn't think so but what would? |
| 13:58.55 | MacDeath | Zeeek : my problem is that since i put my GS onto ilbc, it cant use the zaptel cards |
| 13:59.01 | MacDeath | "No translator path exists for channel type Zap" |
| 13:59.05 | bjohnson | MacDeath: that error sounds like a codec match not being available |
| 13:59.17 | MacDeath | Zeeek : does your flash panel load |
| 13:59.28 | MacDeath | bjohnson : what codec is available for the zaptel cards? |
| 13:59.29 | Zeeek | MacDeath and when you siwtch codecs it works again? |
| 13:59.35 | Zeeek | switch |
| 13:59.56 | MacDeath | well, havent switced back yet, but when it was all on ulaw it used to work, but it uses to omuch b/w |
| 14:00.04 | Zeeek | I use g729 but I remember there was a DTMF issue with iLBC I think |
| 14:00.20 | Zeeek | which has nothing to do with anything |
| 14:00.27 | Zeeek | maybe I didn't try it with ZAP |
| 14:00.48 | Zeeek | MacDeath ya the panel loads but everything flashes, meaning it aint talkin |
| 14:01.17 | MacDeath | is op_panel running |
| 14:01.23 | Zeeek | yup |
| 14:01.32 | MacDeath | and has it connected to the server? |
| 14:01.36 | Zeeek | and it looks normal (debig messages) |
| 14:02.31 | Zeeek | wait a sec |
| 14:02.41 | MacDeath | k |
| 14:02.42 | bjohnson | MacDeath: can you specify a codec in zapata.conf? iirc they only use one specific thing .. but maybe that's for fxo ports. check the codec lines in zapata and change it back to ulaw if it's there |
| 14:03.18 | bjohnson | MacDeath: you can use ilbc from the phones to * and something else from there on .. most people use ulaw on a lan btw |
| 14:03.27 | MacDeath | but cant asterisk talk to zap card on one codec, and to the phone from another |
| 14:03.32 | MacDeath | bjohnson : its not all on lan |
| 14:03.35 | bjohnson | yes |
| 14:03.42 | MacDeath | only 64kb ISDN |
| 14:04.01 | bjohnson | maybe the phones are trying to connect directly to each other |
| 14:04.10 | bjohnson | check into canreinvite settings |
| 14:04.15 | bjohnson | for the sip.conf sections |
| 14:04.27 | MacDeath | it is set as yet |
| 14:04.28 | MacDeath | yes |
| 14:04.36 | bjohnson | try no |
| 14:04.55 | jakepdev | Is * cable of using Lucent CAS for Line Side E1? |
| 14:05.04 | bjohnson | maybe the sip phone is trying to connect directly to the zap instead of keeping * in the middle |
| 14:05.09 | MacDeath | let me change it quick |
| 14:05.14 | bjohnson | jakepdev: no idea |
| 14:05.23 | *** join/#asterisk Jer13261 (~Jer@rdu57-251-152.nc.rr.com) |
| 14:05.26 | bjohnson | jakepdev: what is Lucent CAS? |
| 14:05.38 | jakepdev | probably a variation of CAS |
| 14:06.11 | bjohnson | there are digium E1 cards so I assume that they could connect to any E1 .. but I don't know |
| 14:06.28 | jakepdev | right - i have the E1 T1 card |
| 14:06.42 | jakepdev | the config part is key |
| 14:06.44 | jakepdev | :) |
| 14:07.04 | bjohnson | maybe ask for help with that |
| 14:07.16 | MacDeath | No translator path exists for channel type Zap (native 68) to 256 |
| 14:07.24 | Zeeek | MacDeath - just for completeness, I found what was wrong: I had 127.0.0.1 in the server and it needs the real address apparently, i.e. 192.168... |
| 14:07.39 | MacDeath | ahh :) |
| 14:07.46 | Zeeek | another mental note to add the brain-wiki |
| 14:07.48 | MacDeath | was going to ask what it was listening on |
| 14:08.03 | MacDeath | i have mine on * now |
| 14:08.03 | bjohnson | Zeeek: your apache may not be accepting calls from 127.0.0.1 |
| 14:08.19 | Zeeek | however, note that then an entry had to be added to the manager.conf for 192.168... |
| 14:08.35 | Zeeek | bjohnson yes, it's configured to do so |
| 14:09.05 | MacDeath | bjohnson : going to put ulaw at the top now |
| 14:09.08 | MacDeath | and see what happens |
| 14:09.13 | Zeeek | now the problem is to allow 4445 or whatever it is for remote operation |
| 14:09.49 | Zeeek | MacDeath I've found that when testing a codec, it should be allowed=alone first to remove any ambiguities |
| 14:10.56 | MacDeath | what does allowed=alone? |
| 14:11.00 | MacDeath | mean |
| 14:11.14 | bjohnson | no idea |
| 14:11.38 | bjohnson | I think he means disallow=all and then just allow the one you want to test |
| 14:11.43 | MacDeath | does the order of the codecs in sip.conf make a difference? |
| 14:12.06 | bjohnson | MacDeath: I keep hearing different answers |
| 14:12.19 | bjohnson | MacDeath: latest one I heard is that yes it does matter |
| 14:12.46 | MacDeath | bjohnson : i took g729 out of sip.conf and it all works |
| 14:13.03 | MacDeath | i was using it for passthrough |
| 14:13.05 | bjohnson | it was trying to use g729? |
| 14:13.24 | bjohnson | weird |
| 14:13.27 | MacDeath | mmm, it was using ulaw |
| 14:13.32 | MacDeath | i need it to use ilbc |
| 14:13.57 | Darwin[laptop] | you need a license for g729 unless just using passthrew |
| 14:15.11 | MacDeath | Darwin: yeah, but i want it to use something other than g729 UNLESS it is passthrough |
| 14:15.14 | Zeeek | I just meant allow=ilbc |
| 14:15.20 | Zeeek | disallow all else |
| 14:15.22 | bjohnson | MacDeath: try disallow=all and then just allow=ilbc |
| 14:15.35 | Essobi | ilbc is a heavy codec if you transcode. |
| 14:15.38 | Zeeek | that's what I meant but then a I said it and looked away :) |
| 14:15.38 | bjohnson | in the general section of sip.conf |
| 14:15.45 | bjohnson | get rid of any bandwidth lines |
| 14:16.04 | MacDeath | bjohnson : ilbc is working, changed order of codecs |
| 14:16.12 | MacDeath | im putting g729 at the bottom now |
| 14:16.13 | bjohnson | MacDeath: as Essobi said .. it will require some cpu power |
| 14:16.16 | MacDeath | and will see what it does |
| 14:16.24 | Essobi | show translation on the cli |
| 14:16.29 | MacDeath | Duel Xeon ok? |
| 14:16.34 | bjohnson | MacDeath: unless you bought g729 codecs .. don't list it at all |
| 14:16.50 | MacDeath | i was using them for passthough though |
| 14:17.04 | bjohnson | MacDeath: hardware has to be tested against your uses |
| 14:17.52 | MacDeath | its in experimental stage, will only be about 10 users |
| 14:18.33 | bjohnson | what is passthrough? |
| 14:19.55 | MacDeath | when you have canreinvte=yes |
| 14:20.02 | MacDeath | then you dont need the license |
| 14:23.32 | Essobi | I need some new rings on my 7960s. |
| 14:23.35 | Essobi | MMM... |
| 14:28.32 | bjohnson | MacDeath: I think that is only if the devices support g729 |
| 14:29.11 | *** join/#asterisk Dandan (dandan@234.88.149.195.in-addr.arpa.virt-ix.net) |
| 14:29.45 | BrianR___ | a2the codec negotiation stuff doesn't work very well. |
| 14:30.13 | BrianR___ | ie, if g729 passthrough is enabled asterisk will still select g729 in cases where it actually needs access to the audio stream |
| 14:30.30 | BrianR___ | at least in 1.0.x |
| 14:31.24 | *** part/#asterisk Dabba (~d@ipv6.mfnx.ip6net.net) |
| 14:32.31 | MacDeath | bjohnson : yeah, but the GS (which i am using) all do |
| 14:33.17 | MacDeath | how does * decide which codec to call a phone with |
| 14:33.29 | MacDeath | when there is an incoming call from a zap channel |
| 14:33.42 | MacDeath | what does it use to pass it through to the SIP phone |
| 14:37.13 | Darwin[laptop] | * chooses the codec to use by looking at the exten in sip.conf and in zaptel.conf |
| 14:37.27 | Darwin[laptop] | it looks to see what codecs you have listed |
| 14:37.34 | MacDeath | *looks at zaptel* |
| 14:37.43 | MacDeath | i have them all listed in sip.conf |
| 14:38.18 | MacDeath | where in zaptel.conf do you see codecs? |
| 14:39.02 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 14:39.06 | zoa | there should be ulaw or alaw somewhere in there i think |
| 14:39.24 | zoa | anyway, it uses only sip.conf to see how it has to send it |
| 14:39.39 | Zeeek | speaking of codecs... |
| 14:39.45 | bjohnson | I think he means zapata |
| 14:39.57 | bjohnson | not zaptel |
| 14:40.35 | Zeeek | I have an IAX phone I'm testing. When it calls a queue, no moh. WHen a SIP phone calls it there is moh. Same context. Same queue. Codecs supposedly match (askig for 2, codec=2) |
| 14:40.46 | bjohnson | but I'm not sure it's listed for zap cards at all anyway |
| 14:40.47 | Mimmus | someone can point me to a configuration sample for basic SIP->ISDN PRI gateway? |
| 14:41.09 | Mimmus | sip.conf and extensions.com sample will be appreciated |
| 14:42.18 | Essobi | Is there any GSM 6.10 codecs for windows media player? |
| 14:42.52 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 14:43.51 | Essobi | Google sucks sometimes. |
| 14:44.20 | zoa | Essobi: not that i know of |
| 14:44.24 | zoa | use apple quicktime |
| 14:48.07 | Essobi | It says there is.. HMMm.. |
| 14:49.42 | DrFrancky | there is in sound recorder |
| 14:49.45 | *** join/#asterisk mutilator (~animenodv@65.111.201.79) |
| 14:49.47 | mutilator | mornin all |
| 14:49.52 | DrFrancky | but it's "little" diferent :-)) |
| 14:50.00 | mutilator | i hookup a zultys zip 2 phone up |
| 14:50.07 | mutilator | and i get scrolling errors in asterisk |
| 14:50.17 | mutilator | <PROTECTED> |
| 14:50.21 | *** join/#asterisk spackle (~spackle@209.234.83.19) |
| 14:50.34 | mutilator | why does it try subscribe? |
| 14:51.06 | Essobi | mutilator Bad password/username specified in sip peer |
| 14:51.15 | *** join/#asterisk MikeJ[Laptop] (~icechat5@65.170.43.34) |
| 14:51.15 | mutilator | it's correct tho |
| 14:51.19 | Essobi | umm |
| 14:51.20 | Essobi | no |
| 14:51.22 | Essobi | it's not |
| 14:51.24 | mutilator | umm yes |
| 14:51.28 | mutilator | it's been plugged in for a few days |
| 14:51.33 | Essobi | <PROTECTED> |
| 14:51.35 | mutilator | it just started doing it out of the blue |
| 14:51.40 | MikeJ[Laptop] | so did evertone hear... windows is more secure than linux: http://www.securityinnovation.com/resources/linux_windows.shtml |
| 14:51.41 | mutilator | nothing has changed at all |
| 14:51.42 | MikeJ[Laptop] | :) |
| 14:52.21 | Essobi | Mike Oh god.. I stopped by the microsoft propoganda machine the other day.. read the whitepaper on why Linux has a higher TCO then windows. |
| 14:52.36 | Essobi | mutilator no "sip reloads?" |
| 14:53.00 | Essobi | could be it was borked up and it got reloaded.. the one in memory worked... the one on file didn't. |
| 14:54.22 | Essobi | mutilator Whip out your handy, dandy sniffer and watch the packets.. That always works for me. |
| 14:55.43 | mutilator | i've had ata's do that when they were behind nat before too, not quite the same error message but they won't login |
| 14:55.51 | mutilator | have to unplug em and plug em back in |
| 14:55.56 | *** join/#asterisk pif (ldm@82.66.93.83) |
| 14:56.04 | blackadder | MikeJ[Laptop] Pah! its fixed |
| 14:59.26 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
| 15:00.32 | *** join/#asterisk convey (~test@63.115.106.66) |
| 15:00.34 | *** join/#asterisk fishboy1669 (proxyuser@62.69.81.129) |
| 15:01.07 | fishboy1669 | hi guys |
| 15:01.13 | fishboy1669 | any one any idea on this Huh? an ilbc frame that isn't a multiple of 50 bytes long from RTP (4) |
| 15:01.14 | convey | mornin |
| 15:01.29 | fishboy1669 | hi convey |
| 15:02.12 | *** part/#asterisk lespiggot (~les@217.206.141.131) |
| 15:02.22 | *** join/#asterisk Lee__ (~Lee__@ool-44c26142.dyn.optonline.net) |
| 15:02.33 | *** join/#asterisk CosmicRay (~jgoerzen@2002:4463:7269:1:20e:a6ff:fe66:c5a3) |
| 15:02.39 | convey | anyone using the TE410P card from Digium? |
| 15:03.17 | convey | or the TE405P |
| 15:04.22 | Lee__ | nope, sorry. |
| 15:06.05 | Lee__ | I have found something that appears to be a bug that results in Asterisk crashing. It's 100% reproducible. Where should I go with this? |
| 15:06.24 | *** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc) |
| 15:06.25 | convey | What is it so I can avoid it :) |
| 15:06.35 | *** join/#asterisk elriah (~jfulcrum@adsl-068-209-198-242.sip.bhm.bellsouth.net) |
| 15:06.40 | christo | Lee__ take it to asterisk-developers list I guess |
| 15:06.47 | dwmw2_gone | Lee__: find the simplest way for someone _else_ to reproduce it too |
| 15:06.53 | *** part/#asterisk _Sam-- (sam@ns2.kneedraggers.com) |
| 15:06.53 | Lee__ | anyone game? |
| 15:06.55 | christo | Asterisk-Dev |
| 15:07.34 | elriah | Hey guys, how do I just from one context to another in extensions.conf? Say someone presses '1' and i want them to jump into context [test] where there is another option '1'? |
| 15:07.34 | dwmw2_gone | Lee__: http://www.digium.com/bugguidelines.html |
| 15:07.34 | Lee__ | thanks |
| 15:08.18 | Sedorox | elriah: s,1,goto(text,s,1) |
| 15:08.18 | Sedorox | ? |
| 15:08.23 | christo | elriah- exten => 1,1,Goto(othercontext,s,1) |
| 15:08.31 | Sedorox | er yea.. 1 not a... |
| 15:08.31 | Sedorox | s |
| 15:08.33 | MikeJ[Laptop] | lee- If you are getting craches, post it on mantis... make sure to read the guidelines and get as much information as possible |
| 15:08.33 | Sedorox | :-p |
| 15:08.40 | elriah | Oh, easy.. heh |
| 15:08.40 | *** join/#asterisk _omer (dfsdf@202.147.174.177) |
| 15:08.53 | zoa | LEe, tell me what it is |
| 15:08.57 | zoa | i have te410p cards |
| 15:09.00 | zoa | and te405p cards |
| 15:09.05 | _omer | I always forget the command to check the logged in AGENTS :-/ |
| 15:09.11 | MikeJ[Laptop] | Lee__, if you can, make sure to get a full bt with make valgrind on the crash fromt he start, that will help a lot |
| 15:09.19 | _omer | how to check the logged in agents in asterisk ? |
| 15:09.54 | zoa | bugs.digium.con is where to post it |
| 15:12.26 | mutilator | i dunno Essobi |
| 15:12.29 | mutilator | that phone is logged in |
| 15:12.33 | mutilator | i can make and receive calls on it |
| 15:12.41 | mutilator | it just scrolls that error over and over |
| 15:13.37 | DrFrancky | database show |
| 15:13.40 | *** join/#asterisk mhnoyes (~mhnoyes@user-38lc0k3.dialup.mindspring.com) |
| 15:13.49 | _omer | how to check the logged in agents in asterisk ? whats the command ? |
| 15:13.57 | DrFrancky | _omer: database show |
| 15:14.53 | *** join/#asterisk hobbes (~hobbes@cust143-50.dsl.versadsl.be) |
| 15:14.57 | _omer | let me check |
| 15:15.32 | mutilator | it tries to register 2 times per second too |
| 15:15.45 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
| 15:15.45 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:15.51 | _omer | ok and how to offline registered peers ? |
| 15:15.57 | hobbes | hi, any chan_capi god around ? |
| 15:16.02 | zoa | antony, thnx for chan_spy2 |
| 15:16.04 | zoa | didnt use it yet |
| 15:16.07 | _omer | in my asterisk there are 2 online peers.....I want to kill/disconnect one of them.. |
| 15:16.08 | zoa | but will try it in the future |
| 15:16.28 | zoa | softhangup + remove em from the sip.conf ? |
| 15:16.33 | zoa | or iax.conf ? |
| 15:16.44 | mutilator | or remove their database entry |
| 15:16.54 | mutilator | so it won't do anythin til they reregister |
| 15:17.05 | _omer | alright....... |
| 15:17.15 | hobbes | when I have several lines, how can I restrict incomming calls to one at a time ? |
| 15:17.18 | _omer | :D .thanks |
| 15:18.07 | DrFrancky | _omer: database command will help |
| 15:18.10 | *** join/#asterisk Katty (~angela@68.112.15.110) |
| 15:18.12 | DrFrancky | just chek it |
| 15:18.20 | _omer | yeah I did.... |
| 15:18.23 | Katty | hihi |
| 15:18.38 | *** join/#asterisk mog_home (~mog_home@digium.com) |
| 15:18.55 | _omer | DrFrancky: someone told me a command. in which I could check the queue and registered Agents at the same time...but forgot |
| 15:19.12 | mutilator | _omer... |
| 15:19.14 | _omer | I'm sure ..it wasn't "sip show peers" |
| 15:19.21 | mog_home | can someone confirm for me that the iaxy does not support md5 for auth |
| 15:19.21 | *** join/#asterisk gdh (foobar@bum.net) |
| 15:19.25 | fishboy1669 | where do i find out about * errors |
| 15:19.28 | fishboy1669 | what they mean |
| 15:19.29 | _omer | yes mutilator? |
| 15:19.31 | fishboy1669 | how to fix them |
| 15:19.31 | mutilator | ever seen the wiki? |
| 15:19.36 | mutilator | http://www.voip-info.org/tiki-index.php?page=Asterisk |
| 15:19.39 | fishboy1669 | yes |
| 15:19.41 | mutilator | lots of your answers are there |
| 15:19.44 | *** join/#asterisk Nix (~Nix@dsl81-214-9283.adsl.ttnet.net.tr) |
| 15:19.48 | _omer | aaahh...nope......great...let me check..thanks |
| 15:19.57 | DrFrancky | hehe :-)) |
| 15:20.02 | mutilator | probly answers to stuff ya havn't even thought of yet |
| 15:20.09 | Lee__ | maybe my bug is something common. It seems like it. I'm calling the server from a SIP softphone. The phone is registered and it can send and recieve messages to the server. The phone's context on the server tells it to route all 1700 calls to iaxtel. The client and the server are behind NAT on the same network. When I make a call to an iaxtel number, the server crashes with this message: |
| 15:20.12 | Lee__ | Mar 24 10:19:37 WARNING[1342]: pbx.c:1934 ast_pbx_run: Timeout, but no rule 't' in context 'from-sip' |
| 15:20.24 | gdh | Hola :) Any CAPI gurus around? Am trying to find why the phones on our PBX answer immediately in speakerphone when I dial through CAPI :) |
| 15:20.27 | fishboy1669 | Huh? an ilbc frame that isn't a multiple of 50 bytes long from RTP (4) |
| 15:20.36 | fishboy1669 | whats this mean |
| 15:21.28 | mutilator | the server crashes? |
| 15:21.43 | *** join/#asterisk cjk (~cjk@80.92.64.103) |
| 15:21.46 | Lee__ | yes, the server stops running immediately and I am disconnected from the console. |
| 15:21.58 | mutilator | weird.. |
| 15:22.10 | Lee__ | Asterisk 1.0.5 from Debian testing |
| 15:22.11 | mutilator | whats your dialplan look like? |
| 15:22.35 | mutilator | just remove the t from your dial() |
| 15:22.52 | mutilator | seeing as it's probly not going to timeout and go to voicemail... |
| 15:23.02 | gdh | mutilator: Are you talking to me? |
| 15:23.22 | Lee__ | sip.conf has a section for the client which points to the context called [from-sip] which has only this: |
| 15:23.23 | Lee__ | exten => _1700NXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN}) |
| 15:24.01 | Lee__ | [from-sip] is in extensions.conf |
| 15:24.50 | mutilator | did you ask something? |
| 15:25.08 | fishboy1669 | i have found that ilbc is internet low bitrate codec |
| 15:25.18 | fishboy1669 | but this donesnt make sence as i am using g729 |
| 15:25.20 | *** join/#asterisk shuric (alexander@alexander.office.inter-telecom.net.ru) |
| 15:25.29 | fishboy1669 | is anyone able to help me on this |
| 15:25.31 | fishboy1669 | please |
| 15:25.33 | fishboy1669 | Huh? an ilbc frame that isn't a multiple of 50 bytes long from RTP (4) |
| 15:25.43 | mutilator | gdh: nope i've no idea on your q |
| 15:25.53 | dwmw2_gone | argh. I'm confused. |
| 15:25.53 | gdh | mutilator: OKi, no worries =) |
| 15:26.40 | dwmw2_gone | playing with chan_bluetooth. Incoming audio from the headset is bizarrely distorted. Yet if I make its registered write() function a dummy, and call the _real_ write function with every frame I _read_, I get perfect echo. |
| 15:26.43 | mutilator | fishboy1669: is ilbc explicitly blocked? |
| 15:32.32 | fishboy1669 | no it just comes up warning |
| 15:32.34 | fishboy1669 | Huh? an ilbc frame that isn't a multiple of 50 bytes long from RTP (4) |
| 15:32.42 | fishboy1669 | but the sound dies till the error passes |
| 15:32.47 | fishboy1669 | sorry warnings pass |
| 15:33.24 | fishboy1669 | i have managed to read on web suggesting turning off dtmfmode=rfc2833 but cant understand why this would effect it |
| 15:33.32 | fishboy1669 | help would be greatly recieved |
| 15:33.57 | fishboy1669 | is ilbc something to do with inline? |
| 15:34.01 | fishboy1669 | codec |
| 15:34.08 | *** join/#asterisk akaye (~akaye@i-194-106-46-242.freedom2surf.net) |
| 15:34.13 | fishboy1669 | cos i ait using ilbc so cant see where its coming from |
| 15:34.20 | fishboy1669 | fooooobbbbbaaaarrrr |
| 15:34.51 | fishboy1669 | sometimes i hate my lack of understanding |
| 15:34.53 | fishboy1669 | :( |
| 15:36.56 | Katty | fishboy1669: you just need hugs. |
| 15:37.02 | Katty | fishboy1669: hugs make everything better. |
| 15:37.40 | elriah | If I have an exten => t,1,Goto(s,3) for example, and I want it to only timeout 3 times, then hangup, how would I do that? |
| 15:38.36 | fishboy1669 | cheers katty |
| 15:38.50 | fishboy1669 | u r sweet |
| 15:39.04 | *** join/#asterisk topping (~topping@dhcp024-210-082-196.columbus.rr.com) |
| 15:39.06 | Katty | just sometimes. |
| 15:39.25 | fishboy1669 | lol is that why u have the name katty? |
| 15:39.59 | Katty | fishboy1669: no. Katrina was taken. Karina is a romulan science officer. Add in a t, and you have Kat (i like cats). |
| 15:40.05 | Katty | fishboy1669: so now it's just Katty. |
| 15:40.33 | *** join/#asterisk jeffik (~jeffik@CPE0050bac711e3-CM0012256ead9e.cpe.net.cable.rogers.com) |
| 15:40.34 | tzanger | and women are generally catty to begin with |
| 15:40.44 | Hmmhesays | heh |
| 15:41.08 | fishboy1669 | tzanger carefull tiger she may not hug u for that comment |
| 15:41.08 | Katty | tzanger: what's that mean? |
| 15:41.20 | tzanger | ha |
| 15:41.23 | Katty | pffft |
| 15:41.25 | tzanger | it's all designed to evoke response |
| 15:41.34 | Hmmhesays | that's a nice way of saying fickle and ornery |
| 15:41.37 | tzanger | hahaha |
| 15:41.46 | Katty | well, duh. |
| 15:41.54 | Katty | that's the definition of female. fickle and ornery |
| 15:41.56 | tzanger | women aren't fickle or ornery? That's part of what makes them women... |
| 15:41.59 | tzanger | I never said it was bad |
| 15:42.01 | fishboy1669 | hey wheres my hug katty |
| 15:42.07 | Katty | fishboy1669: i don't hug strangers. |
| 15:42.10 | tzanger | sorry you gotta step up to get one :-) |
| 15:42.12 | mutilator | so does anyone know why this phone is doing this? |
| 15:42.12 | mutilator | Failed to authenticate user Glenn<sip:9895072471@m33access.com>;tag=2730c-7c41 for SUBSCRIBE |
| 15:42.13 | Katty | AND YOU"RE STRANGE |
| 15:42.18 | Hmmhesays | burn |
| 15:42.20 | mutilator | every 500ms i get that error |
| 15:42.26 | Katty | Hmmhesays: i'll hug /you/ in a minute |
| 15:42.28 | mutilator | but the phone is logged in, it can make and receive calls |
| 15:42.28 | tzanger | people are strange... when you're a stranger.. faces seem ugly.. when you're alone |
| 15:42.36 | Hmmhesays | ohhh good song |
| 15:42.38 | Katty | Hmmhesays: just as soon as i'm done playing with gradients |
| 15:42.43 | tzanger | women seem wicked, when you're unwanted... and I dont' remember the rest :) |
| 15:42.43 | fishboy1669 | im strange in what way? |
| 15:42.49 | *** join/#asterisk oej (~oej@apollo.webway.se) |
| 15:42.50 | tzanger | well your nick for starters :_) |
| 15:42.50 | Katty | fishboy1669: in that i don't know you. |
| 15:42.56 | fishboy1669 | aha ok |
| 15:43.08 | Katty | i need second opinion |
| 15:43.09 | fishboy1669 | thought i had three ears or something then |
| 15:43.11 | fishboy1669 | was worried |
| 15:43.12 | fishboy1669 | lol |
| 15:43.21 | Katty | is www.copi-rite.com/connect-rite/textmessaging.asp to...gradienty? |
| 15:43.36 | Hmmhesays | if you had 3 ears I would expect you to be a musician |
| 15:43.39 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 15:43.39 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 15:43.42 | Katty | my --> fade effect is being all omgwtfyellowlolz |
| 15:45.00 | fishboy1669 | now u have lost me! |
| 15:45.09 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 15:45.13 | fishboy1669 | tzanger what film is the song from |
| 15:45.22 | fishboy1669 | it is a wicked film if u know it |
| 15:45.27 | tzanger | oh I dunno, I just know the song |
| 15:45.32 | fishboy1669 | dow |
| 15:45.39 | tzanger | Katty: did you listen to those tracks I pointed out last night? |
| 15:45.40 | fishboy1669 | eeeeeeeeeerrrrrrr you loose |
| 15:45.41 | fishboy1669 | lol |
| 15:45.43 | tzanger | I've got them stuck in my head |
| 15:45.47 | Katty | tzanger: i did not. |
| 15:45.55 | tzanger | you should... :-) |
| 15:46.02 | Katty | tzanger: butbutbut, busy! |
| 15:46.06 | tzanger | if I could play like that I could make women swoon |
| 15:46.11 | tzanger | now I can only do that if I forget to shower |
| 15:46.18 | fishboy1669 | lol |
| 15:46.21 | fishboy1669 | lmfao |
| 15:46.34 | fishboy1669 | the film was lost boys |
| 15:46.40 | fishboy1669 | dam good film |
| 15:46.43 | fishboy1669 | have u seen it |
| 15:46.46 | tzanger | nope |
| 15:46.55 | fishboy1669 | worth getting out on dvd |
| 15:47.02 | fishboy1669 | or dl if u do that |
| 15:48.45 | fishboy1669 | :( |
| 15:48.50 | fishboy1669 | :.( |
| 15:48.56 | fishboy1669 | :...( |
| 15:49.01 | convey | anyone using the TE410P or TE405P card from Digium? |
| 15:49.10 | sivana | convey: yes |
| 15:49.12 | tzanger | yes |
| 15:49.26 | fishboy1669 | feels left out from katty's hugs |
| 15:49.30 | sivana | tzanger: there's a Te411P coming out shortly |
| 15:49.37 | tzanger | sivana: oh yeah? |
| 15:49.40 | sivana | tzanger: do you know the diff? |
| 15:49.40 | convey | Do they perform well? Can you fill evey channel vi T1? |
| 15:49.46 | *** join/#asterisk SpaceBass (~sp@24.125.33.214) |
| 15:49.48 | SpaceBass | hey folks |
| 15:49.50 | tzanger | sivana: not offhand no |
| 15:49.51 | sivana | tzanger: on-board echo can |
| 15:49.54 | sivana | supposedly |
| 15:49.57 | tzanger | nice |
| 15:49.57 | SpaceBass | anyone have expirence with a at-168 or pa168 ata? |
| 15:50.05 | fishboy1669 | hi space |
| 15:50.09 | fishboy1669 | no |
| 15:50.10 | SpaceBass | hey Phishboy |
| 15:50.10 | fishboy1669 | sorry |
| 15:50.14 | `Sauron | Hmmhesays' nick always makes me laugh. |
| 15:50.15 | sambal | tzanger: i guess a little more expensive then.. |
| 15:50.15 | sivana | according to locals around Digium.. heh |
| 15:50.17 | tzanger | I would have figured they coudl do that with the existing hardware with an FPGA firmware update but I'm not sure they have the capability to do that from the drivers |
| 15:50.19 | sambal | tzanger: DSP chips? |
| 15:50.26 | tzanger | sambal: no likely just FPGA stuff |
| 15:50.33 | sivana | tzanger: not sure if the drives will be new |
| 15:50.34 | tzanger | the entire point of Zapata is no DSP |
| 15:50.58 | convey | I have a 96 port application |
| 15:50.59 | sivana | I'm still stuck between a rock and a hard place |
| 15:51.01 | sambal | it's done now by using the cpu power, no? |
| 15:51.08 | *** part/#asterisk blackadder (~sburley@163-177.adsl.totalweb.net.uk) |
| 15:51.12 | convey | I just wanted to make sure the card could handle the app. |
| 15:51.14 | sivana | I'm looking at the IQ1500 softswitch |
| 15:51.26 | sivana | from Versatel Networks |
| 15:51.48 | sambal | tzanger: echo canc. is done by the cpu at the moment in asterisk? |
| 15:52.02 | sivana | sambal: yes, it's weak |
| 15:52.04 | tzanger | sambal: yes |
| 15:52.05 | SpaceBass | anyone ever had a problem with an ata taking 30 - 60 seconds to connect? |
| 15:52.08 | tzanger | weak |
| 15:52.09 | tzanger | hahaha |
| 15:52.13 | sivana | heh |
| 15:52.14 | tzanger | do you watch South Park |
| 15:52.17 | sivana | no |
| 15:52.22 | tzanger | dude that's weak |
| 15:52.24 | *** join/#asterisk ChkDigit (~mike@static65-87-228-18.regina.accesscomm.ca) |
| 15:52.26 | sivana | I can't stand that show |
| 15:52.33 | *** join/#asterisk _queuetue (kyoo@host-216-153-157-227.man.choiceone.net) |
| 15:52.34 | sivana | I watch Sesame Street only |
| 15:52.57 | sivana | ok.. and Family Guy |
| 15:53.01 | robl^ | Mr. Hanky! |
| 15:53.17 | `Sauron | Family Guy++ |
| 15:53.24 | _queuetue | Hi, all. Is there a procedure to unlock the Linksys PAP2? I just picked one up at staples and am a little dissapointed to read it's designed to only work with Vonage... |
| 15:53.38 | sivana | _queuetue: get approved from Linksys to sell/use them |
| 15:53.55 | jeffik | need some help with *@home |
| 15:54.05 | ChkDigit | Can someone point me to some of the FXO channel bank cards that aren't from Digium? |
| 15:54.27 | jakepdev | chkdigit - sagnoma cards? |
| 15:54.28 | ChkDigit | I need between 4 and 12 incoming lines. |
| 15:54.37 | _queuetue | sivana, Is there an unofficial procedure? Becoming a linksys dealer isn't what I had in mind. :) |
| 15:54.38 | SpaceBass | _queuetue from what I have ready you have to get new firmware from linksys and that ain't happening |
| 15:54.38 | tzanger | ChkDigit: Adit600, Rhino, etc |
| 15:54.40 | sivana | _queuetue: supposedly there's a way, but you may be violating the DMCA or whatever |
| 15:55.07 | _queuetue | So I should just return it, then, since it's broken. |
| 15:55.07 | christo | when compiling asterisk-addons, I get the error: "cdr_addon_mysql.c:24:22: asterisk.h: No such file or directory" what's the secret to getting it to see into my asterisk-1.0.5 source directory? |
| 15:55.19 | sivana | _queuetue: it's a bit of a hack that I'm not familiar with exactly, but I know it's been done |
| 15:55.25 | *** join/#asterisk bannerman (~bannerman@209.216.176.42) |
| 15:55.26 | dwmw2_gone | hm. making blt_read() malloc its buffers to prevent the possiblilty that we're scribbling on them later, doens't help. |
| 15:55.28 | bannerman | mornin everyone |
| 15:55.36 | sivana | christo: did you download asterisk as well? :) |
| 15:55.45 | christo | sivana - yup |
| 15:55.52 | _queuetue | sivana, do you know where you heard abou the sucess? Since it's broken anyway, I might as well try to fix it before I return it.:) |
| 15:55.57 | christo | and it's already build and running, sivana |
| 15:56.01 | sivana | christo: use PostgreSQL :) |
| 15:56.13 | dwmw2_gone | subclass and frametype are correct. Yet still what I get back from Echo() is corrupted, while if I call blt_write() immediately with the frame I return from blt_read(), it echoes fine. |
| 15:56.20 | dwmw2_gone | anyone have any clue what else to look for? |
| 15:56.23 | tzanger | sivana: you're such a postgresql whore |
| 15:56.25 | christo | sivana, there's a Postgres() command in asterisk? |
| 15:56.35 | sivana | _queuetue: no idea to be honest |
| 15:56.37 | sivana | hehe |
| 15:56.38 | bannerman | Is there an easy way to make the client hear the phone ringing while they wait in the queue? |
| 15:56.46 | sivana | tzanger: thanks to you :) |
| 15:56.47 | christo | I don't think that's the answer I'm looking for to be honest ;) |
| 15:57.00 | sivana | tzanger: takes one to know one :) |
| 15:57.04 | convey | So will a TE410P or TE405P work for a 96 port T1 app? |
| 15:57.14 | tzanger | convey: yes |
| 15:57.17 | tzanger | sivana: :-) |
| 15:57.23 | tzanger | but the bigger question is... was I wrong? |
| 15:57.29 | sivana | no.. I love it |
| 15:57.32 | tzanger | me too |
| 15:57.44 | convey | tzanger: whould I experience quality issues? |
| 15:58.01 | Juggie | christo, edit the Makefile, fix the asterisk include directory to be the proper one.... and recompile. |
| 15:58.04 | Hmmhesays | ugh I had legacy equipment |
| 15:58.07 | sivana | christo: you may need to check the header paths |
| 15:58.10 | tzanger | convey: you're asking questions iwhtout giving anyone enough infomration about the entire infrastructure. You won't get adequate answers |
| 15:58.18 | Juggie | guys, its very simple |
| 15:58.19 | Hmmhesays | i mean HATE |
| 15:58.20 | Juggie | edit Makefile |
| 15:58.33 | Juggie | change I=../asterisk to I=../asterisk-1.0.5 |
| 15:58.34 | *** join/#asterisk ManxPower (~eric@ip-209-16-83-10.i-55.com) |
| 15:58.37 | Juggie | save, recompile. |
| 15:58.59 | sivana | Juggie: is that an issue with the current stable release? |
| 15:59.00 | antifuchs | and you might want to use -I=../asterisk-1.0.5/include as well |
| 15:59.07 | Juggie | allways has been |
| 15:59.16 | sivana | hehe |
| 15:59.21 | antifuchs | (thanks go to people who assume that everything is going to end up in /usr) |
| 15:59.31 | Juggie | tar balls allways have directory names based on version numbers, but the includes look for no version number. |
| 15:59.32 | convey | tzanger: my system is a Sunfire v40z dual AMD 64 1.8G, SuSe Linux on Asterisk 1.0.5. |
| 15:59.38 | Juggie | if you pull from the cvs, it will work fine. |
| 16:00.00 | sivana | I see |
| 16:00.25 | tzanger | convey: you're already in a strange land, you're using 64-bit x86 |
| 16:01.04 | sivana | tzanger: I dont' think I'm going down this time |
| 16:01.07 | tzanger | convey: the key to getting anything to work right is to know the system... and as it appears that you're an asterisk newbie (not a bad thing, we were ALL there) you're already taking on a significant workload without going to bleeding edge hardware |
| 16:01.11 | tzanger | sivana: eh? |
| 16:01.12 | convey | tzanger: Is AMD64 not recommended? |
| 16:01.29 | tzanger | convey: it's not as well tested as regular old 32 bit x86 |
| 16:01.29 | sivana | tzanger: for Torastricon :) |
| 16:01.47 | tzanger | just as AMD is not as well tested as Intel for the zapata hardware (esp. echo cancellation) |
| 16:01.56 | tzanger | sivana: that's alright I won't be there either :-) Good Friday and all |
| 16:02.11 | sivana | tzanger: ya, next one for sure |
| 16:02.25 | Darwin[laptop] | o hell its easter weekend time to go Rabbit hunting |
| 16:02.30 | sivana | heh |
| 16:02.42 | Darwin[laptop] | get me a big wabbit |
| 16:02.55 | sivana | shhh |
| 16:02.55 | jeffik | anybody familiar with *@home |
| 16:02.58 | convey | tzanger: so should I stick with Intel for T1 apps? |
| 16:04.16 | tzanger | convey: amd works -- don't get me wrong. I'm just saying that intel's far more tested IMO |
| 16:04.21 | *** part/#asterisk Jer13261 (~Jer@rdu57-251-152.nc.rr.com) |
| 16:04.27 | Alexi1 | I have a SJphone and a grandstream, the first can call the GrandS, but SJphone can't be called by the grandstream ! |
| 16:05.03 | Alexi1 | * tells app_dial.c ... unable to create channel of type SIP |
| 16:05.50 | *** join/#asterisk Robbster (~james@wblv-146-243-180.telkomadsl.co.za) |
| 16:05.56 | Robbster | lo all :) |
| 16:06.14 | *** join/#asterisk Moc____ (~mochouina@h66-201-214-109.gtconnect.net) |
| 16:06.47 | christo | ok - fixed the makefile and asterisk-addons builds like a beauty |
| 16:07.02 | SpaceBass | ok... this ata sucks |
| 16:07.10 | SpaceBass | it doesnt seem to recognize all digits |
| 16:07.16 | SpaceBass | and even when it does, its not working |
| 16:08.16 | *** join/#asterisk Uther_P (~uther_p@66.180.120.83) |
| 16:08.43 | Robbster | I'm trying to install asterisk for the 1st time. I've installed the zaptel and libpri sources on a clean installation and when I try to compile asterisk I get the following error:/usr/bin/ld: cannot find -lssl |
| 16:08.59 | Moc____ | Robbster: you need openssl-devel |
| 16:09.09 | Uther_P | can you guys recommend an an Asterisk reference book? I'm looking to budget one, just to have handy |
| 16:09.28 | Robbster | Ahh, I checked that I had the ssl files, not the devel package. Thx Moc____ |
| 16:09.42 | Moc____ | Uther_P: Asterisk change too ofen, it not posible to have a book, but www.voip-info.org is your best ressources |
| 16:09.52 | BrianR___ | How does one specify order of preference for codecs in iax.conf? |
| 16:10.06 | sivana | Moc____: I got your fax, thank you |
| 16:10.19 | Moc____ | allright |
| 16:10.20 | sivana | can't read it, but I got it.. hehe |
| 16:10.22 | Uther_P | yea, I use the wiki all the time... it would be more for the other tech's reference more then mine |
| 16:10.23 | BrianR___ | It seems not to work on 1.0.6 - I specify allow=ulaw, allow=gsm and it prefers gsm.. |
| 16:11.24 | SpaceBass | ahh the root of the problem, the ata doesnt recognize the digit 4 |
| 16:11.29 | SpaceBass | what the F |
| 16:12.38 | fishboy1669 | why is there nothing on the wiki about slin codec? |
| 16:15.23 | *** part/#asterisk Robbster (~james@wblv-146-243-180.telkomadsl.co.za) |
| 16:15.48 | *** join/#asterisk DenisL (~denis@68.148.230.233) |
| 16:16.13 | Uther_P | haha, that sucks... what ata are you using? |
| 16:16.35 | *** join/#asterisk MattH (~matth@noc-wireless.chilitech.net) |
| 16:16.39 | ChkDigit | Does anyone have experience with Audio Codes MP-108 or 124 analogue to SIP adaptors? |
| 16:16.41 | MattH | where do I get mime_construct from to do faxing? |
| 16:17.15 | SpaceBass | what would cause an ATA not to pick up the digit 4? |
| 16:17.24 | johnnyb | Office pranks with asterisk: |
| 16:17.32 | DenisL | How do I assign an extension to a Zap channel? ie) so it shows up in Zap show channels. Been digging around on-line and through the asterisk book I just got, but no such luck yet. |
| 16:17.44 | sambal | Uther_P: http://stores.ebay.com/Signate-Asterisk-Store is the only book around |
| 16:17.45 | johnnyb | <PROTECTED> |
| 16:17.56 | johnnyb | <PROTECTED> |
| 16:18.04 | johnnyb | <PROTECTED> |
| 16:18.38 | johnnyb | DenisL: zap show channels will list it no matter what. |
| 16:19.05 | johnnyb | DenisL: my guess is that it wasn't detected by the driver, or you didn't compile in the zaptel drivers into asterisk. |
| 16:19.06 | DenisL | johnnyb: I do zap show channels, but no extension no's show up... |
| 16:19.26 | johnnyb | DenisL: zap show channels doesn't show extensions, unless they are in use. |
| 16:19.27 | DenisL | johnnyb: The actual channels show up. |
| 16:19.51 | SpaceBass | DenisL what are you trying to do? doubt I can help, just curious |
| 16:19.52 | fishboy1669 | is slinear related to ilbc? |
| 16:19.54 | johnnyb | DenisL: The fact is you can assign multiple extensions to the same channel, or no extension at all. |
| 16:20.52 | fishboy1669 | ? |
| 16:20.54 | _queuetue | Just going to ask one more time before I drive back to the mainland to return this PAP2... Does anyone know of a procedure to unlock this device and make it useable? |
| 16:21.15 | johnnyb | DenisL: think of an extension not as a device, but as a program. Being on extension 2003 doesn't tell me what device I'm on, it tells me which program I'm running. |
| 16:21.27 | _queuetue | I can't believe linksys sells this device without any way of using it without vonage... |
| 16:21.33 | *** join/#asterisk langals (~icechat5@196.7.14.183) |
| 16:21.34 | *** join/#asterisk Frantic (~ab@TechnologicPartners35.dsl.concentric.net) |
| 16:21.38 | DenisL | johnnyb: Perhaps I'm going about this the wrong way. |
| 16:21.41 | SpaceBass | _queuetue thats why they sell an unlocked version too |
| 16:21.44 | langals | hi there.... |
| 16:21.46 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com) |
| 16:21.51 | johnnyb | DenisL: what are you tring to do? |
| 16:21.57 | _queuetue | SpacebarApparently, they no longer do. |
| 16:22.13 | dca[laptop] | hello all, how do i hangup an iax channel from teh cli when soft hangup won't do it? |
| 16:22.20 | SpaceBass | _queuetue i've only seen it on E-bay, so I dont know... but I'd gladly pay for one right now after the nightmare this other ATA i have is |
| 16:22.21 | DenisL | johnnyb: I'm using asterisk at home, and it uses an exten-vm module in extensions_additional.conf to define the actual extensions. |
| 16:22.22 | _queuetue | SpaceBass, that was for you - I'm stuck on a windows box using Mirc and it's not taking care of me like xchat.:) |
| 16:22.38 | *** join/#asterisk bah (048830696@AC887F68.ipt.aol.com) |
| 16:22.47 | johnnyb | DenisL: sorry, never heard of extensions_additional.conf |
| 16:22.53 | DenisL | johnnyb: That macro works great for sip channels but I'm unable to make it work for a zap extension. If I don't use that macro, and just call dial then it can dial the phone but then I have no voicemail and the like. |
| 16:22.53 | langals | I am using SIP with Asterisk - the server seems to request a SIP REGISTER from the client every 15 seconds. Is this a setting somewhere that I can change? |
| 16:23.00 | SpaceBass | johnnyb its from AMP |
| 16:23.06 | _queuetue | Ok, off to return and file a complaint at Staples... Bastards. |
| 16:23.14 | sivana | heh |
| 16:23.58 | DenisL | SpaceBass: Hey, you're here again today... Thanks for your help yesterday... Figured out my contexts, the restart of the server worked I must have been going crosseyed when looking at the output cause I thought it still said from-pstn from the context... |
| 16:24.20 | johnnyb | SpaceBass, what's AMP? |
| 16:24.45 | SpaceBass | DenisL doubt I helped much :) I'm pretty clueless myself :) glad it worked out |
| 16:24.55 | SpaceBass | johnnyb web gui front end... Asterisk Management Portal |
| 16:25.03 | dca[laptop] | soft hangup not working for an iax2 channel, any thoughts? |
| 16:25.15 | johnnyb | DenisL: what does the macro look like? |
| 16:25.28 | DenisL | johnnyb: One sec... |
| 16:25.41 | SpaceBass | (pastebin.ca) |
| 16:25.56 | oej | ~seen kpfleming |
| 16:26.04 | jbot | kpfleming is currently on #asterisk (1h 26m 38s) |
| 16:26.21 | *** join/#asterisk Goshen (~Goshen@c-67-172-238-57.client.comcast.net) |
| 16:26.28 | DenisL | http://pastebin.ca/8143 |
| 16:29.00 | DannyF | anyone here running HEAD? |
| 16:29.28 | tzanger | yup |
| 16:29.47 | DannyF | WaitExten bombs, was it affected by latest changes? |
| 16:30.16 | DannyF | -- Executing WaitExten("SIP/fs_phone140-9fd9", "10") in new stack |
| 16:30.16 | DannyF | <PROTECTED> |
| 16:30.54 | DannyF | could it be the qualify ghost? |
| 16:31.22 | DannyF | nope thats not it |
| 16:32.53 | Uther_P | well. can anyone recommend a book on VoIP standards? |
| 16:33.47 | DannyF | Uther_P, found something on google "VoIP Cookbook" or somthing similiar |
| 16:33.48 | MikeJ[Laptop] | yes, voip for dummies :) |
| 16:34.22 | DannyF | ;) |
| 16:35.37 | Uther_P | try again, I'm no dummy |
| 16:35.39 | Uther_P | heh |
| 16:36.04 | Uther_P | just wanna have a reference book outlining the function and protocol |
| 16:36.26 | Uther_P | I could read the RFC's, but I'm getting the books for the other tech's to read |
| 16:36.36 | MikeJ[Laptop] | there are no really good books on the protocols themselves, not that I have seen at least... the RFC's are a rough read... |
| 16:36.51 | MikeJ[Laptop] | not at a technical level breaking out the bits and such |
| 16:42.38 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 16:43.11 | *** join/#asterisk _Vile (~vile@90.b160.bendtel.net) |
| 16:43.25 | MattH | What do I need to do to avoid having to dial 9 to get out? like just to have asterisk take the digits at face value? (can it figure out extensions and numbers)? |
| 16:44.20 | Blissex | MattH: bad question... |
| 16:44.30 | MattH | oh? |
| 16:44.37 | dca[laptop] | how do i hangup an iax channel from teh cli when soft hangup won't do it? |
| 16:44.46 | dca[laptop] | please don't say stop now |
| 16:44.55 | ChkDigit | MattH: you have to change your extensions.conf to dial direct (don't strip the first digit before dialling). |
| 16:45.01 | Blissex | MattH: Asterisk _always_ takes dialed digits at face value. |
| 16:45.19 | MattH | hrmm ok I'll have to look more at the config cause I thought I did that already |
| 16:45.25 | Blissex | MattH: it is your dialplan that tells Asterisk what to do with those digits. |
| 16:45.50 | Uther_P | dca[laptop]: I don't think there is another way to hang it up :/ |
| 16:46.14 | MattH | Blissex: so with a [dialout-default] and [dialout] (macros) which config file would those be getting called from? |
| 16:46.14 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 16:46.19 | Uther_P | MattH: exten => _NXXNXXXXXX,1,Dial(TECH/${EXTEN},20) |
| 16:46.22 | Blissex | MattH: so if youi prefix extensions with "9" in 'extensions.conf'm then you need to put in a "9"; if you don't, then you don't. |
| 16:47.27 | MattH | ok |
| 16:47.31 | MattH | so: exten => _220NXXX,1,Macro(dialout-default,${EXTEN}) |
| 16:47.36 | MattH | should use dialout-default macro, yes? |
| 16:48.00 | Uther_P | MattH: as long as you aren't stripping a digit off in that macro |
| 16:48.01 | Blissex | MattH: entirely up to you. |
| 16:48.26 | Blissex | MattH: all that Asterisk does is to match extensions with rules. |
| 16:48.31 | MattH | right got that much |
| 16:48.38 | MattH | ok .. I'll look through the configs a bit more... |
| 16:49.13 | Blissex | MattH: I suspect you need to think a bit more about how Asterisk does things, because a couple of concepts in it are not at all obvious. |
| 16:49.54 | MattH | my biggest issue is when dialing like _601Nxxxx I'm getting a 404.. which must be being caused by something elsewhere picking up the 6 ... but I dunno what and debug isn't showing much of usefulness |
| 16:50.36 | Uther_P | MattH: eh? thats 8 numbers |
| 16:50.55 | Uther_P | O_o |
| 16:51.00 | MattH | oh BAH! |
| 16:51.05 | MattH | that's why my dial-plan is screwed up |
| 16:51.05 | MattH | :P |
| 16:51.08 | Uther_P | :P |
| 16:51.09 | MattH | just needed another set of eyes |
| 16:51.12 | Uther_P | heheh |
| 16:51.19 | Blissex | MattH: I have started writing an intro to ''Asterisk concepts'' here: http://www.sabi.co.uk/Notes/linuxIAX2SIP.html |
| 16:51.27 | MattH | hrmm no it wasn't.. I just checked |
| 16:51.28 | MattH | exten => _601NXXX,1,Macro(dialout-default,${EXTEN}) |
| 16:51.31 | MattH | I typoed it wrong |
| 16:51.46 | Uther_P | okie... pastebin your dialout-default macro |
| 16:52.21 | MattH | http://www.pastebin.com/262149 |
| 16:52.40 | MattH | SOME of them are working.. with the same dialout-default like 337/323/etc.. but 601 gives me an error |
| 16:52.45 | Uther_P | MattH: by the way... the first digit of the last 4 numbers of a phone number don't have to be 2-9, they can be 0 or 1 too... just the first number of the area code and the prefix |
| 16:53.02 | MattH | so I should technically have _601XXXX ? |
| 16:53.10 | Uther_P | show me the exten of 337 and or 332 |
| 16:53.26 | MattH | exten => _337NXXX,1,Macro(dialout-default,${EXTEN}) |
| 16:53.30 | MattH | I got it... though.... |
| 16:53.36 | MattH | I think |
| 16:53.37 | Uther_P | MattH: yea, that would work, beacuse what if you tried to dial a number like 601-1356.. it wouldn't match |
| 16:53.44 | MattH | if that N is causing the issue the number 601-1xxx wouldn't work |
| 16:53.49 | MattH | and that is my issue :) |
| 16:53.57 | Uther_P | 601-1XXX or 601-0XXX |
| 16:54.03 | Uther_P | is that the case? |
| 16:54.05 | MattH | right.. got it.. thanks... |
| 16:54.10 | MattH | yuppers those are the numbers that are giving errors |
| 16:54.10 | Uther_P | hehe, no problem |
| 16:54.50 | Uther_P | MattH: are you trying to restrict people to dialing only certain area codes? if not, you can just put one entry in that says _NXXNXXXXXX |
| 16:54.54 | bannerman | hum.. I get = No one is available to answer at this time (1:0/0/0) |
| 16:54.59 | bannerman | whenever I try to dial out |
| 16:55.05 | Uther_P | MattH: and _1NXXNXXXX for long distance |
| 16:55.06 | bannerman | no dialplan changes or anything, using voipjet |
| 16:55.38 | Uther_P | bannerman: are you registered? 'sip show registry' |
| 16:55.48 | bannerman | Uther_P: using iax, and yes, registered |
| 16:55.50 | MattH | Uther_P: so N means it has to be 0 or 1? |
| 16:56.02 | Uther_P | MattH: no, N means 2-9, X means any number |
| 16:56.18 | bannerman | oh, no, I'm not registered. should I be? |
| 16:56.22 | MattH | yes I am trying to restrict people to dialing certain area codes :) |
| 16:56.29 | Uther_P | bannerman: heh, yea I would think so |
| 16:56.51 | bannerman | Uther_P: I thought registration was just if you used them for DID |
| 16:57.30 | Uther_P | I'm not positive.. I'm sure some providers don't require registration, but I know mine does |
| 16:57.44 | Uther_P | do you have a register line in your configuation? |
| 16:57.54 | jakepdev | is there anyway to debug a CAS channel that doesn't seem to come up in *? |
| 16:58.00 | bannerman | Uther_P: ok. I uncommented my register line, reloaded, successfully registered and still get the same thing |
| 16:58.04 | Uther_P | bannerman: I'm going on the notion that the IAX configuration is similar to SIP, because I've never used IAX |
| 16:58.05 | bannerman | X-lite say "403 forbidden" |
| 16:58.52 | Uther_P | bannerman: try turning on debugging for that peer |
| 16:59.05 | MattH | Uther_P: so what would you use a N for? like why use _1NXXNXXXX for LD? |
| 16:59.56 | Blissex | MattH: to prevent dialing of invalid numbers. |
| 17:00.11 | *** part/#asterisk Dandan (dandan@234.88.149.195.in-addr.arpa.virt-ix.net) |
| 17:00.12 | MattH | so basically you CAN'T have a 1 or 0 there.... |
| 17:00.29 | Uther_P | because there are no area codes or prefixes that start with 0 or 1... otherwize they would interfear with long distance, operator or international dialing |
| 17:00.34 | *** join/#asterisk mechn (~mechn@65.164.222.157) |
| 17:00.38 | MattH | Uther_P: got yah |
| 17:00.47 | Blissex | MattH: of course *you* _can_, but the telephone company won't have them. |
| 17:00.56 | MattH | right understood |
| 17:01.14 | bannerman | can I pastebin my debug output, or are the md5 thingies in there giving away my password? |
| 17:02.10 | mechn | can astrisk run without having to be the PSTN gateway |
| 17:02.27 | Blissex | mechn: sure. |
| 17:02.43 | MattH | Blissex: ok.. last question on this then... my dial rules are now working (thanks to removing that N... bah! hehe)... however .. if I dial 96011232 (which is not allowed by the dialout-default) it goes to dialout and tries to dial... where would the 9 be calling the macro from? |
| 17:02.52 | bannerman | http://pastebin.ca/8147 ... pertinent stuff from my iax2 debug |
| 17:03.09 | Blissex | MattH: your questions does not make a lot of sense... |
| 17:03.20 | mechn | will it need to have 2 eth interfaces |
| 17:03.31 | Blissex | mechn: why would it? |
| 17:03.40 | mechn | just making sure |
| 17:03.45 | Blissex | MattH: can you try to rephrase that? |
| 17:03.54 | MattH | blackjack: sorry I will try to rephrase.. in other words.. when I dial "3232166" it works now.. when I dial "6011581" it works... when I dial "96011581" it tries to dial 6011581.. where would I go to disable "9" dialing? |
| 17:04.02 | Uther_P | MattH: pastebin your dialplan |
| 17:04.15 | Blissex | MattH: define «works» |
| 17:04.31 | MattH | works.. rings the numbers |
| 17:04.51 | bannerman | haha, I figured it out. |
| 17:04.54 | Blissex | MattH: my impression, let me repeat, is that you just don't get the ''general'' concepts of how dialing workd. |
| 17:04.59 | bannerman | you have to pay the bill for it to work! |
| 17:04.59 | Uther_P | bannerman: what was it? |
| 17:05.01 | Uther_P | haha |
| 17:05.09 | Uther_P | I was about to say it looks like a problem at the provider |
| 17:05.36 | Uther_P | MattH: it has to be in the dialplan somewhere... can you pastebin your dialplan for me? |
| 17:05.48 | MattH | Uther_P: yeah I know it has to be someplace... I'm just wondering where |
| 17:05.59 | MattH | Uther_P: yeah one second |
| 17:06.00 | Blissex | MattH: you must have a rule somewhere that begins with something like "_9". |
| 17:06.29 | Blissex | MattH: you should not be asking _us_ about a dialplan _you_ have written. |
| 17:06.35 | Uther_P | heh |
| 17:06.46 | Uther_P | without at least pastebin'ing it so we can see it first' |
| 17:06.53 | *** part/#asterisk mechn (~mechn@65.164.222.157) |
| 17:06.57 | MattH | Blissex: well I didn't write it all... and I know there SHOULD be a rule but I'm not seeing it! |
| 17:07.08 | MattH | http://www.pastebin.com/262152 |
| 17:07.10 | Blissex | Uther_P: you are being _too_ helpful, as in ''spoonfeeding''. |
| 17:07.24 | Uther_P | heh, perhaps... but I don't have much better to do anyway |
| 17:07.43 | MattH | Uther_P: the only dialplans with _9's in them direct to dialout-default... |
| 17:07.58 | MattH | and they are for allowing calls to exchanges... |
| 17:07.58 | `Sauron | Grrr. |
| 17:08.04 | `Sauron | dan2, you around? |
| 17:09.08 | *** join/#asterisk Mimmus (~viggiani@ext.pitagora.it) |
| 17:09.43 | Uther_P | MattH: wierd.. I don't see anything in there that should match 9 plus a 7 digit number |
| 17:10.03 | MattH | Uther_P: same here.. I've even grepped the config directory for _9 |
| 17:10.26 | Uther_P | well.. it can be _9, _X, _N or _. |
| 17:10.34 | Mimmus | hi, is it possible to forward incoming call to a different (Microsoft) SIP server? |
| 17:10.53 | Uther_P | eeuuwww |
| 17:11.02 | MattH | ack |
| 17:11.04 | Uther_P | MattH: what context is the phone you are dialing from in? |
| 17:11.11 | Blissex | Uther_P: I worry about the two '#include's at the beginning... |
| 17:11.26 | `Sauron | Grrr. |
| 17:11.26 | Essobi | Anyone know off hand how to get who answered a multi-ring dial command into a CDR? |
| 17:11.34 | MattH | Uther_P: ahhh actually I see what the issue is... it's in another context: |
| 17:11.37 | Blissex | Mimmus: look at the 'switch' directive. |
| 17:11.39 | `Sauron | Nugget, there? |
| 17:11.39 | MattH | [outbound-trunks] |
| 17:11.40 | MattH | exten => _${DIAL_OUT_1}.,1,Macro(dialout,1,${EXTEN}) |
| 17:11.43 | Uther_P | ahh, indeed... what is in the "extensions_additional.conf and extensinos_custom.comf? |
| 17:11.45 | dan2 | `Sauron: yes, but I'm doing something extremely critical right now |
| 17:11.53 | bannerman | Uther_P: while you're in the spoonfeeding business, can you point me in the right direction for generating a ringing tone to the caller when they're waiting in the queue? |
| 17:11.56 | `Sauron | dan2: Then good luck. |
| 17:11.57 | MattH | Uther_P: some other files that are.. er commented out at the moment... but I'm not using them |
| 17:12.06 | `Sauron | critical things always go wrong |
| 17:12.17 | Blissex | MattH: then where is the definition of '${DIAL_OUT_1}' coming from? |
| 17:12.23 | Uther_P | bannerman: sorry, never done anything like that |
| 17:12.28 | bannerman | shoot. |
| 17:12.29 | Nugget | moo |
| 17:12.43 | MattH | from extension_Additional.conf |
| 17:12.44 | MattH | extensions_additional.conf:DIAL_OUT_1 = 9 |
| 17:12.44 | MattH | extensions_additional.conf:DIAL_OUT = ${DIAL_OUT_1} |
| 17:12.49 | `Sauron | Nugget: you have an austin number with your voip provider/ |
| 17:12.50 | `Sauron | ? |
| 17:12.58 | Essobi | bannerman Go read the queue.conf.sample |
| 17:13.01 | Essobi | It's in there.. |
| 17:13.03 | Blissex | bannerman: I think there is an example of that in the [demo] context of the example dialplan |
| 17:13.13 | Essobi | there's an option for MOH or ring |
| 17:13.29 | Mimmus | Blissex: no, I'd like to forward some extensions to SIP accounts on another server |
| 17:13.32 | Uther_P | MattH: well, thats it right there |
| 17:13.40 | MattH | yup... |
| 17:13.40 | Blissex | MattH: and that was one of the '#include'd files, as I suspected. |
| 17:13.47 | Nugget | I have an austin DID through voicepulse, but it has never worked. |
| 17:13.52 | `Sauron | Hum. |
| 17:13.52 | Nugget | I don't recommend them at all |
| 17:13.57 | `Sauron | I see |
| 17:14.01 | Blissex | Mimmus: well, just do it. |
| 17:14.15 | `Sauron | BV just told me my number wasn't portable.. yet |
| 17:14.15 | Essobi | Pshh.. I need to figure out how to get a answered line to show up in cdrs with you're dialing multiple ext... |
| 17:14.22 | `Sauron | So I'm wondering about other providers that actually WORK |
| 17:14.24 | MattH | Blissex: yeah.. it just took some grepping to find it |
| 17:14.33 | Uther_P | MattH: exten => _${DIAL_OUT_1}.,1,Macro(dialout,1,${EXTEN}) ${DIAL_OUT_1} == 9, so you have exten => _9,1,Macro(dialout,1,${EXTEN}) |
| 17:14.38 | Mimmus | Blissex: an example? |
| 17:14.44 | *** join/#asterisk w0w0 (~w0w0@80.26.162.27) |
| 17:14.45 | MattH | Uther_P: yeah I understand what it's doing :) I just couldn't find it earlier |
| 17:14.53 | Uther_P | heh |
| 17:15.04 | MattH | Uther_P: ahhh *sighs* it's all working as it should now... |
| 17:15.25 | MattH | Uther_P: some may say you were spoon feeding.. I see it as a learning experience... asterisk is definately a VERY complex piece of software |
| 17:15.25 | Blissex | Mimmus: well, you can just dial to the corresponding number on the other server... |
| 17:15.25 | Uther_P | heh.. ok, I'm taking flak here, so for the record, next time... .rtfm :) |
| 17:15.31 | MattH | hehe |
| 17:15.34 | MattH | thanks :) |
| 17:15.42 | Uther_P | no problem |
| 17:16.11 | Blissex | MattH: Asterisk is not so complex, there are two issues: it is underdocumented, and like most opensource stuff a lot of clueless n00bs thinks it must be damn easy to setup. |
| 17:16.13 | elriah | Is there a way to play MusicOnHOld until an agi-script is completed? |
| 17:16.26 | Mimmus | Blissex: exten => 1234,1,Dial(SIP/account@otherSIPserver,30,rt) ? |
| 17:16.46 | Blissex | Mimmus: that might well work, add the extension at the end of the ''URI''. |
| 17:17.10 | MattH | Blissex: well this is true... it's complex in that it does alot.. to be honest.. I think it's a whole lot easier then the Nortel system we have here :) |
| 17:17.20 | fishboy1669 | night guys |
| 17:17.23 | fishboy1669 | have fun |
| 17:18.06 | Blissex | MattH: problem is, since it is there and it is available, a lot of people think ''jump in''. Having a scalpel and being able to hold it does not a surgeon make :-) |
| 17:18.34 | Blissex | MattH: it happens all over the place... The most notorious place is #iptables |
| 17:19.19 | Blissex | MattH: for example things like IP routing and firewalling may require a few years of study and training, but people in #iptables assume that since the 'iptables' command is there, why not use it? :-) |
| 17:19.32 | MattH | Blissex: iptables is great... :) |
| 17:19.35 | Mimmus | Blissex: I'm logged in the SIP server as account@domain.it |
| 17:19.49 | Uther_P | Blissex: well... people gotta start somewhere |
| 17:19.50 | MattH | Blissex: but yes I agree.... it's taken me many years to learn how to use iptables right... and even now there is much I don't know about it |
| 17:20.31 | *** join/#asterisk han777 (~jwolf@adsl-64-170-149-161.dsl.sntc01.pacbell.net) |
| 17:21.45 | han777 | do i have to patch CVS 1-0-7 to get it to work with Broadvoice? I can't seem to dial out. It work before Broadvoice made there change to authname. |
| 17:23.30 | *** join/#asterisk PhilM (nwjeki@14.141.8.67.cfl.res.rr.com) |
| 17:27.16 | *** join/#asterisk cgeek (~cgeek@pl6.lawrence.org.uk) |
| 17:28.17 | habakuk | hi any providers here based out of hurricane electric? |
| 17:28.32 | guugmember | who has read Digital Fortress from Dan Brown? |
| 17:28.37 | guugmember | off topic, sorry |
| 17:29.22 | *** join/#asterisk Jearil (~Jearil@216-224-56-213.client.dsl.net) |
| 17:30.37 | *** join/#asterisk GiabboO (~GiabboOo@host101-246.pool8173.interbusiness.it) |
| 17:30.40 | GiabboO | hi everybody |
| 17:31.23 | GiabboO | i have problem with loading Voicemail configuration file into a Mysql DB. |
| 17:31.46 | GiabboO | i specified the voicemail => mysql,mydb,mytable into extconfig.conf |
| 17:32.13 | GiabboO | but when i try to enter a voicemail if channel is busy i get this: |
| 17:32.24 | GiabboO | WARNING[21858]: app_voicemail.c:2227 leave_voicemail: No entry in voicemail config file for '100' |
| 17:32.43 | GiabboO | this mean it seems to be looking for configuration into the voicemail.conf file |
| 17:32.56 | GiabboO | anybody have any idea ? |
| 17:32.59 | Uther_P | is there anyway to tell asterisk to use the hosts returned from an SRV query in order? because it seems to be trying them round robin for each re-registration |
| 17:35.16 | *** join/#asterisk Maxxed (Maxxed@pppte04-317.ght.iadfw.net) |
| 17:35.22 | Maxxed | hey'a :) |
| 17:35.37 | Maxxed | im having a problem modeprobeing the wctdm |
| 17:35.40 | Maxxed | cant find it? |
| 17:35.52 | Maxxed | iv searched the drive and its no were to be found |
| 17:35.56 | Maxxed | am i missing somthing? |
| 17:36.42 | Maxxed | i look in my /etc/modprobe.conf and its not in there |
| 17:37.01 | Maxxed | i can load the old wcfxo fine, but not the "new" module |
| 17:40.28 | *** join/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net) |
| 17:40.52 | Alexi1 | bye bye |
| 17:40.54 | *** part/#asterisk Alexi1 (~alexis@www.trim.it) |
| 17:42.00 | *** join/#asterisk stdio (~stdio@pcp09745793pcs.lncstr01.pa.comcast.net) |
| 17:42.22 | Uther_P | maybe someone knows this problem.... I'm dialing through to a SIP broadworks server. What is supposed to happen is that asterisk sends an invite, broadworks sends a 401 UNAUTHORIZED back, then asterisk is supposed to send an ACK, then another INVITE supplying the credentials... but Asterisk just keeps trying to resend the first INVITE instead of ACK'ing and resending with credentials to the first one |
| 17:42.43 | PhilM | clear |
| 17:42.48 | *** part/#asterisk PhilM (nwjeki@14.141.8.67.cfl.res.rr.com) |
| 17:45.41 | Pj386 | Anybody had their hands on the IAXy ? how do you configure it ? Plug an ethernet cable and you have a separate windows software ? or ? |
| 17:46.21 | JerJer[mobile] | windows :) |
| 17:46.24 | JerJer[mobile] | now that's funny |
| 17:46.57 | Pj386 | Well I'm asking that because I was wondering how well it would work for residential mass market |
| 17:47.22 | Pj386 | Or if it's easy to pre-setup, or.. well, wondering :) |
| 17:47.29 | *** join/#asterisk Maxxed (Maxxed@pppte03-024.ght.iadfw.net) |
| 17:47.29 | tzanger | it's not meant for the residental mass market, obviously |
| 17:47.29 | Pj386 | the data sheet don't say much |
| 17:47.35 | tzanger | needs more work to get to that level |
| 17:47.39 | Pj386 | ok |
| 17:48.06 | Maxxed | grr |
| 17:49.08 | Uther_P | does the set verbose command also change the verbosity of the debug messages? |
| 17:50.19 | *** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
| 17:50.31 | Maxxed | hey i load the wcfxs driver for fxo cards right? |
| 17:50.41 | *** join/#asterisk nix000 (~nix000@66.11.188.165) |
| 17:51.24 | JerJer[mobile] | lol |
| 17:51.31 | JerJer[mobile] | does that even make any sense? |
| 17:51.37 | Maxxed | :\ |
| 17:51.43 | tzanger | lunchtime |
| 17:51.57 | Maxxed | a TDM400P with two FXO modules in it |
| 17:51.59 | JerJer[mobile] | tzanger: smells like a good idea |
| 17:52.03 | JerJer[mobile] | modprobe wctdm |
| 17:52.03 | Maxxed | modprobe zaptel |
| 17:52.09 | Maxxed | modprobe wcfxs |
| 17:52.12 | JerJer[mobile] | um no |
| 17:52.14 | Maxxed | yeah, that dont work for me |
| 17:52.16 | JerJer[mobile] | modprobe wctdm |
| 17:52.21 | JerJer[mobile] | that is all |
| 17:52.24 | stdio | wctdm is only in cvs, no? |
| 17:52.31 | Maxxed | it must be |
| 17:52.44 | stdio | use wcfxs then |
| 17:52.50 | JerJer[mobile] | that is the only code you should be running |
| 17:52.51 | Maxxed | becuse i just compiled 1.0.6 and its not in there |
| 17:53.06 | stdio | yep, confirmed |
| 17:53.25 | Maxxed | ok, so im not 100% stupid, only 99.9999 |
| 17:53.28 | *** join/#asterisk Tili (~Tili@202-133-67-68-dialup.sat.net.pk) |
| 17:53.30 | stdio | we use wcfxs w/ out tdm400p...works fine. |
| 17:53.37 | stdio | s/out/our |
| 17:53.41 | Maxxed | ok thanks |
| 17:53.57 | JerJer[mobile] | cvs co zaptel asterisk ; cd zaptel ; make install && cd ../asterisk && make install |
| 17:53.57 | Maxxed | i read about the reverse signaling, kinda got me mixed up |
| 17:54.03 | johnnyb | stdio: do you ever have problems with people getting connected to the wrong call? |
| 17:54.06 | JerJer[mobile] | accept nothing less |
| 17:54.08 | GiabboO | i have question about voicemail, can anyone help me ? |
| 17:54.24 | stdio | johnnyb: not yet, but we aren't fully in production yet |
| 17:54.43 | Maxxed | we have two analog lines, both going into fxo modules, so just to make sure im on the right track, i modprobe zaptel, then modprobe wcfxs ? |
| 17:54.48 | stdio | jer JerJer[mobile]: you think that thing is stable enough? |
| 17:54.53 | fgravato | is something up with digium ftp |
| 17:55.02 | fgravato | kinda lagging trying to pull down g729 |
| 17:55.04 | johnnyb | stdio: we've been experiencing a lot of wierdness. We replaced our TDM card because line 3 was bad, but we may also need to replace the box itself. |
| 17:55.04 | fgravato | codec |
| 17:55.32 | stdio | johnnyb: what did you replace the tdm with? Another tdm? |
| 17:56.09 | Maxxed | lightning fryed my first tdm400p :( |
| 17:56.44 | stdio | Maxxed: And all of the mods too? |
| 17:56.59 | *** join/#asterisk point (1000@213.27.44.55) |
| 17:57.10 | Maxxed | i belive so |
| 17:57.19 | stdio | man |
| 17:57.22 | stdio | that sucks. |
| 17:57.23 | Maxxed | iv had lightning kill fax machines here too |
| 17:57.33 | Maxxed | yeah, and not in the good street cornor whore way |
| 17:57.34 | stdio | that's like 600 bucks when fully populated |
| 17:57.42 | Maxxed | yep :\ |
| 17:57.44 | stdio | heh. |
| 17:57.51 | Maxxed | hince the 20 dolla arestor |
| 17:57.53 | Maxxed | heh |
| 17:57.55 | Uther_P | you can get surge protectors for phone lines |
| 17:58.03 | Maxxed | yep |
| 17:59.25 | Maxxed | so ok ok, with the fxo modules, i use the fxs drivers right? |
| 17:59.25 | Maxxed | its been a good while since iv messed with this |
| 17:59.45 | stdio | fxs signalling, yep |
| 18:01.07 | *** join/#asterisk a-evol (a-evol@c-8870e353.045-12-6f736c1.cust.bredband.no) |
| 18:01.12 | GiabboO | anybody uses the Realtime Voicemail configuration ? |
| 18:01.13 | *** join/#asterisk ClayReiche123 (fwuser@acxexch1.accxx.com) |
| 18:01.13 | stdio | when you have an fxo card, you need to tell asterisk it's a workstation, the (s) in fxs stands for station, the (o) in fxo stands for Office, aka central office... |
| 18:02.43 | ClayReiche123 | I'm having trouble woth * using SIP and sending the proper codec preference in the INVITE packet with SDP. |
| 18:03.20 | ClayReiche123 | Seems to send my global Allows regardless of my peers allows.... |
| 18:04.17 | ClayReiche123 | specifically, it is sending ULAW and g729 as choices even though I've set disallow=all and allow=ulaw in my peer settings. |
| 18:04.23 | Uther_P | can someone help me debug my SIP problem |
| 18:04.50 | Maxxed | ... |
| 18:06.17 | Uther_P | http://pastebin.ca/8151 |
| 18:06.27 | Uther_P | Asterisk is not responding the way it should |
| 18:07.24 | *** join/#asterisk Maxxed (Maxxed@pppte03-488.ght.iadfw.net) |
| 18:07.30 | stdio | Uther_P: wow, i have no idea what I'm looking at. |
| 18:07.31 | Maxxed | freaking.. |
| 18:07.38 | Maxxed | so ok ok, with the fxo modules, i use the fxs drivers right? |
| 18:07.57 | stdio | yep |
| 18:07.58 | stdio | (12:48:25) stdio: when you have an fxo card, you need to tell asterisk it's a workstation, the (s) in fxs stands for station, the (o) in fxo stands for Office, aka central office... |
| 18:08.06 | Uther_P | when * sends the INVITE, broadworks sends back a 401 UNAUTHORIZED, then asterisk is supopsed to send back an ACK, then another INVITE with the credentials |
| 18:08.10 | Maxxed | stdio: is that a yep to me ? ;) |
| 18:08.16 | stdio | maxxed: yep |
| 18:08.22 | Maxxed | stdio: ah! thanks :) |
| 18:08.28 | Uther_P | instead asterisk just tries to do the initial invite again |
| 18:08.44 | stdio | maxxed: it's termed "signalling" though, fyi.. not drivers |
| 18:08.58 | Maxxed | ah, thanks again :) |
| 18:09.06 | ClayReiche123 | Anyone have a clue on my codec preference problem? |
| 18:09.29 | Uther_P | whats the problem with your codec preference? |
| 18:09.35 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 18:09.46 | Uther_P | does it prefer codec's of the same gender? :P heheh |
| 18:09.51 | *** join/#asterisk dsfr (~dsfr@216.207.244.183) |
| 18:09.56 | ClayReiche123 | specifically, it is sending ULAW and g729 as choices even though I've set disallow=all and allow=ulaw in my peer settings. |
| 18:10.31 | ClayReiche123 | SDP packet is sending my global "allows" and not my peer specific "allows" |
| 18:10.39 | stdio | Maxxed: odd how it's reversed like that, but it does make sense, if you think of how it works... it's similar to a client-server paradigm... asterisk serves up central office functionality when it's told to use fxo signalling, which you would use with an fxs card (fax machine or analog phone).... |
| 18:11.09 | Maxxed | whats the best way to make the modprobe zaptel and modprobe wcfxs happen on boot? /etc/modprobe.conf ? |
| 18:11.32 | Uther_P | ClayReiche123: possible those are being overrided somewhere else? like in the SIP entry or the globals? |
| 18:11.37 | Maxxed | stdio: i got cha, i kinda thought i had it, but i just wanted to make sure :) |
| 18:11.45 | stdio | Maxxed: but, in your case, you want to pretend you're a client to the pstn, which is providing central office functionality. so, you use fxs (s=station) signaling, and an fxo card. |
| 18:12.06 | Maxxed | stdio: the little animated guy tossing a ball on the digium site helped me ALOT! :p |
| 18:12.11 | stdio | hahahahahahahaha |
| 18:12.19 | Maxxed | ;) |
| 18:12.22 | Essobi | lol |
| 18:12.35 | stdio | Maxxed: depends on your dist.... |
| 18:12.45 | stdio | in slackware, rc.local.... |
| 18:12.54 | stdio | in gentoo, rc-update ? |
| 18:13.09 | Maxxed | trustix 2.2, and i jus freshly compiled the 2.6 kernel |
| 18:13.15 | Maxxed | its kinda like redhat |
| 18:13.28 | Maxxed | just real striped down..i guess |
| 18:13.29 | Maxxed | heh |
| 18:13.31 | stdio | ahhh... yep, not sure... not used to redhat... |
| 18:13.42 | Essobi | trustix is strange.. |
| 18:13.50 | Maxxed | il fumble around a bit, im sure il figure it out :) |
| 18:13.54 | stdio | you definitely want to modprobe those two before you start asterisk, though :) |
| 18:14.00 | ClayReiche123 | Uther_P: I'm forcing my device to use ONLY ULAW, I verified that in the initial invite to my *. Then my * sends the invite to our gateway, in THAT invite, it is sending 2 codec choices, ULAW and g729. The gateway sees those choices and decides to pick g729 and the call is broken. |
| 18:14.08 | Maxxed | im asterisk stupid, but i can get around with *nix |
| 18:14.13 | Maxxed | oh yeah ;) |
| 18:14.40 | stdio | Maxxed: same here :) just gimme a shell, damnit |
| 18:15.24 | Maxxed | well its not on a public network :\ |
| 18:15.33 | Maxxed | i can toss ya a shell on jus a generic box if u like :p |
| 18:15.59 | Maxxed | shitty bandwith, but not so bad performance, dual, um i think 1ghz, 512mb ram or some such |
| 18:16.12 | ClayReiche123 | Uther_P: the root question is this: If my SIP device is only giving asterisk 1 codec option, why does it turn around and give the gateway 2 codec options? sounds broken to me. |
| 18:16.33 | Maxxed | its on some flaky buisness dsl, supose to get upgraded to 1mb next month i belive |
| 18:16.42 | Uther_P | ClayReiche123: yea.. intersting |
| 18:16.55 | Uther_P | because I know asterisk isn't wanting to do the translation |
| 18:17.10 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-217-138.dsl.scarlet.be) |
| 18:17.50 | ClayReiche123 | won't do the translation. |
| 18:18.20 | ClayReiche123 | I don't want it to... I just want it to force ULAW withhout having to globally force ULAW.... |
| 18:18.34 | ClayReiche123 | I want to force ULAW on 1 extension. |
| 18:18.40 | GiabboO | <PROTECTED> |
| 18:18.45 | GiabboO | i see this when i start my CLI |
| 18:19.06 | Beirdo | ClayReiche123: because you set the codec preferences per channel |
| 18:19.10 | GiabboO | but when I try to enter a voicemail to leave a message, app_voicemail.c:2227 leave_voicemail: No entry in voicemail config file for '100' |
| 18:19.14 | Beirdo | asterisk -> device is one channel |
| 18:19.21 | Beirdo | asterisk -> gateway is another |
| 18:19.24 | GiabboO | how can I do ? |
| 18:20.21 | ClayReiche123 | Beirdo: In my sip.conf [general] disallow=all, allow=g729, allow=ulaw [8133435400] disallow=all, allow=ulaw |
| 18:20.31 | Beirdo | right |
| 18:20.52 | Beirdo | so the channel to the gateway is using the [general] as you haven't overridden it for the gateway |
| 18:21.05 | Beirdo | the channel to the device has overridden the default |
| 18:21.11 | ClayReiche123 | Beirdo: that is how I have it set... but the SDP packet seems to be taking the global setting anyway and sending both choices. |
| 18:21.22 | Beirdo | uhh |
| 18:21.40 | Beirdo | the 813 one is your device, not the gateway, correct? |
| 18:21.46 | ClayReiche123 | beirdo:exactly... I have no hair left. |
| 18:21.52 | ClayReiche123 | correct |
| 18:22.06 | Beirdo | OK |
| 18:22.17 | Beirdo | where's the setup section for the gateway? |
| 18:23.04 | *** join/#asterisk Bonbon (~bonbon@83.146.53.34) |
| 18:23.23 | *** join/#asterisk marno (~marno@213-182-122-36.teleos-web.de) |
| 18:23.32 | marno | hello together |
| 18:24.03 | ClayReiche123 | beirdo: it's just an entry in my dial plan. (extensions.conf) exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@216.229.127.60) |
| 18:24.09 | Beirdo | right. |
| 18:24.16 | Beirdo | it will use the [general] settings |
| 18:24.29 | Beirdo | if you don't want it to, add a section in the sip.conf for it |
| 18:24.31 | Beirdo | and override |
| 18:24.52 | ClayReiche123 | ahhh... thank you! |
| 18:25.04 | ClayReiche123 | wait.. |
| 18:25.18 | ClayReiche123 | I don't want ALL calls to this gateway to be ULAW. |
| 18:25.39 | Maxxed | hey at what run level would it be good to start zaptel ? |
| 18:26.04 | Uther_P | question... If I have a sip entry that is registered, and it used for my outbound calling... and another entry that has a host name used for my inbound calls.... and the SRV record of the outbound host matches the hostname of my inbound calls... might asterisk use the options from the sip entry for the inbound, since it specifies the actual host that the SRV returns? |
| 18:26.11 | Maxxed | 345 ? |
| 18:26.45 | marno | at first i only registered my sip-account for incomeing calls via "register" and everything worked fine. Then i put the outgoing configuration (peer) ontop. now everytime a calls comes in, asterisk will have (SIP/2.0 407 Proxy Authentication Required.) an registration from the proxy. So i get no incomeing calles |
| 18:26.56 | bannerman | Most numbers work just fine, but certain numbers with touch-tone menu thingies don't accept anything that I push |
| 18:26.57 | *** join/#asterisk jeffik (~jeffik@CPE0050bac711e3-CM0012256ead9e.cpe.net.cable.rogers.com) |
| 18:27.06 | bannerman | and eventually terminate with "A rating error has occured 45" in voice |
| 18:27.18 | bannerman | UPS is one of the numbers that does that |
| 18:27.34 | ClayReiche123 | bennerman:are you using a cisco gateway? |
| 18:27.48 | marno | what can i do to get saterisk to accept incomeing calls without registration? |
| 18:27.56 | bannerman | ClayReiche123: no |
| 18:28.07 | bannerman | ClayReiche: I'm using voipjet, don't know what hardware they use |
| 18:28.36 | ClayReiche123 | bennerman: sounds like they are not using the proper DTMF signaling. |
| 18:29.06 | ClayReiche123 | bannerman: or maybe your device is not... see if you have DTMF settings and play with those. |
| 18:29.11 | *** join/#asterisk Ro[b]ert (~acidburnn@cust.7.204.adsl.cistron.nl) |
| 18:29.17 | marno | any idea???? |
| 18:29.18 | bannerman | ClayReiche123: thanks, I'll check that out on my device |
| 18:29.26 | Ro[b]ert | first of all... VERY GOOD PRODUCT!!! |
| 18:29.39 | ClayReiche123 | newbie... |
| 18:29.46 | bannerman | Why thank you, Robert. |
| 18:29.49 | Ro[b]ert | now i have the following error on my console |
| 18:29.57 | ClayReiche123 | ....did I say that out loud..? |
| 18:30.24 | *** join/#asterisk wow1234 (~wow1234@w038.z064001163.sjc-ca.dsl.cnc.net) |
| 18:30.27 | jeffik | need some help setting up DID for first time |
| 18:30.39 | Ro[b]ert | erro /var/www/html/panel/safe_opserver: line 5: 2250 Terminated ./op_server.pl |
| 18:31.05 | wow1234 | do I need to patch CVS 1-0-7 for broadvoice to work? I can't seem to be able to dial out. |
| 18:31.21 | GiabboO | is it possible that the HEAD CVS have problem with voicemail realtime config ? |
| 18:31.32 | wow1234 | it work before Broadvoice made the change to authname. |
| 18:31.57 | *** join/#asterisk ACiDV (~joel@122-68-181.dr.cgocable.ca) |
| 18:32.13 | wow1234 | does anyone know??? Any help would be great!!! |
| 18:32.41 | elriah | Is there a way to play MusicOnHOld until an agi-script is completed? I saw an email with some command "SET MUSIC ON" but it doesn't seem to work. |
| 18:32.54 | *** join/#asterisk flashnet (~flashnet@200.61.65.203) |
| 18:33.18 | wow1234 | anyone using Broadvoice in this room? |
| 18:33.22 | ACiDV | I do a sip show peer {peername} and I see Callgroup => 1, 2, 33, 34 and Pickupgroup => 1, 33 but the problem is that in my sip.conf, callgroup = 1-2 and pickupgroup = 1. (latest cvs) |
| 18:34.45 | Ro[b]ert | near the bottom .. thats where they talk about DID |
| 18:34.49 | elriah | Or maybe, is there a way to turn MOH on with exten => ??? then run the agi, then turn MOH off with another priority? |
| 18:36.34 | *** join/#asterisk cbachman (~chatzilla@victory.ece.northwestern.edu) |
| 18:36.39 | wow1234 | anyone using CVS 1-0-7 with broadvoice???? |
| 18:37.21 | sivana | lol |
| 18:37.29 | Ro[b]ert | no one for the 2250 Terminated error on line 5 of safe_opserver??? |
| 18:37.41 | sivana | why is everyone having issues with Broadvoice |
| 18:37.51 | sivana | I've been using them for like ever with no problems |
| 18:38.11 | `Sauron | I don't have any problems with them |
| 18:38.17 | `Sauron | other than that they can't port my number |
| 18:38.18 | `Sauron | Grr. |
| 18:38.35 | sivana | why not? |
| 18:38.43 | bannerman | you know.. Turkey is darn good stuff... but who in the world was the first person to look at a turkey and think to themselves, "yum! I'd like to eat that!" |
| 18:39.30 | `Sauron | Dunno, they emailed me today and said that my number wasn't within their "porting footprint" |
| 18:39.44 | sivana | ya |
| 18:39.47 | arbrandes | Broadvoice kinda sucks. |
| 18:39.52 | arbrandes | No compression. |
| 18:39.53 | tzanger | mmmmmmmm my roast is awesome |
| 18:39.56 | arbrandes | Bad support. |
| 18:39.56 | sivana | depends on their CLEC coverage |
| 18:40.00 | bannerman | `Sauron: I actually can't find anyone except teliax who can port my number |
| 18:40.08 | wow1234 | sivana, what cvs version are you running? |
| 18:40.12 | `Sauron | arbrandas: It's called "growth" |
| 18:40.15 | Beirdo | tzanger even |
| 18:40.18 | tzanger | :-) |
| 18:40.18 | Beirdo | doh |
| 18:40.26 | wow1234 | i can get incoming call but no out going call. |
| 18:40.29 | arbrandes | Growth sucks. |
| 18:40.33 | sivana | wow1234: CVS HEAD |
| 18:40.45 | wow1234 | i know the config file is correct |
| 18:40.47 | Beirdo | CVS is giving head again? |
| 18:40.49 | Ro[b]ert | no one for the 2250 Terminated error on line 5 of safe_opserver??? |
| 18:40.58 | Ro[b]ert | i keep trying.... LOL |
| 18:41.04 | wow1234 | what do you mean CVS Head |
| 18:41.23 | sivana | latest version from the CVS HEAD module :) |
| 18:41.40 | sivana | the developmental version |
| 18:41.41 | wow1234 | i see....do you need to patch it. |
| 18:41.47 | sivana | patch it for what? |
| 18:42.03 | `Sauron | Ugh. |
| 18:42.12 | dca[laptop] | is there anything besides soft hangup that will kill a channel (other than stop now) |
| 18:42.12 | `Sauron | Teliax is more expensive |
| 18:42.15 | wow1234 | the broadvoice patch |
| 18:42.28 | `Sauron | It should be in cvs-head |
| 18:42.35 | `Sauron | as of 1.0.6-ish |
| 18:42.40 | sivana | wow1234: forget that patch, it's not needed anymore from what I understand |
| 18:42.52 | bannerman | Teliax scares me. During testing I had my number assigned to another customer. Took them 3 days to respond to me when I asked what was going on. |
| 18:42.54 | `Sauron | or, shortly after 1.0.6 was released |
| 18:43.08 | sivana | wow1234: what version are you running? |
| 18:43.13 | wow1234 | did you have to use authname in the sip.conf |
| 18:43.20 | wow1234 | CVS 1.0.7 |
| 18:43.20 | sivana | yes |
| 18:43.30 | `Sauron | CVS-HEAD-03/08/05-23:05:41 |
| 18:43.31 | dca[laptop] | bannerman: how the quality with Teliax? |
| 18:43.32 | sivana | look on the wiki for broadvoice setup |
| 18:43.43 | `Sauron | If your cvs is newer than that, you have the BV patch already |
| 18:43.56 | sivana | `Sauron: he's using stable |
| 18:44.05 | bannerman | dca[laptop]: Couldn't say, I'm very noob, that was my first provider and it seemed fine to me. Voipjet seems better, probably because I have 30 ping to their west coast server. |
| 18:44.07 | wow1234 | i did....it work for about 6 months without any problem until about 2 weeks ago |
| 18:44.13 | `Sauron | Ugh. |
| 18:44.17 | sivana | wow1234: then you need to add the new parameters |
| 18:44.24 | wow1234 | i did |
| 18:44.32 | wow1234 | and i still can't dial out |
| 18:44.38 | sivana | what's the error you're getting? |
| 18:44.50 | bannerman | Is there a provider with cheap 1-800 DIDs that has particularly good quality? |
| 18:44.54 | wow1234 | from the client side 503 error |
| 18:45.10 | dca[laptop] | bannerman: looks like Teliax has those |
| 18:45.43 | bannerman | dca[laptop]: Yeah, I just can't do business with people who could take up to 3 days to respond in case of total emergency. |
| 18:45.51 | bannerman | At least with nufone, I can come here and bug someone |
| 18:46.23 | ClayReiche123 | Try Volo communications. |
| 18:46.31 | dca[laptop] | hmm, i just signed up with Teliax, did the $10 thing and already got an activation email with my local number |
| 18:46.50 | sivana | wow1234: from *, what's the error msg |
| 18:47.09 | Unrea1 | maybe someone can help me out real quick. I have my asterisk FXO card installed and configured. I can dial into the pbx server once it starts fine and it does everything I need it to do. After I hang up and try to call it again it will answer and go silent |
| 18:47.12 | bannerman | dca[laptop]: yeah, i was real impressed when I signed up too |
| 18:47.14 | wow1234 | unauthorized |
| 18:47.15 | ClayReiche123 | ...AccxxVOIP is a good, new provider. 1-866-voipbox |
| 18:47.15 | Unrea1 | no error messages |
| 18:47.34 | sivana | wow1234: double check your secret with them |
| 18:48.13 | dca[laptop] | bannerman: incoming seems to be working already and sounds pretty good, nto many providers where i'm at either |
| 18:48.17 | wow1234 | it works before and when i enter that same password in x-lite it works |
| 18:48.34 | bannerman | dca[laptop]: yeah, they're one of the few who was able to offer LNP in my area |
| 18:48.37 | bannerman | actually |
| 18:48.39 | bannerman | they were the only |
| 18:48.46 | bannerman | that I could find |
| 18:48.56 | wow1234 | it's not sending the invite correctly....i think |
| 18:49.32 | *** join/#asterisk FirstSword (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
| 18:49.53 | sivana | wow1234: http://pastebin.ca/8156 |
| 18:49.56 | sivana | there's my config |
| 18:50.10 | sivana | that's all I have in sip.conf |
| 18:50.13 | sivana | for BV |
| 18:50.16 | wow1234 | ok, let me take a look at the file |
| 18:50.45 | marno | what can i do to get saterisk to accept incomeing calls without registration? |
| 18:50.50 | marno | at first i only registered my sip-account for incomeing calls via "register" and everything worked fine. Then i put the outgoing configuration (peer) ontop. now everytime a calls comes in, asterisk will have (SIP/2.0 407 Proxy Authentication Required.) an registration from the proxy. So i get no incomeing calles |
| 18:50.55 | Uther_P | I need sip help! * sends an invite, provider sends a 401 UNATHORIZED, at which time asterisk is supposed to send back an ACK, then another INVITE with the MD5 credentials... but asterisk just assumes that first 401 UNAUTHORIZED to be a failure, and errors with " Failed to authenticate on INVITE" |
| 18:51.49 | wow1234 | sivana, you not using authname but using authuser instead? |
| 18:52.03 | sivana | that's the parameter |
| 18:52.08 | sivana | means you didn't read the wiki :P |
| 18:52.37 | *** join/#asterisk Grooby (~Grooby@12.22.232.212) |
| 18:52.42 | wow1234 | i try both and it didn't work.... |
| 18:52.47 | wow1234 | let me try it now |
| 18:53.01 | bannerman | You know, I spend a lot of time on the wiki, and I'd never heard of authuser before |
| 18:53.18 | sivana | the config I pasted is a working config |
| 18:53.25 | Ro[b]ert | when i get Registration rejected... what can be the problem?? |
| 18:53.30 | sivana | bannerman: new Broadvoice requirement |
| 18:53.33 | `Sauron | on the BV wiki page |
| 18:53.41 | bannerman | oh broadvoice |
| 18:53.42 | `Sauron | there's an URL at the top to a diff wiki |
| 18:53.50 | `Sauron | follow the URL and read those instructions |
| 18:53.54 | Ro[b]ert | something to do with resolving the name? |
| 18:54.31 | GiabboO | who know what is this warning about ? app_queue.c:375 changethread: Can't change device '**Unknown**' with no technology! |
| 18:55.02 | wow1234 | sivana, same thing....can't dial out |
| 18:55.17 | Unrea1 | Timeout, but no rule 't' in context 'default' |
| 18:55.17 | Unrea1 | <PROTECTED> |
| 18:55.23 | sivana | wow1234: what's the sip entry title called |
| 18:55.29 | Unrea1 | I am guessing rule t is for terminating the call |
| 18:55.33 | Unrea1 | but what is the syntax? |
| 18:55.35 | wow1234 | sip.broadvoice.com |
| 18:55.45 | sivana | wow1234: [sip.broadvoice.com] ? |
| 18:56.03 | sivana | wow1234: did you reload? :p |
| 18:56.04 | wow1234 | sip entry title |
| 18:56.12 | wow1234 | i did a stop now |
| 18:56.42 | sivana | copy/paste your sip config to pastebin.ca |
| 18:56.55 | sivana | as well as your dialing syntax |
| 18:57.25 | wow1234 | ok |
| 18:57.39 | bannerman | maybe someone could help me with my phone entries, in sip.conf: http://pastebin.ca/8158 is the setup. It seems to me that I'd be able to setup x-lite or my phones to use the username or authuser, but for some reason the only way to connect is to use the extension |
| 18:57.45 | bannerman | it works, but it seems ugly to me |
| 18:58.37 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 18:58.37 | *** mode/#asterisk [+o denon] by ChanServ |
| 18:58.45 | sivana | bannerman: you can only dial by the [] |
| 18:58.58 | sivana | in your case [107] |
| 18:59.33 | sivana | so extension 107 |
| 18:59.54 | bannerman | sure, but for instance, X-lite has (in the SIP proxy settings) an option for username and authorization name. both of those must be 107 in order to connect |
| 19:00.16 | sivana | yes |
| 19:00.53 | sivana | I don't know why :p |
| 19:01.22 | FirstSword | hi all |
| 19:01.24 | wow1234 | sivana - http://pastebin.ca/8160 |
| 19:03.08 | sivana | wow1234: do you have a registeR? |
| 19:03.13 | FirstSword | i wonder if there's any parameter in iax.conf so that asterisk won't bridge channels |
| 19:03.15 | wow1234 | yes |
| 19:03.41 | sivana | FirstSword: what do you mean? |
| 19:03.52 | sivana | FirstSword: notransfer=yes?? |
| 19:03.57 | tzanger | no |
| 19:04.00 | tzanger | that's native transfers |
| 19:04.18 | wow1234 | when i check inside asterisk with sip show registry....broadvoice is register |
| 19:04.25 | tzanger | native bridging has no such config option IIRC |
| 19:04.32 | sivana | wow1234: do you have register statement in your sip.conf? |
| 19:04.42 | wow1234 | i able to get incoming call but no out going call |
| 19:04.46 | wow1234 | yes. |
| 19:05.02 | tzanger | native bridging is when asterisk just takes the voice frame and sends it on its merry way. option 't' or 'T' in the dialplan will make asterisk take it apart in some conditions but I don't think it does at all for IAX |
| 19:05.09 | sivana | wow1234: I don't know, like I said, mine works :) |
| 19:05.18 | sivana | wow1234: I use exten@broadvoice |
| 19:05.52 | sivana | wow1234: I couldn't get it to work using exten@sip.broadvoice.com |
| 19:06.25 | sivana | wow1234: and I also didn't double up the username/secret in the register either |
| 19:06.28 | sivana | so I dunno |
| 19:06.35 | *** join/#asterisk comfrey (~comfrey@208-151-246-153.dq1sn.easystreet.com) |
| 19:06.48 | sivana | now server 2837498 |
| 19:06.52 | sivana | serving |
| 19:07.04 | wow1234 | what do you mean exten@sip.broadvoice.com |
| 19:07.11 | tzanger | hahaha |
| 19:07.17 | sivana | wow1234: you have [sip.broadvoice.com] in your sip.conf |
| 19:07.24 | wow1234 | could i see your extension.conf |
| 19:07.25 | wow1234 | yes |
| 19:07.36 | sivana | wow1234: I have [broadvoice] |
| 19:07.52 | sivana | so when I dial, I use Dial(SIP/exten@broadvoice) |
| 19:08.39 | wow1234 | i had that before and change it to this new setting just to test it |
| 19:08.39 | sivana | about 2 wks ago? |
| 19:08.39 | wow1234 | yes |
| 19:08.39 | sivana | lol |
| 19:08.39 | sivana | change it back |
| 19:08.58 | sivana | you need those three new parameters though |
| 19:08.58 | wow1234 | so three new setting are |
| 19:09.01 | FirstSword | sivana: that asterisk station takes in a call and route to another asterisk station. i've use setgroup to try to limit the # of lines im permitting. but because the 2 channels are iax and using the same codec, they bridged together and released from the routing asterisk |
| 19:09.25 | sivana | that's notransfer=yes you want |
| 19:09.30 | sivana | right tzanger ? |
| 19:09.43 | wow1234 | what are the three new setting just to confirm with you |
| 19:09.56 | Grooby | new braodvoice requirement?!?1 |
| 19:10.01 | sivana | FirstSword: you want it to stay with the routing *? |
| 19:10.03 | `Sauron | grooby: not again |
| 19:10.06 | tzanger | notransfer will prevent asterisk from "dropping out" of the call path |
| 19:10.06 | sivana | lol |
| 19:10.20 | Grooby | sorry..i was just scrolling backup since i saw broadvoice |
| 19:10.22 | FirstSword | i have a question, when this transfer the same as the transfer in dial? like dial(ext,12,tr) |
| 19:10.26 | sivana | wow1234: you'll have to check the wiki.. it's all there |
| 19:10.30 | tzanger | FirstSword: no |
| 19:10.36 | *** join/#asterisk tomtom- (~tomtom@dD5761FDC.access.telenet.be) |
| 19:10.42 | comfrey | hey gang... i am looking to transfer numbars to a voip gateway. anyone know of gateways with good rates and lnp support? |
| 19:10.51 | sivana | comfrey: where? |
| 19:10.56 | comfrey | states |
| 19:11.03 | tzanger | FirstSword: notransfer only affects A-B-C type calls where A and C can "see" each other directly but B was used to set up the call |
| 19:11.12 | sivana | comfrey: never heard of states |
| 19:11.12 | tzanger | sivana: go get yourself a bottle of AC&Cs |
| 19:11.15 | sivana | where is that? |
| 19:11.24 | FirstSword | oic.. |
| 19:11.24 | `Sauron | Hum, outbound calls seem to work through BV, so no problems :) |
| 19:11.31 | comfrey | oh, sorry, the imperial states of america |
| 19:11.44 | sivana | comfrey: ah.. maybe a bit more specific? :) |
| 19:11.49 | comfrey | west coasr |
| 19:11.52 | sivana | lol |
| 19:11.52 | comfrey | caost |
| 19:12.03 | comfrey | nw |
| 19:12.05 | sivana | ok, that narrows it down to 123272434789 providers |
| 19:12.16 | comfrey | portland, OR |
| 19:12.34 | comfrey | sivana, are there that many providers |
| 19:12.35 | tzanger | sivana: don't be silly, it's 12488388483003 providers |
| 19:12.39 | sivana | heh |
| 19:12.45 | FirstSword | tzanger: so i should set every peers and users to notransfer if i don want the call to drop B right? wat are the disadvantages? slow response time (because of more layers)? does it take much more bandwidth though? |
| 19:12.53 | sivana | comfrey: they're everywhere |
| 19:12.53 | *** join/#asterisk l-fy (~pchitescu@l-fy.developer.yate) |
| 19:13.03 | comfrey | i have seen about 10, and of them none i have seen support lnp |
| 19:13.12 | sivana | comfrey: which ones have you seen |
| 19:13.16 | tzanger | FirstSword: disadvantage is that it's another leg in the call path, but that can be a good thing too. |
| 19:13.17 | tomtom- | hi, anyone knows the default admin password of a snom 360? |
| 19:13.31 | comfrey | but.. ok, is there a place to find a more thorough list? |
| 19:13.47 | comfrey | sivana, like voice pulse, and the like... |
| 19:13.50 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 19:13.50 | *** mode/#asterisk [+o denon] by ChanServ |
| 19:13.57 | comfrey | the ones listen on voip-info wiki |
| 19:13.58 | sivana | comfrey: actually, the best to do would be to send an email through the mailing list to find a local CLEC |
| 19:14.03 | sivana | or one as close as possible |
| 19:14.11 | l-fy | is there any girl around here? |
| 19:14.15 | sivana | lol |
| 19:14.19 | tzanger | FirstSword: and you only need it on the potential drops... for example, I have 3 VOIP providers and about 50 customers. I only have notransfer set on the 3 providers since they're the only possible destinations |
| 19:14.28 | sivana | l-fy: if you find tzanger on a Friday night |
| 19:14.28 | tzanger | l-fy: yeah but you scared her off |
| 19:14.32 | comfrey | ok, thanks sivana, i will do that |
| 19:14.33 | tzanger | hahahaha |
| 19:14.34 | sivana | hehe |
| 19:14.35 | tzanger | shush you |
| 19:14.44 | tzanger | buna seora, l-fy |
| 19:15.20 | sivana | comfrey: if you've tried all the familiar national ones, the best is to look for a local or regional one |
| 19:15.57 | FirstSword | tzanger: well. my main purpose is to limit the # of lines, but making the calls bridged can save bandwidth (i think).. is there any other sol'n to limit # of lines other than using setgroup and checkgroup? |
| 19:16.08 | tzanger | bridging does not save bandwidth |
| 19:16.20 | tzanger | bridging simply means that codec in == codec out and * doesn't have to do anything to the audio stream |
| 19:16.33 | tzanger | FirstSword: nope, setgroup and checkgroup is the proper way to do it |
| 19:16.38 | FirstSword | tzanger: oh i see. i mean drop |
| 19:16.40 | l-fy | ja right |
| 19:16.52 | l-fy | tzanger> i'm amazed :) |
| 19:16.54 | l-fy | or NOT |
| 19:16.57 | tzanger | ? |
| 19:17.01 | tzanger | why am I not amazing? |
| 19:17.07 | tzanger | everyone here seems to think so :-) |
| 19:17.15 | FirstSword | tzanger: haha |
| 19:17.16 | sivana | l-fy: I tell you, find him down TO on a friday |
| 19:17.25 | tzanger | shut up sivana or I'll tell them about your phone |
| 19:17.27 | sivana | hehe |
| 19:17.28 | l-fy | tzanger> maybe because i know to say "buna seara" since 20 years now :) |
| 19:17.40 | l-fy | anyway |
| 19:17.45 | tzanger | l-fy: I wasn't trying to impress you with that, but I *am* slowly learning it |
| 19:17.47 | `Sauron | s/since/for |
| 19:17.47 | l-fy | who wanna test a windows program..... :) |
| 19:17.56 | l-fy | ? |
| 19:17.59 | tzanger | send me the url |
| 19:18.07 | sivana | too funny |
| 19:19.01 | tzanger | how old are you anyway, l-fy? |
| 19:19.12 | tzanger | if you've known how to say that for 20 years that'd put you at... oh... 21 maybe? |
| 19:19.17 | l-fy | tzanger > http://yate.null.ro/yategui.exe - to connect to a server use diana.null.ro insted of 192.168 stuf.... |
| 19:19.26 | *** join/#asterisk juice (~juice@mo-69-68-108-44.dyn.sprint-hsd.net) |
| 19:19.26 | sivana | lol |
| 19:19.27 | Unrea1 | Some is dialing into my asterisk box externally through my FXO cards. It rings my IP phone and I can hear them but they are not able to hear me. Whyone have any ideas? |
| 19:19.35 | sivana | ya, run that executable |
| 19:20.45 | Moc____ | woohoo Im off tomorow and monday |
| 19:20.53 | tzanger | Moc____: you must work for the government ;-) |
| 19:21.02 | Darwin[laptop] | it sounds like a code to codec issue |
| 19:21.03 | Moc____ | tzanger: lawfirm |
| 19:21.04 | l-fy | sivana > the executabile is generated with jsmooth and it encapsulate a .jar |
| 19:21.11 | l-fy | if you want the sources you can grab them from cvs |
| 19:21.13 | Darwin[laptop] | translation maybeoff |
| 19:21.15 | l-fy | is a gpl software |
| 19:22.18 | jakepdev | anyone get callerid name to work w/ nufone? |
| 19:23.02 | Nugget | my nufone calls just come in as "Toll Free Call" or something like that. |
| 19:23.07 | *** join/#asterisk _tekati_ (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 19:23.08 | Nugget | I've never bothered to figure out if it's me or them |
| 19:23.13 | Uther_P | why doesn't asterisk show the debug messages of the SIP messages sent FROM my provider, only to them/ |
| 19:23.45 | Uther_P | it gives me the SIP debug message of the packet I sent out, but after that it only says "Failed to authenticate on INVITE to blah blah blah" |
| 19:23.51 | Uther_P | I wanna see the dump |
| 19:23.52 | Uther_P | GRRR |
| 19:24.23 | *** part/#asterisk akaye (~akaye@i-194-106-46-242.freedom2surf.net) |
| 19:29.33 | nix000 | anyone know if sysmaster was litigated because of their (hidden) asterisk usage ? |
| 19:30.10 | *** join/#asterisk zno (~chatzilla@ip-160-79-174-101.autorev.intellispace.net) |
| 19:31.01 | AgiNamu | nix, no, i think they just told digium "hey, we're buying hardware" and digium said "oooh, good point" |
| 19:31.08 | sivana | nix000: why would they? |
| 19:31.16 | AgiNamu | GPL violation., |
| 19:31.32 | sivana | oh ya, for bundling |
| 19:31.49 | jaiger | sivana, for bundling what? |
| 19:32.07 | sivana | Nugget: that's correct, no name on toll-free |
| 19:32.25 | AgiNamu | they take asterisk, modd'd it |
| 19:32.30 | AgiNamu | and then stick it on their servers and sell em |
| 19:32.35 | AgiNamu | but, they dont allow you to get the source. |
| 19:32.39 | AgiNamu | thus violating the GPL. |
| 19:32.42 | sivana | ya |
| 19:32.59 | bannerman | break their keyboards, imo. |
| 19:33.24 | nix000 | AgiNamu, is digium aware ? |
| 19:33.36 | sivana | nix000: yes |
| 19:33.50 | chetan | I believe they're using gnugk as well |
| 19:34.06 | nix000 | sivana, so there is no gpl violations then .. digium allowed it ? |
| 19:34.08 | l-fy | AgiNamu> who cares? |
| 19:34.25 | *** join/#asterisk loick (~loick@APuteaux-151-1-29-222.w82-124.abo.wanadoo.fr) |
| 19:34.31 | l-fy | maybe they brough a proprietary license from Digium |
| 19:34.44 | l-fy | Digium also is selling asterisk proprietary versions :) |
| 19:35.01 | Chuji | Digium reserves the right though |
| 19:35.22 | sivana | nix000: not sure, really up to Digium |
| 19:35.24 | nix000 | AgiNamu, which hardware they are buying specifically ? |
| 19:35.39 | *** join/#asterisk mrobinson (~brimstone@207.111.174.1) |
| 19:36.02 | *** join/#asterisk ikey (ikey@220.226.16.207) |
| 19:36.10 | Chuji | l-fy : the channel is made up of developers who freely contribute code to asterisk. They do not expect people to capitalize off of that withough either paying them, or continuing to release under gpl |
| 19:36.25 | nix000 | AgiNamu, i am looking at their ss7 gateway and wondering if it will work with asterisk ! |
| 19:36.41 | sivana | nix000: Digium is aware, and I'm sure an agreement/settlement has been made. Maybe not publicly, but it's their parrogative |
| 19:36.53 | l-fy | Chuji> i don't care much if Digium is selling or not Asterisk |
| 19:36.58 | Chuji | nix000 : Talk to Steve Underwood, he's got ss7 and asterisk talking |
| 19:37.02 | l-fy | not as long as i don |
| 19:37.05 | l-fy | 't use it :) |
| 19:37.14 | nix000 | Chuji, i already sent him an email. |
| 19:37.18 | Chuji | l-fy : Yes, but many others do |
| 19:37.32 | l-fy | Chuji> and in fact digium ask for disclaimers from developers |
| 19:37.48 | l-fy | so actualy to get your code into mainbranch Digium has to have the right to sell it |
| 19:37.50 | Chuji | l-fy : Digium does, but sysmaster does not |
| 19:37.57 | *** join/#asterisk JohnnyC (~JoaoCorre@81.193.116.63) |
| 19:38.27 | JohnnyC | hello |
| 19:38.33 | ikey | did any one worked on smsc project on ss7 using digium cards |
| 19:38.42 | JohnnyC | anyone can try out making an voIP call to me to try it out ? |
| 19:38.43 | l-fy | Chuji> are you intrested in moral or legal? |
| 19:39.31 | nix000 | l-fy, all i care is if they gateway will talk to asterisk ! |
| 19:39.56 | JohnnyC | can someone test out asterisk with me ? |
| 19:39.57 | Chuji | l-fy I'm a fan of the gpl in general. I have reservations about how Digium treats it, but at least they put it out in the open. |
| 19:40.08 | nix000 | sivana, no one feels betrayed because of that ? |
| 19:40.14 | GiabboO | pfff, who know what variables the app_voicemail pass to the external mail notifyer ? |
| 19:40.24 | l-fy | Chuji> in moral i see no difference bettwen Digium and sysmaster |
| 19:40.37 | l-fy | and at least the sysmaster gnugk is working the gpl dosen't |
| 19:40.52 | Chuji | l-fy : If Sysmaster and Digium had/has an agreement, I agree |
| 19:41.01 | Chuji | when this all came to light many months ago, they did not |
| 19:41.15 | Chuji | Hopefully they do now |
| 19:41.19 | l-fy | Chuji> are you from sysmaster or digium? |
| 19:41.32 | Chuji | Neither |
| 19:41.45 | l-fy | Chuji> then how do you know they don't have an agreement? |
| 19:42.17 | nix000 | l-fy, i would bet they do ! |
| 19:42.23 | Chuji | I don't know that they "do not" now. At the time this came out, it was well known that they didn't |
| 19:42.41 | l-fy | how can you be sure? |
| 19:42.46 | bjohnson | nix000: the general rule for all hardware is .. if it talks a standard that * supports, it will be usable with * |
| 19:42.53 | l-fy | do you think if they will had an agreement they will tell you? |
| 19:42.53 | zno | how do providers "terminate"? we connect to a provider like nufone or voicepulse, but how do they terminate our outgoing calls to the PSTN? |
| 19:42.54 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 19:42.54 | *** mode/#asterisk [+o denon] by ChanServ |
| 19:43.10 | l-fy | in fact the only difference is if they pay Digium or not, is not a GPL issue |
| 19:43.27 | nix000 | l-fy, that is the least they could do respecting the people who submited code |
| 19:43.39 | l-fy | nix000> pay Digium? |
| 19:43.43 | Chuji | l-fy : If they take what is GPL and use it, without paying Digium, it is a GPL issue |
| 19:43.44 | l-fy | i don't get it |
| 19:43.50 | *** part/#asterisk point (1000@213.27.44.55) |
| 19:43.52 | l-fy | no is not |
| 19:44.04 | bjohnson | zno: direct connections to telcos or passing it off to another telco that they have an agreement with |
| 19:44.07 | l-fy | they took a binary version and they didn't pay the license |
| 19:44.15 | l-fy | Digium should handle with them, why should i care? |
| 19:44.17 | nix000 | l-fy, digium has full right .. they can relicence to them ... |
| 19:44.19 | zno | bjohnson: using the iax protocol? |
| 19:44.33 | l-fy | nix000> or couse |
| 19:44.33 | Chuji | They took a GPL'd version, and changed it, and didn't release the source |
| 19:44.38 | Chuji | how is that not in violation? |
| 19:44.41 | bjohnson | zno: likely using PRIs and telco's fibrebackbones |
| 19:44.49 | bjohnson | zno: likely not voip |
| 19:45.00 | l-fy | in this case sysmaster didn't foul the comunity, they just didn't pay to Digium |
| 19:45.10 | nix000 | Chuji, i f digium gave it away (now) then what they took (now) is not gpl anymore ! |
| 19:45.23 | l-fy | Chuji> so in fact they have violated not the GPL but the digium rights |
| 19:46.19 | Nugget | no, that is not a fact. |
| 19:46.28 | nix000 | l-fy, if they violate even the gpl .. only digum can go after them . i.e. violating digium and gpl is the same |
| 19:46.42 | l-fy | i actualy doubt that Digium = GPL |
| 19:46.43 | bjohnson | this is pretty academic since I doubt any of your are enough legal experts to know what the law really would dictate |
| 19:46.49 | Beirdo | violate this! |
| 19:46.50 | l-fy | Digium is so so far away from GPL |
| 19:46.56 | Chuji | It depends on who's eyes you see it through. We who do not represent GPL, but are contributers and supporters, see it as a violation of GPL |
| 19:47.02 | l-fy | Digium is so so far away from open |
| 19:47.16 | zno | bjohnson: so most likely, from the provider's * box, it goes to some non-iax voip to PRI |
| 19:47.18 | Nugget | no, digium is not far away from open. don't be such an OSSHole. |
| 19:47.19 | bjohnson | what I DO know is what I THINK they did makes ME think they are ASSHOLES |
| 19:47.26 | l-fy | Chuji> how much code did you submited lately to asterisk, or libpri or zaptel? |
| 19:47.29 | Beirdo | Chuji: the rights to asterisk are determined by digium and digium alone |
| 19:47.35 | nix000 | l-fy, who cares in copyright term they are the sole owner IIUC |
| 19:47.44 | l-fy | bjohnson> yelling is the fouls wepon |
| 19:48.07 | l-fy | nix000 > the fact that are selling the comunity work, dosen't matter? |
| 19:48.07 | Nugget | digium owns asterisk and reserves the right to give it or sell it to anyone they wish under any license they choose. |
| 19:48.09 | Beirdo | if they are fine with it, it's not our place to complain |
| 19:48.11 | GiabboO | gtg bye all |
| 19:48.12 | *** part/#asterisk GiabboO (~GiabboOo@host101-246.pool8173.interbusiness.it) |
| 19:48.25 | Beirdo | if they aren't fine with it, it's their issue to take up with their lawyers, not ours |
| 19:48.32 | l-fy | what is the difference between digium who is selling legal the community work, and sysmaster that is doing it ilegal? |
| 19:48.42 | l-fy | Beirdo> bingo |
| 19:48.45 | Nugget | asterisk is not "the community's work", it's digium's work. |
| 19:48.45 | Chuji | My very first statement is that if Digium has an agreement with them it's all good |
| 19:48.47 | l-fy | Digium != comunity |
| 19:49.03 | Nugget | anyone who contributed code to asterisk is aware of this fact |
| 19:49.07 | nix000 | l-fy, to me it matters if i contributed code ! |
| 19:49.08 | Beirdo | "community" is irrelevant |
| 19:49.12 | bjohnson | zno: I really don't know .. but they likely connect directly to a telco service of some sort |
| 19:49.19 | l-fy | i don't care if someone take my patches for asterisk and use them, i accept that by defualt when i'm giving it to Digium, which also sell a proprietary version |
| 19:49.30 | Beirdo | nix000: if you contributed code, you agreed to their licensing methodology |
| 19:49.42 | zno | just wondering how these small companies like nufone have such broad access to all the PSTNs |
| 19:49.49 | l-fy | so in fact the name is different insted of beeing Digium who sell my code, is Sysmaster, so what? |
| 19:49.56 | l-fy | do i get anything from that? |
| 19:49.57 | l-fy | no |
| 19:49.59 | JohnnyC | Anyone available to test calling me using IP ? |
| 19:50.02 | nix000 | Beirdo, thats what i mean .. i have not contributed code .. so i cannot claim anything here. |
| 19:50.15 | Beirdo | even if you did, you couldn't :) |
| 19:50.20 | l-fy | well, we did |
| 19:50.24 | tzanger | l-fy: that's precisely WHY digium makes all code contributions signed off to them |
| 19:50.34 | l-fy | who want my patch to use it for free?, is public domain :) |
| 19:50.35 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 19:50.47 | l-fy | tzanger > i know that :) |
| 19:50.47 | jakepdev | sivana - no name on toll-free meaning if use SetCallerID with a local number, I can use a CID name? |
| 19:50.50 | nix000 | l-fy, you should read before submiting then next time ! |
| 19:51.01 | l-fy | for them to grab the money insted of sysmaster :) |
| 19:51.23 | l-fy | nix000> i've wrote my first * application 3 years ago :) |
| 19:51.28 | Beirdo | I should put up pictures of the "after" on my bottle to Te Bheag I've been drinking the last few nights :) |
| 19:52.14 | nix000 | l-fy, whocares you gave it .. it does not matter who makes money from it. but i still think dgium makes some so in essence what you wanted is done. |
| 19:52.32 | *** join/#asterisk heison (~heison@gw-yyz1.somanetworks.com) |
| 19:52.38 | l-fy | nix000 > in fatc you have no idea how little is in asterisk from digium |
| 19:52.48 | jakepdev | anyone able to get Ayava OPX to work with *? |
| 19:52.57 | Shido6 | no jakepdev, its just you |
| 19:52.59 | Shido6 | :) |
| 19:53.06 | Shido6 | dood |
| 19:53.08 | nix000 | l-fy, you are right i have NO idea ... i am the newst kid on the block |
| 19:53.09 | Shido6 | dial in using the pri |
| 19:53.11 | Shido6 | and dial back out |
| 19:53.15 | heison | does anyone know where the code for gotoifTime is? |
| 19:53.25 | jakepdev | greg - can't blame me for asking |
| 19:53.28 | Shido6 | your gear can be up and functioning, and move on to the next project, jakepdev |
| 19:53.28 | heison | i looked under apps, but can't find it |
| 19:53.31 | Shido6 | yeah |
| 19:53.32 | Shido6 | trye |
| 19:53.33 | Shido6 | true |
| 19:53.37 | l-fy | nix000 > i have somewhere a 2 years old version of asterisk |
| 19:53.39 | Shido6 | ~jbot avaya |
| 19:53.40 | jbot | i heard avaya is some big company that equals Micro$oft in phone systems |
| 19:53.41 | l-fy | maybe you want to take a look |
| 19:53.46 | Shido6 | ~jbot opx |
| 19:53.54 | Shido6 | ~jbot google opx |
| 19:54.00 | l-fy | ~jbot h323 |
| 19:54.01 | jbot | hmm... h323 is An ITU-T standard for packet-based multimedia communications systems. This standard defines the different multimedia entities that make up a multimedia system - Endpoint, Gateway, Multipoint Conferencing Unit (MCU), and Gatekeeper - and their interaction. This standard is used for many voice-over-IP applications, and is heavily dependent on other ... |
| 19:54.05 | l-fy | damn |
| 19:54.21 | l-fy | they didn't say anymore that Yate handle much better h323, then asterisk |
| 19:54.23 | l-fy | :) |
| 19:55.08 | jakepdev | I don't want to go analog - it's gotta work |
| 19:57.04 | jakepdev | isn't there any debug util that I can see whats being passed back and forth? |
| 19:57.27 | bjohnson | sip debug |
| 19:57.37 | bjohnson | or enable it in logger.conf |
| 19:57.41 | bjohnson | or set verbose 10 |
| 19:57.49 | elriah | Any AGI gurus in here? I'm trying to do a simple echo "STREAM FILE" and I can't get it to work. Using a bash script. |
| 19:58.01 | bjohnson | or sniff the network packets with ethereal |
| 19:58.07 | *** join/#asterisk SpaceBass (~sp@c-24-125-33-214.hsd1.va.comcast.net) |
| 19:58.15 | jakepdev | NJ -I'm using CAS |
| 19:58.21 | SpaceBass | arrruggg ... my two phones from e-bay arrive... with the wrong power supplies |
| 19:58.34 | SpaceBass | and radio shack doesnt carry a 24v dc supply |
| 19:58.44 | jakepdev | elriah - what's your STREAM FILE command look like? |
| 19:58.58 | Chuji | You guys ever use Max 6000's? |
| 19:59.26 | elriah | echo "STREAM FILE myfile \"#\"" |
| 19:59.33 | jakepdev | bj - so I don't think ethereal will examine the PRI or CAS packets |
| 19:59.44 | elriah | also tried echo "STREAM FILE /the/fqpath/myfile \"#\"" |
| 20:00.00 | elriah | ANd just echo "stream file audiofile" |
| 20:00.11 | jakepdev | what error do you get? |
| 20:00.29 | jakepdev | elriah - should come back with a response code |
| 20:00.30 | *** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com) |
| 20:00.34 | elriah | None. My echo "SET MUSIC ON" works fine. |
| 20:00.38 | jakepdev | 501, 200, etc. |
| 20:00.49 | elriah | In the * CLI? |
| 20:01.06 | jakepdev | if you do agi debug on, you'll see it in the CLI |
| 20:01.12 | elriah | jas |
| 20:01.47 | JohnnyC | Anyone can test Asterisk with me dialing into my extension@IPaddress ? |
| 20:02.06 | Nugget | yes, anyone can do that. |
| 20:02.19 | jakepdev | ~can you |
| 20:02.20 | elriah | 520. hrm. |
| 20:02.27 | jakepdev | jbot can't |
| 20:02.28 | jbot | Yes I can! |
| 20:02.32 | Nugget | see? :) |
| 20:02.37 | jakepdev | ok then |
| 20:02.41 | nix000 | l-fy, you can always give your code to eff ! |
| 20:02.58 | jakepdev | elriah - does it have a description |
| 20:03.20 | l-fy | eff? |
| 20:03.30 | elriah | Invalid command syntax. |
| 20:03.39 | elriah | Looks right to me per the wiki & docs.. hrm... |
| 20:03.42 | jakepdev | that's the issue |
| 20:03.44 | elriah | also tried echo "STREAM FILE /the/fqpath/myfile \"#\"" |
| 20:03.45 | jakepdev | ok |
| 20:03.52 | elriah | Yea. I see that. That's useful, tnx! |
| 20:03.54 | l-fy | nix000 > anyway making my code public also allow the aefirion guys to use it |
| 20:04.07 | *** join/#asterisk denisgalvao (~denis@linux.pesa.com.br) |
| 20:04.27 | jakepdev | STREAM FILE myfile # |
| 20:04.29 | Ro[b]ert | are there people having setup asterisk behind a NAT ? |
| 20:04.38 | Ro[b]ert | i cant seem to get registered. |
| 20:04.56 | jakepdev | Rob - registered w/ who? |
| 20:05.08 | denisgalvao | Someone know how to configure a fxs extension to use DTMF!? |
| 20:05.18 | Chuji | ~SIP+NAT |
| 20:05.36 | Chuji | ~NAT+SIP |
| 20:05.37 | jbot | [nat+sip] just fine if you have the SIP client behind NAT and Asterisk on an official IP..... |
| 20:05.37 | Ro[b]ert | jakepdev: with FWD |
| 20:06.04 | jakepdev | Rob -http://www.voip-info.org/wiki-Asterisk+FWD+NAT+Config+Example |
| 20:06.56 | jakepdev | elriah - did you see the example I put above? |
| 20:07.16 | Ro[b]ert | jakepdev: thanks.. im gonna try that.. my nat has 5060 and 8000 open... |
| 20:07.55 | jakepdev | Rob - np |
| 20:08.03 | elriah | jakepdev: Thanks. Ok, now I'm back where I started. I don't get any errors with echo "STREAM FILE myfile #", but it still doesn't play the file. Weird. |
| 20:08.14 | Unrea1 | I am definatly at a loss with this issue. I am experiancing one way calling when recieving a call through my FXO cards but not dialing from IP phone to IP phone. Where would I begin looking. |
| 20:08.29 | jakepdev | elriah - does myfile exist in your sounds folder? |
| 20:08.34 | elriah | Yep. |
| 20:08.35 | Ro[b]ert | jakepdev: do you use the include sip_additional.conf in sip.conf? |
| 20:08.39 | *** join/#asterisk W1thdraw (~Withdraw@ip70-181-96-254.oc.oc.cox.net) |
| 20:08.45 | elriah | I can play it if I do it with exten => playback.. |
| 20:08.53 | elriah | hrm... |
| 20:09.04 | jakepdev | Rob - doesn't matter as long as sip_additinal.conf is included |
| 20:09.23 | jakepdev | elriah - are you calling Answer first? |
| 20:09.27 | Ro[b]ert | k |
| 20:09.56 | elriah | No, I'm just calling that command only. It's answering the phone and running the script. I'm testing it by calling in. |
| 20:09.58 | jakepdev | Rob - to be more specific sip_additional.conf should be included in sip.conf |
| 20:10.10 | elriah | Thanks for your help, btw. |
| 20:10.39 | *** join/#asterisk W1thdraw (~Withdraw@ip70-181-96-254.oc.oc.cox.net) |
| 20:10.40 | jakepdev | elriah - np. Can you put an Answer step in your dialplan first before you call AGI? |
| 20:11.03 | jakepdev | elriah - not sure if it matters, but it worked for me |
| 20:11.17 | elriah | Sure - |
| 20:11.18 | elriah | jas |
| 20:11.29 | *** join/#asterisk W1thdraw (~Withdraw@ip70-181-96-254.oc.oc.cox.net) |
| 20:12.11 | *** join/#asterisk W1thdraw (~Withdraw@ip70-181-96-254.oc.oc.cox.net) |
| 20:12.32 | Ro[b]ert | jakepdev: isnt the iax_additional.conf the file to include those fwn rules? |
| 20:12.49 | Ro[b]ert | i followed the asteriskathomefwd wiki... |
| 20:13.06 | JohnnyC | How can I call my asterisk server by using SIP without an extension ? |
| 20:13.11 | JohnnyC | like dialing a number ? |
| 20:13.14 | elriah | Didn't make a difference. I can use echo "say number 1234" and it does say the numbers. |
| 20:13.16 | JohnnyC | dial the IP address ? |
| 20:13.17 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 20:13.17 | elriah | hrm... |
| 20:13.23 | shmaltz | helo everybody |
| 20:13.35 | jakepdev | Rob - I think FWD uses SIP, not IAX |
| 20:13.48 | Ro[b]ert | jakepdev: here is a better explanation..http://www.m-networks.net/home/asterisk/ast-fwd.htm |
| 20:14.17 | shmaltz | my provider is asking me if I need a dialtone on my PRI, or if my PBX gives me one, since this will be my first one for an * box, I don't know what to answer. I think that I don't need dialtone from provider but the PBX will give one, is this correct? |
| 20:14.22 | Ro[b]ert | jake - when i follow the instructions online it says to make a iax trunk.. |
| 20:14.30 | *** join/#asterisk Katty (~angela@68.112.15.110) |
| 20:14.30 | *** join/#asterisk dwmw2_gone (dwmw2@baythorne.infradead.org) |
| 20:14.51 | jakepdev | elriah - can you try a different file? |
| 20:14.53 | znoG | jakepdev: you can use IAX with FWD |
| 20:15.03 | elriah | I just did, same thing. Just doesn't like it. Maybe a bug with 1.0.5? |
| 20:15.15 | elriah | I swear I had this working at one point with this version. |
| 20:15.19 | jakepdev | elriah - could be - i got it to work with 1.0 |
| 20:15.55 | *** join/#asterisk Chad-wl (~asdf@207.164.188.10) |
| 20:16.02 | jakepdev | Rob - I believe the example is showing how to to get it working with SIP since your using sip.conf |
| 20:16.24 | Ro[b]ert | znoG: the only diff is that the config is done in iax.conf.???? |
| 20:16.45 | znoG | Ro[b]ert: um, yes, and you have to enable IAX in your FWD account at www.fwd.net |
| 20:16.48 | jakepdev | Rob - IAX is a different protocol requiring a different setup |
| 20:17.15 | elriah | Got an idea - I don't have a responsetimeout set before calling the agi ... |
| 20:17.18 | jakepdev | I don't think you can't just put the SIP config lines and IAX and expect it to work |
| 20:17.27 | Shido6 | zZzZZZ |
| 20:17.32 | jakepdev | elriah - it didn't require all that for me |
| 20:17.33 | Chad-wl | Is it possible to take a full T1 into a Digium card and then split 6 channels to another PRI device? I would like to get 23 channels in for phone and split 6 to our video conferencing device. |
| 20:17.45 | Ro[b]ert | jakepdev: i use this config file from wiki http://www.voip-info.org/tiki-index.php?page=Asteriskathomefwd |
| 20:18.10 | Ro[b]ert | jakepdev: there they talk about host=iax2.fwdnet.net |
| 20:18.10 | Ro[b]ert | type=peer |
| 20:18.10 | Ro[b]ert | username=123456 |
| 20:18.10 | Ro[b]ert | secret=wibble |
| 20:18.17 | Katty | beep |
| 20:18.17 | Ro[b]ert | etc etc |
| 20:18.25 | Shido6 | look at the examples in /usr/src/asterisk/configs/sip.conf.sample |
| 20:18.34 | elriah | That didn't do anything anyway. Let me put the script in default and see if it's my dialplan, possibly. Only thing I can think of. |
| 20:18.47 | jakepdev | Rob - as znog said - if you want to use IAX, make sure it's configued in FWD |
| 20:19.05 | Beirdo | Katty: you beeping at us? :) |
| 20:19.07 | Ro[b]ert | k |
| 20:19.09 | bjohnson | bop |
| 20:19.24 | bjohnson | ziddle zaddle zwoot |
| 20:19.30 | jakepdev | Greg - you couldn't find anyone? |
| 20:19.45 | jakepdev | Lucent, Avaya |
| 20:19.53 | jakepdev | , etc...? |
| 20:20.06 | Shido6 | what? |
| 20:20.18 | Chad-wl | Does anyone know the actual name of the devices they're selling on E-bay as FXO adapters? I am a reseller and would like to get one locally. |
| 20:20.22 | jakepdev | someone has got to be able to get this working with DS1FD\ |
| 20:20.32 | jakepdev | OPX |
| 20:20.36 | Shido6 | err |
| 20:20.37 | Shido6 | cas |
| 20:20.38 | Shido6 | dood |
| 20:20.40 | Shido6 | you can do cas |
| 20:20.44 | jakepdev | ah - all these damn acronyms |
| 20:20.47 | elriah | Sh*t, this sucks. I've committed getting this to work. hrmm... arrghhh!!! ;p |
| 20:20.53 | Shido6 | elriah |
| 20:20.54 | Shido6 | calm down |
| 20:21.10 | Shido6 | pastebin.ca what you have or pm me login details and u can watch me set it up |
| 20:21.14 | bjohnson | Chad-wl: you mean the x100p clones? |
| 20:21.32 | jakepdev | wish there were standard names |
| 20:21.34 | elriah | Shido6: heh, no biggie, I just need to get this proof of concept in front of some folks before they haul off and buy a 300K phone system from nortel... |
| 20:21.34 | Chad-wl | bjohnson: Ya, is there more information on them anywhere? |
| 20:21.41 | zoa | shido6, did you ever modprobe with t1e1override ? |
| 20:21.44 | bjohnson | Chad-wl: nope |
| 20:21.51 | Shido6 | ok |
| 20:21.54 | Unrea1 | Ive got 2 of them in my system |
| 20:21.57 | Unrea1 | seem to work fine |
| 20:21.57 | jakepdev | zoa - we just did that yesterday |
| 20:21.58 | Shido6 | yes zoa |
| 20:22.01 | Shido6 | I love that |
| 20:22.05 | Shido6 | I friggin love that |
| 20:22.05 | *** join/#asterisk SpaceBass (~sp@c-24-125-33-214.hsd1.va.comcast.net) |
| 20:22.09 | zoa | did you get it to work with a quad pri ? |
| 20:22.14 | zoa | it doesnt seem to work all the way |
| 20:22.18 | SpaceBass | if a phone is touted as a "standalone IP phone" what protocal does that imply? |
| 20:22.19 | Shido6 | not a quad but a single |
| 20:22.24 | zoa | i think we did it before but now it doesnt seem to do all spans |
| 20:22.35 | elriah | I'm hung up on a bash agi echo 'STREAM FILE whateverfile #', just doesn't work. No errors returned, just skips that step. If I do an echo "SAY NUMBER 1234", works fine. |
| 20:22.38 | jakepdev | ah greg - could this be the problem? |
| 20:22.38 | zoa | and of course we put the boxes 600km away |
| 20:22.39 | bjohnson | SpaceBass: nobody knows |
| 20:22.45 | jaiger | Spacebar, nothing implied |
| 20:22.57 | Ro[b]ert | jakepdev: i just activated the iax feature at fwd.... |
| 20:23.02 | Ro[b]ert | hope this works |
| 20:23.06 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 20:23.06 | SpaceBass | bjohnson was afaride of that |
| 20:23.09 | Unrea1 | Ill ask this one more time just because there seems to be more people here. Sorry if im being annoying. |
| 20:23.10 | jaiger | SpaceBass, at most tcpip is implied |
| 20:23.11 | Unrea1 | I am definatly at a loss with this issue. I am experiancing one way calling when recieving a call through my FXO cards but not dialing from IP phone to IP phone. Where would I begin looking. |
| 20:23.20 | bjohnson | SpaceBass: maybe that they have a button with a I on it and another with a P on it |
| 20:23.25 | zoa | jakepdev: how many spans did you do ? |
| 20:23.26 | zoa | all 4 ? |
| 20:23.33 | jakepdev | zoa - Shido and I did it with a single |
| 20:23.43 | SpaceBass | lol |
| 20:23.49 | Chad-wl | bjohnson: Ok, I'll try and find a Canadian vendor on E-bay. Thanks, |
| 20:23.50 | jakepdev | SB - no comments |
| 20:23.54 | SpaceBass | looks like the device in question supports h.323 at least |
| 20:24.00 | SpaceBass | cannot find anything on sip or iax2 |
| 20:24.11 | jakepdev | zoa - did switch says E1 - using as a T1 in PRI mode - works |
| 20:24.19 | jakepdev | dip switch |
| 20:24.21 | jaiger | Unrea1, depends on protocol - look at firewall/router rules |
| 20:24.39 | *** join/#asterisk ikey1 (ikey@220.226.28.201) |
| 20:24.41 | jakepdev | ok not dip switch - jumper - same difference |
| 20:24.43 | bjohnson | Unrea1: your config files and your log files |
| 20:24.54 | AgiNamu | spacebass which one? |
| 20:24.56 | bjohnson | Unrea1: then check your drivers |
| 20:25.05 | zoa | ah k |
| 20:25.24 | zoa | did you go from t1 to e1 ? |
| 20:25.26 | AgiNamu | the PA168 is the only H323 phone i knwo of. but i just dont know mnay phones :) |
| 20:25.33 | jakepdev | zoa - from E1 to T1 |
| 20:25.42 | zoa | and what exact param did you change for the t1e1 value ? |
| 20:25.44 | SpaceBass | hey AgiNamu |
| 20:25.48 | zoa | ah i need to do it the other way around |
| 20:25.51 | SpaceBass | AgiNamu the i.picasso 6000 |
| 20:25.58 | *** join/#asterisk Cheng29 (~cheng29@d57-87-253.home.cgocable.net) |
| 20:26.01 | elriah | Shido6: What's weird, all my other agi commands work just fine from this bash script. |
| 20:26.01 | AgiNamu | oh |
| 20:26.05 | SpaceBass | AgiNamu I got the 1.42 for the pa168 works great |
| 20:26.09 | jakepdev | zoa - I beleive thae param was 0 |
| 20:26.21 | SpaceBass | well, not great, the damn thing won't recognize the digit 4, but other than that |
| 20:26.36 | jakepdev | elriah - maybe it's the version -can you try another one... |
| 20:26.57 | jakepdev | zoa - 0 for E1 set to T1 override |
| 20:27.29 | SpaceBass | AgiNamu how tricky is it to set up a h.323 channel |
| 20:27.54 | zoa | and the param for t1 to e1 = 15 ? |
| 20:28.26 | elriah | jakepdev: Yea, probably. Didn't want to go that route if I didn't have to. |
| 20:28.26 | jakepdev | zoa - got me there - digium gave me the command - You can call them and they'll probably know |
| 20:28.55 | zoa | tsss, i would be a wussie if i would call em for tech support :) |
| 20:28.59 | zoa | cant do that |
| 20:29.06 | zoa | never called em |
| 20:29.09 | zoa | cant start doing it now |
| 20:29.10 | zoa | :) |
| 20:29.22 | jakepdev | zoa - it's free hardware config |
| 20:29.37 | SpaceBass | AgiNamu any reason why the pa168 wouldnt recognize some digits from my phone? |
| 20:29.55 | bjohnson | SpaceBass: this http://www.iridia.com/tvse.html seems to imply that it might be SIP |
| 20:30.00 | zoa | jake: i know |
| 20:30.16 | Katty | Beirdo: my hilight went off |
| 20:30.19 | zoa | we have 10 quad e1s and never needed to call em |
| 20:30.26 | zoa | so i dont want to do it now either |
| 20:30.32 | Beirdo | beep :) |
| 20:30.41 | SpaceBass | bjohnson unfortunatly there is not a lot out there on it... i have seen that page though... until i get my POE power supply its anyones guess |
| 20:30.44 | Beirdo | Welcome back, Katty :) |
| 20:30.57 | tzanger | Katty: you missed l-fy she was looking for you |
| 20:31.12 | zoa | hey is this katty the girl i pissed of a long time ago ? :) |
| 20:31.37 | Katty | Beirdo: i'm not here for long. |
| 20:31.41 | Katty | tzanger: why? who's she? |
| 20:31.44 | Katty | zoa: unlikely. |
| 20:31.51 | tzanger | Katty: a developer for YATE |
| 20:31.56 | SpaceBass | who thinks a 24v dc device can take 30v dc? |
| 20:31.57 | SpaceBass | i do |
| 20:31.59 | Katty | tzanger: i....see. |
| 20:32.02 | Katty | tzanger: why? |
| 20:32.38 | Beirdo | SpaceBass: BANG! |
| 20:32.38 | Katty | that must have gone right over my head. |
| 20:32.39 | Beirdo | hehe |
| 20:32.47 | SpaceBass | lol |
| 20:32.48 | SpaceBass | nahhhh |
| 20:32.57 | SpaceBass | it will just have brighter LCDs :) |
| 20:32.57 | jakepdev | zoa - only reason is I don't think that command is documented |
| 20:33.13 | zoa | i will just have someone look at the code |
| 20:33.14 | zoa | :) |
| 20:33.15 | *** join/#asterisk mrgoby (~mrgoby@pcp05308058pcs.wanarb01.mi.comcast.net) |
| 20:33.22 | zoa | we found the value 15 like that |
| 20:33.53 | mrgoby | has anyone had a problem with Redirects in the Manager API ? |
| 20:34.33 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 20:34.39 | mrgoby | it is going to the priority i specify -1, AND, it just hangs, and doesnt actually start the application on that priority definition, but just sits there and then times out in the new context |
| 20:34.50 | bjohnson | SpaceBass: http://www.gcn.com/21_3/reviews/17879-1.html this looks like it is supposed to support h323 AND SIP |
| 20:35.12 | mrgoby | using a zap channel, btw |
| 20:35.25 | *** join/#asterisk bhsx (~Owner@c-67-175-142-179.client.comcast.net) |
| 20:35.35 | *** join/#asterisk dsfr (~dsfr@216.207.244.183) |
| 20:35.56 | bannerman | How do I learn about DTMF? I have a problem where touch tone menu systems don't work (dial an 800 number, hit buttons, nothing happens while the voice drones on and on...) and I've been told it's likely that it's related to DTMF |
| 20:36.14 | Hmmhesays | SANITARIUM! |
| 20:36.25 | SpaceBass | bjohnson that's talking about the server software the manufacturer sold before they went out of business |
| 20:36.31 | mrgoby | what protocol bannerman ? |
| 20:36.42 | SpaceBass | so i'm hoping thats what the phone used too... |
| 20:36.48 | bannerman | SIP phones, IAX2 voip service throughp voipjet |
| 20:36.56 | mrgoby | oh, and also ~docs |
| 20:37.01 | mrgoby | that is |
| 20:37.04 | mrgoby | ~docs |
| 20:37.05 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 20:37.23 | mrgoby | the last one is a good place to start learning about dtmf |
| 20:37.33 | mrgoby | or, next to last rather |
| 20:37.53 | bannerman | I've spent a lot of time surfing the wiki, maybe I'm just not 1337 enough :-/ I'll try more |
| 20:38.14 | bjohnson | I've had luck setting dtmf= in the config files |
| 20:38.23 | bjohnson | 2833 I think is what I use |
| 20:38.28 | mrgoby | are you using outband what? |
| 20:38.29 | mrgoby | ok |
| 20:38.48 | bjohnson | dtmfmode=rfc2833 |
| 20:39.38 | mrgoby | so, you connect to your * with sip basically |
| 20:40.02 | mrgoby | are you having problems dialing the extens with the sip phones ? or through iax too ? |
| 20:42.50 | bannerman | mrgoby: I connect to * with my sip phones, and I can dial extens just fine |
| 20:43.06 | bannerman | its just when I dial like.. other people's phone systems, and then its only some of thme, not all of them |
| 20:43.11 | bjohnson | The Congruency i.Picasso 6000 supports H.323 and the G.729 codec http://www.tmcnet.com/it/0402/0402con.htm |
| 20:43.29 | *** join/#asterisk zipp (~zip@adsl-66-136-35-17.dsl.snantx.swbell.net) |
| 20:43.50 | *** join/#asterisk DenisL (~denis@68.148.230.233) |
| 20:43.59 | DenisL | Anybody running Asterisk on FreeBSD with a TDM400p? |
| 20:44.40 | zipp | anyone running asterisk on solaris |
| 20:44.52 | zipp | trying to get gcc working right |
| 20:47.00 | *** join/#asterisk R3DB0x (nobody@66.142.28.36) |
| 20:47.08 | CosmicRay | that sounds like a question for an entirely different channel :-) |
| 20:47.55 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
| 20:47.59 | zipp | it is, join #solaris, I asked there too |
| 20:48.20 | zipp | I figured if someone here had compiled asterisk on solaris 10, they would know what to do |
| 20:48.21 | bannerman | For instance, if I dial the 800 number for US bank, I can navigate menus just fine. When I dial some of our venders, it just ignores me when I push buttons. I hear a tone when I push the button, but apparently the other side doesn't. With specific numbers. |
| 20:50.29 | AgiNamu | zipp, someone wrote to the dev list a few days ago |
| 20:50.39 | *** part/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com) |
| 20:50.55 | AgiNamu | they were using solaris 10, and had a problem cause the c library on solaris isnt that complete |
| 20:51.06 | *** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
| 20:51.33 | *** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com) |
| 20:52.06 | *** part/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com) |
| 20:52.17 | *** join/#asterisk mrgoby (~mrgoby@pcp05308058pcs.wanarb01.mi.comcast.net) |
| 20:52.30 | tzanger | shmaltz: they want to hook you up with an MF T1?? Just say NO |
| 20:52.42 | *** join/#asterisk Teez (~Tee@cpe-69-75-233-88.san.res.rr.com) |
| 20:52.50 | harryvv | anyone here using voipjet |
| 20:53.20 | *** join/#asterisk zike (~zkm@dial81-131-152-129.in-addr.btopenworld.com) |
| 20:53.22 | tzanger | shmaltz: looks like an ni-2 PRI though... wtf is this "MF" for? |
| 20:54.00 | mrgoby | futher mucker |
| 20:54.04 | *** join/#asterisk mitmit (~pat@CPE000c418b4787-CM001225401fee.cpe.net.cable.rogers.com) |
| 20:55.19 | zipp | AgiNamu, I need to install SUNWCprog somehow |
| 20:55.21 | zipp | just have no idea how |
| 20:55.22 | zipp | ha |
| 20:56.00 | ikey1 | does any one have experiance in installing r2mfc on digium |
| 20:56.01 | ikey1 | ? |
| 20:56.19 | mitmit | hi |
| 20:56.33 | WilliamK | harryvv, I am |
| 20:56.49 | mitmit | bye |
| 20:57.56 | *** join/#asterisk {^DaNi^} (~dcp@eth1.bina-jojma.d1n.net) |
| 20:59.41 | *** part/#asterisk Teez (~Tee@cpe-69-75-233-88.san.res.rr.com) |
| 21:00.23 | *** join/#asterisk montoya (~darklab@200.195.87.218) |
| 21:01.25 | bannerman | bjohnson: that did it. I still don't really know what I'm doing by putting dtmf=whatever in my sip.conf, but it does solve my problem |
| 21:01.26 | montoya | hi all, i need the codec g729.dll compiled for firefly, where i take this? |
| 21:02.09 | *** part/#asterisk ikey1 (ikey@220.226.28.201) |
| 21:02.10 | *** join/#asterisk ORDXpres (~ordxpres@w141.z064001142.chi-il.dsl.cnc.net) |
| 21:02.25 | montoya | i have a g729 source code but i dont have c++ compiler |
| 21:03.28 | ORDXpres | Hello everyone. I have Siemens LP5100 phone. I can call internal extensions but not external phone numbers. I get Cleared - unknown - Can anyone help ? |
| 21:03.43 | Chad-wl | Is it possible to take a full T1 into a Digium card and then split 6 channels to another PRI device? I would like to get 23 channels in for phone and split 6 to our video conferencing device. |
| 21:04.09 | *** part/#asterisk TauReX (~james@colossus.trustmatta.com) |
| 21:04.27 | jakepdev | chad - can't help you but I heard there is a way |
| 21:04.57 | *** join/#asterisk rwj (~rwjblue@63.175.93.3) |
| 21:04.58 | *** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com) |
| 21:05.06 | Uther_P | damn.. I can call in to my phone through my sip provider... and it rings, and gives me the callerid info... and when I pick it up, I can hear the caller for a few seconds, then it disconnects |
| 21:05.08 | Chad-wl | jakepdev: I think so too, it makes sense that you could switch the digium card to do CO signaling for the other devices |
| 21:05.13 | Uther_P | but the caller cannot hear me |
| 21:05.33 | *** join/#asterisk jeffiku (~jeffik@CPE0050bac711e3-CM0012256ead9e.cpe.net.cable.rogers.com) |
| 21:05.34 | Uther_P | only, the calling phone doesn't indicate ringing in the earpiece until after we are disconnected |
| 21:06.10 | *** join/#asterisk SexyKen (~sexyken@c-67-161-5-149.client.comcast.net) |
| 21:06.23 | *** part/#asterisk spackle (~spackle@209.234.83.19) |
| 21:06.47 | bannerman | Uther_P: One of my phones was doing that too. I had to flash the firmware to a different version. Cheap phones :-/ |
| 21:07.03 | Uther_P | this is a sipura 2000 |
| 21:07.15 | bannerman | I was using an ariaVoice atlas |
| 21:07.32 | Uther_P | and its been working fine for everything else, I *know* it has to be a freaking problem between me and my provider |
| 21:07.41 | Uther_P | I've been working with the engineers there for about a week |
| 21:07.47 | Uther_P | and its really starting to piss me the hell of :/ |
| 21:08.01 | bannerman | I suppose you could test that by signing up for a freebie at nufone |
| 21:08.21 | bannerman | or maybe it was voipjet with the freebie |
| 21:09.12 | Shido6 | ok |
| 21:09.19 | Shido6 | now my dell keyboard is acting up |
| 21:09.32 | montoya | somebody has g729.DLL for firefly ? |
| 21:09.38 | Shido6 | right... |
| 21:10.22 | jakepdev | gotta get those old IBM keyboards - they last |
| 21:10.33 | mrgoby | damn straight |
| 21:10.53 | BrianR___ | yay. I wrote a small script to convert my phonelist into config files.. |
| 21:10.57 | jakepdev | ain't nothing like em today |
| 21:11.31 | ORDXpres | Does anyone know Siemens LP5100 (optiPoint 100 Advance) ? |
| 21:11.40 | outtolunc | http://www.virbiage.com/firefly/download/g729.zip |
| 21:12.05 | jontow | neat.. my $15 bellsouth analog phone with integrated callerid works beautifully with the IAXy :)) |
| 21:12.21 | Unrea1 | How do I set it so that I can automaticly dial and extention on the incoming call. Like when someone dials my asterisk server they can just dial an extention any time during the greeting? |
| 21:12.36 | Beirdo | I wrote a little script to convert the NPA-NXX report from dandy to a dialconfig for hylafax for myself :) |
| 21:12.45 | Ro[b]ert | jakepdev: IT WORKS... ahahahahah |
| 21:12.47 | *** part/#asterisk MikeJ[Laptop] (~icechat5@65.170.43.34) |
| 21:12.53 | Beirdo | the joys of being in Toronto. 1282 local exchanges |
| 21:12.54 | jontow | http://www.uselectronics.info/items.asp?item_id=110&variation_id=155 |
| 21:12.54 | AgiNamu | that's not the G729 dll,.... thats just to compile it. |
| 21:12.55 | jontow | that one :) |
| 21:13.01 | Unrea1 | I know its something simple but I cant find it |
| 21:13.10 | jakepdev | Rob - congrats |
| 21:13.20 | outtolunc | and that's all that he should need |
| 21:13.21 | Ro[b]ert | jakepdev: but there is one thing i dont know.. what to put in FWDRINGS ?? |
| 21:13.29 | Ro[b]ert | exten => ${FWDNUMBER},1,Dial(${FWDRINGS},20,r) |
| 21:13.42 | Ro[b]ert | this line is from extensions.conf |
| 21:13.59 | Ro[b]ert | under [from-pstn] |
| 21:14.25 | jeffiku | Bierdo: hi |
| 21:14.35 | jakepdev | FWDRINGS=sip/office ; the phone to ring - accrding to the docs |
| 21:14.49 | jakepdev | sip/office - maybe a context |
| 21:14.52 | jakepdev | don't know for sure |
| 21:14.57 | nix000 | anyone ever done a performance test on * ? |
| 21:15.26 | *** part/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com) |
| 21:15.39 | montoya | exist some softphone with support g729 default ? |
| 21:16.14 | Ro[b]ert | so i can set FWDRINGS=200 ???? |
| 21:16.20 | jakepdev | Rob - i think you can just put your extension in there |
| 21:16.39 | jakepdev | you have a phone you want it to ring when someone calls you? |
| 21:17.27 | Ro[b]ert | jes |
| 21:17.35 | Ro[b]ert | the softphone for now |
| 21:18.21 | jakepdev | I would try the softphone extension in place of FWDRINGS |
| 21:18.34 | jakepdev | then again - i'm no expert on this |
| 21:18.42 | jakepdev | (far from it) |
| 21:20.44 | Ro[b]ert | haha.. ok.. thanks. |
| 21:20.49 | jakepdev | np |
| 21:21.22 | Ro[b]ert | i just want to call myself from a pstn line... |
| 21:21.32 | Ro[b]ert | im still getting the voicemail... |
| 21:21.33 | Ro[b]ert | mmm |
| 21:23.00 | Ro[b]ert | no... now i get .. hungup... |
| 21:23.02 | Ro[b]ert | hahah |
| 21:25.34 | ManxPower | ~docs |
| 21:25.35 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 21:25.36 | ManxPower | ~mailinglist |
| 21:25.37 | jbot | rumour has it, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 21:26.03 | Beirdo | ~beer |
| 21:26.04 | jbot | i guess beer is ummm, ummm good!, or good for you! |
| 21:27.28 | *** join/#asterisk pdracevich (~bob@smtp.aucklandtax.co.nz) |
| 21:27.33 | *** join/#asterisk chetan (freetibet@24-193-188-21.nyc.rr.com) |
| 21:28.02 | pdracevich | hello, all does any one know of a good Call Conferrencing app for asterisk that is not meetme? |
| 21:28.16 | *** join/#asterisk queuetue (~Scott@h69-21-252-54.69-21.unk.tds.net) |
| 21:28.16 | pdracevich | and how do i get ChanSpy |
| 21:28.19 | *** join/#asterisk dersteer (~travis@24.231.151.119.gha.mi.chartermi.net) |
| 21:28.26 | bjohnson | nix000: astertest |
| 21:28.35 | bjohnson | ~port |
| 21:28.36 | jbot | extra, extra, read all about it, port is To port something, you translate the code for a program from one platform to another. You could port a program you wrote on a PC over to a Macintosh, for example. Port |
| 21:28.42 | *** join/#asterisk PhilM (nwjeki@14.141.8.67.cfl.res.rr.com) |
| 21:28.52 | bjohnson | wrong |
| 21:28.59 | AgiNamu | well... I got the PA168 firmware modified... now it responds to POKEs. |
| 21:29.03 | AgiNamu | but now it wont make calls :@ |
| 21:29.21 | *** join/#asterisk chetan (freetibet@24-193-188-21.nyc.rr.com) |
| 21:29.38 | PhilM | I hear there is g729a for Freebsd, where can I find it? |
| 21:29.45 | JohnnyC | Anyone can help with FWD ? |
| 21:29.57 | JohnnyC | I can call the Echo test number but I cant hear my echo |
| 21:30.01 | JohnnyC | my voice |
| 21:30.27 | bjohnson | johnnyb: can you get anything from the time number? |
| 21:30.32 | AgiNamu | it wouldnt be that bad devving for this |
| 21:30.38 | queuetue | Hi - can someone point to a good tutorial to asterisk for a long-time unix/linux admin and programmer? I don't need to learn a lot of remedial stuff about networking or what the CLI is, but I have absolutely zero knowledge of telco interfaces, codecs, pbxes, in general, or asterisk in particular... |
| 21:30.42 | AgiNamu | but I dont have a test harness... i gotta make new firmware, then upload it |
| 21:30.44 | AgiNamu | takes a few minutes |
| 21:30.55 | pdracevich | hello, all does any one know of a good Call Conferrencing app for asterisk that is not meetme? |
| 21:31.01 | bjohnson | ~docs |
| 21:31.02 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 21:31.05 | bjohnson | queuetue: ^ |
| 21:31.09 | *** join/#asterisk chetan (freetibet@24-193-188-21.nyc.rr.com) |
| 21:31.25 | JohnnyC | bjohnson: whats the time number ? |
| 21:31.31 | queuetue | bjohnson, thanks. |
| 21:32.34 | bjohnson | queuetue: it's easier if you playing. you can get a working voip system with nothing but a linux box to play with it |
| 21:33.46 | *** join/#asterisk chetan (freetibet@24-193-188-21.nyc.rr.com) |
| 21:34.21 | pdracevich | hello, all does any one know of a good Call Conferrencing app for asterisk that is not meetme? |
| 21:34.53 | jontow | whats ill with meetme? |
| 21:36.21 | pdracevich | I dont have a Zap card :( |
| 21:36.28 | jontow | so use ztdummy |
| 21:36.29 | doughecka | dont need one |
| 21:36.35 | jontow | all conferencing is going to require timing sources..? |
| 21:36.43 | *** join/#asterisk marno (~marno@213-182-117-45.teleos-web.de) |
| 21:36.54 | pdracevich | ztdummy? |
| 21:36.59 | doughecka | ~google ztdummy site:asterisk.org |
| 21:37.12 | doughecka | huh |
| 21:37.20 | doughecka | ~google ztdummy site:lists.digium.org |
| 21:37.21 | *** join/#asterisk Thus0 (~Thus0@dyn-83-152-162-198.ppp.tiscali.fr) |
| 21:37.27 | bjohnson | pdracevich: yes |
| 21:37.33 | doughecka | time to reboot |
| 21:38.06 | *** join/#asterisk chetan (freetibet@24-193-188-21.nyc.rr.com) |
| 21:39.56 | *** join/#asterisk chetan (freetibet@24-193-188-21.nyc.rr.com) |
| 21:40.22 | *** part/#asterisk ORDXpres (~ordxpres@w141.z064001142.chi-il.dsl.cnc.net) |
| 21:40.26 | pdracevich | I have tryed that this is one of my many problums, when I complie I get an error saying ztdummy.c:219: confused by earlier errors, bailing out |
| 21:40.27 | pdracevich | make: *** [ztdummy.o] Error 1 |
| 21:40.47 | jontow | ok |
| 21:40.52 | jontow | then you need to figure out WHY it is erroring out |
| 21:40.54 | mrgoby | why dont redirects work ? |
| 21:40.57 | jontow | it compiles cleanly on every linux system i've tried |
| 21:41.00 | mrgoby | am i doing something wrong here ? |
| 21:41.02 | *** join/#asterisk p1tst0p (~will@82-38-104-189.cable.ubr03.donc.blueyonder.co.uk) |
| 21:41.03 | harryvv | What is a common cause of this. |
| 21:41.03 | pdracevich | and when i look thourgth the listing i see /usr/include/linux/modversions.h:1:2: #error Modules should never use kernel-headers system headers, |
| 21:41.07 | harryvv | -- Executing Dial("SIP/152-6df7", "IAX2/5555@voipjet/12144465311") in new stack |
| 21:41.07 | harryvv | <PROTECTED> |
| 21:41.07 | harryvv | <PROTECTED> |
| 21:41.07 | harryvv | <PROTECTED> |
| 21:41.07 | harryvv | <PROTECTED> |
| 21:41.08 | harryvv | <PROTECTED> |
| 21:41.15 | xkev | mrgoby must be, because they work :) |
| 21:41.23 | mrgoby | have you used them ? |
| 21:41.37 | harryvv | Trying to figure this out |
| 21:41.40 | jontow | perhaps you are missing USB support in your kernel.. or you are missing CRC32 in your kernel, or something similar |
| 21:41.46 | mrgoby | it is always taking me to priority-1 and always just hangs on the channel |
| 21:41.48 | harryvv | called known numbers that work on voipjet. |
| 21:41.52 | jontow | harryvv; iax2 show peers ? |
| 21:41.55 | xkev | mrgoby, manager redirect right? |
| 21:41.57 | harryvv | hold |
| 21:42.50 | harryvv | voipjet/5555 216.118.117.46 (S) 255.255.255.255 4569 OK (118 ms) |
| 21:42.50 | harryvv | 1 iax2 peers [1 online, 0 offline, 0 unmonitored] |
| 21:43.06 | harryvv | 5555 is not my account number that is a ficticious one. |
| 21:43.34 | bjohnson | 118 is normal for you? |
| 21:43.44 | *** join/#asterisk fenlander_ (~irc@82.152.81.57) |
| 21:43.58 | *** part/#asterisk loick (~loick@APuteaux-151-1-29-222.w82-124.abo.wanadoo.fr) |
| 21:44.16 | mrgoby | xkev: yes |
| 21:44.20 | *** join/#asterisk montag___ (~montag@81-174-23-160.f5.ngi.it) |
| 21:44.29 | mrgoby | xkev: have tried both head and stable |
| 21:44.29 | xkev | I use them extensively, for an operator panel we cooked up in python and C++ |
| 21:44.32 | elriah | Anybody else here running the debian asterisk 1.0.5 package? |
| 21:44.51 | montag___ | hi, i've buy a PLANET VIP-152T phone, but the client arbitraty unregister and end the call after 3-5 minutes, any tips ? |
| 21:45.04 | xkev | Action: Redirect; Channel: Zap/1-1; Context: foo; Extension: 100; Priority: 1 \n\n |
| 21:45.13 | xkev | are you specifying a priority |
| 21:45.16 | mrgoby | yeah |
| 21:45.24 | mrgoby | and it always goes to priority-1 |
| 21:45.27 | mrgoby | using telnet |
| 21:45.33 | xkev | crank verbose up, does it show it falling back to priority 0 |
| 21:45.33 | mrgoby | have tried with php too |
| 21:45.42 | mrgoby | no |
| 21:45.47 | mrgoby | i used show channels |
| 21:45.52 | xkev | so you say Prioity: 400 and it does to 399? |
| 21:45.53 | mrgoby | and it shows me goingg to that channel |
| 21:45.59 | mrgoby | and that priority+1 |
| 21:46.00 | mrgoby | yep |
| 21:46.04 | mrgoby | sorry minus 1 |
| 21:46.10 | mrgoby | and then it just hangs |
| 21:46.18 | mrgoby | i could deal with just adding one to the priority |
| 21:46.20 | mrgoby | nobiggie |
| 21:46.26 | mrgoby | but it hangs there, and then times out |
| 21:46.29 | xkev | that's not normal |
| 21:46.32 | mrgoby | i know |
| 21:46.52 | mgth | Does the ILEC or the CLEC provide phone lines to your place of work? |
| 21:46.52 | mrgoby | this is on an EPIA mini-itx, btw.... i should try on my dell |
| 21:47.07 | xkev | I would say kludge around it by starting your actual action with a gap, and have both 2 and 1 jump to there, so if it fixes itself you won't break. |
| 21:47.07 | mrgoby | though i still dont see why it would do that |
| 21:47.11 | mrgoby | everything else works great |
| 21:47.12 | xkev | it makes no sense |
| 21:47.24 | pdracevich | When I try to complie ztdummy /usr/include/linux/proc_fs.h:193: warning: `create_proc_read_entry' declared `static' but never defined |
| 21:48.05 | mrgoby | what do you mean "gap" and "2 and 1 jump there" ? |
| 21:49.17 | *** join/#asterisk LeoB ([U2FsdGVkX@wireless-79.media.mit.edu) |
| 21:49.34 | mrgoby | shoot, i gotta go pick up my gfriend, i'll be back in a bit... gotta get this working tonight :-P |
| 21:49.39 | mrgoby | thanks for your help |
| 21:49.46 | queuetue | I've set up a broadvoice account, and setup sip.conf with the registration and (I think) correct configuration. When I launch *, I can see the sip registration, and when I call the number from a standard phone, I can see the SIP channell appear. I get the broadvoice voicemail at that point and the channle quickly dissapears from *. Where do I go from here - including a) setting up and getting *my* voicemail to pickup instead of broad |
| 21:49.46 | queuetue | voice's , b) getting it to ring on kphone on my desktop, and c) allowing kphone to dial out through it. |
| 21:49.52 | *** join/#asterisk wow1234 (~wow1234@w038.z064001163.sjc-ca.dsl.cnc.net) |
| 21:49.54 | pdracevich | When I try to complie ztdummy /usr/include/linux/proc_fs.h:193: warning: `create_proc_read_entry' declared `static' but never defined |
| 21:50.50 | wow1234 | could someone tell me why I'm getting "SIP/2.0 404 Not Found" but it show me as being register |
| 21:50.57 | jluk | ~jbot pastebin |
| 21:50.58 | jbot | it has been said that pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 21:51.09 | queuetue | There are just so many docs, and I have no idea what portions I have to learn to get started. |
| 21:51.09 | LeoB | hi, can anyone help me connect Asterisk, FWD and SJphone? |
| 21:51.29 | bjohnson | queuetue: is your * behind a nat router? |
| 21:51.46 | pdracevich | When I try to complie ztdummy /usr/include/linux/proc_fs.h:193: warning: `create_proc_read_entry' declared `static' but never defined |
| 21:51.48 | bjohnson | LeoB: yes |
| 21:51.53 | bjohnson | ~docs |
| 21:51.55 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 21:51.56 | bjohnson | LeoB: ^ |
| 21:51.59 | queuetue | bjohnson, yes, it is. Is that a problem? |
| 21:52.02 | Nugget | LeoB: work through http://voip-info.org/ and come to the channel when you have specific questions. |
| 21:52.06 | bjohnson | queuetue: yes |
| 21:52.17 | queuetue | bjohnson, Even with port-forwarding? |
| 21:52.20 | bjohnson | queuetue: I've never been able to get that to work |
| 21:52.37 | wow1234 | could someone tell me why I'm getting "SIP/2.0 404 Not Found" but it show me as being register |
| 21:52.40 | bjohnson | queuetue: others claim to have it working |
| 21:52.53 | wow1234 | can't seem to dial out but can get incoming call |
| 21:52.56 | bjohnson | queuetue: does bv offer a stun server option? |
| 21:53.03 | *** join/#asterisk mitmit (~mitmit@CPE000c418b4787-CM001225401fee.cpe.net.cable.rogers.com) |
| 21:53.16 | mitmit | hi |
| 21:53.21 | bjohnson | wow1234: do you have it defined as a user? |
| 21:53.51 | wow1234 | bjohnson, as a peer |
| 21:54.02 | bjohnson | wow1234: peer is for incoming calls |
| 21:54.10 | wow1234 | i even try it as a friend |
| 21:54.10 | bjohnson | wow1234: user is for outgoing |
| 21:54.30 | wow1234 | how do you defined as a user |
| 21:54.34 | bjohnson | type=friend often has problems unless you get lucky |
| 21:54.46 | wow1234 | so type=user |
| 21:54.57 | bjohnson | and whatever other config bv needs |
| 21:55.05 | bjohnson | likely some info on bv or on the wiki |
| 21:55.10 | bjohnson | I don't use bv |
| 21:55.23 | wow1234 | let me try it right now |
| 21:57.33 | bjohnson | I'll wait here |
| 21:58.11 | wow1234 | same thing...can't dial out |
| 21:59.31 | pdracevich | when i try to build ztdummy i get /usr/include/linux/modversions.h:5:2: #error to build against the currently-running kernel. |
| 22:00.18 | wow1234 | bjohnson, now i get 407 Proxy Authentication Required.... |
| 22:00.43 | bhsx | hi, i'm sorry, i'm total n00b, but i have a generic x100p and asterisk@home installed. i have the softphone side all working and can call to each other... how do i use the x100p to answer my POTS line? and how do i use the softphones to access POTS? |
| 22:01.01 | file[laptop] | so very very cold |
| 22:01.05 | tzanger | bhsx: you need to set it up in zapata.conf (and zaptel.conf) |
| 22:01.14 | tzanger | and then create a context for incoming calls to go to |
| 22:01.18 | bhsx | can i do it in AMP? |
| 22:01.19 | tzanger | I would suggest something like [pots] |
| 22:01.22 | tzanger | bhsx: no idea |
| 22:01.34 | *** join/#asterisk crich1999 (~crich@217.9.52.18) |
| 22:01.38 | tzanger | I'd consult the *@~ documentation |
| 22:01.47 | wow1234 | does any one get this error message with broadvoice - 407 Proxy Authentication Required |
| 22:02.44 | pdracevich | when i try to build ztdummy i get /usr/include/linux/modversions.h:5:2: #error to build against the currently-running kernel. |
| 22:04.04 | *** join/#asterisk _queuetue (~Scott@h69-21-252-54.69-21.unk.tds.net) |
| 22:04.43 | _queuetue | Oops - Lost everything sfter bjonson asked "does bv offer a stun server option?" - firewall screwup. :) |
| 22:04.59 | bjohnson | wow1234: did you check the wiki or bv help yet? |
| 22:05.00 | AgiNamu | YEY! Got POKE support running on PA168! |
| 22:05.56 | wow1234 | bjohnson, yes....i try everthing.i was working great for about 6 months now then BV made somes changes on there end and now it stop working |
| 22:06.32 | wow1234 | running CVS 1-0-7 |
| 22:06.46 | bjohnson | I think I've got you guys mixed up .. wow1234 isn't trying to use bv |
| 22:07.11 | wow1234 | bv - broadvoice, correct. |
| 22:07.29 | _queuetue | bjohnson, does the fact that the suip registration works and the channel appears when I make the call indicate that the firewall is not causing problems, and the sip->broadvoice connection is working? |
| 22:07.51 | pdracevich | when i try to build ztdummy i get an error /usr/include/linux/modversions.h:5:2: #error to build against the currently-running kernel.? HELP PLEASE! |
| 22:08.52 | harryvv | bj, did you say you had voipjet as a carrier? |
| 22:09.26 | mrgoby | you there xkev ? |
| 22:11.21 | jesster | Im troubleshooting a phone hanging off a FXS port of a channel bank. When the phone is in use, and another call comes in, asterisk cli shows "Zap/106-busy-678259056 is busy" but the caller does not hear a busy signal - it just rings. The callee does not hear a call waiting beep. |
| 22:12.06 | harryvv | This means there are some issues with voipjet? -- Executing Dial("SIP/152-b0c6", "IAX2/1230@voipjet2/12144465300") in new stack |
| 22:12.07 | harryvv | <PROTECTED> |
| 22:12.07 | harryvv | <PROTECTED> |
| 22:12.07 | harryvv | <PROTECTED> |
| 22:12.07 | harryvv | <PROTECTED> |
| 22:12.08 | harryvv | <PROTECTED> |
| 22:12.33 | *** join/#asterisk Uther_P (~uther_p@66.180.120.83) |
| 22:12.36 | Exstatica | i keep getting an error... db.c:177 ast_db_get: Unable to find key '5625551212' in family 'SIP/Registry' |
| 22:12.42 | Exstatica | i have no idea what it means |
| 22:12.53 | *** join/#asterisk Karl_H (~karl@69.177.93.20) |
| 22:13.25 | mrgoby | ~seen xkev |
| 22:13.33 | jbot | xkev is currently on #asterisk (2d 1h 42m 41s). Has said a total of 57 messages. Is idling for 26m 21s |
| 22:13.42 | pdracevich | when i try to build ztdummy i get an error /usr/include/linux/modversions.h:5:2: #error to build against the currently-running kernel.? HELP PLEASE! |
| 22:13.55 | harryvv | I think voipjet has a issue with canada :) |
| 22:14.04 | Beirdo | voipjet has issues |
| 22:14.06 | Beirdo | period |
| 22:14.58 | harryvv | yea. Beirdo, so you seen that one before the no one is aviable to answe? |
| 22:15.11 | bjohnson | wow1234: _queuetue: sounds like you two are both trying to get broadvocie to work. help each other out |
| 22:15.32 | Beirdo | that will mean the other end is busy, I think |
| 22:15.46 | bjohnson | harryvv: I have an accoutn with them and used them for a while but don't use them much now cause they keep giving me huge lags time |
| 22:15.50 | wow1234 | _queuetue, what version of CVS are running |
| 22:15.56 | harryvv | ohh |
| 22:16.09 | harryvv | which of the three servers do you use? |
| 22:16.22 | bjohnson | Exstatica: are you trying to use rsa auth? |
| 22:16.48 | pdracevich | zaptel.c:6225: storage size of `zt_fops' isn't known |
| 22:16.48 | pdracevich | make: *** [zaptel.o] Error 1 |
| 22:16.54 | bjohnson | harryvv: 216.118.117.46 |
| 22:16.59 | harryvv | yea same thing |
| 22:17.02 | pdracevich | these are the errors, please help |
| 22:17.23 | bannerman | bjohnson: I don't have any lag problems with Voipjet's left coast server. maybe some quality issues, can't tell if that's my setup or not though, I'm newb |
| 22:17.29 | bjohnson | harryvv: noormally 42ms but about 2 times and hour it goes over 2000ms |
| 22:17.38 | Beirdo | bjohnson: yeah same here |
| 22:17.44 | Beirdo | they su-diddley-uck |
| 22:17.46 | bjohnson | which is left coast? |
| 22:17.55 | harryvv | I created a iax.conf context with the west coast server and a following extention to match it. And did get a bussy message with a bussy tone. |
| 22:17.59 | bjohnson | from up here left is NY |
| 22:18.16 | CosmicRay | I refuse to use voipjet because of: http://lists.digium.com/pipermail/asterisk-users/2005-March/094229.html |
| 22:18.30 | bannerman | 69.25.60.30 .. my ping there was 30 earlier this month, it's up to 50 now |
| 22:18.45 | bjohnson | I refuse to use vonage .. but doesn't mean that others have the same sense |
| 22:18.51 | Beirdo | left coast is west coast |
| 22:18.57 | *** join/#asterisk mgth (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 22:18.57 | Beirdo | look on a map already :) |
| 22:18.58 | CosmicRay | huh, 112ms for me, 15ms more than livevoip |
| 22:19.00 | *** join/#asterisk Flyboy6440 (~Bobo@192.76.82.89) |
| 22:19.05 | Exstatica | bjohnson, not that i know of |
| 22:19.09 | CosmicRay | bjohnson: well yes, refusing to use vonage goes without saying :-) |
| 22:19.30 | Beirdo | I said look at a map.... |
| 22:19.37 | Flyboy6440 | is there a firmware update to allow iaxy to use dns? |
| 22:19.46 | CosmicRay | heh, iaxy doesn't use dns? |
| 22:19.58 | jontow | cosmicray; no. |
| 22:20.00 | Flyboy6440 | not that i cant tell.. just got it.. |
| 22:20.14 | CosmicRay | I was plesantly surprised recently that my spa-841 supports NTP |
| 22:20.23 | CosmicRay | and also that I can configure them all centrally by placing xml files on tftp servers |
| 22:20.24 | jontow | 'tis one of the drawbacks of keeping your entire 'OS' image in 4kb :) |
| 22:20.48 | _queuetue | wow1234, Not running form CVS. |
| 22:20.56 | harryvv | cosmic the spa 1000-2000 series? its worked 100% for me. |
| 22:21.00 | znoG | CosmicRay: i don't think there is any need to worry too much about their ToS. They probably copied it from somewhere :) I'd like to see a case where they took up legal action against someone for violating the ToS |
| 22:21.06 | CosmicRay | harryvv: yes |
| 22:21.17 | CosmicRay | I got my spa-3000 in recently, going to try hooking it up today |
| 22:21.25 | bannerman | I'm not worried about the TOS, but what |
| 22:21.25 | _queuetue | wow1234, running 1.0.6-BRIstuffed-0.2.0-RC7k |
| 22:21.29 | harryvv | its been a good little unit. |
| 22:21.40 | CosmicRay | znoG: I agree, but why even bother signing up with a company that wants to do all that to people? |
| 22:21.41 | bannerman | 1.2 cents a minute is cheaper than 1.3 cents a minute :) is livevoip pretty reliable? |
| 22:21.54 | bjohnson | CosmicRay: the 3000 is a little tricky |
| 22:21.55 | CosmicRay | bannerman: I've been using them for awhile, had zero problems with them so far |
| 22:21.57 | znoG | CosmicRay: it's so ridiculous anyway their ToS i doubt it would hold any weight in court |
| 22:21.58 | wow1234 | _queuetur, what's your problem....maybe it's like my |
| 22:22.00 | bannerman | neat. |
| 22:22.04 | harryvv | I guess there has been alot of discussion on reliable voip services lately. |
| 22:22.07 | bjohnson | CosmicRay: the fxs is easy .. the fxo has a few tricks |
| 22:22.13 | wow1234 | i can get incoming but no out going |
| 22:22.20 | bjohnson | CosmicRay: info on the wiki though to get you started |
| 22:22.23 | CosmicRay | bjohnson: I've been reading the wiki page about making it go to asterisk without answering the fxo line first |
| 22:22.28 | bjohnson | yes |
| 22:22.31 | wow1234 | i keep getting this error message "407 Proxy Authentication Required" |
| 22:22.37 | harryvv | I have iax.cc anyone vouch for there reliability? |
| 22:22.38 | CosmicRay | bjohnson: what I don't understand is why all the callerid mucking, why not just set it to always forward instead of sometimes forward? |
| 22:22.47 | bjohnson | I've had a few problems with my sipuras but I think is mostly config |
| 22:22.59 | CosmicRay | one thing about the sipuras is that they are extremely configurable |
| 22:23.12 | bjohnson | although I have a 3k at home that wants to ring every hour until I restart the * server |
| 22:23.17 | fgravato | iax.cc = sixtel |
| 22:23.27 | bjohnson | happened twice in a week .. made me look bad to the wife |
| 22:23.29 | CosmicRay | my gripe about the spa-841 was that *67 did not do its conventional duty of disabling caller id for one call, but I found the place to fix it. |
| 22:23.32 | CosmicRay | heh |
| 22:23.40 | CosmicRay | was that related to voicemail? |
| 22:23.42 | _queuetue | Is it possible to run Asterisk communicating via sip to a provider and a sip softphone talking to asterisk on the same computer? |
| 22:23.55 | CosmicRay | bjohnson: over on mythtv people speak of WAF -- Wife Acceptance Factor |
| 22:24.00 | bjohnson | harryvv: i have iax.cc but haven't used them enough to vouch for anything |
| 22:24.07 | CosmicRay | bjohnson: my WAF is high since she discovered the music on hold feature :-) |
| 22:24.20 | bjohnson | CosmicRay: over there I ask wtf I don't get sound |
| 22:24.27 | CosmicRay | mjgeorge: heh |
| 22:24.34 | CosmicRay | err |
| 22:24.35 | CosmicRay | bjohnson: heh |
| 22:24.46 | CosmicRay | nick complete does not go well with typos |
| 22:25.20 | CosmicRay | the only thing I can't do automatically with my spa-841s yet is update the personal directory |
| 22:25.29 | CosmicRay | it looks liek it should be possible but I can't find the configuration field names anywhere |
| 22:25.36 | bjohnson | bannerman: i have livevoip but haven't used them enough to vouch for anything |
| 22:25.41 | Ro[b]ert | is it possible for another user from outsite the internal network to connect to my asterisk?? using xlite |
| 22:26.01 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 22:26.17 | Exstatica | bjohnson, is the key for rsa? I'm only trying to use a username and password |
| 22:26.19 | bjohnson | CosmicRay: I don't have the 841 .. just 2ks and 3ks |
| 22:26.21 | CosmicRay | #1 asterisk list gripe: so many threads full of people that have not read http://www.voip-info.org/wiki-VOIP+Service+Providers+Residential |
| 22:26.33 | CosmicRay | bjohnson: it's a great phone, well worth the money IMHO |
| 22:26.35 | bjohnson | I'm looking for donations of 841s and Linksys PAP2-NAs though |
| 22:26.41 | CosmicRay | bjohnson: and not sucky like the grandstreams |
| 22:26.44 | *** join/#asterisk tommy13v (~Administr@cpe-24-194-110-15.nycap.res.rr.com) |
| 22:26.48 | CosmicRay | err |
| 22:26.56 | stdio | I have 841s... what are you looking to do? |
| 22:26.59 | bjohnson | ahmms for the poor? |
| 22:27.09 | stdio | (sorry, can't donate them :) ) |
| 22:27.12 | CosmicRay | stdio: provision the persional directories from the server |
| 22:27.15 | bjohnson | stdio: have someone give one (or more) to me |
| 22:27.28 | bjohnson | stdio: will you be my sugar daddy? |
| 22:27.32 | CosmicRay | heh |
| 22:27.46 | stdio | sadly, they are not mine to give :) |
| 22:27.57 | CosmicRay | bjohnson: free shipping from the voxilla store |
| 22:28.11 | stdio | provisioning.. now there is something I have yet to do much with... how does one go about setting up a config? |
| 22:28.13 | bjohnson | Ro[b]ert: yes .. if you configure it to allow that |
| 22:28.16 | CosmicRay | nice service over there, unlike the crap service at telephonyware or the inflated shipping prices at voipsupply |
| 22:28.35 | CosmicRay | stdio: I'm going to write up an article about that in a few days, but... first, grab the admin manual and flip to the provisioning section |
| 22:28.45 | bjohnson | CosmicRay: not when I ordered .. the bastard |
| 22:28.47 | bjohnson | s |
| 22:28.55 | stdio | have it... flipping |
| 22:28.56 | CosmicRay | stdio: I don't use the compiled config format because I can't find the source for it, so I use xml files |
| 22:29.15 | CosmicRay | example of the xml format at https://musimi.dk/index.php/forum2/thread/forumid=3/id=2018/ |
| 22:29.22 | bjohnson | gotta go |
| 22:29.25 | CosmicRay | or at http://voxilla.com/forum-viewtopic-t-2828-highlight-xml.html |
| 22:29.25 | bjohnson | cya later |
| 22:29.33 | CosmicRay | cya bjohnson |
| 22:29.38 | bjohnson | good sipura info on the voxilla forums |
| 22:29.56 | CosmicRay | bjohnson: yes, some gold nuggets in there, but some highly-rated posts are crap, I've found |
| 22:30.01 | bannerman | wow, livevoip is cheap |
| 22:30.05 | CosmicRay | but yes, they are the best resource for sipura stuff |
| 22:30.08 | bannerman | wish I'd been sent to them when I first came here |
| 22:30.12 | CosmicRay | bannerman: yeah that $1/mo for 800 DID is tempting |
| 22:30.21 | Beirdo | livevoip just dumped all their DIDs, no? |
| 22:30.27 | CosmicRay | Beirdo: all their non-800 ones |
| 22:30.29 | Beirdo | and are redoing them all? |
| 22:30.35 | CosmicRay | they're still selling toll-free DIDs |
| 22:30.48 | terrapen | i need to figure out how to get PyMusique workin' |
| 22:30.49 | fgravato | sounds like they |
| 22:30.51 | fgravato | killed the did's |
| 22:30.52 | CosmicRay | incoming minutes on them are at the terribly high price of 1.27 cents per minute :-) |
| 22:30.53 | fgravato | or something |
| 22:30.53 | stdio | CosmicRay: recently, the call-info header was patched into asterisk so that the 841's can do auto-answer (effectively, intercom)... any idea when that is gonna get released? |
| 22:30.55 | Beirdo | any company who dumps their DIDs like that is low on my list |
| 22:31.19 | CosmicRay | stdio: well it's not in 1.0.7, so I suspect it's in CVS. Trying out that patch is on my todo list. |
| 22:31.26 | CosmicRay | stdio: because that it a feature I want. |
| 22:31.43 | stdio | stdio: me too |
| 22:31.48 | stdio | argh |
| 22:31.53 | stdio | CosmicRay: me too.... |
| 22:31.58 | CosmicRay | stdio: anyway, once you've got the xml file format, all you need to do is come up with the names for the xml tags |
| 22:32.10 | CosmicRay | most of them are the same as the field names on the web pages, with spaces replaced with underscores |
| 22:32.23 | CosmicRay | the per-line or per-extension ones look like Proxy_1_ or Proxy_2_ |
| 22:32.29 | *** part/#asterisk Flyboy6440 (~Bobo@192.76.82.89) |
| 22:32.31 | CosmicRay | so it's pretty easy |
| 22:33.12 | stdio | CosmicRay: it is in cvs: look for the sipaddheader function |
| 22:33.18 | CosmicRay | nice |
| 22:33.35 | CosmicRay | stdio: how long before cvs turns into a stable asterisk release? |
| 22:33.57 | BrianR___ | rebuilding festival is taking an eternity... lots of yucky c++ :( |
| 22:34.11 | stdio | CosmicRay: that's what I'm wondering |
| 22:34.17 | stdio | *really* want that... |
| 22:34.19 | CosmicRay | stdio: ah, I have no idea |
| 22:34.30 | CosmicRay | stdio: did you see there is a patch to asterisk 1.x? |
| 22:34.38 | CosmicRay | I think the wiki page has a link to it |
| 22:34.53 | CosmicRay | from what I gather from the asterisk bts, the implementation is different but it works |
| 22:35.22 | stdio | asterisk 1.1 |
| 22:35.33 | bannerman | I assume livevoip is sip? |
| 22:35.45 | CosmicRay | oh, that patch is against 1.1, not 1.0? crap. |
| 22:35.53 | CosmicRay | bannerman: I use iax with them. no idea if they do sip. |
| 22:35.57 | bannerman | oh, sweet |
| 22:35.59 | bannerman | I prefer IAX |
| 22:36.09 | bannerman | it sounds cooler |
| 22:36.10 | CosmicRay | if they do, it is non-obvious from their website |
| 22:36.17 | CosmicRay | but then, many things are non-obvious from their website. |
| 22:36.19 | CosmicRay | :-) |
| 22:37.13 | stdio | I believe so. |
| 22:37.16 | stdio | 1.1... |
| 22:37.35 | CoaxD | I can't stop raving..I can't stop raving.. |
| 22:37.45 | *** join/#asterisk mechn (~mechn@65.164.222.157) |
| 22:38.21 | stdio | ...dune? |
| 22:38.42 | mechn | has anyone ever worked with a Quintum Tennor AX |
| 22:39.14 | stdio | CosmicRay: bingo: http://bugs.digium.com/bug_view_page.php?bug_id=0002846 |
| 22:39.58 | *** join/#asterisk wizhippo (~wizhippo@Quebec-HSE-ppp233869.qc.sympatico.ca) |
| 22:40.01 | harryvv | ohh wouldnt that be a little cool the spa series atas ring the phone when the asterisk does not respond |
| 22:40.12 | CosmicRay | nice |
| 22:40.21 | *** part/#asterisk wizhippo (~wizhippo@Quebec-HSE-ppp233869.qc.sympatico.ca) |
| 22:40.24 | stdio | heh |
| 22:40.25 | CosmicRay | but that's not the patch I'm thinking of |
| 22:40.27 | CosmicRay | let me pull it up |
| 22:41.00 | CosmicRay | http://lists.digium.com/pipermail/asterisk-users/2005-January/086453.html |
| 22:41.32 | stdio | yeh, i have that bookmarked too |
| 22:41.38 | *** join/#asterisk flashnet (~flashnet@200.61.65.203) |
| 22:41.54 | stdio | but this will be nearly as easy to use |
| 22:42.06 | CosmicRay | yes, if you're running asterisk cvs :-) |
| 22:42.08 | CosmicRay | which I'm not |
| 22:42.32 | stdio | hehehhe |
| 22:42.37 | stdio | me neither :) |
| 22:42.46 | stdio | I'm tempted to apply this... |
| 22:42.49 | harryvv | if voipjet is not reliable for long distance from canada into the states then what carrier is? |
| 22:42.51 | CosmicRay | stdio: oh, one other thing, have you noticed that after you use the blind transfer feature, the spa-841 provides a weird dialtone the next time you try to use it, although everything else seems to be normal? |
| 22:43.00 | CosmicRay | harryvv: livevoip? :-) |
| 22:43.14 | stdio | what version firmware are you running? |
| 22:43.18 | *** join/#asterisk brycec (~brycec@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 22:43.24 | CosmicRay | stdio: the latest 3.whatever that is on their site |
| 22:43.38 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 22:43.39 | CosmicRay | I dunno, it seems odd to have a phone that comes with 0.9.5 and flash it to 3.something :-) |
| 22:44.35 | ronn | i'm looking to do load balancing for two asterisk boxes.. can any one suggest the best way? |
| 22:44.38 | stdio | stdio.. yeh, that is odd... |
| 22:44.44 | stdio | ARGH |
| 22:44.46 | CosmicRay | heh |
| 22:44.46 | stdio | did it again. |
| 22:44.49 | pigpen | ronn: can't you do clustering? |
| 22:44.55 | CosmicRay | stdio: have you noticed it too? |
| 22:44.58 | greg_work | CosmicRay: yeah they just changed their versioning scheme or something |
| 22:45.12 | stdio | CosmicRay: I never tried blind transfer :( |
| 22:45.14 | wow1234 | Anyone know how to fix this problem..."407 Proxy Authentication Required" Erro |
| 22:45.19 | ronn | pigpen: how do u cluster two asterisk boxes? |
| 22:45.22 | greg_work | CosmicRay: theres a setting under regional for the blindtransfer dialtone |
| 22:45.26 | CosmicRay | stdio: hit xfer, then dial *98 before you dial the destination box |
| 22:45.32 | greg_work | CosmicRay: personally, i turn that off and just tell them to use # |
| 22:45.42 | mrgoby | grrraaarr.... ok, so why does ast_async_goto minus 1 from the priority ? is that the same function goto uses ? |
| 22:45.48 | greg_work | (which makes asterisk do it, instead of the phone itself) |
| 22:46.02 | CosmicRay | greg_work: yeah but I'm not talking about that. I'm talking about after I'm all done with the call and the transfer... hours later, when I'm ready for the next call, the dialtone is weird. |
| 22:46.10 | pigpen | ronn: I haven't done it yet...but I am sure there are suggestions in the wiki |
| 22:46.12 | CosmicRay | greg_work: I haven't really figured out how to implement that |
| 22:46.21 | CosmicRay | greg_work: wouldn't it interfere with foreign voicemail systems and stuff? |
| 22:46.36 | mrgoby | is see it does it to the strings because they want to get rid of the terminating char, but the priority is passed as an int |
| 22:46.40 | greg_work | CosmicRay: its an option you pass to Dial() .. t i think |
| 22:46.47 | CosmicRay | greg_work: oh, t or T? |
| 22:46.51 | CosmicRay | that could be |
| 22:46.56 | greg_work | CosmicRay: when you use t, only the callee can use it |
| 22:47.21 | greg_work | generally when you dial out, you can't do a blind transfer.. only when someone calls you. otherwise you're right, it would interfere with foreign IVr's |
| 22:47.22 | ronn | pigpen: i have looked at a few different solutions .. i wanted to know if anyone is using one here |
| 22:47.25 | CosmicRay | I prefer to use the hard phone featureset.... I think it's easier to remember when you can see the prompts on the display |
| 22:47.25 | *** join/#asterisk leandro_pt (~leandro@bl6-117-235.dsl.telepac.pt) |
| 22:48.41 | bhsx | does anyone have a link (been googling all day) for setting up asterisk as your voicemail system with x100p? i can't seem to get the x100p to pick-up the line |
| 22:49.31 | *** part/#asterisk mrgoby (~mrgoby@pcp05308058pcs.wanarb01.mi.comcast.net) |
| 22:49.32 | CosmicRay | bhsx: well, solve your picking up the line problem before you bother with voicemail |
| 22:49.42 | CosmicRay | bhsx: seems like you want a basic manual on x100p configuration |
| 22:49.49 | bhsx | well, that's what i'm looking for |
| 22:49.58 | DrFrancky | there is one on wiki :-)) |
| 22:50.02 | bhsx | i did the modprobes and so forth |
| 22:50.07 | CosmicRay | well, I've found several zapata tutorials... wiki, asteriskdocs.org maybe, I can't remember |
| 22:50.11 | CosmicRay | did you run ztcfg? |
| 22:50.31 | bhsx | yup |
| 22:50.40 | CosmicRay | and everything is cool in syslog? |
| 22:50.49 | DrFrancky | bhsx: and what is saying zttool ? |
| 22:51.25 | bhsx | unconfigured ztdummy1/1 |
| 22:51.32 | *** join/#asterisk Flyboy6440 (~Bobo@192.76.82.89) |
| 22:51.54 | CosmicRay | did you modprobe wcfxo and set up your /etc/zaptel.conf befure running ztcfg? |
| 22:52.22 | Flyboy6440 | before i call and bug digium support, does anyone here know if they will be adding dns support to the iaxy? |
| 22:52.23 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 22:52.24 | bhsx | i didn't edit zaptel.conf by hand |
| 22:52.44 | zoa | bhsx, if you want and you have paypal we could have a look on your machine :p |
| 22:52.52 | bhsx | heh |
| 22:52.59 | zoa | but i suggest you read the zaptel and zapata config files |
| 22:53.05 | CosmicRay | well, you probably need to tell it at least fxsks=1, but seriously... rtfm |
| 22:53.16 | CosmicRay | I rftm'd and basically did this all with no trouble |
| 22:53.20 | CosmicRay | my first time |
| 22:53.22 | zoa | its not going to work buy just plugging it in |
| 22:53.41 | CosmicRay | though I have found the x100p cards to be low-quality and unstable, but then I suppose that's all I can expect for something I got for $7 off ebay :-) |
| 22:53.46 | bhsx | zaptel.conf is fine |
| 22:53.47 | zoa | look at loopstart, kewlstart etc |
| 22:53.51 | zoa | hmm |
| 22:53.51 | dan2 | CosmicRay: ;) |
| 22:53.55 | zoa | is it a real x100p ? |
| 22:53.59 | dan2 | CosmicRay: the spa3k work nicely as an fxo interface |
| 22:54.00 | CosmicRay | probably not |
| 22:54.02 | bhsx | and yes i did modprobe wfxco |
| 22:54.03 | zoa | because those were 100$ |
| 22:54.08 | CosmicRay | dan2: that is exactly why I ordered one |
| 22:54.10 | bhsx | no, it's generic |
| 22:54.12 | dan2 | :) |
| 22:54.14 | CosmicRay | dan2: setting it up tonight :-) |
| 22:54.26 | dan2 | CosmicRay: gotta show you this |
| 22:54.27 | pigpen | Anyone on doing clustering with * ? |
| 22:55.09 | zoa | pigpen: you mean like this: http://www.astertest.com/downloads/scx-testlab.jpg ? |
| 22:55.20 | CosmicRay | I figure, why pay $100+ for a PCI card when I could pay $99 and get something that does all the echo cancelation in hardware |
| 22:55.35 | pigpen | Dude...you need some cable ties... |
| 22:55.42 | DrFrancky | hehe |
| 22:55.42 | pigpen | Yeah...like that! |
| 22:56.05 | DrFrancky | it works just fine with current cabling |
| 22:56.06 | pigpen | ok..so what package are you using for clustering? |
| 22:56.16 | zoa | some custom built modules |
| 22:56.21 | stevekstevek | hey, zoa |
| 22:56.25 | zoa | hey steve! |
| 22:56.31 | zoa | tomorrow we continue with the JB |
| 22:56.35 | file[laptop] | zoa: I wanted to ask you, totally forgot, what exactly are those... |
| 22:56.35 | stevekstevek | cool. |
| 22:56.50 | zoa | file: those are 21 asterisk machines |
| 22:56.52 | stevekstevek | I have someone here also working on it. We'll have some updates soon. |
| 22:56.58 | zoa | aha great! |
| 22:56.59 | file[laptop] | zoa: I know that dingo |
| 22:57.05 | zoa | very cool! |
| 22:57.32 | file[laptop] | zoa: I meant hardware, like what are they... name of them... |
| 22:57.32 | zoa | slav will be online tomorrow |
| 22:57.32 | zoa | aaah |
| 22:57.32 | file[laptop] | cause they look like cute little machines |
| 22:57.32 | zoa | via nemehiah or so |
| 22:57.32 | zoa | they suck |
| 22:57.32 | file[laptop] | awwww |
| 22:57.32 | pigpen | zoa: cool...I am wanting to do things right...so mind sharing? |
| 22:57.32 | stevekstevek | there's some problems we'll fix including: it does the wrong thing if you give it packets with duration 20, 40, 20, 40 [...]. |
| 22:57.49 | zoa | super! |
| 22:57.55 | zoa | talk to slav about it tomorrow |
| 22:58.00 | zoa | or send him an email |
| 22:58.09 | stevekstevek | that happens if you have * in the middle of a call that comes in iLBC, and goes out with a 20ms codec. |
| 22:58.11 | zoa | the biggest problem is the high load shit |
| 22:58.26 | stevekstevek | today, there was a problem on the dev call with it. |
| 22:58.32 | zoa | aha |
| 22:58.36 | stevekstevek | After being great last time, and good for about an hour here.. |
| 22:58.46 | stevekstevek | it started reporting 25% incoming loss, and we heard lots of PLC. |
| 22:58.58 | zoa | hehe |
| 22:58.59 | zoa | funky |
| 22:59.02 | stevekstevek | but, mtr didn't report any loss, and turning it off sounded fine. |
| 22:59.14 | zoa | i think im going to bed |
| 22:59.18 | stevekstevek | Also, there's this guy who had some problem with Monitor() with it. |
| 22:59.19 | file[laptop] | bye zoa |
| 22:59.22 | stevekstevek | don't know what that was. |
| 22:59.25 | stevekstevek | bye, Zoa. |
| 22:59.27 | zoa | yeah i read those reports |
| 22:59.32 | zoa | we used monitor with it |
| 22:59.38 | zoa | (only with the sip version though) |
| 22:59.42 | zoa | to make those sample files |
| 22:59.46 | zoa | and we did not have that issue |
| 22:59.47 | pigpen | Any recommended links for * in clusting environment? |
| 22:59.59 | zoa | pigpen: not really |
| 23:00.11 | zoa | its not as easy as you might expect it to be |
| 23:00.17 | zoa | you have to do it for every protocol |
| 23:00.19 | *** join/#asterisk lancey (Shady@support.net1.cc) |
| 23:00.22 | lancey | hi guys! |
| 23:00.24 | zoa | hey lancey |
| 23:00.28 | lancey | oooooooooo |
| 23:00.30 | lancey | zoa |
| 23:00.31 | lancey | wow! |
| 23:00.31 | lancey | :) |
| 23:00.36 | pigpen | so I guess you are not using any of the big clustering projects... |
| 23:00.42 | lancey | i'm JUST looking for PSTN termination |
| 23:00.42 | lancey | :) |
| 23:00.54 | zoa | hehe |
| 23:00.58 | zoa | im not doing it atm |
| 23:01.10 | lancey | xex too bad |
| 23:01.21 | pigpen | k...I will do some reading...and come back more educated.... |
| 23:01.30 | zoa | look at linuxha |
| 23:01.33 | zoa | and ultramonkey |
| 23:01.36 | *** join/#asterisk montag___ (~montag@81-174-23-160.f5.ngi.it) |
| 23:01.36 | zoa | some do it like that |
| 23:01.40 | zoa | also look at astertest.com |
| 23:01.45 | zoa | there is a recording from astricon on there |
| 23:01.50 | zoa | that might give you some information |
| 23:01.50 | pigpen | yeah..that is what I was thinking... |
| 23:01.54 | pigpen | thanks... |
| 23:01.58 | montag___ | i've buyed a welltech 302 IP Phone, but this phone unregister after 5 minutes during call, any tips ? |
| 23:02.09 | mechn | where is the list of Asterisk codecs |
| 23:02.17 | zoa | http://www.astertest.com/forum/viewtopic.php?t=8 |
| 23:03.01 | *** join/#asterisk bjohnson (~bjohnson@66.11.165.161) |
| 23:03.22 | Ro[b]ert | bjohnson: are ther for that some special configs?? i just created an extension.. |
| 23:03.52 | Ro[b]ert | bjohnson: i mean for some internet user who needs to use my asterisk... |
| 23:04.05 | stevekstevek | I didn't know cisco supports speex.. |
| 23:04.18 | stevekstevek | (from some e-mail on -users, someone says they do..). |
| 23:04.36 | _queuetue | Is it kosher to run Asterisk communicating via sip to a provider and a sip softphone talking to asterisk on the same computer? |
| 23:06.24 | *** part/#asterisk Flyboy6440 (~Bobo@192.76.82.89) |
| 23:09.45 | *** part/#asterisk Ro[b]ert (~acidburnn@cust.7.204.adsl.cistron.nl) |
| 23:09.46 | *** join/#asterisk Ro[b]ert (~acidburnn@cust.7.204.adsl.cistron.nl) |
| 23:10.19 | *** join/#asterisk TrevorSHarrison (~trevorsha@24.49.36.218) |
| 23:12.35 | TomL | yes. not only is it kosher but its low sodium and tastes great |
| 23:12.58 | MacDeath | ummm |
| 23:12.59 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 23:13.07 | MacDeath | i think i have a very simple question |
| 23:13.13 | MacDeath | i have 2 * servers |
| 23:13.22 | TomL | the answer is 42 |
| 23:13.26 | MacDeath | both with 2 sip clints |
| 23:13.28 | MacDeath | :P |
| 23:13.42 | MacDeath | 1001, 1002 on the one and 2001 and 2002 on the other |
| 23:13.47 | *** join/#asterisk dca (~dca@c-67-166-37-218.client.comcast.net) |
| 23:14.06 | MacDeath | i just want to be able to dial 2001 and it must connect to the other * server |
| 23:14.22 | MacDeath | what goes in iax.conf/ |
| 23:14.26 | TomL | you need 2-way trunking between the * servers |
| 23:14.35 | *** join/#asterisk Skurpy (~merlyn@83.220.199.54) |
| 23:14.57 | MacDeath | TomL, and how would I do that? |
| 23:16.56 | *** join/#asterisk lilshtz (~lilshtz@static-70-19-113-140.ny325.east.verizon.net) |
| 23:16.57 | lancey | guys |
| 23:16.57 | TomL | I've never done it, so I'm not sure |
| 23:17.04 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 23:17.06 | jakepdev | MD - couldn't you just modify the dialplan to dial SIP on the other server |
| 23:17.07 | lancey | anyone here recommend a good PSTN termination provider? |
| 23:17.10 | lancey | preferably in europe? |
| 23:17.20 | lancey | i need sip or iax, and caller-id transfer? |
| 23:17.27 | jesster | Im troubleshooting a phone hanging off a FXS port of a channel bank. When the phone is in use, and another call comes in, asterisk cli shows "Zap/106-busy-678259056 is busy" but the caller does not hear a busy signal - it just rings. The callee does not hear a call waiting beep. |
| 23:18.47 | bhsx | ok... one step closer... it's (the x100p) now picking up and dialing out... but when it picks up it just hangs up right away... any ideas? |
| 23:19.37 | xeet2 | is it better to use scsi or sata, in regards to any performance hit or issues using asterisk and zaptel cards? |
| 23:20.03 | mmlj4 | i'm looking for a SIP "hard phone" for my home (http://www.voip-info.org/wiki-VOIP+Phones#id405505) --- suggestions? |
| 23:20.20 | mmlj4 | less than $00, if possible |
| 23:20.22 | ManxPower | SIPura SPA-841 |
| 23:20.26 | ManxPower | Which is about $100 |
| 23:20.27 | mmlj4 | er, $100 |
| 23:20.29 | lancey | xeet2 i don't think asterisk demands high hard-disk performance |
| 23:20.55 | jakepdev | MacDeath - http://www.voip-info.org/wiki-Asterisk+-+dual+servers |
| 23:21.01 | bhsx | i saw an chinese oem on ebay selling nice-featured ones for $60+shipping |
| 23:21.04 | JunK-Y | lancey: if u record and stream a lot, maybe. |
| 23:21.10 | lancey | mmlj4: why don't buy something like LinkSys PAP2-NA |
| 23:21.18 | lancey | and connect a regular phone to it? |
| 23:21.21 | lancey | it costs $63 |
| 23:21.26 | lancey | and has two lines :) |
| 23:21.39 | mmlj4 | lemme look at it |
| 23:21.51 | MacDeath | jakepdev : thx |
| 23:22.28 | jakepdev | np |
| 23:22.31 | xeet2 | lancey: I know it doesn't need hard drive performance, I was more asking about pci bus issues, like with dma and such |
| 23:22.34 | bannerman | mmlj4: So far I've been getting along pretty well with ariaVoice's atlas phone. www.ariavoice.com. |
| 23:22.40 | xeet2 | if scsi provides any advantages on a * box |
| 23:22.55 | lancey | xeet2 i don't think any modern system |
| 23:23.01 | lancey | could experience such issues |
| 23:23.22 | xeet2 | well, I've had it happen in the past, I had to turn alot of things on and off, but I've only ever built a * box with ide |
| 23:23.30 | mmlj4 | xeet2: i seem to remember reading that RAID 1 is best, IDE even... RAID 5 is slow for * |
| 23:23.38 | xeet2 | mmlj4: not even raid |
| 23:23.52 | lancey | mmlj4: RAID5 is ALWAYS faster than RAID1 |
| 23:23.59 | PatrickDK | raid5 is always slow |
| 23:24.01 | lancey | i think... |
| 23:24.04 | lancey | slow? |
| 23:24.05 | johnnyb | mmlj4: incorrect. |
| 23:24.06 | xkev | lancey what? |
| 23:24.07 | jakepdev | RAID 0 should be real fast |
| 23:24.08 | PatrickDK | it's the slowest thing arround |
| 23:24.09 | xkev | not on writes |
| 23:24.15 | lancey | well, i have a 3Ware RAID5 |
| 23:24.16 | mmlj4 | RAID 0 is stupid |
| 23:24.18 | PatrickDK | raid0 is fast |
| 23:24.20 | lancey | and it's faster than RAID1 |
| 23:24.21 | jakepdev | but don't turn off your machine |
| 23:24.23 | PatrickDK | raid10 is best |
| 23:24.33 | xeet2 | raid5 can be really slow if you're doing software raid, |
| 23:24.40 | lancey | xex :) |
| 23:24.44 | johnnyb | mmlj4: RAID1 has faster reads, because the read can be divided between the mirrored disks. |
| 23:24.46 | PatrickDK | raid5 can only be speed up by write caching |
| 23:24.49 | xeet2 | raid5 can be really fast with hardware raid with a good card, ie the 3ware or the broadmax stuff |
| 23:24.50 | PatrickDK | but that is dangerous |
| 23:25.01 | pigpen | lsi is pretty good. |
| 23:25.02 | PatrickDK | xeet2, no it can't |
| 23:25.11 | PatrickDK | in order to do raid5 |
| 23:25.13 | xeet2 | patrick: oh it can't? go read all the tests done on it then |
| 23:25.17 | PatrickDK | and write a single byte |
| 23:25.21 | jakepdev | (mmlj4 - what is stupid about RAID 0? |
| 23:25.22 | PatrickDK | you have to read from every drive |
| 23:25.25 | Bacon | All these generalizations about different raid levels are dangerous. |
| 23:25.26 | xeet2 | you can speed up your read speed a hell of alot with raid 5 |
| 23:25.28 | PatrickDK | calc parity |
| 23:25.33 | PatrickDK | and write changed drives |
| 23:25.33 | johnnyb | jakepdev: it's not redundant. |
| 23:25.42 | Bacon | Its just a matter of knowing the right tool for the job. |
| 23:25.44 | mmlj4 | jakepdev: so you hate your data, eh? |
| 23:25.47 | xeet2 | bacon: right, it is, done with this |
| 23:25.50 | johnnyb | xeet2: RAID 5 doesn't speed up your RAID nearly as much as RAID1 |
| 23:25.57 | PatrickDK | fuck read speed, every server I have does 4 times the amout of writes than it does reads |
| 23:25.57 | jakepdev | if you backup - you can use RADID 0 |
| 23:25.58 | xeet2 | johnnyb: lol |
| 23:26.00 | johnnyb | s/RAID/read |
| 23:26.14 | PatrickDK | raid1 doubles read rate |
| 23:26.16 | mikegrb | PatrickDK: in can still be fast with an expensive controller that has large ammounts of battery backed write cache |
| 23:26.21 | xeet2 | anyway |
| 23:26.23 | johnnyb | xeet2: RAID1 can split up your read among multiple disks, RAID5 cannot |
| 23:26.32 | mikegrb | we use raid1 at work, still not fast enough reads |
| 23:26.33 | lancey | johnnyb: incorrect |
| 23:26.41 | johnnyb | lancey: in what way. |
| 23:26.42 | mikegrb | going to fiber channel stuff most likely |
| 23:26.47 | lancey | RAID5 speeds up you reads |
| 23:26.50 | PatrickDK | mikegrb, not as fast, but can get closer, but per drive added to a single raid5 the slower it gets |
| 23:26.51 | Bacon | So many people saying so many incorrect things.... |
| 23:26.54 | jakepdev | RAID 1 is known for redundancy - not speed |
| 23:27.10 | mikegrb | PatrickDK: yeah |
| 23:27.10 | johnnyb | lancey: how does RAID5 speed up reads more than RAID1? |
| 23:27.34 | lancey | johnnyb at least it's not slower |
| 23:27.34 | johnnyb | jakepdev: it's known for both, but it takes almost twice as many disks as RAID5, which is why it's not often used. |
| 23:27.38 | Bacon | RAID5 should be faster for reads than RAID1. |
| 23:27.41 | lancey | all disks read |
| 23:27.42 | johnnyb | lancey: yes it is. |
| 23:27.49 | lancey | johnnyb no way :) |
| 23:27.58 | jakepdev | I'd say go with 10 (1+0) if you can |
| 23:27.58 | PatrickDK | bacon, only for sequenual large reads |
| 23:28.12 | lancey | johnnyb if you speak of software raid, then maybe |
| 23:28.14 | xeet2 | johnnyb: raid 5 does NOT require a read to be run across all drives, it can determine the value of some of the bits without actually having to read it, raid 5 is, with hardware, not as fast as raid 0, but it has fault protection, which 0 does not |
| 23:28.17 | PatrickDK | for random reads or small reads, raid1 is better |
| 23:28.30 | johnnyb | lancey: if you have RAID5, then you only have one copy of your data. You have a checksum, but it's only usable if you compare it to ALL disks on the array. |
| 23:28.38 | Bacon | PatrickDK: Large sequentail reads are the common case. Look at how RH tunes the IO elevators in rhel. |
| 23:28.48 | lancey | johnnyb you have 1 copy |
| 23:28.53 | lancey | but it's being read faster |
| 23:28.56 | lancey | :) |
| 23:28.59 | bannerman | argh. when I try to log an agent in, it tries to use a context that doesn't even exist anymore-- not in queues.conf, sip.conf, iax.conf, extensions.conf... |
| 23:29.01 | PatrickDK | bacon, large sequential reads are common? hell no, unless your streaming video |
| 23:29.08 | lancey | all hard disks read |
| 23:29.08 | johnnyb | lancey: in RAID1, you have multiple _usable_ copies of your data. What would be only a single-disk read in RAID5 can be a multi-disk read in RAID1 |
| 23:29.09 | lancey | :) |
| 23:29.09 | xeet2 | johnnyb: yeah, you have one copy thats distributed across drives, and you can loose several drives and it can still determine what the values are |
| 23:29.14 | bannerman | is there something you have to do, in order to reset your agents or something? |
| 23:29.21 | Bacon | PatrickDK: Or using MySQL. |
| 23:29.24 | xeet2 | johnnyb: go read about raid dude, you're quite wrong |
| 23:29.26 | lancey | johnnyb data is split on the disks |
| 23:29.27 | johnnyb | xeet2: yes, but it's not useful for speed. |
| 23:29.32 | lancey | there's no single read on RAID5 |
| 23:29.34 | PatrickDK | even with elevator io, it is not sequential, it's just in order |
| 23:29.39 | Bacon | Most DB apps are poorly tuned and do lots of full table scans. |
| 23:29.40 | lancey | all hard disks always write/read |
| 23:29.42 | PatrickDK | sequiential is 1 then 2 then 3 |
| 23:29.43 | xeet2 | johnnyb: in a fault situation it will be slow, but when everything is up its quite fast |
| 23:29.44 | PatrickDK | not 1 then 3 |
| 23:30.12 | lancey | xeet2 i had a hard drive crash on my 3ware, it's pretty fast when it's faulty, too |
| 23:30.15 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 23:30.30 | xeet2 | anyway, scsi and ide, I know a pc will communicate with the controllers differently, does anyone know if scsi has any advantages when used on a box with * and zaptel cards? |
| 23:30.34 | johnnyb | xeet2: You're missing entirely what I'm saying. |
| 23:30.38 | jakepdev | ~RAID |
| 23:30.39 | jbot | i heard raid is (Redundant Array of Inexpensive Drives) A manipulation of SCSI technology that allows you to combine several hard drives for use as one. The benefits can be either speed increase or failsafe (so that if one drive dies or crashes you don't lose your data), or a combination of both. |
| 23:30.47 | Bacon | It all depends on what you are doing. Server applications do logs of sequential reads. Workstations are the oppsite. |
| 23:30.54 | PatrickDK | xeet2, only if you do lots of voicemail |
| 23:30.58 | johnnyb | xeet2: in RAID1 you ALWAYS have 2 ACTIVE, FAST copies of the data. |
| 23:31.16 | johnnyb | xeet2: in RAID5, you have 1 ACTIVE, FAST copy of the data, and ONE as a checksum |
| 23:31.22 | jakepdev | johnnyb - but your writes slow |
| 23:31.28 | PatrickDK | bacon, none of my servers do sequential reads |
| 23:31.34 | PatrickDK | I stat that info |
| 23:31.46 | lancey | johnnyb you ARE wrong |
| 23:31.52 | johnnyb | xeet2: therefore, for reads, RAID1 can be much faster, because the card can schedule which sectors come from which disks much better. |
| 23:31.54 | PatrickDK | the closest I get to sequential read are compiling new kernels |
| 23:31.59 | Bacon | PatrickDK: Asterisk servers? Or all servers in general? |
| 23:32.13 | lancey | johnnyb data is being chunked |
| 23:32.13 | PatrickDK | bacon, all servers |
| 23:32.13 | lancey | both with RAID1 and RAID5 |
| 23:32.17 | PatrickDK | asterisk shouldn't be doing i/o |
| 23:32.19 | lancey | with equal number of drives |
| 23:32.20 | Bacon | I haven't been working with asterkisk long enough to know what its typical use case looks like. |
| 23:32.22 | PatrickDK | except for voicemail and logs |
| 23:32.23 | xeet2 | johnnyb: then... why do raid 1 arrays 99% of the time run slower than not running raid at all? |
| 23:32.27 | lancey | both perform the same |
| 23:32.29 | lancey | on reads |
| 23:32.38 | xeet2 | on reads, and of course writes |
| 23:32.42 | johnnyb | xeet2: on what data do you base this? |
| 23:32.54 | lancey | writes too |
| 23:32.57 | xeet2 | johnnyb: many many performance test results out on the net |
| 23:33.00 | PatrickDK | xeet2, hardware or software raid1? |
| 23:33.00 | johnnyb | xeet2: and are you talking about reads or writes? |
| 23:33.05 | mmlj4 | lancey: do you use that linksys box? |
| 23:33.05 | lancey | johnnyb follow this: |
| 23:33.10 | lancey | mmlj4 yes |
| 23:33.19 | lancey | it's not worse than a Cisco ATA |
| 23:33.19 | lancey | :) |
| 23:33.26 | mmlj4 | * has no issues registering the phones? |
| 23:33.30 | lancey | nopes |
| 23:33.37 | lancey | works like a charm |
| 23:33.37 | mmlj4 | is there a page on it? |
| 23:33.47 | lancey | www.linksys.com? |
| 23:33.52 | PatrickDK | http://www.acnc.com/raid.htmlvvv |
| 23:33.55 | PatrickDK | http://www.acnc.com/raid.html |
| 23:33.56 | mmlj4 | i mean an * page? |
| 23:34.02 | lancey | johnnyb imagine we have a 4 hard drive array |
| 23:34.06 | lancey | if using RAID1 |
| 23:34.10 | jakepdev | http://www.adaptec.com/worldwide/product/markeditorial.html?sess=no&prodkey=quick_explanation_of_raid |
| 23:34.21 | lancey | let's say each HDD can write at 10 MB/s |
| 23:34.23 | lancey | if using RAID1 |
| 23:34.28 | lancey | we write with 20 MB/s |
| 23:34.37 | lancey | (we have 2 mirrors) |
| 23:34.40 | lancey | if using RAID5 |
| 23:34.45 | xeet2 | so you're doing raid 10 then |
| 23:34.46 | lancey | we write with 30 MB/s |
| 23:34.52 | lancey | actual data is on 3 disks |
| 23:34.54 | lancey | not on 2 |
| 23:34.58 | lancey | get the point now? |
| 23:35.00 | PatrickDK | lancey, no |
| 23:35.06 | Bacon | Ugh. |
| 23:35.13 | lancey | PatrickDK where am i wrong? |
| 23:35.14 | PatrickDK | raid5 3disks, is only 20mb/s - calc time |
| 23:35.19 | johnnyb | lancey: If you actually read what I was writing, I was talking about READS not WRITES. |
| 23:35.25 | lancey | PatrickDK i speak of hardware raid |
| 23:35.32 | PatrickDK | lancey, so am I |
| 23:35.35 | lancey | johnnyb on reads |
| 23:35.38 | lancey | in both situations |
| 23:35.39 | PatrickDK | I only use hardware raid |
| 23:35.44 | lancey | they both perform the same |
| 23:35.59 | lancey | PatrickDK i'm not willing to agree with you :) |
| 23:36.05 | *** join/#asterisk JerJer[mobile] (~jj@feth100-fw.fament.net) |
| 23:36.06 | johnnyb | lancey: they do not, because w/ RAID1 you have 2 USABLE alternatives to pull the data from, and it can be scheduled better. |
| 23:36.07 | lancey | what i AMM sure is |
| 23:36.09 | PatrickDK | lancey, only perform the same for sequential reads |
| 23:36.16 | PatrickDK | with random raid1 is faster |
| 23:36.17 | lancey | RAID5 is FASTER than RAID1 in all cases |
| 23:36.25 | lancey | no way |
| 23:36.33 | JerJer[mobile] | you all are wrong MFM is uber fast |
| 23:36.37 | PatrickDK | why? raid5 you can only get part 1 or 3 at once |
| 23:36.43 | PatrickDK | raid1 you can get both at the same time |
| 23:36.44 | xeet2 | jerjer: lol |
| 23:36.53 | lancey | no matter what RAID u are using |
| 23:36.56 | PatrickDK | since on raid5 they are on the same drive |
| 23:36.57 | lancey | data is chunked |
| 23:37.03 | PatrickDK | not on raid1 :) |
| 23:37.05 | lancey | and everything behaves like 1 big HDD |
| 23:37.06 | xkev | is there an app for testing if an member interface exists in a queue? |
| 23:37.17 | jakepdev | RAID5 should be faster than RAID 1 since you're reading different chunks at the same time |
| 23:37.27 | johnnyb | lancey: yes, but in RAID1, you have 2 ALTERNATIVES and choose whichever is less busy |
| 23:37.30 | lancey | so everything gets queued as it will on a normal hard drive |
| 23:37.32 | *** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
| 23:37.35 | xkev | jakepdev, faster or reads, but writes are awful |
| 23:37.38 | lancey | johnnyb define less busy |
| 23:37.42 | lancey | if i have raid |
| 23:37.47 | lancey | i assume all hard drives work |
| 23:37.48 | jakepdev | xkev agreed |
| 23:37.48 | xkev | read, calculate, write <- a write |
| 23:37.49 | lancey | not sleep |
| 23:37.49 | lancey | :) |
| 23:38.07 | debaser | you know, arstechnica has a great raid faq that all of you should probably read. |
| 23:38.12 | lancey | that's what it's made for :)) |
| 23:38.19 | johnnyb | lancey: it seems you are assuming only one task is accessing the disk at a time. |
| 23:38.37 | xkev | jakepdev raid1 reads can only be theoretically 200%, where a 3 disk raid 5 would be 200% as well, but striped (loss of one spindle for parity) |
| 23:38.38 | lancey | [01:36] <lancey> RAID5 is FASTER than RAID1 in all cases |
| 23:38.40 | lancey | definitely |
| 23:38.54 | lancey | unless using less than 4 drives |
| 23:39.03 | xkev | 4 drive raid 1 is stupid |
| 23:39.04 | xkev | :) |
| 23:39.11 | xeet2 | can anyone explain exactly what ide dma is? |
| 23:39.18 | lancey | xkev: yes it is |
| 23:39.18 | lancey | :) |
| 23:39.21 | xeet2 | and why scsi doesn't use dma? or does it? |
| 23:39.38 | AgiNamu | xeet2, DMA is direct memory access |
| 23:39.39 | xkev | but a 4 drive 0+1 is not :) |
| 23:39.39 | PatrickDK | xeet, ide uses dma cause ide sucks |
| 23:39.48 | PatrickDK | scsi never needed it cause it already had it |
| 23:39.51 | AgiNamu | 4 drive raid 1 isnt stupid. |
| 23:39.51 | xkev | just wasteful, but if you do more than 30% writing, raid5 blows |
| 23:39.54 | xeet2 | ahhh |
| 23:39.55 | AgiNamu | It's just extra redundant. |
| 23:40.07 | xeet2 | so, in a * box with lots of traffic to zaptel cards, scsi would be better |
| 23:40.09 | xeet2 | right? |
| 23:40.12 | xkev | aginamu, 3 drives is extra redundant. 4 is stupid. :) |
| 23:40.29 | PatrickDK | xeet, no difference, except if you keep log files and voicemail on it |
| 23:40.38 | PatrickDK | cause all asterisks work in done in memory, not harddrive |
| 23:40.45 | xkev | xeet2 what you really need is a good apic board to handle all the interrupt traffic |
| 23:40.47 | PatrickDK | and sound files will get cached |
| 23:40.47 | *** part/#asterisk whmok (~acidBurn@219.94.82.55) |
| 23:40.55 | *** join/#asterisk durex (~ironman@weber.anpa.org.br) |
| 23:41.01 | xkev | if I slaughter my system with full debug logging on, I get bad audio |
| 23:41.08 | xkev | due to disk io |
| 23:41.10 | *** part/#asterisk mechn (~mechn@65.164.222.157) |
| 23:41.13 | PatrickDK | need scsi for that |
| 23:41.20 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 23:41.26 | xeet2 | xkev: ok thats what I was looking for... does anyone have any opinion on dell poweredge servers? do they normally have good apic's that can handle things well? |
| 23:41.30 | AgiNamu | there's no need for SCSI today |
| 23:41.32 | AgiNamu | no need at all. |
| 23:41.49 | PatrickDK | hmm, I have toasted 3 computers with ide |
| 23:41.53 | jakepdev | which distro is best |
| 23:41.56 | PatrickDK | my scsis still work, and are ALOT faster |
| 23:42.03 | jakepdev | jk |
| 23:42.05 | PatrickDK | ide seek time 8.9ms, scsi seek time 2.9 |
| 23:42.18 | AgiNamu | seek time is unreleated to SCSI or IDE |
| 23:42.19 | xeet2 | patrickdk: which scsi? |
| 23:42.19 | AgiNamu | duh. |
| 23:42.23 | PatrickDK | agi, I know |
| 23:42.24 | ariel_ | jakepdev, that is a loaded question here. I like Centos |
| 23:42.27 | xkev | patrickdk, and if you don't get WD raptors, you don't get full QA on each IDE drive |
| 23:42.32 | PatrickDK | but you find a ide faster than 8.9 |
| 23:42.58 | AgiNamu | at any rate, you're paying the damn price |
| 23:43.00 | terrapen | if you are doing serious heavy lifting, don't even fuck with IDE |
| 23:43.04 | xkev | IDE is just a bus, whereas SCSI is an architecture |
| 23:43.04 | AgiNamu | so you better well get a 1ms seek time!! |
| 23:43.10 | AgiNamu | I use SATA |
| 23:43.18 | AgiNamu | going to buy an 8x SATA 3ware RAID card. |
| 23:43.18 | xkev | command queueing, blah blah etc (which some sata has now) |
| 23:43.22 | AgiNamu | run a RAID 10 array |
| 23:43.37 | PatrickDK | sata of 1ms? na |
| 23:43.46 | xeet2 | sata2 can do it |
| 23:43.49 | AgiNamu | im saying 1ms is what you should get on SCSI for the price. |
| 23:43.52 | xeet2 | with raid 5!!! |
| 23:43.55 | AgiNamu | and i can throw on some nice Seagates 300 or 400GB |
| 23:44.14 | ariel_ | seagates argh |
| 23:44.24 | durex | hello *s, i'm having a few doubts about analog lines |
| 23:44.29 | _queuetue | Can anyone take a look at this sip history and tell me if it looks right? If I call the broadvoice number, this happens: http://deadbeefbabe.org/paste/83 but the call does not get transferred to asterisk voicemail - it gets bounced back to bv voicemail. Can anyone tell by looking at this log if it's a sip config proble, a vm config problem, or a firewall problem? |
| 23:44.33 | durex | does anybody can gimme some help ? |
| 23:44.53 | _queuetue | durex, No need to doiubt them - they exist! |
| 23:44.59 | AgiNamu | im sooo thrilled. i made my first firmware today |
| 23:45.13 | AgiNamu | I actually added some features to the IAX2 implementation of the PA168 |
| 23:45.19 | ariel_ | durex, ask a question |
| 23:45.21 | stevekstevek | AgiNamu: is the source to PA168 firmware free? |
| 23:45.32 | stevekstevek | or do you work for the PA168-people.. |
| 23:45.33 | AgiNamu | it's available if you own a device |
| 23:45.42 | stevekstevek | the source is? |
| 23:45.43 | AgiNamu | it's not "free as in crackhead thinking" |
| 23:45.47 | AgiNamu | yes, the source is available. |
| 23:45.51 | PatrickDK | here is something too |
| 23:45.54 | AgiNamu | you have a PA168? |
| 23:46.01 | stevekstevek | Not presently. |
| 23:46.01 | PatrickDK | I keep movies on my 8 drive raid5 3ware |
| 23:46.04 | AgiNamu | oh. |
| 23:46.07 | stevekstevek | I'm just interested in what they did. |
| 23:46.08 | AgiNamu | get one, then shoot me an email |
| 23:46.12 | PatrickDK | when I transfer it to my raid1 scsi 15k drives |
| 23:46.19 | PatrickDK | the 3ware can't keep up |
| 23:46.21 | stevekstevek | how do you know if I have one. |
| 23:46.29 | stevekstevek | wait. There it is. I have one now! |
| 23:46.32 | AgiNamu | you're honest. |
| 23:46.33 | AgiNamu | :) |
| 23:46.34 | _queuetue | "free as in crackhead thinking"? WFT does that mean? |
| 23:46.36 | stevekstevek | heh. |
| 23:46.47 | AgiNamu | you havce to tell me the hardware tag. thats how. |
| 23:46.56 | AgiNamu | which doesnt mean much |
| 23:47.01 | AgiNamu | cause you could just say "PA168Q" |
| 23:47.07 | ariel_ | question has anyone used sip to talk between two asterisk. Yes I know iax2 is better but I need to get sip working between two asterisk servers. |
| 23:47.11 | AgiNamu | _queutue, ever meet stallman? :) |
| 23:47.12 | stevekstevek | How about $59.95 |
| 23:47.32 | durex | well... I have 10 external lines (POTS), and 30 internal lines, analogic phones, and I wanna change my old PBX to Asterisk, and then change my old PBX of other office and put Asterisk to work. In both offices I have 10 external linas (POTS) and 30 internal lines, analog phones. What kind of hardware should I use? |
| 23:47.43 | stevekstevek | are you from iaxtalk.com? |
| 23:47.49 | _queuetue | AgiNamu, Yes, I have, actually. I've got quite a bit of respect for him - was that your point? |
| 23:48.04 | AgiNamu | _queuetue, yea. it's a "cheap shot" |
| 23:48.22 | AgiNamu | stevek me? no. |
| 23:48.33 | AgiNamu | i just needed some IAX2 devices for a Voip rollout |
| 23:48.35 | xeet2 | durex: where are you located? |
| 23:48.36 | stevekstevek | I should bug them to at least post the source to their "iaxLite softphone", which is based on a lot of my code, licensed under the LGPL, which they distribute in violation of the license :) |
| 23:48.38 | AgiNamu | and the PA168 is the only one I know of. |
| 23:48.39 | ariel_ | durex, easy. Use some adtran 750/850 channel banks and TE110p card 2 of them on each server. |
| 23:48.49 | AgiNamu | stevek, who is this? |
| 23:48.50 | AgiNamu | iaxtalk? |
| 23:48.52 | stevekstevek | maybe they'll apologize and send me a whone :) |
| 23:48.55 | stevekstevek | iaxtalk.com |
| 23:49.02 | stevekstevek | s/whone/phone :) |
| 23:49.04 | AgiNamu | whore? |
| 23:49.06 | AgiNamu | oh. phone. |
| 23:49.10 | durex | xeet2 I'm in Brazil |
| 23:49.24 | durex | ariel_ where I found it? |
| 23:49.32 | AgiNamu | at any rate... im so happy. |
| 23:49.38 | AgiNamu | now I can use qualify=yes |
| 23:49.40 | AgiNamu | wai wai! |
| 23:49.50 | xeet2 | durex: if its cheap enough you might want to consider using a pri or a t1 from your telco |
| 23:49.53 | stevekstevek | is their iax2 implementation based on libiax2? |
| 23:49.54 | ariel_ | durex, adtrans are about 400 to 500 dollars on ebay. |
| 23:50.02 | stevekstevek | because we fixed that in libiax2 a few months ago.. |
| 23:50.23 | AgiNamu | stevek, never say libiax2. is it based on "sessions"? |
| 23:50.28 | durex | ok, and I have to connect POTS or internal lines to adtrans? |
| 23:50.30 | stevekstevek | yeah. |
| 23:50.30 | xeet2 | out here in this part of the us, a pri is the same price as 8 "business" lines from most clecs |
| 23:50.35 | AgiNamu | I never say libiax2, i should say. |
| 23:50.39 | durex | xeet2 maybe I can use a T1 from my telco |
| 23:51.31 | ariel_ | durex, yes you can, see if they can switch you out. But for your 30 internal phones you will still need a channel Bank |
| 23:52.36 | lancey | anyone recommend good PSTN termination provider? |
| 23:52.49 | durex | ok... and If I just stay with my PBX, and connect * to my PBX |
| 23:52.57 | durex | how should I do it? |
| 23:53.27 | xeet2 | lancey: nufone, broadvoice, txlink are the ones I use |
| 23:54.33 | xeet2 | lancey broadvoice isn't very flexible and use sip only but offer unlimited calling, txlink and nufone do iax |
| 23:54.58 | xeet2 | there are plenty more out there too |
| 23:55.02 | terrapen | iax.cc uses txlink for some of their DIDs i think |
| 23:56.10 | durex | so... if I don't change my PBX, I would have a diagram like it: PBX conect to POTS and internal analog lines. Asterisk connects to PBX as a internal analog line or POTS? And of course, PBX connects to Internet |
| 23:56.21 | ariel_ | durex, do you have available ports as fxs or fxo on the pbx or do you have a t1/pri or e1 connection from it. |
| 23:57.03 | durex | ariel_ no, fxo and fxs ports on PBX |
| 23:57.16 | lancey | xeet2 |
| 23:57.20 | lancey | u happy with nufone? |
| 23:58.57 | dca | lancey: i'm pretty happy with Teliax |
| 23:59.22 | lancey | i need something in europe, preferably |
| 23:59.27 | lancey | as i'm in bulgaria |
| 23:59.31 | lancey | ping to USA = 200 ms :) |
