| 00:00.29 | *** join/#asterisk Qwell (~north@70-32-102-18.ontrca.adelphia.net) |
| 00:00.38 | Grant2 | Tutorials? |
| 00:00.46 | *** join/#asterisk jerryh (~jhicks@67.141.135.121) |
| 00:01.04 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 00:01.21 | DevilFish | ManxPower - the other day I mentioned that I was having big troubles with disconnections |
| 00:01.26 | *** join/#asterisk Qwell (~north@70-32-102-18.ontrca.adelphia.net) |
| 00:01.33 | DevilFish | you told me to upgrade to the new version of asterisk |
| 00:01.42 | DevilFish | which I did, do I thankyou for that :) |
| 00:01.47 | ManxPower | DevilFish, And I told you to set callprogress=no and busydetect=no? |
| 00:01.54 | DevilFish | still having troubles though |
| 00:01.56 | ManxPower | DevilFish, Upgrading fixed it? |
| 00:01.57 | JohnnyC | thanks ManxPower |
| 00:01.58 | DevilFish | oh really |
| 00:02.02 | DevilFish | then I missed that |
| 00:02.06 | ta[i]nted | ManxPower you mean putting the NoOp after the AGI call? I thought AGI is asynchronous? would i still get correct results? |
| 00:02.15 | ManxPower | ta[i]nted, before the AGI call |
| 00:02.25 | DevilFish | nah I'm getting lines like thatione in debug |
| 00:02.38 | DevilFish | Mar 2 15:44:00 DEBUG[7647]: Bridge stops because we're zombie or need a soft hangup: c0=SIP/251-37e5, c1=SIP/sip_metaswitch_out-d35b, flags: No,No,No,Yes |
| 00:02.51 | ManxPower | DevilFish, Upgrading did not help? |
| 00:02.55 | ManxPower | Weird. |
| 00:03.01 | DevilFish | nope I still got it |
| 00:03.11 | ManxPower | DevilFish, Try setting "canreinvite=no" in sip.conf. |
| 00:03.11 | DevilFish | we're using PolyCom IP500s |
| 00:03.17 | DevilFish | not sure that matters though |
| 00:03.26 | *** join/#asterisk Ridgeback (~Ridgeback@ppp245-134.lns2.adl2.internode.on.net) |
| 00:03.38 | Ridgeback | hello |
| 00:03.43 | ManxPower | DevilFish, If that doesn't fix it, post to bugs.digium.com |
| 00:03.47 | DevilFish | also using a MetaSwitch .. I got a guy looking at the debug logs for it as well |
| 00:03.54 | Ridgeback | anyone here have any experience with IaxTalk hard phones? |
| 00:04.02 | DevilFish | but I thought I'd look here again as well |
| 00:04.22 | jerryh | hrm. i suppose this is mostly a support forum? |
| 00:04.37 | ManxPower | Ridgeback, They ship from Taiwan. There is another distributer in the USa that has those phones. |
| 00:04.53 | ManxPower | Ridgeback, They are a bitch to set up, but not hard, once you understand it. |
| 00:04.54 | DevilFish | ManxPower, have you ever seen the zombie softhangup thing in a debug log? |
| 00:04.55 | ariel_ | jerryh, yes it is mainly for that. |
| 00:05.05 | *** part/#asterisk clord (~Clint@166.70.78.242) |
| 00:05.06 | ManxPower | Also do NOT try to upgrade them unless you are on the same local LAN. *SOB* |
| 00:05.18 | ManxPower | DevilFish, Only when I was running an old Asterisk |
| 00:05.39 | Ridgeback | ManxPower, hmmmm ok i just wondered if they were any good. I was looking for a good iax hard phone... any others out there? |
| 00:05.58 | ManxPower | Ridgeback, They are all based on the same chips. |
| 00:06.00 | DevilFish | hmmm, the CLI tells me I got new version ... is it possible I hosed up the upgrade ? |
| 00:06.00 | jerryh | ariel_: where to find 31337 hax0rs? :) |
| 00:06.07 | *** join/#asterisk yaboo (~jsirucka@220.245.131.131) |
| 00:06.11 | jerryh | hehe |
| 00:06.13 | tzanger | jerryh: 31337 ones? Right here |
| 00:06.17 | jerryh | kewl |
| 00:06.21 | DevilFish | I just did a make all, and make install of the asterisk tar.gz |
| 00:06.27 | tzanger | the elite ones are elsewhere but until you show you really need them I won't say where :-) |
| 00:06.43 | jerryh | i was told there would be free beer? |
| 00:06.45 | epoch | man, thirty-one thousand three-hundred and thirty-seven hax0rs... that's a hell of a lot of hax0rs... |
| 00:06.57 | jerryh | epoch: it's a MOVEMENT :) |
| 00:07.01 | tzanger | jerryh: heh |
| 00:07.05 | epoch | a... bowel movement? |
| 00:07.07 | Ridgeback | ManxPower, regarding setting up. they offer web server on the phone. is that how you set them up? |
| 00:07.07 | *** join/#asterisk Katty (~angela@68.112.15.110) |
| 00:07.08 | jerryh | nonono |
| 00:07.15 | Katty | hihi |
| 00:07.18 | epoch | ;) |
| 00:07.33 | ta[i]nted | ManxPower more weirdness.. bad phone number $DIALSTATUS returns 'CANCELLED' while $NoOp-Dialstatus returns '', answered phone call $DIALSTATUS returns 'noresponse', $NoOp-Dialstatus returns 'noresponse' |
| 00:07.54 | ariel_ | Katty, hello |
| 00:07.55 | Katty | that snom190 sure was /dreamy/ |
| 00:08.14 | Katty | hihi ariel_ |
| 00:08.21 | epoch | dreamy? |
| 00:08.23 | epoch | how so? |
| 00:08.30 | Dr-Linux | any body knows about NAT etc for asterisk ? |
| 00:08.31 | Katty | the polycom 500s were nice too |
| 00:08.31 | twisted[work] | hmm... |
| 00:08.37 | Katty | but.. |
| 00:08.37 | twisted[work] | snom190 is okay. |
| 00:08.59 | epoch | I wanna get a soundstation ip4000 |
| 00:09.01 | twisted[work] | i lubbor my 7960's |
| 00:09.16 | epoch | twisted[work]: I actually prefer my ip500 to my 7960 |
| 00:09.21 | twisted[work] | epoch, that's nice ;0 |
| 00:09.22 | epoch | except for the minibrowser thing |
| 00:09.22 | twisted[work] | er ;) |
| 00:09.28 | ariel_ | for the ip500 at 200 dollars it's a great phone. |
| 00:09.37 | *** part/#asterisk opinsky (~opinsky@200.204.120.198) |
| 00:09.38 | Katty | those poly 500s were sorta confusing |
| 00:09.38 | epoch | oh yeah. |
| 00:09.49 | Katty | so much stuff in there i didn't recognize |
| 00:09.50 | epoch | confusing?? |
| 00:09.52 | jerryh | You guys really got John "Maddog" Hall fired up. |
| 00:09.56 | Katty | including the 'sample bits. |
| 00:10.03 | Katty | there was a .wav listed in one. |
| 00:10.04 | jerryh | He says Asterisk will be bigger than Linux. |
| 00:10.07 | Moc | epoch, me too (but I prefer my ip 600 to my ip 500 ;) |
| 00:10.09 | JohnnyC | Hey is there any Mobile phone interface so we can use Mobile Phone cards instead of PTSN ? |
| 00:10.09 | Katty | but i didn't see a way to change it. ah well |
| 00:10.12 | JohnnyC | GSM ? |
| 00:10.26 | Katty | epoch: very confusing to me (= |
| 00:10.29 | epoch | Katty: yeah, I have the super mario brothers 3 "power up" sound as my ringtone ;) |
| 00:10.43 | Katty | i don't have asterisk up yet, right now i'm just dialing ips direct...no outside/incoming |
| 00:10.43 | Moc | Katty, you just need to take the default config, and only change what you want to change.. |
| 00:10.45 | epoch | Moc: heh, I'd also like one of those ;) |
| 00:10.54 | Katty | short of the firewall forward port requests |
| 00:10.57 | jerryh | epoch: beats the heck out of a "Dig-Dug" or "Joust" rington |
| 00:11.02 | epoch | hahahaha |
| 00:11.02 | jerryh | flap flap flap |
| 00:11.12 | Katty | so, how do you change the samples eh? |
| 00:11.15 | epoch | the cool thing about the polycoms is the ringtones are done by a soft-synth |
| 00:11.17 | Nugget | the joust respawn sound would be cool. |
| 00:11.19 | Katty | i didn't see a pretty browse button |
| 00:11.27 | jerryh | Nugget: true true |
| 00:11.29 | ariel_ | I like the joust flap flap. |
| 00:11.36 | epoch | and in _theory_ you can set up any chords you want |
| 00:11.45 | mesi | Is anybody willing to connect his asterisk to mine? |
| 00:11.54 | Dr-Linux | Moc: can u ? |
| 00:11.56 | ta[i]nted | mesi where are u located? |
| 00:11.58 | tuxinator_linux | hey Katty |
| 00:11.59 | Goshen | mesi: why? |
| 00:11.59 | Katty | epoch: yesyesyes, but how do you change? |
| 00:12.00 | epoch | Katty: just set the filename in the ipmid.cfg file on the bootserver |
| 00:12.21 | epoch | Katty: and make sure the file's on the FTP site |
| 00:12.22 | Katty | epoch: gosh, that went right over my head |
| 00:12.28 | Katty | oh |
| 00:12.31 | Katty | duh |
| 00:12.37 | Katty | i thought you meant on asterisk |
| 00:12.56 | mesi | ta[i]nted: Germany, Duesseldorf |
| 00:13.00 | Dr-Linux | Moc: i my friend using x-lite can hear me, but i can't hear him, can you help with ? |
| 00:13.02 | Katty | i need a giggly ringtone |
| 00:13.04 | epoch | Katty: heh, no... on teh polycom boot server :) |
| 00:13.12 | mesi | Goshen: Don't know. 'Cause it's fun? :-) |
| 00:13.20 | Katty | epoch: :> |
| 00:13.27 | Juggie | hahaha |
| 00:13.32 | ariel_ | mesi, are you looking for a dundi setup? |
| 00:13.34 | yaboo | whats the best way/method to do video conferencing? |
| 00:13.37 | Dr-Linux | grrrrrrrr |
| 00:13.52 | Juggie | "Agent on Agent/1000 hungup on the customer. They're going to be pissed." |
| 00:13.53 | Dr-Linux | grrrrrrrrrrrr |
| 00:13.59 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 00:14.12 | modulus_ | perl -pli.orig -e 's/CFLAGS\+\=-D_GNU_SOURCE/CFLAGS\+\=-D_GNU_SOURCE\nCFLAGS\+\=-DMYSQL_LOGUNIQUEID\n/' asterisk-addons/Makefile |
| 00:14.13 | modulus_ | fjear |
| 00:14.18 | mesi | ariel: Don't remeber what dundi is. I am looking for just a connection by iax. |
| 00:14.45 | ariel_ | mesi, to do what? |
| 00:14.47 | Katty | do the ringtones have to be .wav, epoch? |
| 00:14.59 | epoch | Katty: yeah, but the format has to be pretty specific |
| 00:15.11 | epoch | Katty: G.711 uLaw, mono, 8khz |
| 00:15.12 | Katty | will a midi work? |
| 00:15.25 | Katty | k |
| 00:15.26 | footnote | kewl, eye yam footnote |
| 00:15.29 | epoch | Katty: if you convert it to a waf in that format, yeah :) |
| 00:15.39 | *** part/#asterisk Grant2 (~grant@adsl-39-232.swiftdsl.com.au) |
| 00:15.51 | Katty | and if i forget what format it needs to be in, is there a comment somewhere? |
| 00:15.53 | mesi | ariel: Make calls to each other. Just for fun. But in fact I can offer "dundi" for you, to all ptsn in Germany. At zero cost :-) |
| 00:15.56 | ariel_ | mesi, take a look at #dundi they are made for use with asterisk. |
| 00:16.09 | epoch | Katty: yeah, check voip-info.org -- that's where I found it |
| 00:16.15 | mesi | ariel: thanks! |
| 00:16.17 | modulus_ | perl -pli.orig -e 's/#define DATE_FORMAT/#define MYSQL_LOGUNIQUEID\n#define DATE_FORMAT/' asterisk-addons/cdr_addon_mysql.c |
| 00:16.19 | modulus_ | fjear |
| 00:16.20 | epoch | Katty: I think they even have the appropriate sox line to convert it with |
| 00:16.22 | Katty | epoch: kk |
| 00:16.34 | epoch | anyways, I have to go home now... good luck ;) |
| 00:17.04 | Katty | buh bye |
| 00:17.59 | footnote | I see there is a cdr module for postgres? |
| 00:20.48 | modulus_ | there sure is |
| 00:21.44 | footnote | this is the neatest package i've run across in a long time :) |
| 00:21.54 | modulus_ | no it isn't |
| 00:22.03 | footnote | hrm |
| 00:22.09 | footnote | *pop* |
| 00:22.15 | footnote | didja hear that? |
| 00:22.28 | footnote | my bubble burst :( |
| 00:23.41 | *** join/#asterisk Matt-E- (~Matt-E-@66-224-125-137.atgi.net) |
| 00:27.59 | tuxinator_linux | hmm, time to raid the fridge |
| 00:28.19 | ariel_ | tuxinator_linux, what a great Idea. |
| 00:28.46 | tuxinator_linux | the sad part is I know there is nothing in there, but I will go check anyways |
| 00:28.49 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 00:29.33 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 00:29.40 | ariel_ | tuxinator_linux, not in mine. There is lots of stuff there. |
| 00:30.12 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 00:30.25 | *** join/#asterisk djMax (~djMax@artsalliancelabs.com) |
| 00:31.17 | modulus_ | SHE BANG |
| 00:31.19 | modulus_ | SHE BANG |
| 00:31.31 | tzanger | I like women who bang |
| 00:31.36 | Katty | mmm, i have the intro to kylie minogue's can't get you out of my head for my ringtone |
| 00:31.42 | tzanger | haha |
| 00:31.50 | tzanger | I hate hate hate HATE musical ringtones |
| 00:32.09 | Katty | it's better than our currnt ERRRRRRRRRRRRRRRRRRRRRRRRRR noise when we get buzzed |
| 00:32.14 | djMax | any ideas why * and 0 wouldn't work in VM? I have operator=yes and such |
| 00:32.20 | Katty | our current phone system is older than me! |
| 00:32.27 | tzanger | Katty: sure but vibrate mode is the best... totally unobtrusive to everyone else |
| 00:32.40 | Katty | tzanger: i can't get the ip phone into my pocket |
| 00:32.46 | Katty | it's too *big* |
| 00:32.52 | tzanger | Katty: what, her name is Gladys and she works in Switchboard? :-) |
| 00:32.56 | ManxPower | djflux, The voicemail app is on a macro and you don't have exten => o,1,blah in that macro |
| 00:33.06 | tzanger | hmm |
| 00:33.09 | tzanger | ok it's decided |
| 00:33.12 | tzanger | I'm not a fan of thai food |
| 00:33.15 | Katty | :< |
| 00:33.18 | Katty | more for me :> |
| 00:33.19 | tzanger | red curry chicken's not bad |
| 00:33.24 | Katty | oh |
| 00:33.26 | tzanger | I do NOT like pad thai |
| 00:33.29 | Katty | :< |
| 00:33.43 | djMax | but I thought * was "built in" to the Voicemail app? |
| 00:33.50 | tzanger | I love korean food though... dolsot bibimbap and beef gulgogi |
| 00:33.51 | tzanger | mmmmmmmmmmmmmmmmmmmmmm |
| 00:33.53 | djMax | I'm trying to figure out how to call out once in VM |
| 00:33.58 | tzanger | and vietnamese food |
| 00:34.03 | tzanger | although I've only had it once |
| 00:34.09 | tzanger | and bubble tea gets nauseating after a while |
| 00:34.23 | JohnnyC | My number apears Unkwown number how can I send my Number ? |
| 00:34.33 | ta[i]nted | tzanger lol u been talking about food all day |
| 00:34.33 | ariel_ | I am with tzanger except the Korean food. Can't stand the smell. Even when I was in Korea I did not like it. |
| 00:35.06 | tzanger | ta[i]nted: :-) |
| 00:35.29 | tzanger | ariel_: well bibimbap doesn't smell and gulgogi...well it smells just like sirloin beef since that's exactly what it is :-) |
| 00:35.33 | tzanger | I love miso |
| 00:35.35 | tzanger | and miso |
| 00:35.40 | tzanger | miso orny |
| 00:36.55 | ariel_ | tzanger, ever been to Korea? |
| 00:36.57 | tzanger | nope |
| 00:37.02 | tzanger | never been out of north america |
| 00:37.04 | tzanger | ugh |
| 00:37.05 | tzanger | screw this |
| 00:37.14 | tzanger | I ate the chicken bits and now I'm gonna warm up some 3-day-old chili |
| 00:37.15 | ariel_ | Well that place took my view of food away. |
| 00:37.29 | ta[i]nted | how do i change CLI verbosity for remote connects? |
| 00:37.35 | tzanger | ta[i]nted: set verbose |
| 00:37.44 | JohnnyC | how can I enable the others viewing my number ? |
| 00:37.44 | ta[i]nted | permanently as well? |
| 00:37.50 | djMax | so when I hit * in VoiceMailMain, who is supposed to handle where that goes? |
| 00:37.51 | ariel_ | I found some leftover Fried rice and teriyaki chicken... going to the microwave with it. |
| 00:37.54 | tzanger | well until the next set or startup of * |
| 00:38.10 | ta[i]nted | is there a .conf file somewhere to set remove verbosity? |
| 00:38.16 | ManxPower | djflux, The dialplan needs an exten => a, I think. |
| 00:38.24 | tuxinator_linux | I found some toquitos |
| 00:38.53 | tzanger | toquitos? |
| 00:39.04 | tuxinator_linux | rolled tacos |
| 00:39.27 | tzanger | my son had quesedillas (however you spell it) with his mom tonight |
| 00:39.46 | Matt-E- | toquitos are awsome |
| 00:39.49 | tzanger | he pronounced it "quilly dillahs" |
| 00:40.21 | ta[i]nted | ManxPower any ideas on how to set remote CLI verbosity permanently? |
| 00:40.25 | Juggie | bah, theres no good way to log agents out. |
| 00:40.32 | tzanger | remote needs a LOT of work |
| 00:40.35 | TSCHAK | *sigh* |
| 00:40.55 | tzanger | for starters, not polluting EVERY attached CLI with "-- remote connection established" and "-- remote connection closed" |
| 00:41.08 | tzanger | and -rx needs to give proper output, not output from "around the time of the command execution" |
| 00:41.25 | ta[i]nted | i must say.. asterisk wiki is improving a lot |
| 00:41.32 | tzanger | ta[i]nted: yeah? |
| 00:41.45 | KalD|Work | ~docs |
| 00:41.46 | 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 |
| 00:42.20 | tuxinator_linux | tzanger: http://i2.peapod.com/c/OD/ODUSV.jpg |
| 00:42.24 | ManxPower | ta[i]nted, start Asterisk with the verbosity you want. |
| 00:42.36 | ta[i]nted | ManxPower but for remote CLI |
| 00:42.46 | ManxPower | ta[i]nted, You can't. |
| 00:43.04 | *** join/#asterisk tclark (~TC@S0106000c413a1c61.gv.shawcable.net) |
| 00:43.11 | ManxPower | If "permanently" you mean "after an asterisk restart" |
| 00:43.51 | tzanger | terracon: ahh |
| 00:44.04 | *** join/#asterisk julianjm (~julianjm@250.Red-80-59-67.pooles.rima-tde.net) |
| 00:44.10 | ta[i]nted | yea |
| 00:44.26 | ta[i]nted | ManxPower i figured out what the dialstatus issue was |
| 00:44.32 | ManxPower | ta[i]nted, What was it? |
| 00:44.42 | ta[i]nted | ManxPower DeadAGI() vs AGI() |
| 00:44.49 | ManxPower | ta[i]nted, Ah. |
| 00:45.45 | ta[i]nted | ManxPower I don't get the concept of 'no live channels' behind DeadAGI().. since the user is still using a channel.. |
| 00:46.19 | Juggie | anyone have any idea of how an automated agent login/logout could be done... i can do a like 5*#### where #### = their agent id, then ask for a pin and logged in |
| 00:46.45 | Juggie | but logging out is not so easy.... i would like to do something similar but i cant see anyway, anyone have any ideas? |
| 00:46.47 | ta[i]nted | or by channel do they mean that two or more parties (not including asterisk) |
| 00:47.51 | tzanger | jesus |
| 00:47.56 | tzanger | I thought I flipped on a porn channel |
| 00:48.02 | tzanger | but it was a commercial for an ashanti cd |
| 00:50.23 | Katty | same thing |
| 00:50.24 | *** join/#asterisk afe ([9U5keEKcE@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 00:50.31 | tzanger | of course |
| 00:50.37 | tzanger | my chilli bowl hasn't been washed in 3 days either |
| 00:50.41 | tzanger | adds to the flavour I htink |
| 00:50.41 | Katty | i have a tofu pup! |
| 00:50.56 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 00:51.11 | JohnnyC | How can I show my Caller ID to another Phone ? Im using ISDN CAPI |
| 00:52.10 | tzafrir_home | any IRC channel for FOP? |
| 00:52.11 | ariel_ | I just love left over Teriyaki chicken and fried rice. Now I need a cup of coffee. |
| 00:52.34 | ariel_ | tzafrir_home, not that I know of. |
| 00:52.36 | Katty | mmm, fried rice |
| 00:52.50 | ariel_ | Katty, yes home made great. |
| 00:52.57 | tzanger | coffee at this time of night? |
| 00:53.35 | ariel_ | tzanger, yes in fact the really hard stuff. Cuban Coffee... |
| 00:53.44 | tzafrir_home | anyway, for the debianinsts among you, my op-panel debs now seem to be useful enough for the public. http://tzarir.org.il/rapid/APT.html |
| 00:53.53 | tzanger | wow |
| 00:54.11 | PyroSteve | hello my fellow * users |
| 00:54.12 | ta[i]nted | do u guys ever play with music sampling in ivr intros? |
| 00:54.16 | ariel_ | tzanger, its just a little stronger the Espresso. |
| 00:54.22 | ta[i]nted | PyroSteve hello pyrosteve |
| 00:54.34 | PyroSteve | i wanted to remove asterisk and so I did rm * |
| 00:54.37 | PyroSteve | opps |
| 00:54.40 | PyroSteve | just joking |
| 00:54.52 | PyroSteve | hehe get it ? rm * <asterisk> |
| 00:55.00 | PyroSteve | im sure I have a flame coming on |
| 00:55.00 | ReVoK | nice :) |
| 00:55.07 | tzanger | PyroSteve: heh |
| 00:55.19 | PyroSteve | now why would i want to remove asterisk |
| 00:55.22 | PyroSteve | anyway |
| 00:55.24 | ariel_ | see you all later. It's feeding time for my baby... |
| 00:55.41 | PyroSteve | ariel_: breast feeding ?? :) |
| 00:55.58 | tzanger | dammit PyroSteve |
| 00:56.05 | tzanger | I damn near spit chili at the screen |
| 00:56.16 | ManxPower | ariel_ is a multi-talented guy. |
| 00:56.17 | PyroSteve | ok ill stop being obnixous |
| 00:56.22 | PyroSteve | (spelling) |
| 00:56.27 | PyroSteve | ?? |
| 00:56.31 | PyroSteve | ok ok ok |
| 00:56.34 | Katty | PyroSteve: ariel_ is a male, dear. |
| 00:56.36 | PyroSteve | I need some advice |
| 00:56.45 | tzanger | Katty: he knows, I htink |
| 00:56.46 | tzafrir_home | no ispell support in jbot yet? |
| 00:56.49 | Katty | tzanger: k |
| 00:56.52 | tzanger | which is why I almost spit chili at the screen :-) |
| 00:56.55 | PyroSteve | hehe |
| 00:56.56 | Katty | ya |
| 00:56.58 | Katty | i return... |
| 00:57.02 | Katty | and i all i read is breasts |
| 00:57.03 | ta[i]nted | tzafrir_home your link doesn't work |
| 00:57.07 | Katty | everything else just...doesn't get read |
| 00:57.08 | tzanger | hahaha |
| 00:57.19 | tzanger | so you're a boob girl then huh |
| 00:57.24 | tzanger | I didn't think man boobs were attractive |
| 00:57.26 | Katty | girls are great |
| 00:57.45 | tzanger | I agree |
| 00:57.53 | ta[i]nted | i have a feeling a/s/l is coming up sometime soon |
| 00:57.54 | tzanger | we may not be able to eat together but we can go bird watching anytime |
| 00:57.56 | Katty | let's make a club |
| 00:57.57 | ManxPower | Reminds me of the person that didn't get their question answered on the mailing list, so they resent it with a "Subject: Hot breasts!" or something like that. |
| 00:58.10 | ariel_ | PyroSteve, it's just a bottle with milk and other stuff call formula that baby's drink.... |
| 00:58.10 | tzanger | ManxPower: hahaha |
| 00:58.10 | PyroSteve | i have several users using my asterisk server. I want to setup my dialplan so it will throw in the area code that is local to the caller when the caller dial only 7 digits |
| 00:58.11 | ta[i]nted | haha .. classic |
| 00:58.16 | tzanger | that happens on every list at some point |
| 00:58.26 | tzanger | PyroSteve: ok, so do it |
| 00:58.27 | Katty | ariel_: i bet it's soy! |
| 00:58.28 | tzafrir_home | http://tzafrir.org.il/rapid/APT.html |
| 00:58.29 | tuxinator_linux | tzanger: My jaw droped the first time I saw an had for http://www.hottrimmer.com/. My wife and I couldn't stop laughing. |
| 00:58.39 | tzanger | oh dear |
| 00:58.40 | tzafrir_home | (with the right domin name) |
| 00:58.47 | ariel_ | Katty, no real stuff made from Enfimil. |
| 00:59.05 | tzanger | HAHAHAHAHA |
| 00:59.07 | Katty | ariel_: real stuff? from...enfimil? is that a brand or something? |
| 00:59.09 | ta[i]nted | good lord |
| 00:59.10 | PyroSteve | tzanger: so I need to use a database that holds the local area code for each sip user ... so my question is ... |
| 00:59.37 | PyroSteve | should I use the built-in database that asterisk comes with |
| 00:59.38 | Qwell | ariel_: welcome me to the club... :p |
| 00:59.40 | Katty | there's something wrong about powdered breast milk |
| 00:59.46 | tzanger | PyroSteve: why wouldn't you |
| 00:59.47 | PyroSteve | or should I use an sql server ? |
| 00:59.51 | Qwell | its not powdered breast milk |
| 00:59.58 | Qwell | its powdered formula... |
| 01:00.02 | tzanger | PyroSteve: do you need any kind of relational data model? |
| 01:00.14 | ManxPower | Dial(Zap/g1/${CALLERIDNUM:-7}${EXTEN}) or something like that |
| 01:00.17 | tzanger | PyroSteve: if not, do you need any kind of ability to query and add to the database with 3rd party tools? |
| 01:00.21 | Katty | i thought ariel_ said it was /real/ stuff |
| 01:00.24 | tzanger | PyroSteve: if not, use the regular db |
| 01:00.31 | Qwell | ManxPower: Whats the :-7 do? |
| 01:00.37 | ta[i]nted | tuxinator_linux there was an ad for this on television!?! |
| 01:00.39 | tzafrir_home | PyroSteve: the standard dialplan rules should suffice |
| 01:00.39 | Katty | but i don't have kids, so obviously i wouldn't know |
| 01:00.41 | Qwell | remove from the front, or something? |
| 01:00.43 | *** join/#asterisk jcims (~jcims@cpe-69-135-121-57.columbus.rr.com) |
| 01:00.55 | Qwell | Katty: "real formula" |
| 01:01.11 | epoch | what counts as pretend formula, I wonder |
| 01:01.12 | JohnnyC | anyone from Germany or Spain ? |
| 01:01.17 | ariel_ | Katty, it's as close as I can get. I add real cows milk to it. |
| 01:01.22 | Katty | ahh, k |
| 01:01.26 | Qwell | ariel_: seriously? |
| 01:01.29 | tzanger | powdered breast milk?? |
| 01:01.33 | ManxPower | Qwell, It's my too lazy to look up the correct notation for "strip off last 7 digits" |
| 01:01.42 | Qwell | ManxPower: ahh, gotcha |
| 01:01.45 | PyroSteve | ManxPower: good idea |
| 01:01.52 | jcims | are there any minipci fxo or fxs boards out there? |
| 01:01.54 | Katty | ariel_: what's the first three ingredients? |
| 01:02.06 | Qwell | ariel_: What do you do, just put less powder? odd |
| 01:02.11 | PyroSteve | ManxPower: My first thought was how I would pass vars from sip.conf |
| 01:02.23 | KalD|Work | Ok when you hit # to transfer a caller and dial that extension - what is the name of that extension that you enter to have the remote caller transfered to? It isnt CONTEXT_TRANSFER =( |
| 01:02.25 | PyroSteve | and the caller id would be one solution |
| 01:02.43 | Qwell | Katty: milk, lactose, and some oil |
| 01:02.50 | Katty | kk |
| 01:02.54 | Qwell | safflower oil? dunno |
| 01:03.02 | *** join/#asterisk NeonLevel (~otto@200.52.142.184) |
| 01:03.15 | Katty | i doubt it's safflower oil |
| 01:03.20 | Qwell | Thats what it says |
| 01:03.24 | Katty | mad |
| 01:03.29 | Qwell | I put the question mark, because I don't know what it is. :p |
| 01:03.38 | Katty | oh |
| 01:03.42 | Katty | oil made from safflowers |
| 01:03.46 | ManxPower | PyroSteve, Always reuse information when you can. |
| 01:03.50 | Qwell | yeah...duh |
| 01:03.52 | Qwell | What are safflowers? |
| 01:04.04 | ManxPower | lactose: *shudder* |
| 01:04.10 | ariel_ | oh god the smellll it's changing time.....argh |
| 01:04.22 | tzanger | ariel_: I'll be emitting some of those smells in a while |
| 01:04.25 | Qwell | ariel_: I just joined the club a few days ago. heh |
| 01:04.28 | tzanger | hopefully without any texture |
| 01:04.29 | Katty | http://www.dfg.ca.gov/lands/wa/region2/graylodge/graylodgeslides/images/15%20Safflower%200703.jpg |
| 01:04.31 | Qwell | just brought him home today, actually |
| 01:04.50 | CoaxD | cats suck. Unless you get one of those extremely lovey dovey ones |
| 01:04.52 | ariel_ | Qwell, say good by to your sleep |
| 01:05.00 | Qwell | ariel_: So I've been warned |
| 01:05.00 | Katty | aww |
| 01:05.00 | tzanger | CoaxD: no way |
| 01:05.02 | CoaxD | i had 2 cats. they took over my being |
| 01:05.03 | tzanger | I prefer the aloof ones |
| 01:05.05 | Katty | new baby in the house :> |
| 01:05.14 | tzanger | this current one is a lovey dovey one |
| 01:05.19 | CoaxD | and then when my kid was diagnosed with asthma (and also, a cat allergy) the cats went on to a new home ;) |
| 01:05.21 | Katty | cats++ |
| 01:05.22 | tzanger | and he shits on the floor if I don't give him enough attention |
| 01:05.33 | tzanger | so he's currently chilling is ass in -17oC weather |
| 01:05.37 | tzanger | for that stunt |
| 01:05.45 | Qwell | -170c? |
| 01:05.47 | CoaxD | tzanger: That right there is reason to put a bullet in its head |
| 01:05.54 | Qwell | hmm, bbl |
| 01:06.07 | tzanger | no not a hundred and seventy below, seventeen below zero, degrees C |
| 01:06.09 | CoaxD | tzanger: (if you've tried training and failed, of course) |
| 01:06.24 | tzanger | CoaxD: first rule of cat ownership: you don't "train" them |
| 01:06.25 | *** join/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu) |
| 01:06.26 | ManxPower | The only time my cats don't use the litter box is when I don't clean the litter box. |
| 01:06.27 | tzanger | you develop an understanding |
| 01:06.40 | CoaxD | tzanger: Put that cat on a bare tile floor (probably bathroom) until it shits in the litter box. THEN and ONLY THEN, let it out |
| 01:06.50 | KalD|Work | tzanger, I trained my cat to shit in the toilet - now she comes running in asking for the morning paper tho =( |
| 01:06.50 | tzanger | CoaxD: no he uses the litter box just fine |
| 01:06.51 | CoaxD | tzanger: thats probably true to some degree |
| 01:06.58 | CoaxD | tzanger: (Cant "train" cats) |
| 01:07.11 | tzanger | but like I said he tends to try and train me and I'm not down with that flow |
| 01:07.15 | footnote | CoaxD: can't herd cats either |
| 01:07.26 | tzanger | KalD|Work: *rolls eyes* |
| 01:08.09 | ManxPower | One of my cats has a thing for any water I touch. |
| 01:08.21 | tuxinator_linux | I just bought my cat a LitterMaid. I hear they crap out, but I was going to try it anyways |
| 01:08.26 | ManxPower | Tries to drink my bathwater, tries to drink out of any cup of water I have. |
| 01:08.29 | CoaxD | tuxinator: Oh god dude |
| 01:08.35 | CoaxD | tuxinator: Your $90 is wasted |
| 01:08.39 | ManxPower | tuxinator_linux, Mine lasted about a year. |
| 01:08.41 | footnote | Cats are from outer space btw |
| 01:08.42 | PyroSteve | ManxPower: In my case, my sip users will be making calls out to the pstn via Broadvoice |
| 01:08.43 | ManxPower | I live my littermail. |
| 01:08.51 | PyroSteve | ManxPower: which means the caller id string is useless |
| 01:08.57 | par | the cat from outer space rules |
| 01:09.00 | CoaxD | tuxinator: too much litter, damn thing wont work. Too little, and the cats get shit everywhere |
| 01:09.07 | JohnnyC | How can I disable Private Number on ISDN ? |
| 01:09.13 | par | with the collar that flashes when he talks |
| 01:09.13 | CoaxD | tuxinator: they are horrid, bad smelling little pieces of shit. And hard to clean, too |
| 01:09.16 | PyroSteve | ManxPower: but I would still be able to use it my area code trick, right ? |
| 01:09.19 | ManxPower | PyroSteve, Set the callerid in sip.conf. |
| 01:09.26 | tzanger | wtf is a littermaid? |
| 01:09.27 | PyroSteve | ManxPower: got ya |
| 01:09.30 | tuxinator_linux | She hasn't used it yet, I have the old one next to it |
| 01:09.34 | CoaxD | tzanger: an automated litter scooper |
| 01:09.41 | ManxPower | tzanger, It's a "self cleaning litterbox" |
| 01:09.48 | tuxinator_linux | http://littermaid.com/ |
| 01:09.49 | CoaxD | tzanger: it is robotic. Cat craps or pees, and 10 minutes later, it scoops |
| 01:09.50 | tzanger | ugh |
| 01:09.55 | tzanger | that's idiotic |
| 01:09.57 | footnote | tzange: high-tech turd handling |
| 01:10.04 | ManxPower | tzanger, It's great! |
| 01:10.06 | CoaxD | tzanger: the idea is, at the end of it, you just throw out a thingie of turds |
| 01:10.13 | CoaxD | tzanger: And you never actually have to scoop |
| 01:10.17 | CoaxD | personally? I'D RATHER SCOOP |
| 01:10.20 | ManxPower | Clean the bin where it scoops stuff about every 5 days or so. |
| 01:10.21 | NeonLevel | what is the best way to learn AGI? |
| 01:10.25 | tuxinator_linux | but you have to buy the stupid turn holders |
| 01:10.37 | CoaxD | tuxinator: yeah, the turd holders are expensive, too |
| 01:10.37 | tzanger | me too |
| 01:10.44 | CoaxD | tuxinator: You can reuse them, but taht really does defeat the purpose |
| 01:10.49 | tzanger | "turd holders" |
| 01:10.50 | tzanger | hahahahaha |
| 01:10.54 | JohnnyC | ManxPower: caler id in sip.conf , didnt work |
| 01:11.02 | tuxinator_linux | I am temped to stick a bag inside and toss the bad |
| 01:11.05 | ManxPower | I can leave for up to 5 days on business trips and my apartment doesn't smell like litterbox when I come back. |
| 01:11.13 | CoaxD | Manx: Mine sure did |
| 01:11.14 | ManxPower | JohnnyC, Huh? |
| 01:11.26 | CoaxD | Manx: even after about 2 days, the goddamn cats had the bathroom smelling like a war zone |
| 01:11.37 | CoaxD | Manx: (Complete with the stench of 2 week old dead bodies in tow) |
| 01:11.48 | JohnnyC | ManxPower: callerid=yes or no by user or by [Â ] |
| 01:11.49 | JohnnyC | ? |
| 01:11.57 | ManxPower | JohnnyC, What are you trying to do? |
| 01:12.03 | tuxinator_linux | I use Arm and Hammer Cat litter, the littermaid rakes through that fine. |
| 01:12.13 | CoaxD | I used Tidy Cats |
| 01:12.22 | CoaxD | and one other; cant remember the name, but it was good |
| 01:12.27 | tzanger | bah |
| 01:12.57 | CoaxD | I just dont' understand why ANYONE would want to own cats |
| 01:13.01 | JohnnyC | ManxPower: Im trying to call from my SIP phone using Asterisk and an ISDN line |
| 01:13.01 | CoaxD | they are fucking HORRID little bastards |
| 01:13.08 | CoaxD | especially the ones I had. Fuck. |
| 01:13.16 | ManxPower | JohnnyC, I can't help you. |
| 01:13.16 | JohnnyC | it calls ok , but the number doesnt get identified |
| 01:13.22 | tuxinator_linux | My wife wanted a baby or a cat. We got the cat. |
| 01:13.23 | JohnnyC | ok |
| 01:13.27 | ManxPower | CoaxD, Better than a codependent dog. |
| 01:13.31 | CoaxD | Thank god my kid had allergies. I was thinking I was going to have to holdonto those bastards for 20 years |
| 01:13.41 | CoaxD | Manx: I dunno. Dogs are excellent creatures |
| 01:13.48 | CoaxD | Manx: And they *ARE* codependant. By nature, they live in packs. |
| 01:14.06 | CoaxD | Manx: Cats do have families, but generally,they're a little less formed than that |
| 01:14.28 | KalD|Work | ManxPower, ok - I've looked at the docs - I am going to ask this question please if you can point me to the info: I need to know what variable holds the number the user enters for the transfer request... i.e. I hit # and I'm asked for the exten and I dial 3123 - what variable is that number stored in? |
| 01:14.30 | CoaxD | Manx: Only trouble is, you cant leave your dogs alone for weeks at a time or its too much stress for them |
| 01:14.31 | tzanger | bkw_: a little testy? :-) |
| 01:14.53 | tuxinator_linux | Any of you use a sharper images air purifier? http://www.sharperimage.com/us/en/catalog/productview/sku=SI830SNX/catid=676889373/pcatid=22 |
| 01:14.57 | CoaxD | okay, home time for me |
| 01:15.21 | footnote | music on hold the right way! |
| 01:15.58 | Juggie | hrm queue ringall doesnt work with grouping... |
| 01:17.42 | ManxPower | KalD|Work, ${EXTEN} |
| 01:19.12 | KalD|Work | ManxPower, Ok - I guess I don't understand how tranfser works then- If I call 3911 and answer the phone then hit # then dial 3901 it looks for 3911 in my context - so I'm doing 3911,1,Dial(iax2/(whatever I wanted to transfer to),,r) but if I put ${EXTEN} in then it just loops... does that make sense? |
| 01:20.19 | *** join/#asterisk myloforreal (~mylo@c211-30-18-157.thorn1.nsw.optusnet.com.au) |
| 01:21.43 | ManxPower | KalD|Work, Um, use exten => 3911,1,Dial(iax2/(whatever I wanted to transfer to),,r) |
| 01:22.02 | ManxPower | Why are you using the # hack anyway? Is your SIP device too braindead to do it's own transfers? |
| 01:22.15 | KalD|Work | I'm doing IAX |
| 01:22.24 | myloforreal | USB soundcards - opinions? |
| 01:22.32 | ManxPower | Do your IAX phone is too stupid to do it's own transfers? |
| 01:22.33 | KalD|Work | dial from a cell into an iax client - then wanna hit # to transfer it to another |
| 01:22.40 | KalD|Work | yeah |
| 01:23.04 | ManxPower | I don't understand what the problem is. |
| 01:24.19 | ta[i]nted | is AMP any good? |
| 01:24.52 | Mavvie | somebody here with a Cisco call manager? |
| 01:25.09 | tuxinator_linux | It's 6:30 PM here, I think it is time to shower |
| 01:25.30 | Mavvie | just wondering if your unity box has an http proxy on port 8080 and an NNTP server running. |
| 01:25.44 | *** join/#asterisk zignig (~simon@203.217.15.10) |
| 01:25.52 | shmaltz | Does anybody know where I can get an mpg123 for AMD 64 arch? |
| 01:26.09 | *** join/#asterisk yaboo (~jsirucka@220.245.131.131) |
| 01:26.09 | KalD|Work | ok this is how i'm trying to make it work (maybe I'm just nuts): cell phone dials into main number: hits 1 - calls my iax2 client at 3911 - I decide to pass the call to friend - I hit # dial 3901 - * should then send the call to IAX2/3901... so what I have is [inbound] exten => 1,Dial(iax2/3911,30,tr) -- so when I dial 3901 it looks for [inbound] 3911,1 and if it finds it it goes on else it drops the call - so my question is what do I p |
| 01:26.10 | KalD|Work | ut in 3911,1,Dial(iax2/{extn},30,tr) to make it pick up the ext I put in for the transfer? does that make more sense? |
| 01:26.10 | ManxPower | shmaltz, "make mpg123" in the Asterisk source directory. |
| 01:26.26 | shmaltz | thanks ManxPower |
| 01:26.46 | ManxPower | KalD|Work, Think of # as Dialing a new call. |
| 01:26.51 | KalD|Work | ok |
| 01:27.11 | epoch | man, I hate the # option |
| 01:27.19 | ManxPower | KalD|Work, You don't "make it pick the extension to dial" Whatever you dial after # will act just like you dialed the phone. |
| 01:27.20 | KalD|Work | so the ${EXTEN} will be the new number not the exten => 3911,1 exten |
| 01:27.32 | ManxPower | KalD|Work, IGNORE ${EXTEN}! |
| 01:27.35 | KalD|Work | ...ok |
| 01:27.47 | *** part/#asterisk tuxinator_linux (~tuxinator@ip68-109-146-168.ph.ph.cox.net) |
| 01:27.49 | zignig | has anyone got sphinx voice recognition working with asterisk ? |
| 01:28.27 | zignig | voice commands would be cool. |
| 01:28.29 | Juggie | ManxPower, should agent groups be working with ringall/roundrobin w/memory etc... |
| 01:28.47 | Juggie | if i specify every agent that stuff works, but if i use the group, no such love. |
| 01:28.49 | Mavvie | zignig: voice recognition is more difficult than DTMF recognition |
| 01:28.51 | ManxPower | Juggie, I don't know anything about queues and agents other than what I've read on the mailing list. |
| 01:29.25 | epoch | zignig: I dunno if you've ever used voice recognition stuff, but it's the most annoying thing *ever* ;P |
| 01:29.39 | epoch | ever talked to Bell Canada's Emily, for example? |
| 01:30.03 | Juggie | epoch, thats likely nuonce |
| 01:30.04 | ManxPower | A year or so ago someone got Sphynx working with Asterisk, but only using ulaw or alaw codec and it's accuracy was terrible. |
| 01:30.06 | shmaltz | ManxPower, make mpg123 is giving me tons of errors |
| 01:30.09 | epoch | I usually just end up yelling some gibberish until she lets me use DTMF |
| 01:30.16 | epoch | Juggie: nuonce? |
| 01:30.17 | zignig | epoch: Telstra (.au) has one, and yes it's terrible |
| 01:30.25 | Juggie | epoch, its a company that produces voice rec |
| 01:30.28 | epoch | ah |
| 01:30.35 | Juggie | its decent if u are looking for keywords |
| 01:31.03 | zignig | i'd just like to be able to pick upthe phone , press 0 and say 'music' and have the mp3 juke box arc up :) |
| 01:31.03 | epoch | well, anyways, my point is I really don't like the idea of having to say things into the phone when I could just as well be pressing buttons |
| 01:31.06 | myloforreal | telstra in general = terrible |
| 01:31.23 | epoch | cuz when you get used to a certain phone system, you can navigate the menus *way* faster using DTMF |
| 01:31.28 | zignig | myloforreal: yep. |
| 01:31.55 | myloforreal | zignig: any decent ip carriers in australia? |
| 01:32.00 | KalD|Work | ManxPower, this is what is confusing me: (pasting 3 lines) -- Goto (INCOMING,3911,1) |
| 01:32.00 | KalD|Work | Mar 2 16:23:19 WARNING[18653]: pbx.c:1970 ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension '3911' in context 'INCOMING', but no invalid handler |
| 01:32.00 | KalD|Work | <PROTECTED> |
| 01:32.02 | epoch | zignig: wouldn't doing something like picking up the phone and dialing *1 or something be a lot faster to get an mp3 player going? ;) |
| 01:32.24 | *** join/#asterisk faaa_ (faceoff@devel.acdbddh.eu.org) |
| 01:32.26 | *** part/#asterisk faaa_ (faceoff@devel.acdbddh.eu.org) |
| 01:32.34 | *** join/#asterisk faaa_ (faceoff@devel.acdbddh.eu.org) |
| 01:32.39 | faaa_ | hello |
| 01:32.40 | ManxPower | KalD|Work, Set relaxdtmf=no |
| 01:32.44 | faaa_ | cypromis are you there? |
| 01:32.48 | ManxPower | KalD|Work, You have a DTMF problem |
| 01:32.56 | ManxPower | Turn on LONG TONES in your cell phone too |
| 01:33.13 | KalD|Work | ManxPower, ok I'll give that a shot - THX! =) |
| 01:35.11 | *** join/#asterisk bratner (~bratner@bzq-179-152-71.pop.bezeqint.net) |
| 01:36.22 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 01:36.25 | *** join/#asterisk marc324 (~marc32344@69-28-224-214.dsl.teksavvy.com) |
| 01:37.15 | bratner | i wounder , how come digium E1 card costs at least twice less then the intel one? |
| 01:37.27 | tzanger | bratner: telecom has always been expensive |
| 01:37.35 | tzanger | digium isn't in bed with the telecoms |
| 01:38.09 | bratner | tzanger: ok. |
| 01:39.02 | bratner | tzanger: i just have a potential project with asterisk and my competition according to the rumors uses intel cards. |
| 01:39.03 | tzanger | bratner: I have great successes with their T100P and TE405P cards |
| 01:39.08 | ManxPower | bratner, The Dialogoc/Intel cards have built in CPUs for sound processing. Digium's cards do not, they use the computer CPU for htat. |
| 01:39.14 | tzanger | bratner: what are your specific requirements |
| 01:39.24 | Goldenear | I'm testing the fwd iax2 service. I have two PC here, one with XP the other with Linux. IAXComm is installed on both and each PC/IAXCOM has its own account to iax2.fwdnet.net. I'm going thru a NAT router. Is it normal that I have a very long delay between speaking on one PC and receiving the sound on the other one ? |
| 01:39.47 | Goldenear | (about 6 sec) |
| 01:40.22 | bratner | tzanger : ability to handle 30 paralel calls to asterisk based IVR. basically a playback system. |
| 01:40.41 | tzanger | bratner: that's not very specific but yes, asterisk should have no issues doing that iwth a TE110P |
| 01:40.52 | moonwick | depends, are you using a satellite internet link? |
| 01:40.56 | tzanger | bratner: a word of advice: don't skimp on the processor or motherboard |
| 01:41.12 | Goldenear | I'm in France, so I guess the "voice" have to go to USA and go back to France |
| 01:41.29 | Goldenear | I use a cable modem connexion to the internet |
| 01:41.32 | *** part/#asterisk JohnnyC (~JoaoCorre@81.193.116.63) |
| 01:41.34 | moonwick | I wouldn't think it'd take that long |
| 01:41.46 | Goldenear | is this a normal delay ? |
| 01:41.46 | bratner | tzanger: what should i know to make the data more specific? |
| 01:42.17 | bratner | tzanger: what are the "parameters"? |
| 01:42.21 | tzanger | bratner: do you have to do any codec conversion? CAS E1 or E1 PRI? what switchtype? what are you playing back? |
| 01:42.34 | Goldenear | is this the delay I have to expect for a Voip call from one side of the planet to the other ? |
| 01:42.55 | ariel_ | back baby is feed and now sleeping... |
| 01:43.21 | tzanger | ariel_: that's about all the do for the first while... eat, shit, sleep with brief periods of cooing, giggling and crying |
| 01:43.57 | ariel_ | tzanger, yes but after awhile they start doing more... |
| 01:44.16 | ManxPower | Yeah, like getting pregnant and crashing the family car. |
| 01:44.20 | tzanger | bratner: as I said, when you spec out your system, I *strongly* recommend Intel P4 and a GOOD motherboard |
| 01:44.45 | *** join/#asterisk MikeJ[Jayden] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
| 01:44.47 | Goldenear | moonwick: is this asterisk depedend or a normal delay for any VoIP system ? |
| 01:44.52 | bratner | tzanger: i was thinking towards dual xeon $2500 server |
| 01:44.58 | tzanger | AMD works, and so do the cheap little all-in-one motherboards but unless you are experienced in working through the little issues that tend to crop up, stick with Intel and good quality systems |
| 01:45.19 | tzanger | bratner: you won't need that much processor -- get a single xeon maybe (dual capable mobo) and if you need it you can always add more |
| 01:46.06 | tzanger | single T1 I can use with a P3/700 system without much effort, but get it working THEN optimize |
| 01:46.15 | tzanger | early optimization is the reason for the fall of the roman empire |
| 01:46.45 | footnote | tzanger: I blame the christians |
| 01:46.46 | *** join/#asterisk redder86 (~lee@gateway.howardsilvan.com) |
| 01:46.59 | tzanger | footnote: who do you think convinced the romans to optimize early? |
| 01:47.03 | footnote | aha! |
| 01:47.08 | Mavvie | hi redder86. |
| 01:47.14 | zignig | footnote: heh, I blame the myan civilization myself. |
| 01:47.25 | footnote | zignig: and the aztecs! |
| 01:47.26 | Mavvie | redder86: Any chance you can recommand a PRI card which works with HylaFax to me? |
| 01:47.46 | redder86 | Mavvie: Patton 2977 |
| 01:48.09 | zignig | footnote: splitters! |
| 01:48.28 | footnote | hehe |
| 01:48.35 | bratner | where can i get more info on the telecom tech-terminology "PRI/CAS...." ? |
| 01:48.36 | Mavvie | redder86: any idea if it works with isdn4linux ? |
| 01:48.58 | tzanger | bratner: google will give you some but it's a lot to wade through |
| 01:49.04 | redder86 | Mavvie: uh, don't use isdn4linux, use the Patton tty drivers... you'll get your average tty interface to the channels |
| 01:49.09 | *** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net) |
| 01:49.21 | Mavvie | redder86: aha, it's own TTY drivers. Coolies. Thanks! |
| 01:49.37 | redder86 | Mavvie: your other option is the Eicon Diva Server... it also has it's own tty drivers |
| 01:49.52 | redder86 | Mavvie: I like the Pattons better, though. |
| 01:50.19 | redder86 | Mavvie: I like the technology on the Pattons better, but the Eicons may be better supported by Eicon than Patton supports theirs. |
| 01:50.26 | Mavvie | as usual, bloody expensive. |
| 01:50.36 | Mavvie | redder86: we now have a quad BRI eicon card. |
| 01:50.39 | redder86 | Mavvie: you can find Patton 2977s on eBay |
| 01:50.41 | bratner | tzanger: what is the internal codec of TE110P card? |
| 01:50.51 | Mavvie | redder86: but it's running at 25% of the time on 7 and 8 channels in use. |
| 01:50.51 | tzanger | slinear |
| 01:50.53 | redder86 | Mavvie: I just picked up a 24-port model for $125 |
| 01:50.55 | tzanger | but your E1's all alaw |
| 01:51.03 | footnote | I remember getting in a big fight with our Eicon rep over Linux support. |
| 01:51.05 | tzanger | so you'll be converting between slinear and alaw which is fine |
| 01:51.12 | tzanger | pretty much 0 cpu time |
| 01:51.13 | footnote | "We will NEVER support Linux!" |
| 01:51.19 | footnote | bwahahaha |
| 01:51.21 | epoch | Eicon's still around!? |
| 01:51.26 | bratner | should all the audio files be in alaw then? |
| 01:51.26 | redder86 | Mavvie: if you're in a crunch and don't want to play the bid game then you can buy-it-now for like $600 |
| 01:51.36 | footnote | epoch: apparently |
| 01:51.43 | Mavvie | I'm surfing to ebay right now to have a look. |
| 01:51.48 | footnote | Hey, heads up. |
| 01:51.52 | footnote | Avoid Sangoma |
| 01:51.53 | epoch | crazy. |
| 01:52.02 | tzanger | footnote: for someone who wants to play around with ISDN BRI? |
| 01:52.10 | tzanger | footnote: what would you suggest |
| 01:52.14 | Mavvie | redder86: well, it's the digium cards I compare it too. they're cheap++, but no support with regarding to TTY drivers. |
| 01:52.18 | footnote | tzanger: they screwed me hard a couple of weeks ago |
| 01:52.20 | tzanger | (sorry that iddn't come out in english, I swear it's my first language) :-) |
| 01:52.40 | footnote | tzanger: They don't *really* have SS7 |
| 01:53.17 | TSCHAK | garrrrr |
| 01:53.22 | bratner | isn't SS7 optic fibers? |
| 01:53.25 | redder86 | Mavvie: and, well, the retail price on the 2977 is something like $3000... and that probably far-exceeds the other options in price |
| 01:53.27 | footnote | nonono |
| 01:53.32 | tzanger | bratner: not necessarily |
| 01:53.47 | redder86 | Mavvie: the other options being a channel bank and 24 analog modems |
| 01:53.48 | ManxPower | SS7 is a protocol telcos use between their own switches. |
| 01:54.03 | Mavvie | redder86: no. no analogue for me again. *ever* |
| 01:54.23 | footnote | It's used for MTSO's and other apps too. |
| 01:55.39 | KalD|Work | ManxPower, I figured it out - I'm a moron - I was playing with DNID - and had left that in my Dial() statement - so whenever I'd dial a number to transfer it'd STILL dial the number that got me into the context in the first place (which is my own extension) thx again! |
| 01:56.38 | redder86 | Okay, so my problem today is that I can't figure out why I can't match extension "5660" with "_566[0]" in the dialplan. |
| 01:56.53 | redder86 | exten => _566[0],1,MusicOnHold will not work |
| 01:57.01 | redder86 | _5660 works |
| 01:57.08 | redder86 | _566X works |
| 01:57.23 | redder86 | it's very confusing |
| 01:57.40 | ManxPower | redder86, Prolly a bug, since logically _5660 and _566[0] are the same. |
| 01:57.49 | redder86 | ManxPower: yeah |
| 01:57.58 | redder86 | ManxPower: I'm using current CVS 1_0 |
| 01:58.00 | ManxPower | Nobody sane would use _566[0] |
| 01:58.20 | redder86 | ManxPower: well, I'm really trying to get _566[02] to work |
| 01:58.21 | ManxPower | redder86, report it as a bug. |
| 01:58.23 | redder86 | 0 or 2 |
| 01:58.30 | ManxPower | THAT should work. |
| 01:58.36 | redder86 | ManxPower: it doesn't |
| 01:59.04 | ManxPower | It should. |
| 01:59.07 | redder86 | ManxPower: I loathe reporting bugs with Asterisk... all of the hassle that the bug people put me through... it's a nightmare |
| 01:59.15 | *** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net) |
| 01:59.27 | ManxPower | redder86, No. I was recommending you report _566[0] as a bug. |
| 01:59.45 | *** part/#asterisk jcims (~jcims@cpe-69-135-121-57.columbus.rr.com) |
| 01:59.47 | ManxPower | If you are trying _566[02] and it's not working, that's a config error. |
| 01:59.50 | Mavvie | '_*2[0]' <- works here. |
| 02:00.05 | redder86 | ManxPower: _566[0] doesn't work, either |
| 02:00.14 | Mavvie | that's on 1.0.6 |
| 02:00.24 | redder86 | ManxPower: what's wrong with _566[02] ? |
| 02:03.21 | *** join/#asterisk Jayden (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
| 02:04.25 | redder86 | Hrmm... _566[0] works on *this* server |
| 02:04.34 | redder86 | but not on the other |
| 02:07.42 | Juggie | redder86, it works for me |
| 02:07.45 | Juggie | but only sometimes. |
| 02:07.46 | Juggie | eg. |
| 02:07.51 | Juggie | if my dialplan has |
| 02:08.05 | Juggie | exten => _402X .... |
| 02:08.07 | Juggie | and |
| 02:08.13 | Juggie | exten => _4029 |
| 02:08.17 | Juggie | that works like you'd expect |
| 02:08.22 | Juggie | if its 4029, it does whatever |
| 02:08.25 | Juggie | and if its anything else |
| 02:08.27 | Juggie | it does that. |
| 02:08.42 | Juggie | however, if you do exten => _402[89] |
| 02:08.48 | Juggie | and leave the first exten |
| 02:08.56 | Juggie | it allways goes to the first exten |
| 02:09.06 | Juggie | remove the frist and keep just exten => _402[89] |
| 02:09.08 | Juggie | and it works |
| 02:09.17 | Mavvie | the one on ebay is going for US$ 700. the price in the shop is US$ 16,000 |
| 02:09.44 | Juggie | maybe for this case you'd need to funk around with the ordering using includes or something. |
| 02:11.24 | mikegrb | Juggie: no |
| 02:11.25 | mikegrb | need |
| 02:11.27 | mikegrb | to |
| 02:11.29 | mikegrb | type |
| 02:11.30 | mikegrb | so |
| 02:11.32 | mikegrb | little |
| 02:11.35 | mikegrb | per |
| 02:11.38 | mikegrb | line |
| 02:13.39 | shmaltz | I'm getting chopped audio with MOH what could it be? |
| 02:14.07 | tzanger | shmaltz: gentoo :-) |
| 02:14.13 | shmaltz | :( |
| 02:14.21 | tzanger | shmaltz: what are you listening to MOH with |
| 02:14.39 | hermie | tzanger: his ear, I would hope :) |
| 02:14.39 | shmaltz | tzanger, I couldn't get version r of mpg123 to compile with amd 64 |
| 02:14.43 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 02:14.46 | shmaltz | so I installed 59r |
| 02:14.52 | shmaltz | i mean 59s |
| 02:15.00 | tzanger | shmaltz: check the lists then |
| 02:15.14 | shmaltz | for what? for s or amd 64? |
| 02:15.22 | tzanger | both |
| 02:16.05 | shmaltz | thanks |
| 02:17.07 | redder86 | Juggie: in my case I have a _566[0] and then later in the same context a _566X, and it appears that the latter is taking precedence over the former |
| 02:17.45 | redder86 | Juggie: yeah, that's what it is |
| 02:18.30 | redder86 | Am I wrong in thinking that the dialplan is read top-down and the first extension that matches is used? |
| 02:18.52 | tzanger | redder86: typically yes |
| 02:18.59 | ManxPower | redder86, In theory the CLOSEST match will happen. |
| 02:19.05 | shmaltz | and mp3player gives me received udp packet with bad checksum |
| 02:19.14 | tzanger | redder86: but it's been said that that's not always the case, as as ManxPower just said, the closest match will hit |
| 02:19.17 | ManxPower | But that assumption can be screwed up by include => |
| 02:19.56 | ManxPower | Maybe you need _566[89] AND _566[0-7] |
| 02:20.11 | shmaltz | ManxPower, whats that? |
| 02:20.15 | tzanger | ManxPower: yuck |
| 02:20.22 | ManxPower | or maybe _566[89] AND _566[0123456] |
| 02:20.27 | tzanger | [89] is considered more "specific" than [0-7] ??? |
| 02:20.30 | ManxPower | tzanger, I agree. |
| 02:20.41 | ManxPower | no, but now the patterns CANNOT overlap. |
| 02:20.51 | shmaltz | tzanger, nothing on the lists |
| 02:20.52 | tzanger | ManxPower: ahhhhhhhh |
| 02:20.56 | TSCHAK | is there a channel for #gnugk ? |
| 02:20.57 | tzanger | ManxPower: I understand now |
| 02:21.03 | ManxPower | No string of digits will match both _566[89] AND _566[0-7] |
| 02:21.12 | tzanger | shmaltz: I know that the required mpg123 version is **VERY** specific |
| 02:21.22 | tzanger | you're already in unchartered territory with amd64 |
| 02:21.28 | tzanger | er |
| 02:21.29 | tzanger | uncharted |
| 02:22.02 | shmaltz | but it doesn't compile on my system, I had problems with it before (like a while ago), but then I found one somewhere that had it all , but now I'm out of luck |
| 02:22.12 | shmaltz | meaning for AMD 64 |
| 02:22.18 | tzanger | you're on the bleeding edge already |
| 02:22.18 | shmaltz | or x86_64 |
| 02:22.24 | shmaltz | :( |
| 02:22.29 | ManxPower | redder86, Idealy _566X and _566[89] would work as expected. You might want to file a bug report. |
| 02:22.31 | tzanger | if you're not willing or able to bugfix this as a developer then it's time to fall back |
| 02:22.57 | ManxPower | I'm a big fan of bounties for this sort of thing. |
| 02:24.37 | KalD|Work | anyone know if the mpg123 that comes w/ slackware is the correct mpg123 to use? |
| 02:24.45 | tzanger | KalD|Work: don't think it is |
| 02:24.56 | tzanger | KalD|Work: but "make mpg123" in asterisk should make the rightv ersion to you |
| 02:25.03 | KalD|Work | k |
| 02:25.24 | tzanger | KalD|Work: remember to removepkg mpg123 so you don't have the old version hanging around causing troubles |
| 02:25.57 | ManxPower | tzanger, I think CVS-HEAD will complain if it seens a wrong version of mpg123 |
| 02:26.19 | redder86 | ManxPower: 90% of the time filing a bug report goes nowhere unless 1) a patch is supplied or 2) it's a blatant bug or 3) someone else agrees with you and writes a patch |
| 02:26.20 | *** join/#asterisk Goshen (~Goshen@c-67-172-238-57.client.comcast.net) |
| 02:26.27 | shmaltz | make mpg123 just errors out on me |
| 02:26.39 | shmaltz | does mpg123 rquire a sound card? |
| 02:26.40 | Goshen | I didn't have to make mpg123 on my machine |
| 02:26.40 | ManxPower | redder86, The problem should be pretty simple to reproduce. |
| 02:26.47 | shmaltz | or a /dev/dsp? |
| 02:26.52 | redder86 | ManxPower: I'd rather save myself the frustration of trying to convince someone that _566[0] is more specific than _566X |
| 02:26.52 | ManxPower | shmaltz, no |
| 02:27.08 | tzanger | shmaltz: as I said, you're in uncharted territory. If you're not willing or able to fix it as a developer you will need to either fall back or as Manxpower suggests, put up a bounty |
| 02:27.11 | zignig | shmaltz: mpg123 outputs onto stdout , that asterisk grabs. |
| 02:27.16 | redder86 | ManxPower: I'd say that the #1 flaw with * development is the bug handling |
| 02:27.27 | zignig | although debian comes with mpg321 which does not seem to work. |
| 02:27.27 | *** join/#asterisk MikeJ[Jayden] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
| 02:27.51 | ManxPower | redder86, That's prolly because there are too few developers, too many bugs, too many new features, and users that file bug reports that are not bugs. |
| 02:28.13 | MikeJ[Jayden] | hey, does anybody know how to configure the asterisk :) |
| 02:28.36 | shmaltz | MikeJ, I dont think so :) |
| 02:28.37 | tzanger | shmaltz: if you're not able or willing to work on * as a developer, don't use bleeding edge hardware. This is the same in any project. |
| 02:28.37 | *** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-175.modem.logical.net) |
| 02:28.45 | ManxPower | Here's a question: Where the hell can I order the Asterisk Book online. |
| 02:28.56 | shmaltz | tzanger, but it was working in FC3 |
| 02:29.00 | MikeJ[Jayden] | asterisk book? |
| 02:29.07 | MikeJ[Jayden] | I saw one on ebay today |
| 02:29.11 | ManxPower | Yes, someone wrote an asterisk book. |
| 02:29.15 | MikeJ[Jayden] | ebay |
| 02:29.16 | Carp1 | Yeah |
| 02:29.18 | Carp1 | I see it all the time. |
| 02:29.20 | tzanger | shmaltz: shmaltz I am willing to put good money on fc3 not using "stock" mpg123 |
| 02:29.22 | Carp1 | It's like $60 on ebay |
| 02:29.23 | tzanger | for amd64 |
| 02:29.47 | ManxPower | I ordered mine at the same time I ordered my sipura (from the same company). That was 3 months ago. Got the SIPura, waiting for the book. |
| 02:29.58 | ManxPower | I'd rather buy it new. |
| 02:30.03 | MikeJ[Jayden] | amazon? |
| 02:30.05 | tzanger | shmaltz: I know for a FACT that gentoo patches the shit out of their 'ports' collection since they feel they know better |
| 02:30.13 | shmaltz | but it worked, not with what FC3 gave me, but I found a RPM that was ported to x86_64 |
| 02:30.23 | ManxPower | MikeJ[Jayden], I was looking for a specific answer, not another question. |
| 02:30.30 | tzanger | shmaltz: so grab the source for htat RPM |
| 02:30.32 | MikeJ[Jayden] | :) |
| 02:30.44 | tzanger | shmaltz: as I said -- THIS IS WORK FOR * DEVELOPERS OR THOSE WHO WANT TO DEVELOP |
| 02:30.46 | shmaltz | i've been looking for it, but I can't find it |
| 02:30.47 | MikeJ[Jayden] | well why didn'y you say so |
| 02:30.52 | tzanger | shmaltz: you don't seem to be accepting that |
| 02:31.13 | MikeJ[Jayden] | work we love to get paid for :D |
| 02:31.21 | shmaltz | tzanger, at the moment I'm not programming, but I WILL GET THIS SYSTEM TO WORK, I'm just trying to speed things up by asking |
| 02:31.39 | KalD|Work | ick - how do I make mpg123 play louder? -g <?> what is a good gain level |
| 02:31.39 | tzanger | shmaltz: ok. You've already got your answer from those online at this time... |
| 02:32.15 | ManxPower | Wow! Amazon didn't have it when I checked a couple of months ago! |
| 02:32.16 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 02:32.18 | MikeJ[Jayden] | bkw_ you provide free support don't you?? :) |
| 02:32.27 | shmaltz | thanks |
| 02:32.30 | MikeJ[Jayden] | how much/ |
| 02:32.31 | MikeJ[Jayden] | ? |
| 02:32.51 | MikeJ[Jayden] | yum,installing 1/278... nap time |
| 02:34.56 | *** join/#asterisk jsolares (~jsolares@200.12.33.64) |
| 02:37.07 | Trepalium | I've always just used mp3gain to change the volume of my MOH stuff. When using it, I use -m 7, and the volume seems alright over SIP. Dpm |
| 02:37.16 | Trepalium | Don't yet have PSTN hardware to try that one. |
| 02:37.52 | *** join/#asterisk tessier_ (~treed@222.253.65.164) |
| 02:42.01 | cbachman | hmmmm... latest Microsft TechNet newsletter mentions speech activated password resets for helpdesks... |
| 02:42.19 | footnote | cbachman: it's good to ignore those |
| 02:43.13 | jsolares | hehe |
| 02:43.23 | cbachman | footnote... we rarely have to do password resets for our users. they state 30% of calls are password related. yeesh! |
| 02:43.38 | footnote | Win-ders rot they minds |
| 02:44.20 | tzafrir_home | zignig: actually Debian has mpg123, but in non-free. See also http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=292260 |
| 02:45.12 | zignig | tzafrir_home: I know , but i don't use non free on my machines. |
| 02:45.16 | zignig | :) |
| 02:45.22 | *** join/#asterisk jgaviria (~jgaviria@201.245.167.78) |
| 02:45.53 | tzafrir_home | zignig: read the report and look at http://mpg123.org/ |
| 02:45.59 | tzafrir_home | :-( |
| 02:47.29 | jgaviria | hi i have a big problem, mi * box have 100% CPU usage when i do a sip or iax call, i got this every 10 seconds, and the voice sound choppy, somebody could help me? |
| 02:47.46 | *** join/#asterisk Corydon76-home (beige@pcp08665860pcs.500ash01.tn.comcast.net) |
| 02:48.10 | tzanger | jgaviria: what hardware |
| 02:48.15 | *** join/#asterisk dalabera (~dalabera@adsl-9-164-104.mia.bellsouth.net) |
| 02:48.19 | *** join/#asterisk WGFreewill (~chatzilla@24-75-221-174.miamfl.adelphia.net) |
| 02:48.20 | ManxPower | jgaviria, That's been beten to death on the mailing list. |
| 02:48.26 | ManxPower | Seems to be redhat specific, IIRC |
| 02:48.37 | MikeJ[Jayden] | 100% cpu will make lousy calls for sure, what is pegging the cpu? |
| 02:48.52 | zignig | tzafrir_home: if they claim it is GPL why is it in non-free |
| 02:48.55 | zignig | ? |
| 02:48.59 | ManxPower | MikeJ[Jayden], That's been beten to death on the mailing list. |
| 02:49.11 | MikeJ[Jayden] | mailing list, what's that :) |
| 02:49.12 | jgaviria | tzanger: TDM04 in a DELL with Celeron 2.53Ghz 256k cache |
| 02:49.19 | MikeJ[Jayden] | jbot: Mailing List |
| 02:49.23 | ManxPower | ~mailinglist |
| 02:49.24 | jbot | well, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. |
| 02:49.27 | jsolares | ManxPower, grab a 4x4" and hit me repeteadly, the whole reason my zap wasnt working was beacuse i'm only using 2 of the 4 ports on the card, and i stubbornly believed 3-4 to be 1-2.... |
| 02:49.27 | jgaviria | tzanger: with just 1 call it happens |
| 02:49.38 | MikeJ[Jayden] | o.. that mailing list... :) |
| 02:49.53 | tzanger | jgaviria: is wctdm sharing interrupts with anything else? Look at /proc/interrupts |
| 02:50.28 | shmaltz | what does thsi mean (i'm getting it when compiling mpg123): |
| 02:50.30 | shmaltz | undefined reference to xfermem |
| 02:50.51 | tzanger | shmaltz: it means it wants to call a function xfermem() but doesn't know anything about it |
| 02:51.01 | shmaltz | so?????????????????/// |
| 02:51.14 | jgaviria | tzanger: upsss... yes 18: 1853587404 IO-APIC-level usb-uhci, wctdm |
| 02:51.31 | tzanger | jgaviria: try shuffing it to another PCI slot if you can |
| 02:51.33 | jgaviria | tzanger: this couldbe the problem right? |
| 02:51.48 | tzanger | shmaltz: you need some basic C programming experience here |
| 02:52.01 | MikeJ[Jayden] | tzanger, I think he was looking for more from that answer.... I know because of how many ??? he used :) |
| 02:52.06 | shmaltz | and that I can take from you I suppose |
| 02:52.08 | tzanger | jgaviria: could be, I don't have enough info from that |
| 02:52.09 | shmaltz | tzanger |
| 02:52.26 | jgaviria | tzanger: maybe i can disabling usb in the bios |
| 02:52.30 | tzanger | MikeJ[Jayden]: I've told him probably 5 times now that he's got more work ahead of him than I think he's ready for |
| 02:52.32 | shmaltz | Thanks MikeJ |
| 02:52.39 | MikeJ[Jayden] | :) |
| 02:52.54 | tzanger | jgaviria: you can try but that doesn't physically disconnect the device |
| 02:52.58 | shmaltz | tzanger, never underestimate anybodys' will |
| 02:53.08 | MikeJ[Jayden] | why not use native MOH? |
| 02:53.15 | *** join/#asterisk atmel (~vlad@ip68-4-101-199.oc.oc.cox.net) |
| 02:53.18 | tzanger | shmaltz: believe me, I don't -- I am a very tenatious person |
| 02:53.36 | tzanger | shmaltz: but I am not prepared to educate someone on basic C programming to solve a problem I will never encounter myself |
| 02:53.59 | shmaltz | never? how do you know? tzanger? |
| 02:54.01 | WGFreewill | shmaltz: why are you compiling mpg123? |
| 02:54.15 | MikeJ[Jayden] | you know what, I like make config... I just had to say |
| 02:54.17 | shmaltz | b/c I want to play moh WGFreewill |
| 02:54.28 | Trepalium | Might be easier to install an mpg123 package if you aren't strong with C programming. |
| 02:54.29 | WGFreewill | doesnt your distro have mpg23 |
| 02:54.30 | WGFreewill | 132 |
| 02:54.34 | MikeJ[Jayden] | wtf is WGFreewill |
| 02:54.49 | shmaltz | but the one that comes with gentoo doesn't work with asterisk |
| 02:54.50 | Trepalium | Most distros include mpg321 instead of mpg123. |
| 02:54.51 | WGFreewill | something like MikeJJayden |
| 02:55.00 | tzanger | shmaltz: a) I don't do MOH. b) I'm not using 64bit x86. c) I have over a decade of C programming experience and would be able to solve the problem if I spent a bit of time on it if a) or b) were to vanish |
| 02:55.00 | WGFreewill | time for a real distro |
| 02:55.11 | MikeJ[Jayden] | o... I get it |
| 02:55.27 | shmaltz | yikes |
| 02:55.29 | tzanger | shmaltz: I'm not trying to be mean or an ass but I've told you many times now what you need but you instead decide to continue asking the same questions and ignoring what I say |
| 02:55.39 | WGFreewill | smaltz what distro do you run |
| 02:55.42 | Trepalium | You don't program, and you're using Gentoo... Ouch. |
| 02:56.05 | MikeJ[Jayden] | ok, so back to, why not native MOH instead of mpg123? |
| 02:56.08 | shmaltz | tzanger, you did not. all you said was that I should get a 32 bit install, and not try to play ahaed of myself |
| 02:56.15 | tzanger | shmaltz: no |
| 02:56.27 | shmaltz | MikeJ what do you mean by native? |
| 02:56.28 | tzanger | shmaltz: I said if you are unable or unwilling to solve the program AS A DEVELOPER then it is time ot step back or put up a bounty |
| 02:56.31 | tzanger | shmaltz: THAT is what I said |
| 02:56.33 | *** join/#asterisk Ayano (~erik_leee@adsl-66-51-208-150.dslextreme.com) |
| 02:56.37 | tzanger | shmaltz: and I stand behind my statement 100% |
| 02:56.50 | shmaltz | tzanger, you r right, but that was stone ages ago |
| 02:56.50 | tzanger | as I am sure any of the other developers hanging out here watching this |
| 02:57.01 | shmaltz | :( |
| 02:57.02 | jsolares | http://chema.solaresmaldonado.com/files/code/app_waitforsound.c <- if anyone has any use for it, it's kind of like the inverse of waitforsilence |
| 02:57.10 | tzanger | shmaltz: they're perhaps more tactful than I am and have been smart enough to keep quiet :-) |
| 02:57.20 | shmaltz | :( |
| 02:57.30 | WGFreewill | apt-get install mpg123 |
| 02:57.32 | tzanger | shmaltz: what has changed in the last 90 minutes to make my statement invalid? |
| 02:57.53 | shmaltz | nothing tzanger, but my memmory |
| 02:57.54 | MikeJ[Jayden] | read: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20musiconhold.conf |
| 02:57.57 | Ayano | I have a 7940 that keeps getting sip registration failed. I remember I had to have a patch to fix it but I can't find the patch now. Can someone help? |
| 02:58.15 | WGFreewill | 7940 works out of the box for me |
| 02:58.18 | WGFreewill | with 1-0 and head |
| 02:58.28 | MikeJ[Jayden] | my 7060 is fine w/ head |
| 02:58.40 | WGFreewill | damn 7970 |
| 02:58.48 | WGFreewill | where is my SIP! |
| 02:58.53 | phaded | hiding |
| 02:59.02 | MikeJ[Jayden] | and stable 1.0.5 now that you mention it, I was on one of those yesterday too |
| 02:59.15 | tzanger | hahaha |
| 02:59.17 | Zaw | what's the cheapest, most reliable phone to use with asterisk? |
| 02:59.21 | tzanger | "there's a reason there are no time-shares in Somalia" |
| 02:59.32 | tzanger | Zaw: does not compute |
| 02:59.36 | Ayano | budgetone 101 |
| 02:59.38 | tzanger | Zaw: restate your question |
| 02:59.57 | MikeJ[Jayden] | I also really like PHPConfig btw... |
| 02:59.59 | phaded | Zaw: analog phones are almost free |
| 03:00.22 | MikeJ[Jayden] | cuz I recently remembered how much I dislike vi. |
| 03:00.31 | phaded | yay for pico |
| 03:00.31 | Zaw | tzafrir: normally in technology it is 'cheap, reliable, fast' pick any two. |
| 03:00.38 | WGFreewill | sipura 841 |
| 03:00.40 | Zaw | er tzanger |
| 03:00.41 | WGFreewill | nice phone |
| 03:00.53 | Trepalium | Analog phones are all three, but connecting them to * may not be cheap. |
| 03:00.54 | tzanger | Zaw: exactly |
| 03:00.54 | Zaw | phaded: so get something like the linksys PAP2 and an analog phone? |
| 03:01.01 | tzanger | cheap and most reliable don't compute |
| 03:01.15 | WGFreewill | the 841 is nice |
| 03:01.18 | WGFreewill | and cheap |
| 03:01.22 | Trepalium | On the other hand, analog phones have the worst features out of all phones. |
| 03:01.24 | WGFreewill | no 2 ethernet ports |
| 03:01.27 | Trepalium | e.g. none. |
| 03:01.33 | tzanger | Trepalium: depends on what you need for features |
| 03:01.37 | MikeJ[Jayden] | you should be able to connect an analog for less than $40 per port, no matter how few you have |
| 03:01.37 | Zaw | tzanger: 'cheap, reliable, feature rich' pick any two is along the lines of what i was thinking. with 'cheap and reliable' being my picks :) |
| 03:01.53 | WGFreewill | networkhardware.com had some 7960 non G for around 200 USD |
| 03:02.01 | tzanger | my ADSI phone has a display, soft buttons, assisted call transfer, call waiting, it's all there |
| 03:02.14 | MikeJ[Jayden] | what phone? |
| 03:02.19 | Trepalium | tzanger: True enough. You usually only get basic features on analog, though |
| 03:02.20 | tzanger | Zaw: TDMx0P with analog phones then |
| 03:02.24 | Ayano | budgetone 101s are like 100 |
| 03:02.27 | tzanger | x being the # of FXS ports you need, up to 4 |
| 03:02.31 | Zaw | MikeJ[Jayden]: where do you get the VOIP adapters for analog phones for under $35 at? |
| 03:02.33 | loko | Zaw, thats how it is for me as well, but in the sense of, "loko, early, attentive, pick any two" |
| 03:02.41 | tzanger | Ayano: bt101s aren't necessarily reliable |
| 03:02.42 | dalabera | could anyone give me suggestion of a piece of a script that I wrote on perl -> http://www.pastebin.com/248636 |
| 03:02.43 | Zaw | loko: hehe |
| 03:02.51 | Zaw | tzanger: ok, thanks |
| 03:02.52 | Ahewes | newbie hosting question: I'm tasked with hosting a large application, where can I find the lowest cost inbound 800 for 100,000 minutes |
| 03:03.01 | MikeJ[Jayden] | spa2000, you can get them for about $70 US if you look around (2 ports) |
| 03:03.09 | footnote | Hrm. It's possible that Asterisk may hurt the sales of Glenayre and Comverse. |
| 03:03.15 | Ayano | really, I've never had a problem with them tz. |
| 03:03.17 | footnote | (we can all hope) |
| 03:03.17 | WGFreewill | 2c on broadvoice |
| 03:03.37 | Ahewes | Anyone here selling inbound 800 is more than welcome to msg me |
| 03:03.41 | MikeJ[Jayden] | ahewes, what do you mean hosting? |
| 03:03.57 | *** part/#asterisk redder86 (~lee@gateway.howardsilvan.com) |
| 03:04.02 | Ahewes | I'm hosting an IVR application for a customer |
| 03:04.34 | MikeJ[Jayden] | so you are just looking for 800 inbound over IAX or sip? |
| 03:04.54 | Ahewes | MikeJ[Jayden] sip, preferrably |
| 03:05.16 | Goshen | Ahewes: nufone, iax.cc |
| 03:05.27 | footnote | <PROTECTED> |
| 03:05.30 | footnote | hehe |
| 03:05.35 | MikeJ[Jayden] | check the wiki, there is quite a list, with 100,000 minutes, I would make some phone calls to discuss rates, |
| 03:05.36 | Ahewes | I'll be using asterisk for now, but I may have to switch to a VoiceXML platform at some point. |
| 03:05.40 | tzanger | footnote: hahaha |
| 03:05.42 | bjohnson | Ahewes: livevoip |
| 03:05.51 | footnote | Ahewes: VXML < * |
| 03:05.55 | Ahewes | MikeJ[Jayden] thanks! |
| 03:05.58 | dsmouse | wow. zaptel |
| 03:06.06 | Ahewes | thanks all for the info. |
| 03:06.22 | dsmouse | wow. zaptel/TDM400P really doesn't like fc3. FC2 is great tho |
| 03:06.24 | dalabera | could anyone give me suggestion of a piece of a script that I wrote on perl -> http://www.pastebin.com/248636 |
| 03:06.45 | loko | Ahewes, i personally say clear of iax.cc but thats just me |
| 03:06.47 | tzanger | dalabera: dpm |
| 03:06.51 | footnote | dalabera: nobody agrees about perl style :) |
| 03:06.56 | *** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc) |
| 03:06.58 | tzanger | dalabera: don't spam here. put it in a message to -users and let us bash you openly there :-) |
| 03:07.19 | tzanger | wow |
| 03:07.30 | tzanger | just got finished watching the true story of black hawk down... WOW |
| 03:07.32 | bjohnson | MikeJ[Jayden]: eezeephone has pap2-na for $60 |
| 03:07.47 | jsolares | dalabera: that's php not perl, they might be very similar, but they are not the same thing :) |
| 03:07.48 | bjohnson | loko: why? |
| 03:08.14 | MikeJ[Jayden] | pap2 has issues as to who they sell those too.. no end users |
| 03:08.18 | MikeJ[Jayden] | I beleive |
| 03:08.19 | loko | bjohnson, I just found their service to be shitty, my DIDs were down weekly, and customer support takes days to reply to emails |
| 03:08.59 | bjohnson | MikeJ[Jayden]: I don't know the ins and outs of the licensing .. they sell the -na for $60 including one month of service |
| 03:09.10 | bjohnson | I don't know what the service is |
| 03:09.18 | footnote | Still too many crappy companies around. |
| 03:09.32 | footnote | Like HP, etc |
| 03:10.05 | MikeJ[Jayden] | ~troutslap tzanger |
| 03:10.09 | jbot | ACTION slaps tzanger around with a large trout |
| 03:10.09 | MikeJ[Jayden] | :) |
| 03:10.10 | loko | bjohnson, is eezeephones pap2-na locked at all? |
| 03:10.12 | *** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au) |
| 03:10.20 | tzanger | hahaha |
| 03:10.20 | zignig | footnote: HP are cool, Sir billy and his merry men are the danger...... |
| 03:10.29 | WGFreewill | who cant waut for some AOL voip? |
| 03:10.33 | footnote | zignig: Carly ruined HP! |
| 03:10.37 | tzanger | some school academy caught a bunch of kids in an orgy |
| 03:10.47 | footnote | zignig: she was hot for billy |
| 03:11.18 | zignig | footnote: aren't we all <NOT> |
| 03:11.41 | zignig | footnote: carley has gone now , so we will see. |
| 03:11.49 | footnote | nice parachute |
| 03:12.02 | footnote | wreck a company, get rich. |
| 03:12.07 | *** join/#asterisk Rick_Hunter (~rhunter@02-119.008.popsite.net) |
| 03:12.25 | WGFreewill | yep shes out to destroy the entire free world |
| 03:12.32 | footnote | mean ole bat |
| 03:13.31 | footnote | I'm just mad because HP stopped using X on their logic analyzers |
| 03:13.38 | footnote | #$^%$# |
| 03:16.09 | footnote | Hrm. R2 is supported by Asterisk. Flexible? |
| 03:16.51 | zignig | footnote: don't forget to read yourself ( much usefull information in that little bit at the bottom ) :-) |
| 03:17.04 | footnote | gotit! |
| 03:17.20 | footnote | footnote* |
| 03:17.42 | footnote | zignig |
| 03:17.53 | footnote | It said "Ignore all footnotes". |
| 03:17.59 | footnote | :-/ |
| 03:18.42 | shmaltz | MikeJ the native format doesn't work for me |
| 03:19.02 | Trepalium | Isn't HP just Compaq in disguise these days? |
| 03:19.08 | *** join/#asterisk SirK (~SirK@dsl-202-173-146-192.qld.westnet.com.au) |
| 03:19.18 | zignig | Trepalium: other way round |
| 03:20.37 | Trepalium | I mean, they dropped the HP desktops, and servers for the Compaq models (just with an HP logo instead of Compaq). Their printers are getting more and more cheaply built. Their calculators are pretty much dead now, etc. |
| 03:22.57 | MikeJ[Jayden] | do sipura ata's support callwaiting? |
| 03:23.19 | Trepalium | Frankly, I think the HP Netservers were far better built than the Compaq Proliant servers ever were. |
| 03:24.53 | *** join/#asterisk mikeirc (~mikeirc@6532112hfc30.tampabay.res.rr.com) |
| 03:25.14 | *** join/#asterisk riksta (~rick@host217-42-22-145.range217-42.btcentralplus.com) |
| 03:25.36 | *** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-138.modem.logical.net) |
| 03:29.35 | SirK | hrm. ztdummy is panicing on my 2.6.10-xenU kernel. anyone want to take a stab at why ? zaptel loads fine (although i dont have any hardware) but ztdummy dies in the arse with http://203.149.72.134/crash.txt as the result. |
| 03:30.47 | Trepalium | xen? |
| 03:31.02 | *** join/#asterisk pimpwell (~pimpwell@ool-44c6ab45.dyn.optonline.net) |
| 03:31.13 | tzanger | xen rocks |
| 03:31.24 | tzanger | SirK: do you have the rtc exposed to the xenu? |
| 03:31.53 | porkchop | My * server is a xen slice. |
| 03:32.09 | SirK | well. unprivledged kernels dont have the option to compile in rtc... |
| 03:32.11 | porkchop | got too many machines. didnt need another. |
| 03:32.23 | pimpwell | not that I want to, but I could change the caller ID message everytime I place a call with a simple change of a variable? |
| 03:32.31 | SirK | porkchop: does it work well ? i'm noticing 'issues' when making more than one call via an IAX trunk. |
| 03:32.34 | pimpwell | that doesn't sound right but |
| 03:32.42 | SirK | which i figured wasthe lack of a timer ? |
| 03:32.46 | tzanger | SirK: so what makes you think that ztdummy will work then if it can't access the rtc?? |
| 03:32.56 | shmaltz | tzanger, I was able to compile it using make linux-devel, but I'm still having problems that the sound is choppy |
| 03:33.05 | shmaltz | I don't know why |
| 03:33.07 | porkchop | SirK: I have not noticed any problems, but I only have 3 users. |
| 03:33.26 | pimpwell | what kind of line are you guys working with |
| 03:33.28 | BrianR___ | Is there anti-tromboning support in IAX? |
| 03:33.30 | pimpwell | T1? |
| 03:33.34 | tzanger | shmaltz: my next suggestion would be to use a stock kernel instead of hte patched up "we know better than linus" gentoo kernel |
| 03:33.54 | SirK | porkchop: sip clients -> iax to a provider who terminates ontothe pstn. |
| 03:33.55 | shmaltz | tzanger, you got it, moving to slackware |
| 03:34.09 | tzanger | shmaltz: and after that I'd suggest packet captures to see if the packets are being sent out badly |
| 03:34.17 | MikeJ[Jayden] | I can load an * box over and over again on FC2 no problem... |
| 03:34.25 | porkchop | SirK: same thing here. Might be talking about the same provider even. |
| 03:34.28 | BrianR___ | tzanger: Oh. I got a script set up to automagically set MWI's on Norstar sets when the user gets a voicemail on asterisk. |
| 03:34.34 | tzanger | shmaltz: I didn't say that, slackware requires you to have some decent programming experience too to fix niggly little things |
| 03:34.53 | shmaltz | tzanger, I'm not scared |
| 03:35.11 | Trepalium | Gentoo generally does, too. |
| 03:35.13 | tzanger | BrianR___: send that shit my way :-) |
| 03:35.17 | tzanger | shmaltz: I didn't say you were scared |
| 03:35.33 | shmaltz | Trepalium, and I got gentoo to work fine |
| 03:35.33 | tzanger | ... naive perhaps, but not scared. I am actually very happy to see that you're not afraid to dive in |
| 03:35.41 | shmaltz | :) |
| 03:35.51 | SirK | porkchop: entirely possible :) |
| 03:35.59 | SirK | whenever i make more than one call i get Mar 3 03:35:19 WARNING[999]: channel.c:523 ast_channel_walk_locked: Avoided deadlock for 'SIP/5000-4c44', 10 retries! |
| 03:36.06 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 03:36.16 | porkchop | I dont get those messages. |
| 03:36.18 | MikeJ[Jayden] | MWI on the norstar sets.. tell me more, I am about to replace a norstar w/ * |
| 03:36.19 | SirK | it goes choppy as then dies. |
| 03:36.46 | mikeirc | What is this "Config Edit" I see talked about in some of the documents on Asterisk? |
| 03:37.01 | MikeJ[Jayden] | mike: huh? |
| 03:37.29 | shmaltz | tzanger, I do have some programming experience from the past. But not C |
| 03:37.51 | tzanger | shmaltz: ok |
| 03:37.52 | shmaltz | also I havn't done any programming in a while |
| 03:38.03 | MikeJ[Jayden] | shmaltz: what languages? |
| 03:38.11 | *** join/#asterisk Ron-Na (~ronald@203.70.36.126) |
| 03:38.13 | mikeirc | Yeah... I don't know... I've seen this alot: " First use the Config Edit to edit extensions.conf" ...? Well I was doing it from the console...but what are they talking about? |
| 03:38.24 | Ron-Na | Hi, ... |
| 03:38.26 | shmaltz | VB and assembly |
| 03:38.43 | Ron-Na | What is wrong with that statement: Dial,SIP/${NUFONEUSER}@NuFone/${EXTEN:${TRUNKMSD}} |
| 03:38.48 | Grooby | *yawn* |
| 03:38.55 | BrianR___ | tzanger: on its way. |
| 03:38.59 | MikeJ[Jayden] | mike: you can edit with whatever suits your fancy... |
| 03:39.06 | tzanger | Ron-Na: well it's SIP with nufone for starters... :-) |
| 03:39.07 | tzanger | BrianR___: danke |
| 03:39.08 | MikeJ[Jayden] | BrianR: Tell me more.... |
| 03:39.10 | *** join/#asterisk joaovianna (naturalvoi@node-40247a6a.ewr.onnet.us.uu.net) |
| 03:39.28 | Ron-Na | tzanger, it should be, but it does not work!!! |
| 03:39.34 | mikeirc | MikeJ: yah.. I was just curious what they were refering to though.. Thanks though. |
| 03:40.01 | BrianR___ | MikeJ[Jayden]: Essentially it lets you put a file with a list of norstar extensions in each user's voicemail directory. When the box gets a message, it uses a fxo port connected to an ATA on the norstar to go around and enable the MWI lights. |
| 03:40.04 | MikeJ[Jayden] | Ron: @NuFone is prolly not a valid host |
| 03:40.23 | Ron-Na | this works: Dial,IAX2/${NUFONEUSER}@NuFone/${EXTEN:${TRUNKMSD}} (another account with nufone) |
| 03:40.24 | Goshen | Ron-Na: use IAX not sip |
| 03:40.35 | MikeJ[Jayden] | ooooo... me to... please brian |
| 03:40.38 | Goshen | I thought it was Dial(SIP |
| 03:40.52 | BrianR___ | MikeJ[Jayden]: WHats your email address? |
| 03:41.40 | MikeJ[Jayden] | PM |
| 03:41.48 | Goshen | Ron-Na: do you have your nufone info in your sip.conf? |
| 03:42.03 | Ron-Na | yes |
| 03:42.28 | BrianR___ | forgot to mention, you should probably do FEATURE-4 (Flash *4) on the port the ATA is connected to and forward it to an extension on asterisk that runs VoiceMailMain. |
| 03:42.52 | tzanger | shmaltz: huh?? |
| 03:43.02 | shmaltz | I'm thank you for your help |
| 03:43.12 | tzanger | I didn't help much, you still haven't got a working system |
| 03:43.18 | *** join/#asterisk Trionnis (buffy@12-203-113-15.client.insightBB.com) |
| 03:43.24 | shmaltz | it's working just MOH is choppy |
| 03:43.27 | Trionnis | any sipura gurus around? :) |
| 03:43.32 | BrianR___ | I'll try to comment that script tomorrow and post it on the wiki under a free license... |
| 03:43.42 | shmaltz | Trionnis, what sipura device? |
| 03:43.44 | mikeirc | Anyone hear familiar with the Asterisk@Home distro? I was thinking of wiping my current setup and trying it out. |
| 03:43.49 | BrianR___ | Oh.. And fix the remaining hardcoded things in it. |
| 03:43.50 | Trionnis | spa-2k |
| 03:43.58 | MikeJ[Jayden] | mikeirc: yes |
| 03:44.01 | Trionnis | need a bit o' help with the dialplan stuff |
| 03:44.05 | Trionnis | not on * |
| 03:44.15 | BrianR___ | tzanger: Currently it's hardcoded to use the 4th zap interface |
| 03:44.22 | shmaltz | I have the spa3k but its the same with the difference on interface so go ahead |
| 03:44.30 | mikeirc | MikeJ: Any good? |
| 03:44.31 | MikeJ[Jayden] | Trionnis: huh? |
| 03:44.45 | Trionnis | well, I don't use any really funky prefix dialing |
| 03:45.01 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 03:45.10 | Trionnis | so I would just use (reinvite=no |
| 03:45.10 | Trionnis | canreinvite=no |
| 03:45.12 | Trionnis | gah |
| 03:45.13 | Trionnis | crap |
| 03:45.15 | Trionnis | sorry |
| 03:45.30 | Trionnis | (xxxxxxxxxxxx.) <- that? |
| 03:45.32 | shmaltz | what are you trying to do Trionnis? |
| 03:45.44 | mikeirc | MikeJ: Any reason why I shouldn't go for it...like bugs, etc? |
| 03:45.46 | MikeJ[Jayden] | mike: ummm, yes and no, quick to get up and running, if you spend a little time with it, it is really easier to do some stuff jsut in the dialplan than with AMP, but it is nice enough to set a bunch of stuff up for you... I would suggest it for a first play around box, but you will rebuild it by hand soon after that |
| 03:45.54 | Trionnis | map port 1 to a pots phone in the same fashion as my xlite phone |
| 03:45.55 | MikeJ[Jayden] | it's stable |
| 03:46.17 | Trionnis | I don't need the 2k to do any actual routing |
| 03:46.22 | Trionnis | just pass the digits |
| 03:46.29 | *** join/#asterisk Inv_arp (junya@adsl-3-247-135.mia.bellsouth.net) |
| 03:47.02 | Goshen | Trionnis: have you been drinking? |
| 03:47.08 | Trionnis | laf |
| 03:47.10 | Trionnis | not lately |
| 03:47.13 | MikeJ[Jayden] | Trionnis: no more clear than the first time I asked... wtf are you talking about |
| 03:47.16 | Trionnis | just tired as crap |
| 03:47.17 | Trionnis | ok |
| 03:47.18 | MikeJ[Jayden] | drugs? |
| 03:47.24 | Goshen | Dude, go to bed |
| 03:47.31 | mikeirc | MikeJ(Jayden): Thanks for the info |
| 03:47.34 | Trionnis | in the config for line1 |
| 03:47.35 | MikeJ[Jayden] | np |
| 03:47.37 | Trionnis | at the bottom |
| 03:47.45 | Trionnis | there's a section for dialplan |
| 03:47.48 | kram | wow |
| 03:47.54 | Goshen | line1 of what? you are totally out of it |
| 03:47.58 | MikeJ[Jayden] | tri:ok, so you are talking about the sipura dialplan |
| 03:48.00 | Trionnis | ............ |
| 03:48.01 | MikeJ[Jayden] | got ya |
| 03:48.03 | Trionnis | finally |
| 03:48.06 | Trionnis | :D |
| 03:48.08 | MikeJ[Jayden] | Kram: wow what? |
| 03:48.25 | kram | i figured out that on x86 systems asterisk could only support ~300 threads |
| 03:48.26 | Trionnis | yes, the dialplan in the spa-2000 |
| 03:48.56 | MikeJ[Jayden] | Tri, depends on what you are possibly going to hit, be specific so you don't have to wait for digit timeout on the spa |
| 03:49.08 | kram | and now i'm fixing it :) |
| 03:49.11 | Trionnis | the thing on voxilla that I'm reading says to use a bunch of prefixes |
| 03:49.11 | MikeJ[Jayden] | kram: but did you figure out how to fix it :( |
| 03:49.13 | MikeJ[Jayden] | :) |
| 03:49.16 | Trionnis | and I don't need to do that |
| 03:49.19 | Trionnis | so |
| 03:49.20 | Trionnis | ... |
| 03:49.40 | MikeJ[Jayden] | what are you going to dial? possibly? |
| 03:49.55 | MikeJ[Jayden] | for the initial call, not digits after the call is connected |
| 03:49.56 | Trionnis | 11 digit, and possibly international |
| 03:50.13 | MikeJ[Jayden] | what about ext? |
| 03:50.17 | Trionnis | no |
| 03:50.26 | Trionnis | none other than voicemail |
| 03:50.48 | MikeJ[Jayden] | I can't remember if that uses n's and x's or just x's, anyone |
| 03:50.50 | MikeJ[Jayden] | ?? |
| 03:51.01 | Trionnis | the manual just shows x's |
| 03:51.34 | Trionnis | oh hell |
| 03:51.37 | Trionnis | ignore me |
| 03:51.40 | Trionnis | I see it now |
| 03:51.49 | Trionnis | wasn't quite clicking how it went together |
| 03:51.57 | Trionnis | I'll just go sit in the corner now |
| 03:51.58 | MikeJ[Jayden] | so do some simple *123 or somthing for vm, and then your dialplan would be *123|1xxxxxxxxxx |
| 03:52.10 | Trionnis | ya |
| 03:52.16 | MikeJ[Jayden] | ~troutslap trionnis |
| 03:52.19 | jbot | ACTION slaps trionnis around with a large trout |
| 03:52.19 | Trionnis | that'll work |
| 03:52.19 | Trionnis | .. |
| 03:52.24 | Trionnis | ugh |
| 03:52.33 | MikeJ[Jayden] | kram: done yet |
| 03:52.36 | Trionnis | deserved... |
| 03:52.39 | Trionnis | ;) |
| 03:52.39 | MikeJ[Jayden] | :) |
| 03:53.01 | MikeJ[Jayden] | ok.. night night time for me.... |
| 03:57.31 | Ron-Na | Is anybody awake who can help me with ASTCC ? |
| 03:57.46 | mikeirc | Is it just me or is there ALOT you CANT do from AMP??? |
| 04:01.09 | Goshen | mike: I tried to install AMP, but it wanted to take over my machine, which I can't allow I need it for other applications, so I scrapped it, and learned how to work config files |
| 04:02.23 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 04:02.40 | *** join/#asterisk tuxinator_linux (~tuxinator@ip68-109-146-168.ph.ph.cox.net) |
| 04:07.06 | Goshen | 2.6.11 kernel is out :) |
| 04:10.30 | tuxinator_linux | whats new with 2.6.11? |
| 04:10.39 | *** join/#asterisk beto75 (~ha@201.133.240.35) |
| 04:10.45 | beto75 | hello guys |
| 04:10.53 | pimpwell | is it true caller ID will be anything I specify? |
| 04:11.11 | pimpwell | so if I have a user make a call, I can have their last name on it |
| 04:11.27 | pimpwell | or will it relfect my box |
| 04:11.37 | pimpwell | and always have my company name on t |
| 04:11.40 | beto75 | guys ins extensions.conf I need to call a extension which is a public sip user user@sipdomain.com , how do I call it? |
| 04:12.00 | Sedorox | exten => user,1,blah |
| 04:12.03 | Sedorox | and setup dns srv |
| 04:12.05 | Sedorox | I believe |
| 04:12.21 | Nugget | beto75: Dial(SIP/user@sipdomain.com) |
| 04:12.34 | Sedorox | ops.. read wrong.. soryr |
| 04:12.39 | beto75 | thanks Nugget sounds like it could worl let me try |
| 04:13.38 | tuxinator_linux | pimpwell: I think that is only true on PRI lines. |
| 04:14.32 | pimpwell | hmmm |
| 04:14.51 | pimpwell | the caller ID part is real important |
| 04:14.58 | pimpwell | hmm |
| 04:15.19 | Nugget | many voip providers will let you set whatever callerid you want, but only the number will actually make it out to the "real" phone network. the receiver's end will do a lookup on the number for the text part of the id. |
| 04:15.36 | Nugget | voicepulse connect lets you set whatever callerid you want, for instance. |
| 04:16.02 | pimpwell | but it wont work half the time is what you're saying |
| 04:16.12 | pimpwell | when the call makes it to someone's kitchen via copper line |
| 04:16.14 | Nugget | no, I'm saying that only half of it will work. |
| 04:16.21 | Nugget | not that it only works half the time. |
| 04:16.30 | pimpwell | the text part is what I mean |
| 04:16.35 | pimpwell | sorry for not being clear |
| 04:16.36 | Nugget | that part will not work. |
| 04:16.54 | pimpwell | crap |
| 04:18.59 | pimpwell | know any providers that will let me drop .call files in their outbound directory for a per charge cost? |
| 04:19.13 | pimpwell | I need a backup plan for what I'm doinf |
| 04:19.15 | Nugget | no clue, sorry. |
| 04:19.18 | pimpwell | doing* k thx |
| 04:21.32 | Grooby | anyone here uses speex? |
| 04:21.47 | *** join/#asterisk carbon60 (~adam@CPE001217cdb3c5-CM000f9fa6ba66.cpe.net.cable.rogers.com) |
| 04:21.48 | Grooby | more specifically speex with IAX |
| 04:21.56 | carbon60 | Is RealTime support in 1.0.5? |
| 04:22.30 | Goshen | Nugget: does a lookup for the text part where? |
| 04:22.42 | Nugget | hell if I know. I just know that's how it works. :) |
| 04:22.50 | Juggie | no |
| 04:23.14 | Nugget | I just know that when I spoof a number, the text that shows is whatever the number would normally show. |
| 04:23.15 | file | uses SS7, database lookup on phone number for the name |
| 04:23.46 | carbon60 | Juggie: That for me? |
| 04:24.00 | file | realtime support is only available in cvs head |
| 04:24.02 | file | NEXT!!! |
| 04:24.04 | Juggie | yes. |
| 04:24.07 | carbon60 | Ah, thanks. |
| 04:25.07 | tuxinator_linux | snuggle tackle? |
| 04:25.12 | sivana | heh |
| 04:25.15 | file | yessss |
| 04:25.44 | tuxinator_linux | hey file, is your cousin named folder? |
| 04:25.48 | mishehu | bah. |
| 04:25.59 | file | folder? pfft, you Windows user |
| 04:26.02 | file | directory :p |
| 04:26.04 | tuxinator_linux | cheezy humor or the night |
| 04:26.05 | Inv_arp | heg |
| 04:26.07 | Inv_arp | heh |
| 04:26.35 | tuxinator_linux | true, they are directories |
| 04:26.37 | Nugget | linux is poo. |
| 04:26.45 | tuxinator_linux | poo? |
| 04:26.49 | carbon60 | I see that app_voicemail.c has USEPOSTGRESVM and what looks like some pgsql support though. |
| 04:26.51 | tuxinator_linux | ~poo |
| 04:26.52 | jbot | poo is smelly |
| 04:26.55 | carbon60 | Anyone know of doco for that? |
| 04:28.02 | Inv_arp | hey whats the code for PRI_CAUSE for discconected # |
| 04:28.46 | bjohnson | for those with a WRT54G .. can they be made to act as a wifi bridge? |
| 04:29.12 | bjohnson | ie. access point + wifi client at same time |
| 04:30.56 | Grooby | in a WDS mode? |
| 04:33.20 | *** join/#asterisk kbot (~mejii@hosted-pat.7ticks.com) |
| 04:35.03 | kbot | good evening... Has anybody encountered a problem with the cisco 7960 phones where it gets stuck in a loop where it constantly reboots itself? |
| 04:35.30 | Nugget | usually that means your tftp server is supplying inconsistent firmware instructions |
| 04:36.07 | Nugget | pay particular attnetion to what's in OS79XX.TXT |
| 04:36.18 | Juggie | i hate those damn phones |
| 04:36.22 | Juggie | mitel sip phones are so much better |
| 04:36.28 | Nugget | make sure it agrees with image_version |
| 04:36.43 | kbot | I upgraded to firmware 6.2 from 4.4 when this started happening |
| 04:37.20 | Juggie | 7.3 is the latest |
| 04:37.37 | Juggie | if its still downloading files from your tftp you are in good shape |
| 04:37.44 | Juggie | you just have to configuire the files propetly |
| 04:37.46 | kbot | it keeps pulling the OS79XX.TXT file and then the P00-3-02-00.sbn over and over again |
| 04:38.03 | Nugget | what image_version is in SIPDefault? |
| 04:38.11 | Juggie | 3 02 sounds like an older version |
| 04:38.13 | Juggie | and not 6 2 |
| 04:38.18 | `Sauron | what up nugget |
| 04:38.23 | Nugget | moo |
| 04:38.31 | `Sauron | meow |
| 04:38.48 | kbot | image_version: "P0S3-06-2-00" |
| 04:39.07 | `Sauron | kbot: there's a doc on puck.nether.net that explains a LOT about how the cisco phones work |
| 04:39.09 | `Sauron | basically |
| 04:39.15 | Juggie | whats in OS79XX.txy |
| 04:39.17 | Nugget | I think there are some issues with doing non-sequential upgrades on those old cisco versions. |
| 04:39.17 | Juggie | er, .txt |
| 04:39.24 | Nugget | but I dunno. I started out with 7.x |
| 04:39.24 | `Sauron | they'll try to download the file for sccp, then the file for mgcp, then the file for sip |
| 04:39.31 | `Sauron | and whichever one it finds, it boots as |
| 04:40.01 | kbot | I am using SIP, figures it's the last one it looks for |
| 04:40.01 | *** join/#asterisk marc32344 (~marc32344@69-28-224-214.dsl.teksavvy.com) |
| 04:40.13 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 04:40.55 | ManxPower | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=5756064723&ssPageName=ADME:B:SS:US:1 SoundPoint IP 500 $190. |
| 04:41.33 | Juggie | kbot, is P0S3-06-2-00 in your os79xx.txt? |
| 04:41.39 | kbot | yeppers |
| 04:41.41 | `Sauron | I dunno where the link was, though |
| 04:41.42 | Juggie | ok |
| 04:41.48 | Juggie | i'm pretty sure i know the problem :) |
| 04:41.56 | kbot | speak up |
| 04:41.59 | kbot | :) |
| 04:41.59 | Juggie | make a copy of all the files |
| 04:42.03 | Juggie | and rename them to 8.3 standard |
| 04:42.19 | `Sauron | https://puck.nether.net/pipermail/cisco-voip/2005-January/002738.html |
| 04:42.24 | `Sauron | go read that |
| 04:42.28 | joaovianna | provisioning |
| 04:42.29 | Juggie | so 8letters.3ext |
| 04:42.32 | joaovianna | gps |
| 04:42.33 | joaovianna | pga |
| 04:42.53 | Juggie | older firmware had a limit on file name lenght |
| 04:42.57 | Juggie | once u do that u should be good to do |
| 04:43.16 | Juggie | *go |
| 04:43.20 | Juggie | i would think anyways |
| 04:43.28 | Juggie | you may as well get the 7.3fw and upgrade to that |
| 04:43.32 | Juggie | assuming u have a cisco account |
| 04:43.36 | kbot | I'll give that at try and read the article that Sauron provided |
| 04:43.41 | kbot | yep, got a cisco account |
| 04:43.47 | Juggie | then download 7.3 |
| 04:43.59 | kbot | roger 10-4 |
| 04:52.20 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 04:52.57 | mikeirc | just wiped amp from my system... feels better already... |
| 04:54.15 | Juggie | amp is lame. |
| 04:54.21 | Juggie | you'd be much better of playing with head and using realtime. |
| 04:55.02 | Grooby | lol |
| 04:55.12 | Grooby | you can really read that sentence out of context ;) |
| 04:55.24 | Juggie | yeah |
| 04:55.27 | Juggie | i thought about that after |
| 04:55.33 | Juggie | but decided not to bring attention to it :P |
| 04:55.37 | Juggie | i have head skis too.... |
| 04:55.42 | Juggie | so :) |
| 04:57.22 | Grooby | hehe |
| 04:57.49 | Grooby | so has anyone here use IAX and start hearing high pitch noise after couple seconds? |
| 04:58.54 | shido6 | leave that crack alone |
| 04:59.43 | Grooby | huh? |
| 05:00.02 | *** part/#asterisk kb8ude (~me@c-24-2-127-157.client.comcast.net) |
| 05:00.59 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
| 05:06.31 | `Sauron | dum di dum |
| 05:06.34 | `Sauron | I love fighting with usb |
| 05:08.03 | *** join/#asterisk terrapen (~cjs@cpe-66-25-86-139.satx.res.rr.com) |
| 05:09.33 | Grooby | what's' usb? |
| 05:13.20 | *** join/#asterisk Bentley (~rbc@S01060080c8135e6a.cg.shawcable.net) |
| 05:15.14 | joaovianna | Anyone working with GAPSlite from Grandstream ? |
| 05:16.21 | *** join/#asterisk shepherd (matt@pcp01541028pcs.huntsv01.al.comcast.net) |
| 05:24.55 | *** join/#asterisk Koshatul (~evangelio@202.9.38.223) |
| 05:27.03 | *** join/#asterisk NormAst (HydraIRC@Ottawa-HSE-ppp4119108.sympatico.ca) |
| 05:27.37 | *** join/#asterisk mady (~root@61.11.24.250) |
| 05:28.27 | mady | hello kram |
| 05:28.40 | mishehu | gah, I have the build problem with codec_speex again. |
| 05:28.40 | mady | can i talk to u kram |
| 05:28.51 | mishehu | I can't remember the cc comman to link it... |
| 05:29.11 | mishehu | something like cc -fPIC -XLinker -o codec_speex.so codec_speex.o |
| 05:29.16 | mady | Hello kram r u there? |
| 05:29.22 | kram | going home |
| 05:29.42 | mady | ok bye. |
| 05:30.25 | *** join/#asterisk provolone (~root@70-32-14-201.frdrmd.adelphia.net) |
| 05:30.27 | *** join/#asterisk Nukemizer (~Nuke@66.237.85.58.ptr.us.xo.net) |
| 05:30.34 | mishehu | somebody remember what parameters I need to use again? |
| 05:31.15 | *** join/#asterisk Othello (Othello@nusnet-156-21.dynip.nus.edu.sg) |
| 05:32.22 | provolone | how can I connect a analog telephone to a soundcard or modem ? |
| 05:33.09 | *** join/#asterisk jtodd (~jtodd@ti.fox-den.com) |
| 05:34.56 | mishehu | provolone: plug it into the modem? |
| 05:35.37 | provolone | this will work with asterisk ? |
| 05:36.36 | mishehu | uhm. that seriously depends. |
| 05:36.40 | Nukemizer | I am trying to fix my T1 problems and looking for some advice on places to search. '1 WCT1/0/1 Clear (In use)' seems to be the state of the span I do not seem to get more than 1 irq miss. if I am on a B channel doing a conference 'meetme" the channel status does not change. most of the time i am lucky to get a b channel to come up.. |
| 05:36.52 | *** part/#asterisk Bentley (~rbc@S01060080c8135e6a.cg.shawcable.net) |
| 05:38.02 | beto75 | Guys...... anyone know of someone that can sell me inbound DID (for calling card) in Spain |
| 05:38.25 | Nukemizer | I realize much detail is missing from the above message but that would go on forever if I typed it out here. I guess I am hoping someone can offer ways that I can dig this monster out of its hole |
| 05:40.24 | NormAst | Nukemizer: I am using PRI (23+1D).. What type of protocal are you using on your T1? |
| 05:40.53 | provolone | I just want to capture the audio from the telephone for voip |
| 05:41.37 | Nukemizer | NormAst: I am using NI2 |
| 05:41.45 | beto75 | provolone,, better look for a FXS card for that |
| 05:41.48 | NormAst | Okay.. I am using Ni2 .. |
| 05:41.57 | NormAst | esf |
| 05:42.26 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
| 05:42.35 | beto75 | provolone: cheaper,,, a softphone with your pc speakers and mic. |
| 05:42.38 | Nukemizer | signalling=pri_net - span=1,0,0,esf,b8zs |
| 05:42.53 | NormAst | should be pri_cpe |
| 05:43.03 | *** join/#asterisk clive- (~pirch@myw-stp-66-18-83-119.sentechsa.net) |
| 05:43.11 | NormAst | the telco is pri_net |
| 05:43.30 | beto75 | thank you guys see you other day ,, BFN |
| 05:43.33 | *** part/#asterisk beto75 (~ha@201.133.240.35) |
| 05:43.38 | Nukemizer | I am connecting PBX to Asterisk using T1 |
| 05:44.03 | NormAst | I have a customer doin that.. PRI in to * and then out of * to PBX. |
| 05:44.09 | NormAst | and recording in the middle. |
| 05:45.04 | Nukemizer | yep have heard of setup that way |
| 05:45.05 | Juggie | norm did u get your trunking working? |
| 05:45.33 | NormAst | I haven't had time to play with it. |
| 05:45.47 | Juggie | if you need work done :P |
| 05:46.15 | *** join/#asterisk odie_flocon (~chatzilla@S01060011953994ee.cg.shawcable.net) |
| 05:46.18 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:46.27 | odie_flocon | Hello all. |
| 05:46.52 | odie_flocon | Hey Tuxinator. |
| 05:47.09 | NormAst | juggie: Thanks keep it in mind. |
| 05:47.16 | *** join/#asterisk footnote (~jhicks@67.141.135.121) |
| 05:50.17 | ManxPower | ~doc |
| 05:50.24 | ManxPower | ~docs |
| 05:50.25 | 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 |
| 05:50.42 | footnote | ~sccp |
| 05:50.53 | footnote | hrm |
| 05:51.02 | ManxPower | ~doc |
| 05:51.03 | jbot | from memory, doc is the command is ~docs, moron. |
| 05:51.08 | footnote | haha |
| 05:51.22 | footnote | ~SCCP |
| 05:51.29 | footnote | ok, the bot hates me |
| 05:52.36 | ManxPower | ~sccp |
| 05:52.37 | jbot | i guess sccp is Proprietary protocol used between Cisco Call Manager and Cisco VOIP phones. Also supported by some other vendors. |
| 05:52.59 | footnote | I wish Cisco hadn't chosen that acronym. |
| 05:53.16 | odie_flocon | Man you should see the New stuff that Cisco is doing. |
| 05:53.28 | clive- | odie what are they doing |
| 05:53.32 | *** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
| 05:53.32 | footnote | odie_flocon: I used to work in Petaluma :) |
| 05:53.45 | footnote | (formerly Cerent, acquired by Cisco) |
| 05:54.08 | odie_flocon | they are building an appliance with IVR/Voicemail etc. |
| 05:54.15 | ManxPower | ~sccp |
| 05:54.16 | jbot | i guess sccp is Proprietary protocol used between Cisco Call Manager and Cisco VOIP phones. Also supported by some other vendors. Also Signaling Connection Control Part (SCCP), a routing protocol in SS7 protocol suite in layer 4, provides end-to-end routing for TCAP messages to their proper database. |
| 05:54.19 | odie_flocon | it's nice lookin. |
| 05:54.47 | footnote | There goes Glenayre, Comverse. |
| 05:55.00 | odie_flocon | it's integrated in their switch, which also does VPN/Routing/Firewall etc. |
| 05:56.41 | footnote | Wonder when * supports OC192 :) |
| 05:56.44 | footnote | hehe |
| 05:57.24 | odie_flocon | OC192. that's not telecom. |
| 05:57.28 | footnote | erm |
| 05:57.42 | odie_flocon | that's Fibre, not standard Telecom. |
| 05:58.01 | odie_flocon | the question might be when will * support SS7. |
| 05:58.08 | odie_flocon | ~ss7 |
| 05:58.09 | jbot | somebody said ss7 was something that * does not support right now..... given time, who knows :-\ |
| 05:58.09 | footnote | That's what I'm here for... |
| 05:58.38 | odie_flocon | your here to implement ss7? |
| 05:58.45 | footnote | yes |
| 05:58.50 | odie_flocon | hmmm. |
| 05:59.01 | odie_flocon | it's an interesting protocol isn't it. |
| 05:59.02 | odie_flocon | :D |
| 05:59.08 | footnote | it's a PITA :) |
| 05:59.14 | footnote | but valuable |
| 05:59.29 | footnote | We're four disgruntled ex Glenayre engineers. |
| 05:59.30 | clive- | I belive ss7 is available for *, if you pay |
| 05:59.38 | clive- | I am not sure who you pay though |
| 05:59.44 | brc_ | bkw.digiweb.com/file1.gsm 1 through 4 |
| 05:59.49 | footnote | clive-: we're doing an open source version. |
| 05:59.56 | clive- | footnote, mabe poeple will pay you for ss7 in * :) |
| 05:59.56 | ManxPower | clive-: search the archives for "coppice" |
| 06:00.09 | clive- | ok, :) |
| 06:00.20 | ManxPower | Steve Underwood either has a commercial SS7 for Asterisk, or knows who does. |
| 06:00.22 | clive- | footnote, an open source would be a killer app for * |
| 06:00.27 | brc_ | steve underwood, aka coppice has commercial ss7 |
| 06:00.39 | brc_ | ^^what he said |
| 06:00.52 | Inv_arp | what bout 0penss7? |
| 06:00.56 | footnote | We've got probably $500mil worth of SS7 supported systems in the field that we did. |
| 06:01.07 | footnote | Inv_arp: not ready for primetime |
| 06:01.09 | ManxPower | Inv_arp: OpenSS7 has been discussed to DEATH on the mailing lists. |
| 06:01.11 | footnote | maybe never |
| 06:01.12 | ManxPower | ~mailinglist |
| 06:01.13 | jbot | hmm... mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. |
| 06:01.24 | odie_flocon | hmmm. |
| 06:01.35 | brc_ | <PROTECTED> |
| 06:01.37 | brc_ | waittttttttttt |
| 06:01.38 | footnote | Personally I believe Brian blew it by going to LiS |
| 06:02.21 | footnote | Sockets won, STREAMS died :) |
| 06:03.35 | clive- | footnote sounds like you know ss7 backwards:) |
| 06:03.44 | footnote | I'm actually the low-level guy |
| 06:03.58 | footnote | The others I work with do the upper parts of the stack |
| 06:04.34 | footnote | And DSP... |
| 06:04.47 | odie_flocon | I used to support a box that converted ss7 to PRI. |
| 06:04.47 | footnote | I took the easy job :) |
| 06:05.50 | footnote | We believe * is gonna be HUGE |
| 06:06.07 | Inv_arp | is SS7 basically SIP for PSTN? |
| 06:06.14 | footnote | more than that |
| 06:06.23 | footnote | but similar roles |
| 06:06.44 | footnote | There is a lot of concern with redundancy, failover, etc |
| 06:07.01 | footnote | It uses something of a funked out HDLC on the wire |
| 06:07.10 | *** join/#asterisk pratik (~root@202.149.48.205) |
| 06:07.19 | Inv_arp | footnote: ahh so its not tcp/ip based |
| 06:07.39 | mishehu | uff. |
| 06:07.40 | footnote | Well, SIGTRAN is all about interworking |
| 06:07.49 | mishehu | 1.0.6 still craps out with speex 1.1.6 |
| 06:08.22 | pratik | i am facing problem in configuring phones for asterisk |
| 06:08.31 | footnote | We hope to bring a lot of this stuff into the asterisk environment over the next year or two |
| 06:09.20 | *** join/#asterisk redder86 (~lee@gateway.howardsilvan.com) |
| 06:09.31 | redder86 | Mavvie: there? |
| 06:10.01 | footnote | hrm, anybody playing with using SCTP+Asterisk? |
| 06:10.44 | mishehu | hmm... I seem to have a deadlock.. |
| 06:12.13 | pratik | how to record aclls using the cdr |
| 06:13.27 | pratik | i have 2 internal phones so how to record those two phones |
| 06:15.58 | *** join/#asterisk paulc (paulc@S010600062586a0b4.vc.shawcable.net) |
| 06:16.50 | *** part/#asterisk redder86 (~lee@gateway.howardsilvan.com) |
| 06:21.22 | mishehu | Mar 3 00:20:57 DEBUG[27189]: Dropping voice to exceptionally long queue on IAX2/alHuti@alHuti/4 |
| 06:21.40 | kpfleming | footnote: yeah, we need SCTP (in the Linux kernel but experimental) and M2UA/M3UA as a start |
| 06:21.47 | wildcard0 | paulc, hey. how's it going? |
| 06:21.49 | mishehu | that's what I get on a call that goes IAX2-g711ulaw->IAX2-speex |
| 06:21.50 | wildcard0 | <--- ex calltrex |
| 06:22.04 | footnote | kpfleming: Yah, I've been testing it on 2.6 |
| 06:22.10 | footnote | It rocks for us so far |
| 06:22.18 | kpfleming | nice to hear |
| 06:22.57 | footnote | SCTP is useful even apart from M2UA ya know |
| 06:23.06 | wildcard0 | whats the advantage of sctp? |
| 06:23.12 | kpfleming | yes, i learned about it a moht or two ago... very nifty protocol |
| 06:23.14 | footnote | redundant links |
| 06:23.20 | footnote | among other things |
| 06:23.22 | wildcard0 | ah. that's cool |
| 06:23.33 | kpfleming | diverse routing, parallel delivery, etc. |
| 06:23.36 | footnote | yeap |
| 06:23.49 | wildcard0 | ah. so kinda like the old mbone stuff, but right this time |
| 06:23.51 | paulc | wildcard0 aka.. umm.. tall bloke, Daniel's mate, foreign, who's name escapes me right now.. umm.. |
| 06:23.53 | paulc | or.. someone else? |
| 06:24.05 | kpfleming | sort of... SCTP is pretty much for nailed-up connections, not dynamic ones |
| 06:24.36 | wildcard0 | paulc, that was markus...who is actually getting married soon. this is mike. we talked a bunch of times over...something asterisk related :) |
| 06:24.58 | paulc | Markus.. that's right.. Mike rings a bell.. we emailed some stuff back and forth AGES ago right? |
| 06:25.12 | wildcard0 | ya like >1 yr ago |
| 06:25.30 | paulc | Time flies + small world eh? :-) |
| 06:25.35 | paulc | What you up to these days? |
| 06:25.36 | wildcard0 | totally. |
| 06:25.51 | *** join/#asterisk Bentley (~Bentley@S01060080c8135e6a.cg.shawcable.net) |
| 06:26.18 | wildcard0 | right now, working with some interconnects |
| 06:26.33 | wildcard0 | did 24/7 communications get in touch with you reciently? |
| 06:27.17 | footnote | kpfleming: we figure SCTP to be pretty useful for intra-system ethernet plumbing but all we've done so far is a handful of test programs |
| 06:27.20 | *** part/#asterisk Bentley (~Bentley@S01060080c8135e6a.cg.shawcable.net) |
| 06:27.22 | *** join/#asterisk Bentley (~Bentley@S01060080c8135e6a.cg.shawcable.net) |
| 06:27.46 | kpfleming | footnote: yeah, i have in mind some other uses for it too, related to some future asterisk<->asterisk communication |
| 06:27.57 | footnote | indeedy :) |
| 06:28.05 | paulc | wildcard0: Doesn't ring a bell.. should they have? |
| 06:28.51 | wildcard0 | they're one of the places im working with. mentioned they needed some additional help and i mentioned you. i don't know if he actually called anyone tho |
| 06:29.59 | footnote | kpfleming: you familiar with the cvsup setup of the project? |
| 06:30.05 | paulc | Hmm.. well thanks for remembering me :-) No contact from them so far but I'm around.. wonder if they emailed and got hit with the TMDA spam filter thing, I haven't checked that for a while.. |
| 06:30.23 | kpfleming | footnote: nope, i'm a BK diehard, i avoid CVS like the plague |
| 06:30.23 | *** part/#asterisk Bentley (~Bentley@S01060080c8135e6a.cg.shawcable.net) |
| 06:30.30 | footnote | gotit |
| 06:30.38 | wildcard0 | np :) there arent's many of us around this area so i figure we should at least attempt to stick together :) |
| 06:30.42 | footnote | understandable |
| 06:30.47 | *** join/#asterisk |Vulture| (~Vulture@109.238.204.68.cfl.res.rr.com) |
| 06:30.47 | `Sauron | boo hiss |
| 06:30.51 | `Sauron | svn all the way |
| 06:31.00 | `Sauron | :) |
| 06:31.03 | mishehu | Mar 3 00:30:45 WARNING[27513]: codec_speex.c:166 speextolin_framein: Out of buffer space |
| 06:31.22 | footnote | Anything but ClearCase is fine by me :) |
| 06:31.22 | mishehu | grrr |
| 06:31.23 | `Sauron | dum di dum |
| 06:31.24 | `Sauron | man, this is taking forever to recompile |
| 06:31.43 | footnote | well, PVCS is suck too |
| 06:31.54 | kpfleming | VSS, VSS, VSS! |
| 06:32.07 | `Sauron | who needs versioning |
| 06:32.19 | `Sauron | just edit the text files on seperate machines |
| 06:32.19 | *** join/#asterisk Bentley (~rbc@S01060080c8135e6a.cg.shawcable.net) |
| 06:32.23 | Qwell | pvcs, vss, and clearcase are...crap. |
| 06:32.23 | footnote | `Sauron: I use cat as my primary editor! |
| 06:32.23 | `Sauron | and muddle it together :) |
| 06:32.40 | `Sauron | hehe |
| 06:32.43 | footnote | ed is bloatware, etc |
| 06:32.58 | Qwell | footnote: ed is the standard text editor! |
| 06:33.03 | footnote | bloat! |
| 06:33.21 | Qwell | http://www.gnu.org/fun/jokes/ed.msg.html |
| 06:33.22 | Qwell | heh |
| 06:33.22 | footnote | I actually did a LOT of programming under ed :) |
| 06:33.47 | Qwell | any time somebody asks for a good editor, I tell them ed or emacs... |
| 06:33.53 | Qwell | I'm such an asshole. :( |
| 06:33.58 | footnote | Qwell: xemacs! |
| 06:34.10 | Qwell | footnote: I assume you've read the above link? |
| 06:34.14 | footnote | yeah |
| 06:34.18 | footnote | ed man! man ed! |
| 06:34.23 | Qwell | pretty funny. heh |
| 06:34.29 | footnote | It's old as dirt :) |
| 06:34.36 | Juggie | whats the best editor for windows? :) |
| 06:34.42 | Qwell | likely older then me |
| 06:34.42 | footnote | Juggie: fdisk |
| 06:34.43 | Juggie | cuz we all know some of us edit code on winblows |
| 06:34.44 | Qwell | Juggie: notepad |
| 06:34.51 | Juggie | accept it or not :) |
| 06:34.55 | Qwell | actually, ultraedit is pretty good |
| 06:34.58 | Juggie | ucant get away from it |
| 06:35.03 | Juggie | i like PSPad |
| 06:35.04 | Qwell | Thats what I always used in the past. |
| 06:35.05 | footnote | Already did! |
| 06:35.06 | Juggie | its freeware |
| 06:35.08 | Juggie | look it up |
| 06:35.15 | Qwell | uedit is freeware too, isn't it? |
| 06:35.21 | Qwell | it was when I used it, I think |
| 06:35.33 | Qwell | I actually use nano in cygwin nowadays |
| 06:35.51 | footnote | Juggie: I was forced to spend a summer at Bldg 6 in Redmond |
| 06:35.54 | *** join/#asterisk pygmy (~pygmy@141.110.15.200) |
| 06:35.59 | footnote | That cured me of the M$ disease |
| 06:36.10 | Juggie | nope, its written by some christian who says god told him to write ultraedit |
| 06:36.16 | Qwell | heh |
| 06:36.18 | Qwell | seriously? |
| 06:36.22 | Juggie | yes read the webpage |
| 06:36.25 | Qwell | reading |
| 06:36.49 | Qwell | I must have pirated it then. |
| 06:36.55 | kpfleming | hey, if you haven't written PL/1 for 48 hours straght using a VIP7801 green-screen and Emacs, you haven't lived :-) |
| 06:36.55 | Qwell | but its okay, God told me to. |
| 06:36.56 | footnote | If he's devout he'll prolly think he will burn in hell for creating bugs |
| 06:37.17 | footnote | kpfleming: PL/1 makes me want to slash my wrists :) |
| 06:37.29 | Juggie | the web design is new from the orange one |
| 06:37.34 | Juggie | so i dunno if its still there |
| 06:37.42 | kpfleming | we did a lot of good stuff with it back in the day, it was way ahead of its time :-) |
| 06:37.45 | footnote | yeah |
| 06:37.49 | footnote | but WORDY |
| 06:37.58 | footnote | i barely remember the syntax |
| 06:38.18 | Juggie | http://www.ultraedit.com/index.php?name=Content&pa=showpage&pid=7 |
| 06:38.25 | Juggie | there u go, read that. |
| 06:38.36 | footnote | I never worked in Ada but I know people that swear by it. |
| 06:38.45 | kpfleming | yes, it was wordy, although not as much as COBOL and FORTRAN |
| 06:39.00 | Qwell | Juggie: just read it..sad |
| 06:39.11 | kpfleming | i'm just amazed at how Unix/Linux continue to evolve to have the features we had on Multics in the 70s/80s |
| 06:39.16 | footnote | yeap |
| 06:39.25 | footnote | Never got a chance to work on Multics |
| 06:39.34 | marc32344 | not one amd motherboard i can find that is compatible with the te410p |
| 06:39.44 | footnote | I learned C on V7 |
| 06:40.00 | tuxinator_linux | You guys are a bit older than me? |
| 06:40.03 | kpfleming | cripes, another firefox update was just released |
| 06:40.08 | kpfleming | tuxinator_linux: probably so, yes |
| 06:40.09 | Juggie | its a good editor even so :) |
| 06:40.13 | Qwell | tuxinator_linux: many are probably considerably so |
| 06:40.14 | footnote | tuxinator_linux: i'm a grampa |
| 06:40.15 | tuxinator_linux | 24 here |
| 06:40.18 | Juggie | none the less, for windows use pspad |
| 06:40.20 | Juggie | it is great |
| 06:40.28 | pygmy | I'm trying to set up IAX2 peering to (at this point) use a softphone to make local calls through my provider. I have created 2 entries for the provider as both peer and user in my iax.conf and updated the IAXINFO and TRUNK entries in extensions.conf. I still have nothing listed in iax2 show channels. What might I be missing? |
| 06:40.31 | footnote | tuxinator_linux: same age as my son :) |
| 06:40.49 | Qwell | /nick the_youngest_guy_in_telecom |
| 06:41.00 | Qwell | probably not. |
| 06:41.03 | Juggie | pygmy, the iax trunk will only be up if there is an active call |
| 06:41.06 | Juggie | at least how i have it setup |
| 06:41.12 | mishehu | #1200 0x0806294b in ast_translate (path=0x819200c, f=0x8194c50, consume=135866892) at translate.c:189 p = (struct ast_trans_pvt *) 0x8194c50 out = (struct ast_frame *) 0xb7787ad0 delivery = {tv_sec = 775043377, tv_usec = 135877316} Previous frame inner to this frame (corrupt stack?) |
| 06:41.14 | footnote | young is good. i've known some *really* dynamite teenage developers. |
| 06:41.16 | JamesDotCom | Qwell: how old? |
| 06:41.17 | Juggie | maybe u can keep it persistant not sure |
| 06:41.24 | mishehu | anybody able to decipher that bt full ? |
| 06:41.28 | Qwell | JamesDotCom: not young enough to say what I said, while being serious. |
| 06:41.36 | JamesDotCom | haha |
| 06:41.40 | JamesDotCom | well |
| 06:41.42 | tuxinator_linux | mishehu: is that greek? |
| 06:41.45 | JamesDotCom | i just turned 21 a week ago |
| 06:41.51 | Qwell | though, if I get my son to config my next asterisk box...maybe he can be |
| 06:41.55 | Juggie | mishehu, if its a crash post it to the buglist |
| 06:41.58 | footnote | JamesDotCom: want a tip? |
| 06:42.00 | JamesDotCom | but i keep forgetting i'm getting considerably older |
| 06:42.02 | Qwell | "my son"...that still sounds freaking weird |
| 06:42.05 | marc32344 | james-- it will g faster from here |
| 06:42.15 | footnote | JamesDotCom: 36 year old wimmen are hawt for 21 year olds |
| 06:42.20 | JamesDotCom | hahahaha |
| 06:42.28 | JamesDotCom | i got me a 19yo |
| 06:42.30 | Qwell | footnote: baggage... |
| 06:42.32 | footnote | I wish someone had told ME that |
| 06:42.36 | Juggie | id prefer be 36 year old man, with a 21 year old woman :) |
| 06:42.36 | tuxinator_linux | footnote: 36 old wimmen, eww |
| 06:42.43 | pygmy | Juggie: When I try to initiate a call from my softphone: I get either A) No authority found (if set up for authentication) or B) Invalid user name (when not sending credentials) |
| 06:42.54 | footnote | tuxinator_linux: you won't always feel that way whippersnapper :) |
| 06:43.05 | Juggie | sounds like you just dont have your iax setup properly. |
| 06:43.06 | mishehu | Juggie: nobody likes me on the list ;-) the dev list I suppose? |
| 06:43.10 | pygmy | On the asterisk end I get A) |
| 06:43.11 | tuxinator_linux | My wife is 2 years younger |
| 06:43.14 | Qwell | footnote: well, yeah, when you're 60...you'd be lucky to get a 36 year old, heh |
| 06:43.14 | pygmy | Rejected connect attempt from $IP |
| 06:43.24 | *** join/#asterisk bofh42 (~bofh42@pD95EDCC6.dip.t-dialin.net) |
| 06:43.26 | footnote | Qwell: for your information |
| 06:43.27 | pygmy | or B) No registration for peer 'pygmy' |
| 06:43.30 | footnote | my gf is 26 :) |
| 06:43.39 | Qwell | footnote: I wasn't saying you were 60 or anything. heh |
| 06:43.40 | JamesDotCom | i wouldnt mind finding a sugarmummy |
| 06:43.41 | JamesDotCom | hahaha |
| 06:43.53 | Qwell | I picked a random age. My apologies if I hit the mark... |
| 06:43.55 | footnote | JamesDotCom: I recommend one with a pool. |
| 06:44.16 | Qwell | Thats only 16 years... |
| 06:44.18 | provolone | I am 21 and I have been dating a 30 yo |
| 06:44.23 | Qwell | eww |
| 06:44.36 | JamesDotCom | sif eww |
| 06:44.39 | JamesDotCom | 30s still decent |
| 06:44.45 | tuxinator_linux | provolone: is she Swiss? |
| 06:44.46 | footnote | 35 is decent! |
| 06:44.47 | paulc | I'm 30 and dating a 22 year old.. who acts more like 24/25 - definite "head screwed on right" case.. |
| 06:45.02 | tuxinator_linux | robbing the cradle |
| 06:45.07 | JamesDotCom | paulc: yeah i got one of them too, which is why i can handle that she's 19 |
| 06:45.07 | JamesDotCom | hehe |
| 06:45.13 | footnote | There are even hawt 40 year old wimmens (if they rich) |
| 06:45.19 | tuxinator_linux | One of my friends is 49, he married at 19 girl |
| 06:45.20 | Qwell | rich ~= hawt |
| 06:45.25 | paulc | and that said.. I only just turned 30 a couple of weeks back.. |
| 06:45.46 | footnote | paulc: amen brother |
| 06:46.18 | footnote | ok, time for gramps to take his geritol and crash |
| 06:46.22 | tuxinator_linux | boy, I tought all you guys were a bunch of highschoolers, the way you talk in here |
| 06:46.39 | footnote | tuxinator_linux: old people didn't mean to end up that way... |
| 06:46.47 | Qwell | No offense to anybody for my following statement. |
| 06:46.56 | tuxinator_linux | I think I will be a cool old fart |
| 06:46.58 | Qwell | tuxinator_linux: I've found that many people in the telecom industry are a little aged. |
| 06:47.01 | JamesDotCom | Qwell: shutup and tell us your age, heheh |
| 06:47.04 | Qwell | Very "young at heart" though |
| 06:47.04 | marc32344 | any amd mbs for te410p? |
| 06:47.07 | Qwell | JamesDotCom: 22 :p |
| 06:47.07 | pygmy | Does anyone have suggestions for me? |
| 06:47.11 | JamesDotCom | ahh |
| 06:47.11 | JamesDotCom | see |
| 06:47.12 | tuxinator_linux | Qwell: I see that too |
| 06:47.13 | JamesDotCom | older than me |
| 06:47.25 | footnote | Qwell: the ones that survived the blowout anyway... |
| 06:47.34 | tuxinator_linux | former telecom is a dying art... |
| 06:47.39 | Qwell | footnote: and/or years of ring voltage... |
| 06:47.43 | footnote | yeah |
| 06:47.44 | tuxinator_linux | replaced by new digital stuff |
| 06:47.46 | Qwell | You know what I mean, of course. :p |
| 06:48.03 | provolone | is there a way to route an incoming call to a modem into voip ? |
| 06:48.05 | footnote | tuxinator_linux: some of us were always multi-disciplinary :) |
| 06:48.10 | Qwell | ring voltage to the teeth isn't pretty |
| 06:48.20 | footnote | nite folks |
| 06:48.38 | tuxinator_linux | I was shocked by a ring once |
| 06:48.42 | tuxinator_linux | kinda fun |
| 06:49.03 | tuxinator_linux | Qwell: Why was it in your month |
| 06:49.06 | Qwell | tuxinator_linux: I'm sure there are several people here who have been holding the cable in their mouths at the time |
| 06:49.06 | tuxinator_linux | mouth |
| 06:49.13 | Qwell | because its convenient...you only have two hands |
| 06:49.21 | tuxinator_linux | true |
| 06:49.26 | tuxinator_linux | I will have to try that |
| 06:49.37 | tuxinator_linux | Gotta do everything, once. |
| 06:50.20 | tuxinator_linux | Jeritol? |
| 06:51.38 | tuxinator_linux | ~geritol |
| 06:52.12 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 06:52.27 | tuxinator_linux | jbot doesn't have an answer |
| 06:52.35 | tuxinator_linux | ~what is geritol? |
| 06:52.37 | jbot | tuxinator_linux: I think you lost me on that one |
| 06:52.50 | tuxinator_linux | jbot: me too |
| 06:52.52 | jbot | moi? |
| 06:53.04 | AgiNamu | How can I start a process in the background, and redirect stderr AND stdout to /dev/null? |
| 06:53.04 | tuxinator_linux | jbot: yes you |
| 06:53.05 | jbot | i'm a broken bot |
| 06:53.14 | tuxinator_linux | jbot: fix yourself |
| 06:53.16 | jbot | ACTION takes yourself to the vet for a "special" visit. |
| 06:53.22 | Qwell | heh |
| 06:53.31 | tuxinator_linux | jbot: poor jbot |
| 06:53.43 | tuxinator_linux | jbot: no baby bots |
| 06:53.56 | AgiNamu | seems like things screw up with the & to run in background |
| 06:54.11 | Qwell | AgiNamu: been a while, gimme a sec |
| 06:54.24 | AgiNamu | mono x.exe &>/dev/null <-- works fine |
| 06:54.32 | AgiNamu | mono x.exe & &>/dev/null <-- doens't work |
| 06:54.42 | Qwell | are you trying to background it? |
| 06:54.50 | AgiNamu | yea, bacgrkound, with everything sent to /dev/null |
| 06:54.52 | Qwell | What about mono x.exe &>>/dev/null & |
| 06:54.53 | Qwell | ? |
| 06:55.10 | pashah | morning |
| 06:55.29 | AgiNamu | oh, hey |
| 06:55.31 | AgiNamu | that works :) |
| 06:55.34 | AgiNamu | thanks qwell! |
| 06:55.43 | Qwell | technically all of that is one 'command' |
| 06:56.45 | tuxinator_linux | ~weather KGEU |
| 06:57.02 | *** join/#asterisk Muiz (someone@dhcp185-1-186.dsl.ucc-net.ca) |
| 06:57.04 | pashah | I have several boxes, every now and then one of the processes freezes. I can asterisk -rvvvc to it, but reload cmd does nothing and * does not route calls, kill -9 and restart asterisk process helps |
| 06:57.27 | Muiz | hi there...does anybody know what happened to the loligo site? I can't get in ths evening and I wanted their sound files |
| 06:57.33 | AgiNamu | pashah, are you leaving it open on a disconnected console? |
| 06:57.52 | pashah | AgiNamu: nope |
| 06:57.59 | AgiNamu | pashah, how do you start asterisk? I get a similar problem :\ |
| 06:58.13 | AgiNamu | he CLI "responds" but doesnt do anything, eh? |
| 06:58.14 | pashah | AgiNamu: helps only kill -9 |
| 06:58.20 | pashah | AgiNamu: exactly |
| 06:58.27 | AgiNamu | i'd sure like to know :P |
| 06:58.33 | AgiNamu | you running a recent head? |
| 06:58.34 | *** join/#asterisk nirs (~nirs@62.90.49.115) |
| 06:58.35 | AgiNamu | or stable? |
| 06:58.36 | pashah | AgiNamu: I have the problem with every cvs head ive tryed |
| 06:58.49 | pashah | AgiNamu: a week old i guess |
| 06:58.53 | AgiNamu | me too. |
| 06:59.02 | AgiNamu | I need to go back to stable. I just gotta test my own patches again :@ |
| 06:59.08 | |Vulture| | anyone know a good program to burn in/test a new server? I tried Seti@Home but couldn't get above 1.1load... it is Dual Xeon 2.8 |
| 06:59.17 | AgiNamu | cause *someone* doesn't understand VSP scenario :\ |
| 06:59.39 | Qwell | |Vulture|: dd if=/dev/urandom of=/dev/null |
| 06:59.42 | marc32344 | prime?? |
| 06:59.44 | Qwell | should get it to 8-10 or so |
| 06:59.59 | |Vulture| | Qwell: Ill give that a try |
| 07:00.02 | AgiNamu | vulture, what about the BIOS burn in tests? |
| 07:00.24 | |Vulture| | AgiNamu: BIOS burn in? |
| 07:00.35 | Qwell | |Vulture|: it doesn't hit the HD or anything though, so you may always want to try it with of=/path/to/some/file |
| 07:00.35 | AgiNamu | you know what sucks? Thinking you're SSH'd into a lan box and doing "route delete default", and then suddenly realising that you were actually logged into a machine 2000km away. |
| 07:00.38 | |Vulture| | its a Dell I didn't see anything in there |
| 07:00.49 | AgiNamu | Vulture, yea, some BIOSes /MBs have a burn-in |
| 07:00.53 | AgiNamu | my Intel PERL board does |
| 07:01.03 | |Vulture| | AgiNamu... wow that sucks |
| 07:01.04 | tuxinator_linux | AgiNamu: oops |
| 07:01.10 | AgiNamu | yea :P |
| 07:01.13 | Juggie | |Vulture|, is the voicemail patch on mantis yours? |
| 07:01.25 | AgiNamu | I panicked and went to reboot... then I realised, I have another box in the same subnet :D |
| 07:01.27 | |Vulture| | Juggie: can you link me to it Ill throw it on right now |
| 07:01.34 | |Vulture| | I am running HEAD |
| 07:01.42 | AgiNamu | at least I didn't do iptables -f |
| 07:01.44 | Qwell | AgiNamu: Don't Panic |
| 07:02.00 | AgiNamu | Qwell, I don't. I have glasses that turn dark when stuff like this happens. |
| 07:02.04 | |Vulture| | AgiNamu: I was on a 2k3 machine and all of a sudden my VNC and RDC stopped... and I just though... oh shit |
| 07:02.04 | Qwell | :p |
| 07:02.23 | |Vulture| | AgiNamu: KVMoIP :) |
| 07:02.24 | Qwell | |Vulture|: oh, thats normal. It's called "crashing" |
| 07:02.29 | Trepalium | You know, it's errors like this that make me love Microsoft software. "The Microsoft Exchange System Attendant services failed to start because of the following error: The operation completed successfully." |
| 07:02.37 | AgiNamu | W2K3 Rocks with RPD. network issues dont make any problem. |
| 07:02.38 | Qwell | rofl |
| 07:02.41 | AgiNamu | trep LOL |
| 07:02.55 | |Vulture| | Qwell: ah... oky that makes a lot sense I think I read something about it in the 2k3 manual... something about common accurances |
| 07:03.12 | AgiNamu | I'm just getting over the fact that I need a MAPI program (Outlook) for SQL Server to email me with alerts :@ |
| 07:03.37 | AgiNamu | seriously, whens the last time anyone here has had a W2k3 machine crash? |
| 07:03.48 | AgiNamu | I've never seen it. And XP only crashes on dumbass hardware |
| 07:03.55 | Trepalium | With or without me doing something obviously stupid? |
| 07:03.58 | |Vulture| | AgiNamu: 2k3... not very common I actually had a few 2k boxes crash though |
| 07:04.15 | AgiNamu | yea, 2k had a few issues here and there... but XP and above seem very solid |
| 07:04.28 | |Vulture| | yea I enjoy 2k3/XP |
| 07:04.32 | AgiNamu | and as far as hangs... lets just say that Gnome gives me a lot more hassle than XP :D |
| 07:04.36 | |Vulture| | ....did I just say that outloud? |
| 07:04.44 | AgiNamu | up2date freezes a lot for me, for instance. |
| 07:04.46 | AgiNamu | lol |
| 07:04.55 | |Vulture| | I am running yum right now on my new server |
| 07:05.10 | tuxinator_linux | |Vulture|: you're making me hungry |
| 07:05.12 | |Vulture| | takes... forever... |
| 07:05.15 | |Vulture| | ;P |
| 07:05.23 | Trepalium | Yeah, but if Gnome freezes, you can still generally access your server. |
| 07:05.25 | Juggie | vulture, the reason i asked is because i thin kthere needs to be a better way for agents to log out |
| 07:05.30 | Juggie | theres no reason to ask for a pin |
| 07:05.34 | Juggie | or a # |
| 07:05.36 | Juggie | to log out |
| 07:05.48 | |Vulture| | to log out of voicemail? |
| 07:05.52 | AgiNamu | i dont need to run yum. RHEL is set for autoupdates |
| 07:05.55 | Juggie | and if it was your patch i was gonna say you know the code, can u make it happen ;) |
| 07:06.01 | Juggie | no, log out of an agent |
| 07:06.01 | AgiNamu | and it just handles itself. I think RedHat emails me if there's an issue too. |
| 07:06.12 | |Vulture| | ah Ill take a look at it |
| 07:06.28 | |Vulture| | all sorts of fun stuff going on |
| 07:06.31 | tzafrir | AgiNamu, the email is sent quite a while after apt2date notices the issue |
| 07:06.43 | AgiNamu | oh :( |
| 07:06.50 | tzafrir | this is up2date on RHEL, not the yum wrapper on Fedora, right? |
| 07:06.57 | Juggie | Vulture, a new function called like AgentCallBackLogout could be good |
| 07:07.00 | AgiNamu | well.. things are still working, and no problems yet |
| 07:07.26 | Juggie | which accepts the agent # and a flag to ask for password or not |
| 07:07.50 | Juggie | i tried looking at the code today, but if you already know it its better you take a look. |
| 07:07.54 | tzafrir | THe mail is often sent a couple of hours later. If you don't care about such a time frame, you can ignore it |
| 07:09.36 | Juggie | does anyone know the intended operation of agent groups? if i do like member => Agent/@1 |
| 07:09.41 | Juggie | and my queue is set to ring all |
| 07:09.47 | Juggie | should it ring everyone in that group? |
| 07:10.06 | Juggie | trying to decide if its intended behavior or a bug, because for me it does not ring everyone. |
| 07:10.26 | AgiNamu | wow |
| 07:10.46 | AgiNamu | time for bed. My Asterisk -> .NET CDR Mediation and status update just worked. |
| 07:11.51 | AgiNamu | sorta amazing actually. all these moving parts (mono, cdr-csv patches, etc.) |
| 07:12.01 | AgiNamu | and all the data just appears in SQL... gets billed... works |
| 07:12.41 | *** join/#asterisk idnar (mithy@idnar.user) |
| 07:13.31 | AgiNamu | night all |
| 07:13.56 | idnar | hrm... every day or two, I have to reload the kernel module for my Digium Wildcard TDM400P, because it goes a little haywire; has anyone experienced the same sort of problem? any suggestions on how to fix the problem? |
| 07:14.58 | Qwell | bed time |
| 07:28.30 | tzafrir | idnar, you better state the version of zaptel you use and the kernel version. |
| 07:29.21 | tzafrir | Also, what do you mean by "haywire"? Could you describe the symptoms? |
| 07:31.11 | *** join/#asterisk harryvv (~none@S010600055d210201.vs.shawcable.net) |
| 07:31.17 | tzafrir | IIRC you should stop asterisk to unload zaptel. Any chance the problem is with Asterisk getting "haywire" (whatever that means)? |
| 07:33.05 | harryvv | I came home about a hour ago and found my switchs lights on the * port peged out at max. Did a top on my debian system found that * was gobbling up 98% cpu time. Any ideas where I can go to find out why it was doing this? there was no phone traffic on it at the time. Did a quick stop start and the network actvity came to a halt. |
| 07:36.15 | tzafrir | harryvv, strace is your friend |
| 07:36.56 | tzafrir | strace -p <asterisk's pid> |
| 07:37.51 | tzafrir | Also look at 'netstat -nu' to see active udp traffic . or netstat -nt, or -ntv or whatever |
| 07:37.53 | *** join/#asterisk ipso (~ipso@d207-81-249-35.bchsia.telus.net) |
| 07:38.15 | ipso | Anyone know what would cause this error when starting Asterisk? loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_retrieve_call_to_death |
| 07:38.49 | wildcard0 | looks like you didn't load a module that needs to be loaded |
| 07:38.58 | tzafrir | ipso, the module failed to load. Pobably a non-matching library |
| 07:39.11 | wildcard0 | think so? it's an ast_ function... |
| 07:39.43 | wildcard0 | i'm gonna go with "didn't load correct modules first" :) |
| 07:39.47 | ipso | tzafrir: I'm trying to use the ztdummy module, I have zaptel and ztdummy loaded, do I need to load additional kernel modules? |
| 07:39.52 | tzafrir | hmmm... and old left-over module? maybe chan_zap was not built and this is from an older installation? |
| 07:40.02 | wildcard0 | that could be |
| 07:40.33 | wildcard0 | well, either way, it's definitely a library issue...either one that asterisk loads or a module it loads |
| 07:40.34 | harryvv | tz well it was obvios something was going on with asterisk and not some one that hacked into it. Dont know what caused the problem. |
| 07:40.36 | tzafrir | ipso, I refer to asterisk's modules (like chan_zap.so), not to kernel modules. |
| 07:41.11 | tzafrir | harryvv, you kind-of destroyed the evidences... |
| 07:41.17 | harryvv | I know |
| 07:41.23 | tzafrir | you could have also ran: |
| 07:41.36 | tzafrir | asterisk -rvvv and get some tracing |
| 07:42.02 | harryvv | I actually did asterisk -r into cli and saw nothing showing up. |
| 07:42.27 | tzafrir | harryvv, 'set verbose' to something more resonable than 0 |
| 07:42.37 | harryvv | I know |
| 07:42.38 | harryvv | ;) |
| 07:42.50 | harryvv | will remember to do that next time. |
| 07:43.07 | tzafrir | But then again, netstat should tell you where it is connecting to. Also: if someone has abused your PBX to get free calls, it should appear on the CDR |
| 07:43.17 | *** join/#asterisk landrocker (~landrocke@port-222-152-54-115.fastadsl.net.nz) |
| 07:43.50 | wildcard0 | especially if you were smart enough to put your db on another box and only allow inserts (no updates, deletes or selects) |
| 07:44.50 | landrocker | Hey all, I'm playing round with asterisk and want to have a look at 3way calling etc, but I'm just using Xlite and it doesn't appear to have a 'flash' button - I vaguely recall there was a config option so that * or # could be used instead of flash, can anyone enlighten me? Thanks :) |
| 07:44.53 | harryvv | I checked the cdr came up with nothing unusuall |
| 07:45.02 | tzafrir | wildcard0, or the least smart to run Asterisk as non-root. But that is indeed a good point about "inserts-only" user |
| 07:46.31 | wildcard0 | i had to harden an asterisk box a few weeks ago. a friend and i took turns hardening it and trying to break into it |
| 07:46.39 | wildcard0 | ran across a number of small things like that |
| 07:47.43 | harryvv | wild interesting :) |
| 07:48.46 | wildcard0 | it was a good process for us. i learned a lot |
| 07:48.58 | wildcard0 | stuff you normally wouldn't think of...like that inserts-only thing |
| 07:49.25 | harryvv | wild what distro are you running |
| 07:50.32 | harryvv | wildcard0 what distro of linux are you running |
| 07:51.03 | wildcard0 | oh sorry. i have a bunch of them running. but i've been using fc2 for asterisk |
| 07:51.21 | wildcard0 | mostly cause of it's ability to pxe boot/kickstart install -very- easily |
| 07:52.30 | harryvv | yea i just downloaded fc3 but for somone reason it says that something about long like it does not support longfile so could not install the iso. |
| 07:52.34 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 07:52.41 | harryvv | you have debian? |
| 07:52.41 | *** join/#asterisk Tarox (someone@pD9E7ABCB.dip.t-dialin.net) |
| 07:52.57 | wildcard0 | i have 2 debian boxes. but neither of them are running asterisk |
| 07:53.04 | harryvv | okay |
| 07:53.23 | wildcard0 | debian/stable generally doesn't have new enough tools for me. and testing/unstable aren't gaurenteed secure |
| 07:53.33 | wildcard0 | (not to say the debian team doesn't do everything they can to make them so) |
| 07:53.38 | harryvv | you feel fd2 is better for asterisk then |
| 07:54.03 | wildcard0 | i went with fc2 really cause it was easy to install on many boxes quickly and because it had everything i needed up to date |
| 07:54.42 | harryvv | I am running it on debian 2.4.27. And mabey next time its going to be installed on a fedora box :) |
| 07:55.09 | harryvv | anyway im off to bed. |
| 07:55.11 | harryvv | long day. |
| 07:55.23 | wildcard0 | im not distro-religious. do what works for you |
| 07:55.24 | wildcard0 | night |
| 08:00.43 | *** join/#asterisk Newbie___ (some@60.48.51.187) |
| 08:01.37 | *** join/#asterisk oej (~oej@apollo.webway.se) |
| 08:04.47 | landrocker | hmm, anybody? |
| 08:11.06 | *** join/#asterisk ta[i]nted (~ta_i_nted@65-60-70-243-cust.telepacific.net) |
| 08:11.31 | ta[i]nted | what does it usually mean if i can call out on a SIP channel but cannot receive calls |
| 08:11.34 | ta[i]nted | firewall? |
| 08:14.31 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 08:14.41 | r0d3nt|m | boo. |
| 08:15.34 | RaYmAn-Bx | ta[i]nted: most likely either NAT (i.e. server doesn't know where to send calls) or firewall...depending on the actual problem..(i.e. does asterisk receive the call at all or?) |
| 08:18.48 | ta[i]nted | RaYmAn-Bx asterisk doesn't seem to receive the call |
| 08:19.09 | ta[i]nted | RaYmAn-Bx it's on an external IP with 5060 open so firewall shouldn't be the issue |
| 08:19.42 | ta[i]nted | i get hung up in 2 seconds after placing the call.. no ringing |
| 08:19.59 | *** join/#asterisk oej (~oej@apollo.webway.se) |
| 08:21.28 | RaYmAn-Bx | ta[i]nted: try logging the packets sent when it registers with the server (I assume you do actually have a register => line for the server :P) |
| 08:26.32 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 08:28.46 | *** join/#asterisk FryGuy- (fryguy@c-24-23-19-33.client.comcast.net) |
| 08:29.24 | |Vulture| | anyone try using AMP? |
| 08:34.05 | Zeeek | no |
| 08:34.23 | *** join/#asterisk w0w0 (~w0w0@80-28-171-221.adsl.nuria.telefonica-data.net) |
| 08:37.08 | Newbie___ | i did |
| 08:40.21 | |Vulture| | ah nvm I got it... had to create a passwd file for the web interface |
| 08:40.28 | |Vulture| | just checking it out... looks interesting |
| 08:40.46 | *** join/#asterisk fitzel (~flint@p3EE3986A.dip0.t-ipconnect.de) |
| 08:50.50 | *** join/#asterisk TrevMeister (~thammonds@ip68-4-223-70.oc.oc.cox.net) |
| 09:00.21 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 09:02.08 | *** join/#asterisk djin (~djin@62.58.40.196) |
| 09:04.57 | *** join/#asterisk heka (~fasada@82.114.68.126) |
| 09:06.03 | *** join/#asterisk Savage-S (~savage@c514701e0.cable.wanadoo.nl) |
| 09:06.57 | Zeeek | good morning quand même |
| 09:11.15 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 09:12.20 | *** join/#asterisk Jas_Williams (~Jason@host81-155-66-178.range81-155.btcentralplus.com) |
| 09:14.50 | *** join/#asterisk bassie (~bas@datarack.xs4all.nl) |
| 09:15.01 | bassie | 'morning |
| 09:15.37 | bassie | any european or dutch guys present that use asterisk on ISDN? |
| 09:17.11 | idnar | tzafrir: hmm... response lag ;) |
| 09:17.22 | Jas_Williams | Yes... |
| 09:17.25 | Jas_Williams | Uk |
| 09:17.56 | idnar | tzafrir: by haywire, I mean that audio I get from the channels is garbled; I'll get noise on top of the actual audio, or the channel will just be dead, that sort of thing |
| 09:18.10 | bassie | @jas_williams: what kind of gateway do you use? |
| 09:18.12 | idnar | tzafrir: reloading the module (after stopping asterisk) fixes the problem, but then after running for another day or two, it'll usually go wrong |
| 09:18.39 | Jas_Williams | Native Asterisk 4 port PRI cards |
| 09:19.02 | idnar | tzafrir: I'm using 2.4.26 with zaptel 1.0.2 from debian sid |
| 09:19.41 | tzafrir | first of all, there should be zaptel 1.0.4 in Debian SID |
| 09:20.14 | tzafrir | you can find may take on the packages at http://tzafrir.org.il/rapid/APT.html |
| 09:20.26 | idnar | I'm building 1.0.4 modules as we speak |
| 09:20.54 | tzafrir | I have 1.0.6 over there, if you want to test |
| 09:21.08 | tzafrir | It includes the bristuff package. |
| 09:21.09 | idnar | yeah, the problem is that testing is such a lengthy procedure |
| 09:21.43 | bassie | ah ok, I'm busy with a cisco 1720 router |
| 09:21.53 | bassie | and a ISDN BRI i/f |
| 09:22.08 | bassie | but I guess I'll buy an ISDN card today :) |
| 09:22.17 | tzafrir | How do you build the zaptel module? do you use make-kpkg? |
| 09:22.21 | idnar | yes |
| 09:22.57 | fitzel | Is anybody using (sucessfully) a datafire 4port isdn card? |
| 09:22.59 | tzafrir | If you have an existing and configured kernel tree it shouldn't take long |
| 09:23.13 | idnar | tzafrir: oh, building it doesn't take long; but the problem usually takes more than a day to manifest |
| 09:23.31 | *** join/#asterisk Stonekeeper (~Stonekeep@195.173.212.2) |
| 09:23.33 | *** part/#asterisk pashah (~pashah@relay.patentica.com) |
| 09:24.29 | tzafrir | Have you tried earlier to just restart asterisk without unloading the zaptel module? (just trying to verify that it is not an * issue) |
| 09:26.19 | fitzel | bassie, I am using it on isdn, too. Unfortunately, I tried to compile a new version (I was on 0.7) and I am not able at all to get chan_capi running again. |
| 09:28.05 | idnar | tzafrir: I haven't tried it for a while, but the first few times I ran into this problem (I've basically been having it since I started using Asterisk a few weeks ago) I did try restarting just Asterisk |
| 09:28.54 | idnar | tzafrir: I've actually only had the "horrible noise" problem once (and I didn't try restarting just Asterisk that time), the other few times the card just wasn't responding to incoming calls, and picking up a channel just resulted in slightly noisy silence |
| 09:29.00 | idnar | well, not "few" |
| 09:29.09 | *** join/#asterisk Fishwaldo (~Fish@bb219-74-243-126.singnet.com.sg) |
| 09:29.13 | Jas_Williams | bassie: Are you looking at doing bri or pri |
| 09:29.59 | Jas_Williams | fitzel: Which version of * are you using |
| 09:30.30 | fitzel | I tried the latest cvs and the stable version. |
| 09:31.00 | fitzel | Both was not sucessful. I was able to compile the chan_capi 0.3.5 |
| 09:31.00 | Fishwaldo | does anyone know if its possible to get asterisk to continue through priorities even if a app return -1?. I'm trying to write some scripts around VoiceMail, if the user uses # at the end of his message, everything goes fine, but if they hang up, then it all stops! |
| 09:31.02 | Jas_Williams | with cvs head there is a patch to apply to chan_capi_3.5.0 let me find a link |
| 09:31.22 | fitzel | Ah, that could be helpful. Thank you. |
| 09:31.22 | *** join/#asterisk afe ([BvwPCemLJ@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 09:31.43 | Jas_Williams | http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2 |
| 09:31.46 | fitzel | At the moment, I did a workaround with chan_modem but that is not the same |
| 09:32.38 | Jas_Williams | Or I can tar you up my version for cvs head I changed another couple of calls that have become outdated |
| 09:33.10 | *** join/#asterisk amir2 (~amir@shield.guindehi.ch) |
| 09:33.28 | Jas_Williams | regarding the way config fles are read |
| 09:33.55 | fitzel | I try that first, just a moment. |
| 09:36.13 | fitzel | Hm, I still get: structure has no member named `delivery' during chan_capi compilation |
| 09:36.53 | Jas_Williams | Have you updated libpri |
| 09:37.11 | fitzel | I think, but I am not sure which one is now in use here. |
| 09:38.57 | fitzel | Maybe I should anyway update the whole machine where its running on. Its now up since 370 days. |
| 09:40.06 | Jas_Williams | Yes update zaptel libpri and asterisk remake src and then make clean make chan_capi |
| 09:40.47 | *** join/#asterisk pashah (~pashah@relay.patentica.com) |
| 09:41.03 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 09:44.26 | *** join/#asterisk jeofrey (~jeofrey@espeed19-153.brunet.bn) |
| 09:44.31 | jeofrey | hi all |
| 09:44.48 | jeofrey | anyone can help me please |
| 09:44.55 | jeofrey | i install asterisk |
| 09:45.07 | jeofrey | how i can connect the quintum to asterisk |
| 09:45.25 | jeofrey | how i setup the oh323 to accept connection from quintum |
| 09:45.32 | jeofrey | h323 protocol |
| 09:48.12 | *** join/#asterisk Red_6 (~alex@m174.net81-66-29.noos.fr) |
| 09:48.39 | jeofrey | hello |
| 09:48.44 | jeofrey | help please |
| 09:48.53 | jeofrey | ;( |
| 09:49.28 | jeofrey | <PROTECTED> |
| 09:49.29 | jeofrey | xten 's' |
| 09:49.29 | jeofrey | <PROTECTED> |
| 09:49.29 | jeofrey | <PROTECTED> |
| 09:49.29 | jeofrey | <PROTECTED> |
| 09:49.29 | jeofrey | <PROTECTED> |
| 09:49.31 | jeofrey | <PROTECTED> |
| 09:49.33 | jeofrey | <PROTECTED> |
| 09:49.35 | jeofrey | ocal user) |
| 09:49.51 | jeofrey | i always got that meesege |
| 09:49.53 | Zeeek | jeofrey you'll want ot use the pastebin for that |
| 09:49.55 | jeofrey | messege |
| 09:49.58 | Zeeek | ~pastebin |
| 09:49.59 | jbot | [pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 09:50.01 | jeofrey | ok ok |
| 09:50.08 | Zeeek | just so you know |
| 09:50.12 | jeofrey | sorry |
| 09:50.19 | Zeeek | np |
| 09:50.45 | bassie | anybody using a cisco router as a pstn gateway here? |
| 09:50.51 | Jas_Williams | jeofrey: looks like you do not have an extension to handle 0036738829118 in the default context |
| 09:51.00 | jeofrey | http://www.pastebin.com/248742 |
| 09:51.06 | jeofrey | i have |
| 09:51.24 | jeofrey | wait i will paste pastebin |
| 09:53.07 | jeofrey | ahhhh ok ok |
| 09:53.11 | jeofrey | i will try again |
| 09:54.21 | bassie | someone willing to share their pstn config in extensions.cnf? |
| 09:55.47 | Jas_Williams | bassie: I all depends where in the world you are but there are loads availiable on the web |
| 09:56.05 | Jas_Williams | opps it |
| 09:56.37 | bassie | the info I'm looking for is the pattern that makes the call outbound to the gateway |
| 09:57.33 | Jas_Williams | How are you connected to your gateway sip h323 iax ? |
| 09:59.40 | bassie | sip |
| 09:59.50 | bassie | it's a cisco 1720 series |
| 09:59.59 | bassie | it's registered in Asterisk |
| 10:00.13 | bassie | sip show users shows the gateway |
| 10:01.04 | bassie | pardon, it's a 1750 |
| 10:01.10 | bassie | :) |
| 10:01.40 | Jas_Williams | Do you whant your outbound calls prefixed by a number eg 9 ? |
| 10:01.46 | bassie | yes |
| 10:01.47 | Zeeek | Hi Jas |
| 10:02.17 | Jas_Williams | morning Zeeek fixed your voice quality issues for the farfon ? |
| 10:03.01 | Jas_Williams | bassie: Can you post your current extensions.conf to pastebin |
| 10:03.40 | Jas_Williams | Is the 1750 registered as a use peer or friend |
| 10:03.45 | *** join/#asterisk RaYmAn-Bx (user@213.237.12.147.adsl.vby.tiscali.dk) |
| 10:03.45 | Jas_Williams | user ? |
| 10:04.39 | bassie | @Jas, sure mom |
| 10:05.15 | Zeeek | Jas no I'm waiting for word from Farfon - sudden radio silence fromthem |
| 10:05.35 | Zeeek | but (thanks to you) at least the phone works in both directions now |
| 10:06.27 | bassie | Jas, check post 6766 |
| 10:07.27 | bassie | the prefix # we want to use is 0 btw |
| 10:08.17 | *** join/#asterisk shaZwaz (~chatzilla@203.81.196.167) |
| 10:09.22 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 10:09.32 | Jas_Williams | this line is wrong exten => 0X,1,Dial(SIP/gateway,20,Ttr) |
| 10:10.28 | bassie | ok, what should it be? |
| 10:10.42 | bassie | :) |
| 10:10.56 | Zeeek | ~lart RoyK |
| 10:11.04 | jeofrey | http://www.pastebin.com/248751 |
| 10:11.18 | jeofrey | i have this extensions.conf setup |
| 10:11.26 | bassie | this is _the_ most important thing on our list now, internal calls/transfers/music on hold etc works |
| 10:11.39 | bassie | now we want to call outside the building :) |
| 10:11.45 | *** join/#asterisk Red_6 (~alex@m174.net81-66-29.noos.fr) |
| 10:11.52 | jeofrey | but if i call from quintum cannot hit the context that i assign for quintum |
| 10:12.35 | RoyK | ~kill Zeeek |
| 10:12.37 | jbot | ACTION shoots a magneto-ionized neutron gun at Zeeek |
| 10:13.18 | Zeeek | RoyK you're testing farfon? |
| 10:13.43 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 10:13.59 | jeofrey | http://www.pastebin.com/248752 |
| 10:14.45 | Jas_Williams | it should read something like exten => 0X,1,Dial(SIP/${EXTEN:1}@gateway,20,Ttr) |
| 10:14.54 | Jas_Williams | bassie it should read something like exten => 0X,1,Dial(SIP/${EXTEN:1}@gateway,20,Ttr) |
| 10:15.06 | bassie | thanks |
| 10:15.18 | Jas_Williams | opps also you need some more |
| 10:15.19 | bassie | but ${EXTEN} looks like a variable? |
| 10:15.39 | Jas_Williams | exten => _0X.,1,Dial(SIP/${EXTEN:1}@gateway,20,Ttr) |
| 10:15.55 | *** join/#asterisk meppl (~mephisto@pD9E6861E.dip.t-dialin.net) |
| 10:15.59 | meppl | guten morgen |
| 10:16.05 | Jas_Williams | EXTEN is a variable a system variable for the number you dialled on the phone |
| 10:16.19 | Jas_Williams | the :1 deleats the leading 0 |
| 10:16.57 | bassie | ok |
| 10:17.22 | RoyK | ~seen wasim |
| 10:17.23 | jbot | wasim <~wasim@203.81.213.118> was last seen on IRC in channel #asterisk, 2d 20h 42m 6s ago, saying: 'yay! fresh feta cheese!'. |
| 10:17.27 | RoyK | argh |
| 10:17.31 | RoyK | where's that fscking paki |
| 10:17.39 | jeofrey | anyone help me please to this problem http://www.pastebin.com/248752 |
| 10:17.40 | Jas_Williams | That cheese was bad then |
| 10:18.10 | shaZwaz | Hi Jas , Zeeek , RoyK |
| 10:18.23 | Zeeek | RoyK are you testing these fones too? |
| 10:18.35 | Zeeek | looks like you had the same problem |
| 10:18.37 | bassie | @Jas, thanks, we made progress |
| 10:19.08 | Jas_Williams | jeofrey: You need to change the context of your h323 endpoint to quintum rather than default |
| 10:19.17 | bassie | we can dial to the outside world, the phone says it's ringing, I hear it ringing, but the called phone does not... |
| 10:19.27 | Jas_Williams | OK |
| 10:20.01 | RoyK | Zeeek: yep |
| 10:20.03 | Jas_Williams | do an isdn debug on the router and see what is going on there |
| 10:20.13 | RoyK | Zeeek: oh. the noise? bug 13? |
| 10:20.17 | Zeeek | whatd'ya think RoyK? |
| 10:21.15 | bassie | @jas, we're already logging in :) |
| 10:21.33 | Zeeek | I think that noise is a hardware issue |
| 10:21.58 | Zeeek | RoyK if you have a minute I'd like to compare notes? #farfon or proivate |
| 10:23.42 | modulus_ | hi |
| 10:25.50 | *** join/#asterisk Mother_ (~m@53.Red-217-126-93.pooles.rima-tde.net) |
| 10:25.53 | Mother_ | hi all |
| 10:26.06 | Zeeek | how ah yuh mutha? |
| 10:26.46 | Mother_ | alwrigt thankya |
| 10:26.53 | Mother_ | :) |
| 10:26.58 | Zeeek | kewl |
| 10:27.03 | Mother_ | got the double-ring thing sorted eventually |
| 10:27.27 | Zeeek | the secret decoder ring? |
| 10:27.32 | Mother_ | for some reason, the IAX2 call was getting stuck on the hangup extension when it ended, and the SIP phones all started ringing again |
| 10:27.53 | bassie | Jas, say I'd like to use H323 between the router and asterisk, I'd just replace SIP with H323? |
| 10:28.06 | Mother_ | so it was doing s,1,Dial... once, then h,1,Dial... when either the remote IAX or the local SIP phone hangup |
| 10:28.19 | Zeeek | and the reason was? |
| 10:28.50 | Mother_ | I have no idea, but since these people are in use of the PBX, I cannot experiment much other than on a sunday, so it'll have to wait |
| 10:29.06 | Zeeek | i c |
| 10:29.16 | Jas_Williams | bassie: Stick with SIP but in theory yes build an h323 channel and change the dial line to suit |
| 10:29.48 | Mother_ | in the extensions.conf context, there was an entry to playback an unavailable message when nobody answered, removing this killed the double ring |
| 10:29.55 | Mother_ | so it's really really spooky |
| 10:29.58 | Jas_Williams | Mother_: sounds like you have a catch all that is doing it some thing like _. |
| 10:30.13 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 10:30.25 | Mother_ | Jas_Williams: yes, that's true, the context also had _., and I changed it to s |
| 10:30.51 | Mother_ | so either of the two was causing the problem, I'll have to skip the bbq on sunday and come down to investigate |
| 10:31.53 | *** join/#asterisk yaboo (~jsirucka@220.245.131.131) |
| 10:32.07 | Mother_ | and now the telco disables callerid on the pstn lines when they are configured to jump...so a new battle |
| 10:33.11 | Zeeek | configured to jump? |
| 10:33.15 | Zeeek | eh? |
| 10:33.24 | yaboo | hello, anyone able to tell me if any linux clients can use usb handsets |
| 10:33.40 | Jas_Williams | Mother_: change it to _X. then it will not catch h nor s |
| 10:34.10 | Mother_ | Jas_Williams: thanks for the tip, I'll try that |
| 10:34.27 | Zeeek | interesting Jas - never occurred to me that ._ trap! |
| 10:35.02 | Mother_ | Zeeek: I'm not sure of the english name for it, but basically it's when you have a header number, and if it's busy, the switch calls other lines you have coming in to the PBX |
| 10:35.25 | phaded | hunting |
| 10:35.34 | Mother_ | i.e. 12345 is the number people call you on, and if it's busy, the switch sends it to 23456 which is a second line you have, and so on |
| 10:35.50 | Mother_ | hunting then :) |
| 10:39.06 | Mother_ | question: do I define pickupgroups in sip.conf? i.e. pickupgroup=x for each SIP phone I want on the group? |
| 10:39.46 | Mother_ | or is pickupgroup only for zaptel interfaces? |
| 10:41.10 | Zeeek | i believe that's the case |
| 10:41.53 | Zeeek | I hear it won't workacross different technologies in any case |
| 10:42.17 | Zeeek | I kow it doesn't when I tried to pickup a call to SIP from the IAX |
| 10:44.07 | Mother_ | hmmmmkay |
| 10:44.29 | Jas_Williams | *8 pickup works fine under SIP |
| 10:44.42 | Mother_ | so, the problem now is that if a SIP phone is busy, then hangs up, it doesn't ring if there is a call already ringing on other SIP phones that weren't busy when the call started |
| 10:44.45 | *** join/#asterisk Nebukadneza (~daddel9@i3ED6E583.versanet.de) |
| 10:44.46 | Nebukadneza | hi |
| 10:45.04 | Mother_ | so just like that, *8 will pickup in any case, or do I have to specify callgroups? |
| 10:45.23 | Jas_Williams | Mother_: You need to have all of the phones in the callgroup |
| 10:45.54 | Zeeek | I don't think that'll work |
| 10:46.11 | Nebukadneza | would MP#Player(http://any.web.stream/) work? |
| 10:46.14 | Mother_ | Jas_Williams: OK, I'll try adding that in sip.conf |
| 10:46.15 | Zeeek | I think when the phone is busy, it won't start ringing until a new call comes in |
| 10:46.35 | Zeeek | I was told this is normal pbx behavior |
| 10:46.37 | Mother_ | Zeeek: that sucks then, because sometimes one of the workers is alone in the office |
| 10:46.55 | Zeeek | Of course, I do understand - workaround = pickup groups |
| 10:46.57 | Mother_ | there is something called late entry I remember on some PBXs |
| 10:47.41 | Mother_ | on this really cheapo made-in-Taiwan PBX it would ring the extensions when they hangup with no extra configuration |
| 10:47.59 | Zeeek | disassemble the code :) |
| 10:49.49 | Zeeek | Jas_ do you have a tftp server running all the time? |
| 10:49.49 | Mother_ | lol |
| 10:49.56 | Mother_ | I don't think it even has code :D |
| 10:49.59 | Mother_ | j/k |
| 10:50.06 | Zeeek | it has wires |
| 10:50.09 | Mother_ | it's so crappy |
| 10:50.13 | Mother_ | lol |
| 10:50.26 | Mother_ | I could start a crappy PBX museum |
| 10:50.39 | Zeeek | at least you shpould take some photos |
| 10:50.45 | Mother_ | yeah I will |
| 10:50.58 | Mother_ | at least the wiring is not that badly done |
| 10:51.26 | Zeeek | inclidentally (OT) heres the weather yesterday and today in Paris: |
| 10:51.27 | Zeeek | http://www.myspeakerscorner.com/forum/user_images/winter2.jpg |
| 10:52.03 | Zeeek | I like the optimism of the sidewalk café chairs and tables |
| 10:52.19 | Zeeek | no one is sipping a Pernod though |
| 10:55.02 | *** join/#asterisk [ro]nic3try (~iancu@81.181.199.39) |
| 11:00.40 | RoyK | http://gfx.dagbladet.no/pub/artikkel/4/42/425/425085/kaldt4_858.jpg |
| 11:00.50 | Mother_ | that weather sucks |
| 11:00.55 | Mother_ | you in Paris now? |
| 11:02.18 | Zeeek | yes |
| 11:02.31 | Zeeek | it's about 2°C here and still snowing |
| 11:02.44 | Mother_ | whoa |
| 11:02.44 | Zeeek | the second photo is RoyK |
| 11:02.56 | Mother_ | in Barcelona it was snowing a couple of days ago, but it's better now |
| 11:02.57 | RoyK | it's around -10 here with a clear blue sky |
| 11:03.14 | Zeeek | I hear it snowed a lot in Portugal |
| 11:03.53 | Mother_ | yeah, same here in some places |
| 11:04.00 | Mother_ | I was snowed in monday morning |
| 11:04.23 | RoyK | Mother_: where? |
| 11:04.29 | Mother_ | Barcelona |
| 11:04.45 | Mother_ | near of, some 20km away |
| 11:05.31 | Mother_ | hrmmm this is nice, adding the pickupgroups now causes the SIP phones to ring all by themselves when hanged up, no need for *8.... |
| 11:05.48 | Zeeek | Mother, explain fully again? |
| 11:06.03 | Zeeek | all phones are SIP? |
| 11:06.08 | Mother_ | yes |
| 11:06.38 | Zeeek | so pickupgroups make the phones ring after they are hung up on a incoming call in the process of ringing? |
| 11:06.56 | Mother_ | I added callgroup=1 to zapata and pickupgroup=1 to each entry in sip.conf |
| 11:07.11 | Mother_ | Zeeek: it would appear so, I'll fully test it a few more times to be sure |
| 11:11.24 | Mother_ | this is strange - I have touched nothing else, and now it appears that it's doing it via call waiting, on the busy SIP phone, the second call appears, and starts ringing when it hangs up |
| 11:12.07 | Mother_ | and I haven't touched any call waiting setting on either phones nor * |
| 11:14.42 | Nebukadneza | is there a way to let asterisk play webradio (as suggested in http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf) without letting asterisk play the webradio interally the whole time? |
| 11:15.37 | Inv_arp | Nebukadneza: agi? |
| 11:15.50 | Nebukadneza | huh? what do you mean by agi? |
| 11:16.00 | Nebukadneza | ah kay -.- |
| 11:17.07 | Nebukadneza | is the python interface more or less stable? |
| 11:18.13 | Inv_arp | Nebukadneza: i use php... doesnt really matter since communication is done by stdin/out |
| 11:19.25 | Nebukadneza | hm ... |
| 11:19.58 | Nebukadneza | so i just send the raw audio data to stdout from my script? |
| 11:21.33 | *** join/#asterisk meppl (~mephisto@pD9E6861E.dip.t-dialin.net) |
| 11:23.38 | Nebukadneza | hm arghs |
| 11:24.06 | Nebukadneza | how can i send audio data from a agi to asterisk (let the agi play ... in this case the stream) |
| 11:26.26 | Nebukadneza | ;ah found it |
| 11:29.41 | *** join/#asterisk lespiggot (~les@217.206.141.131) |
| 11:31.37 | *** join/#asterisk Gronker (~Gronker2@adsl-220-79-68.ags.bellsouth.net) |
| 11:32.13 | lespiggot | Hi, I've just installed on openwrt but the SIP support has been disabled as it cannot find the IP address of the box. in my sip.conf externalip has been set as well as localnet variables. Any suggestions I can try? |
| 11:32.43 | lespiggot | forgot to ay its asterisk version 1.0.5 |
| 11:33.43 | *** join/#asterisk Martohtar (Martohtar@82.196.218.130) |
| 11:36.25 | *** join/#asterisk pinhead (pinhead@pinhead.hitv.ru) |
| 11:44.25 | ta[i]nted | what port(s) do incoming SIP calls use? 5060? |
| 11:46.37 | moonwick | yes |
| 11:47.16 | ta[i]nted | hmm |
| 11:47.25 | ta[i]nted | i don't understand why outgoing is working but not incoming |
| 11:47.37 | ta[i]nted | i'm on external ip with 5060 open... |
| 11:48.14 | Nebukadneza | argh! |
| 11:48.26 | Nebukadneza | is there a way to send audio data to asterisk via AGI? |
| 11:49.53 | fitzel | can I supress the "attempt native bridging" for SIP, too? I did it for iax already. |
| 11:50.22 | *** join/#asterisk amir (~amir@shield.guindehi.ch) |
| 11:52.56 | jeofrey | anyone can help me please |
| 11:54.08 | jeofrey | what port do i have to open in firewall for sip and h323..... |
| 11:54.08 | jeofrey | i have try to open the 5060 and 1720 |
| 11:54.34 | jeofrey | but the other party cannot hear me |
| 11:54.34 | jeofrey | but i can heard them |
| 11:54.47 | jeofrey | i open also the rtp 10000 to 20000 |
| 11:55.11 | Zeeek | is the other end behind NAT? |
| 11:55.24 | Zeeek | they may need to forward |
| 11:55.33 | jeofrey | no |
| 11:55.48 | jeofrey | the other end is using public ip |
| 11:56.50 | jeofrey | but it i put the asterisk in public ip its working both side can communicate |
| 11:59.28 | jeofrey | any idea that can share about this problem |
| 12:00.33 | *** join/#asterisk mozrat (~sm@194.203.40.131) |
| 12:00.35 | roamer323 | joefrey - is the other side able to call you (i.e. will your phone ring?) |
| 12:00.46 | jeofrey | nooo |
| 12:00.55 | jeofrey | the other side cannot call me |
| 12:01.19 | roamer323 | jeofreg - but you can call out and get voice 1 side |
| 12:01.24 | jeofrey | yess |
| 12:02.07 | roamer323 | jeofrey - is it a home router you're going through? Linksys/D-link etc? |
| 12:02.25 | jeofrey | im using linksys |
| 12:03.06 | jeofrey | wrt54gs |
| 12:03.36 | jeofrey | is this kind of router cannot support the voip ?? |
| 12:04.10 | Zeeek | jeofrey I have one of those |
| 12:04.17 | Zeeek | works great |
| 12:04.26 | roamer323 | double and triple check the port mapping - and turn on debug to make sure all the UDP packets are getting through - sounds like the ports are not mapped right |
| 12:04.39 | Zeeek | you say you "opened" ports - what d'you mean? |
| 12:04.51 | Zeeek | you want to *forward* ports |
| 12:04.57 | jeofrey | forward |
| 12:05.09 | Zeeek | 5060 -> asterisk |
| 12:05.19 | Zeeek | 100000-10100 -> asterisk |
| 12:05.31 | Zeeek | 4569 -> asterisk if you use IAX2 |
| 12:05.36 | jeofrey | the asterisk is inside the nat using 192.168.1.1 |
| 12:05.48 | Zeeek | no it isn't - the router uses that ip |
| 12:06.02 | Zeeek | oh you mean asterisk is running in the router softwar? |
| 12:06.09 | jeofrey | yessssss |
| 12:06.13 | roamer323 | hehe - Zeeek - you're onto him :-) over and out. |
| 12:06.18 | Zeeek | ok |
| 12:06.26 | jeofrey | thanks reoamer |
| 12:06.33 | Zeeek | in that case you'll have to find someone that's done that. I haven't |
| 12:06.43 | jeofrey | ohmmmm |
| 12:06.58 | jeofrey | or if i use the asterisk in public ip |
| 12:07.21 | jeofrey | then i want to enable the firewall what are the ports that i have to open |
| 12:07.35 | jeofrey | then how i do that because im new in linux |
| 12:09.49 | tzafrir | for what protocol? SIP/ IAX? |
| 12:09.58 | jeofrey | sip |
| 12:11.42 | jeofrey | ok ok |
| 12:11.56 | jeofrey | i think no one yet do that setup here |
| 12:12.36 | jeofrey | just forget about that nat setup problem |
| 12:12.38 | RoyK | fuck |
| 12:12.49 | RoyK | seems asterisk doesn't read the queues.conf on reaload |
| 12:13.17 | jeofrey | just help me how to connect the quintum using h323 to asterisk |
| 12:13.32 | jeofrey | http://www.pastebin.com/248752 |
| 12:15.11 | RoyK | h323 support in asterisk is really shitty |
| 12:16.11 | Nebukadneza | re |
| 12:17.36 | jeofrey | ahhhhhhhh |
| 12:17.47 | jeofrey | no one want to answer me ;( |
| 12:18.13 | jeofrey | they are all bussy |
| 12:18.46 | RoyK | jeofrey: as I said, h323 support in asterisk is really shitty |
| 12:19.07 | jeofrey | heheheh |
| 12:19.08 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 12:19.26 | *** join/#asterisk Astinus_ (~abba@213.167.111.138) |
| 12:19.39 | *** part/#asterisk Astinus_ (~abba@213.167.111.138) |
| 12:19.47 | Nebukadneza | hm ... is there a reason why MP3Player(http://64.236.34.67:80/stream/8022) isnt working? |
| 12:20.25 | lespiggot | <jeofrey> I'm currently installing asterisk on my wrt, but sip is down as it can not determine external IP, any suggestions on how to activate it? |
| 12:21.26 | *** join/#asterisk w0w0 (~w0w0@80-28-171-221.adsl.nuria.telefonica-data.net) |
| 12:22.18 | jeofrey | from port range forwarding open the port 5060 then point to internal ip of ur asterisk |
| 12:22.36 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 12:22.49 | jeofrey | hahaha but you can heard the voice of other side then you cannot heard them |
| 12:22.56 | jeofrey | thats my problem |
| 12:22.57 | jeofrey | hahahahaha |
| 12:27.01 | *** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk) |
| 12:28.14 | *** join/#asterisk DEVILoper (~x@202.5.145.50) |
| 12:29.14 | DEVILoper | hi any asterisk programmer here ? |
| 12:29.24 | DEVILoper | or Developer |
| 12:31.24 | Zeeek | you are ! |
| 12:31.40 | jeofrey | ur the deviloper here |
| 12:31.45 | DEVILoper | yeah i am but have to discuss few ideas |
| 12:31.47 | jeofrey | maybe you can help us |
| 12:32.04 | Zeeek | ideaz? whazzzat? |
| 12:32.28 | jeofrey | idea how to solve the nat problem |
| 12:33.23 | ta[i]nted | jeofrey i've got similar issue |
| 12:33.30 | ta[i]nted | i can make outgoing.. but cannot get incoming |
| 12:33.43 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 12:33.57 | puzzled | hi all |
| 12:35.35 | jeofrey | ta <--- we are they same |
| 12:35.37 | jeofrey | heheheheh |
| 12:35.47 | jeofrey | have you manage to solve it??? |
| 12:36.44 | ta[i]nted | no |
| 12:36.46 | Nebukadneza | *grml* |
| 12:36.54 | jeofrey | ;( |
| 12:36.54 | Nebukadneza | is there a way to let asterisk play webstreams (without writing a mpg123 wrapper or MusicOnHold())? |
| 12:37.05 | DEVILoper | <PROTECTED> |
| 12:38.29 | jeofrey | {public ip}--->{linksys}-----> {asterisk} |
| 12:39.45 | puzzled | Nebukadneza: iirc there was a way/patch to use an external music streamer app. check voip-info.org and search for MusicOnHold |
| 12:40.18 | ta[i]nted | jeofrey i'm on public ip :( |
| 12:40.37 | jeofrey | me also |
| 12:40.46 | jeofrey | using public already in asterisk |
| 12:40.47 | jeofrey | hehehehe |
| 12:40.49 | Nebukadneza | checked already - nothing :( |
| 12:40.53 | ta[i]nted | yea.. strange |
| 12:41.00 | ta[i]nted | so u get audio but they can't hear u? |
| 12:41.17 | Mother_ | nat problem? what nat problem? ;) |
| 12:41.19 | ta[i]nted | my incoming calls don't even work |
| 12:41.53 | Nebukadneza | puzzled: musiconhold lets the stream play the whole time |
| 12:41.53 | jeofrey | yahhhhhhhh |
| 12:42.00 | Mother_ | I also have a problem, all my ports are natted OK, sip goes through from a 7960 to * both behind their own NATs, I can call and receive calls, but RTP is only working one way |
| 12:42.07 | jeofrey | you can heard them but they cannot heard you |
| 12:42.16 | Mother_ | ha! exactly my problem |
| 12:42.17 | Nebukadneza | consumes just too much bandwidth for a dsl line |
| 12:42.30 | Zeeek | jeofrey what phone is trying to call your asterisk? |
| 12:42.39 | Zeeek | soft? hard? what type? |
| 12:42.51 | Mother_ | let me guess - Cisco |
| 12:42.51 | jeofrey | normal pstn |
| 12:42.55 | Mother_ | ah ok |
| 12:42.57 | jeofrey | no |
| 12:43.01 | ta[i]nted | same here |
| 12:43.04 | jeofrey | im using grandstream |
| 12:43.06 | ta[i]nted | except i don't get connection |
| 12:43.07 | jeofrey | for sip |
| 12:43.09 | Zeeek | they can't hear you on pstn? |
| 12:43.20 | jeofrey | they cannot heard me in pstn |
| 12:43.24 | Mother_ | same here |
| 12:43.35 | jeofrey | but i can heared them in my grandstream |
| 12:43.35 | Zeeek | what hardware to pstn? |
| 12:43.44 | Mother_ | TDM400P |
| 12:44.01 | Mother_ | (this is multi-user simulproblemsolving) |
| 12:44.11 | Zeeek | where is your grandstream? same side of NAT? |
| 12:44.33 | Zeeek | look if all the ports are forwarded to asterisk, how cxan you GS get the audio RTP? |
| 12:46.19 | jeofrey | noooo |
| 12:46.37 | jeofrey | the grandstream is in other dsl connection |
| 12:46.45 | Mother_ | same here |
| 12:46.58 | jeofrey | hahaha |
| 12:47.11 | Mother_ | Cisco <- NAT DSL -> Internet <- NAT DSL -> Asterisk |
| 12:47.56 | ta[i]nted | jeofrey i thought u said u were on public ip |
| 12:48.18 | jeofrey | now i setup on public ip |
| 12:48.33 | jeofrey | becuase i cannot get thru the voice in linksys |
| 12:48.35 | jeofrey | hahaha |
| 12:48.52 | jeofrey | but we still need to solve this kind of nat issue |
| 12:48.53 | jeofrey | hahaha |
| 12:49.16 | jeofrey | sasme block diagram we have Mother... |
| 12:49.20 | jeofrey | same setup |
| 12:49.23 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 12:49.30 | Mother_ | yep, and same problem too, RTP works one way only |
| 12:49.43 | Mother_ | and I double-checked the mappings and they *should* work |
| 12:49.53 | jeofrey | yessssss |
| 12:50.08 | Zeeek | Attention Target shoppers.... |
| 12:50.15 | Mother_ | today I'm sticking a hub inbetween phone and router and will be sniffing |
| 12:50.19 | jeofrey | actually i open the dmz already |
| 12:50.24 | jeofrey | still theysame problem |
| 12:50.40 | Zeeek | If you ever use X-Lite, when "transmit noise" is set to "no" (default) it results in the person not hearing you |
| 12:50.57 | Mother_ | yes, I was aware of that with X-Lite |
| 12:51.20 | Mother_ | it breaks the RTP stream AFAIK |
| 12:51.30 | jeofrey | i use also xlite same problem with my grandstream |
| 12:51.53 | RoyK | hmmmmmmmm |
| 12:52.04 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 12:52.05 | RoyK | it seems asterisk doesn't reload the queues.conf file on RELOAD |
| 12:52.09 | RoyK | fuck |
| 12:52.32 | Zeeek | restart g<tab> |
| 12:53.16 | RoyK | Zeeek: right.. |
| 12:53.23 | Zeeek | I use X-Lite and GS BT100 on the same NAT router to an asterisk behind another NAT router with no problem |
| 12:53.33 | jeofrey | Mother what router you are using??? |
| 12:53.36 | RoyK | is restart gracefully == restart when convenient? |
| 12:53.36 | Zeeek | X-Lite is set to use 5016 |
| 12:53.39 | Mother_ | reload is very limited, I've seen a few cases already of people pulling their hair because of this |
| 12:53.48 | Zeeek | restart when no one is talking |
| 12:53.51 | *** join/#asterisk sangee (ravi@209.250.129.135) |
| 12:53.52 | Mother_ | jeofrey: on both sides, a USR 9015 |
| 12:53.53 | RoyK | same then |
| 12:54.03 | Zeeek | yeah i tink zo |
| 12:54.12 | RoyK | Zeeek: only this system is busy something like 20 hours a day |
| 12:54.34 | Zeeek | well, then "when convenient" will mean in 2.46 hours :) |
| 12:54.44 | jeofrey | ohmmmmmm |
| 12:54.57 | RoyK | 10 hours, more likely....... |
| 12:55.20 | Zeeek | or you could somehom transfer all channels to a recording "ATTENTION, the pbx has decided it needss to restart, please call each other back." |
| 12:55.29 | Mother_ | LOL |
| 12:55.37 | Zeeek | "or not" |
| 12:55.53 | jeofrey | Zeek what are the ports that you open for NAT in asterisk side?? |
| 12:56.04 | *** join/#asterisk innerweb (~innerweb@pcp0010181839pcs.columbus.in.indy.comcast.net) |
| 12:56.20 | Zeeek | my grandson wrote this up in a blog |
| 12:56.21 | Zeeek | http://willypick.mindsay.com/?entry=10 |
| 12:57.05 | Zeeek | stirred, not shaken..... forwarded, not open |
| 12:57.23 | *** join/#asterisk LarsAC (~chatzilla@134.130.124.227) |
| 12:57.53 | LarsAC | where can I find information about filenames and dir structure of voiceboxes ? |
| 12:58.08 | Zeeek | it's pretty obvious if you take a peek Lars |
| 12:58.16 | Mother_ | maybe the problem is in the router, although in the USR you can forward either TCP, UDP or both, etc. so it's quite convenient - some other routers I've seen don't even allow you to forward to a different internal port than the external port |
| 12:58.56 | Zeeek | <PROTECTED> |
| 12:59.12 | Zeeek | "etc" meaning and so on, not "/etc" |
| 12:59.50 | Mother_ | Zeeek: so you are *not* forwarding any RTP port specifically??? |
| 13:00.00 | Zeeek | who my grandson? |
| 13:00.02 | Mother_ | on the client side? |
| 13:00.07 | Mother_ | well yes, your grandson :D |
| 13:00.17 | Zeeek | yes he did that. Did he forget to document it? |
| 13:00.22 | Zeeek | kids! |
| 13:00.34 | jeofrey | hahahahaha |
| 13:00.39 | Mother_ | lol |
| 13:00.45 | jeofrey | your a very lucky grnad father |
| 13:00.50 | jeofrey | grand |
| 13:00.50 | Zeeek | IIRC he forwards 10000-10050 to one phone and 10051-10100 to the other on the client side |
| 13:00.56 | LarsAC | Zeeek: the number of the caller and the extension called are in the filename ? |
| 13:01.01 | Mother_ | OK |
| 13:01.02 | innerweb | Finally got the X100P *generic clone* to show up. Now if I can only get a channel to appear. (zaptel and zapata are set up - maybe not correctly). |
| 13:01.05 | LarsAC | dont have it up and running, thats why |
| 13:01.06 | Zeeek | but he knows more about SIP than I do (literally!) |
| 13:01.12 | Mother_ | hahaha |
| 13:01.24 | jeofrey | hahahaha |
| 13:01.32 | Zeeek | every time I tell someone double NAT works for me, they say "You're very lucky" |
| 13:01.46 | jeofrey | Mother lets try this setup of the grandson |
| 13:02.03 | Mother_ | the problem with the Cisco is that no matter how much I try it always defaults to something like 16000 to 32767 as the RTP range |
| 13:02.17 | Mother_ | and it always picks a random port, not the first one (duh!) |
| 13:02.23 | Zeeek | I gotta SMS him to get the setup documented right! |
| 13:02.29 | Zeeek | when he gets a phone |
| 13:02.32 | Mother_ | lol |
| 13:02.40 | Zeeek | which will prolly be in a year |
| 13:03.12 | jeofrey | hahahaha |
| 13:03.37 | jeofrey | i will be on silent mode |
| 13:04.00 | jeofrey | i will try to play with the setup of Zeek grandson |
| 13:04.05 | jeofrey | hahahhaa |
| 13:04.06 | *** join/#asterisk garfield13 (~dmorris@nsc69.38.48-79.newsouth.net) |
| 13:04.54 | garfield13 | New to IRC. I have a question about 3-way calling. Can someone out there help me? |
| 13:05.39 | Mother_ | are the other two parties involved in the 3-way call here too? |
| 13:06.31 | Mother_ | <legpull mode=on> |
| 13:06.58 | innerweb | John, odie? |
| 13:07.22 | *** join/#asterisk BuzzBud (~adoroar@90.62-97-254.bkkb.no) |
| 13:07.22 | garfield13 | No. It is an asterisk configuration question. |
| 13:07.58 | garfield13 | Here is what I am doing,,,, step by step..... |
| 13:08.11 | garfield13 | I use my IP phone to call my cell phone. |
| 13:08.18 | garfield13 | I answer my cell phone. |
| 13:08.33 | garfield13 | I then flash the ip phone and call my other cell phone. |
| 13:08.39 | garfield13 | I answer that cell phone. |
| 13:09.23 | garfield13 | I click the tranfer button on the ip phone and all 3 people are talking to each other (cell 1, cell 2, and the ip phone). |
| 13:09.34 | garfield13 | This is where my question comes into play. |
| 13:10.06 | jeofrey | conference call |
| 13:10.29 | garfield13 | Apparently, through the steps I just defined, asterick is actually using 2 different external lines to call each cell phone. |
| 13:11.08 | garfield13 | (FYI, we have many external phone lines here). |
| 13:11.10 | *** join/#asterisk shaZwaz (~adnans@203.81.196.167) |
| 13:11.27 | Zeeek | Mother_ "Donner, party of 50, Donner" |
| 13:11.42 | garfield13 | What I want to happen is when I do this, I only want to use one phone line because our external provide has given us 3-way calling |
| 13:11.47 | garfield13 | capability on each line. |
| 13:12.14 | garfield13 | How do I tell asterick to let the telco handle the 3-way calling instead of asterisk doing it itself? |
| 13:12.16 | Zeeek | garfield13 what PSTN harware connection? |
| 13:14.46 | garfield13 | I am trying to find out. We have a T1 through a provider called NewSouth or Nuvox |
| 13:15.11 | Mother_ | hehehe |
| 13:15.21 | innerweb | If someone would not mind, can someone please tell me what the simplest channel setup for a X100P card would be? |
| 13:15.59 | garfield13 | Digium T1 card |
| 13:17.08 | Zeeek | innerweb |
| 13:17.11 | Zeeek | Starter tutorial: |
| 13:17.11 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 13:17.11 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 13:17.11 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 13:17.11 | Zeeek | THE reference of the moment: |
| 13:17.12 | Zeeek | http://www.asteriskdocs.org |
| 13:17.15 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 13:17.21 | Zeeek | innerweb read the second link above |
| 13:17.56 | innerweb | I have been through all of that, and I have copied the files given in those. I keep getting ztcfg with no channel. |
| 13:18.09 | Zeeek | garfield I think you need to flash the phone line and I believe you can do this with a T1 - check the mailing list and/or wiki |
| 13:18.11 | innerweb | THe card is seen in dmesg, modules are loaded (lsmod) |
| 13:18.25 | Zeeek | innerweb did you install the zaptel drivers? |
| 13:19.01 | Zeeek | pastebin you zaptel.conf |
| 13:19.05 | innerweb | I have this for now: zaptel.conf loadzone=us; defaultzone=us; fxsks=1; |
| 13:19.23 | jeofrey | conference call |
| 13:19.23 | garfield13 | Zeeek, the reason I know this is not working is that the next step is that I want to transfer both cell phones to a meet-me conference. |
| 13:19.37 | innerweb | lsmod showswcfxo 12640 0 |
| 13:19.37 | innerweb | zaptel 222180 1 wcfxo |
| 13:19.37 | innerweb | crc_ccitt 1920 1 zaptel |
| 13:19.40 | garfield13 | When I do the transfer now, it only moves one. |
| 13:19.51 | Zeeek | inner looks right |
| 13:20.04 | garfield13 | The other is still camped on my phone and I have to transfer it separately. |
| 13:20.09 | innerweb | That is what I though ;-) Unfortunately, * does not agree with me. |
| 13:20.13 | Zeeek | and ztcfg -vvvvv |
| 13:20.24 | Zeeek | pastebin |
| 13:20.26 | innerweb | 0 channels configured. |
| 13:20.28 | Zeeek | not in line |
| 13:20.31 | Zeeek | ok |
| 13:20.33 | Zeeek | clear enuf |
| 13:20.36 | garfield13 | Ultimately, I want to move the two callers simultaneously since, according to the telco, it is just one inbound call. |
| 13:21.09 | innerweb | garfield - We've come a long way since the original quorum board for meetings. |
| 13:21.32 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 13:21.46 | Zeeek | innerweb you sure there are no issues between versions and locations of zaptel and * and all that ? |
| 13:21.47 | innerweb | Does anyone have any ideas beyond the docs of what to look for? I am not used to be being stumped like this on a comp. |
| 13:22.29 | innerweb | Zeeek - No I am not sure. I have recompiled everything under 2.6.10, so the modules and the kernel use the came base and compiler. |
| 13:22.31 | Zeeek | I've never heard of ztcfg not working when the drivers appear to be installed right - but then I know little |
| 13:22.44 | innerweb | You know more than me about * my friend. |
| 13:22.44 | Zeeek | 2.6 - why didn't you say so right away? |
| 13:23.06 | innerweb | I did not know there was another issue with it. Many I have talked to using it have reported no problems. |
| 13:23.07 | shaZwaz | its compiles differently under 2.6 |
| 13:23.16 | innerweb | Ohh. Someone left that out ... ;-) |
| 13:23.30 | innerweb | are you talkeing aobut the make linux26? |
| 13:23.38 | shaZwaz | yup |
| 13:23.43 | innerweb | got that 8-) |
| 13:23.59 | shaZwaz | see the wiki page on compiling zaptel under 2.6 for more |
| 13:24.06 | innerweb | I have a nice fresh baked cherry pie for the seargent if hw has any useful information. |
| 13:24.16 | Zeeek | se the wiki page on comp^H^H^H^H^H^H^H^H^H^H |
| 13:24.48 | Zeeek | wow inner, you're either over 25 or in an area where syndication is still in effect :) |
| 13:25.11 | innerweb | lol... Guess which one. |
| 13:25.34 | Zeeek | frankly yopu seem to exhibit the stability of an older person |
| 13:25.38 | garfield13 | zeek, can you see a reason that I cannot do what I want to? |
| 13:25.52 | shaZwaz | fate |
| 13:25.55 | innerweb | Funny, I keep falling off my chair. I would hardly call that stable. |
| 13:25.56 | Zeeek | I'd put your age at 25.1 to 34 |
| 13:26.09 | Zeeek | +/- 20% |
| 13:26.27 | shaZwaz | how about 25 |
| 13:26.30 | innerweb | Very close, but I score on the high end of that range (including the margin) |
| 13:26.47 | Zeeek | yeah well the falling off chair is good for another +20% |
| 13:26.53 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 13:26.54 | Mother_ | I had many issues with 2.6 kernels, now I run CentOS with 2.4.something and I'm very happy indeed :) |
| 13:27.04 | Zeeek | so let's say around 40 |
| 13:27.10 | innerweb | Hmmm... I may have to go back a little in the OS dept. |
| 13:27.13 | Zeeek | +/- 80% |
| 13:27.15 | innerweb | Close. |
| 13:27.29 | innerweb | Just think moon launch |
| 13:27.34 | Zeeek | I'll bet 10 IAXys your younger than I |
| 13:27.37 | Zeeek | am |
| 13:27.46 | bjohnson | I haven't had problems I can blame specifically on a 2.6 kernel except for the Digium S100U .. but I agree that Centos is a nice OS for a server |
| 13:27.46 | Zeeek | 67? |
| 13:27.51 | Zeeek | 70? |
| 13:27.54 | innerweb | 66 |
| 13:27.55 | Mother_ | Zeeek: that's not hard if you have a blogging grandson |
| 13:27.58 | innerweb | DOB = 66 |
| 13:28.13 | Zeeek | Mother_ :) |
| 13:28.17 | shaZwaz | I use 2.6 fine on a RH9 |
| 13:28.18 | garfield13 | Hey, guys, I hate to interrupt this informative repartee, but.... scan WAAAAYYY back to my last question requesting a reply/ |
| 13:28.18 | RoyK | 42! |
| 13:28.24 | Mother_ | so you win 10 IAXys I guess :D |
| 13:28.44 | Mother_ | garfield13: are you completely sure both calls are happening on the same line? |
| 13:28.44 | BoRiS | 28! |
| 13:28.44 | Zeeek | well we're from the same centruy but not the same hakf! |
| 13:28.46 | innerweb | Age is such an incosequential thing once you have your head on straight, and before that, it still does not matter. |
| 13:29.01 | Zeeek | BoRiS this is seniors day, sorry you'll have to leave |
| 13:29.03 | Mother_ | ooops gotta go |
| 13:29.06 | bjohnson | I didn't see the question |
| 13:29.09 | garfield13 | The problem is they are NOT happening on the same line. I want them to be. |
| 13:29.13 | BoRiS | Awwww, damn! |
| 13:29.15 | innerweb | lol Zeee - so long as it si the same codebase, I am not worried. What gets me are these visitors with adiffrent codebase. |
| 13:29.20 | BoRiS | lol |
| 13:29.24 | Zeeek | the problem is HEAD is more rare as you get older, but I digress |
| 13:29.38 | RoyK | asterisk HEAD meets asterisk ARSE |
| 13:29.43 | Zeeek | heh |
| 13:30.00 | innerweb | lol... Normally I do not get to have so much fun figuring tech stuff out. Takes the sting out of being lost. |
| 13:30.20 | garfield13 | Mother, thank you for at least acknowledging me... :) |
| 13:30.21 | Zeeek | This is the #asterisk-comedyChannel |
| 13:30.39 | BoRiS | This is #asterisk35+ |
| 13:30.42 | Zeeek | so I now have Farfon with the correct time |
| 13:30.45 | shaZwaz | ~seen zx8 |
| 13:30.57 | jbot | shaZwaz: i haven't seen 'zx8' |
| 13:30.57 | innerweb | When you are working with bleeding edge, it is good to laugh at the booboos. |
| 13:30.57 | Zeeek | yeah Boris like the teen chat rooms |
| 13:30.57 | Jas_Williams | garfield13: What is connectivity to the provider also what proces do they say yo have to use to d 3 way calling on their line |
| 13:31.13 | shaZwaz | ~seen nc0 |
| 13:31.14 | jbot | shaZwaz: i haven't seen 'nc0' |
| 13:31.41 | Zeeek | ~seen The Light |
| 13:31.42 | jbot | i haven't seen 'the light', Zeeek |
| 13:31.47 | Zeeek | no shit! |
| 13:31.48 | garfield13 | The provider has supplied us a T1 line that we connect into a Digium T1 card. At this site. |
| 13:32.36 | garfield13 | What were are trying to do is prototype a solution that we will use else where that will have a single analog line connected to an FXO card that has 3-way calling just like at your house. |
| 13:33.14 | Jas_Williams | garfield13: have you 3way calling enabled in zapata.conf |
| 13:33.20 | innerweb | Boris may not be used to all of the serious talk in here. |
| 13:33.29 | innerweb | I(f he is like most youngsters. |
| 13:33.59 | garfield13 | Jas-Williams, looking.... |
| 13:34.18 | innerweb | What versions of * are each of you using? |
| 13:34.25 | Jas_Williams | HEAD |
| 13:34.32 | Zeeek | 1.0.3 |
| 13:34.52 | shaZwaz | anyone used 1.0.6 |
| 13:35.01 | shaZwaz | howz it going ? |
| 13:35.09 | garfield13 | jas_Williams, right now it is off. It was on and the same behavior occurred both ways. Should it be on? |
| 13:35.11 | innerweb | I seem to have 1.0.5 |
| 13:35.48 | garfield13 | Jas, OK, it is back on.... |
| 13:36.01 | innerweb | NOt well. I can see my card, dmesg shows my card, * chokes on :chan_zap.c:769 zt_open: Unable to specify channel 1: No such device or address |
| 13:36.20 | Jas_Williams | garfield13: what type of phone are you using ? |
| 13:36.25 | Zeeek | * can't work until ztcfg works |
| 13:36.36 | garfield13 | snom190 |
| 13:36.40 | innerweb | I figured that.. Hmm, sunny day aoutside, rainy day in. |
| 13:37.04 | Zeeek | snowing here! Look for ztcfg on wke/mlist and see if you can come up with something? |
| 13:37.27 | Jas_Williams | garfield13: ah well then there is your problem, 3 way calling using the Zap interface is only availiable to ZAP phones not SIP |
| 13:37.29 | DEVILoper | any idea how to use external modem with asterisk ?? |
| 13:37.46 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 13:38.06 | shaZwaz | Jas works partially with GS phone |
| 13:38.15 | innerweb | zttool even sees it. must be something with my config.. ? |
| 13:38.33 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 13:38.49 | Jas_Williams | innerweb: what is the output from ztcfg -vvvv |
| 13:39.02 | innerweb | Zaptel Configuration |
| 13:39.02 | innerweb | ====================== |
| 13:39.06 | innerweb | Channel map: |
| 13:39.06 | innerweb | <PROTECTED> |
| 13:39.08 | innerweb | 0 channels configured. |
| 13:39.09 | innerweb | <PROTECTED> |
| 13:39.24 | garfield13 | That is the answer I was looking for. If I hook up an analog phone to an iaxy and do the 3-way calling, it should work? |
| 13:39.36 | innerweb | zttoll shows: |
| 13:39.37 | innerweb | Current Alarms: No alarms. │ |
| 13:39.37 | innerweb | <PROTECTED> |
| 13:39.37 | innerweb | <PROTECTED> |
| 13:39.37 | innerweb | <PROTECTED> |
| 13:39.37 | innerweb | <PROTECTED> |
| 13:39.45 | innerweb | <PROTECTED> |
| 13:40.02 | Jas_Williams | garfield13: NO it will not work as you whant using IAX |
| 13:40.31 | Jas_Williams | innerweb: post your zapata.conf to pastebin.ca |
| 13:43.31 | Jas_Williams | garfield13: No you will need some kind of script in your dialplan that you can transfer the call to that will then work out which channel the incomming call is connected to and then send that channel a flash hook and any other commands required by the telco. this is not easy and many people just use the asterisk to handle the calls as the cost is the same but it does use phone lines |
| 13:43.50 | Jas_Williams | in and out of asterisk |
| 13:44.08 | shepherd | huh? |
| 13:45.20 | innerweb | Cool: never used that before -> http://pastebin.ca/6776 |
| 13:45.21 | Darwin35 | grrr I have to figure all the pkkgs needed for * and make a mini install |
| 13:45.30 | *** join/#asterisk carbon60 (~adam@CPE001217cdb3c5-CM000f9fa6ba66.cpe.net.cable.rogers.com) |
| 13:45.33 | innerweb | Thanks Jas |
| 13:45.36 | carbon60 | Morning all. |
| 13:45.43 | garfield13 | jas, problem is, I will only have one. I was thinking, if I plug an analog phone into an FXS port, then will it work? |
| 13:45.50 | Darwin35 | asterisk@home does not like my hardware |
| 13:46.27 | *** join/#asterisk gpowers (~glenn@static-68-162-84-101.phil.east.verizon.net) |
| 13:46.39 | *** part/#asterisk lespiggot (~les@217.206.141.131) |
| 13:48.33 | carbon60 | Anyone using the PGSQL application? |
| 13:48.56 | garfield13 | Still with me, Jas_Williams |
| 13:49.05 | innerweb | carbon60 -> Not until I get out of my buckyball. |
| 13:49.10 | carbon60 | It's pretty nice and simple, but the commas in my queries are getting turned into pipes when sent to the database. |
| 13:49.16 | Jas_Williams | garfield13: Yes it will but you need something like the TDM400P for the Analogue phone |
| 13:49.21 | carbon60 | innerweb: Heh? |
| 13:49.47 | shepherd | or an iaxy |
| 13:50.12 | Jas_Williams | shepherd: an iaxy does not support ZAP 3way calling.... |
| 13:50.18 | innerweb | I am stuck in my own little box until I get a few things worked out (mysql for now), then I get to do a new server with pgsql (server I am bldg next is buckyball). |
| 13:50.31 | carbon60 | Hehe. LOL |
| 13:50.34 | shepherd | are you takling about programmatically? |
| 13:50.44 | shepherd | i can make 3way calls from my iaxy :) |
| 13:50.49 | garfield13 | Jas_Williams, yes, I was going to use one of those with an FXO for the telco and an FXS for the analog phone. |
| 13:50.50 | Jas_Williams | innerweb: you have some duplication in your zapata.conf |
| 13:51.03 | carbon60 | Is the commas-to-pipes thing a standard part of the API? |
| 13:51.06 | innerweb | Ok... I suppose (sounding ignorant) that is bad. lol |
| 13:51.23 | Jas_Williams | garfield13: that should work then but get the hardware and test/build |
| 13:51.33 | garfield13 | Shepherd, the problem is, that requires to external phone line. |
| 13:52.37 | shepherd | i guess i don't understand |
| 13:52.53 | Jas_Williams | innerweb: Possibly bad why not remove all of the redundat stuff then at least you can see what is going on without working your way through a 477 line config |
| 13:53.03 | shepherd | but i'm pretty sure iaxy will support 3 way calls (even if you have no external phone liens) |
| 13:53.13 | innerweb | Removed. Restarting... |
| 13:53.26 | garfield13 | Jas_Williams, let me clarify one last thing. Once I have my 3 parties talking to each other, will I then be able to transfer them both, as a unit, to a meet-me conference room? Would that be considered a 3rd transfer? |
| 13:53.37 | innerweb | Mar 3 03:01:06 WARNING[2325]: chan_skinny.c:2584 reload_config: Unable to get our IP address, Skinny disabled |
| 13:53.37 | innerweb | <PROTECTED> |
| 13:53.37 | innerweb | <PROTECTED> |
| 13:53.37 | innerweb | Mar 3 03:01:06 WARNING[2325]: chan_zap.c:769 zt_open: Unable to specify channel 1: No such device or address |
| 13:53.38 | innerweb | Mar 3 03:01:06 ERROR[2325]: chan_zap.c:6202 mkintf: Unable to open channel 1: No such device or address |
| 13:53.40 | innerweb | here = 0, tmp->channel = 1, channel = 1 |
| 13:53.42 | innerweb | Mar 3 03:01:06 ERROR[2325]: chan_zap.c:9149 setup_zap: Unable to register channel '1' |
| 13:53.44 | innerweb | Mar 3 03:01:06 WARNING[2325]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 |
| 13:53.46 | innerweb | <PROTECTED> |
| 13:53.48 | innerweb | <PROTECTED> |
| 13:53.50 | innerweb | Mar 3 03:01:06 WARNING[2325]: loader.c:440 load_modules: Loading module chan_zap.so failed! |
| 13:54.03 | Jas_Williams | innerweb: Use pastebin please !!!!! |
| 13:54.03 | RoyK | ~pastebin? |
| 13:54.04 | jbot | extra, extra, read all about it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 13:54.16 | RoyK | ~lart innerweb |
| 13:54.24 | shepherd | gar: i don't know if you can do that.. you would have to individually transfer them to meetme |
| 13:54.37 | innerweb | Sorry. Bad habit. I will stop (now). |
| 13:54.43 | shepherd | which you can also do from an iaxy :) |
| 13:54.54 | garfield13 | But, according to the telco, It is only one call. |
| 13:55.09 | Jas_Williams | innerweb: post /etc/zaptel.conf to pastebin.ca |
| 13:55.13 | shepherd | oh.. so you have a 3 way useing bell? |
| 13:55.20 | garfield13 | yes. |
| 13:55.28 | shepherd | not a problem :) |
| 13:55.38 | garfield13 | I don't want askterisk to manage the 3-way, I want the local telco to do it. |
| 13:55.42 | phantam | hmmm |
| 13:55.42 | shepherd | but you would have to do the 3 way stuff with the fxo |
| 13:55.52 | *** join/#asterisk w0w0 (~w0w0@80-28-174-152.adsl.nuria.telefonica-data.net) |
| 13:55.57 | phantam | thats a sexy concept pc from intel |
| 13:56.05 | innerweb | http://pastebin.ca/6777 |
| 13:56.07 | garfield13 | That is what instigated this entire conversation. |
| 13:56.11 | phantam | you know if intel sold pc's apple would have a good competitor |
| 13:56.12 | phantam | lol |
| 13:56.38 | innerweb | What if AMD or Sun did? |
| 13:56.40 | innerweb | lol |
| 13:56.52 | Jas_Williams | innerweb: ztcfg -vvvv |
| 13:57.35 | Jas_Williams | Until that shows a channel then asterisk will not work |
| 13:57.55 | innerweb | http://pastebin.ca/6778 |
| 13:58.11 | garfield13 | Jas-Williams, I know you are still working with innerweb, but do you think my last scenario if feesible (sp?). |
| 13:58.20 | tzanger | morning |
| 13:58.40 | Jas_Williams | garfield13: Yes possible but will require some work. |
| 13:59.00 | innerweb | Jas - I figured that. I just wish I could figure out why it does not show a channel. |
| 13:59.04 | Jas_Williams | Why do you want ameetme room as both parties can then talk via the telco line |
| 13:59.31 | innerweb | dmesg shows:Found a Wildcard FXO: Generic Clone |
| 14:00.22 | innerweb | My box? |
| 14:00.41 | Jas_Williams | Can I ssh into your server and fix it for you ? |
| 14:00.47 | garfield13 | On top of these two people, I am also going to open up the meetme to people coming in thru the internet. |
| 14:01.24 | garfield13 | So I need a meetme that will hold 2 people that don't have computers and everyone else that does. |
| 14:01.43 | *** part/#asterisk Pinhole (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com) |
| 14:01.52 | *** join/#asterisk Pinhole (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com) |
| 14:02.05 | Jas_Williams | garfield13: You would have issues in doing this as you will no longer have control of the call |
| 14:03.26 | garfield13 | After I transfer into meetme, I wouldn't need control anymore. To make a long story short, I have a telco line with 3-way calling. Shouldn't I be able to put those 2 people into a meetme? |
| 14:03.42 | Pinhole | Occasionally, I get a channel that is heard in other channels. It's happening right now. No matter which extension I call, I hear another conversation mixed in (badly) with the current conversation. how do I fix / debug this? |
| 14:04.24 | dsmouse | pinhead: is this POTS or voip? |
| 14:04.34 | pinhead | what ? |
| 14:04.46 | Pinhole | it's all sip phone -> asterisk -> sip phone |
| 14:04.46 | dsmouse | analog or digital phones? |
| 14:04.53 | Zeeek | is this love or just confusion? |
| 14:04.54 | Jas_Williams | manually from the fxs attached phone yes, then you could recall into the meet me room but this sounds a pain. whynot just host the service on a large system rather than using a soho solution * is not that |
| 14:05.14 | shaZwaz | what signalling does a digital phone ususally work on ? |
| 14:05.52 | garfield13 | Not familiar with the acronym SOHO. What is it? |
| 14:05.59 | Jas_Williams | shaZwaz: what type of digital phone, most PBX ones are propriatory |
| 14:06.24 | Jas_Williams | SoHO Small Office Home Office |
| 14:06.30 | shaZwaz | anyone |
| 14:07.02 | garfield13 | What do you mean by host the service on a large system? Like what? |
| 14:07.54 | *** part/#asterisk Nebukadneza (~daddel9@i3ED6E583.versanet.de) |
| 14:09.24 | *** join/#asterisk riksta (~rick@host217-42-22-145.range217-42.btcentralplus.com) |
| 14:09.52 | Pinhole | I have a channel that is open to my phone even though I have hung up. If I call anybody, I get the new channel and that old channel mixed together. how can I prevent this from happening? |
| 14:11.37 | garfield13 | Jas_Williams, so you know my environment, I am trying to set this up for my church so that the perishoners that are sick or home bound can listen in on the services. Most have computers and using the voip with the meetme capability provides the perfect solution for them. However, we have a few people that do not have computers so I am trying to include a solution for them. Incidentally, one of the members of the meetme is a phone plugg |
| 14:12.00 | Grooby | morning peeps |
| 14:13.04 | garfield13 | From that standpoint, cost is an issue. So when you say large system, costs start rolling around in my head. |
| 14:13.28 | garfield13 | Things start getting blurry. |
| 14:13.36 | garfield13 | and I start to babble. |
| 14:13.42 | garfield13 | Like now..... :) |
| 14:14.19 | innerweb | <PROTECTED> |
| 14:15.08 | garfield13 | I originally looked at that solution but again, it only works for the computer capable. |
| 14:17.42 | garfield13 | I do realize I can easily make all this work if I forego the 3-way calling and just put in twice as many phone line. Just seems impractical to spend an extra $30 a line when I can get a 2nd line on each one (3-way call) for 3.95. |
| 14:18.12 | innerweb | Garfield -> Where are you located (AC,)? |
| 14:18.23 | garfield13 | NC |
| 14:18.40 | innerweb | Have you talked to your local telco about a trunk for NOprof? |
| 14:19.39 | innerweb | I know that here a non-prof gets a nice discount on telco lines. A trunk might be more cost effective in that case. |
| 14:20.09 | innerweb | How close are you to the closest POP? |
| 14:22.20 | garfield13 | No. I hadn't thought of that. I am talking to my local telco at this moment. |
| 14:22.23 | garfield13 | POP? |
| 14:22.47 | innerweb | Point Of Presence. It is where the telephone company runs the land lines form. THe clser you are to one, the cheaper trunks are. |
| 14:23.33 | jobi | hi |
| 14:23.35 | garfield13 | Oh, I have no idea on that regard. There are lines running along the street but I think you are talking about a more specific type of line. |
| 14:23.51 | BrianR___ | garfield13: How is the internet connection? What about bringing in channels over IAX? |
| 14:24.02 | jobi | I keep getting a undefined symbol: pri_suspend_acknowledge |
| 14:24.06 | jobi | when starting asterisk |
| 14:24.17 | jobi | libpri is properly installed |
| 14:24.23 | BrianR___ | garfield13: iax.cc and nufone.net let you have a toll free number for $0.02/minute. If it's only used for a few hours a month then it'll be much cheaper than bringing in pots lines. |
| 14:25.02 | innerweb | Good question - How many hours do you plan on putting on this thing? I am used to working on these things 60 hours per week./ |
| 14:25.24 | garfield13 | We are talking 80 hours a month with an average of 4 people listening in. |
| 14:25.41 | ariel_ | jobi, did you configure your pri settings in the zapata.conf and zaptel.conf files? |
| 14:25.53 | garfield13 | We have 2 congregations that meet. Each using 5 hours a week. |
| 14:26.04 | innerweb | Is that 4 people connecting, or 4 people without an Internet conection? |
| 14:26.25 | *** join/#asterisk Stitch (~magliett@cpe-69-204-16-160.rochester.res.rr.com) |
| 14:26.54 | ariel_ | garfield13, you can get if they provide in your area via VoicePulse an local did number that allows up to 4 incoming channels to your asterisk. |
| 14:27.11 | ariel_ | cost is around 7.95 per month unlimited inbound. |
| 14:27.23 | Stitch | hello all -- is there anyone here that could help me with a rather puzzling RTP problem =) |
| 14:27.43 | ariel_ | Stitch, post the problem? |
| 14:27.53 | garfield13 | Let us say, 2 need the phone access. Most others could use the internet. And sometimes we go up to 6 but we tell the people on the phone to 3-call others and daizy chain them together. That is how we do it now with our one phone line connected to our PA. |
| 14:28.35 | *** join/#asterisk Zaw (zaw@zaw.subneural.net) |
| 14:29.20 | innerweb | LOL. I love ingenuity. So, you might need 4 lines if you want two available. THen, the rest can use SIP to access through the Internet. If they are listening and you have enough outbound bandwidth, that might work. |
| 14:29.42 | Stitch | [asterisk]->[NAT]->[Internet]->[NAT]->[xLite] The RTP is arriving at the xLite phone and can be captured and played back with ethereal but the xLite phone does not play any incoming audio even though the stream is there. Other sip calls work from other locations behind NATs |
| 14:30.06 | garfield13 | I am expecting a call back from my telco sales rep. I will ask him these questions. |
| 14:30.13 | Stitch | doesn't matter if the xLite phone dials or if a phone hosted on the asterisk fails |
| 14:30.21 | Stitch | also broken with pulver.Comunicator |
| 14:30.32 | Jas_Williams | garfield13: Why not use an IAX provider to ring in the inbound phone lines rather than a telco and have no PSTN lines |
| 14:30.34 | Stitch | fwd works |
| 14:30.41 | jobi | ariel_: yes |
| 14:31.10 | innerweb | Jas is right. I am still thinking in old_speak. |
| 14:31.49 | *** join/#asterisk jcims (~jcims@cpe-69-135-121-57.columbus.rr.com) |
| 14:31.53 | innerweb | I might keep one land line though (for emergency uise, like power outages.) |
| 14:32.43 | garfield13 | This is a new concept to me. An IAX provider. Would my local telco provide that or someone else? |
| 14:32.45 | Essobi | WOOOOP WOOOP ROOOOOOT HAAAAAS LAAAAANDED. |
| 14:32.52 | Essobi | Morning peeps. |
| 14:33.15 | Essobi | garfield13 Maha.. No, likely not your local Telco.. been to http://www.voi-info.org? |
| 14:33.29 | Essobi | http://www.voip-info.org |
| 14:33.31 | Essobi | even.. |
| 14:34.43 | garfield13 | OOOOHHHH, like vonage? |
| 14:34.48 | Jas_Williams | garfield13: In this the telco has lots of PSTN lines and gives you a number that you point over the internet to your asterisk server as many people as you have internet bandwidth can call in |
| 14:35.15 | Jas_Williams | and costs you a lot less than pstn rental |
| 14:35.33 | Essobi | Stitch Hmm. I'm doing exactly what you're talking about, and it works here. |
| 14:36.01 | Essobi | Albeit with a 7960 and it took a lot of dicking around with sip.conf and the fone configs to get it right... |
| 14:36.50 | garfield13 | Guys, is Vonage an example of what you are talking about? |
| 14:36.54 | Stitch | essobi: I know it is very perplexing ~ the SDP negotiates out the correct channels |
| 14:37.05 | Stitch | but still no incoming audio at the xLite phone |
| 14:37.06 | *** join/#asterisk SkySky (~Jack@host6614613596.biz.tor.fcibroadband.com) |
| 14:37.12 | Essobi | Jas_Williams That's if bandwidth is cheap er then PSTN. :) |
| 14:37.45 | Essobi | Umm.. I was getting control channels and RTP was landing and everything.. but the phone or * would just ignore it. |
| 14:38.01 | Essobi | So.. It could be X-lite it sound. |
| 14:38.01 | pimpwell | what happens with an automated call when it gets an answering machine? |
| 14:38.05 | Essobi | What version of *? |
| 14:38.06 | Jas_Williams | Essobi: Internet access to the box was planned anyway so should not affect the cost model much |
| 14:38.06 | pimpwell | Is there a way to detect this? |
| 14:38.07 | *** join/#asterisk santiago (~santiago@63.245.86.105) |
| 14:38.25 | Essobi | "much" |
| 14:38.32 | Essobi | Depends on what codecs he uses. |
| 14:38.43 | Essobi | I'd call 711 significant. |
| 14:38.45 | Stitch | essobi: if I take the nat away -- and connect, the xlite works |
| 14:38.49 | Essobi | And is he planning on doing faxing? |
| 14:38.59 | Stitch | its almost like there is something in the RTP packet that is confusing the phone |
| 14:39.00 | Essobi | Stitch TADA! It's MAGIC! |
| 14:39.10 | Essobi | Naah, RTP IS RTP. |
| 14:39.19 | Essobi | the phone has to be smart enough to deal with it. |
| 14:39.26 | *** join/#asterisk visik7 (~ciao@visik7.user) |
| 14:39.33 | Essobi | what ver of * Stitch ? |
| 14:39.34 | Stitch | hehe -- yeah -- xlite and smart -- oxymoron =) |
| 14:39.35 | garfield13 | HAVE I BEEN DROPPED? ALL MY PROBLEMS ARE SOLVED??? :) |
| 14:39.41 | Stitch | let me look at ver |
| 14:39.52 | Jas_Williams | garfield13: Yes Vonage is such a company |
| 14:40.04 | Stitch | Asterisk CVS-HEAD-02/17/05-16:51:45 built by root@switchbord on a i686 running Linux |
| 14:40.09 | garfield13 | I now have a much clearer picture. |
| 14:40.10 | ariel_ | garfield13, but it's the wrong way to go. |
| 14:40.11 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
| 14:40.20 | garfield13 | SHOOOOOOT |
| 14:40.22 | garfield13 | why? |
| 14:40.59 | *** join/#asterisk jsolares (~jsolares@200.30.141.85) |
| 14:41.02 | ariel_ | vonage is one or 2 lines inbound at most with there ata device. You need one that goes via the internet to your asterisk server. |
| 14:41.21 | Jas_Williams | garfield13: Use nuphone instead |
| 14:41.23 | Essobi | Stitch What's the X-lite diagnositc logs showing you? |
| 14:41.36 | ariel_ | Jas_Williams, he needs local inbound did |
| 14:41.52 | Jas_Williams | Depends where local is :) |
| 14:41.57 | Stitch | they just show the sip conversation -- unless there is some other logs that are not in the diag window that comes up |
| 14:42.06 | Stitch | all of the IP's in there look good too |
| 14:42.06 | visik7 | how can I forse * to open audio channel before the other parts brings up the telephone ? |
| 14:42.16 | visik7 | s/forse/force |
| 14:42.22 | pimpwell | my system is making an automated call time to time for a reminder, I was wondering if there is a way to recognize an answering machine and what does it take to do so. |
| 14:42.36 | pimpwell | can this be done alone in the .call file? |
| 14:42.47 | ariel_ | Stitch, do you have in the asterisk sip.conf extenalip= and localip= in the settings. |
| 14:42.56 | Stitch | yes |
| 14:43.05 | Essobi | pimpwell No idea |
| 14:43.20 | Stitch | and other phones (cisco 7940) work in the same situation |
| 14:43.31 | Stitch | as does PSTN access ;) |
| 14:43.39 | Essobi | welp |
| 14:43.41 | Essobi | I dunno. |
| 14:43.48 | Jas_Williams | garfield13: Where are you and then maybe areil_ can recomend a suitable provider of inbound iax or SIP |
| 14:43.49 | Essobi | I never had to run X-lite in a double nat |
| 14:43.51 | Stitch | ... told you it was perplexing =) |
| 14:43.56 | Essobi | Nat's are evil enough. |
| 14:44.02 | Essobi | Let alone a double nat. |
| 14:44.13 | Essobi | I could probably tell you with a sniffer, but I got to get back to work. |
| 14:44.25 | Stitch | NAT is fun =) Protocol fixups are evil =D |
| 14:44.36 | Essobi | Nat is EVIL. |
| 14:45.27 | Essobi | garfield13 what are of the US? |
| 14:45.54 | Essobi | area |
| 14:46.10 | Essobi | How much are vonage's sip rates, btw? |
| 14:46.17 | Essobi | and nuphone? |
| 14:46.21 | garfield13 | I am in central NC. |
| 14:47.03 | ManxPower | ~docs |
| 14:47.04 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 14:47.29 | Essobi | garfield13 Check your personal messages. |
| 14:48.21 | *** part/#asterisk Stitch (~magliett@cpe-69-204-16-160.rochester.res.rr.com) |
| 14:48.39 | *** join/#asterisk Stitch (~magliett@cpe-69-204-16-160.rochester.res.rr.com) |
| 14:48.49 | Zeeek | charlotte NC ? |
| 14:49.56 | *** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 14:50.43 | Goshen | Essobi: Vonage is a closed system that makes you use their adapters I beleive |
| 14:50.59 | Goshen | Essobi: nufone.com |
| 14:51.31 | jobi | in sip.conf, is secret= used to ask authentication from incoming call or for asterisk to authenticate outgoing calls? |
| 14:51.40 | Essobi | Well DUH.. I figured jer or someone would have chimed in reguarding rates. :) |
| 14:53.47 | *** join/#asterisk zno (~chatzilla@ip-160-79-174-99.autorev.intellispace.net) |
| 14:54.16 | garfield13 | Greensboro, Zeeek |
| 14:54.20 | Goshen | Essobi: well duh, why waste his time when you can find out in two clicks? |
| 14:55.01 | Essobi | Cause I'm lazy, and if you knew the rate, you just expended more enegery bitching then it would have taken to type 0.03. :) |
| 14:55.31 | *** join/#asterisk adjacent (~scott@office.bftwave.com) |
| 14:56.59 | Essobi | Anyways, back to work. |
| 14:57.24 | ariel_ | garfield13, lets get back to what your posible solution is. |
| 14:57.48 | ariel_ | 1) do you have an asterisk server for your PBX or do you have some other pbx system in the Church? |
| 14:58.01 | garfield13 | So let's do the math. 80 hours a month * 4 people * .02/minutes comes to a monthly change of just under $400. |
| 14:58.28 | garfield13 | Right now, nothing. I am building an asterisk server since I believe it is my best solution. |
| 14:58.41 | ariel_ | garfield13, I am not saying to use nufone for 800 number service. |
| 14:58.54 | ariel_ | Good that you have no pbx setup yet. |
| 14:58.59 | garfield13 | Actually, I have a phone with 3-way calling, a little box between the phone and the PA and the PA. |
| 14:59.02 | Goshen | nufone doesn't charge for outgoing 800 |
| 14:59.16 | ariel_ | Goshen, he needs inbound. |
| 14:59.38 | garfield13 | I need inbound, local. All callers are within 5 miles of the host. |
| 14:59.52 | Goshen | iax.cc has 1.43 inbound 1800 with $200 prepay |
| 15:00.04 | garfield13 | It would almost be cheaper to just run a phone cable to everyone house. |
| 15:00.08 | Goshen | oh so you need a flat rate local incoming |
| 15:00.11 | ariel_ | ok now the box that you have to the pa you can plug it into the asterisk server. |
| 15:00.37 | Goshen | what state? |
| 15:01.02 | garfield13 | Our solution is to hook the phone that is plugged into that box, (Goshen, NC) into the asterisk box via a iaxy. |
| 15:01.03 | *** join/#asterisk mountie (~mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com) |
| 15:01.19 | *** join/#asterisk Es_PagAn (~EP@ppp18-adsl-237.ath.forthnet.gr) |
| 15:01.27 | Es_PagAn | hello ppl |
| 15:01.34 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
| 15:01.35 | Goshen | I like that city :) |
| 15:01.41 | Goshen | the name anyhow |
| 15:01.47 | ariel_ | In the asterisk server you should get a tdm400 board for the inbound pots line and the box that you have. |
| 15:01.49 | pif | hi, what is the latest cisco 7960 sip version? |
| 15:01.50 | garfield13 | Put the iaxy between the box and * |
| 15:01.56 | garfield13 | High Ponit |
| 15:01.57 | ariel_ | no need for a iaxy. |
| 15:02.02 | garfield13 | oops, High Point, NC |
| 15:02.08 | Es_PagAn | anyone to help out a newbie? |
| 15:02.37 | ManxPower | ~docs |
| 15:02.38 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 15:02.38 | ManxPower | There |
| 15:02.40 | ManxPower | s your help |
| 15:02.45 | Goshen | iax.cc offers Unlimited Local Number $10.49/month 0¢/min and their map shows NC available |
| 15:03.02 | ManxPower | Goshen: LiveVoip is only $4 or $7/month |
| 15:03.03 | garfield13 | Yes. We have decided on the TDM400 with a FXO port and we were going to use the iaxy instead of the FXS port. |
| 15:03.07 | ariel_ | Goshen, so does VoicePulse 7.95 unlimited inbound. |
| 15:03.14 | Goshen | Manx: for unlimited incoming IAX? |
| 15:03.25 | Es_PagAn | well it doesn' answer a specific question, I have already RTFM |
| 15:03.29 | ariel_ | garfield13, why? |
| 15:03.35 | garfield13 | I certainly hope there is a way I can print this conversation when we are all done. |
| 15:03.37 | ariel_ | Goshen, yes |
| 15:03.38 | ManxPower | Goshen: The $6/month is "low usage" |
| 15:03.43 | Goshen | Es: ask specific questions, and you get specific answers :) |
| 15:03.49 | ManxPower | The $4/month is even lower usage. |
| 15:03.53 | Es_PagAn | ok there we go then |
| 15:04.12 | ManxPower | I signed up for Livevoip last night, my account is not active yet. |
| 15:04.17 | Goshen | have either of you used LiveVoip or VoicePulse? |
| 15:04.20 | Goshen | ok |
| 15:04.22 | ariel_ | Via voicepulse connect I have there 7.95 unlimite inbound number works great I can have up to 4 calls with it. |
| 15:04.29 | *** join/#asterisk DrRighteous (~DrRighteo@ool-182c867b.dyn.optonline.net) |
| 15:04.32 | Goshen | 4 calls at a time? |
| 15:04.36 | ManxPower | Goshen: Voicepulse has had a history of call quality problems |
| 15:04.55 | Es_PagAn | If I want to use asterisk with 2 incoming analog lines and 25 phones in the company, what hardware is needed? |
| 15:04.56 | ariel_ | Goshen, yes |
| 15:05.04 | Goshen | wow, thats nice :) |
| 15:05.04 | *** join/#asterisk marshall (~test@S0106000f66563988.wp.shawcable.net) |
| 15:05.11 | ariel_ | and ManxPower yes they have but since the last 4 months I have had no problems with them. |
| 15:05.15 | Es_PagAn | the phones are analog ones |
| 15:05.53 | Goshen | ariel: that sounds really neat..I will go have a look |
| 15:05.56 | ariel_ | Es_PagAn, are you phones analog phones? |
| 15:06.03 | *** part/#asterisk Moc____ (~mochouina@64.235.210.66) |
| 15:06.06 | Es_PagAn | yes :) |
| 15:06.42 | ariel_ | you will need at least one channel bank like an adtran 750 which you can get from ebay used for about 400 to 500 dollars. |
| 15:06.42 | Goshen | channel bank? |
| 15:06.47 | *** join/#asterisk Moc____ (~mochouina@64.235.210.66) |
| 15:07.04 | ariel_ | Then an t100p card for the asterisk and you might need another tdm400b for the other analog port. |
| 15:07.09 | garfield13 | Let me recap....... VoicePulse will provide 4 calls at a time? |
| 15:07.57 | Goshen | VoicePulse provide IAX config files(looking at them now) neat :) |
| 15:08.18 | Es_PagAn | ariel_: was that an answer to me? you mean TDM400P? |
| 15:08.19 | ManxPower | I believe LiveVoIP also supports IAX. |
| 15:08.29 | DrRighteous | Yes they do |
| 15:08.29 | ariel_ | Goshen, there is a 2nd area for the low rate inbound did. |
| 15:09.15 | ManxPower | This is funny: http://livevoip.com/index.php?subject=1&content=CanadianDIDS |
| 15:09.36 | tzanger | http://portlandmercury.com/2002-05-02/feature4.html |
| 15:09.41 | tzanger | no THAT's funny |
| 15:09.58 | Es_PagAn | this will only provide 4 analog phones |
| 15:10.04 | ariel_ | Es_PagAn, as a 2nd card you will need it as well as the t100p for the c/b |
| 15:10.37 | ariel_ | In the us the channelbank is a 24 port fxs and you said you need 25 |
| 15:11.11 | ariel_ | now you also need to connect your pots lines to the asterisk Es_PagAn correct. how many do you have? |
| 15:11.30 | Es_PagAn | aha, so that means that one tdm400p will provide for 24 analog phones? |
| 15:11.41 | ariel_ | Es_PagAn, no no |
| 15:11.54 | ariel_ | tdm400 will only provide 4 ports. |
| 15:12.11 | Es_PagAn | so how do i support 25 phones? |
| 15:12.20 | ariel_ | the t100p connected to a Channel Bank will provide you the 24 analog ports. |
| 15:12.24 | Es_PagAn | let's say I have 4 pots lines |
| 15:13.10 | ariel_ | Es_PagAn, you have a system that is an in between different setups. So it's going to have to have a special setup. |
| 15:13.18 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 15:13.22 | Es_PagAn | ok so the tdm is for the public network ant the t100p will be connected on the channel bank right? |
| 15:13.26 | ariel_ | It might be less expensive to change at least one phone out to sip |
| 15:13.40 | ManxPower | Es_PagAn: The Digium TxxxP and TExxxP cards plus a channel bank will convert the T-1 channels into analog. |
| 15:13.41 | ariel_ | Es_PagAn, close |
| 15:13.57 | Es_PagAn | so i got it (i think) |
| 15:14.40 | Es_PagAn | ok let's do some searchin' |
| 15:14.59 | ariel_ | Es_PagAn, a tdm04b for your inbound pots lines. a t100p for the channel Bank 24 ports for analog phones |
| 15:15.14 | Es_PagAn | yes, i got that, thank you |
| 15:15.25 | ariel_ | The next step is for the 25th phone which in your case it can be done via a sip phone or a sip ata device. |
| 15:15.46 | eKo1 | Hmm...I come in this morning and I see 32 active channels, yet nobody is on the phones. |
| 15:15.59 | eKo1 | Then I do a 'soft hangup' on those channels and get: Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 172.16.0.151 |
| 15:16.19 | Es_PagAn | yeah maybe 24 analog phones will be used, was not clear about that as of yet |
| 15:16.28 | eKo1 | Anybody experienced such nonsense before? |
| 15:17.20 | ariel_ | eKo1, stop asterisk and restart it. |
| 15:17.45 | eKo1 | ariel_: eh, that doesn't solve the problem. It just fixes it temporarily. |
| 15:18.09 | ctooley | Ok, I got my IAXy with no instructions, no information, nothing but the device and the power adapter. How am I supposed to notice the thing on the network to configure it. |
| 15:18.10 | ariel_ | well then read your logs to see what errors you got. |
| 15:18.18 | eKo1 | There are no errors. |
| 15:18.21 | ariel_ | ctooley, yes |
| 15:18.46 | eKo1 | The interesting things is that all those dead 'active' channels are from the same sip device. |
| 15:19.13 | ManxPower | ctooley: check the mailing list archives and download using cvs "iaxprov" |
| 15:19.22 | eKo1 | So I'm guessing I have to replace that PoS device. |
| 15:19.37 | garfield13 | Goshen , what is the url for iax.cc? With the map you mentioned earlier? |
| 15:20.06 | Goshen | www.iax.cc :) |
| 15:20.29 | Goshen | they advertise on pastebin.ca thats were I saw them, I haven't tried their service yet |
| 15:21.06 | *** join/#asterisk k_bot (~mejii@hosted-pat.7ticks.com) |
| 15:22.12 | zno | anyone try xorcom ? |
| 15:23.19 | garfield13 | Goshen, thanx... :O |
| 15:23.25 | ariel_ | garfield13, do you have any more questions or are you on the right track yet? |
| 15:24.58 | ariel_ | eKo1, I had something similar and it was a device that had a dialing string incorrect and it kept doing a round robin dialing got up to 34 channels in use once on me. |
| 15:25.30 | eKo1 | ariel_: Care to say what device that was? |
| 15:25.38 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
| 15:25.38 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:25.39 | ariel_ | I got to go for a few will return in about 10 |
| 15:26.32 | garfield13 | I want to thank everyone for their help. I am looking at the web sites for iax.cc (thanx Goshen :) ), voicepulse and livevoip. |
| 15:27.20 | garfield13 | I don't the 7.95 includeing unlimited and 4 concurrent call on the voicepulse site so I am not sure what you were talkeing about,ariel_? |
| 15:27.20 | jobi | am I right that ${EXTEN:1} should be the current extension, stripped from the first digit? |
| 15:27.45 | Es_PagAn | thanx again, off now :) |
| 15:28.39 | tzafrir | heck, nobody here tried Rapid? |
| 15:28.43 | *** join/#asterisk wrmem (~wrmem@monnin-win.cso.uiuc.edu) |
| 15:29.35 | garfield13 | I do want to thank all of you for the help, Jas_Williams that got me started, innerweb & zeeek, and then you ariel_ and Goshen really rounding out the input to my solution. I have a lot of investigating to do. |
| 15:29.46 | ctooley | ManxPower, Ok, so how am i supposed to devine what IP address the IAXy got from my DHCP server? I'm honestly expected to go through the leases and try to find it without them even providing a MAC address? |
| 15:30.19 | ctooley | Oh, well, guess I'll make the telephone call. |
| 15:31.17 | ariel_ | garfield13, http://connect.voicepulse.com/ |
| 15:31.18 | garfield13 | Does anyone know in mIRC how I can save this conversation to a file? |
| 15:31.29 | afe | does anyone has an example of how to automatically mail monitored call files to an email address? |
| 15:32.21 | garfield13 | ariel_, ah hah. Thanx again. |
| 15:32.30 | sambal | garfield13: ALT-E, IRC, Logging |
| 15:32.59 | Beirdo | gah |
| 15:33.13 | Beirdo | freeworldtel just renamed to link2voip |
| 15:33.54 | ariel_ | garfield13, they don't tell you that they have 4 calls at once on there site. But I have tested it and it works for 4 inbound calls. |
| 15:34.00 | tzafrir | BTW: are logs of this channel kept anywhere? |
| 15:34.13 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 15:34.49 | ariel_ | eKo1, I had a bad dialing rule that kept dialing it's own number over and over again. so it took all my channels up on my pri and my back provider. |
| 15:35.18 | mishehu | oops |
| 15:35.18 | mishehu | heh |
| 15:35.53 | garfield13 | sambal, this is not intuitive to me. I want to save this entire conversation. Logging automatically is now set to None. |
| 15:35.55 | jobi | I'm using Dial(ZAP/g1/${EXTEN:1},20,tr) in an extension |
| 15:35.56 | adjacent | anyone interested in buying a full voip solution? all cisco? |
| 15:35.57 | roamer323 | beirdo - do u use freeworldtel/linkwvoip? |
| 15:36.13 | Beirdo | was planning on it for a US50/Canada toll free |
| 15:36.16 | jobi | but it dials the whole extension, without dropping the first digit |
| 15:36.35 | Beirdo | wow. and voipjet was flopping around like a fish on a dock this morning |
| 15:36.49 | adjacent | two cisco 2600's with PRI interface, and t1/e1 MUX cards, and around 40-50 ATA 186's, over half are still NIB |
| 15:36.49 | roamer323 | beirdo - the guy is out of edmonton, but his connection point is in dallas it seems |
| 15:36.55 | Beirdo | yep |
| 15:37.28 | roamer323 | beirdo - does he have unlimited incoming DIDs - do you know? site is vague. |
| 15:37.48 | Beirdo | he does for some area codes, none Canadian |
| 15:37.59 | zno | Beirdo: do you find that voipjet is normally consistent? Voicepulse is very flaky |
| 15:38.22 | Beirdo | zno: no, I find that voipjet buggers up more than once a day |
| 15:38.58 | zno | The thing that really prevents me from going to voipjet is having to manually use Paypal for invoices |
| 15:39.02 | mishehu | Beirdo: how often do you call via voipjet, is it buggering up to specific countries, and are you sure it's not a problem on your route? |
| 15:39.02 | roamer323 | anyone knows how * "unwinds" a call going from A->B->A ? A and B are * interconnected via IAX2 |
| 15:39.12 | mishehu | I generally have good usage with voipjet. |
| 15:39.24 | Beirdo | it's buggering up by latencies hitting 2s once in a while |
| 15:39.25 | mishehu | but I'm not doing a heavy load of calls yet. |
| 15:39.27 | Beirdo | it's not my link |
| 15:39.35 | Beirdo | it's likely theirs |
| 15:39.48 | mishehu | Beirdo: or somewhere between |
| 15:39.56 | Beirdo | I have had not a single glitch with FWD's gateway |
| 15:39.56 | zno | Beirdo: he guarantees 27 ms to tier 1 tho, is that in some SLA? |
| 15:40.11 | Beirdo | no |
| 15:40.17 | *** join/#asterisk lyroy (~lyroy@picachou.csaffluents.qc.ca) |
| 15:40.18 | Beirdo | you think he'll give an SLA? |
| 15:40.32 | garfield13 | Jsa_Williams, are you still there? |
| 15:40.42 | lyroy | What is the easyest way to rate phone calls in the cdr database? |
| 15:40.48 | Beirdo | wishful thinking, zno :) |
| 15:41.04 | mishehu | lyroy: cdr database table has nothing to do with how to route calls |
| 15:41.10 | Jas_Williams | garfield13: Yes |
| 15:41.13 | zno | voip is still a fetus, no real good providers |
| 15:41.23 | mishehu | not unless you're doing something wonky with app mysql or something |
| 15:41.32 | Beirdo | I doubt ANY VOIP provider would be rash enough to give an SLA at this point |
| 15:41.36 | zno | I'm scouring weekly for a good one. Everyone competes on price but not on service |
| 15:41.41 | mishehu | zno: it's more like a toddler I'd say. |
| 15:42.02 | garfield13 | Do you have a way to log this conversation back to the when I logged in and send it to me. I gained a lot of information in all this and I don't want to lose it and I can't get it to save using mIRC. |
| 15:42.12 | mishehu | it's improved so much since the early 90s |
| 15:42.21 | lyroy | mishehu how can I rate phone calls? |
| 15:42.29 | Beirdo | ~docs |
| 15:42.30 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 15:42.43 | mishehu | lyroy: oh sorry, I was lysdexic |
| 15:42.43 | roamer323 | beirdo - do you think the cable provider like shaw or rogers may give a tap into their private voip networks for a price -with an SLA ? |
| 15:42.49 | mishehu | lyroy: I thought you typed "route" |
| 15:42.54 | Jas_Williams | No i do not have most of it I only have a small scrollback buffer |
| 15:42.55 | lyroy | k |
| 15:43.16 | lyroy | is there a way to put a price in the cdr database? |
| 15:43.18 | garfield13 | shoot.... |
| 15:43.29 | eKo1 | lyroy: Yes. |
| 15:43.30 | Beirdo | roamer323: I doubt they'd give an SLA, it's too easy to get screwed that way |
| 15:43.31 | mishehu | lyroy: I heard rumor about something called "rate engine" |
| 15:44.06 | lyroy | eKo1 can you point me to documentation? |
| 15:44.20 | eKo1 | lyroy: Check the wiki. |
| 15:44.28 | mishehu | Wiki Tiki Tavi |
| 15:44.55 | lyroy | eko1 is custom code or you found something that do this? |
| 15:45.03 | mishehu | hahah I love this, on /., an article about a microsoft robot that watches children... is this Microsoft dba Sirius Cybernetics Corporation? |
| 15:46.09 | roamer323 | haha - if that wiki's database server ever crashes unrecoverably (like livejournal, etc) - it will set back voip progress by 5 years :-D |
| 15:47.02 | mishehu | roamer323: 5? |
| 15:47.04 | mishehu | try 10 |
| 15:47.05 | mishehu | heh |
| 15:47.23 | mishehu | we'll have to call up vocaltel |
| 15:47.46 | Juggie | has anyone done much work with agents & queues.... i'm trying to decide if i have a bug, or this functionality is intended... |
| 15:47.59 | mishehu | Juggie: state the problem... |
| 15:48.08 | Damin | Well.. |
| 15:48.16 | mishehu | I can't claim to be an expert on them, but it's worth you to spit it out |
| 15:48.16 | Beirdo | roamer323: livejournal, etc weren't unrecoverable :) |
| 15:48.21 | Damin | Much to my surprise, Asterisk builds "like buttah" on OSX! :) |
| 15:48.47 | eKo1 | lyroy: I do it using a DB trigger function. |
| 15:48.50 | roamer323 | beirdo - "somewhat" recoverable with "great pain" - I've heard :-D |
| 15:49.30 | Beirdo | so I might hook it up tonight |
| 15:50.25 | lyroy | eKo and you custom made that script...? |
| 15:50.38 | roamer323 | anyone knows how * "unwinds" a call going from A->B->A ? A and B are * interconnected via IAX2 |
| 15:51.35 | tzanger | roamer323: "unwinds" ? |
| 15:51.42 | tzanger | roamer323: as in does it drop B out? |
| 15:51.57 | roamer323 | tzanger - I have observed it |
| 15:52.08 | roamer323 | tzanger - and I don't know why/how |
| 15:52.18 | Zeeek | Someone was here an hour ago with this router: |
| 15:52.20 | Zeeek | http://www.pbs.org/cringely/pulpit/pulpit20040527.html |
| 15:52.34 | Zeeek | <PROTECTED> |
| 15:52.41 | garfield13 | sambal, you still there? |
| 15:52.44 | Damin | I have about 50 of those Linksys WRT54 units.. |
| 15:52.47 | Damin | They rock! |
| 15:52.48 | sivana | ~seen normast |
| 15:52.51 | jbot | normast <HydraIRC@Ottawa-HSE-ppp4119108.sympatico.ca> was last seen on IRC in channel #asterisk, 10h 5m 42s ago, saying: 'juggie: Thanks keep it in mind.'. |
| 15:52.53 | tzanger | Damin: they're alright |
| 15:53.05 | Damin | tzanger: What's better for $60? :) |
| 15:53.33 | tzanger | 200MIPS ARM isn't really all that powerful but yes for what they do they're pretty good little routers |
| 15:53.34 | Zeeek | " This same firmware runs on Belkin, ASUS, and Buffalotech routers today. The source code comes from Broadcom, not Linksys. " |
| 15:54.02 | Zeeek | " There is an obvious business opportunity here, especially for VoIP providers like Vonage, Packet8 and their growing number of competitors. If I was running a VoIP company ,I'd find a way to sell my service through all these new Wireless ISPs. |
| 15:54.19 | Zeeek | talk about hitting the mainstream? PBS ? |
| 15:54.25 | Damin | tzanger: Hmm.. let's see.. 5 Ethernet Ports, 200 Mips, 16 megs ram, 8 megs flash.. :) |
| 15:54.39 | garfield13 | Before I leave, thank you again to everyone. You are a friendly wealth of knowledge. |
| 15:54.45 | Damin | Rock on dude! :) |
| 15:54.56 | tzanger | Damin: as I said if it works of ryou it's great. And they do work reasonably well |
| 15:55.11 | *** join/#asterisk meppl (~mephisto@pD9E6861E.dip.t-dialin.net) |
| 15:55.19 | spackle | Anybody have some hints on how to register the netweb eezee phones with IAX? |
| 15:55.30 | Damin | tzanger: I actually deploy the WRT54GS units, w/ 233 mips, 32 megs of ram and 16 of flash w/ the Sveasoft firmware.. |
| 15:55.57 | Juggie | mishehu, the problem is that when i use groups, aka 3 agents in a group, ringall on the queue only rings one person in the group |
| 15:55.57 | Damin | tzanger: That adds a ton of crap... |
| 15:55.58 | tzanger | yeah sveasoft's not bad |
| 15:55.59 | Juggie | not all, as i would expect. |
| 15:56.47 | zipp | openwrt++ |
| 15:56.48 | Damin | Hell.. my entire house is wired up w/ WRT's using WDS as a wireless backbone. I can go anywhere within about 500 feet of my house and get a solid 54 meg siganl. :) |
| 15:57.04 | Damin | And each room has 5 ethernet ports in it. :) |
| 15:57.16 | Damin | Alright.. |
| 15:57.19 | Damin | must go to lunch meeting.. |
| 15:57.23 | Damin | BBL.. |
| 15:57.24 | *** join/#asterisk gtigene (~gnadenx@208.239.206.195) |
| 15:57.26 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
| 15:57.52 | *** join/#asterisk julianjm (~julianjm@250.Red-80-59-67.pooles.rima-tde.net) |
| 15:57.52 | Beirdo | Damin: your neighbours must love the free internet you are providing :) |
| 15:58.23 | mikegrb | Beirdo: I do. |
| 15:58.48 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 15:59.14 | gtigene | How can I tell how many calls have been abandoned in my ACD agent queue? |
| 16:01.27 | jontow | manager API has that info, gtigene.. |
| 16:01.35 | Juggie | events |
| 16:01.41 | jontow | look at the "Action: Queues" and "Action: QueueStatus" |
| 16:01.54 | gtigene | jontow, thank you |
| 16:02.46 | SkySky | hi, i wonder if anyone here uses oss or alsa for paging? Im using fc3 and when loading up chan_oss.so, i also get a warning msg saying that oss device is busy or not installed. But when i Dial(CONSOLE/dsp) it's working properly. |
| 16:03.02 | Juggie | again, has anyone used agent queues and can tell me if they noticed similer behaviour.... when i use agent groups.... ringall only rings one phone in the group, not all... random allways rings the same phone in the group.... is this intended or broken? |
| 16:03.51 | *** join/#asterisk Tarox (someone@pD9E7B727.dip.t-dialin.net) |
| 16:04.00 | SkySky | also, after i've loaded chan_oss.so, I can't use oss in other applications. Can somebody help me? |
| 16:04.13 | Juggie | if i specify each agent individually in queues.conf instead of just pointing at the group then it works like you would expect. |
| 16:05.12 | SkySky | Juggie: i believe u really should special each agent right? |
| 16:05.23 | SkySky | Juggie: *specify* |
| 16:05.43 | Juggie | whats the purpose of a group then if i have to specify each? |
| 16:06.23 | *** join/#asterisk JerJer[mobile] (~jj@feth100-fw.fament.net) |
| 16:07.06 | SkySky | Juggie: ?? im sorry.. but i never uses agent b4. right now, im using member. for each queue line, i have member => SIP/ext01 member => SIP/ext02 |
| 16:07.28 | SkySky | Juggie: sorry. im not sure about wat's group for agent. |
| 16:07.43 | ManxPower | SkySky: That is correct. Asterisk requires exclusive access to OSS. |
| 16:08.16 | tzanger | ManxPower: huh? |
| 16:08.20 | tzanger | 11:14 < ManxPower> SkySky: That is correct. Asterisk requires exclusive access to OSS. |
| 16:09.07 | SkySky | ManxPower: something weird when i tried to unload chan_oss.so then reload again. asterisk crashes after a while. "Segment fault" or something like that |
| 16:09.11 | ManxPower | tzanger: "Correct. Asterisk requires exclusive access to your sound device if you want to use chan_oss or chan_alsa." Better? |
| 16:09.16 | tzanger | ohhhhhhh |
| 16:09.24 | tzanger | OSS as in sound, not OSS as in open source software |
| 16:09.32 | ManxPower | tzanger: Yeah. |
| 16:09.32 | tzanger | my apologies, I only caught the first part of that |
| 16:09.35 | SkySky | ManxPower: and also, do i need to worry about that warning msg asterisk is giving me when loading chan_oss.so? |
| 16:09.53 | ManxPower | SkySky: Without knowing the warning message, I can't say. |
| 16:10.18 | SkySky | ManxPower: yep. let me get that msg .. pls wait me one sec~ thx |
| 16:11.26 | *** join/#asterisk Rick_Hunter (~rhunter@03-185.008.popsite.net) |
| 16:12.21 | SkySky | ManxPower: at the first time i load chan_oss.so. I got this msg: "WARNING[5468]: chan_oss.c:239 sound_thread: Read error on sound device: Resource temporarily unavailable" |
| 16:13.06 | Zeeek | Windows machine: "can not load Open Source Software" |
| 16:13.18 | Zeeek | "Please Install Office" |
| 16:13.36 | *** join/#asterisk fugitivo (~ajf@201.255.100.146) |
| 16:13.51 | [TK]D-Fender | Zeeek : Careful, Redmond may be listening...... |
| 16:14.09 | Zeeek | They're stilll stinging from the EU rulings |
| 16:14.31 | Zeeek | anyway, Windows has its place. That's something many asterisk users should learn |
| 16:14.33 | Beirdo | yeah, they can't just ignore them like the DOJ :) |
| 16:14.46 | Beirdo | Windows is great for playing games |
| 16:14.47 | [TK]D-Fender | The same way you car stings when it hits a bee at 70mph |
| 16:14.59 | Zeeek | Personally I think MS is a little too easy a target and too much fun to attack |
| 16:15.45 | Zeeek | Windows is a fact, it's a huge user base installed in most offices. It needs to be considered and has its uses. Like non electric can openers |
| 16:16.08 | Zeeek | But I don't defend it, I just realize it's a necessary tool some people need and use |
| 16:16.23 | zipp | surely those tools can be replicated and improved |
| 16:16.29 | Zeeek | go ahead |
| 16:16.34 | [TK]D-Fender | Actually I find electric can-openers absolutely worthless. I can do the job faster and easier myself without the dependance. |
| 16:16.42 | SkySky | ManxPower: but dial(console/dsp) works without problem after this warning msg. i'm worrying if it will crashes asterisk sometimes. |
| 16:16.57 | Zeeek | agreed - which is why it's no good pushing linux on people happy with Windoze |
| 16:17.03 | ctooley | Ok, I'm stumped, how do I make my IAXy or Asterisk wait until I'm done dialing before trying to dial "1" |
| 16:17.06 | [TK]D-Fender | Zeeek : I agree for the most part. I advocate XP w/ OSS software at the office myself. |
| 16:17.34 | ctooley | This is a simple problem I'm sure of it, I just don't use these kinds of devices as a rule. |
| 16:17.39 | Zeeek | so on this topic, anyone using OpenVPN w/ Win XP + Linus ? |
| 16:17.55 | ctooley | Zeeek, I have some customers doing that |
| 16:18.07 | Zeeek | ctooley I think it's a matter of timeout as in digit |
| 16:18.20 | Zeeek | and the order of evaluation of your dialplan |
| 16:18.27 | [Sim] | evening |
| 16:18.34 | ctooley | Zeeek, in the iax.conf for the iaxy or in the dialplan? |
| 16:18.41 | Zeeek | I installed OpenVPN on two XP boxes and never got them talking |
| 16:18.48 | Zeeek | extensions.conf - the dialplan |
| 16:18.56 | [Sim] | anyone noticed asterisk (stable) doing bad audio over iax links on multiple nic machines ? |
| 16:19.08 | [Sim] | I'm trying to get a grip on this thing |
| 16:19.10 | ctooley | Zeeek, there's only one extension in the context that the the IAXy uses. |
| 16:19.19 | ctooley | _1NXXNXXNXXX |
| 16:19.30 | Zeeek | and what happens when you start dialing? |
| 16:19.36 | Zeeek | it dials the 1 right away? |
| 16:19.38 | ctooley | I dial 1 and get a busy signal |
| 16:19.41 | ctooley | yeah |
| 16:19.53 | Zeeek | ok, can't be. It sees more extensions somehow |
| 16:20.06 | Zeeek | the IAXy is in its own context? |
| 16:20.31 | ctooley | yes |
| 16:20.33 | zipp | [Sim], what codec? |
| 16:21.29 | [Sim] | zipp: g711 all the way |
| 16:21.33 | [Sim] | plenty of bandwith |
| 16:21.42 | [Sim] | jitterbuffer on or off makes little difference |
| 16:21.44 | Zeeek | ctooley how many lines in the context? can you patebin it? |
| 16:21.49 | zipp | [Sim], try gsm |
| 16:21.58 | ctooley | Zeek, yeah... just a sec |
| 16:22.05 | [Sim] | zipp: can't - faxes on the line |
| 16:22.07 | zipp | I don't really think it is anything to do w/ multiple interfaces |
| 16:22.12 | [Sim] | but even if I could: why should it matter ? |
| 16:22.14 | [Sim] | well |
| 16:22.30 | [Sim] | we recently started introducing multihomed servers in the network |
| 16:22.45 | [Sim] | and we dont seem to have it on the singlehomed boxes |
| 16:22.47 | ctooley | http://pastebin.ca/6787 |
| 16:22.52 | *** join/#asterisk marc324 (~marc32344@69-28-224-214.dsl.teksavvy.com) |
| 16:22.52 | zipp | [Sim], ipv6 :) |
| 16:22.58 | [Sim] | right |
| 16:23.43 | Zeeek | ctooley what about CLI output for that? |
| 16:23.51 | zipp | [Sim], ping flood 100%? |
| 16:23.58 | [Sim] | nothing |
| 16:24.06 | [Sim] | its all nice and quick ethernet |
| 16:24.10 | [Sim] | no congestions, no packetloss |
| 16:24.19 | ctooley | aha! |
| 16:24.52 | ctooley | forgot to make the 2 line entry in extensions.conf to make it look for the gnumberfax context in Realtime |
| 16:24.58 | *** join/#asterisk point (1000@213.27.44.55) |
| 16:25.09 | Zeeek | I don't know anything about realtime anyway |
| 16:25.21 | Zeeek | just looking at it gives me chills |
| 16:25.22 | JerJer[mobile] | Zeeek: good, you don't want to know |
| 16:25.47 | Zeeek | I'm an old solder and rosin guy, I don't want to add all those layers to my little pbx |
| 16:25.57 | Zeeek | no X, no mysql |
| 16:26.11 | Zeeek | I do allow Apache but it's independent |
| 16:26.11 | flewid | sup |
| 16:26.55 | Katty | hmmmmmm. |
| 16:27.01 | ctooley | all better |
| 16:27.05 | Katty | hi, etc. |
| 16:27.08 | Zeeek | what does mobile mean, JerJer ? You logged in on a cellphone on the Freeway or something? |
| 16:27.14 | [Sim] | zipp: any thoughts ? |
| 16:27.23 | Katty | isn't there something happened today? |
| 16:27.31 | tzanger | Katty: I got out of bed... |
| 16:27.37 | Katty | tzanger: excellent |
| 16:27.43 | Katty | Zeeek: :> |
| 16:27.56 | Zeeek | God I love when you say it like that |
| 16:27.58 | Katty | hmm, dev conf 1pm cst march 3rd |
| 16:28.04 | Zeeek | yes, YES! |
| 16:28.06 | Katty | that's today |
| 16:28.09 | Katty | and I"M HERE |
| 16:28.11 | Zeeek | so it is |
| 16:28.16 | Katty | but not for long |
| 16:28.17 | Zeeek | so you is |
| 16:28.26 | Zeeek | so you was |
| 16:28.35 | Katty | i have upload pretty .wav mario brothers stuff to the polycom sip 500 |
| 16:28.44 | tzanger | Katty: hahaha |
| 16:28.47 | flewid | haha |
| 16:28.50 | flewid | that's gold |
| 16:28.54 | Zeeek | *you have an ip500? |
| 16:28.57 | Katty | yes |
| 16:28.58 | tzanger | Katty: put the wavs up |
| 16:29.02 | Katty | 14 of them, actually |
| 16:29.03 | Zeeek | are you liking it? |
| 16:29.03 | eKo1 | mario bros. 1? |
| 16:29.04 | *** join/#asterisk Nukemizer (~Nuke@65.103.231.133) |
| 16:29.07 | Zeeek | them? |
| 16:29.10 | Katty | Zeeek: they're all teh fluffy |
| 16:29.11 | flewid | i was considering putting the theme song for 'the saint' |
| 16:29.19 | Katty | Zeeek: yes, i have 14 sip 500s and a snom 1990 |
| 16:29.20 | flewid | so all my agents could look around mysteriously before answering the line |
| 16:29.20 | tzanger | flewid: ugh |
| 16:29.21 | Katty | erm, 190 |
| 16:29.24 | tzanger | musical ringtones suck |
| 16:29.29 | tzanger | flewid: hahaha |
| 16:29.33 | Zeeek | yeah they get old REALLY fast |
| 16:29.35 | tzanger | flewid: ok you're fogiven |
| 16:29.39 | flewid | tzanger: :) |
| 16:29.39 | tzanger | forgiven even |
| 16:29.55 | spackle | Anybody have some hints on how to register the netweb eezee phones with IAX? |
| 16:30.02 | flewid | spackle: |
| 16:30.02 | Zeeek | you all read that Martha Stwerat threatened to fire her broker if he didn't change the moh on his pbx? |
| 16:30.06 | flewid | i wrote up a huge howto on the wiki |
| 16:30.17 | tzanger | Zeeek: haha |
| 16:30.17 | flewid | look for atcom on the wiki |
| 16:30.18 | Zeeek | She's getting out of jail RealSoonNow |
| 16:30.23 | spackle | is it listed under that phone or another? |
| 16:30.24 | zipp | [Sim], do you use spandsp? |
| 16:30.25 | Katty | MOooooooooooOO!!!!!!!! <- next ringtone |
| 16:30.25 | Zeeek | like tomorrow, I think |
| 16:30.30 | tzanger | Zeeek: I heard that she's lost 15lbs on the tossed salad diet... |
| 16:30.37 | flewid | spackle: atcom |
| 16:30.42 | spackle | k, thanks. |
| 16:30.44 | Zeeek | and the excercize |
| 16:30.45 | flewid | np |
| 16:30.51 | Katty | i was thinking about setting up 'moo moo....moo moo' etc for ringtone as well |
| 16:31.11 | flewid | i have a question - i have * running on a box, but every time i call in it gives me gsm frametolin errors |
| 16:31.21 | flewid | but this is from pstn directly to zap, why is it auto defaulting to using gsm? |
| 16:31.24 | Zeeek | we had a guy installing a floor at the office - he had the greatest cellphone ring ever |
| 16:31.26 | flewid | all i hear are a series of fast clicks |
| 16:31.29 | flewid | then * hangs up on me |
| 16:31.39 | Zeeek | sounded like a 100 pound alarmclock bell |
| 16:31.47 | JerJer[mobile] | flewid: something is playing a gsm file then or somethng |
| 16:32.08 | *** join/#asterisk folsson (~filip@h87n2fls31o985.telia.com) |
| 16:32.11 | flewid | JerJer[mobile]: well, i'm just testing it out eh, and i put in the incoming context 'playback(goodbye)' |
| 16:32.17 | flewid | just for testing, where that should work shoudl it not? |
| 16:32.18 | Zeeek | flewid and how do your IRQs look? |
| 16:32.27 | flewid | 4 for tdm 1, 7 for tdm2 |
| 16:32.34 | flewid | sec i'll get the actual error |
| 16:32.54 | Zeeek | clicking on POTS hw is often IRQ conflicts |
| 16:33.07 | Zeeek | or let's say vice verse |
| 16:33.29 | [Sim] | zipp: only on one of the boxes |
| 16:33.49 | flewid | zeek: but it seems to be when it's playing gsm files, and it's working fine if i call in and it auto rings zap/4 |
| 16:34.57 | flewid | yeah it's the |
| 16:35.10 | flewid | 'file codec_gsm.c gsmtolin_framein: invalid gsm data |
| 16:35.20 | flewid | (using the * supplied sounds) |
| 16:37.38 | Essobi | Has anyone done a rule of thumb or live burnins for the number of SIP channels an * box can maintain with SIP re-invite running? |
| 16:37.55 | Essobi | I'd be interested in finding out.. |
| 16:37.55 | Katty | what is the SNTP server? |
| 16:38.07 | flewid | your ntp server |
| 16:38.15 | Katty | better be more clear |
| 16:38.17 | Katty | what does it handle? |
| 16:38.18 | flewid | Essobi: have you checked out astertest.org? |
| 16:38.28 | flewid | Katty: network time protocol |
| 16:38.30 | flewid | port 123 |
| 16:38.30 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 16:38.36 | Essobi | flewid I heard about it. |
| 16:38.38 | Katty | flewid: k |
| 16:38.47 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || 1.0.6 Released || Dev Conf 1PM CST MARCH 3rd -> IAX2/guest@66.250.68.194/996 || ClueCon Dev Conf Aug 3rd - 5th |
| 16:38.57 | Katty | hi bkw_ (= |
| 16:39.04 | flewid | astertest might have some info for ya |
| 16:39.19 | ariel_ | hello Katty |
| 16:39.20 | flewid | ok time for coffee |
| 16:39.23 | Katty | hi ariel! |
| 16:39.28 | Katty | uh oh. |
| 16:39.32 | Katty | kid problems? |
| 16:39.40 | ariel_ | yes |
| 16:39.43 | Katty | bkw_: let me know when you have more info about the chicago thing (= |
| 16:39.46 | Katty | ariel_: sick? :< |
| 16:40.16 | ariel_ | no she just wants to be held. I can't work she yells and just wants to be pickup all the time today. |
| 16:40.21 | pif | bad day? have a beer |
| 16:40.31 | Katty | ariel_: uh oh... |
| 16:40.31 | Zeeek | doing that now |
| 16:40.32 | ariel_ | pif, don't drink. |
| 16:40.38 | pif | you should |
| 16:40.38 | Katty | ariel_: i bet someone isn't feeling well :< |
| 16:41.05 | pif | PROSIT! |
| 16:41.09 | Katty | gosh |
| 16:41.10 | Zeeek | salut |
| 16:41.12 | Zeeek | santé |
| 16:41.16 | Zeeek | à la tienne |
| 16:41.17 | RoyK | skål |
| 16:41.21 | Zeeek | mud in your eye |
| 16:41.26 | Zeeek | sköl |
| 16:41.29 | Zeeek | I thought |
| 16:41.31 | Katty | s/mud/mudd |
| 16:41.38 | pif | bud? |
| 16:41.45 | *** join/#asterisk heison (~heison@ns.somanetworks.com) |
| 16:41.50 | Zeeek | sadder budweiser |
| 16:42.17 | Katty | so, what's something fun to do with this sip500 |
| 16:42.24 | Katty | mass spam everyone with text messages? |
| 16:42.30 | Zeeek | there's only one thing better than cheesy nachos and an ice cold beer, and I'm not gonna go into that here |
| 16:43.09 | Zeeek | "Le numéro le plus fort qu'une biere puissse vous faire" |
| 16:43.10 | pif | getting a blowjob with a beer in hand is my next task |
| 16:43.19 | Katty | gosh |
| 16:43.28 | Katty | Zeeek: :< |
| 16:43.41 | Zeeek | we're getting roudy again |
| 16:43.51 | Katty | you are |
| 16:44.13 | Katty | ariel_: bring me the kid, i'll hold (+ |
| 16:44.14 | NotchJohnson | heh |
| 16:44.14 | Katty | (= |
| 16:44.37 | NotchJohnson | wake up wife, bad. Very bad. |
| 16:45.01 | NotchJohnson | Drums stop, not good. |
| 16:45.04 | pif | let Mr Johnson wake up wife |
| 16:45.18 | innerweb | A russian au pair? That sounds very intriguing. |
| 16:45.19 | NotchJohnson | <groucho voice> depends on whose wife |
| 16:45.45 | Essobi | maha |
| 16:45.45 | NotchJohnson | innerweb are you the one with the WRT54g with asterisk? |
| 16:45.57 | NotchJohnson | hi cousin |
| 16:46.06 | NotchJohnson | where's BJ ? |
| 16:46.21 | innerweb | Uh, no, no WRT54g, more like a WRTG radio station. 8-) |
| 16:46.25 | Essobi | You poor bastard. |
| 16:46.37 | NotchJohnson | no that was the other young fellow |
| 16:46.47 | flewid | i need to get an fm transmitter for my icecast server |
| 16:46.51 | innerweb | I have a friend with an au pair from Georgia (of ols USSR). |
| 16:46.54 | flewid | it'd be nice to have 9 radiostations in the house |
| 16:46.54 | big_bjohnson | NotchJohnson: few people running * on linksys. Jerjer is one |
| 16:47.02 | Essobi | No shit? |
| 16:47.11 | Essobi | how many channels can you get on a linksys? |
| 16:47.17 | big_bjohnson | 4 he says |
| 16:47.18 | innerweb | She does thier laundry, cooking, child care and a few other things.. |
| 16:47.19 | NotchJohnson | there was someone on a couple hours ago asking - jeofrey I think |
| 16:47.23 | pif | au pair's are horny I can tell you |
| 16:47.25 | big_bjohnson | gotta go |
| 16:47.27 | Essobi | 4 What? |
| 16:47.28 | Essobi | SIP? |
| 16:47.39 | Essobi | 711? &29? |
| 16:47.39 | pif | they usually are pretty lonely |
| 16:47.50 | innerweb | Ona linksys? Oh, cool. What piece of equip? |
| 16:47.57 | NotchJohnson | WRT54g |
| 16:48.09 | Katty | is there a pretty little soft phone that can dial an ip (for connecting via VPN and then dialing an internal IP) |
| 16:48.10 | NotchJohnson | I have one a few feet away |
| 16:48.10 | zipp | g/gs |
| 16:48.27 | zipp | Katty, for sip or IAX2? |
| 16:48.31 | Katty | sip |
| 16:48.34 | NotchJohnson | They really should have put a switch on the radio part of these routers tho. I always leave it off until I need it |
| 16:48.51 | innerweb | Really? I never knew (nor tried to know). SHe is very friendly (a Phd in Physics, could not get a good job), multi-lingual and can sing. |
| 16:48.52 | zipp | anyone have any comments on the spa2100? |
| 16:48.56 | zipp | I ordered 1 |
| 16:48.57 | NotchJohnson | but I have to boot something up to turn the wifi part on |
| 16:48.59 | Katty | i see. |
| 16:49.06 | Katty | i'll take that as a no. |
| 16:49.31 | Katty | does anyone /else/ know of a pretty little software phone for dialing an ip (for vpn and internal ip) |
| 16:49.45 | NotchJohnson | X-Lite does that real nice |
| 16:49.52 | Katty | k |
| 16:50.00 | flewid | anyone here using that pulver communicator thing? |
| 16:50.06 | NotchJohnson | tried it, didn't like it |
| 16:50.07 | flewid | it seems to want 5060 and 5082 |
| 16:50.15 | innerweb | I have always wanted a phone system that tied my computer into so I could do some *neat* things. Now, I guess I want a phone that I can attach to my computer. lol |
| 16:50.16 | flewid | i can't get it to traverse my firewall |
| 16:50.20 | pif | Katty : asterisk does it too:" > dial 6465465465" |
| 16:50.21 | NotchJohnson | that's their STUN server or proxy |
| 16:50.32 | flewid | did you have to forward anything? |
| 16:50.32 | TSCHAK | say I have a voicemail extension 1234 on an asterisk server...how do I tell Cisco Callmanager to route through the SIP trunk to that specific extension ? |
| 16:50.35 | zipp | flewid, welcome to SIP |
| 16:50.36 | zipp | ha |
| 16:50.39 | flewid | zipp: heh |
| 16:50.41 | pif | ok, it's command line but it's FAST |
| 16:51.12 | pif | and it's GEEKY, and it's SEXY, hummm never mind... |
| 16:51.27 | NotchJohnson | flewid I don't think I waited long enough to get it to work |
| 16:51.29 | pif | beer++ |
| 16:51.33 | flewid | NotchJohnson: hehehe |
| 16:51.41 | tzanger | command lines are sexy? you *definitely* need to get out more |
| 16:51.44 | NotchJohnson | I tried the beta |
| 16:51.51 | flewid | yeah it looks kina weird, but it says it supports msn/aim/et al |
| 16:51.58 | tzanger | flewid: what's weird |
| 16:51.58 | flewid | so i was toying with replacing trillian with it |
| 16:52.00 | *** join/#asterisk hajekd (~hajekd@mail.idoox.com) |
| 16:52.05 | tzanger | I use psi for my IM |
| 16:52.08 | pif | tzanger : yeah, command line is the epitome of sexiness |
| 16:52.10 | NotchJohnson | but the only real advantage is that it groups FWD, AIM, IM, CQ and all that into one client |
| 16:52.13 | innerweb | Nothing sexier in the world than a very smart lady who is in love with you. |
| 16:52.15 | tzanger | multiplatform and jabber |
| 16:52.16 | zipp | is there a simple sip library, like IAX client? |
| 16:52.20 | flewid | psi is nice |
| 16:52.27 | tzanger | innerweb: she's got to be attractive too or it all falls apart |
| 16:52.27 | flewid | tzanger: what jabber server you using? |
| 16:52.28 | innerweb | I have been married for 14 years, and I know beautiful. |
| 16:52.30 | NotchJohnson | whazzzat? (psi) |
| 16:52.36 | tzanger | you need brains and looks and the ability to communicate |
| 16:52.40 | tzanger | psi.affinix.com |
| 16:52.42 | flewid | NotchJohnson: it's a jabber client |
| 16:52.47 | innerweb | Would ex-boardwalk model work? |
| 16:52.48 | NotchJohnson | k |
| 16:52.49 | flewid | tzanger: we were running uhm |
| 16:52.51 | boch | innerweb im agree with you |
| 16:52.54 | [Sim] | zipp: any ideas about my Q ? |
| 16:52.58 | tzanger | innerweb: I am dating one of those actually |
| 16:53.01 | zipp | [Sim], nope |
| 16:53.02 | zipp | sorry |
| 16:53.03 | flewid | shit now i forget the name of the server we were trying |
| 16:53.17 | [Sim] | drat |
| 16:53.36 | innerweb | My wife used to model, she can tell you the chemical compositions of most cosmetics, and what they do to you, she can tell you wha tmost vitamins/nutrients are and how they work in the body. SHe can cook (traditional polish),... |
| 16:53.40 | flewid | jive messenger |
| 16:53.45 | tzanger | I told her she needs to go back to sicily though to get her head back togrether (she's had some tough times lately) |
| 16:53.56 | NotchJohnson | innerweb - dobjé |
| 16:53.57 | tzanger | innerweb: sounds really, *really* good |
| 16:54.03 | pif | and my gf is a prima ballerina at the paris opera |
| 16:54.04 | tzanger | innerweb: you're a very fortunate man :-) |
| 16:54.05 | innerweb | And just for kicks, she used to model on the runway. However, she is kind of tall. lol |
| 16:54.21 | NotchJohnson | and looks like Claudia Schiffer, right? |
| 16:54.40 | NotchJohnson | or you mean like Boeing 747 runway |
| 16:54.52 | innerweb | I dated a ballerina From Norway when I was a teen. I wans not smart enough to make the needed effort for a relationship. SHe became a member of the royal ballet. She was a sweet as they came (and very smart). |
| 16:54.53 | MajestiK | 2 SPA-2000 questions: Is it possible to have one locked to a provider, like a linksys PAP2? And, how many phones can you put on each of the phone jacks? |
| 16:55.19 | tzanger | I've dated all kinds of women and I find I'm naturally attracted to the ones who you can actually hold a conversation with |
| 16:55.31 | hajekd | someone here has sipura 841? |
| 16:55.38 | innerweb | Notch 0-> Nowhere near. Tall, dirty blonde, green eyes, pale (like Polish), size 6 to 8 (we are not kids anymore). |
| 16:55.58 | Zeeek | you're old I know |
| 16:55.59 | hajekd | is it possible to have backlight on the lcd turned on? |
| 16:56.26 | Zeeek | how do you like that phone, hajekd? |
| 16:56.31 | pif | ballerinas tend to be quite clever, independent, strong-willed gals, and these incredible extensions when making love, it's scary! |
| 16:56.32 | innerweb | Now the good stuff. Reads about one book a week on nutrition (not one of those fad books), cosmetics or health per week, and knows how to apply it for the family. |
| 16:56.54 | Zeeek | innerweb, ok, let's exchange wives for week |
| 16:56.59 | innerweb | pif -> I would not know. I never touched Nina that way. I just was lucky enough to be in her compnay. |
| 16:57.07 | hajekd | zeek: I'm using it just shortly for few mins.. |
| 16:57.09 | tzanger | innerweb: sounds like my gf... Dr in psychology |
| 16:57.11 | pif | call her some day |
| 16:57.13 | eKo1 | Any use mediatrix fxo gateways? |
| 16:57.26 | Juggie | is anyone going to von canada in april? |
| 16:57.30 | tzanger | Juggie: doubtful |
| 16:57.37 | tzanger | I don't find conferences useful as a general rule |
| 16:57.43 | hajekd | zeek: but looks good, but don't know if it is possible to turn on the lcd backlight |
| 16:57.45 | tzanger | if you're gonna meet up with a bunch of people there that's one thing |
| 16:57.48 | k_bot | Juggie: thanks for helping me out last night... your suggestions fixed my problem |
| 16:57.48 | Juggie | well, work may send me for free :) |
| 16:57.55 | tzanger | but trade shows are just pomp |
| 16:57.56 | innerweb | Zeeek -> Thanks anyway, It has taken me this long to get to know her, and one week with a stranger would leave me a nervous wreck (I am really quite shy). |
| 16:58.00 | Juggie | k_bot, no problem... thats whet were here for :) |
| 16:58.24 | Zeeek | ok, it was just a thought; a lot of women would put up with me for a week in Paris |
| 16:58.30 | innerweb | Zeeek-> Yeah, older, not old. |
| 16:59.13 | flewid | Juggie: my boss was planning on sending me |
| 16:59.20 | innerweb | tzanger -> A smart woman is much better any day of the week than a woman who is not smart. Especially if she is tight with the money. |
| 16:59.21 | flewid | but i think i'm going to india instead |
| 16:59.24 | pif | french girls tend to use MSN messenger a lot to date |
| 16:59.28 | eKo1 | AH! I can't believe I forgot my usb flash drive... |
| 16:59.33 | *** join/#asterisk phantam (~phantam@72.252.15.235) |
| 16:59.33 | tzanger | innerweb: amen to that. |
| 16:59.35 | phantam | ok guys |
| 16:59.40 | phantam | im getting a headache from this |
| 16:59.42 | phantam | oh323 doesnt work well |
| 16:59.49 | Zeeek | pfi t'es en France? |
| 16:59.51 | phantam | and u said h323 should work if i follow the directions |
| 16:59.52 | tzanger | h323 doesn't work well. :-) |
| 16:59.53 | phantam | well guess what |
| 16:59.54 | innerweb | She hates French. Two of her old bosses were French and they were very bad envoys. |
| 16:59.54 | phantam | SF.net |
| 17:00.02 | phantam | doesnt even have the stupid "You must run Open H.323 v1.15.1 and PWLib v1.8.1. All other versions are not supported. |
| 17:00.05 | phantam | " |
| 17:00.20 | phantam | goto http....................... .... well guess what at that site those versions dont even exist |
| 17:00.21 | zipp | french people suck |
| 17:00.23 | zipp | ha |
| 17:00.30 | Zeeek | well, Poles often will tell you that both the French and the American constitutions are based on theirs |
| 17:00.37 | eKo1 | zipp: you mean stink. |
| 17:00.46 | zipp | yea, that too |
| 17:00.48 | innerweb | zipp -> I have a friend of mine dating a girl named Loli. He agrees. And he will not let us meet her. |
| 17:00.55 | tzanger | stereotypes abound |
| 17:00.58 | flewid | yeah really |
| 17:01.04 | *** join/#asterisk Red_6 (~alex@m174.net81-66-29.noos.fr) |
| 17:01.07 | pif | is french bashing still on in the US of A ? |
| 17:01.14 | tzanger | pif: appears so |
| 17:01.14 | flewid | pif: when hasn't it been |
| 17:01.16 | Zeeek | even kidding around, it's too bad to use those |
| 17:01.26 | zipp | pif, it will never end |
| 17:01.39 | flewid | pif: people like to make fun of those they don't understand |
| 17:01.43 | *** join/#asterisk JunK-Y (~grepmoo@65.39.228.5) |
| 17:01.46 | innerweb | LOL -> I know many French (we hava French company up the road). We neer got into it here (except the rednecks). |
| 17:01.48 | zipp | pif, yea |
| 17:02.12 | hajekd | sipura 841 does not have display backlight...so silly |
| 17:02.21 | flewid | my ip500 doesn't have a backlight either |
| 17:02.27 | innerweb | Most of the French I know are very nice. They have very strong opinions, but who does not? |
| 17:02.33 | phantam | let me guess jerjer isnt here |
| 17:02.33 | jontow | nor my cisco 7960 (?) |
| 17:02.37 | phantam | nor any of the developers |
| 17:02.41 | sambal | nor my cisco 7940 ;) |
| 17:02.46 | flewid | heh |
| 17:02.55 | flewid | i hear the farfon will have 320/240 backlight |
| 17:02.55 | sambal | but my grandstream has :D |
| 17:02.57 | jontow | better question: what phones DO have backlights? |
| 17:02.57 | flewid | tha'tll be shaweet |
| 17:03.06 | jontow | sambal; same here ;) |
| 17:03.07 | flewid | jontow: the pa-168s bsed (yuxin) |
| 17:03.08 | pif | and most USians i know are cool ppl, so why all that hatred? |
| 17:03.11 | sambal | blue |
| 17:03.15 | flewid | jontow: and the grandstreams |
| 17:03.17 | pif | it's stupid really |
| 17:03.23 | Zeeek | farfon has no backlight that I've noticed |
| 17:03.23 | flewid | and cisco 794 or whatever |
| 17:03.25 | flewid | that 1k one |
| 17:03.25 | Beirdo | jontow: my Vista 350 does :) |
| 17:03.30 | flewid | Zeeek: no, not yet |
| 17:03.33 | Beirdo | but it's not SIP, it's ADSI |
| 17:03.34 | flewid | Zeeek: it's planned |
| 17:03.35 | innerweb | I do not know. I am not a typical American. |
| 17:03.36 | sambal | 7970? |
| 17:03.38 | zipp | flewid, when is farfon _actually_ going to be released? |
| 17:03.48 | sambal | and maybe that new fancy motorola phone will have |
| 17:03.48 | flewid | zipp: they've shipped beta units afaik |
| 17:03.48 | Zeeek | zipp I have one now in testing |
| 17:04.01 | flewid | Zeeek: how is it? |
| 17:04.06 | Zeeek | interesting |
| 17:04.10 | flewid | i'm using the pa-168s based atcom phone and it's alright |
| 17:04.16 | pif | what is the latest version of 7960 SIP software? |
| 17:04.17 | flewid | a little bit of static on the pickup of the headset |
| 17:04.19 | flewid | but it's fine otherwise |
| 17:04.20 | Zeeek | from iaxtalk? |
| 17:04.22 | innerweb | I guess fear. Our country is no longer teh powerhouse it once was. And people are not living as well as they used to, and jobs are getting harder to come by and providing less (overseas competition and bad bosses). |
| 17:04.24 | jontow | .. gonna test jeff rizzo's netbsd wcfxo drivers :) |
| 17:04.26 | flewid | Zeeek: naw, from seshu |
| 17:04.29 | flewid | (netweb) |
| 17:04.50 | jontow | has anyone tried them out? |
| 17:04.51 | Zeeek | I'm considering an order from iaxtalk |
| 17:04.57 | sambal | http://broadband.motorola.com/consumers/products/ojo/index2.html |
| 17:04.59 | sambal | fancy phone |
| 17:04.59 | flewid | Zeeek: where you located? |
| 17:05.03 | Zeeek | Paris, France |
| 17:05.05 | flewid | ah |
| 17:05.09 | flewid | yeah iaxtalk would be your best bet |
| 17:05.16 | Zeeek | $40 shipping |
| 17:05.21 | flewid | i got mine from seshu like 2 days after i ordered it |
| 17:05.23 | flewid | shit |
| 17:05.25 | Zeeek | but I wonder if the phones are worth it |
| 17:05.27 | flewid | i paid 5$ shipping |
| 17:05.34 | Zeeek | where are you, Taiwan ? |
| 17:05.39 | flewid | canada |
| 17:05.46 | *** part/#asterisk Stitch (~magliett@cpe-69-204-16-160.rochester.res.rr.com) |
| 17:05.47 | sambal | Zeeek: same in holland + 15 euros tax handling costs by the post office :S |
| 17:05.49 | Zeeek | where is shshu? |
| 17:05.51 | innerweb | Zeeek -> How do most French feel about US with GW in office? |
| 17:05.57 | flewid | zeek: he's in new jersey |
| 17:06.02 | flewid | (about 12 hours south of me) |
| 17:06.02 | Zeeek | I'm only 1/2 naturalized FR |
| 17:06.04 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 17:06.17 | innerweb | I am not natural anything (besides human). |
| 17:06.26 | flewid | i'm not even natural human |
| 17:06.34 | Zeeek | ok I'm coming back to the US in May - maybe I'll order one then along with my polycom and sipura |
| 17:06.43 | innerweb | I am pretty sure that I am not an alien plant. |
| 17:06.50 | Zeeek | innerweb but to asnwer your question |
| 17:07.01 | flewid | Zeeek: the atcoms talk to you too eh |
| 17:07.01 | flewid | heh |
| 17:07.06 | flewid | if you press the 'my ip' button |
| 17:07.18 | flewid | the phone says 'your ip is x x x . x x x . x x x . x x x |
| 17:07.21 | flewid | it's pretty funny |
| 17:07.44 | Zeeek | oh yeah, my mobo says that "booting you opah-rate-ing system" |
| 17:07.50 | flewid | haha |
| 17:07.51 | Zeeek | cute |
| 17:07.52 | flewid | damn asus |
| 17:08.03 | innerweb | I am writing perl code to link the asterisk into a PIM, Calendat, order managemtn system, etc. For a chain of medical supply offices. |
| 17:08.07 | Zeeek | I had to go to ASUS board and ask what the mobo was saying when it booted! |
| 17:08.15 | flewid | haha |
| 17:08.20 | flewid | you can turn that off eh |
| 17:08.22 | pif | how much does that OJO phone cost? |
| 17:08.28 | flewid | is the ojo the video phone? |
| 17:08.36 | pif | yep |
| 17:08.39 | flewid | i emailed their sales distributor, and he said they don't have pricing yet |
| 17:08.43 | flewid | and that it would be out q2 05 |
| 17:08.47 | pif | it looks really nice |
| 17:08.47 | flewid | but i ahven't heard anything more from him |
| 17:09.11 | flewid | i should be getting an advance test unit in the mail soon tho |
| 17:09.20 | Zeeek | innerweb et al, everything you read about France and Europe is bullshit. People are people. The French have always liked and admired Americans and frankly, most intellignet Amreicans admire and like the French - and they have been here to know what it's like |
| 17:09.24 | flewid | some company in ip telephony magazine is giving them away if you call and bullshit a bit |
| 17:09.27 | flewid | and ask for more information |
| 17:09.31 | innerweb | has anyone tied * into anything like PIM, RCving, Shipping, etc? |
| 17:09.33 | flewid | :) |
| 17:09.55 | flewid | zeeek |
| 17:10.12 | Zeeek | innerweb and GW B is a passing thing, it's over in 4 years. I have a customer whose business is 400 years old? Thats' older than nuphone, boys, hello? |
| 17:10.13 | Juggie | all this stuff being said i'd like to point out that it is best to keep on topic as best we can here. |
| 17:10.20 | flewid | hehe |
| 17:10.24 | Juggie | and that this isnt the place for french vs amercains and such. |
| 17:10.26 | Zeeek | nufone even |
| 17:10.42 | TSCHAK | hmm |
| 17:10.51 | Zeeek | there is no conflict here - which is exactly what I'm saying - ask Mark - he comes here often |
| 17:10.55 | TSCHAK | has anyone in here worked with asterisk/cisco callmanager? |
| 17:11.07 | Beirdo | you guys should just make a meetme to bullshit about politics :) |
| 17:11.12 | pif | yeah, mark has an appartment in paris |
| 17:11.12 | *** join/#asterisk jeofrey (~jeofrey@espeed19-153.brunet.bn) |
| 17:11.21 | pif | well in la Defense |
| 17:11.21 | stepcutHM | I had dreams about VoIP last night :( |
| 17:11.26 | Juggie | zeek, let it go. |
| 17:11.34 | Zeeek | no one with an IQ > 50 believes what they see on TV, especially Fox tv or what your read in the paper. It's all balooney |
| 17:11.40 | flewid | haha fox |
| 17:11.50 | flewid | it's like the cartoon network of news |
| 17:11.50 | Zeeek | Ok, I just answered a question - we're in a conversation here |
| 17:11.51 | flewid | :) |
| 17:11.56 | zipp | we know who Zeeek voted for :) |
| 17:11.59 | pif | fox tv? is that about cute girls? |
| 17:12.03 | Zeeek | end of free-speach |
| 17:12.04 | fitzel | Zeek, but most are below 50 |
| 17:12.06 | innerweb | Zeeek -> lol. Yeah, I know that. Most people over here do not seem to though (except those that bother to learn.) I do telecomute work for people all over the world, and most ask me why we hate so much. I always say I don't know, I don't hate. |
| 17:12.13 | flewid | pif: no, it's about bill o'reilly shooting off his mouth |
| 17:12.14 | flewid | :) |
| 17:12.41 | innerweb | pif -> How old are you? |
| 17:12.50 | pif | 39 |
| 17:13.21 | pif | I know, I sound younger :) |
| 17:13.31 | innerweb | pif-> Too bad, I could have referenced you to a lady in Italy (she does not go over thirty years olds) who is a techno-hack and loves tech guys. |
| 17:13.48 | Zeeek | We are on topic |
| 17:13.50 | pif | but my dick is 29 |
| 17:13.51 | flewid | Juggie: i got a q since you're so interested in on topic stuff :) |
| 17:13.56 | Zeeek | asterisk the Open Source pbx |
| 17:13.56 | innerweb | She is a masseuse with a wild life style. I would never date her (I coulnd not keep up), but she is SMART. |
| 17:14.08 | flewid | Juggie: fresh install of asterisk, when i all up via zap on the tdm card, anytime i try and play a sound |
| 17:14.24 | flewid | all i hear from the phone is fast clicking, and asterisk debugging shows me that gsm frametolin isn't working |
| 17:14.26 | flewid | and it can't play the gsm file |
| 17:14.35 | flewid | (this is stock asterisk sounds, stock asterisk install, irq's aren't shared) |
| 17:14.53 | innerweb | I have written code for her before, and she pays well. She is thinking about setting up a PBX for her Salon. |
| 17:14.58 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 17:15.18 | pif | innerweb : thanks for the thought though |
| 17:15.19 | Juggie | flewid, irq conflict? |
| 17:15.30 | flewid | Juggie: tdm1 is on irq 4, tdm2 is on irq 7 |
| 17:15.31 | Zeeek | I already said that hours ago |
| 17:15.38 | flewid | using ACPI , no apm |
| 17:15.40 | flewid | kernel 2.6.10 |
| 17:15.44 | flewid | gentoo latest |
| 17:16.14 | flewid | it works if i just have the incoming context dial a zap channel directly |
| 17:16.17 | flewid | i can pick up and talk fine |
| 17:16.30 | flewid | but if i try and make an ivr and play soome sounds, no workie |
| 17:17.26 | Zeeek | Juggie have you been around here for ages, like a year or many months? |
| 17:19.05 | flewid | Juggie: any other ideas? |
| 17:19.09 | flewid | i'm kinda at a loss here |
| 17:19.11 | Zeeek | asterisk is great! |
| 17:19.15 | flewid | i need to get this server shipped to india in 2 days |
| 17:19.46 | Zeeek | my wife just called - she's leaving the office. I cover the phone from home thanks to Digium (tm) and asterisk |
| 17:20.21 | Zeeek | flewid what format are you using for ivr sound files? |
| 17:20.25 | *** join/#asterisk Nebukadneza (~daddel9@i3ED6E583.versanet.de) |
| 17:20.49 | phantam | where he hell do i get the proper pwlib and openh323 from? |
| 17:20.50 | phantam | for h323 |
| 17:21.17 | flewid | Zeeek: they are just the stock gsm format ones that come in asterisk-sounds |
| 17:21.23 | flewid | my dialplan is simple, someone calls in |
| 17:21.32 | flewid | it Playback(all-your-base) |
| 17:21.33 | Nebukadneza | hi |
| 17:21.42 | flewid | then Playback(thank-you-cooperation) |
| 17:21.43 | flewid | then hangup |
| 17:21.58 | Zeeek | and you get noise on this |
| 17:22.02 | flewid | yeah |
| 17:22.03 | Zeeek | no frame eror msgs |
| 17:22.05 | flewid | rapid clicking |
| 17:22.08 | RoyK | 5400 context switches per second... that a lot? |
| 17:22.08 | Zeeek | yeah |
| 17:22.22 | Zeeek | not if you're talking about my brain |
| 17:22.25 | Nebukadneza | is SIP also "p2p" (i mean - 2 people got a sip client or somethin running and jep ... just phone by the ip of each other) |
| 17:22.33 | Zeeek | I do that in a typical conversation |
| 17:22.43 | machinehd | how hard is it to stumble accross sip formware for the cisco 7960? |
| 17:22.47 | Zeeek | Nebu yes you can do that |
| 17:22.50 | Zeeek | get X-Lite |
| 17:22.54 | Zeeek | or Skype |
| 17:23.09 | Nebukadneza | (or asterisk and kphone) |
| 17:23.21 | Nebukadneza | skype isnt sip is it? |
| 17:23.23 | pif | plugging * on the skype service should be a no-brainer, right? |
| 17:23.25 | Zeeek | if the two are not behind NAT asterisk can be used to arrange the sessions and bridge the two and leave them |
| 17:23.39 | Zeeek | no, no way to bridge skype and asterisk |
| 17:23.45 | Zeeek | pif are you in France? |
| 17:23.48 | pif | isn't it SIP? |
| 17:23.51 | Nebukadneza | nah - i hate closedsource stuff like skyp |
| 17:23.51 | Nebukadneza | e |
| 17:23.56 | pif | Zeeek : yeah |
| 17:23.58 | Zeeek | no skype is not SIP at all |
| 17:24.02 | Zeeek | pif where? |
| 17:24.07 | Nebukadneza | i mean - i got my asterisk set up as a sip2capi and capi2sip gateway |
| 17:24.11 | pif | paris, why? |
| 17:24.18 | Nebukadneza | now i thought about maybe i can call other sip users too |
| 17:24.19 | Zeeek | pif try wengo - it's SIP |
| 17:24.23 | Zeeek | works well with astrisk |
| 17:24.29 | Zeeek | asterisk too |
| 17:24.33 | Zeeek | and asterix even |
| 17:24.43 | Katfood | tata |
| 17:24.45 | pif | Zeeek : thanks but when I need a softphone I use * |
| 17:24.56 | Zeeek | wengo is a SIP voIP provider |
| 17:25.07 | pif | oh sorry |
| 17:25.13 | Zeeek | unlimited calling in FR for 8eu§mo |
| 17:25.25 | Zeeek | they do have an Open Source SIP client |
| 17:25.30 | pif | pretty good |
| 17:25.38 | pif | URL? |
| 17:25.39 | TSCHAK | is someone able to dial 1234@64.74.186.249 ? |
| 17:25.44 | *** join/#asterisk rkjpl (~rk@66-214-188-218.gln-eres.charterpipeline.net) |
| 17:25.50 | Zeeek | and right now, "for a limited time only" a 1 euro beta program |
| 17:25.56 | Zeeek | Guess! |
| 17:26.03 | Zeeek | http://www.wengo.fr |
| 17:26.05 | pif | you work for them? |
| 17:26.16 | Zeeek | no but I was one of the first to connect asterisk to it |
| 17:26.22 | Zeeek | itr works pretty well |
| 17:26.35 | pif | sounds better than a freebox call? |
| 17:27.00 | Zeeek | and considering that a lot of people have unlimited dialing with freebox here, they have 087 numbers - free with wengo, 2c/min with FT |
| 17:27.13 | Zeeek | sounds like voIP - it ISvoIP |
| 17:27.36 | pif | well freebox is voip (sorta) and sound like shit |
| 17:27.39 | Zeeek | check out the forum - a lot of stuff posted, plenty about asterisk |
| 17:27.50 | Zeeek | not sorta, freebox is voIP |
| 17:27.56 | pif | checking it out as we talk |
| 17:27.58 | Zeeek | wengo is voIP (9telecom) |
| 17:28.10 | Zeeek | pif you are ze French people ? |
| 17:28.30 | pif | freebox is voip over atm, that's why "sorta" |
| 17:28.39 | Zeeek | bon tu vas pouvoir lire tout ça sans problèmes |
| 17:28.51 | Zeeek | atm, schma-tm |
| 17:28.55 | *** part/#asterisk TSCHAK (tschak@cuodan.net) |
| 17:29.00 | Zeeek | it ends up as voIP |
| 17:29.11 | Zeeek | especially if I'm on the other end with my own voIP |
| 17:29.24 | *** join/#asterisk Mother_ (~mother@93.Red-80-32-127.pooles.rima-tde.net) |
| 17:29.30 | Zeeek | freebox 20mbits ATM, 16 TCP/IP |
| 17:29.36 | Mother_ | hiyup |
| 17:29.40 | Zeeek | Mutha |
| 17:29.49 | Mother_ | Zeeeeeeeeeeeeeeek |
| 17:29.57 | Mother_ | :) |
| 17:30.25 | Mother_ | I'm having trouble with a 7960 accepting the RTP port range, it keeps going back to the defaults |
| 17:30.36 | SkySky | may i ask if u need to declare anything in order for sip calls to have music on hold enable? (right now, i have 2 local sip phones calling each other, and there's no music which holding) |
| 17:30.41 | Mother_ | no matter if I change it in the conf via TFTP or directly on the phone |
| 17:30.42 | Zeeek | what are the defaults? |
| 17:31.01 | Mother_ | 16384 to 32766 |
| 17:31.05 | Zeeek | SkySky you got moh working anywhere on the system? |
| 17:31.12 | pif | "37075679 bytes transferred in 24 seconds (1.47M/s)" that's the freebox effect! |
| 17:31.20 | pif | linux-2.6.11 |
| 17:31.21 | *** join/#asterisk stevekstevek (~stevekste@slim-eth0.horizonlive.net) |
| 17:31.24 | Zeeek | Mother_ if it's a major prb, you can change the asterisk ports |
| 17:31.36 | *** join/#asterisk calvinhp (~calvinhp@cpe-65-29-88-222.indy.res.rr.com) |
| 17:31.36 | *** join/#asterisk jarrod (jarrod@dipole.informationwave.net) |
| 17:31.37 | Mother_ | I've tried that too, but to no avail |
| 17:31.52 | Zeeek | pif you are on freebox? |
| 17:31.54 | rkjpl | Has Asterisk been used successfully in a pay-per-call [900, psyc, adult, etc..] model? |
| 17:32.08 | Mother_ | if I debug SIP on the * cli, I see the 7960 picks any port in that range, no matter what's in rtp.conf |
| 17:32.14 | SkySky | Zeeek: yes, its working on queue |
| 17:32.14 | rkjpl | Using phone-bill, cheque, credit card? |
| 17:32.20 | jarrod | what are people using for IAD devices to integrated voip into legacy phone systems? (anything besides cisco?) |
| 17:32.37 | pif | yep |
| 17:32.48 | pif | Zeeek : can you fax with wengo? |
| 17:32.57 | Zeeek | you happy with it? |
| 17:33.03 | pif | very happy |
| 17:33.06 | Zeeek | you can fax with asterisk |
| 17:33.30 | Zeeek | but I don't know about wengo - I have other solutions for fax when we need it |
| 17:34.03 | Zeeek | pif what are you doing here? Working in telcom biz? |
| 17:34.34 | pif | open source software integration, my own company of 5 dudes |
| 17:34.47 | Zeeek | you use asterisk |
| 17:34.56 | Zeeek | you met Mark when he was here? |
| 17:35.03 | pif | and we push * on our existing customers too |
| 17:35.13 | pif | yeah, briefly |
| 17:35.33 | *** part/#asterisk jcims (~jcims@cpe-69-135-121-57.columbus.rr.com) |
| 17:35.37 | pif | and you? |
| 17:35.39 | Zeeek | in Bordeaux or Paris? |
| 17:35.50 | pif | paris I told you |
| 17:35.51 | *** join/#asterisk powD (~atob100@83.146.53.34) |
| 17:36.01 | Zeeek | no I meant Mark - he was in Bdx last year |
| 17:36.10 | pif | nope in paris |
| 17:36.13 | Zeeek | there have been two lunch meetings with him |
| 17:36.28 | Zeeek | more, but I was at the last two |
| 17:36.35 | pif | the first on at "frogs and ..?" |
| 17:36.49 | Zeeek | no I wasn't there |
| 17:37.01 | Zeeek | anyway - though we might have met |
| 17:37.05 | Zeeek | thought |
| 17:37.10 | phantam | argggggggggggggg |
| 17:37.13 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-197-79.dsl.scarlet.be) |
| 17:37.17 | mogorman | rkjpl i have seen people using it there |
| 17:37.19 | pif | what do you do? |
| 17:37.23 | phantam | how do u build pwlibbbbbbbbb |
| 17:37.26 | phantam | it keeps erroring |
| 17:37.28 | phantam | arggg |
| 17:37.38 | Zeeek | I have a compnay here - we manage web sites for Bordeaux wine producers |
| 17:37.47 | Mother_ | can the rtp ports be limited in sip.conf? or only in rtp.conf? |
| 17:37.56 | pif | nice, you in bordeaux? |
| 17:37.57 | Zeeek | i'd say rtp only |
| 17:38.03 | Mother_ | ok |
| 17:38.05 | Mother_ | thanks |
| 17:38.06 | Zeeek | Paris but we go to Bodeaux often |
| 17:38.22 | Zeeek | Mother_ unauthorative answer |
| 17:38.25 | Mother_ | invented even |
| 17:38.31 | Zeeek | so use IAX |
| 17:38.36 | zipp | yes |
| 17:38.37 | Mother_ | with a 7960? |
| 17:38.41 | *** join/#asterisk zapa (~zapa@201.135.137.236) |
| 17:38.41 | Mother_ | heh |
| 17:38.51 | mogorman | i havent seen that patch... |
| 17:38.54 | Zeeek | yeah buy a US Army surplus converter |
| 17:39.08 | nestAr | wonder what non-answer they'll give me this time. |
| 17:39.20 | Zeeek | the nature of tech support is non asnwers |
| 17:39.28 | mogorman | ouch |
| 17:39.33 | JerJer[mobile] | cisco won't let me release an IAX firmware for the 7960s |
| 17:39.35 | ManxPower | Has anyone seen a problem where when calling a Polycom phone the Polycom doesn't ring? |
| 17:39.51 | mogorman | jerjer you have iax firmware? |
| 17:39.51 | Mother_ | lol |
| 17:39.56 | mogorman | indeed |
| 17:39.56 | *** join/#asterisk beto75 (~ha@201.133.240.35) |
| 17:39.57 | Zeeek | let's bring all this to light BEFORE I buy one! |
| 17:40.02 | ManxPower | JerJer[mobile]: I don't suppose you could accidently leave the firmware in an unsecured directory? |
| 17:40.06 | *** part/#asterisk jarrod (jarrod@dipole.informationwave.net) |
| 17:40.09 | zipp | JerJer[mobile], please |
| 17:40.10 | zipp | ha |
| 17:40.16 | beto75 | hello guys |
| 17:40.46 | mogorman | morning beto |
| 17:41.32 | *** join/#asterisk lyroy (~lyroy@modemcable007.224-203-24.mc.videotron.ca) |
| 17:41.44 | Mother_ | nice nice |
| 17:41.49 | lyroy | Does someone here who use rate-engine to rate calls? |
| 17:41.54 | Mother_ | I hear the Cisco Police knocking down some doors |
| 17:42.06 | Mother_ | and black helicopters overhead |
| 17:42.09 | Zeeek | he's making the unprotected directories now |
| 17:42.10 | nestAr | lol |
| 17:42.15 | nestAr | NOT THE BLACK HELICOPTERS! |
| 17:42.18 | mogorman | lol |
| 17:42.25 | Zeeek | the MIB |
| 17:43.09 | beto75 | Guys, I need to have autocreate peers for 2 diferent contexts is there a way (easy) to have different ports in order to differentiate |
| 17:43.22 | *** join/#asterisk Kokomo (~databoot@wbb30.fwa3.jaring.my) |
| 17:43.34 | rkjpl | Sorry, using it where, mogor? Was afk. |
| 17:43.55 | mogorman | I met some one at von who was using it as a chat line |
| 17:44.02 | Kokomo | Greeting all. I have a question here. I'm trying to do a server sizing for Asterisk deployment, anyone can assist me in how to do a server sizing ? |
| 17:44.04 | mogorman | and had a 1900 chat line as well |
| 17:44.25 | *** join/#asterisk brc__ (~brian@brc.base.supporter.pdpc) |
| 17:44.30 | mogorman | jerjer where you around when that guy showed it off? |
| 17:44.46 | zipp | who was laughing @ 66.250.68.194/996 |
| 17:44.46 | zipp | ? |
| 17:44.58 | rkjpl | Any hint as to the success? There would be a significant amount of DB work for that application. |
| 17:45.13 | Kokomo | anyone ? |
| 17:45.48 | pif | Zeeek : what codecs wengo accept? |
| 17:45.58 | *** join/#asterisk RaYmAn-Bx (user@213.237.12.147.adsl.vby.tiscali.dk) |
| 17:46.01 | *** join/#asterisk DJ-Pyro (~dj-pyro@12-202-92-37.client.insightBB.com) |
| 17:46.07 | mogorman | i think he put a good bit of work in, but once its setup he has probably made a good bit of money, i imagine 1900 numbers are real easy way to make cash esp if you have a well spelled word |
| 17:46.15 | mogorman | but all the tools are there |
| 17:46.33 | Kokomo | anyone can assist me in asterisk server sizing ? |
| 17:46.56 | mogorman | what do you need to know kokomo |
| 17:47.04 | mogorman | and dont spam channel, is rude... ^_^ |
| 17:47.14 | Kokomo | sorry...i don't intent to spam |
| 17:47.15 | |Vulture| | just ask thats the best way to find the answer... |
| 17:47.15 | rkjpl | 900 is less complex than 800 credit card stuff. I can see it handling the 900 no problem. Kewl. |
| 17:47.30 | Mother_ | Kokomo: http://www.voip-info.org/wiki-Asterisk+dimensioning |
| 17:47.31 | ariel_ | Kokomo, post what you want to do and we might be able to help but for sizing more info is needed. |
| 17:47.48 | mogorman | well credit card stuff is fairly simple implementation to check and what not |
| 17:47.59 | mogorman | it really depends on what you need done |
| 17:48.05 | rkjpl | <nod> |
| 17:48.06 | Kokomo | i'm planning to implement asterisk for quite a number of branches...114 in total.... |
| 17:48.15 | Kokomo | need to do some hardware calculation for that. |
| 17:48.27 | |Vulture| | I got a dual 2.8 xeon with 1gb ram from dell for $1400 that thing will handle everything I can throw at it :) |
| 17:48.30 | mogorman | but writing an agi script to get the credit card number out of asterisk is easy |
| 17:48.34 | mogorman | then just verify it |
| 17:49.01 | Zeeek | pif, look at the forum, it's all there, I do,n't remember |
| 17:49.17 | |Vulture| | Kokomo: how many phones connected to each server, thats really the question... |
| 17:49.18 | rkjpl | <nod> Then sending off via Xpacket to, for example, a CC processor and keeping a CDR. |
| 17:49.35 | Zeeek | pif but my guess would be ulaw |
| 17:50.02 | Kokomo | vulture: on aaverage, there will be 35 client per site...with a total of 114 site. |
| 17:50.31 | |Vulture| | Kokomo: you trying to have them connect to 1 server, or 1 server at each localtion interconnected? |
| 17:50.40 | ariel_ | Kokomo, the question is how your going to handle your sites. The main location and the remotes. |
| 17:50.43 | *** join/#asterisk Syncros (~sysop@noc.routermonkey.net) |
| 17:50.52 | mogorman | dont know what xpacket is, but you can get the credit card number in a string outside of asterisk and then do anything with it, you do get a cdr by defualt and can use the cdr mysql stuff for a better one |
| 17:50.58 | Kokomo | it will be a mesh connection |
| 17:51.08 | |Vulture| | not sure I follow |
| 17:51.09 | Juggie | woo... |
| 17:51.15 | Juggie | von canada is confirmed for me |
| 17:51.25 | Juggie | 5 days in toronto on the tax payers money |
| 17:51.31 | Juggie | fantastic :P |
| 17:51.34 | rkjpl | Packet of data with CC holder info for scrubbing and verification...tis all :) |
| 17:51.49 | *** join/#asterisk carbon60 (~adam@gw.techsupport.ca) |
| 17:51.54 | rkjpl | <nod> |
| 17:51.57 | Kokomo | vulture: each server will be connected to a regional server, and each regional server will connect to each other. |
| 17:52.14 | mogorman | so yeah you can do it easy peasy |
| 17:52.29 | ariel_ | Kokomo, it's something that should be easy to do if you lay out a good mapping. |
| 17:52.29 | mogorman | lots of people in community work on it for you for cash too as will digium |
| 17:52.34 | Zeeek | Juggie so you are in Canada? |
| 17:52.54 | |Vulture| | wont need anything that powerful |
| 17:53.00 | carbon60 | Once ztdummy is loaded under Linux 2.6, nothing special needs to be done for Asterisk to use it for timing, correct? |
| 17:53.02 | Kokomo | ariel: yes, true. so now I'll need to do some server sizing for each of the site server |
| 17:53.12 | mogorman | well you have to have asterisk built after zaptel |
| 17:53.20 | mogorman | so that you will have chan_zap module |
| 17:53.25 | mogorman | but you can check |
| 17:53.28 | Kokomo | searching for sites that can assist me in those calculation |
| 17:53.30 | mogorman | type zap show channels |
| 17:53.35 | mogorman | and you will see psuedo |
| 17:53.50 | carbon60 | mogorman: I do see pseudo. That's it? |
| 17:53.58 | mogorman | then you have dummy |
| 17:54.04 | carbon60 | Great, thanks. |
| 17:54.18 | mogorman | no problem |
| 17:54.35 | *** join/#asterisk Shido (~greg@d57-87-253.home.cgocable.net) |
| 17:55.17 | zipp | lo Shido |
| 17:55.20 | bkw_ | YO YO YO |
| 17:55.21 | bkw_ | HO HO HO |
| 17:55.32 | tzanger | MO MO MO |
| 17:55.35 | mogorman | THIS GUY |
| 17:55.35 | Zeeek | where's the bottle of rum? |
| 17:55.38 | mogorman | RIGHT HERE |
| 17:55.38 | Shido | hey |
| 17:55.42 | Shido | yager man |
| 17:55.43 | mogorman | bkw def has it... |
| 17:55.45 | Shido | drink the yager |
| 17:55.47 | bkw_ | haha |
| 17:55.51 | bkw_ | YOU KNOW IT |
| 17:55.59 | mogorman | I DIDNT KNOW ANY C |
| 17:56.06 | zipp | Shido, good stuff |
| 17:56.11 | tzanger | reminds me of the simpsons last night |
| 17:56.17 | bkw_ | haha |
| 17:56.19 | |Vulture| | oh no its bkw_! |
| 17:56.19 | Juggie | zeek, yes i am in canada. |
| 17:56.22 | tzanger | homer and lisa are building homes for hte homeless and homer paints something on the wall |
| 17:56.26 | bkw_ | |Vulture|, 1pm CST |
| 17:56.31 | Kokomo | thanks everyone...... |
| 17:56.34 | tzanger | lisa says "what are you doing" and homer said "well if Bart can be El Barto, I can do this" |
| 17:56.38 | zipp | http://www.andrewhodel.com/index.php?page=viewimage&id_src=5853 <- mi casa :) |
| 17:56.43 | tzanger | then the angle pans and you see "El Homo" on the wall |
| 17:56.44 | Zeeek | funny how there are so many canadians - I notice because they are looking for voIP providers |
| 17:56.45 | mogorman | heh |
| 17:56.46 | |Vulture| | bkw_: yea I am gunna go grab some food so I can be charged! |
| 17:56.54 | tzanger | as is Sivana |
| 17:57.05 | Zeeek | et tu bruté tzanger |
| 17:57.09 | *** join/#asterisk harryvv (~comming@S010600055d210201.vs.shawcable.net) |
| 17:57.12 | carbon60 | mogorman: I was wondering because I get a lot of "monmp3thread: Request to schedule in the past?!?!" errors. |
| 17:57.16 | bkw_ | el homo? |
| 17:57.20 | Zeeek | I've been to Canada |
| 17:57.22 | carbon60 | But the system isn't loaded at all. |
| 17:57.23 | tzanger | bkw_: yup |
| 17:58.14 | Goshen | Hey ariel_: where is that second page on voicepulse that details their asterisk plan? |
| 17:58.19 | bkw_ | mogorman, which one were you? |
| 17:58.35 | mogorman | matt O |
| 17:58.47 | bkw_ | picture |
| 17:58.49 | bkw_ | so I can place ya |
| 17:58.55 | Nugget | poose menis. |
| 17:58.56 | mogorman | we went to waffle house |
| 17:59.00 | Zeeek | Goshen "Voicepulse connect" |
| 17:59.03 | mogorman | with those guys |
| 17:59.05 | bkw_ | oh ya |
| 17:59.08 | bkw_ | with the loud guy |
| 17:59.09 | mogorman | who were so drunk |
| 17:59.13 | Zeeek | <PROTECTED> |
| 17:59.15 | ariel_ | Goshen, botton of there site go to connect |
| 17:59.20 | bkw_ | that guy was funny |
| 17:59.21 | *** join/#asterisk Stitch (~magliett@cpe-69-204-16-160.rochester.res.rr.com) |
| 17:59.21 | mogorman | and they learned to speak english better through the night |
| 17:59.30 | SkySky | excuse me, i wonder if there will be any log in asterisk ctl when one sip user press hold while they're in a call |
| 17:59.31 | bkw_ | he acted more gay than I and wanted to go where the girls were |
| 17:59.46 | JunK-Y | woa, that link's cool. http://elgoog.rb-hosting.de/index.cgi |
| 17:59.49 | SkySky | *cli* |
| 18:00.01 | mogorman | yup |
| 18:00.07 | mogorman | he was chrazy |
| 18:00.12 | bkw_ | thats fo sure |
| 18:00.42 | Goshen | ahh, ok http://connect.voicepulse.com/ |
| 18:00.43 | provolone | There has got to be a cheaper solution than buying an fxs card for ATA. Could I build a line voltage simulator and use a modem ? |
| 18:00.49 | Goshen | ~voicepulse |
| 18:01.03 | Zeeek | VoicePulse has DID in 27 states - I just counted |
| 18:01.10 | Goshen | jbot: voicepulse is http://connect.voicepulse.com/ |
| 18:01.11 | jbot | Goshen: okay |
| 18:01.14 | Nugget | area code 512 doesn't count. |
| 18:01.17 | Zeeek | I said that twice |
| 18:01.31 | *** join/#asterisk riddlebox (~blah@GC1.24.171.46.188.charter-stl.com) |
| 18:01.34 | Nugget | voicepulse in area code 512 is worthless. avoid. |
| 18:01.45 | Zeeek | what is that? Like Zone 69? |
| 18:01.58 | Zeeek | are you in Canada as well? |
| 18:02.02 | Katfood | mmm, full |
| 18:02.05 | Katfood | *pample* |
| 18:02.06 | zipp | Nugget, you are in austin? |
| 18:02.08 | Nugget | yes |
| 18:02.30 | riddlebox | is Digium the only manufacturer for voice boards that you can use with Asterisk? |
| 18:02.30 | *** join/#asterisk gpowers (~glenn@static-68-162-84-101.phil.east.verizon.net) |
| 18:02.38 | mogorman | carbon60 http://www.voip-info.org/wiki-Asterisk+Request+to+schedule+in+the+past |
| 18:02.40 | mogorman | check that? |
| 18:02.52 | carbon60 | mogorman: I did. Let me check it again to be sure. |
| 18:02.53 | mogorman | no but digium is the coolest... |
| 18:02.59 | mogorman | you unload oss? |
| 18:03.35 | carbon60 | Yeah, it was never loaded. noload in modules. |
| 18:04.26 | mogorman | which version of mpg123 installed? |
| 18:04.46 | Zeeek | I get that mesage on FreeBSD without chanoss loaded |
| 18:04.56 | carbon60 | mogorman: Version 0.59r (1999/Jun/15). |
| 18:06.57 | *** join/#asterisk jas_williams (~Jason@host81-155-66-178.range81-155.btcentralplus.com) |
| 18:07.08 | Nebukadneza | hehe - i just won against mpg123 :) |
| 18:08.09 | provolone | I want to hook up the following: analog telephone -> telco line voltage simulator -> modem -> voip |
| 18:09.41 | *** join/#asterisk MikeJ[Jayden] (~ircatjerr@65.170.43.34) |
| 18:09.42 | Zeeek | make sure you have rosin when you solder them |
| 18:09.50 | mogorman | hmmm |
| 18:09.53 | MikeJ[Jayden] | anyone using te110p? |
| 18:09.56 | phantam | yo |
| 18:09.57 | phantam | how do i do make |
| 18:10.03 | provolone | is this feasible? |
| 18:10.12 | mogorman | whats up mike? |
| 18:10.15 | phantam | but i need to do the equivilent of filter-flags -fstack-protector |
| 18:10.19 | phantam | how do i do that from command line |
| 18:10.26 | *** part/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 18:10.37 | Goshen | yea, my sipura 2100 from BSDmall just came...man that was fast! |
| 18:10.42 | *** join/#asterisk mikayle (~talamar@akro-static4-cs250.pool.dslohio.net) |
| 18:11.00 | MikeJ[Jayden] | uhhh... put a new te110p in and it's borking... I think I just have it configed wrong it zaptel.conf, was looking for a quick config example to check against |
| 18:11.06 | heison | who's that kid at in the back? |
| 18:11.27 | mogorman | what are you running? |
| 18:11.28 | Goshen | this sipura is tiny! it looks so big in the pictures :) |
| 18:11.29 | mikayle | anyone have problems with putting people on hold and not getting them back? I can't find anything about that in wiki |
| 18:11.29 | mogorman | pri? |
| 18:11.32 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 18:11.35 | MikeJ[Jayden] | y, |
| 18:12.02 | phantam | anyone? |
| 18:12.27 | MikeJ[Jayden] | I have span=1,1,0,esf,b8zs |
| 18:12.27 | MikeJ[Jayden] | <PROTECTED> |
| 18:12.27 | MikeJ[Jayden] | <PROTECTED> |
| 18:12.41 | mogorman | that is correct for pri and taking timing |
| 18:12.58 | boch | is there a manual of programming * applications or modules? |
| 18:13.00 | MikeJ[Jayden] | hmmm.. ok, won't enven do a zaptel start |
| 18:13.04 | carbon60 | mogorman: Me? SIP-to-SIP. AudioCodes on one end and Polycom IP500s on the other. |
| 18:13.37 | MikeJ[Jayden] | boch: there are those that would say the source code is the manual, others would say, that manual is still being written |
| 18:13.41 | jas_williams | MikeJ[Jayden]: Have you compiled and installed zaptel and libpri |
| 18:13.45 | MikeJ[Jayden] | y |
| 18:13.59 | MikeJ[Jayden] | no problem on that at all |
| 18:14.13 | jas_williams | MikeJ[Jayden]: what error are you getting on modprobe zaptel |
| 18:14.28 | MikeJ[Jayden] | pastebin, give me a sec to pull it up |
| 18:14.34 | boch | mikayle im looking at prepaid applications source codes, but i want to know hot to compile them |
| 18:14.36 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 18:14.47 | boch | MikeJ[Jayden] sory |
| 18:15.19 | *** join/#asterisk Nukemizer (~Nuke@65.103.231.133) |
| 18:15.24 | MikeJ[Jayden] | http://pastebin.ca/6794 |
| 18:16.15 | MikeJ[Jayden] | that is diff now... hmmm |
| 18:17.14 | bkw_ | ok we have #clueniversity COME ONE COME ALL!!! |
| 18:17.16 | MikeJ[Jayden] | getting kobject_register failed for t1xxp |
| 18:17.22 | MikeJ[Jayden] | hehe |
| 18:17.27 | bkw_ | MikeJ[Jayden], you too |
| 18:17.33 | bkw_ | go on over to the clueniversity |
| 18:17.57 | tuxinator_linux | bkw_: What will we find at clueniversity? |
| 18:18.27 | MikeJ[Jayden] | Tux: a clue |
| 18:19.30 | bkw_ | go see |
| 18:19.32 | *** join/#asterisk bjohnson_ (~bjohnson@jecinc.tor.istop.com) |
| 18:20.31 | bjohnson_ | I'm having trouble with my SPA 3k .. I think it may be my fxs port since call-ins seems to work but call-outs don't |
| 18:20.44 | *** join/#asterisk Red_6 (~alex@m174.net81-66-29.noos.fr) |
| 18:20.57 | bjohnson_ | is it possible to test the fxo outgoing with a call file? |
| 18:21.05 | jsolares | isnt an fxs port connected to a phone? |
| 18:21.10 | bjohnson_ | yes |
| 18:21.24 | phantam | bkw_: how do i do the equivilent of flag-filter -fstack-protector from an ebuild in a makefile |
| 18:21.36 | bjohnson_ | call-in = fxo - * - fxs |
| 18:21.42 | phantam | seems my damn old ass pwlib wont build because of -fstack-protector |
| 18:21.45 | phantam | acccording to bugs.gentoo |
| 18:22.16 | jsolares | call-out = fxs - * - fxo ? |
| 18:23.40 | *** part/#asterisk mikayle (~talamar@akro-static4-cs250.pool.dslohio.net) |
| 18:23.42 | bjohnson_ | jsolares: exactly |
| 18:23.44 | *** join/#asterisk riddlebox (~blah@GC1.24.171.46.188.charter-stl.com) |
| 18:23.59 | jsolares | one works but not the other... weird |
| 18:24.03 | bjohnson_ | jsolares: exactly |
| 18:24.06 | jsolares | kick it a bit, that helps sometimes XD |
| 18:24.27 | bjohnson_ | I've rebooted both * and the SPA 3k numerous times |
| 18:24.40 | riddlebox | damn I just got a couple of dialogic D/81A boards for free and I was hoping on using them with asterisk but it doesnt seem like it will |
| 18:25.07 | phantam | does anyone know |
| 18:25.11 | phantam | jsolares: do u know lol |
| 18:25.29 | bjohnson_ | here's something I just noticed .. the log files used to get the same level of logging as shown in the CLI .. now they get less |
| 18:25.58 | mogorman | that only happens if some code has printf |
| 18:26.04 | mogorman | instead of ast_verbose |
| 18:26.13 | mogorman | i don't think there any left in the cvs |
| 18:26.17 | jsolares | phantam: i didnt have a problem compiling h323 or oh323 |
| 18:26.28 | jsolares | i couldnt get either to work with netmeeting tho |
| 18:26.29 | mogorman | unless you turned off some form of messages in your messages |
| 18:26.32 | jsolares | so i gave up |
| 18:26.52 | *** join/#asterisk stan (~stan@82.152.179.36) |
| 18:27.36 | *** join/#asterisk HeadachesAbound (user79@wsip-68-99-73-32.tu.ok.cox.net) |
| 18:27.43 | bjohnson_ | mogorman: didn't touch logger.conf .. but did upgrade to recent stable cvs yesterday |
| 18:28.01 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 18:28.09 | *** join/#asterisk pcm (~pcm@user-69-73-0-22.knology.net) |
| 18:28.11 | HeadachesAbound | Can asterisk take advantage of a dual processor setup? |
| 18:28.18 | bjohnson_ | yes |
| 18:28.36 | bjohnson_ | HeadachesAbound: system examples on the wiki somewhere |
| 18:28.44 | bjohnson_ | ~docs |
| 18:28.45 | 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 |
| 18:29.16 | jas_williams | MikeJ[Jayden]: Can you post your /etc/zaptel.conf |
| 18:29.33 | bjohnson_ | so .. I've got a ssh connection to the home * box .. can I initiate an outgoing call and play something somehow to test? |
| 18:29.54 | *** join/#asterisk DevilFish (~me@staff211.qtm.net) |
| 18:30.24 | *** join/#asterisk girx (~roche@216.194.173.2) |
| 18:30.30 | jas_williams | bjohnson: does Dial work in the CLI |
| 18:30.40 | DevilFish | I'm trying to get the native asterisk format_mp3 working for music on hold as defined at http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf |
| 18:30.45 | DevilFish | anyone get this working? |
| 18:30.49 | DevilFish | not working for me |
| 18:30.56 | girx | I want to use speex codec there are any documentation about ? |
| 18:31.06 | *** join/#asterisk yashax (~yasha_x@69.15.218.218) |
| 18:31.06 | jas_williams | DevilFish: what version of code and what phones ? |
| 18:31.13 | DevilFish | asterisk 1.0.6 |
| 18:31.19 | DevilFish | phones are polycom IP500s |
| 18:31.56 | DevilFish | I followed the directions exactly as the website example was typed but I get nothing |
| 18:32.01 | jas_williams | There is a bug in 1.0.6 get latest verion of 1.0.6 from cvs this was fixed yesterday |
| 18:32.07 | Katfood | i need an ip to call |
| 18:32.16 | DevilFish | its like asterisk is still trying to use the mpg123 binary |
| 18:32.34 | Katty | someone must have an ip to call that's not behind a firewall |
| 18:32.52 | DevilFish | is there a place to get the latest cvs in tar.gz ? |
| 18:32.53 | yashax | Guys, what is the GSM file name in Asterisk@Home for greeting? |
| 18:32.53 | jas_williams | DevilFish: that was not the error, So maybe you have a config problem |
| 18:32.59 | DevilFish | ah |
| 18:33.06 | DevilFish | ok.. well hmm |
| 18:33.20 | jas_williams | DevilFish: do you see any error |
| 18:33.32 | DevilFish | lemme do a quick debug |
| 18:33.47 | Katty | :< |
| 18:33.59 | *** part/#asterisk girx (~roche@216.194.173.2) |
| 18:34.10 | bjohnson_ | use HEAD |
| 18:34.26 | yashax | bjohnson: what is the GSM file name in Asterisk@Home for greeting? |
| 18:34.37 | bjohnson_ | for greeting? |
| 18:34.48 | DevilFish | just getting this in the debug log |
| 18:34.49 | DevilFish | Unable to start music on hold (class 'default') on channel SIP/261-5ef8 |
| 18:34.53 | mikegrb | ha ha asterisk@home |
| 18:35.03 | yashax | yes |
| 18:35.11 | *** join/#asterisk calvinhp (~calvinhp@rrcs-24-123-25-236.central.biz.rr.com) |
| 18:35.25 | yashax | we recored the file through web gui and need to know what is the name |
| 18:35.38 | yashax | so we can replace it with professional recording... |
| 18:36.07 | DevilFish | I do have [classes] setup in musiconhold.conf with a default => /directory/of/mp3s |
| 18:36.40 | mikegrb | if you use asterisk@home you are on your own |
| 18:36.50 | bjohnson_ | yashax: check under /usr/src/asterisk-sounds/sounds |
| 18:36.55 | mikegrb | the override normal asterisk conventions with crap replacements |
| 18:37.15 | yashax | yeah, I am in this dir right now, but it has 100's of files... |
| 18:37.23 | *** join/#asterisk slak- (~alex@dsl093-236-218.hfd1.dsl.speakeasy.net) |
| 18:37.28 | bjohnson_ | jas_williams: how do you dial from cli? it won't take dial() |
| 18:37.39 | DevilFish | dial SIP/EXT |
| 18:37.48 | DevilFish | works fer me |
| 18:37.51 | bjohnson_ | Katty: most people test against fwd |
| 18:38.00 | beto75 | guys I am lost (in voip-info) can you please guide me how to autocreate extensions while (if I dial XXX menans Dial(sip/XXX) |
| 18:38.38 | bjohnson_ | yashax: use the find command then to look for mtime -1 |
| 18:39.03 | bjohnson_ | yashax: something like find / -mtime -1 -name '*.gsm' |
| 18:39.06 | jas_williams | bjohnson: it only works if your server has a sound system unfortunately why not make an iax connection from one server vto the other to originate a call or use firefly or other iax client |
| 18:39.10 | bjohnson_ | (assumming it is a gsm file) |
| 18:39.40 | slak- | hey is hardware echo cancel enabled in latest zaptel stable? |
| 18:39.53 | mogorman | for which card? |
| 18:39.57 | slak- | tdm |
| 18:40.01 | yashax | bjohnson: yeah, there are 100's of gms files in that directory |
| 18:40.01 | slak- | 04b |
| 18:40.06 | bjohnson_ | jas_williams: yeah I'll have to make an extension or something .. can't do a remote connection in very well since both ends behind nat and port forwards interfere |
| 18:40.26 | slak- | for digium cards |
| 18:40.46 | nestAr | beto75: sounds like you want a macro |
| 18:40.53 | *** join/#asterisk bacondoublechz (~bacon@cein176210.ceinetworks.com) |
| 18:40.56 | sivana | ~seen normast |
| 18:40.59 | jbot | normast <HydraIRC@Ottawa-HSE-ppp4119108.sympatico.ca> was last seen on IRC in channel #asterisk, 12h 53m 50s ago, saying: 'juggie: Thanks keep it in mind.'. |
| 18:41.10 | *** join/#asterisk montoya (montoya@200.195.87.20) |
| 18:41.17 | montoya | jumento |
| 18:41.58 | bacondoublechz | anyone have any suggestions for good sip/iax providers? |
| 18:42.18 | tzanger | bacondoublechz: are you proficient with asterisk? |
| 18:42.20 | bacondoublechz | for say.. 20 lines? |
| 18:42.22 | nestAr | beto75: look at this for an example.. |
| 18:42.24 | nestAr | beto75: http://pastebin.ca/6795 |
| 18:42.43 | bacondoublechz | tzanger, yea, I run it now with a T1 card for about 8 lines |
| 18:42.53 | tzanger | nufone |
| 18:43.09 | bacondoublechz | tzanger, but we're getting a 10 mbit circuit to our colo and want to get rid of the t1 |
| 18:43.12 | tzanger | they won't hold your hand but I run about 5k min/mo through them for the past year |
| 18:43.21 | beto75 | thansk Nestar |
| 18:43.30 | nestAr | np |
| 18:43.30 | yashax | bjohnson: Thanks buddy.... Found the file at /var/lib/asterisk/sounds/custom |
| 18:43.31 | beto75 | looking @ it |
| 18:43.35 | bacondoublechz | tzafrir, nice, I'll take a look |
| 18:43.38 | slak- | Connected to Asterisk CVS-v1-0-01/28/05-17:36:50 |
| 18:43.41 | slak- | do i need to upgrade? |
| 18:43.54 | nestAr | i <3 macros |
| 18:44.01 | tzanger | bacondoublechz: and the website's kind of goofy, but jerjer's in here as is shido6... they Just Work for me. |
| 18:44.14 | tzanger | (nufone just works, not jerjer and shido working for me, heh) |
| 18:44.18 | nestAr | lol |
| 18:44.22 | nestAr | just works(tm) |
| 18:44.37 | bacondoublechz | sweet |
| 18:44.44 | bacondoublechz | yea, website is a little starange |
| 18:45.04 | tzanger | they're telco guys, not web guys |
| 18:45.35 | bjohnson_ | hmm .. seems my problem is really my fxo |
| 18:45.59 | bacondoublechz | tzanger, so can I transfer my existing numbers to them? |
| 18:46.05 | bjohnson_ | on outgoing, logs say it's ringing but caller just gets dial tone |
| 18:46.31 | tzanger | bacondoublechz: kind of doubt it, they're mostly a termination company not an origination one, although they can originate SOME DIDs, it's just limited |
| 18:47.03 | bacondoublechz | hrmm |
| 18:47.03 | tzanger | bacondoublechz: best to get a PRI in your data centre for incoming only (they're cheaper nad if the data center is with telcos they can get you cheap PRIs since there's no loop cost) |
| 18:47.23 | tzanger | bacondoublechz: I have not found a stable origination company with DIDs in the areas I need |
| 18:47.38 | *** join/#asterisk facek (faceoff@devel.acdbddh.eu.org) |
| 18:47.42 | facek | hi |
| 18:48.09 | *** join/#asterisk TSCHAK (tschak@cuodan.net) |
| 18:48.33 | Katty | is it one yet |
| 18:48.35 | Katty | heh |
| 18:48.44 | Katty | tzanger: i sure hope i learn something today! |
| 18:48.51 | tzanger | Katty: whyzat |
| 18:48.59 | Katty | tzanger: see topic (= |
| 18:49.00 | bacondoublechz | tzanger, hrmm, I'll have to look into that |
| 18:49.07 | beto75 | Nestar? are you here? |
| 18:49.50 | *** part/#asterisk stan (~stan@82.152.179.36) |
| 18:50.51 | nestAr | yes |
| 18:51.18 | *** join/#asterisk Blackvel (~blackvel@dsl-213-023-033-012.arcor-ip.net) |
| 18:52.16 | Blackvel | ohohoh |
| 18:52.23 | Blackvel | DISA doesn't pickup the call anymore |
| 18:52.24 | Blackvel | :) |
| 18:52.42 | pcm | where ? |
| 18:53.40 | bjohnson_ | hmm PSTN Tone: None .. maybe I need to add some delay before going off hook |
| 18:53.51 | Blackvel | ah |
| 18:53.52 | Blackvel | now |
| 18:53.59 | Blackvel | hmmm |
| 18:54.43 | *** join/#asterisk citats (~james@duff.gnuinter.net) |
| 18:55.02 | Blackvel | I still can't dail anymore with bristuff (even with latest 1.0.6) from my pbx (e.g #9130 picks up DISA, #91309110000 would dail sipgate test number extension _9110000) |
| 18:55.19 | *** join/#asterisk bobx (~bobx@206.124.165.14) |
| 18:55.44 | Blackvel | is there any way to debug, if the extension 9110000 gets sent to asterisk internals? |
| 18:55.59 | bjohnson_ | well I'll be !! my SPA gain from PSTN lowered the volume enough the unit couldn't detect dial tone |
| 18:56.44 | bjohnson_ | Blackvel: watch the cli with set verbose 3 |
| 18:57.07 | Blackvel | bjohnson: no informations there :( |
| 18:57.11 | bjohnson_ | can someone dial my fwd number? |
| 18:57.15 | Blackvel | maybe I can enable bri debugging |
| 18:57.24 | Blackvel | whats a span? |
| 18:57.36 | jas_williams | bjohnson: shoot me the number |
| 18:57.37 | Blackvel | a channel? |
| 18:58.07 | bjohnson_ | jas_williams: 541735 |
| 18:58.36 | *** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no) |
| 18:59.00 | *** join/#asterisk carbon60 (~adam@gw.techsupport.ca) |
| 18:59.21 | carbon60 | Help! <ZOMBIE>'s are after me! |
| 18:59.30 | jas_williams | failled did you see anything :-?( |
| 18:59.51 | bjohnson_ | no |
| 19:00.06 | bjohnson_ | that's what's got me puzzled .. has been working before |
| 19:00.13 | carbon60 | I just setup an AudioCodes MP-108 on a system that used to be pure-IP. Now when an incoming call is transferred from one SIP device to another, I get a <ZOMBIE> and the call drops. Any ideas, of the top of your heads? |
| 19:00.21 | bjohnson_ | my fwd account shows you as a missed call |
| 19:00.29 | bjohnson_ | but nothing in the * cli |
| 19:00.35 | carbon60 | I'm also getting some voicemail messages that are cut off. |
| 19:00.39 | *** part/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 19:00.40 | jas_williams | I seem to be imediately disconnected |
| 19:00.44 | carbon60 | Maybe that is caused by the same problem? |
| 19:01.05 | slak- | http://www.userfriendly.org/cartoons/archives/99nov/uf001202.gif |
| 19:01.07 | *** join/#asterisk clark_ (~bclark@64.171.107.5) |
| 19:01.42 | jas_williams | carbon60: What version of code there was some RTP stream changes fixed yesterday |
| 19:01.45 | jontow | hmmm |
| 19:01.55 | carbon60 | jas_williams: This is 1.0.2. |
| 19:02.05 | *** part/#asterisk clark_ (~bclark@64.171.107.5) |
| 19:02.09 | jontow | if i can get SMDI working for MWI, i get to replace this crappy APEX VM server with * :) |
| 19:02.09 | jas_williams | Thats not it then :) |
| 19:02.15 | carbon60 | Only change is the AudioCodes. |
| 19:02.37 | RoyK | :%s/asterisk/aefirion/gi |
| 19:02.44 | bjohnson_ | jas_williams: iax2 show users gives iaxfwd Key: freeworlddialup 000000000000004 incoming-public No |
| 19:02.50 | bjohnson_ | what is A/C ? |
| 19:03.11 | Blackvel | Can I see the incoming extension in asterisk while debugging? I call from my analog/isdn pbx into asterisk, where DISA picks up the call and I am missing some extensions |
| 19:04.52 | bjohnson_ | with set verbose 3 you should see incoming call attempts in the cli |
| 19:05.18 | slak- | hey i built my * on 1/28/05, any reason to upgrade? |
| 19:05.28 | slak- | and is hardware echo cancel in stable yet? |
| 19:05.32 | slak- | for digium tdm cards |
| 19:05.49 | bjohnson_ | jas_williams: can I try your fwd number? |
| 19:05.55 | jas_williams | bjohnson: Your incoming-public context can you post this |
| 19:06.06 | bjohnson_ | ~pastebin |
| 19:06.07 | jbot | [pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 19:06.23 | jas_williams | bjohnson: Sure just hold on I'll change the destination |
| 19:06.50 | bjohnson_ | http://pastebin.ca/6796 |
| 19:07.12 | bjohnson_ | basically it just sets the group and then goes to [public] where my pstn calls come in and ring |
| 19:07.13 | Qwell | Anybody know much about the vonage softphone account? ie; what protocol is used? Can an IP phone be used? |
| 19:07.49 | RoyK | fsck |
| 19:07.49 | bjohnson_ | Qwell: I'm told you can connect to vonage if you pay the $5 extra /month to get a SIP connection |
| 19:07.50 | DevilFish | anyone got an example of asterisk working with shoutcast? |
| 19:08.05 | RoyK | with 1.0.3, I can't make queue exist by dtmf work |
| 19:08.06 | bjohnson_ | Qwell: likely more info on the wiki (I think I saw a sample config) |
| 19:08.17 | jas_williams | bjohnson: call me now |
| 19:08.18 | Qwell | great, I'll look. Any keyword suggestions? |
| 19:09.11 | bjohnson_ | jas_williams: must be my iax.conf .. won't do outgoing either |
| 19:09.28 | bjohnson_ | I was trying to split incoming and outgoing into separate user/peer sections |
| 19:09.29 | *** join/#asterisk rje (~rje@c-67-160-57-179.client.comcast.net) |
| 19:09.30 | modulus_ | how much do ip500's cost? |
| 19:09.33 | jas_williams | post away and i'll look blank those passwords |
| 19:09.38 | bjohnson_ | modulus_: $180 USD |
| 19:09.40 | slak- | anyone have asterisk-stat working? |
| 19:09.46 | modulus_ | that's too expensive |
| 19:09.59 | bjohnson_ | heh |
| 19:11.02 | bjohnson_ | http://pastebin.ca/6798 |
| 19:11.05 | slak- | i have asterisk running on my linux box, i want to replace this desk phone connected to it with a softphone running on win2k3 |
| 19:11.09 | slak- | any suggestions? |
| 19:11.14 | slak- | whats the best softphone for windows |
| 19:11.26 | bjohnson_ | slak-: idax and firefly and xten seem popular |
| 19:11.29 | Inv_arp | slak-: xtite i orefer |
| 19:11.35 | Inv_arp | err prefer |
| 19:11.35 | zipp | slak-, use an iax phone if you are connecting to asterisk |
| 19:11.36 | slak- | thanks |
| 19:11.36 | bjohnson_ | the first one should be diax |
| 19:11.44 | *** join/#asterisk Nukemizer (~Nuke@65.103.231.131) |
| 19:12.09 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
| 19:12.32 | slak- | eh wait ill need a microphone |
| 19:12.37 | slak- | heh |
| 19:13.05 | carbon60 | Should it be safe to upgrade from 1.0.2 debs to 1.0.5 debs? |
| 19:13.29 | modulus_ | # wc -l extensions.conf |
| 19:13.30 | modulus_ | <PROTECTED> |
| 19:13.38 | modulus_ | holy crip he's a crapple! |
| 19:14.10 | modulus_ | my extensions need to be in a database |
| 19:14.11 | modulus_ | thanks |
| 19:14.42 | slak- | hydrogen:/etc/asterisk# wc -l extensions.conf |
| 19:14.42 | slak- | 260 extensions.conf |
| 19:14.49 | slak- | how bout mine? |
| 19:15.02 | modulus_ | nah |
| 19:15.11 | modulus_ | you're just doing puny bits |
| 19:15.12 | slak- | i wanna play wiht mysql man but its too fucking confusing |
| 19:15.21 | slak- | i want asterisk-stat areski working |
| 19:15.25 | slak- | but it needs mysql |
| 19:15.26 | slak- | ;/ |
| 19:15.38 | slak- | and theres no instructions on how to create a table.setup mysql |
| 19:15.39 | bjohnson_ | # wc -l extensions.conf |
| 19:15.39 | bjohnson_ | 810 extensions.conf |
| 19:16.01 | bjohnson_ | me wants to play too |
| 19:16.11 | bjohnson_ | but I missed the slash |
| 19:16.19 | jas_williams | bjohnson: can't see anything obvious try turning iax2 debug and looking at what is going on |
| 19:16.20 | slak- | hey i built my * on 1/28 what version does that make it damnit |
| 19:16.51 | bjohnson_ | jas_williams: iax2 show peers lists fwd and all but one of my voip providers as unreachable .. maybe I'll upgrade and restart * |
| 19:16.52 | Juggie | depends on if its stable or head. |
| 19:16.56 | slak- | stable |
| 19:17.00 | Inv_arp | carbon60: why are u upgrading? if everything works u shouldnt |
| 19:17.03 | Juggie | 1.0.4 or around there likely. |
| 19:17.15 | slak- | Juggie, any reason to upgrade? |
| 19:17.19 | Juggie | yes |
| 19:17.22 | Juggie | see the change log |
| 19:17.30 | slak- | is hardware echo cancel enabled in new zaptel stable? |
| 19:17.33 | carbon60 | Inv_arp: I'm having some weird behaviour. |
| 19:17.36 | slak- | for digium tdm cards |
| 19:17.39 | Juggie | althought music on hold is broken 1.0.6 |
| 19:17.43 | *** join/#asterisk outsidefactor (barf@203-173-32-225.dyn.iinet.net.au) |
| 19:17.44 | Juggie | so if u go to stable, get cvs stable. |
| 19:17.45 | slak- | we dont have music on hold |
| 19:17.58 | carbon60 | I just setup an AudioCodes gateway, and users transferring incoming calls are getting them dropped. |
| 19:17.58 | slak- | i always get cvs stable |
| 19:18.03 | modulus_ | slak, asterisk documentation sucks |
| 19:18.05 | Juggie | well then do that. |
| 19:18.08 | slak- | modulus_, for sure |
| 19:18.10 | modulus_ | slak, i had to read source to get mysql working |
| 19:18.21 | modulus_ | with cdr/sip configs |
| 19:18.21 | Juggie | the wiki has everything i've ever needed. |
| 19:18.27 | modulus_ | the wiki sucks |
| 19:18.32 | Juggie | the wiki is fine. |
| 19:18.38 | Juggie | just search it with google |
| 19:18.40 | modulus_ | wiki is OK for looking up basic dialplan functions |
| 19:18.40 | Juggie | not the wiki search |
| 19:18.43 | *** join/#asterisk afe ([1HoUNVB5A@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 19:18.55 | eKo1 | The wiki rocks. |
| 19:19.04 | citats | show ip bgp sum |
| 19:19.08 | citats | ack |
| 19:19.11 | modulus_ | cisco |
| 19:19.18 | Juggie | asterisk & database is quite basterdized at t he moment |
| 19:19.22 | Juggie | there's code everywhere |
| 19:19.26 | Juggie | in head its cleaner |
| 19:19.29 | Juggie | but still not perfect. |
| 19:19.30 | modulus_ | yeah and too many ways to do it |
| 19:19.51 | eKo1 | Talk to anthm about that. |
| 19:20.03 | Juggie | also |
| 19:20.10 | Juggie | i dont agree with files having compile time directives. |
| 19:20.18 | Juggie | eg, voicemail and odbc support |
| 19:20.25 | Juggie | cdr and unique id support |
| 19:20.25 | carbon60 | Man, I can't figure this out at all. |
| 19:20.34 | Juggie | they should be config options. |
| 19:20.40 | modulus_ | juggie, i agree |
| 19:20.57 | *** join/#asterisk buddah (~hnic@208.179.86.5) |
| 19:21.05 | carbon60 | Incoming call via AudioCodes, Dial to SIP extension. Speak to them. Have them transfer the call to another SIP extension. Call drops. Channel ZOMBIE's. |
| 19:21.06 | modulus_ | /usr/bin/perl -pli.orig -e 's/CFLAGS\+\=-D_GNU_SOURCE/CFLAGS\+\=-D_GNU_SOURCE\nCFLAGS\+\=-DMYSQL_LOGUNIQUEID\n/' Makefile |
| 19:21.13 | modulus_ | /usr/bin/perl -pli.orig -e 's/#define DATE_FORMAT/#define MYSQL_LOGUNIQUEID\n#define DATE_FORMAT/' cdr_addon_mysql.c |
| 19:21.23 | modulus_ | those are my compile time directives |
| 19:21.29 | carbon60 | BRB. |
| 19:21.43 | sivana | anyone know why I would get congestion on a dial with the 44? http://pastebin.ca/6799 |
| 19:21.57 | buddah | anyone know how toll free numbers that charge your phone bill for stuff, i.e. voting for mtv vma awards, or american idol, would be handled if used over a VoIP line? would it charge to our bill with our carrier? |
| 19:22.08 | buddah | or even 900 numbers |
| 19:22.29 | *** join/#asterisk Mitch22 (~mitch@83.243.117.41.static.cablesurf.de) |
| 19:22.31 | Mitch22 | Hi |
| 19:22.57 | Mitch22 | Is it possible to configure asterisk in a way that the clients alone may chose the codecs (audio and video) used? |
| 19:23.49 | |Vulture| | yea... they choose and you just have a row of dissallow=all;allow= |
| 19:24.21 | Mitch22 | |Vulture|: on the CLI a "show codecs" doesn't show all codecs my SIP-client softphone knows |
| 19:24.30 | DevilFish | this is in my musiconhold.conf techno => mp3:/var/lib/asterisk/empty,http://64.236.34.196:80/stream/1003 any idea why I'm just getting silence? |
| 19:24.33 | jas_williams | sivana: that is a problem with asterisk matching order place the disallowed codes in their own context and include that context in international |
| 19:24.45 | *** join/#asterisk znoG (gs@200.115.216.109) |
| 19:24.48 | |Vulture| | Mitch22: well if * doesn't support it you can't have it on the client to * |
| 19:24.57 | |Vulture| | Mitch22: what codec is in question? |
| 19:25.03 | slak- | hm am i better of waiting til tonite to upgrade cvs-stable? |
| 19:25.07 | eKo1 | DevilFish: Use default instead of techno. |
| 19:25.10 | slak- | like after 5pm |
| 19:25.12 | DevilFish | same deal |
| 19:25.19 | DevilFish | same result |
| 19:25.24 | |Vulture| | slak-: why not just 1.0.6? |
| 19:25.26 | Mitch22 | |Vulture|: i think it's speex wideband that's causing problems |
| 19:25.35 | eKo1 | Well, maybe it can't access that stream. |
| 19:25.36 | slak- | Vulture i prefer cvs stable |
| 19:25.51 | |Vulture| | Mitch22: did you install speex? |
| 19:25.52 | *** join/#asterisk cbachman (~chatzilla@victory.ece.northwestern.edu) |
| 19:25.57 | slak- | Devil 1.0.6 has mousic on hold broken |
| 19:26.00 | slak- | use cvs stable |
| 19:26.04 | Mitch22 | |Vulture|: wait a second |
| 19:26.21 | |Vulture| | slak-: ah I use 1.0.5 in production |
| 19:26.23 | Juggie | does anyone have a listen only dialin # for the conference? |
| 19:26.25 | slePP | http://pastebin.ca/6800 -- has anyone seen this crash before? |
| 19:26.35 | DevilFish | what version is cvs stable at? |
| 19:26.40 | Blackvel | who has some tipps for 1.0.2 to 1.0.5/1.0.6? |
| 19:26.51 | Nugget | cvs stable is "at" cvs stable. |
| 19:26.54 | Blackvel | my extension getse not anymore into DISA |
| 19:26.56 | DevilFish | uh huh |
| 19:27.00 | DevilFish | thanks then |
| 19:27.03 | Blackvel | how can I track this down or debug it? |
| 19:27.12 | slak- | im a little histant about taking * down in the middle of the day at work |
| 19:27.13 | slak- | heh |
| 19:27.17 | slak- | i might make some people unhappy |
| 19:27.54 | slak- | make install copy config and start...20seconds if all goes right |
| 19:28.06 | slak- | oh unload/load zaptel ;/ |
| 19:28.15 | modulus_ | damnit Wait() doesn't process dtmf? |
| 19:28.22 | *** join/#asterisk cstdenis (voiptest@zz197059.cipherkey.net) |
| 19:28.31 | slePP | modulus_: WaitExten() |
| 19:28.43 | *** join/#asterisk wizhippo (~wizhippo@Quebec-HSE-ppp233869.qc.sympatico.ca) |
| 19:28.54 | modulus_ | slepp, yeah i just saw that in the wiki |
| 19:28.58 | *** join/#asterisk _tekati_ (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 19:28.58 | jas_williams | Blackvel: do you get the dial tone ? |
| 19:29.03 | modulus_ | wiki is only good for functions() |
| 19:29.40 | slePP | it used to continue on in the dialplan if no exten was entered, so it acted identical to Wait |
| 19:29.43 | slePP | but they buggered it |
| 19:29.43 | Blackvel | jas_williams: if I dail from my pbx telephone #9130 yes |
| 19:29.44 | Mitch22 | can I use "G.722 Wideband" audio codec with asterisk? if yes, how? |
| 19:29.56 | slePP | in the version i've got, it hangs up on them after the wait or something. |
| 19:29.58 | Blackvel | but before with bristuff and asterisk 1.0.2 I could dail #9130910000 |
| 19:30.03 | slePP | so now i have WaitExtenOrContinue :> |
| 19:30.09 | Katty | hmm |
| 19:30.16 | Katty | how do i get to that seminar thing? |
| 19:30.19 | Katty | or is it not in here? |
| 19:30.21 | jas_williams | Blackvel: what is the disa line from extensions.conf |
| 19:30.22 | Blackvel | then i got the dail tone and disa passed the extension 9110000 to asterisk |
| 19:30.27 | Blackvel | now nothing happens anymore |
| 19:30.28 | carbon60 | Sorry, got pulled away. |
| 19:30.30 | carbon60 | Incoming call via AudioCodes, Dial to SIP extension. Speak to them. Have them transfer the call to another SIP extension. Call drops. Channel ZOMBIE's. |
| 19:30.34 | *** join/#asterisk MasterYoda (~mnicholso@dhcp-155.digium.com) |
| 19:30.42 | bjohnson_ | can you connect to your * yet? |
| 19:30.43 | Blackvel | there seem to be some code changes anywhere in the answering mechanism |
| 19:30.44 | carbon60 | canreinvite is no everywhere. Same subnet. |
| 19:30.45 | eKo1 | Mitch22: You need to make a format_g722.so module for *. |
| 19:30.54 | Blackvel | if i hit #9130 |
| 19:30.57 | bjohnson_ | make a exten in extensions.conf for it |
| 19:30.57 | eKo1 | As well as a codec_g722.so. |
| 19:31.01 | Blackvel | disa picks the call up and i get a dail tone, right |
| 19:31.04 | carbon60 | A call coming in via a SIP provider, to the same dialplan, doesn't have this problem. |
| 19:31.14 | bjohnson_ | ie exten => 8,1,Dial(IAX2/guest@66.250.68.194/996) |
| 19:31.17 | Blackvel | i can then call 911000, disa waits some time and moves on |
| 19:31.25 | *** part/#asterisk MasterYoda (~mnicholso@dhcp-155.digium.com) |
| 19:31.25 | Blackvel | but i cant pass anymore in a simple call |
| 19:31.34 | Mitch22 | eKo1: why do the audio codecs I can use depend on asterisk and not on the clients only? |
| 19:31.34 | Katty | i guess i don't know enough to get there. heh |
| 19:31.35 | Blackvel | i really wonder why and how I can track this problem done |
| 19:31.35 | *** join/#asterisk zapa (~zapa@201.135.137.236) |
| 19:31.44 | Blackvel | is it a asterisk problem, is it a zaphfc problem? dunno :( |
| 19:31.48 | cstdenis | I have 3 asterisk servers and am having some sip codec problems. Call path is server1 (ulaw/alaw) -> server2 (ulaw/alaw/g729A) -> server3 (g729A). However, server2 tries to directly connect them together it seems and the call fails because it can't make the codec's compatable. How do I make it do the codec translation? |
| 19:32.01 | eKo1 | Mitch22: Because * is in the middle of the media path. |
| 19:32.15 | bjohnson_ | any idea why I can ping fwd at 35ms but can't get iax2 show peers to show a connection? |
| 19:32.33 | Mitch22 | eKo1: but canreinvite=yes leads to pure peer-to-peer communication, doesn't it? |
| 19:32.48 | *** part/#asterisk MikeJ[Jayden] (~ircatjerr@65.170.43.34) |
| 19:32.55 | eKo1 | Yeah, sure. |
| 19:33.13 | Mitch22 | eKo1: why does it depend on * then what codecs I can use? |
| 19:33.39 | *** join/#asterisk andrewh (~Andrew_He@andrew.1ec.aaisp.net.uk) |
| 19:33.50 | eKo1 | Well, say you have an IVR and the files are all gsm... |
| 19:34.00 | bjohnson_ | Katty: can you connect to your * yet? |
| 19:34.01 | eKo1 | and your phones only do ilbc... |
| 19:34.03 | Mitch22 | eKo1: ivr? sorry, i'm noob |
| 19:34.12 | *** part/#asterisk powD (~atob100@83.146.53.34) |
| 19:34.15 | *** join/#asterisk powD (~atob100@83.146.53.34) |
| 19:34.47 | eKo1 | Mitch22: The bottom line is, if you are going to do 'pure peer-to-peer communication' then you don't really need *. |
| 19:35.13 | Mitch22 | eKo1: mhhh. i see. |
| 19:35.25 | bjohnson_ | I think * will do pass through of any codec that it doesn't support .. but can't access the stream of course |
| 19:35.53 | eKo1 | bjohnson_: Yes, it will but I've had problems with that before * does the reinvite. |
| 19:36.16 | eKo1 | You'll always get an 'unknown codec' before the reinvite happens. |
| 19:36.41 | BrianR___ | Is there any easy way to identify in-state vs. nationwide 800 numbers? |
| 19:36.41 | bjohnson_ | wouldn't that depend if you answer or not? |
| 19:36.55 | bjohnson_ | ie .. I don't have an IVR .. incoming calls just rings phones |
| 19:37.16 | bjohnson_ | so I don't answer until someone picks up or it times out to vm |
| 19:37.25 | Mitch22 | eKo1: mhhh... My problem is that if I'm using eyebeam peer-to-peer (without asterisk), the audio and video quality is mich better than when using asterisk |
| 19:37.54 | eKo1 | Mitch22: eh, I don't see that as a problem... |
| 19:38.07 | eKo1 | bjohnson_: Well, I'm just telling you from experience. |
| 19:38.16 | Mitch22 | eKo1: may I send you a screenshot please? |
| 19:38.21 | bjohnson_ | eKo1: I'm just guessing |
| 19:38.31 | Mitch22 | eKo1: it's the Xten Eyebeam codec selection dialog |
| 19:38.48 | *** join/#asterisk riksta (~rick@host217-42-22-145.range217-42.btcentralplus.com) |
| 19:38.49 | eKo1 | How is showing me the screenshot going to help? |
| 19:38.53 | slak- | asterisk can do video |
| 19:38.53 | slak- | heh |
| 19:38.56 | slak- | i didnt kniow |
| 19:39.11 | *** join/#asterisk Pantanero (~Pantanero@a213-22-84-159.netcabo.pt) |
| 19:39.18 | Mitch22 | eKo1: maybe you can tell me what codec performs best and works with asterisk? |
| 19:39.19 | ariel_ | asterisk can do some limited video via sip. |
| 19:39.20 | *** join/#asterisk _tekati_ (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 19:39.29 | bjohnson_ | Mitch22: ulaw |
| 19:39.44 | ariel_ | Mitch22, ulaw is best but gsm is also very good in asterisk. |
| 19:40.03 | bjohnson_ | some people like ilbc too |
| 19:40.03 | slak- | wee only use asterisk for ptsn here |
| 19:40.04 | Mitch22 | bjohnson: mhhh. in peer to peer mode I can use e.g. a codec called "speex wideband" which imo is much better than ulaw |
| 19:40.05 | eKo1 | * can do video via h.263 i think. |
| 19:40.11 | bjohnson_ | most people use ulaw on the lan |
| 19:40.13 | slak- | wee only use asterisk for ptsn here, id like to see what else it can do |
| 19:40.18 | Mitch22 | video works great with eyebeam |
| 19:40.20 | slak- | maybe set up sip for long d |
| 19:40.23 | Blackvel | who can help me? DISA doesn't pickup anymore extensions after picking up the dailin :( |
| 19:40.42 | bjohnson_ | slak-: iax is a popular protocol for voip providers |
| 19:40.45 | ariel_ | argh ulaw is best any where but it requires a higher b/w |
| 19:40.55 | eKo1 | Say, is anybody participating in todays dev. conf.? |
| 19:40.57 | ariel_ | Blackvel your using stable? |
| 19:41.04 | *** part/#asterisk yashax (~yasha_x@69.15.218.218) |
| 19:41.09 | bjohnson_ | eKo1: participating or eavesdropping? |
| 19:41.12 | *** join/#asterisk lev1athan (~jan@dD5769252.access.telenet.be) |
| 19:41.15 | Mitch22 | ariel_: ok thx |
| 19:41.16 | Blackvel | bristuff 0.2.0RC7f which is 1.0.6 based |
| 19:41.18 | eKo1 | Doesn't matter. |
| 19:41.24 | slak- | bjohnson so id purchase a sip/iax account and route my long distance thru them? |
| 19:41.29 | Blackvel | its not working anymore since bristuff which uses 1.0.5 :( |
| 19:41.37 | gpowers | eKo1: yes |
| 19:41.46 | Blackvel | 1.0.2 worked fine |
| 19:41.56 | *** join/#asterisk GiabboO (~GiabboOo@host101-246.pool8173.interbusiness.it) |
| 19:41.57 | Blackvel | I just need a way to debug it |
| 19:42.00 | GiabboO | hi all! |
| 19:42.01 | gpowers | disc on docs, was manager interface |
| 19:42.07 | Blackvel | what should I debug to check the passed extension? |
| 19:42.14 | Blackvel | the error can be anywhere |
| 19:42.16 | Blackvel | libpri |
| 19:42.18 | Blackvel | bristuff |
| 19:42.21 | Blackvel | (zaphfc) |
| 19:42.21 | ariel_ | hummm I am using it here on 1.0.5 but I use zap channels |
| 19:42.23 | Blackvel | asterisk |
| 19:42.24 | Blackvel | zap |
| 19:42.50 | Blackvel | you know what I mean, don't you ariel? |
| 19:42.57 | Blackvel | instead of waiting for the DISA dailtone |
| 19:43.09 | Blackvel | you pass DISA the complete extension BEFORE dailing into the system |
| 19:43.17 | GiabboO | 1st of all i want to send my congrats to asterisk creators and community, i installed it yesterday and im doin praticize to build a good working asterisk pbx |
| 19:43.22 | lev1athan | when i use the gentoo ebuild or i use te cvs repositry i always get en error while starting asterisk "[res_features.so]Mar 3 20:41:41 WARNING[12706]: loader.c:301 __load_resource: /usr/lib/asterisk/modules/res_features.so: undefined symbol: adsi_available" what can be the problem for that? |
| 19:43.43 | slak- | dont use gentoo |
| 19:43.45 | slak- | ;) |
| 19:43.46 | mikegrb | lev1athan: ddon't use gentoo |
| 19:43.47 | lev1athan | :p |
| 19:43.51 | GiabboO | i use asterisk in gentoo |
| 19:43.58 | GiabboO | but i told im very new to this |
| 19:44.08 | JerJer[mobile] | lev1athan: load res_adsi.so b4 res_features.so |
| 19:44.39 | GiabboO | can anyone help me to set my asterisk to answer calls from /dev/ttyI0 ? |
| 19:44.59 | *** part/#asterisk Nebukadneza (~daddel9@i3ED6E583.versanet.de) |
| 19:45.13 | JerJer[mobile] | mikegrb: fire away, i've got unlimited data baby |
| 19:45.25 | mikegrb | heh me too |
| 19:45.32 | mikegrb | latency sucks tthough :/ |
| 19:45.40 | JerJer[mobile] | and i've been keeping AT&T to that usage of the word unlimited |
| 19:45.52 | mikegrb | was killer ssh'ing into a few servers to fix problems with 500 ms ping times |
| 19:46.09 | mikegrb | silly airport wifi wanting $10 for an hour |
| 19:46.10 | *** join/#asterisk folsson_ (~filip@h87n2fls31o985.telia.com) |
| 19:46.11 | JerJer[mobile] | they tried to charge me some crazy fee cuz i hooked up my laptop |
| 19:46.11 | mikegrb | bah! |
| 19:46.24 | JerJer[mobile] | per kilobyte |
| 19:46.24 | mikegrb | ahh... I only use my laptop |
| 19:46.27 | bjohnson_ | hmm .. I can connect to some servers but not others and my remote server at home can connect to the ones I can't here. Any ideas? |
| 19:46.54 | JerJer[mobile] | but when I signed up for the unlimited data I specifically asked the lady if there was any restrictions on this and she said no |
| 19:46.59 | JerJer[mobile] | and I kept her contact info :) |
| 19:47.21 | *** join/#asterisk |Vulture| (~Vulture@109.238.204.68.cfl.res.rr.com) |
| 19:47.41 | JerJer[mobile] | latency does suck if you arent in an EDGE area... its pretty ok if you are |
| 19:47.47 | lev1athan | JerJer[mobile]: how can i load res_adsi.so b4 res_features.so? |
| 19:47.55 | *** join/#asterisk zapa (~zapa@201.135.137.236) |
| 19:47.56 | JerJer[mobile] | modules.conf |
| 19:47.59 | lev1athan | ah |
| 19:48.01 | lev1athan | thx |
| 19:48.13 | Inv_arp | man 3.3 kbytes/sec each way into the conf ilbc.... nice |
| 19:48.15 | zipp | JerJer[mobile], I have used att EDGE service |
| 19:48.27 | JerJer[mobile] | so who here can jive Chinese? |
| 19:48.28 | zipp | seems to work fine w/ IAX2{gsm} |
| 19:49.00 | GiabboO | JerJer[mobile], can u help me to set my asterisk to be able to answer calls incoming on modem channel ? |
| 19:50.22 | Blackvel | hm |
| 19:50.31 | Blackvel | how to debug incoming extensions on zap? |
| 19:50.41 | *** part/#asterisk Mitch22 (~mitch@83.243.117.41.static.cablesurf.de) |
| 19:50.54 | Blackvel | how can I make sure zaphfc passes them to DISA which passes them to asterisk? |
| 19:51.33 | *** join/#asterisk adjacent_ (~scott@office.bftwave.com) |
| 19:54.36 | jas_williams | GiabboO: Do not use the modem channel get a proper pstn interface |
| 19:55.20 | GiabboO | jas_williams: thx for answer, i only have this stupid ISDN TA Winbond 6692 |
| 19:55.36 | GiabboO | jas_williams: all looks working good, except incoming calls |
| 19:56.04 | *** join/#asterisk RevK ([U2FsdGVkX@excalibur.bracknell.aaisp.net.uk) |
| 19:56.28 | jas_williams | GiabboO: Make sure you have your MSN's correct. |
| 19:57.36 | GiabboO | jas_williams: i setted up MSN in modem.conf, using my 2 numbers, separated by a "," |
| 19:57.45 | *** part/#asterisk gtigene (~gnadenx@208.239.206.195) |
| 19:57.48 | *** join/#asterisk afe ([QJJGILocN@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 19:57.52 | Blackvel | someone plz teach me debugging |
| 19:57.55 | Blackvel | thx :) |
| 19:58.12 | eKo1 | eh, no. |
| 19:58.30 | GiabboO | jas_williams: when i do Dial(Modem/ttyI0:anynumber) calls get established using the first MSN i entered in modem.conf |
| 19:58.41 | jas_williams | Blackvel: what do you get in the cli when the call arrives :) |
| 19:59.19 | GiabboO | jas_williams: so i think my modem.conf is properly setted up |
| 19:59.55 | GiabboO | jas_williams: there is a way to check up if /dev/ttyI0 ring ? |
| 19:59.55 | jas_williams | Blackvel: Why do you want to use disa, whynot route the calls directly |
| 20:00.15 | jas_williams | GiabboO: check syslog messages dmesg for incomming messages |
| 20:00.26 | GiabboO | jas_williams: ok i take a look wait |
| 20:00.59 | GiabboO | jas_williams: ok i found, ty man |
| 20:01.13 | gpowers | Anyone have a URL for the "gatekeeper" patch manager? |
| 20:01.17 | andrewh | Hello! I'm having real problems with getting zaptel working on a new machine (been trying all day!) . it's FC2 2.6.10-1.770_FC2smp, on a dual XEON, running CVS 1.0 with an TE110P. When starting zaptel I get Running ztcfg: ZT_SPANCONFIG failed on span 1: No such device or address (6) |
| 20:02.58 | JerJer[mobile] | ztcfg - vv |
| 20:03.14 | jas_williams | andrewh: post your /etc/zaptel.conf and /etc/asterisk/zapata.conf to pastebin |
| 20:03.18 | *** part/#asterisk rje (~rje@c-67-160-57-179.client.comcast.net) |
| 20:03.19 | JerJer[mobile] | and lsmod |
| 20:03.27 | JerJer[mobile] | and cat /proc/interrupts |
| 20:04.05 | andrewh | ok hang on... |
| 20:05.03 | phantam | JerJer[mobile]: dude pwlib wont compile |
| 20:05.38 | phantam | i read something about 3.4.3 gcc when compiling |
| 20:05.51 | phantam | about disabling -fstack-protection or something but cant figure out how to do it |
| 20:06.01 | GiabboO | jas_williams: can I pvt ? |
| 20:06.09 | jas_williams | sure |
| 20:06.09 | slak- | can i safely tar up my config dir from 1.0.4 and use it for 1.0.6 |
| 20:06.10 | slak- | ? |
| 20:06.38 | andrewh | jas_williams, I've pasted the stuff to http://pastebin.ca/6801 |
| 20:06.39 | jas_williams | slak-: mostly may be some changes in sip.conf |
| 20:08.21 | slak- | jas_williams, is hardware echoo cancel enabled in 1.0.6? |
| 20:08.24 | phantam | JerJer[mobile]: ? |
| 20:08.28 | slak- | foo digium tdm cards |
| 20:08.30 | slak- | for |
| 20:10.48 | jas_williams | andrewh: can you see the card in lspci |
| 20:11.13 | RevK | Yes he can. |
| 20:11.36 | jas_williams | k |
| 20:11.37 | phantam | JerJer[mobile]: i know ya hate h323 now but please help :( |
| 20:11.38 | RevK | He has also changed the card |
| 20:11.50 | RevK | [just in case]... I'm watching on a web cam. |
| 20:12.01 | |Vulture| | h323 is the devil :P |
| 20:12.23 | eKo1 | ditto |
| 20:12.25 | jas_williams | andrewh: The card does not seem to have an interrupt which is worrying |
| 20:13.20 | andrewh | jas_williams, lspci is added to the pastebin, http://pastebin.ca/6804 |
| 20:13.55 | RevK | Ah, perhaps a BIOS funny??? |
| 20:14.01 | andrewh | jas_williams, is the Jens Schoenfeld Intel 537 the card? |
| 20:14.23 | andrewh | jas_williams, RevK is working with me on this too |
| 20:16.12 | andrewh | jas_williams, I'll go check some bois settings, but it's pretty much default with PnP off |
| 20:17.15 | *** join/#asterisk RoyK (~roy@80.239.107.69) |
| 20:21.23 | jas_williams | andrewh: rather than using the startup script why not modprobe each component separately |
| 20:21.27 | phantam | JerJer[mobile]: u around? |
| 20:21.31 | phantam | why do i always miss him |
| 20:21.31 | phantam | argg |
| 20:21.43 | TSCHAK | blahhh |
| 20:21.43 | GiabboO | cya all community, thx to jas_williams for support |
| 20:21.50 | *** part/#asterisk GiabboO (~GiabboOo@host101-246.pool8173.interbusiness.it) |
| 20:22.13 | TSCHAK | now it seems that Callmanager is interefering with GNUGK.... |
| 20:22.33 | andrewh | jas_williams, OK, I'll try that... |
| 20:23.06 | jas_williams | andrewh: modprobe zaptel |
| 20:23.35 | jas_williams | andrewh: modprobe wcte11xp |
| 20:24.19 | andrewh | jas_williams, zaptel was OK, but wcte11xp gives: FATAL: Error inserting wcte11xp (/lib/modules/2.6.10-1.770_FC2smp/misc/wcte11xp.ko): No such device |
| 20:24.21 | andrewh | FATAL: Error running install command for wcte11xp |
| 20:24.30 | slak- | hey did zaptel.conf change from 1.0.2 to 1.0.6 |
| 20:24.31 | slak- | ? |
| 20:24.36 | *** join/#asterisk file (~file@mctnnbsah25-142166093180.nb.aliant.net) |
| 20:24.38 | slak- | anything about hardware echo cancel added? |
| 20:24.47 | *** join/#asterisk Mearth (~Mearth@65.38.211.195) |
| 20:25.08 | jas_williams | andrewh: Hmm without that loading your going nowhere |
| 20:25.24 | BrianR___ | Does the person who runs iax.cc hang out here? |
| 20:25.48 | *** join/#asterisk bprice20 (~brandon@Dynamic-216.120.224.151.hrnoc.net) |
| 20:27.29 | andrewh | jas_williams, modprobe wcte11xp worked ok now (after stopping zaptel, as modules were already loaded) |
| 20:28.13 | citats | so whats happening everyone? been a while since i've been around |
| 20:28.57 | *** join/#asterisk mgomes_mpg (1000@Froes.microlink.com.br) |
| 20:29.15 | *** join/#asterisk mgomes_mpg (1000@Froes.microlink.com.br) |
| 20:29.16 | jas_williams | andrewh: oh now ztcfg -vvv |
| 20:30.14 | Mearth | i have a problem with the sound... on the clients machines... there is a weard sound comming back when i dial to the latenci test... is there any win32 application i can download to use with asterisk for it to adjust the sound automatically? |
| 20:30.53 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
| 20:30.54 | andrewh | jas_williams, yeah, that loaded |
| 20:31.11 | Mearth | the asterisk is on linux and the workstations are on windows |
| 20:31.32 | Mearth | i mean the server in on linux and the clients on windows |
| 20:31.56 | jas_williams | andrewh: then modify your rc.local to load the modules like that zaptel wcte11xp ztcfg -vvv and then safe_asterisk and all should be well |
| 20:32.23 | jas_williams | do not use the zaptel start script |
| 20:32.51 | Mearth | can someone help me with the sound here? |
| 20:32.51 | andrewh | jas_williams, ok.... |
| 20:34.42 | Mearth | i have a T1 running with 14 clients on a call center.. the clients are getting cutoff in the call. the call comes in and goes out comes in and goes out... and so on.. can some one help on this? |
| 20:34.47 | mgomes_mpg | Hello all, is Chris or anybody from Digium staff here ( issue : 16035 ) ? |
| 20:35.13 | *** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl) |
| 20:35.59 | Mearth | is there a way to adjust the sound atomatically on the clients... a program for it? wich i can install to adjust the sound of the microphone and the headset...? |
| 20:37.25 | mgomes_mpg | Hello I am having some problems to take the Telco E1/PRI clock, can anyone help me ? |
| 20:40.20 | jas_williams | andrewh: is your pri now working ? |
| 20:41.22 | andrewh | jas_williams, it did, then we rebooted, and now re putting it back in! I think we're getting there though! |
| 20:41.51 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 20:42.03 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 20:42.25 | *** join/#asterisk YoYo (~YoYo@pool-151-199-125-240.roa.east.verizon.net) |
| 20:43.51 | *** join/#asterisk Lashkar (~johann@12.44.215.11) |
| 20:44.02 | neopher | in the sip.cong is there a way to do a single line allow with multiple codecs eg. something like allow= g729 ulaw alaw ? |
| 20:44.23 | jas_williams | neopher: allow=all |
| 20:44.32 | eKo1 | no, you have multiple allow= with the different codecs |
| 20:44.49 | neopher | hmmm, thats what i thought |
| 20:46.05 | eKo1 | Man, I didn't know a dev. conf. on the phone could be so boring... |
| 20:46.09 | neopher | ok, i likes all, tnx, though that would cause a 488, but i guess not |
| 20:49.18 | bprice20 | I'mmmm baaack |
| 20:51.37 | zipp | so what makes a quality phone, quality? |
| 20:51.55 | zipp | other then sound quality, ease of configuration, and good looks? |
| 20:52.27 | Essobi | Hey guys.. If I plug a PBX into an asterisk box VIA a zaptel card.. |
| 20:52.30 | eKo1 | It's difficult to say. If your customers don't complain about the phone, then it's all good. |
| 20:52.54 | Essobi | Will the PBX (if setup) get handed CallerID name set in the dialplan in the Q931 messages to the pbx? |
| 20:53.24 | Beirdo | zipp: having it not feel like a flimsy POS would be a good thing too |
| 20:53.47 | RevK | Looks like we needed to cut the modules down to say wcfxs and wcte11xp only and it is fine... |
| 20:53.59 | zipp | Beirdo, like the budgetone? |
| 20:54.06 | RevK | Note that RMODULES has wcte11xp at the end and it shoudl surely be at the start... |
| 20:54.14 | Essobi | anyone? |
| 20:54.16 | Beirdo | I haven't handled one, but probably |
| 20:54.22 | Essobi | Q931, and caller ID names? |
| 20:54.39 | neopher | i have a budgetone 101, its a preatty nice phone |
| 20:55.04 | zipp | neopher, have you used any other phones |
| 20:55.06 | neopher | has a nice webpage ui |
| 20:55.14 | zipp | you don't have loss of registration problems w/ the budgetone? |
| 20:55.32 | neopher | yes, cisco, sipara |
| 20:55.54 | jas_williams | RevK: I would just load the modules you need and comment out the others, why waste memory on modules you do not need |
| 20:55.55 | neopher | nope, budgetone works great, no probs at all |
| 20:56.12 | RevK | More to the point, it simply would not work with the full list! |
| 20:56.17 | zipp | neopher, mine loses registration, and won't ring unless I make a call so * knows it is there |
| 20:56.39 | neopher | hmm, what firmware you have? |
| 20:56.40 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 20:57.05 | zipp | 1.0.5.22 |
| 20:57.20 | neopher | Software Version: Program--1.0.5.16 Bootloader--1.0.0.21 HTML--1.0.0.41 VOC--1.0.0.7 |
| 20:57.35 | neopher | that may be why |
| 20:57.53 | neopher | the last stable release was 1.0.5.16 |
| 20:58.06 | zipp | interesting |
| 20:58.06 | jas_williams | RevK: I just don't use the init script I do it manually I know where I am then |
| 20:58.24 | zipp | neopher, everything else is the same |
| 20:58.40 | neopher | point your tftp to 168.75.215.189 |
| 20:59.14 | neopher | it will get the lastest stable image on there |
| 21:00.07 | Lashkar | is there a wiki article on connecting multiple asterisk boxes together? |
| 21:00.30 | mgomes_mpg | Hello I am having some problems to take the Telco E1/PRI clock, can anyone help me ? |
| 21:02.17 | ManxPower | from LiveVoIP: |
| 21:02.21 | ManxPower | Sorry no plans to do so now. |
| 21:02.21 | ManxPower | > It sure would be nice if you supported G726. (Speex would be nice too, but that's pretty CPU intensive) |
| 21:02.32 | jas_williams | mgomes_mpg: post your zaptel.conf to pastebin.ca |
| 21:02.35 | ManxPower | They are not even planning on supporting G726. |
| 21:02.48 | jas_williams | mgomes_mpg: also cat /proc/interrupts |
| 21:02.57 | *** join/#asterisk Mavvie (edwin@edwin.adsl.barnet.com.au) |
| 21:03.19 | mgomes_mpg | jas _> I will open a pvt msg with you |
| 21:03.28 | buddah | Mar 3 13:05:29 NOTICE[32041]: app_dial.c:927 dial_exec_full: Unable to create channel of type 'SIP' (cause 3) |
| 21:03.33 | buddah | any idea about that? |
| 21:03.38 | *** join/#asterisk Legend (~Legend@office.bgcfreedom.com) |
| 21:03.47 | buddah | should i stop/restart asterisk, would that clear it up? |
| 21:04.09 | *** join/#asterisk zagaya972 (~d2s-compa@APointe-a-Pitre-102-1-13-214.w81-248.abo.wanadoo.fr) |
| 21:04.21 | *** join/#asterisk darby_t (~tom@dof148.neoplus.adsl.tpnet.pl) |
| 21:05.53 | Essobi | Anyone? |
| 21:05.58 | bprice20 | I have a problem with my dialplan, both incoming as well as outgoing calls match _1NXXNXXXXXX, so asterisk always gois with the action that comes firs (as it should) |
| 21:06.03 | buddah | hmm, that didnt fix it either |
| 21:06.30 | ManxPower | I can't say I'm impressed with their support. The tech says "G726 is too CPU intensive" |
| 21:06.41 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
| 21:06.44 | Essobi | mehe |
| 21:06.49 | bprice20 | my problem is how do i make asterisk handle incoming and outgoings differently without making users dial a number like "9" to dial out |
| 21:07.14 | Essobi | ManxPower Any idea is 4ESS supports sending a caller ID name in Q931, and more importantly does * support it? |
| 21:07.16 | Nugget | edit your dialplan so users don't need to dial 9 to dial out. |
| 21:07.19 | Essobi | s/is/if |
| 21:07.19 | Lashkar | dialing 9 to get out is the way pretty much any pbx handles it not just asterisk |
| 21:07.24 | bprice20 | I want calls to be switched to a different context based on weather they are an incoming our outgoing call |
| 21:07.50 | Essobi | bprice20 so land them appropriately |
| 21:08.00 | bprice20 | If we want to be a real live voip carrier I can't ask users to dial 9 before they dial whoever |
| 21:08.10 | Essobi | mahaha |
| 21:08.13 | bprice20 | Essobi "land them?" |
| 21:08.21 | Nugget | I want calls to be switched to a different context based on weather. if it's sunny out, I don't want to take any calls. |
| 21:08.21 | Essobi | You're a "voip carrier"? :) |
| 21:08.29 | Essobi | hahahaaa |
| 21:08.33 | xkev | on the polycoms, what is the tftp option to make them actually work right for the server address |
| 21:08.42 | ManxPower | Essobi: I don't know, but I would assume it does. |
| 21:08.45 | bprice20 | Essobi did you not read "want to be" |
| 21:09.02 | xkev | ..and can one set the password somehow in a config after the initial update? |
| 21:09.03 | bprice20 | you guys aren't helping |
| 21:09.06 | Luhiwu | Nugget: load app_weather.so |
| 21:09.11 | Nugget | bprice20: the answer to every single question you've just asked is the same. "learn how to edit extensions.conf" |
| 21:09.16 | ManxPower | bprice20: A carrier doesn't nee "dial 9" since a carrier would require 10 or 11 digit dialing for ALL calls. |
| 21:09.18 | Essobi | ManxPower Umm. cool.. I got a pbx that I want to feed caller name to over a PRI I'm going to hand them from an * box. |
| 21:09.26 | bprice20 | I do |
| 21:09.29 | ManxPower | The need for dial 9 is if you have local extensions. |
| 21:09.37 | bprice20 | ManxPower I do |
| 21:09.41 | Luhiwu | Nugget: or just come to Argentina 10 years ago... on not-so-sunny days the phones didn't work. |
| 21:09.52 | bprice20 | Manxpower take a look at my extensions.conf and you will see what I mean |
| 21:10.04 | ManxPower | bprice20: bprice20 Do you require 10 digit or 11 digit dialing? |
| 21:10.23 | Essobi | ManxPower Some carriers ofer local 7's, the problem is you have to assume the local 7 is the same area code as the originating ANI, which doesn't always work. |
| 21:10.31 | bprice20 | 11 |
| 21:10.43 | Luhiwu | is there any standar about how many digits should be for each country? |
| 21:10.50 | ManxPower | bprice20: then you just have to make sure no exten => begins with "1" for regular extensions/IVR and you should be fine. |
| 21:10.57 | ManxPower | Luhiwu: no. |
| 21:11.18 | Luhiwu | ManxPower: thanks |
| 21:11.40 | Essobi | Another way to work it is a NPA/NPXX to local calling area SQL tables and whip up a program to make assumptions about 7 digit dialing.. which won't always work either if there is overlapping dids as in my local 7 calling area. |
| 21:11.53 | *** join/#asterisk SkySky (~Jack@host6614613596.biz.tor.fcibroadband.com) |
| 21:11.56 | bprice20 | Manxpower http://pastebin.ca/6806 |
| 21:12.01 | Essobi | Nope.. No started lengthy at all |
| 21:12.05 | bprice20 | take a look and you will see what I mean |
| 21:12.10 | Essobi | standard |
| 21:13.00 | bprice20 | whats wrong with just pulling it from the callerid |
| 21:13.09 | *** join/#asterisk Darwin35 (~Darin@c-24-3-241-22.client.comcast.net) |
| 21:13.11 | bprice20 | if you can trust the callerid, or if your the carrier |
| 21:13.14 | bprice20 | exten => _NXXXXXX,1,Dial(SIP/${CALLERIDNUM:0:4}${EXTEN}${TRUNK},60,rtT) |
| 21:13.14 | ManxPower | bprice20: Yup! You have overlapping, non unique extension patterns. |
| 21:13.32 | bprice20 | ManxPower that i know |
| 21:13.49 | bprice20 | overlapping is new but non unique i knew |
| 21:14.44 | ManxPower | How does Asterisk know when you are finished dialing when you have _NXXXXXX and 5XXX? |
| 21:14.47 | ManxPower | It doesn't. |
| 21:14.48 | lilo | hi all |
| 21:14.54 | lilo | is this channel intentionally +s? |
| 21:14.55 | bprice20 | wouldn't the code I just pasted allow you to pull the area code from the caller id before dialing out |
| 21:15.02 | ManxPower | So anytime you dial 5XXX you have to wait for DigitTimeout before the call starts. |
| 21:15.11 | bprice20 | therefore allowing 7 digit dialing on any area code |
| 21:15.15 | Beirdo | lilo: I'd bet no |
| 21:15.27 | ManxPower | bprice20: I dunno. Try it. |
| 21:15.47 | ManxPower | But honestly, I can't see WHY you are using rtT. |
| 21:15.49 | bprice20 | it works. People in here were asking how to do it |
| 21:16.01 | ManxPower | Do you want the person that is called to be able to transfer themselves to international numbers? |
| 21:16.04 | *** mode/#asterisk [-s] by bkw_ |
| 21:16.25 | Beirdo | there we go :) |
| 21:16.31 | Darwin35 | -s ? |
| 21:16.43 | Beirdo | yes |
| 21:16.53 | Darwin35 | I forget what it is |
| 21:16.54 | bprice20 | ManxPower I see what you mean about digittimeout, i'll look into that, right now i am looking for some help on the non unique extension pattern issue |
| 21:17.07 | Beirdo | +s makes the channel secret |
| 21:17.13 | Darwin35 | ahh ok |
| 21:17.31 | *** join/#asterisk elric (~kavit@ppp114-10.static.internode.on.net) |
| 21:17.33 | Beirdo | which is why I was betting it wasn't intentional here :) |
| 21:17.42 | bprice20 | Manxpower, actually yes |
| 21:17.56 | bprice20 | Manxpower the calee no |
| 21:18.02 | bprice20 | call-ee |
| 21:18.12 | BrianR___ | Unless it's to keep the spambots out... |
| 21:18.24 | jas_williams | A secret channel with registered users only keeps the spam bots out |
| 21:18.43 | BrianR___ | Although the channels is also set to identified required, which is supposed to inhibit spambots somewhat.. |
| 21:18.45 | Beirdo | just registered only will keep out the majority of them |
| 21:19.01 | BrianR___ | pesky spammers... |
| 21:19.26 | *** join/#asterisk HeadachesAbound (user79@wsip-68-99-73-32.tu.ok.cox.net) |
| 21:19.36 | Beirdo | too bad there's no mode for "keep out total idiots"... I'm sure some channels would love that one |
| 21:20.04 | Darwin35 | ok asterisk @home is made for what hardware x86 or sparc |
| 21:20.12 | Darwin35 | or ppc |
| 21:21.19 | *** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl) |
| 21:21.39 | bprice20 | asterisk can run on x86 and ppc, not sure about sparc |
| 21:21.54 | bprice20 | asterisk@home ? probably x86 |
| 21:22.17 | bprice20 | yep x86 |
| 21:23.08 | *** join/#asterisk Goshen (~Goshen@c-67-172-238-57.client.comcast.net) |
| 21:27.32 | flewid | sup |
| 21:28.03 | flewid | i have two tdm cards (04/40) in a box, and using a default stock asterisk installation, when i call into the box via pstn, i get gsm errors if i try and play a soundfile |
| 21:28.16 | flewid | i hear maybe .2 seconds of the soundfile, then the screen just scrolls and scrolls gsm errors |
| 21:28.21 | flewid | all the caller hears is a bunch of clicking |
| 21:28.30 | flewid | cards are both using seperate irq's |
| 21:28.37 | eKo1 | What gsm erros? |
| 21:28.38 | flewid | tried two cards, same thing happens with each |
| 21:28.47 | flewid | sec i'll try it again |
| 21:29.20 | *** part/#asterisk RevK ([U2FsdGVkX@excalibur.bracknell.aaisp.net.uk) |
| 21:29.50 | flewid | WSWarning warningwfff |
| 21:30.08 | eKo1 | huh? |
| 21:30.19 | flewid | erm |
| 21:30.25 | flewid | sorry, my connection almost died:) |
| 21:30.52 | flewid | warning: format_gsm.c:164 gsmtolin_framein: invalid gsm data |
| 21:31.08 | flewid | warning: format_gsm.c:145 gsm_read: short read (6) no such file or directory |
| 21:31.20 | flewid | but the files exist, and it seems to be playing like .2-.6 seconds |
| 21:31.22 | *** join/#asterisk fugitivo (~ajf@201.255.105.251) |
| 21:31.30 | flewid | then goes into screenfulls of these errors |
| 21:31.38 | BrianR___ | Is there a WHat's New for 1.0.6 somewhere? |
| 21:32.08 | BrianR___ | (aside from reading the cvs logs) |
| 21:33.09 | eKo1 | flewid: What file? |
| 21:33.24 | eKo1 | Maybe the files is corrupted? |
| 21:35.15 | jas_williams | flewid: What type of HDD system RAID ? Has the system enough interrupt time to service the disk read ? |
| 21:36.30 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 21:37.02 | *** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl) |
| 21:37.55 | Darwin35 | Boris Baby |
| 21:38.47 | flewid | eKo1: i've tried multiple files |
| 21:38.49 | BoRiS | Darwin!!!!! |
| 21:38.52 | flewid | all the stock sounds that come with asterisk |
| 21:38.55 | BoRiS | Wasssssssssup? |
| 21:38.56 | flewid | so i don't think the files are borked |
| 21:39.12 | flewid | jas_williams: hdd is a removable drive |
| 21:39.18 | eKo1 | flewid: Are you using head or stable? |
| 21:39.23 | flewid | head |
| 21:39.28 | flewid | doesn't matter with either tho |
| 21:39.52 | jas_williams | flewid: How is the removable drive attached to the system USB ? |
| 21:39.57 | YoYo | can anyone reccomend a GOOD player to listen to my voicemail in linux? prefer gui so that I can quickly pause and/or rewind. I've tried gmplayer, but it garbles the audio if I touch the slider bar |
| 21:40.24 | Juggie | why not a sip/iax phone |
| 21:40.28 | Juggie | to call the vm system |
| 21:40.48 | eKo1 | flewid: So on stable, it does the same thing? |
| 21:40.54 | YoYo | Juggie: because we get voicemail via email |
| 21:41.28 | Juggie | ah.... for me winamp plays them find |
| 21:41.30 | Juggie | *fine |
| 21:41.58 | YoYo | winamp... hrrm... wonder if they still make a linux version |
| 21:42.18 | eKo1 | get xamp |
| 21:42.41 | fugitivo | xmms |
| 21:43.21 | |Vulture| | Anyone using Asterisk Flash Operator Panel? |
| 21:43.33 | JerJer[mobile] | does anyone know if one can set the TFTP server option on a 7690 via the telnet interface? (dumb user alert) |
| 21:45.46 | flewid | jas_williams: removable drive is attached via internal connection - looks to be 24 pin female slot of some type |
| 21:47.57 | Juggie | does anyone have a link to that activex sip client that xten produces? |
| 21:48.10 | jas_williams | JerJer[mobile]: I don't think so |
| 21:49.05 | flewid | yeah it's a superrack |
| 21:49.11 | flewid | removeable docking modules for ide or usb devices |
| 21:49.15 | flewid | looks to be ata10-0 |
| 21:49.17 | flewid | er 100 |
| 21:50.16 | |Vulture| | Anyone using Asterisk Flash Operator Panel? I get it to connect to * but it isn't viewing active/inactive extensions |
| 21:50.18 | bjohnson_ | damn. can't figure out why half of my iax peers are unreachable |
| 21:50.34 | jontow | vulture; i just set it up yesterday.. if that counts :) |
| 21:50.37 | bjohnson_ | wooo hooo .. ordered my WRT54G |
| 21:50.45 | |Vulture| | jontow: worked fine? |
| 21:50.51 | jontow | yeah, few/no problems |
| 21:50.52 | |Vulture| | bjohnson: you will be happy |
| 21:51.03 | |Vulture| | jontow: HEAD or v1-0? |
| 21:51.04 | bjohnson_ | found out that it can work as wireless bridge too |
| 21:51.07 | jontow | v1-0 |
| 21:51.14 | |Vulture| | odd... |
| 21:51.25 | jontow | my biggest problems are getting multiple-server-monitoring working |
| 21:51.26 | andrewh | night all, thanks again to jas_williams |
| 21:51.43 | jontow | bbiaf, cig break :) |
| 21:52.43 | Goshen | does anyone have a sipura 2100 that can share their configuration? |
| 21:53.09 | |Vulture| | jontow: can you take a look at http://pastebin.ca/6808 and see if that looks like your example? |
| 21:53.17 | Goshen | I pick up the phone, and the green phone light(on sipura) blinks and I don't hear a dialtone |
| 21:54.03 | *** join/#asterisk znoG (gs@200.115.216.109) |
| 21:55.11 | *** part/#asterisk andrewh (~Andrew_He@andrew.1ec.aaisp.net.uk) |
| 21:55.28 | flewid | hmm |
| 21:55.32 | flewid | okay i just tested by putting |
| 21:55.38 | flewid | s,1,VoicemailMain |
| 21:55.44 | flewid | and i can check my voicemail and it plays the sounds fine |
| 21:55.54 | flewid | so i think that removes the chance that it's the removable drive's fault |
| 21:57.13 | jas_williams | k |
| 21:58.11 | bjohnson_ | can a iax.conf section name include a dash for a type=peer and still do dial(iax2/section-name/exten) ? |
| 21:58.41 | flewid | i don't get it - two tdm cards and still getting this |
| 21:58.42 | flewid | hmm |
| 21:58.56 | bjohnson_ | Goshen: with my 2000 that usually mean it isn't registered to the server |
| 22:00.37 | jontow | vulture; i don't see anything immediately wrong with that, compared to mine.. |
| 22:00.42 | Goshen | bjohnson_: I got it to register now, and that fixed it thank you, have to played with many of the settings? there are so many settings! |
| 22:01.08 | jontow | whats the server output say? :) |
| 22:01.49 | bjohnson_ | Goshen: find the wiki page that gives an url on voxilla to a spa 2000 howto. if * and spa are on same lan, only about 4 things to set to work (and then you can tweak volume, etc) |
| 22:02.17 | Goshen | I used the vocilla howto to configure it |
| 22:02.42 | Goshen | It doesn't go over any of the settings though |
| 22:02.55 | Goshen | other then the basic...get it to works settings |
| 22:03.04 | |Vulture| | jontow: Ill pastebin |
| 22:03.19 | Goshen | Where can I find sipura dial plan info? |
| 22:03.52 | |Vulture| | jontow: http://pastebin.ca/6810 |
| 22:04.22 | Zaw | does anyone use iaxtel? i'm trying it but the sound quality is horrible. choppy, laggy, etc. and that's when i can even get through. |
| 22:04.27 | jontow | i mean the output from op_server.pl |
| 22:04.29 | jontow | not the CLI :) |
| 22:05.52 | jontow | for example: evento regstatus SIP/132|ip|192.168.2.103|YYYY-0|SIP/132-XXXX |
| 22:06.04 | |Vulture| | jontow: http://pastebin.ca/6811 |
| 22:06.30 | jontow | it never shows anything else? |
| 22:06.59 | |Vulture| | nope |
| 22:07.03 | jontow | odd |
| 22:07.04 | |Vulture| | its strange |
| 22:07.08 | Inv_arp | Zaw: with what provider? |
| 22:07.16 | |Vulture| | Ive done it on 2 servers now... same result |
| 22:08.05 | jontow | ok |
| 22:08.11 | jontow | what browser are you using, and what version of flashplugin? |
| 22:08.12 | |Vulture| | ah I have debug off |
| 22:08.19 | jontow | so do i ;) |
| 22:08.36 | |Vulture| | IE6.. it works fine with the demo |
| 22:08.50 | jontow | i've tested it working with IE6+FP7, and Mozilla/Linux with flashplugin7 |
| 22:09.13 | *** join/#asterisk NirS (Nir@l192-117-110-178.cable.actcom.net.il) |
| 22:09.56 | |Vulture| | I have it on max debug now |
| 22:10.07 | jontow | what IP is your browser running from? (adding you to my ACLs to test mine) |
| 22:10.18 | |Vulture| | it is seeing when I dial etc. |
| 22:10.29 | *** join/#asterisk dimmik (~dimmik@ip-92-216.first.gr) |
| 22:10.31 | jontow | ok, just not seeing client IPs or anything like that? |
| 22:11.06 | jontow | one annoying thing i find with that panel is that when i hit alt+F# to switch workspaces, it pops up the queue/call window ;) |
| 22:11.10 | |Vulture| | strange it sees everything |
| 22:11.37 | dimmik | Is there a way to deny sip transfers to a specific sip extention ? |
| 22:11.53 | jontow | dimmik; contexts? |
| 22:11.57 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 22:11.58 | Zaw | Inv_arp: i'm not sure i know what you're asking, with iaxtel.com and their 1700 network |
| 22:12.17 | *** part/#asterisk DrRighteous (~DrRighteo@ool-182c867b.dyn.optonline.net) |
| 22:12.31 | dimmik | jontow: I don't think so |
| 22:13.04 | *** join/#asterisk muckdog (~jay@coal.acdmis.com) |
| 22:13.13 | dimmik | jontow: I want a sip agent to be able to call an external call but not transfer to an external call |
| 22:13.34 | dimmik | jontow: context will prevent both |
| 22:13.54 | jontow | ok, sounds like you're gonna get waist deep in extension logic for that one then ;) |
| 22:14.06 | dimmik | :) |
| 22:15.24 | dimmik | I dont think that this is posible with an extention logic |
| 22:15.33 | jontow | .. you'd be surprised what is possible with extension logic then |
| 22:15.44 | dimmik | since sip transfers are part of chan_sip |
| 22:15.46 | jontow | what i would like to see on the voip-info wiki is a section devoted to people's crazy examples |
| 22:16.04 | jontow | specifically extension logic |
| 22:16.19 | dimmik | that would be nice |
| 22:16.41 | jontow | good existing example is the chan_agent-less agent channels ;) |
| 22:17.03 | jontow | someone didn't like chan_agent, so they rewrote it purely in the dialplan |
| 22:17.10 | jontow | its brilliant, long, and complicated.. but thats what i'd like to see |
| 22:17.31 | jontow | lots of people learn better by example than by theory |
| 22:17.41 | fugitivo | i agree |
| 22:17.47 | dimmik | me too |
| 22:18.34 | MajestiK | How can you tell if a SPA-2000 is locked or unlocked? |
| 22:18.47 | Beirdo | look at the keyhole? |
| 22:18.51 | nestAr | lol |
| 22:18.53 | flewid | fric |
| 22:18.59 | flewid | sorry guys, damn wireless at work sucks |
| 22:19.02 | flewid | back on wired connect now |
| 22:19.11 | Beirdo | hey now... :) |
| 22:19.15 | flewid | so yeah, i'm still getting this same gsm problem, no matter what i try |
| 22:19.23 | flewid | anyone have any ideas besides throwing the computer out the window? |
| 22:19.24 | flewid | :) |
| 22:19.35 | *** join/#asterisk afe ([lb+bCl9I4@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 22:20.39 | *** join/#asterisk Legend (~legend@24.244.142.133) |
| 22:22.50 | roamer323 | MajestiK - can you access the web server on the spa2k? |
| 22:23.11 | MajestiK | roamer323: no idea, I'm trying to buy the thing, just want to make sure it's not locked before I shell out money for it |
| 22:23.50 | roamer323 | MajestiK - I don't think you can tell until you actually get your hand on the unit :-( |
| 22:24.07 | MajestiK | Well, the guy who has it could, and I could get him some directions :) |
| 22:24.34 | MajestiK | Of course, I'd have to believe him.. but I can accept that.. or get him to throw it on a public IP :) but that's probably a bad idea |
| 22:25.02 | BrianR___ | Is the iLBC codec in asterisk Free software? |
| 22:25.42 | *** part/#asterisk wrmem (~wrmem@monnin-win.cso.uiuc.edu) |
| 22:26.29 | flewid | okay, just to make sure it's not cause it's a removable drive, i've now put my gsm file into /etc/asterisk/ and tried Playback(./goodbye) |
| 22:26.33 | flewid | and i still get these errors |
| 22:26.36 | flewid | w t f |
| 22:27.20 | Goshen | In my sipura 2100, this dialplan is sending local numbers to asterisk missing the first digit |
| 22:27.21 | Goshen | (xxx|xxxx|xxxxx|xxxxxx|xxxxxxx|1xxxxxxxxxx|61xxxxxxxxxx|7*1xxxxxxxxxx|7**1xxxxxxxxxx) |
| 22:27.34 | Goshen | so if I dial 2556690 asterisk sees 556690 |
| 22:27.40 | Goshen | any ideas? |
| 22:28.02 | Goshen | it wasn't passing anything to asterisk until I added a seven digit xxxxxxx |
| 22:32.28 | *** part/#asterisk wizhippo (~wizhippo@Quebec-HSE-ppp233869.qc.sympatico.ca) |
| 22:34.00 | eKo1 | I think your spa-2100 is bad. |
| 22:36.16 | *** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com) |
| 22:40.59 | *** join/#asterisk Mike (~mike@201.135.48.217) |
| 22:47.48 | Goshen | I missed a exten:1 in my dialplan, my bad |
| 22:49.30 | Inv_arp | have 2 sip phones on a conversation, how can i bridge them together |
| 22:50.20 | modulus_ | ooh |
| 22:50.21 | modulus_ | ip600 |
| 22:50.23 | modulus_ | *eviLGrin* |
| 22:51.19 | *** join/#asterisk NirS (Nir@l192-117-110-178.cable.actcom.net.il) |
| 22:53.55 | TSCHAK | hmm |
| 22:54.16 | TSCHAK | say I want the # 6647 to go to extension 6647 in extensions.conf |
| 22:54.24 | Inv_arp | where do i start to setup conferencing? |
| 22:55.36 | eKo1 | Inv_arp: features.conf |
| 22:56.35 | Inv_arp | thx |
| 22:57.54 | *** join/#asterisk ast_freak (~ast_freak@hades-out.universalsystems.net) |
| 22:58.03 | *** part/#asterisk Stitch (~magliett@cpe-69-204-16-160.rochester.res.rr.com) |
| 22:58.12 | *** join/#asterisk mhess (~mhess@66.185.98.148) |
| 22:58.47 | TSCHAK | how do I route calls with asterisk-oh323? |
| 22:59.14 | TSCHAK | I've got it registered with a gatekeeper, and I want to use a terminal, also connected to the gatekeeper, to dial extension 1234 on the asterisk box. |
| 22:59.24 | *** join/#asterisk Ahewes (~rsb@209.81.2.58) |
| 22:59.25 | cstdenis | I have 3 asterisk servers and am having some sip codec problems. Call path is server1 (ulaw/alaw) -> server2 (ulaw/alaw/g729A) -> server3 (g729A). However, server2 tries to directly connect them together it seems and the call fails because it can't make the codec's compatable. How do I make it do the codec translation? |
| 22:59.36 | *** join/#asterisk JohnnyC (~JoaoCorre@81.193.116.63) |
| 22:59.54 | JohnnyC | anyone has an Asterisk working so we/she can call be using IP ? |
| 23:00.00 | JohnnyC | Asterisk->Asterisk ? |
| 23:00.21 | Ahewes | no, no one. |
| 23:00.46 | Ahewes | JohnnyC just be more specific about your problem and we'll try to help. |
| 23:01.16 | JohnnyC | Ahewes: I just wanted to test asterisk with someone calling IP to another VoIP system |
| 23:02.04 | afe | Anyone using cidstart=polarity in zapata.conf? |
| 23:02.07 | ariel_ | JohnnyC, try calling your system like extennumber@yourIP for the test |
| 23:02.20 | JohnnyC | hmm oki doki Ariel Thanks |
| 23:02.37 | JohnnyC | btw can you point out a script to call IP addresses ? |
| 23:03.02 | ariel_ | oki doki sounds like from the Rollie Pollie Ollie. |
| 23:03.47 | Goshen | sweet, message waiting indicator works |
| 23:04.02 | Goshen | stutter tone anyhow |
| 23:04.44 | JohnnyC | It works ! |
| 23:04.52 | JohnnyC | ariel_: it works ! :) |
| 23:04.55 | *** join/#asterisk FryGuy- (fryguy@c-24-23-19-33.client.comcast.net) |
| 23:05.25 | ariel_ | Goshen, yes it does work here. |
| 23:05.26 | afe | I tried reporting a bug but it was closed as a support issue, and I haven't yet got an answer from support ... |
| 23:05.53 | ariel_ | what was the problem? |
| 23:06.26 | afe | ariel: my TDM11B calls back to my incoming context after hanging up a call |
| 23:06.35 | JohnnyC | ariel_: instead of calling extension@ipaddress how can I call general IP address ? |
| 23:06.53 | JohnnyC | what I have to put in extensions.conf ? |
| 23:07.17 | *** join/#asterisk Grooby (~Grooby@12.22.232.212) |
| 23:08.52 | TSCHAK | does any1 in here know how to configure asterisk-oh323 ? |
| 23:09.04 | nestAr | ugh |
| 23:09.08 | nestAr | i tried |
| 23:09.08 | ariel_ | afe, mine does that some times as well. |
| 23:09.10 | nestAr | gave up |
| 23:09.13 | TSCHAK | i've got it configured |
| 23:09.16 | TSCHAK | i mean, compiled |
| 23:09.22 | TSCHAK | and it works fine... it registers with my GK |
| 23:09.27 | ariel_ | JohnnyC, use the wiki it's there. A dialing plan for ip address. |
| 23:09.30 | ariel_ | ~docs |
| 23:09.31 | 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 |
| 23:09.32 | TSCHAK | but I want to know how to configure the endpoint parameters for it. |
| 23:09.40 | JohnnyC | oki |
| 23:09.42 | *** join/#asterisk tuxinator_linux2 (~tuxinator@67.155.218.2) |
| 23:09.43 | nestAr | got father than i ever got |
| 23:09.53 | nestAr | further** |
| 23:10.07 | afe | ariel_: I increased RING_DEBOUNCE from 64 to 128 in wctdm.c and it doesn't happen as much after that, but still ... |
| 23:10.13 | ariel_ | h323 works but it's a bare to configure. I don't use it at all anymoe. |
| 23:10.26 | ariel_ | ./anymoe /any more |
| 23:10.31 | TSCHAK | ariel_ the h323 channel or the oh323 one? |
| 23:10.53 | TSCHAK | I just need to know how to route calls. |
| 23:10.55 | ariel_ | TSCHAK, either one |
| 23:11.09 | afe | I tried with RING_DEBOUNCE at 256 ms, but astersi wouldn't start properly after that :) |
| 23:11.09 | ariel_ | you route them by context |
| 23:11.44 | TSCHAK | ariel_ like...say I have asterisk registered via gatekeeper... I want whenever somebody calls 6647 to go to mailbox 6647.. how would I do that? |
| 23:11.50 | TSCHAK | the docs are um... nonexistant. |
| 23:12.59 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 23:14.49 | *** join/#asterisk ipso (~ipso@d207-81-249-35.bchsia.telus.net) |
| 23:15.04 | *** join/#asterisk JerJer (~jj@feth100-fw.fament.net) |
| 23:16.10 | ariel_ | TSCHAK, JerJer is the person to ask about h323 |
| 23:16.22 | TSCHAK | ok, I found some smidgen of docs |
| 23:16.28 | TSCHAK | I'm going to read them and try some stuff first |
| 23:16.53 | TSCHAK | this LOOKS simple enough....although...heh, |
| 23:17.03 | shido6 | heh |
| 23:17.20 | shido6 | its not that difficult - the README tells you just about everything u need to know |
| 23:18.03 | dimmik | does moh work with 1.0.6? |
| 23:18.10 | Inv_arp | what are the options called to setup conferencing? |
| 23:18.16 | Inv_arp | dimmik: heard its broken |
| 23:18.31 | dimmik | sounds correct |
| 23:18.41 | ariel_ | dimmik, yes |
| 23:18.47 | ariel_ | Inv_arp, meetme? |
| 23:19.07 | Inv_arp | meetme? is for general three way conferencing? |
| 23:19.18 | dimmik | ariel_: yes it works? |
| 23:19.32 | Inv_arp | dimmik: no it doesnt |
| 23:19.33 | JerJer[mobile] | Inv_arp: more like 100 way conferencing |
| 23:19.37 | ariel_ | yes it does at least it's working on my setup. |
| 23:20.08 | dimmik | ariel_: using 1.0.6? |
| 23:20.10 | Goshen | dimmik: works on my system |
| 23:20.12 | Inv_arp | heh well i just need to setup it up where do i start? meetme.conf or somethin? |
| 23:20.16 | ariel_ | dimmik, yes |
| 23:20.27 | ariel_ | I got it this morning. |
| 23:20.53 | dimmik | so is it just me and Inv_arp ? |
| 23:20.56 | Inv_arp | hmm MOH must have been fixed recently |
| 23:21.13 | Inv_arp | yesterday everypne was complaining it wasnt working |
| 23:21.25 | JohnnyC | ariel_: I have a BRI , when I call my self I see Asterisk on the SIP Phone ! how can I put something else ? |
| 23:22.04 | dimmik | Inv_arp: so is there another 1.0.6? |
| 23:22.23 | ariel_ | JohnnyC, use setcallid http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20SetCallerID |
| 23:22.47 | Inv_arp | dimmik: prob a simple fix to it today *shrug* |
| 23:23.35 | ariel_ | Inv_arp, look for info on setting up your meetme on the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20meetme.conf |
| 23:24.28 | *** join/#asterisk mikeirc (~mikeirc@6532112hfc30.tampabay.res.rr.com) |
| 23:24.39 | JohnnyC | ariel_: setcallid is set where ? sip.conf in general ? |
| 23:24.44 | Inv_arp | ariel_: yea in there now thx |
| 23:24.51 | Goshen | How do you have a generic extension for users to check their mailbox and pass the extension they are calling from to the voicemailmamain2? |
| 23:25.01 | Inv_arp | JohnnyC: extension.conf |
| 23:25.30 | ariel_ | I gess people don't use the wiki much today. |
| 23:25.47 | mikeirc | Hey..just a quick question... If I'm doing JUST SIP... do I still need to check and compile zaptel (with ztdummy enabled) in order to have music on hold?? |
| 23:26.01 | Goshen | I have this much... exten => *98,1,VoicemailMain2 what variable goes on the end to specify the voicemail box of the extension that is dialing in? |
| 23:26.01 | Inv_arp | Goshen: with {$EXTEN} var and setting up a exten=> to go to voicemail i assume |
| 23:26.33 | ariel_ | Goshen, it's just a dialing rule http://www.voip-info.org/wiki-Asterisk+VoiceMail |
| 23:27.01 | ariel_ | voicemailmain2 is not needed any more |
| 23:27.11 | Goshen | just following the wiki |
| 23:27.14 | Goshen | what is used now? |
| 23:27.44 | ariel_ | Goshen, exten => 850,1,VoicemailMain |
| 23:27.44 | ariel_ | exten => 850,2,Goto(incoming,s,1) |
| 23:27.44 | ariel_ | exten => 855,1,Voicemailmain(s${CALLERIDNUM}) |
| 23:27.44 | ariel_ | ; |
| 23:27.44 | nestAr | VoicemailMain(${CALLERID}) |
| 23:27.48 | nestAr | yeah |
| 23:27.49 | nestAr | num |
| 23:27.54 | nestAr | my bad |
| 23:28.21 | ariel_ | instead of 850 and 855 use *98 if you want. |
| 23:28.27 | Goshen | thank you |
| 23:28.30 | *** join/#asterisk cbachman (~chatzilla@victory.ece.northwestern.edu) |
| 23:28.51 | Goshen | wow, asterisk keeps dialing your sipura extension with one ring to let you know you have voicemail ehh? neat :) |
| 23:29.15 | ManxPower | ~docs |
| 23:29.16 | 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 |
| 23:29.18 | ariel_ | Goshen, yes it's not asterisk it's the sipura that does that. it's a setting you can change on it. |
| 23:29.26 | Goshen | ahh I see |
| 23:29.28 | Goshen | I like it |
| 23:29.42 | ariel_ | ManxPower, how are you doing today? |
| 23:30.01 | Goshen | sitting here with my grandstream flashing at me(voicemail waiting) and sipura cordless ringing |
| 23:30.08 | ariel_ | Seems your right about asking before reading.... |
| 23:30.20 | flewid | hello again :) |
| 23:30.49 | flewid | when dialing multiple extensions (call groups) and i have Dial(SIP/EXT1&SIP/EXT2 etc |
| 23:30.55 | ariel_ | ok it's dinner time and I have to make it tonight. See you all later. |
| 23:31.01 | Goshen | night |
| 23:31.11 | flewid | can i add in a &IAX2/server:password@host/ext@context,24,twm) |
| 23:31.12 | flewid | in there? |
| 23:31.22 | flewid | or is there a better way to dial users connected via a slave server |
| 23:31.34 | zapa | hi does anybody know how to set echo cancell in a polycom 300 |
| 23:32.46 | TSCHAK | ok, I added an [ic] section into extensions.conf .. with one extension --- s,1,voicemail(u665) .... I then added under [register] in oh323.conf the context=ic and alias=665 .... anything else? |
| 23:35.40 | *** join/#asterisk dsfr (~dsfr@216.207.244.183) |
| 23:36.11 | JohnnyC | When I call via Asterisk I get private number if receiving on a Mobile Phone and "asterisk" when receiving on an IP PHONE! |
| 23:36.14 | JohnnyC | how can I change this ! |
| 23:39.09 | JamesDotCom | callerid in your sip.conf? |
| 23:39.22 | Goshen | who are you dialing out from? |
| 23:39.46 | Goshen | what service provider? some don't allow you to set the caller ID |
| 23:40.11 | JohnnyC | Goshen: Im from Portugal |
| 23:40.24 | Goshen | nice :) |
| 23:40.33 | JohnnyC | But if I connect the ISDN line to a "normal" phone and call my Mobile I can see the number |
| 23:40.40 | Goshen | JohnnyC: how are you placing your call? |
| 23:40.45 | JohnnyC | 1 min |
| 23:40.55 | JohnnyC | exten => _XXX.,2,Dial(CAPI/@265541786:${EXTEN}|30|T) |
| 23:41.22 | Goshen | I no knothing about CAPI lines or where you put the caller ID for those |
| 23:41.49 | Goshen | how about doing this... |
| 23:42.29 | mikeirc | Ok... It seems that I need zaptel/ztdummy for the timing if I want music on hold even though this is a SIP ONLY setup. I just wanted to make sure this was the case though before I go compiling it in.. Thanks for any info. |
| 23:43.04 | JohnnyC | Goshen: Im using a ISDN line |
| 23:43.13 | JohnnyC | my modem is AVM Fritz |
| 23:43.31 | bonez41 | question: My wife has an 800 number which is pointed to her cell phone...when the call is picked up..on the cell, it displays 1 111 111 1111... never the actual number dialed incoming...why won't the 800 # provider forward that information? |
| 23:43.35 | afe | I would like to automatically send all my monitored call files to an email address. |
| 23:43.54 | afe | Anyone has an example of how to do that? |
| 23:44.03 | bonez41 | could they, technically, if they would...forward that information? |
| 23:44.06 | Goshen | [globals] |
| 23:44.06 | Goshen | MYNAME=My Name |
| 23:44.06 | Goshen | MYPHONE=8015559999 |
| 23:44.14 | Goshen | [Dialing Context] |
| 23:44.14 | Goshen | exten => s,1,SetCallerID(${MYPHONE}) |
| 23:44.14 | Goshen | exten => s,2,SetCIDName(${MYNAME}) |
| 23:44.14 | Goshen | exten => _XXX.,3,Dial(CAPI/@265541786:${EXTEN}|30|T) |
| 23:44.17 | Goshen | How about that? |
| 23:45.02 | Goshen | mikeirc: I read that you need ztdummy if you don't have a card in your system |
| 23:45.06 | JohnnyC | let me try |
| 23:45.19 | *** join/#asterisk MikeJ[Jayden] (~ircatjerr@65.170.43.34) |
| 23:45.32 | MikeJ[Jayden] | anyone around using head that can update and test somthing? |
| 23:46.09 | dimmik | I think it's just _XXX.,2,Dial(CAPI/265541786:${EXTEN}|30|T) |
| 23:46.13 | mikeirc | Goshen: Yeah, I read that too...I just wanted to confirm with someone before I go including extra stuff I don't need. Thanks. |
| 23:46.21 | afe | I found some script on the web that use mutt to email a file, but my shell programming knowledge is very limited |
| 23:46.30 | dimmik | and 265541786 must be in your capi.conf |
| 23:46.43 | afe | so I was wondering if anyone here did something similar :) |
| 23:47.16 | JohnnyC | Goshen: same thing it doesnt identifies the number :( |
| 23:47.16 | JohnnyC | |
| 23:49.07 | dimmik | Goshen: in capi.conf put msn=265541786 and then use _XXX.,2,Dial(CAPI/265541786:${EXTEN}) |
| 23:49.53 | Goshen | dimmik: tell JohnnyC :) I was just trying to help |
| 23:50.04 | JohnnyC | Thanks dimmik |
| 23:50.06 | dimmik | :) |
| 23:50.07 | JohnnyC | I'll try it |
| 23:50.43 | dimmik | missed at copy & paste |
| 23:51.20 | JohnnyC | Thanks dimmik ! |
| 23:51.26 | dimmik | np |
| 23:51.29 | JohnnyC | You'r the best ! :) thanks its working |
| 23:51.33 | JohnnyC | yeahhhhhh |
| 23:52.36 | *** join/#asterisk footnote (~jhicks@67.141.135.121) |
| 23:52.39 | JohnnyC | BTW Im using Mac OSX and the X-Ten phone is very poor in features ! I can't transfer a call for example. Is there any soft phone you know ? |
| 23:52.44 | dimmik | so, moh is broken on 1.0.6? I saw a posting on the list saying upgrade to cvs stable... |
| 23:53.04 | footnote | hrm |
| 23:53.16 | Grooby | isn't transfer in x-ten the "#" key? |
| 23:53.18 | Goshen | 1.0.6 MOH works fine for me.. |
| 23:53.31 | Grooby | x-lite i mean |
| 23:53.56 | Sedorox | Grooby: only if its enabled on the dial line in *, I think |
| 23:53.59 | footnote | Anybody here familiar with the setup of the cvsupd? |
| 23:54.05 | Grooby | i c i c |
| 23:54.16 | Grooby | the t flag |
| 23:54.21 | JohnnyC | X-lite , how do you transfer calls ? |
| 23:54.29 | dimmik | Goshen: I have upgarded from 1.0.5. It was working fine. Now it's gone (maybe only from the sip side) |
| 23:54.36 | Grooby | if you have the T flag in your dialplan |
| 23:54.39 | Grooby | you can hit the # key |
| 23:54.45 | Goshen | dimmik: I installed 1.0.6 on a mandrake 10.0 system running 2.6.10 kernel |
| 23:55.28 | Goshen | dimmik: I just dialed extension 23 on my server from my sip phone and I hear MOH |
| 23:55.33 | footnote | Under FreeBSD I'm used to keeping a local copy of the repo with cvsupd. Having a local copy of the Asterisk repo would be sorta handy... |
| 23:55.54 | Goshen | dimmik: did you compile and install the asterisk addons? |
| 23:56.07 | dimmik | Goshen: no |
| 23:56.21 | Goshen | isn't the MOH in there? |
| 23:56.26 | dimmik | Goshen: I have the same behaviour as here: http://lists.digium.com/pipermail/asterisk-users/2005-March/092470.html |
| 23:56.37 | afe | anyone with a TDM499 that could take a look at bug http://bugs.digium.com/bug_view_page.php?bug_id=0003700 and see if they solved it (I know the bug is closed) :) |
| 23:56.49 | afe | that would be a TDM400 |
| 23:57.29 | afe | and I meant it you had a solution - not Digium ... |
| 23:57.35 | Goshen | oh so, MOH extensions work ok? but not others? |
| 23:57.48 | Goshen | I get MOH when I dial one extension from another extension in 1.0.6 |
| 23:59.56 | dimmik | Goshen: can you try this: exten => _16,1,WaitMusicOnHold(200) |
| 23:59.56 | shido6 | then u have |m or ,m there at the end |
| 23:59.56 | mgth | afe: I have the same issue and I have called support and they have told me to recompile my kernel |
| 23:59.56 | shido6 | at the end of the Dial exten line u can add m instead of r |
