00:00.18 | *** join/#asterisk Weezey (WeezeyD@206.210.109.233) |
00:00.55 | *** join/#asterisk iceyp (~icepick@202.150.105.150) |
00:01.01 | Nugget | I guess I just prefer less mainstream films. |
00:01.07 | PTG123 | ah |
00:01.32 | iceyp | are budgetones still the cheapest phones out there |
00:01.34 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
00:01.49 | shmaltz | how can I find out on a blind transfer who is dialing the number? |
00:01.55 | shmaltz | I need it for billing |
00:02.00 | shmaltz | using SIP |
00:02.14 | Nugget | three great films I've watched in the past month which are not on your list: amelie, the station agent, and house of sand and fog. |
00:02.25 | shmaltz | or is there any way I could find out that it is a blind transfer |
00:02.47 | PTG123 | never even heard of those |
00:02.57 | *** join/#asterisk jhowardPA (~jhoward@12.25.177.120) |
00:03.12 | jhowardPA | Hello people! |
00:03.19 | Nugget | Hello jhowardPA! |
00:04.02 | jhowardPA | I've got a problem with some cisco 7940's - know anything about 'em? |
00:05.48 | Nugget | what's the problem? |
00:06.18 | jhowardPA | Well, when I make a call from one to another, the call is muted until I hold and resume it. |
00:07.41 | *** join/#asterisk likwid-- (likwid@nc-67-77-138-97.dyn.sprint-hsd.net) |
00:08.13 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
00:08.39 | Nugget | strange, I've never heard of that. any suspicious messages in the asterisk console? |
00:08.52 | jhowardPA | Nothing obvious... :\ |
00:09.05 | jhowardPA | damn, phone call ;) |
00:10.42 | PTG123 | anyone need any servers, got a couple of AMD 1700s and some other ones gonna sell for $200-250 a piece? :) |
00:10.44 | PTG123 | rackmount |
00:12.11 | *** join/#asterisk NormAst (~NormAst@toronto-HSE-ppp3959569.sympatico.ca) |
00:12.20 | Sedorox | specs? |
00:12.37 | Weezey | If anyone's looking for a nice yet cheap headset, the Plantronics M175 is gret. |
00:12.38 | PTG123 | dual 60gig ide drives, 512 memory, 2u |
00:12.40 | Weezey | great too. |
00:13.02 | PTG123 | got some dual 1u 1gz ones too.. even got a scsi based with a dpt V raid card 2u dual 1gz but i may keep that |
00:13.04 | Sedorox | not bad specs |
00:13.27 | PTG123 | i got a stack of servers, just trying to get rid of them first come first serve basis :) |
00:13.31 | Sedorox | too bad i dun have the money.. i could use one |
00:13.31 | PTG123 | from my hosting days |
00:13.53 | Sedorox | hehe |
00:14.06 | PTG123 | hrmph |
00:14.14 | PTG123 | go rape an pillage your neighbors |
00:14.15 | PTG123 | :) |
00:14.17 | robl^ | a little off topic -- but -- http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=1469&item=5570085385&rd=1&tc=photo#ebayphotohosting |
00:14.20 | Sedorox | take payment plans? :-p |
00:14.25 | Hogie | is there like a ringing sound, or something I could use for intercom notification inside the sounds dir? I dont see anything |
00:14.31 | PTG123 | hah afraid not :) too much work |
00:14.54 | robl^ | Hogie, I just use the beep.gsm :) |
00:15.18 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l03m-16-26.d4.club-internet.fr) |
00:15.24 | shmaltz | how do I add a new page to the wiki? |
00:15.26 | Hogie | robl^: I was hoping for like a 3beep, lol |
00:15.40 | robl^ | Hogie, play beep 3 times |
00:15.41 | shmaltz | I know how to edit, but I want to add a new page, how do I do that? |
00:15.50 | Sedorox | hmmm |
00:15.51 | Hogie | I can't with dial... |
00:16.06 | Sedorox | too bad I just got a new cell... coulda been able to do ~200... |
00:16.21 | Hogie | I want it played when the receiving phone (which is setup for auto answer on that instance) picks up |
00:16.49 | PTG123 | well damn you :) |
00:17.12 | Hogie | PTG123: i'd buy, but im putting $700/month into flight lessons, sorry:( |
00:18.12 | Sedorox | hey.. if your willing to hold one I could pay you in about a month.. but that isn't a option :-p |
00:18.14 | Sedorox | I gues |
00:21.04 | PTG123 | heh not really trying to clean out my garage/storage |
00:21.40 | Sedorox | Actually... |
00:21.41 | Sedorox | *thinks* |
00:21.53 | Sedorox | can I PM you? |
00:22.13 | *** part/#asterisk xai (~pasta@cpe-70-112-17-10.austin.res.rr.com) |
00:22.38 | PoWeRKiLL | !seen HellHound |
00:23.09 | niZon | PTG123: any free stuff? :P |
00:23.20 | Sedorox | ahah |
00:27.09 | Sedorox | PTG123: if you can do paypal... I wanna chat about it.. hehe |
00:28.32 | L|NUX | how can i set md5 password in asterisk and how can i create md5 pass ? |
00:29.10 | JunK-Y | L|NUX: md5 app? |
00:29.25 | L|NUX | md5 app ? |
00:29.30 | JunK-Y | show applications like MD5 |
00:30.09 | L|NUX | ok |
00:30.44 | L|NUX | NuFW*CLI> show applications like MD5 |
00:30.47 | L|NUX | <PROTECTED> |
00:31.07 | JunK-Y | which version? |
00:31.16 | L|NUX | 1.0.7 |
00:31.16 | JunK-Y | use HEAD |
00:31.56 | JunK-Y | goto in bug tracker and install these apps in u need just these. |
00:32.06 | JunK-Y | but i recommend head |
00:34.03 | *** join/#asterisk locoast (~locovox@218-153-89-200.fibertel.com.ar) |
00:34.23 | locoast | hi, anyone using simpletelecom.com? |
00:34.53 | niZon | ooo iax termination |
00:35.31 | locoast | I can call continental US but cant call intl. They telling me I have to use g729 (which i dont have) or g723.1 |
00:35.38 | Sedorox | seems kinda expensive... |
00:35.49 | locoast | when I configure the sip part to use g723.1 |
00:35.58 | locoast | asterisk says it cant make it compatible with my zapata |
00:36.02 | locoast | regular phone |
00:36.15 | *** join/#asterisk Hackett (~chatzilla@cuscon1882.tstt.net.tt) |
00:36.25 | locoast | Apr 13 21:32:33 WARNING[12267]: channel.c:2170 ast_channel_make_compatible: No path to translate from Zap/1-1(68) to SIP/simpleconnect-sip-be4f(1) |
00:36.25 | locoast | Apr 13 21:32:33 WARNING[12267]: app_dial.c:1260 dial_exec_full: Had to drop call because I couldn't make Zap/1-1 compatible with SIP/simpleconnect-sip-be4f |
00:36.32 | harryvv | simple g729 is a per port licenced codec. It also sounds the best |
00:36.57 | locoast | I'm trying to use g723. |
00:36.57 | malbech | Anyone konws a SoftSwitch Solution for a low cost ? |
00:37.15 | locoast | so I configured the SIP connection to use g723 |
00:37.24 | locoast | however the ZAP/1-1 is not working |
00:37.35 | file[mac] | G723.1 is not officially supported in asterisk |
00:37.37 | harryvv | get zap to work before anything else |
00:37.39 | locoast | with g723 and i dont know how to make it work |
00:37.43 | file[mac] | there is no official transcoder |
00:37.55 | file[mac] | so therefore, if asterisk has to convert between two codecs (in your case signed linear and G723) then it won't work |
00:37.57 | locoast | zap works if i use ulaw with simpletelecom |
00:38.14 | file[mac] | however, if both sides are using G723.1 then asterisk can act as a passthru and allow the audio to traverse to each |
00:38.22 | file[mac] | Thank you for choosing Asterisk. Have a peachy day! |
00:38.44 | locoast | how do i configure ZAP to use G723.1? |
00:38.51 | file[mac] | it can't |
00:39.14 | locoast | so I can't use G723 with simpletelecom... |
00:39.20 | file[mac] | correct |
00:39.26 | *** join/#asterisk obsidianr (~obsidianr@pcp03266395pcs.waldlk01.mi.comcast.net) |
00:39.38 | harryvv | first you need to know what zap is. Locoast how long have you spent learning asterisk |
00:40.17 | harryvv | file, by chance do you know if telcos use any known form of compression that is non propriatory? |
00:40.29 | file[mac] | define 'non proprietary' |
00:40.34 | file[mac] | like give me an example of a codec |
00:40.44 | harryvv | other then the ones we know of |
00:40.51 | harryvv | 7xx and gsm |
00:41.02 | harryvv | ilib I think was another |
00:41.02 | locoast | what are you talking about? |
00:41.12 | file[mac] | ilbc? nah... |
00:41.13 | locoast | 711u 711a gsm? |
00:41.16 | harryvv | locoast its a off subject |
00:41.18 | file[mac] | G729 and G723 are ths tandard |
00:41.20 | file[mac] | er the standard |
00:41.26 | file[mac] | G723 less because of the CPU needed for it |
00:41.26 | harryvv | for telcos? |
00:41.47 | file[mac] | well, for TDM equipment |
00:41.54 | harryvv | I am talking non voip carrier grade voice compression. |
00:42.55 | file[mac] | harryvv: your question confuses me though |
00:43.05 | JunK-Y | file: nice |
00:43.09 | mmlj4 | cluecon? |
00:43.22 | file[mac] | I'll be speaking about something or other |
00:43.28 | PTG123 | they need to make a daughterboard for g729 compression |
00:43.38 | locoast | any recommendation for the best/cheap SIP provider? |
00:43.45 | JunK-Y | ~seen jkerdev |
00:43.47 | jbot | JunK-Y: i haven't seen 'jkerdev' |
00:43.50 | blitzrage | locoast: I can provide SIP termination |
00:43.52 | JunK-Y | what's his nick again? |
00:44.00 | harryvv | file[mac]: Do telcos like singular varizon telus use voice compression in there standard every day Centeral offices |
00:44.03 | JunK-Y | ~seen jakevdev |
00:44.04 | jbot | i haven't seen 'jakevdev', JunK-Y |
00:44.11 | mmlj4 | blitzrage: details? |
00:44.17 | locoast | blitzrage, details? |
00:44.18 | file[mac] | harryvv: I highly doubt it |
00:44.21 | blitzrage | mmlj4: well, depends what you need :) |
00:44.29 | locoast | blitzrage: any web site? |
00:44.31 | mmlj4 | that's fair |
00:44.43 | locoast | SIP Providers... any other options? |
00:44.56 | niZon | blitzrage, how about termination to 204 in manitoba canada? :P |
00:45.28 | locoast | blitzrage, hurry wife calling |
00:45.45 | blitzrage | locoast: lol, sorry, the website isn't up right now I guess (I just started working for these guys :)) |
00:45.58 | blitzrage | locoast: drop me an email to leif@leifmadsen.com and I can get you details if you're interested |
00:46.24 | blitzrage | I just admin the network, not really doing sales :) |
00:46.57 | *** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au) |
00:47.05 | niZon | then you can sneak in some free options ;) |
00:47.17 | blitzrage | file[mac]: hehe... remember my postgres problems yesterday? Well, they are resolved, but because I kept installing overtop of the DB, it thinks there's users that aren't there :) |
00:47.29 | blitzrage | niZon: well... "I" get free options :D |
00:47.38 | niZon | lol |
00:47.46 | niZon | can you terminate to area code 204? |
00:47.49 | blitzrage | hey, anyone have 905 DID's in Oakville? |
00:48.06 | blitzrage | I can't seem to find anyone. I can 905 in Hamilton, but that does me no good |
00:48.17 | file[mac] | no DID for you! |
00:48.21 | blitzrage | *gasp* |
00:48.24 | file[mac] | and silly blitzrage for mucking with the database |
00:48.32 | blitzrage | file[mac]: well, its a test box, so no hard done |
00:48.34 | *** join/#asterisk lilneon (~tj_r3@cuscon12298.tstt.net.tt) |
00:48.35 | blitzrage | harm* |
00:48.39 | lilneon | good night all |
00:48.40 | blitzrage | LOL |
00:48.40 | JunK-Y | blitzrage: u fixed ur psql issues? |
00:48.41 | file[mac] | hardon? :p |
00:48.45 | file[mac] | muahahaha |
00:48.52 | blitzrage | file[mac]: I knew someone was going to catch that :D |
00:48.56 | blitzrage | JunK-Y: yep! thanks for the help last night |
00:49.00 | file[mac] | yeah yeah twisted minds think alike |
00:49.09 | JunK-Y | blitzrage: y owe me 1 beer. |
00:49.13 | blitzrage | JunK-Y: it was the version problem :) |
00:49.16 | blitzrage | JunK-Y: done and done |
00:49.34 | blitzrage | JunK-Y: that's cheap. 2 hours of service for 1 beer. Can I hire you? :) |
00:49.49 | blitzrage | a 2-4 ever 48 hours worth of work! |
00:49.51 | want561or772did | me want DID |
00:50.03 | JunK-Y | im like a dealer, when ya gonna be addict, i'll increase my rate :) |
00:50.12 | blitzrage | JunK-Y: LOL! |
00:50.14 | L|NUX | lol |
00:50.31 | want561or772did | here i am. rock you like a hurricane |
00:50.31 | jhowardPA | I still can't find any details on why my 7940's are muted when a call's connected, until I hold and resume it. Any new ideas? |
00:50.47 | blitzrage | jhowardPA: that sounds like a very odd problem |
00:50.54 | jhowardPA | It is ;) |
00:51.03 | obsidianr | anyone setup a siemens optipoint 100 advanced before? i'm having trouble |
00:51.04 | jhowardPA | I wish I knew where to start... |
00:51.40 | blitzrage | jhowardPA: hrmmmm... packet traces :) |
00:52.11 | jhowardPA | Yeah, that was step #2 after I verified that it is, in fact, a very odd problem. ;) |
00:52.12 | blitzrage | jhowardPA: do you have it nailed down to whether its the phone or asterisk? |
00:52.27 | jhowardPA | No sir. I'm new to Asterisk, but I'm a quick learner. |
00:52.37 | blitzrage | jhowardPA: thats what they all say! :) |
00:52.46 | blitzrage | jhowardPA: explain again what is happening? |
00:53.20 | riksta | crack pipe |
00:53.38 | jhowardPA | When I initiate a call between extn 200 and 204 (or any others), it rings normally. When I pick up, both ends are fully muted until one end holds and resumes the line. |
00:53.52 | *** join/#asterisk zotz (~zotz@24.231.32.109) |
00:54.40 | riksta | jhowardPA: pastebin your sip.conf and your phone .cnfs |
00:55.08 | jhowardPA | Doesn't matter which end does the hold->resume, either will suffice to unmute both. That's datum number one that leads me to suspect the server. |
00:55.16 | jhowardPA | ok |
00:55.44 | blitzrage | jhowardPA: and the relevant sections of your dialplan (extensions.conf) |
00:55.54 | riksta | yeah was just typin that |
00:55.59 | jhowardPA | [general] |
00:55.59 | jhowardPA | port = 5060 ; Port to bind to (SIP is 5060) |
00:55.59 | jhowardPA | bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) |
00:55.59 | jhowardPA | disallow=all |
00:55.59 | jhowardPA | allow=ulaw |
00:55.59 | jhowardPA | allow=alaw |
00:56.01 | jhowardPA | context = from-sip-external ; Send unknown SIP callers to this context |
00:56.03 | jhowardPA | callerid = Unknown |
00:56.08 | jhowardPA | #include sip_nat.conf |
00:56.09 | jhowardPA | #include sip_additional.conf |
00:56.12 | jhowardPA | There's my sip.conf |
00:56.32 | *** join/#asterisk TheEmperor (~user@203.121.47.165) |
00:56.40 | jhowardPA | here's one entry from sip_additional.conf (from AMP): |
00:56.43 | jhowardPA | [200] |
00:56.43 | jhowardPA | username=200 |
00:56.43 | jhowardPA | type=friend |
00:56.43 | jhowardPA | secret=111 |
00:56.43 | jhowardPA | qualify=no |
00:56.44 | jhowardPA | port=5060 |
00:56.46 | jhowardPA | pickupgroup= |
00:56.48 | jhowardPA | nat=never |
00:56.50 | jhowardPA | mailbox=200@default |
00:56.52 | jhowardPA | host=dynamic |
00:56.54 | jhowardPA | dtmfmode=rfc2833 |
00:56.56 | jhowardPA | disallow= |
00:56.58 | jhowardPA | context=from-internal |
00:57.00 | jhowardPA | canreinvite=no |
00:57.02 | jhowardPA | callgroup= |
00:57.03 | niZon | jhowardPA: http://www.pastebin.ca |
00:57.04 | jhowardPA | callerid="Jon Howard" <200> |
00:57.06 | jhowardPA | allow= |
00:57.12 | jhowardPA | Thanks, sorry. |
00:57.17 | blitzrage | eek! |
00:57.18 | Sedorox | .. |
00:57.23 | L|NUX | flood :D |
00:57.29 | Sedorox | now see what you've gone and done! |
00:57.36 | jhowardPA | I'm retarded. Sorry! |
00:57.42 | Sedorox | lol |
00:57.46 | blitzrage | lol |
00:57.47 | jhowardPA | Stress, lack of sleep, and too much coffee. |
00:57.50 | harryvv | yea |
00:57.55 | JunK-Y | jhowardPA: stop flooding and use www.pastebin.ca |
00:57.55 | blitzrage | jhowardPA: I've seen dumber :) |
00:58.02 | Sedorox | I'm Sofa King We Tar Did |
00:58.29 | *** join/#asterisk zilas (~1@adsl-158-98-233.mia.bellsouth.net) |
00:58.38 | zilas | hello all |
00:59.53 | zilas | one quick Q: what function you should use when you call lets say your voicemail and you dont hear the begining? |
00:59.56 | jhowardPA | http://pastebin.ca/9511 |
01:01.11 | jhowardPA | Flood-free! :D |
01:01.41 | Sedorox | This time.... |
01:01.46 | *** join/#asterisk PBXtech (~nik@70-58-41-173.slkc.qwest.net) |
01:02.00 | jhowardPA | First hit's free, gotta pay me for more. |
01:02.06 | Sedorox | lol |
01:02.43 | PBXtech | Stupid T1 cards |
01:03.08 | L|NUX | lol |
01:03.18 | JunK-Y | PBXtech: which card? |
01:03.29 | PBXtech | quad 3.3 thinger |
01:03.58 | PBXtech | doesnt play to nicely with ESI telephone system for some reason |
01:04.37 | L|NUX | support@digium.com |
01:05.01 | PBXtech | yea going to call support at ESI first |
01:06.00 | *** join/#asterisk ManxPower (~eric@adsl-35-236-60.msy.bellsouth.net) |
01:06.10 | ManxPower | ~docs |
01:06.11 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
01:06.13 | ManxPower | ~mailinglist |
01:06.15 | jbot | i heard mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
01:06.42 | PBXtech | just buggin me had it on the v5 card and it was dying once per week, now its on a v3.3 card and its dying daily. time to call ESI :) |
01:06.59 | Sedorox | Does FWD pass CID on? |
01:07.03 | jhowardPA | blitzrage: any ideas? |
01:07.20 | file[laptop] | Sedorox: yup |
01:07.29 | Sedorox | hmmmm |
01:07.33 | blitzrage | jhowardPA: looking |
01:07.39 | riksta | back |
01:07.44 | riksta | looking too |
01:07.44 | jhowardPA | thank you, sir |
01:07.45 | *** join/#asterisk chaoscon (~ph33r@chaoscon.user) |
01:07.59 | riksta | pastbin.ca is so slow |
01:08.05 | jhowardPA | blitzrage: I was afraid my rampant foolery offended you off. |
01:08.07 | jhowardPA | ;) |
01:08.25 | *** join/#asterisk NewSole (~david@i216-58-44-245.avalonworks.net) |
01:08.26 | riksta | wtf |
01:08.30 | riksta | where is the rest of the data |
01:09.02 | riksta | jhowardPA: ? |
01:09.23 | riksta | did you read what i asked for? |
01:09.34 | jhowardPA | riksta: which part do you need? |
01:09.48 | slePP | er.. 'slaps' |
01:09.52 | slePP | not lapes. wtf lapes is, i don't know |
01:10.08 | jhowardPA | riksta: second phone, same as the first, with a diff set of extn numbers. |
01:10.23 | blitzrage | jhowardPA: extensions.conf contexts and Asterisk console output would be handy |
01:10.38 | jhowardPA | lemme fetch that... one sec. |
01:10.51 | riksta | jhowardPA: was asking for the sip suff and extention stuff |
01:11.15 | riksta | no such word slePP :) |
01:11.20 | slePP | damn |
01:11.25 | |Vulture| | whats the best method to sync config files from a centralized server to individual * boxes? |
01:11.32 | |Vulture| | rsync? ssh scripting? |
01:11.33 | riksta | nfs? |
01:11.50 | riksta | or is it remote |
01:11.50 | blitzrage | switch => |
01:12.20 | |Vulture| | riksta: I want to be able to push all the config files from my windows box down to all my linux * boxes |
01:12.45 | riksta | ew :) dunno, i'd probably say rsync, but ask others! |
01:12.48 | Sedorox | CID doesn't get passed through queues?? |
01:12.49 | blitzrage | switch => |
01:12.51 | riksta | slePP: thats just mean |
01:12.55 | slePP | i know :> |
01:12.59 | slePP | i'm bad that way |
01:12.59 | riksta | and, i'm not jewish, fool |
01:13.02 | slePP | how'd that project go? |
01:13.05 | Sedorox | slePP: :-p |
01:13.06 | slePP | skullcap being the top of yer head :P |
01:13.08 | shmaltz | anybody here that has app_valetparking that works with latest CVS HEAD? |
01:13.09 | slePP | Sedorox: 'lo. |
01:13.13 | Sedorox | how be thee? |
01:13.13 | slePP | Sedorox: the 15th |
01:13.15 | blitzrage | pssst... I think switch => would work |
01:13.16 | riksta | i got it in yeah, hopefully shud get a good mark |
01:13.19 | slePP | not bad. bit out of sorts today |
01:13.33 | Sedorox | when you get the thingy installed? |
01:13.37 | slePP | 15th |
01:13.44 | Sedorox | nice |
01:13.51 | slePP | we finally got confirmation today |
01:13.56 | Sedorox | kewl |
01:13.58 | riksta | oohh, that skullcap :) |
01:14.10 | Sedorox | the other place is finally working on the number... I'm thinking about just telling them to forget it and refund me.. hehe |
01:14.18 | slePP | that'd be a good idea :> |
01:14.21 | jhowardPA | riksta: here's my extensions.conf http://pastebin.ca/9512 |
01:14.36 | slePP | i should give a prize to the 10,000th pastebin poster |
01:14.44 | slePP | i should remember to ship the prize for the draw winner, too |
01:14.53 | blitzrage | slePP: need my address right? |
01:14.56 | riksta | you'll be there forever slePP ..it loads too slow ;))))) |
01:14.57 | Sedorox | ahahah |
01:15.08 | slePP | riksta: 8ms for me ;> |
01:15.14 | riksta | bah |
01:15.20 | slePP | blitzrage: heh. you didn't win :P |
01:15.30 | blitzrage | slePP: well... I got the email that said I did |
01:15.38 | riksta | ping times are ok across the pond here |
01:15.42 | slePP | i didn't send an e-mail. i sent a message on MSN :> |
01:15.43 | riksta | but your server responds so slowly |
01:15.48 | riksta | rtt min/avg/max/mdev = 141.862/145.076/190.753/9.387 ms |
01:15.49 | blitzrage | slePP: it said, "YOU BIG WINNAH!" |
01:15.53 | slePP | see, that's weird, riksta. i dunno why.. |
01:16.00 | slePP | blitzrage: y'sure it wasn't for v14gr4? |
01:16.06 | blitzrage | slePP: lol |
01:16.17 | jhowardPA | riksta: the sip.conf is http://pastebin.ca/9513 |
01:16.19 | riksta | slePP: i might try without privoxy, just incase |
01:16.25 | blitzrage | slePP: I was impressed with the use of ASCII art in a spam message today |
01:16.35 | slePP | really? pastebin it :> |
01:16.40 | riksta | jhowardPA: ill have a look |
01:16.50 | PBXtech | asterisk@home has a bug if you call a phone that isnt registered it goes into a loop |
01:17.01 | slePP | riksta: it may possibly be the javascript bit.. i could drop that and see if it changes anything |
01:17.10 | riksta | jhowardPA: are you using any macros currently |
01:17.17 | riksta | slePP: yeah would you like to try? |
01:17.19 | *** join/#asterisk rrk (~chatzilla@rrcs-67-53-9-175.west.biz.rr.com) |
01:17.20 | slePP | k |
01:17.36 | slePP | riksta: try now |
01:17.39 | riksta | firefox just sits "Waiting for pastebin.ca" tho |
01:17.43 | blitzrage | slePP: damn... after I saw it marked as spam and its gone |
01:17.48 | riksta | same thing |
01:17.49 | jhowardPA | I'm using Asterisk@Home, so it's pretty likely. I'm just trying to get my hands dirty with it, then move to a clean setup on debian. |
01:17.51 | slePP | blitzrage: :< |
01:17.58 | riksta | lemme put your pastebin.ca in the list of non proxy hosts |
01:18.01 | Sedorox | riksta: I've had that problem before with .ca |
01:18.05 | jhowardPA | riksta: sorry, that was meant for you |
01:18.08 | slePP | some .ca lookups can be damn slow |
01:18.11 | slePP | for some odd reason |
01:18.12 | blitzrage | pastebin.ca slooooow for me |
01:18.12 | rrk | lot of luck with the clean debian |
01:18.17 | riksta | see |
01:18.20 | riksta | :) |
01:18.21 | jhowardPA | rrk: whaddya mean? |
01:18.25 | blitzrage | slePP: yep, thats basically what I'm seeing |
01:18.28 | slePP | someone run a dnstrace on that |
01:18.39 | slePP | my lookup is 100ms |
01:18.40 | rrk | i tried for three days to move it to debian |
01:18.53 | rrk | with no sucess |
01:18.53 | riksta | jhowardPA: i would try without the macros, first of all |
01:19.07 | jhowardPA | rrk: was it an asterisk versioning problem, or a debian problem? |
01:19.17 | rrk | i assume you are interested in keeping amp |
01:19.29 | riksta | slePP: seems to be faster now |
01:19.29 | jhowardPA | riksta: you got it. I'll see what happens. |
01:19.34 | file[laptop] | ooh la la |
01:19.36 | jhowardPA | rrk: yeah, I'd like to. |
01:19.36 | *** join/#asterisk luke-jr_ (~luke-jr@207.192.219.246) |
01:19.38 | slePP | riksta: i didn't change anything :> |
01:19.40 | jhowardPA | rrk: won't build? |
01:19.49 | riksta | slePP: i just put pastebin.ca in firefox's no proxy list |
01:19.51 | riksta | cuz i use privoxy |
01:19.53 | riksta | lemme take it out |
01:19.55 | riksta | and try again |
01:20.14 | riksta | slooow |
01:20.15 | rrk | problems with spandsp patch for the fax and some of the config files |
01:20.17 | riksta | must be privoxy man |
01:20.31 | Sedorox | the internet here at school has been sucking balls lately |
01:20.31 | riksta | strange |
01:20.48 | riksta | yep it's privoxy, slePP , ill look at its log file |
01:20.51 | slePP | k |
01:20.59 | slePP | maybe revdns? |
01:21.07 | riksta | Apr 14 02:18:02 Privoxy(-1235059792) Request: pastebin.ca/ads/freestyle-1.gif crunch! |
01:21.26 | riksta | Apr 14 02:14:16 Privoxy(-1209631824) Request: config.privoxy.org/send-stylesheet crunch! |
01:21.28 | slePP | wtf is 'crunch'? |
01:21.46 | riksta | it takes out your gay ads |
01:21.51 | riksta | :) |
01:21.54 | slePP | :> |
01:21.57 | slePP | those ads keep the site alive |
01:22.24 | rrk | the zaptem would not build right with the 2.4.27 stable but would with 2.6.10 and 11.7 |
01:22.25 | Sedorox | does anyone know if queue's don't pass on CID... seems when I call in.. and into the queue.. the agent doesn't get the cid info |
01:22.46 | jhowardPA | riksta: I can't see how to del those macros without basically moving to a fresh install of asterisk. Is that the goal? |
01:22.47 | zilas | how can I eliminate this gap when I call voicemail from sip phone and before I hear something * server already starts talking so I dont hear like first 5 seconds |
01:23.22 | zilas | no maybe 2 seconds |
01:23.22 | riksta | jhowardPA: just get the incoming context to Dial() the phone |
01:23.28 | riksta | directly, for testing |
01:23.49 | PTG123 | <PROTECTED> |
01:23.58 | Sedorox | lol |
01:23.58 | PTG123 | damn space :) |
01:24.07 | jhowardPA | riksta: I said I was quick, but I also said I was new to Asterisk. Could you elaborate a bit for me? ;) |
01:24.13 | Sedorox | PTG123: if you can take paypal for one server.. I'm willing to talk :-p |
01:24.25 | jhowardPA | Sorry, I don't mean to be a goon, but I've had insufficient study. |
01:24.30 | PTG123 | hah paypal is prefered :) |
01:24.36 | PTG123 | msg me :) |
01:24.37 | Sedorox | can I pm ya? |
01:24.38 | Sedorox | aha |
01:24.46 | riksta | jhowardPA: one second let me read the config a bit more |
01:24.47 | PTG123 | and you better hurry people are emailing me like mad on the list :) |
01:24.48 | PTG123 | yah pm me |
01:24.53 | Sedorox | ahah |
01:25.07 | jhowardPA | riksta: Thanks :) |
01:25.59 | L|NUX | slePP : can you tell me how can i create md5 password for * |
01:26.12 | file[laptop] | PTG123 is gonna be rich |
01:26.16 | Sedorox | lol |
01:26.34 | PTG123 | hah |
01:26.43 | *** join/#asterisk ta[i]nted (~tainted@adsl-69-108-114-226.dsl.irvnca.pacbell.net) |
01:27.11 | Sedorox | I need a decent system to either colo (eventually...) or just to run for * |
01:27.26 | file[laptop] | I run asterisk on a wide variety of things really |
01:27.36 | riksta | L|NUX: can you not just use perl's crypt() ? |
01:27.47 | JunK-Y | linux: i told u to use the application. |
01:27.48 | file[laptop] | Celerons, Xeons, AMDs, Geodes |
01:28.05 | L|NUX | JunK-Y : but there is not listed MD5 :( |
01:28.11 | L|NUX | in asterisk application |
01:28.35 | JunK-Y | use head |
01:28.40 | L|NUX | i used |
01:28.43 | L|NUX | but same |
01:28.48 | *** join/#asterisk bah (048830696@AC8C1316.ipt.aol.com) |
01:28.49 | L|NUX | recompiled |
01:28.51 | L|NUX | but same problem |
01:28.51 | JunK-Y | huh? |
01:28.54 | L|NUX | really |
01:28.54 | L|NUX | wait |
01:29.03 | Sedorox | hehe.. right now I have it on a celery 333, which doesn't have a working CPU fan.. ahah |
01:29.08 | L|NUX | http://digium-cvs.netmonks.ca/viewcvs.cgi/asterisk/md5.c?rev=1.11&view=auto |
01:29.13 | JunK-Y | debian*CLI> show applications like md5 |
01:29.13 | JunK-Y | <PROTECTED> |
01:29.13 | JunK-Y | <PROTECTED> |
01:29.13 | JunK-Y | <PROTECTED> |
01:29.13 | JunK-Y | <PROTECTED> |
01:29.18 | Sedorox | I'm asking for a death sentance :-p |
01:29.32 | JunK-Y | is that what u want? |
01:29.43 | niZon | Sedorox: do you smell burning yet? :P |
01:30.00 | riksta | perl -e 'print crypt("password", "salt"),"\n"' ? |
01:30.07 | L|NUX | wait |
01:30.14 | riksta | is that md5? |
01:30.18 | Sedorox | nope :-p |
01:30.24 | L|NUX | JunK-Y : but not showing in my * |
01:30.28 | L|NUX | i recompiled |
01:30.50 | riksta | ahh ok, this is md5 http://sial.org/howto/perl/password-crypt/ |
01:30.57 | JunK-Y | ya've a app_md5.c in ur /usr/src/asterisk/apps/ ? |
01:31.10 | drumkilla | it's only in cve head |
01:31.12 | drumkilla | cvs* |
01:31.24 | JunK-Y | yes, i already told him that. |
01:31.28 | L|NUX | hmm |
01:31.29 | L|NUX | wait |
01:31.37 | L|NUX | can you give me a link ? |
01:31.41 | JunK-Y | ls -l /usr/src/asterisk/apps/app_md5.c |
01:31.52 | JunK-Y | type that in ur shell |
01:31.56 | L|NUX | k |
01:32.05 | L|NUX | [root@NuFW asterisk]# ls -l apps/app_md5.c |
01:32.05 | L|NUX | ls: apps/app_md5.c: No such file or directory |
01:32.07 | L|NUX | O_o |
01:32.14 | L|NUX | i was doing with wrong file :$ |
01:32.15 | L|NUX | shit |
01:32.15 | L|NUX | wait |
01:37.50 | harryvv | you would be better to just wait and not type anything that does not fill up the window..one ..line ...at ...a ..time. |
01:38.17 | riksta | slePP: that privoxy crap is weirdddd |
01:38.30 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
01:38.30 | *** mode/#asterisk [+o bkw_] by ChanServ |
01:39.14 | riksta | beautiful hostname bkw_ ... lol |
01:41.13 | |Vulture| | is there a way to write an SSH script that will log into a server and execute commands, such as put and say restart *? |
01:41.49 | JunK-Y | rsh ? |
01:42.02 | JunK-Y | that's why rsh (remote shell) is there. |
01:42.16 | |Vulture| | hmmm |
01:42.32 | dec | or you can setup SSH authorized keys and just do 'ssh hostname sudo /etc/init.d/asterisk restart' |
01:42.53 | |Vulture| | never used that |
01:43.14 | |Vulture| | I want to run these scripts from within windows... |
01:43.28 | |Vulture| | I tried using putty's pscp |
01:43.44 | |Vulture| | it sends the files fine, but duno if it will let me execute commands |
01:43.56 | malbech | Anyone konws a SoftSwitch Solution for a low cost ? |
01:44.09 | mmlj4 | pscp is putty's scp client, useful mainly for sending files, as noted |
01:44.21 | *** join/#asterisk mog_home (~mog_home@146.229.178.196) |
01:44.25 | mmlj4 | putty itself can let you log in and run commands |
01:44.45 | mmlj4 | you might be able to do something from windows itself if you install cygwin |
01:44.47 | |Vulture| | mmlj4: but I am trying to automated it all so I just have to click a batch file etc. and it will all do it |
01:45.00 | mmlj4 | yeah, cywin.com, go install that |
01:45.18 | *** part/#asterisk jhowardPA (~jhoward@12.25.177.120) |
01:45.42 | mmlj4 | your "batch files" are going to be regular shell scripts... so you can do 'ssh hostname sudo /etc/init.d/asterisk restart' by clicking an icon, for example |
01:46.08 | |Vulture| | okay |
01:46.17 | mmlj4 | hint; do a default cygwin install, then run the setup again to add ssh, etc. |
01:46.30 | mmlj4 | what country are you in, |Vulture| ? |
01:46.38 | |Vulture| | USA |
01:46.57 | mmlj4 | ok... choose either the nasa or kernel.org ftp sites, to do the cygwin install |
01:49.46 | |Vulture| | okay all done going to try and get ssh now |
01:51.08 | *** join/#asterisk Dovid (~hirisk@pool-138-89-170-224.mad.east.verizon.net) |
01:51.19 | harryvv | is iax.cc down? |
01:51.26 | harryvv | It shows me as not registered |
01:53.13 | niZon | make a call |
01:54.17 | harryvv | i did |
01:54.36 | harryvv | said == Everyone is busy/congested at this time (1:0/1/0) |
01:54.42 | niZon | weird |
01:54.49 | harryvv | did a iax2 show registry and shows me as not registered |
01:55.06 | harryvv | this is of course not the first time thay have had problems. |
01:55.12 | Dovid | hi |
01:55.20 | niZon | they're slow at getting DIDs |
01:55.33 | *** join/#asterisk tessier (~treed@203.210.216.187) |
01:55.35 | harryvv | Im just using there outbound pstn |
01:55.48 | Dovid | i have x-lite. what ports do i need to open on my firewall at my home to access it ? |
01:56.20 | harryvv | doe! Apr 13 18:53:04 NOTICE[7098]: chan_iax2.c:7090 iax2_poke_noanswer: Peer 'sixtel' is now UNREACHABLE! Time: 2892 |
01:56.50 | niZon | harryvv: drop them an IM |
01:56.55 | harryvv | :) |
01:56.55 | L|NUX | Dovid : 5060, 9999-20001 |
01:57.11 | harryvv | noZon, what is there IM |
01:57.16 | Hogie | bleh blah |
01:57.44 | niZon | harryvv: sixtel9 on aim, and msn@sixtel.net for msn (I think) |
01:57.59 | Dovid | L|nux: Anyway that i can set it work with other ports ? |
01:58.28 | L|NUX | hmm |
01:58.37 | L|NUX | think so |
01:58.38 | L|NUX | you can |
01:58.50 | Dovid | how ? |
01:58.57 | L|NUX | <PROTECTED> |
01:59.03 | L|NUX | and sip.conf |
01:59.50 | harryvv | - Registered IAX2 to '205.234.133.203', who sees us as 24.81.64.126:4569 |
01:59.59 | harryvv | looks like thay are now up |
02:00.23 | ta[i]nted | harryvv how is their service |
02:00.46 | ta[i]nted | better or worse than BV |
02:01.01 | harryvv | actually im not registered |
02:01.20 | Dovid | L|nux: Thanks |
02:01.32 | L|NUX | Dovid : Welcome |
02:04.55 | *** join/#asterisk tawker (tawker@d154-5-108-14.bchsia.telus.net) |
02:04.59 | tawker | hi |
02:05.23 | L|NUX | hey |
02:05.59 | *** join/#asterisk bugbot (~bugbot@d141-234-145.home.cgocable.net) |
02:06.03 | tawker | i was looking in the doc project |
02:06.05 | blitzrage | wb bugbot |
02:06.16 | tawker | and I can't seem to find anything close to a voicemail map |
02:07.31 | tawker | i've also gone through voip-info |
02:09.52 | tawker | any ideas |
02:11.13 | *** part/#asterisk lilneon (~tj_r3@cuscon12298.tstt.net.tt) |
02:11.18 | tawker | hi |
02:11.23 | *** part/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
02:11.55 | tawker | ok, maybe this is a bad time |
02:11.59 | tawker | i'll try again later |
02:12.00 | ta[i]nted | yea |
02:12.07 | ta[i]nted | it's rather idle bout now |
02:12.15 | tawker | yes, for the number of people in here |
02:12.21 | tawker | i'd expect it to be a lot more lively |
02:12.26 | ta[i]nted | it's weird |
02:12.39 | ta[i]nted | i think most of them are pulling their hair out configuring stuff |
02:13.18 | tawker | ok, anyways, i'll scram for a bit |
02:13.24 | tawker | talk to you later |
02:13.31 | ta[i]nted | good times |
02:13.42 | Qwell | Have you guys ever seen a problem with an Avaya PBX (only when you call TO an 8xx DID) giving a congestion signal if you call FROM an 8xx DID? |
02:14.24 | ta[i]nted | carrier issue? |
02:14.27 | Qwell | If I change my CIDNum to a non 8xx number, it works great |
02:14.40 | Qwell | its kinda weird... |
02:14.49 | ta[i]nted | u sure its pbx |
02:15.01 | ta[i]nted | could be some kind of call blocking |
02:15.18 | Qwell | yeah, I can call the direct number to people in there, and if I spoof CID, it works fine |
02:15.25 | ta[i]nted | is ORIG 8xx PRI? |
02:15.34 | Qwell | its from nufone |
02:15.40 | ta[i]nted | lol |
02:15.43 | ta[i]nted | case closed |
02:15.50 | Qwell | Its not a nufone problem. |
02:15.57 | Qwell | I can call other 8xx DIDs fine, spoofed or not |
02:16.00 | *** join/#asterisk iq (~IQ@70-57-182-73.omah.qwest.net) |
02:16.07 | iq | hi |
02:16.15 | ta[i]nted | probably carrier issue |
02:16.20 | ta[i]nted | how are u spoofing CID |
02:16.26 | Qwell | SetCIDNum, heh |
02:16.33 | ta[i]nted | what kind of line |
02:16.34 | ta[i]nted | PRI? |
02:16.35 | harryvv | looks like one of my work places got a suspected anthrax letter that was opened up. |
02:16.47 | Qwell | ta[i]nted: the number I'm calling? |
02:16.51 | ta[i]nted | no |
02:16.57 | ta[i]nted | the line u are spoofing from |
02:17.06 | Qwell | ta[i]nted: my nufone account |
02:17.13 | Qwell | maybe "spoofing" is the wrong word? |
02:17.16 | harryvv | <PROTECTED> |
02:17.16 | harryvv | Apr 13 19:13:55 NOTICE[7098]: chan_iax2.c:7090 iax2_poke_noanswer: Peer 'sixtel' is now UNREACHABLE! Time: 729 |
02:17.23 | harryvv | tonights not the nite to use iax.cc |
02:17.26 | ta[i]nted | u can set callerid to whatever u want with nufone? |
02:17.29 | Qwell | ta[i]nted: yeah |
02:17.36 | ta[i]nted | what a bunch of jackasses |
02:17.57 | sivana | I don't think the accountcode entry works from iax.conf.. is that right? |
02:18.00 | *** join/#asterisk alegh (~ag10@OL217-17.fibertel.com.ar) |
02:18.07 | Qwell | ta[i]nted: never seen such an issue though? |
02:18.13 | ta[i]nted | Qwell i have. |
02:18.22 | ta[i]nted | but it was carrier issue (XO specifically) |
02:18.46 | ta[i]nted | tell nufone u have this issue |
02:18.49 | Qwell | is it possible that Avaya would be sending back a fuckoff signal, just because of my CID? |
02:18.52 | ta[i]nted | they can look into their profider |
02:18.56 | *** join/#asterisk Treemole (Treemole@evvlinlwt-nas-07-s126.cinergycom.net) |
02:19.46 | ta[i]nted | try spoofing the 8xx from you nufone line |
02:19.53 | Qwell | the one I'm calling? |
02:19.55 | ta[i]nted | and calling the avaya box |
02:19.57 | Qwell | hmm |
02:20.10 | Qwell | like, SetCIDNum(${EXTEN}), ? |
02:20.29 | sivana | I don't think Nufone accepts CIDNum for outbound toll-free |
02:20.50 | sivana | sorry.. toll-free CIDnum |
02:20.53 | Qwell | well, it works if I change 800 to 951 |
02:21.05 | sivana | you can't have toll-free CIDNum |
02:21.24 | Qwell | Then what am I gonna use? heh |
02:21.28 | ta[i]nted | Qwell are other 8xx able to call the avaya pbx |
02:21.46 | Qwell | ta[i]nted: not sure. lemme try changing it to its own number |
02:21.55 | ta[i]nted | i think u need to try all the possibilities |
02:22.05 | niZon | iax.cc let me set my cidnum to 1337 :P |
02:22.33 | Qwell | heh, I confused the remote provider, or something |
02:22.39 | Qwell | "We're sorry, your call could not go through." |
02:23.01 | Qwell | oops, actually... |
02:23.20 | Qwell | yeah, fast busy still |
02:23.22 | *** join/#asterisk jbAU (~johnblade@61.8.110.41) |
02:23.46 | jbAU | Can anyone recommend a USA IAX service for PSTN calls ? |
02:23.57 | Qwell | jbAU: nufone is great |
02:24.48 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
02:24.51 | shido6 | boink |
02:25.00 | Qwell | shido6: ;] |
02:25.11 | Qwell | shido6: Any idea about the above? |
02:25.17 | jbAU | Qwell: hmm can't seem to find the url for nuphone |
02:25.25 | Qwell | jbAU: Fone...nufone.net |
02:25.37 | Qwell | says they aren't accepting orders...but talk to my man shido here |
02:25.56 | Qwell | oh, hmm, missed the join...right |
02:26.01 | jbAU | ahh ok |
02:26.16 | jbAU | i wonder if i click the 'let me in' button :) |
02:26.24 | Qwell | jbAU: Thats how customers login |
02:26.56 | niZon | shido6 = nufone employee? |
02:27.28 | Qwell | jbAU: basically, 2c/minute outgoing, and you can either get a MI DID for $8-9/month(I always forget), or a us48 tollfree, for 2c/min incoming |
02:27.43 | Qwell | niZon: something like that :p |
02:27.58 | niZon | ok :P |
02:28.06 | jbAU | Qwell: that's pretty damn schweet |
02:28.12 | Qwell | jbAU: yeah, nufone is great |
02:28.42 | jbAU | Qwell: excellent... now if only they were available for business. :) |
02:29.05 | Qwell | jbAU: if shido here doesn't respond soon (that might've been sent automatically when he joined), shoot an email to greg@nufone.net |
02:29.10 | Qwell | He should be able to care of you |
02:29.24 | Qwell | tell him Qwell sent you... |
02:29.25 | jbAU | excellent - i'll send him a message |
02:29.44 | jbAU | sounds fair. :) |
02:30.18 | Qwell | jbAU: I think he said that if people can use paypal for the first order, he can start an account for them |
02:30.34 | jbAU | Qwell: ahh ok - that shouldn't be a problem |
02:30.41 | Qwell | yeah...it rarely is |
02:31.01 | jbAU | Qwell: i don't expect to have too much use at the moment, as i don't have -that- much business in the states |
02:31.19 | Qwell | jbAU: well, nufone also has good international rates |
02:31.27 | tzanger | yeah nufone is not taking new customers at the moment but if you msg shido6 he may be bale to set you up manually if you've got a paypal acct |
02:31.45 | *** join/#asterisk Dovid (~hirisk@pool-138-89-169-188.mad.east.verizon.net) |
02:31.53 | jbAU | shido6: can you help me out with a nufone account please ? |
02:32.03 | jbAU | shido6: qwell sent me :) |
02:32.04 | Dovid | anyone know any sip phone that u can specify what ports u want it to use ? |
02:32.10 | Qwell | jbAU: might be better off sending him an email |
02:32.16 | jbAU | Qwell: indeed |
02:35.35 | dec | is there some sort of no-audio timeout on IAX? on an IAX connection i'm testing, there's only audio going one way... and the connection keeps dropping out |
02:37.42 | *** join/#asterisk MrBelvedr (~tt@ip68-227-209-110.dc.dc.cox.net) |
02:38.46 | *** join/#asterisk riksta (~rick@81-178-248-131.dsl.pipex.com) |
02:39.05 | Dovid | anyone know of a sip phone that i can set what ports it should use ? |
02:39.55 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
02:41.38 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
02:42.01 | file[laptop] | ...achoo |
02:42.23 | Qwell | file[laptop]: cover your mouth |
02:42.29 | shmaltz | anybody here has problems with quad span interrupts? |
02:42.31 | file[laptop] | but then you won't get infected |
02:43.11 | Qwell | file[laptop]: Make my work fix their PBX, would ya? |
02:43.26 | file[laptop] | Qwell: what? |
02:43.29 | *** part/#asterisk Pinhole (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com) |
02:43.33 | Qwell | dunno, its b0rked |
02:43.39 | file[laptop] | that made absolutely no sense |
02:43.45 | Qwell | If my CIDNum is 8xx, I get fast busy. Otherwise, its fine |
02:44.07 | Qwell | and its only the 800 DIDs at my work...if I direct dial to somebody, it works fine |
02:44.27 | file[laptop] | oh depending on the provider and stuff if you set your caller id number to a toll-free number it may or may not work |
02:44.39 | Qwell | meh...thats my only phone number. heh |
02:44.40 | file[laptop] | the provider where the number you're calling... |
02:44.57 | file[laptop] | it's just the way it is, I've found it out through my own... testing and junk |
02:45.13 | Qwell | file[laptop]: but...the npanxx numbers at my work are off the same provider |
02:45.35 | file[laptop] | usually when you call toll-free numbers that it happens too... |
02:45.45 | Qwell | ahh |
02:45.48 | Qwell | well, hell...heh |
02:45.54 | Hogie | is digium in SC? |
02:46.00 | file[laptop] | Hogie: Huntsville, Alabama |
02:46.06 | *** join/#asterisk PBXtech (~nik@70-58-41-173.slkc.qwest.net) |
02:46.12 | Qwell | I'm gonna have to spoof my CIDNum then...Simpsons 939 number, here I come |
02:46.21 | file[laptop] | Qwell: tee he he |
02:47.23 | Hogie | Im trying to figure out where my vendor had my card drop shipped from |
02:48.14 | *** join/#asterisk mountie (~mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com) |
02:49.18 | robl^ | Hogie, India ;) |
02:49.38 | Hogie | lol, its coming from SC |
02:49.38 | MrBelvedr | i am trying to test the performance on my system. Can somebody tell me a process that I can run that will consume all my ram and force the system to start swapping? |
02:50.05 | MrBelvedr | i only have 64 mb of ram so it should not be hard |
02:50.50 | robl^ | MrBelvedr, Windows :) |
02:50.51 | harryvv | anyone here make there extensons failover from sip/iax service to zap? I have tried it before and did not seem to want to advanced to the next zap priority and make the call. Anyone done this before? seems simple to setup but only if it works. |
02:51.38 | harryvv | sixtel is now up! |
02:51.55 | tzanger | harryvv: sure |
02:52.10 | tzanger | if you get CHANUNAVAIL or CONGESTION, dial out the zap |
02:52.58 | Hogie | harryvv: N+1 on the priority of the extension after a Dial has worked for me |
02:53.39 | harryvv | tzanger thats the idea |
02:53.56 | tzanger | it's not just the idea, it's how it's done :-) |
02:54.26 | harryvv | :) its more of a wife bitching thing :) |
02:54.27 | harryvv | heheh |
02:55.48 | tzanger | harryvv: http://pastebin.ca/9517 |
02:57.15 | PBXtech | anyone have one of those pulver wifi phones? |
02:57.15 | Hogie | tzanger: does that only work on cvs, not stable? Ive not seen the -CANCEL stuff and such so far |
02:58.01 | tzanger | you need to use extension priority numbers rather than n but it should work otherwise |
02:58.19 | tzanger | Hogie: if you really needed to you could use GotoIf()s to work around it but I think it works just fine on standard |
02:58.22 | tzanger | PBXtech: yeah they suck |
02:59.06 | Hogie | any of the wifi phones good tzanger? or is it too early? |
02:59.19 | tzanger | Hogie: I've only tried the pulver one but the others look like ass |
02:59.30 | Nugget | the pulver/zyxel ones are horrid. |
02:59.32 | tzanger | I have heard of one by an egyptian company and another from mexico that look better |
02:59.37 | tzanger | and might work better |
02:59.40 | Nugget | vaewyn has the hitachi one and seems to really like it |
02:59.40 | docelmo | Anyone in here from the NYC or 2 hour north area? |
02:59.57 | tzanger | I'm ~2.5hrs from NY but not NYC |
02:59.58 | mishehu | "talk like an egyptian..." |
03:00.02 | tzanger | hehe |
03:00.07 | Hogie | Im trying to find one for work... Im never at my desk, and that would be nice with the new phone system |
03:00.19 | PBXtech | whats a good wimax phone then? |
03:00.38 | tzanger | uh PBXtech we just said we don't really know of any |
03:00.39 | *** join/#asterisk bugbot (~bugbot@d141-234-145.home.cgocable.net) |
03:00.53 | PBXtech | oh |
03:01.24 | *** join/#asterisk ScythelX (Fleb@pc-24-181-176-181.sbi.ct.charter.com) |
03:02.32 | *** join/#asterisk Poemius (~poemius@adsl-70-48-192-81.adsl.iam.net.ma) |
03:02.38 | Poemius | howdy everyone :) |
03:02.52 | ScythelX | hello all - can someone take a look at this - im having a problem with incoming calls only - its very choppy but outgoing calls work fine - i have a cisco 79xx phone and using nufone as my upstream provider- i have a cable modem with an upload speed of 384 / http://pastebin.ca/9380 |
03:03.36 | docelmo | ScythelX, whats your network topology look like? |
03:03.52 | Poemius | ScythelX : not sure if it is an issue... on some configs, X causes choppiness, try turning off X |
03:04.01 | ScythelX | not using X |
03:04.16 | ScythelX | the * box is at my house where the cable modem is and the cisco phone is located at my friends house |
03:04.20 | ScythelX | whom also has a cable modem |
03:04.50 | Poemius | shouldn't be an issue, maybe latency |
03:04.51 | docelmo | So your cable -> router -> *? |
03:04.59 | *** join/#asterisk libpcp (libpcp@210.16.20.5) |
03:05.31 | ScythelX | outgoing call -> cisco phone -> router -> my house ( * box ) -> nufone to pstn |
03:05.43 | ScythelX | the * box is directly connected to the internet |
03:05.45 | ScythelX | no router |
03:06.23 | ScythelX | i meant incoming call above |
03:06.34 | ScythelX | outgoing calls are crystal clear |
03:07.06 | docelmo | What codec, dial plan are you using? Do you have a default conext setup/ |
03:07.24 | ScythelX | did you look at that pastebin |
03:07.28 | docelmo | yes |
03:07.38 | ScythelX | its supposed to be using gsm |
03:07.45 | ScythelX | but then again the cisco phones dont support gsm |
03:08.27 | Nugget | yeah, you might really want to consider buying a couple g729 licenses. |
03:08.28 | docelmo | Change your codecs to g729 or ULAW that may fix the problem. Is it transcoding? |
03:08.34 | Moonwick | for what they cost, you'd expect them to be able to display full motion video on those ginormous LCDs |
03:09.39 | docelmo | true.. |
03:09.41 | Poemius | or maybe do the dishes automatically :) |
03:10.06 | docelmo | 10 buks.. I just told my company I needed to order 100 and they about died when I told them the cost |
03:11.46 | Poemius | well it depends if it is intra... ilbc/ speex might do the trick |
03:12.31 | Poemius | or you may not need 100, it is the number of active lines at one time |
03:12.45 | docelmo | I dunno.. My network is about to go live and I am doing g729 only well for the most part.. I have to terminate to my down stream provider at ULAW.. |
03:13.02 | docelmo | I run at any given point in time about 500ish |
03:13.15 | ScythelX | well right now hte phone is set to use g711ulaw |
03:13.25 | docelmo | Set * to do the same |
03:13.31 | Poemius | :) much bigger than my tiny weeny setup :)... but I guess it makes sense :) |
03:13.40 | ScythelX | hmm ok |
03:13.50 | docelmo | with NuFone |
03:13.56 | ScythelX | the config in ast is g711ulaw |
03:13.57 | harryvv | just tested my call its not failing over |
03:13.57 | Nugget | you're not allowed to call it a "tiny weeny" setup unless your asterisk server is a mac mini. :) |
03:14.01 | ScythelX | or just g711 |
03:14.11 | docelmo | g711 |
03:14.13 | Poemius | or a linksys :) |
03:14.14 | docelmo | or ULAW |
03:14.28 | docelmo | Well Im running Dual 248 Opterons w/ 4GB ram |
03:14.30 | ScythelX | does it matter which one |
03:14.34 | ScythelX | g711 or ULAW |
03:14.39 | docelmo | They are the same |
03:14.41 | ScythelX | okie |
03:14.59 | PatrickDK | they are almost the same |
03:15.04 | PatrickDK | g711 can also be alaw |
03:15.10 | docelmo | This is true |
03:15.18 | docelmo | Which is a fricken Euro standard.. :( |
03:15.39 | PatrickDK | alaw is suppost o give better sound quality |
03:15.44 | Poemius | but compared to the price of a setup this size though, the price of the licences should not be that much |
03:15.48 | PatrickDK | personally though, I do perfer the metric sytem |
03:15.50 | PatrickDK | system |
03:16.01 | docelmo | I run ALAW from Australia and it doesnt sound any better than the ULAW I use up here |
03:16.13 | Poemius | it's true a decimal system makes a lot of things easier |
03:16.30 | PatrickDK | docelmo, suppost to sound better, in mathmatical terms it does |
03:16.34 | docelmo | Anyone know of any good A-Z providers? |
03:16.39 | docelmo | heheh :) |
03:16.40 | PatrickDK | in actually use, well, you never know how that works out |
03:16.45 | Qwell | docelmo: A-Z? |
03:16.53 | Poemius | termination |
03:16.55 | docelmo | all international countries |
03:17.09 | Qwell | nufone doesn't? |
03:17.09 | docelmo | I have domestic.. Just bought 2.5 DS3's.. |
03:17.16 | docelmo | I dont like nufone |
03:17.20 | Poemius | at my size voipjet seems to be the cheapest |
03:17.27 | Qwell | Whats wrong with nufone? |
03:17.32 | docelmo | I pled the 5th |
03:17.51 | Poemius | I haven't tried them, but voipjet looks cool |
03:17.59 | Qwell | docelmo: nufone is great |
03:18.01 | docelmo | Poemius, gimme a month or so.. You will see things MUCH cheaper.. |
03:18.02 | Poemius | at least from a cost point of view |
03:18.37 | Poemius | :) docelmo, I am definitely interested :) |
03:19.18 | docelmo | .02c for Domestic 48? Wicked expensive.. I was looking at charging around 1c or less depending on volume |
03:19.38 | docelmo | And I will have origination in 15 markets about 300 rate centers |
03:19.40 | Poemius | dicelmo I think voipjet is at 1.4 |
03:20.02 | docelmo | ya.. I use them now until my network is finished |
03:20.14 | docelmo | Im building at 60 hudson right now |
03:20.49 | Poemius | sounds very interesting :) |
03:21.02 | docelmo | Poemius where do you live? |
03:21.11 | Poemius | :) Casablanca, Morocco |
03:21.23 | Poemius | I reckon, it's a bit late here (3 am) |
03:21.41 | docelmo | ohh |
03:21.55 | docelmo | No DID's there YET.. But I have 750 Rate Centers |
03:22.04 | docelmo | In the states |
03:22.16 | Poemius | lol :) I do need a DID in CT |
03:22.19 | mishehu | if only I could afford a single ds3... |
03:22.24 | Poemius | I currently use broadvoice |
03:22.42 | niZon | docelmo: get some in canada :D |
03:22.44 | Poemius | but it's not flexible enough in my own taste |
03:23.34 | Poemius | not really sure, but I get the impression it is not as crowded in canada |
03:23.42 | Poemius | (voip competition wise) |
03:23.42 | docelmo | Closest I have to CT is MASS |
03:24.20 | docelmo | niZon, CA is next on my list. Working some deals up there. Trying to find a provider that will do SS7 and not rape me |
03:24.30 | niZon | lol cool |
03:24.51 | Poemius | :) that's one of the downsides of no competition |
03:25.16 | docelmo | DID's are gonna run about 9.50 for unlimited.. Once I build up the network things will get MUCH cheaper.. |
03:25.20 | Poemius | here in morocco, no competition, they charge 30 cents a minute to europe or US |
03:25.39 | ScythelX | docelmo: who are you leasing your ds3 from |
03:25.47 | docelmo | Me to know.. :) |
03:25.50 | niZon | docelmo: sounds good |
03:25.57 | ScythelX | or is it in a telco hotel |
03:26.04 | docelmo | Hotel.. 60 Hudson |
03:26.05 | Poemius | docelmo: 9.50 for incoming is about twice as most |
03:26.10 | ScythelX | figured |
03:26.18 | docelmo | Peo, unlimited? |
03:26.25 | docelmo | Not metered |
03:26.40 | docelmo | Voice Pulse is 11 something |
03:26.42 | Poemius | ah ok... it's true it's more interesting for big guys |
03:27.03 | docelmo | AND I offer LNP |
03:27.13 | docelmo | Along with Caller ID Name |
03:27.17 | ScythelX | xo communications is the cheapest i think in 60 hudson |
03:27.26 | Poemius | definitely interesting |
03:27.29 | ScythelX | although they suck balls |
03:27.42 | Poemius | although u should have an option later for standard users |
03:27.51 | docelmo | the company I work for now uses them.. I have no complaints.. They do what I want when I need it. |
03:28.23 | docelmo | Poe once I am established prices will change.. Right now I need to cover costs.. |
03:28.25 | Poemius | it definitely sounds like a big and great project |
03:29.01 | docelmo | And the 1st DID is gonne be around 9.50 the additional's will only be $2.50 more |
03:29.07 | Poemius | agreed... it's a good idea... plus if you target big companies that yet don't have voip, |
03:29.34 | Poemius | ah ok, definitely more interesting with that scheme |
03:29.43 | MrBelvedr | whenver I startup asterisk it sometimes gets to the CLI, sometimes it does not. but it always says 'Killed.' Why is this happening? |
03:29.47 | ScythelX | most big companies would have their own data lines |
03:30.06 | jbAU | MrBelvedr: try starting up with asterisk -vvvgc to see if there's an error |
03:30.13 | MrBelvedr | k |
03:30.26 | Poemius | :) ScythelX, not talking about fortune 500 |
03:30.27 | ScythelX | ie: american eagle |
03:30.32 | ScythelX | uses all voip |
03:30.57 | docelmo | Im targeting larger VOIP carriers as clients for termination or origination and small med business along with "bring your own gear" type of accounts and of course the end users |
03:31.53 | Poemius | docelmo: definitely with multiple dids, your offer becomes much more interesting |
03:33.00 | docelmo | Im going to be competitive in the market.. Like I said.. I am targeting bigger clients like Million Minutes a month + |
03:33.12 | docelmo | But for the smaller guys a good service also |
03:33.24 | ScythelX | do you have any clients |
03:33.28 | *** join/#asterisk jakepdev (~jakepdev@pool-68-236-58-19.phil.east.verizon.net) |
03:33.34 | ScythelX | or are you just shelling out money for your ds3 |
03:33.54 | docelmo | I have quite a few. I am not paying for anything yet. Still waiting to move into my cage |
03:34.38 | jakepdev | hello everyone |
03:35.08 | Poemius | :) as we all know, there is a large potential :)... the key is targeting things right :) |
03:35.47 | docelmo | Well I am using * as my backend highly modified |
03:35.54 | Poemius | and from what u say, it sounds like a very nicely thought plan |
03:36.01 | docelmo | so I will support IAX/SIP/H323 |
03:36.20 | docelmo | I have been engineering this network for about 8 months and its just now coming to light |
03:36.38 | *** part/#asterisk Treemole (Treemole@evvlinlwt-nas-07-s126.cinergycom.net) |
03:37.10 | Poemius | it seems like you've done ur homework well, I wish you good luck, and maybe get bigger than vonage :) |
03:37.24 | docelmo | Vonage is being bought |
03:37.29 | docelmo | so I am not worried abou tthem |
03:37.32 | *** join/#asterisk lotku (~hadme@210.213.173.53) |
03:37.52 | Poemius | well if u get the size of vonage and get bought :) |
03:37.54 | docelmo | besides there is enough of a market for VOIP and I am not going to target the same clients as vonage |
03:38.00 | Poemius | I wouldn't worry much :) |
03:38.08 | Poemius | retire on a beach and stuff :) |
03:38.21 | docelmo | I live in Tampa FL.. Im already on the beach.. :) |
03:38.39 | docelmo | Anyone going to Astricon in Atlanta in September? |
03:39.01 | Poemius | wave on the beach next time, I'm right across the ocean :) |
03:39.42 | docelmo | Im in the Gulf.. |
03:39.53 | docelmo | Its in october now? Geesh |
03:40.12 | Poemius | ah ok :) I thought u were in the atlantic |
03:40.26 | Poemius | I was gonna say... you dive in the atlantic, go towards south east, avoid a couple of hungry sharks...and boom, you're in casablanca |
03:40.34 | *** join/#asterisk odie_flocon_ (~chatzilla@S01060011953994ee.cg.shawcable.net) |
03:40.44 | odie_flocon_ | hello all. |
03:40.51 | Poemius | hi odie :) |
03:41.12 | odie_flocon_ | how goes it Poe |
03:41.28 | *** join/#asterisk dash (washort@68.212.221.181) |
03:42.48 | docelmo | There's a bootcamp in Tampa.. WOO HOO! 3000 buks tho? sigh |
03:43.27 | dash | Hi. I'm trying to add some remote users to my Asterisk setup at an office; I need to be able to transfer calls to them, and have them transfer calls, etc. If I set them up with an IAXy, will this be convenient? |
03:43.50 | dash | i don't know much about the user interface to the iaxy |
03:46.17 | docelmo | Um, just use Flash and type the extension |
03:46.20 | docelmo | Thats about it. |
03:46.28 | *** join/#asterisk Poemius_ (~poemius@adsl-70-48-192-81.adsl.iam.net.ma) |
03:46.31 | dash | okay :) |
03:46.39 | Poemius_ | @^#~{@~^{#~@{~# battery on my laptop |
03:46.45 | dash | docelmo: is that attended transfer? |
03:46.59 | Poemius_ | (message censored for sensitivity purpose) |
03:47.00 | Poemius_ | :) |
03:47.14 | dec | is there a way to find out what version of asterisk is installed on a server without being able to connect to the console? :P |
03:47.33 | dash | dec: call someone who is able to connect to the console? ;) |
03:47.41 | dec | no one can :) |
03:47.50 | dec | i have ssh and full root access |
03:47.50 | dec | but |
03:48.01 | dec | asterisk -cr won't connect because its not listening on 127.0.0.1 |
03:48.11 | docelmo | hehe.. Kill the binary |
03:48.15 | Poemius | not sure if there is a -v param |
03:48.19 | docelmo | or check the source directory |
03:48.24 | Poemius | asterisk -v? |
03:48.30 | docelmo | dash yes |
03:48.32 | docelmo | should be |
03:48.41 | dash | -V actually |
03:48.44 | dash | docelmo: Awesome. |
03:48.45 | Mavvie | [root@mercury root]# asterisk -V |
03:48.46 | Mavvie | Asterisk CVS-HEAD-02/10/05-22:44:29 |
03:49.01 | Poemius | nyep, -V :) |
03:49.30 | Poemius | mine is more recent than yours :) |
03:49.36 | dec | Asterisk CVS-04/28/04-15:16:48 |
03:49.38 | dec | got it, thanks |
03:49.39 | dec | :) |
03:49.42 | Mavvie | but for your information, asterisk -r looks for /var/run/asterisk.ctl, not for a TCP socket. |
03:49.57 | Mavvie | Poemius: that's my test machine, the real one runs 1.0.7 |
03:50.21 | dec | Mavvie: oh okay. does it only look in /var/run ? I have the asterisk.ctl file, but its in another location... |
03:50.30 | Poemius | you're entitled to even use older versions too |
03:50.31 | Mavvie | where do you have it? |
03:50.41 | Poemius | as long as they have the moose penis sound :) |
03:51.48 | dec | Mavvie: /var/horizon/server1/logs/asterisk.ctl and /var/horizon/server2/logs/asterisk.ctl |
03:51.57 | dec | Mavvie: (I didn't set it up this way, blame someone else :P) |
03:52.16 | docelmo | uhh ya.. someone didnt use the default.. |
03:52.26 | docelmo | Dec if you need access kill the server and restart it. |
03:52.47 | Poemius | kill -9 the sucker :) |
03:52.54 | dec | It's done like that because there's two asterisk instances on the same box |
03:52.55 | Poemius | yeah, hehe, killing is coool :) |
03:52.59 | Poemius | hehhe hehe |
03:53.06 | dec | I don't need to kill it... i wanted to get console access to check the debug output :P |
03:53.16 | dec | But its okay, I think I've fixed the problem anyway |
03:54.24 | Poemius | too bad, no killing |
03:54.28 | dec | hehe |
03:54.55 | Poemius | better luck next time |
03:55.33 | dec | oooh, I could symlink /var/run/asterisk.ctl to the asterisk.ctl from the one I want to connect to :) |
03:58.18 | Poemius | yep, if you do that often, even make a script that does all for u |
04:00.12 | docelmo | Look at your physical log |
04:00.34 | Poemius | good idea |
04:01.14 | docelmo | :) I try |
04:01.23 | docelmo | even tho I am tired as hell.. |
04:01.32 | *** join/#asterisk cp5 (~samy@chcgil2-ar7-4-3-040-086.chcgil2.dsl-verizon.net) |
04:01.34 | cp5 | hi |
04:02.10 | cp5 | has anyone ran into zaptel/asterisk dropping ALL calls when on a span when the asterisk machine is the network, not the CPE? i have a bunch of different, strange error messages |
04:02.13 | cp5 | it happens maybe twice a day |
04:02.17 | odie_flocon | Hmm now I got 2 tdm400p cards. |
04:02.25 | *** join/#asterisk AlexCeli (~Alex@200.37.85.95) |
04:02.45 | AlexCeli | hi |
04:02.46 | odie_flocon | and a Hitachi wireless IP phone. |
04:03.06 | odie_flocon | and a polycom ip600 |
04:03.31 | docelmo | I have 500 PAP2-NA's and RT31P2-NA's.. :) |
04:03.44 | Poemius | p o w e r :) |
04:03.51 | odie_flocon | must be nice. |
04:05.11 | AlexCeli | i have a problem, my provider sent me the cisco firmware 7.4 for the 7960 cisco phone, but my default firmware is the 3.2 and I got a lot of "Disk Full Errors", i checked on digium list that I need another older firmware, exist a way to resolv it? |
04:05.59 | docelmo | Download a different IOS |
04:06.40 | AlexCeli | docelmo: Which version i need? |
04:06.45 | *** part/#asterisk dash (washort@68.212.221.181) |
04:08.13 | docelmo | What IOS does Digium say you need? |
04:08.51 | *** join/#asterisk Poemius (~poemius@adsl-70-48-192-81.adsl.iam.net.ma) |
04:09.43 | docelmo | Im looking at the IOS's now |
04:11.49 | *** join/#asterisk tylorflys (~tylorflys@ip68-104-178-155.ph.ph.cox.net) |
04:11.51 | docelmo | AlexCeli, I looked at the site says 7.x works well with * |
04:12.11 | *** join/#asterisk NewSole (~david@i216-58-44-245.avalonworks.net) |
04:12.33 | AlexCeli | docelmo: but i'm upgrading from 3.2 |
04:12.57 | AlexCeli | i don't have 4.X, 5.X, only i have 6.X and 7.X |
04:14.23 | docelmo | Use 6.0 seems most stable of them all |
04:15.06 | docelmo | Alex can you download or do you have 6? |
04:16.09 | *** join/#asterisk afrosheen (~afro@c-67-166-172-141.hsd1.tx.comcast.net) |
04:16.22 | afrosheen | hello again |
04:20.27 | docelmo | nite all.. work calls.. well sleep then work |
04:21.19 | afrosheen | yeah |
04:23.11 | harryvv | well, to test my failovr dial plan for calls into the states my sixtel does not failover to zap when the ether cord is disconected to the cable modem. |
04:23.16 | harryvv | hi afrosheen |
04:23.22 | afrosheen | hey harryvv |
04:23.37 | harryvv | seems simple enough with the N+1 |
04:24.36 | harryvv | what is the default timeout that asterisk kills a iax session where it looses connectivity? |
04:25.01 | harryvv | then advances to the next priority? |
04:28.21 | *** join/#asterisk jtodd (~jtodd@garthim.fox-den.com) |
04:30.12 | *** join/#asterisk kimosabe (~nat@201.129.75.182) |
04:30.33 | afrosheen | man it's dead in here tonight |
04:30.43 | afrosheen | there are flies zooming around below the vultures |
04:30.45 | kimosabe | can you run a sipura device with 8k with g711 |
04:31.13 | jakepdev | which sipura device? |
04:31.21 | kimosabe | 2000 model |
04:32.02 | AlexCeli | runs good..!!! |
04:32.18 | jakepdev | http://www.sipura.com/Documents/SPA-2000.pdf |
04:32.23 | kimosabe | the thing is i have several voip setups for interoffice comunications |
04:32.40 | kimosabe | works great only some times it gets choppy |
04:33.10 | jakepdev | oh - you asked a trick question |
04:33.21 | jakepdev | you already knew the answer |
04:33.32 | AlexCeli | I have another error with the Cisco 7969, the tftp looks for P0S3-07-4-00.sbn but i only have with the firmware the P003-07-4-00.sbn, it's the same file? |
04:36.41 | *** join/#asterisk ptblank (~MURDER1@68-169-176-137.lmdaca.adelphia.net) |
04:37.20 | AlexCeli | upss 7960* |
04:37.44 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
04:40.45 | afrosheen | AlexCeli: welcome to the wonderful world of cisco |
04:43.14 | *** join/#asterisk bimbo (~konversat@200.66.20.198) |
04:43.26 | bimbo | hello, I'm very newb to this voip thing... |
04:43.34 | afrosheen | bimbo: hello |
04:43.47 | bimbo | so I only have a few questions to ask you in order to clarify some things |
04:43.57 | afrosheen | bimbo: don't ask to ask, just go |
04:44.16 | bimbo | the first question is: is it possible to make a pc to phone call with asterisk? |
04:44.23 | bimbo | if so, what do I need for this? |
04:44.47 | afrosheen | bimbo: pc to phone call? like use your computer to call a phone line? |
04:44.55 | Sedorox | Yes, a softphone, a computer running asterisk, and either a VoIP PRovider. or a FXO card... |
04:44.57 | Sedorox | NEXT... |
04:44.58 | Sedorox | :-p |
04:45.05 | jakepdev | ~docs |
04:45.07 | 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 |
04:45.07 | bugbot | docs is assigned nothing and reported nothing. |
04:45.47 | bimbo | hmmm difference between a voip provider and a fxo card |
04:46.16 | jakepdev | ~fxo |
04:46.17 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
04:46.17 | bugbot | fxo is assigned nothing and reported nothing. |
04:46.31 | AlexCeli | afrosheen: lol, i'm understanding the beautifull world of Cisco. |
04:46.48 | afrosheen | AlexCeli: oh it's beautiful alright..if you don't have to touch it :) |
04:46.56 | jakepdev | http://www.voip-info.org/wiki-VOIP+Service+Providers |
04:54.14 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
04:54.41 | *** join/#asterisk three55ml (trilluser@cpe-66-25-89-157.satx.res.rr.com) |
04:54.52 | three55ml | Hey everyone |
04:56.15 | three55ml | Quiet tonight? |
04:56.32 | bimbo | ok lets say I choose to use the fxo card... I install it on my computer and then, would I be able to do what I'm thinkinig? or do I ened something else? |
04:56.41 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
04:56.56 | Sedorox | you can use it then.. just gotta learn howto setup asterisk |
04:57.52 | afrosheen | bimbo: you can do it with or without the fxo card..or an iaxy adapter, or a ton of other ways |
04:58.04 | jakepdev | anyone know how to get the call back after this: http://www.voip-info.org/wiki-Asterisk+cmd+MusicOnHold |
04:58.10 | afrosheen | bimbo: at the cheapest you can install Xlite or other soft client and get subscribed with a voip provider. |
04:58.27 | three55ml | jakepdev: Take them off hold :) |
04:58.35 | jakepdev | how? |
04:58.36 | three55ml | jakepdev: That just sets the MusicOnHold to use |
04:58.39 | bimbo | afrosheen: for some reason I don't want a voip provider... |
04:58.49 | jakepdev | no - it plays the music on hold |
04:58.54 | jakepdev | doesn't it? |
04:58.59 | three55ml | Let me look |
04:59.06 | afrosheen | bimbo: then I guess you wanna pay through the nose on long distance and world calling ;) |
04:59.08 | *** part/#asterisk DaLion (~DaLion@Toronto-HSE-ppp3770773.sympatico.ca) |
04:59.14 | *** join/#asterisk DaLion (~DaLion@Toronto-HSE-ppp3770773.sympatico.ca) |
04:59.25 | bimbo | afrosheen: if I buy the fxo card... this problem is solved I guess |
04:59.28 | *** join/#asterisk dca (~dca@c-67-166-37-218.hsd1.co.comcast.net) |
04:59.53 | three55ml | jakepdev: You're right, sorry |
04:59.57 | *** join/#asterisk DaLion (~DaLion@Toronto-HSE-ppp3770773.sympatico.ca) |
05:00.07 | afrosheen | bimbo: the fxo just gives you an interface for your soft phone to use..i.e. an expensive phone jack. |
05:00.08 | DaLion | syle u there ? |
05:00.19 | jakepdev | np - but Q still remains - there's gotta be a way |
05:00.29 | DaLion | tryiong to find how to compile or even how to get mysql-max |
05:00.30 | three55ml | jakepdev: What exactly you trying to do? |
05:00.48 | jakepdev | pipe music down while doing FastAGI stuff |
05:01.14 | jakepdev | when I'm done all data operations, start speaking variables |
05:01.15 | three55ml | Ah |
05:01.16 | bimbo | hmmmm so if I want to do pc to phone calls I have to expend some money |
05:01.47 | jakepdev | it's doing several AGI calls, so what happens now is music , pause, music, pause, etc.... |
05:01.48 | afrosheen | bimbo: yeah at some point you will |
05:01.49 | three55ml | I'm sure there's an AGI command to put the user on hold. |
05:01.57 | three55ml | Oh |
05:02.15 | jakepdev | not the end of the world, but would be nice if there's a way |
05:03.20 | three55ml | Yeah, off the top of my head I can't think of anything easy. I'm sure there's a few ways to do it though. |
05:04.14 | jakepdev | wish I could understand the C code enough to figure it out... oh well. |
05:04.54 | DaLion | anyone know how to fucking compile mysql-max ? |
05:05.13 | jakepdev | wow - someone had a rough day |
05:05.19 | DaLion | yes |
05:05.47 | bimbo | afrosheen: can I PM you? |
05:05.49 | DaLion | darn site is lame |
05:05.56 | three55ml | I had a worse one. My flight got cancelled and I got a speeding ticket. |
05:06.10 | DaLion | nothing about it they all point my ass to downloads page..only binairies are for friggin freebsd 4.7 |
05:06.13 | DaLion | i got 5.3 |
05:06.25 | three55ml | Have you looked at freebsd-ports? |
05:06.26 | DaLion | so i say ok ill do from source |
05:06.28 | jakepdev | plane too slow - car too fast |
05:06.36 | jakepdev | should equal out |
05:06.43 | DaLion | lol three55ml i NEVER USE PORTS... |
05:06.48 | DaLion | too many oold shit |
05:07.02 | three55ml | jakepdev: Well the lady messed up last week on the outbound flight and gave me a boarding pass for someone else, so it was all her fault :) |
05:07.04 | DaLion | an d i need to customize a lazy makinstall isnt enought for me |
05:07.25 | three55ml | Why you so intent on MySQL-MAX now? |
05:07.43 | jakepdev | that's 2 mistakes she made then? |
05:07.45 | three55ml | Last night you were running 5.x. I would just stick to 4.x in a production environment unless you're doing things it doesn't support. |
05:08.03 | jakepdev | one for you and someone else |
05:08.08 | three55ml | jakepdevL: Well it cascaded from there. She basically inadvertantly cancelled my return flight. |
05:08.13 | jakepdev | ugh |
05:08.27 | three55ml | And I found out at 7AM :) Luckily there were seats. |
05:08.35 | three55ml | I did not want to be stuck in Kansas a day longer. |
05:08.38 | DaLion | ? |
05:08.42 | DaLion | new box |
05:08.42 | jakepdev | hehe |
05:08.52 | three55ml | DaLion: Ah |
05:08.52 | jakepdev | guess you're not in Kansas anymore |
05:08.58 | DaLion | freebsd 5.3 .. and im trying to figure how to friggin compile max |
05:09.02 | jakepdev | sorry - had to go there |
05:09.02 | DaLion | its same source ? |
05:09.14 | DaLion | does it make automaticaly 3 deamons ? |
05:09.28 | DaLion | the fuck theyse assholes cant write a darn install like it should |
05:09.29 | three55ml | No |
05:09.33 | *** part/#asterisk Enigma8121 (~Enigma812@pcp02587377pcs.shlb1201.mi.comcast.net) |
05:09.36 | odie_flocon | sure |
05:09.40 | odie_flocon | why is me |
05:09.42 | DaLion | ok one good answer in 3 days form any forum post etc i saw |
05:09.42 | odie_flocon | soft |
05:09.49 | Nugget | what do you expect? it's mysql-related, it's bound to be full of misguided and malintentioned crap. |
05:09.54 | DaLion | ahha |
05:09.58 | DaLion | ok humor me |
05:10.05 | DaLion | how does one get max made then ? |
05:10.08 | Nugget | mysql attracts people who mean well but have no clue what to do. |
05:10.16 | Mavvie | :-) |
05:10.18 | odie_flocon | hey when I dial with an extention * doesn't do anything? |
05:10.26 | DaLion | i prolly have anough shit with * as it is without this mysql crap |
05:10.40 | DaLion | odie_flocon get used to * barfing on you |
05:10.40 | jakepdev | odie - sure it doesn't do anything? did you run a debug? |
05:10.44 | three55ml | DaLion: If it was me I would use 4.x and optimize it correctly. |
05:10.54 | three55ml | DaLion: 4.x has a proven record in production. |
05:11.03 | DaLion | oh and funny thing they invented RPM's for fucker who cant make shit |
05:11.20 | DaLion | three55ml i need clustered solution on 4 servers |
05:11.23 | DaLion | so i need -max |
05:11.38 | DaLion | all i can google gets RPM shit pages |
05:11.52 | DaLion | ~jbot RPM |
05:11.53 | jbot | Red Hat's package management system. URL: http://www.rpm.org/ |
05:11.53 | bugbot | jbot RPM is assigned nothing and reported nothing. |
05:11.54 | jbot | ...but rpm is already something else... |
05:12.05 | DaLion | lol |
05:12.10 | DaLion | ~jbot SPM |
05:12.11 | bugbot | jbot SPM is assigned nothing and reported nothing. |
05:12.12 | jbot | okay, bugbot |
05:13.01 | jakepdev | why are these two bots in a converversaion - looks like a bug |
05:13.25 | jakepdev | ~SPM |
05:13.26 | jbot | spm is, like, assigned nothing and reported nothing. |
05:13.26 | DaLion | man |
05:13.26 | bugbot | SPM is assigned nothing and reported nothing. |
05:13.30 | jakepdev | haha |
05:13.34 | DaLion | this i s totla shit |
05:14.36 | jakepdev | isn't there an IRC channel for this MySQL Max? |
05:14.54 | DaLion | no |
05:15.08 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
05:15.34 | DaLion | >> WHO #mysql-max |
05:15.34 | DaLion | [01:15]> #mysql-max End of /WHO list. |
05:16.43 | three55ml | It'll be the same channel as MySQL |
05:16.53 | odie_flocon | what company is that? |
05:16.53 | three55ml | Probably on EFNet |
05:17.20 | odie_flocon | Internet provider? |
05:17.23 | Qwell | umm |
05:17.33 | Qwell | DaLion: Which version of MySQL did you download? |
05:17.39 | Mavvie | I would try freenode.net myself first, but okay. |
05:17.49 | odie_flocon | where do you want? |
05:18.23 | odie_flocon | gt? |
05:18.27 | odie_flocon | no that's not GT |
05:18.32 | odie_flocon | that's MTS. |
05:18.32 | three55ml | Mavvie: Yeah, #mysql on FreeNode |
05:18.52 | odie_flocon | is that IP to pot's termination? |
05:18.56 | DaLion | 4.1.11 |
05:19.54 | Qwell | DaLion: and did you compile it? |
05:20.08 | DaLion | no |
05:20.23 | Qwell | then how are you going to have a mysqld-max binary? |
05:20.26 | DaLion | im trying to ./cponfigure it for max or something but nothing in docs and i read EVERY line of htem |
05:20.38 | DaLion | always talk about RPm shit |
05:20.39 | Qwell | So, you also ready INSTALL-SOURCES? |
05:20.47 | DaLion | ALL of it yes |
05:20.52 | DaLion | even the windozes SHIT |
05:20.59 | Qwell | and? |
05:21.19 | DaLion | now you agree they should of splitted install.unix and install.windoz.. or are theyre that much lamers out htere / |
05:21.21 | Qwell | You didn't read that mysqld-max is PART of mysql? |
05:21.33 | DaLion | nope |
05:21.55 | Qwell | well, it makes it fairly obvious |
05:22.00 | *** join/#asterisk ubergoober (~ubergoobe@c-24-16-110-117.hsd1.ca.comcast.net) |
05:22.09 | DaLion | well im to tired then to do this stuff |
05:22.35 | Qwell | Why not just get it from portage? |
05:22.37 | DaLion | where ( line) does it say the mysqld mysql-max etc will be compiled in same time |
05:22.44 | DaLion | portage ? |
05:22.47 | Qwell | DaLion: Where does it say that it doesn't? |
05:23.16 | Qwell | You're using FreeBSD, right? |
05:23.34 | DaLion | ./usr/ports/databases/mysql41-server |
05:23.36 | DaLion | its thetres |
05:23.53 | Qwell | So why not just use portage? |
05:24.15 | DaLion | 4.1.5 |
05:24.21 | DaLion | not 4.1.11 |
05:24.27 | Qwell | valid reason |
05:24.32 | Qwell | so, compile away |
05:24.36 | DaLion | hehe |
05:24.38 | DaLion | doing |
05:24.44 | Qwell | Generally, you're supposed to try something first, before complaining... |
05:24.51 | Qwell | at least, thats how we like it done |
05:25.03 | DaLion | well i looked into my other 5 sebrers running mysql and no max binarie so |
05:25.18 | DaLion | i `assumed` that it needed a compile flag |
05:25.27 | DaLion | and still do btw |
05:25.47 | Qwell | well, most of the packages have a -max package |
05:25.55 | Qwell | I'm fairly certain the source does not |
05:26.03 | DaLion | binaries builds yes..but only for bsd 4.XX |
05:26.07 | DaLion | not bsd 5.XX |
05:26.33 | DaLion | soi m fucked ? |
05:26.44 | Qwell | no, just compile it |
05:26.48 | DaLion | ill see.. should finish compiling in 34 -35 days max |
05:26.58 | DaLion | ;) |
05:27.01 | Qwell | mysql takes like 30 minutes, tops |
05:28.40 | Nugget | trying to run asterisk on non-linux platforms is not for the meek. |
05:28.52 | Nugget | it's a damn shame, but that's how it is. |
05:28.54 | DaLion | nope |
05:29.03 | DaLion | expesially zap |
05:29.07 | Nugget | yeah |
05:29.23 | DaLion | cant use only linux box we run if zap sevrers |
05:31.01 | Nugget | cant understand what you just tried to say |
05:31.05 | *** join/#asterisk remmo (~rem@smack.isp.net.au) |
05:33.31 | DaLion | compiled |
05:33.37 | DaLion | 4 minutes not bad |
05:34.25 | DaLion | now where should it be lol |
05:34.46 | Nugget | DaLion: are you under the mistaken impression that the mysql server port is version 4.1.5? |
05:35.01 | DaLion | it is |
05:35.03 | Nugget | no it is not |
05:35.39 | DaLion | bash-2.05b# cd /usr/ports/databases/mysql41-server |
05:35.40 | DaLion | bash-2.05b# make |
05:35.46 | DaLion | >> mysql-4.1.5-gamma.tar.gz doesn't seem to exist in /usr/ports/distfiles/. |
05:35.58 | DaLion | hmm that looks like 4.1.5 to me |
05:35.59 | Nugget | so update your damn ports. |
05:36.05 | DaLion | lol |
05:36.05 | Nugget | that's your fault. |
05:36.09 | Nugget | the port is at 4.1.11 |
05:36.18 | DaLion | yeah i compiled 4.1.11 |
05:36.26 | DaLion | still cant find where it makes mysql-max |
05:36.28 | Nugget | but you could have just used the port. |
05:36.39 | Qwell | mysqlD-max |
05:36.41 | Nugget | no wonder you complain that ports is all full of old stuff, you seem to not know how to update it. |
05:36.42 | Qwell | d, d, d |
05:37.00 | three55ml | I think I hear an echo in here, I remember saying to use ports 30 minutes ago :) |
05:37.12 | Qwell | three55ml: yeah, so did I, about 10 minutes ago |
05:37.21 | DaLion | so ? |
05:37.36 | DaLion | what the diff ? i got 4.1.11 sources and compiled.. wont change anything would it ? |
05:37.53 | three55ml | DaLion: Not trying to be rude, but why are you using things you're not familiar with? Last night you were in here asking how to fix MySQL because your max connections was 100, and now you're setting up a 5-server cluster all of a sudden? |
05:38.02 | DaLion | ahah |
05:38.04 | Nugget | I see several patches in the mysql port. no clue how important they are, but yeah, it does chance things. |
05:38.34 | DaLion | updating ports then |
05:38.47 | Nugget | not to mention the added convenience of the startup/shutdown scripts being automatically set up by the port. |
05:39.39 | Nugget | and what the hell did you change root's shell to bash for? :) |
05:39.41 | Qwell | and the updating |
05:39.45 | Nugget | that's just evil |
05:39.58 | Qwell | bash with root is evil on fbsd? |
05:40.00 | DaLion | didnt chagne root cshell just my own |
05:42.00 | DaLion | <PROTECTED> |
05:42.03 | DaLion | ? |
05:42.16 | Nugget | is that supposed to be a question? |
05:42.18 | Qwell | gonna be a long night... |
05:42.23 | Qwell | and a long rest of your life. ;/ |
05:42.26 | DaLion | ahahah |
05:43.09 | Nugget | you pasted a random command and then added a question mark. that's not a question. |
05:43.18 | DaLion | my bad |
05:43.26 | Nugget | I can only presume that the "?" means "is this what I want to do" but how the hell should we know -- what do you want to do? |
05:43.54 | Nugget | and how on earth did we get roped into the obligation of teaching you how to use your operating system anyway? |
05:44.01 | Nugget | isn't there a #freebsd somewhere? |
05:44.09 | DaLion | no it was wtf is this doing in a file called UPDATE when it has nothing to do with updating |
05:44.14 | Nugget | this has absolutely nothing to do with asterisk. |
05:44.25 | DaLion | heu well yes it updates the index of ports lol but not the ports |
05:44.29 | Nugget | that command has a great deal to do with updating. |
05:44.43 | Qwell | emerge sync <3 |
05:44.44 | DaLion | yeah sorry bye |
05:44.47 | *** part/#asterisk DaLion (~DaLion@Toronto-HSE-ppp3770773.sympatico.ca) |
05:44.53 | Qwell | my god |
05:45.05 | Nugget | damn. |
05:45.08 | Qwell | I've never even used freebsd, and I know more then him... |
05:45.12 | Nugget | heh |
05:45.20 | Nugget | for the record, "portage" is gentoo. "ports" is bsd. :) |
05:45.25 | Qwell | whatever, heh |
05:45.51 | Qwell | will remember that though |
05:46.53 | *** join/#asterisk heison (~heison@206.16.163.128) |
05:48.48 | Dovid | hello all |
05:48.57 | Dovid | anyone know of a good free pocker pc phone |
05:48.57 | Dovid | ? |
05:49.08 | denon | sjphone? |
05:49.25 | Dovid | didnt work |
05:49.29 | Dovid | anything else ? |
05:49.47 | denon | it works |
05:50.04 | marlowe | ~softphone |
05:50.06 | jbot | something that should be drug out into the street and shot |
05:50.06 | bugbot | softphone is assigned nothing and reported nothing. |
05:50.18 | marlowe | ~x-ten |
05:50.20 | bugbot | x-ten is assigned nothing and reported nothing. |
05:50.22 | *** join/#asterisk _Eagle_ (nick@wasteland.net) |
05:50.23 | marlowe | ok i give up |
05:50.28 | marlowe | try x-ten xlite |
05:50.29 | Dovid | x-ten charges for thiers |
05:50.35 | Dovid | i need a free one for testing |
05:50.42 | marlowe | x-ten light is free |
05:50.42 | Nugget | you didn't ask for a free one. :) |
05:50.45 | Nugget | oh, you did. :) |
05:50.48 | Nugget | nevermind me |
05:50.57 | marlowe | oh wait |
05:51.01 | marlowe | you said pocker pc |
05:51.06 | marlowe | first of all, its pocket :) |
05:51.12 | marlowe | sorrry nevermind then |
05:51.25 | marlowe | x-ten on pocket pc sucks |
05:51.25 | _Eagle_ | can anyone tell me how the Transfer() application works? i cant seem to get it to do what i want |
05:51.29 | marlowe | sjlite is a lot better |
05:51.40 | marlowe | sjphone omg |
05:51.46 | marlowe | im geting my terms mixed up |
05:51.50 | Nugget | heh |
05:51.56 | Nugget | no more beer for marlowe. |
05:52.02 | marlowe | :-/ |
05:52.08 | marlowe | im sick, and it sucks. |
05:52.30 | marlowe | i had to postpone my lasik surgery :( |
05:52.45 | Nugget | sjphone is useful but kinda klunky. x-lite is pretty, but it's cumbersome to use and intentionally limited. eyeBeam is bloated, but the only option if you don't want to use a handset or headset. |
05:53.01 | Nugget | and I've never even touched a pocke[rt] pc, so I have no idea what is viable on the platform, sorry. |
05:53.15 | marlowe | x-ten just sucks on a pocket pc |
05:53.21 | marlowe | I dont know why - but it does |
05:53.36 | marlowe | when I do use a softphone on my pc i personally use x-ten pro w/ g.7229 |
05:53.40 | marlowe | g.729 too |
05:53.47 | Nugget | it's a shame they don't sell x-pro any more. |
05:53.55 | marlowe | they dont? |
05:53.57 | marlowe | where was i? |
05:53.58 | Nugget | nope. just eyebeam |
05:54.05 | marlowe | that's dumb |
05:54.08 | Nugget | I agree. |
05:54.15 | _Eagle_ | what exactly is "Transfer()" expecting as an argument? every time i try to use an extension, it gives an error |
05:54.21 | marlowe | not everyone wants eyebeam |
05:54.30 | Nugget | _Eagle_: "show application transfer" |
05:54.36 | Qwell | Nugget++ |
05:54.41 | _Eagle_ | nugget: dont you think i already tried that? |
05:54.41 | Nugget | or google "site:voip-info.org transfer" |
05:54.49 | Nugget | no, I don't think you already tried that. |
05:54.50 | marlowe | wtf |
05:54.51 | _Eagle_ | i tried both of those, nugget |
05:54.57 | marlowe | They want to outsource X-PRO development |
05:55.00 | _Eagle_ | it doesnt help me at all |
05:55.00 | marlowe | I hate that |
05:55.20 | Nugget | nobody in here *ever* tries those things first. |
05:55.20 | Nugget | :) |
05:55.34 | marlowe | Nugget:I do :( |
05:55.45 | Qwell | Nugget: sadly, that includes the people who offer that as advice to others. :p |
05:55.45 | marlowe | I kung foo asterisk |
05:55.46 | _Eagle_ | nugget: ive been using asterisk for over 2 years now.. i know how to RTFM :-) |
05:55.54 | marlowe | Whats a manual? |
05:56.02 | Nugget | it's what cool cars use. |
05:56.11 | AlexCeli | the P0S3-07-4-00.sbn is the same file with P003-07-4-00.sbn? i can rename it? |
05:56.25 | _Eagle_ | the docs i found dont explain the problem i'm having |
05:56.36 | marlowe | _Eagle_: Why dont you try explaining the problem. |
05:56.38 | _Eagle_ | they say Transfer(extension) |
05:56.49 | _Eagle_ | but it doesnt like it when i give an extension |
05:56.56 | marlowe | Doesnnt like it? |
05:57.01 | marlowe | * doesnt have a personality |
05:57.03 | marlowe | Unfortunately |
05:57.19 | Nugget | bkw_ has enough personality for all of us. |
05:57.38 | _Eagle_ | the *only* time it liked any of the 300 things i tried was when i did Transfer(SIP/1000@sipmachine) then it sent the call to a different machine |
05:57.46 | Nugget | heh |
05:57.49 | _Eagle_ | but i want to transfer to another extension on the same machine |
05:58.20 | three55ml | Try including the context |
05:58.21 | Nugget | I don't know the answer. |
05:58.24 | three55ml | Like 101@internal |
05:58.27 | _Eagle_ | so i'm asking, what exactly does Transfer want? |
05:58.40 | Qwell | are the extensions you're trying included in the current context? |
05:58.41 | Nugget | check the source, I guess. |
05:58.43 | _Eagle_ | three: ok... but shouldnt it use th current context if i dont specify one? |
05:58.44 | marlowe | It wants a valid extension |
05:58.49 | marlowe | Make sure it's included in the context |
05:58.58 | marlowe | Wooops Qwell beat me |
05:59.17 | marlowe | I keep sneezing on the screen - I gotta go to bed |
05:59.18 | _Eagle_ | qwell: i tried that, yes |
05:59.38 | _Eagle_ | i created a new extension inside the current context just for testing that |
05:59.38 | three55ml | Well technically you need to include the type. If you want to sent to an extension in that, or even a differnt context, use Goto(context, exten, 1) |
05:59.57 | _Eagle_ | no... i need to use Transfer i think |
06:00.04 | _Eagle_ | Goto wont do what i need |
06:00.05 | Qwell | and...what happens? |
06:00.32 | _Eagle_ | qwell: that was one out of a hundred tries.. i dont remember the exact error... it just didnt work |
06:00.47 | _Eagle_ | ill try again, if it will help |
06:01.22 | _Eagle_ | what i'm trying to accomplish is using Dial() with the M() option to run a macro.... |
06:01.41 | _Eagle_ | then during that Macro, send the other side to an extension, then continue the original call where Dial left off |
06:01.55 | _Eagle_ | i need to "bring another person into the system" from the outside |
06:02.15 | _Eagle_ | while continuing the original caller's call independently after the second person is online |
06:03.25 | _Eagle_ | the only way ive seen to bring an outside into the system is with /var/spool/outgoing... but as far as i can tell, /var/spool/outgoing doesnt give any sort of call progress or info if the call fails/ is busy or no answer, etc |
06:03.37 | Qwell | wait, you want to run something AFTER a Dial(), while its still active? |
06:04.22 | _Eagle_ | i want caller A to Dial() caller B... send Caller B to extension 1, and caller A to extension 2, independent of eachother |
06:04.33 | Qwell | after the Dial()? |
06:04.41 | _Eagle_ | after, during, whatever |
06:05.18 | _Eagle_ | i need to be able to bring a third party into the system... without them dialing in |
06:05.52 | _Eagle_ | picture for example a conferencing system... and the moderator does a dialout to another phone number... and wants to add that person into the conference..... |
06:06.02 | _Eagle_ | thats just one example of what this could do |
06:06.11 | Dovid | can onyone help me set up sjphone for my pockt pc ? |
06:06.17 | Dovid | i am having a bit of trouble |
06:06.27 | dec | that will be great if you can get it to work _Eagle_ :) |
06:06.50 | _Eagle_ | supposedly it would work if i could get Transfer() to work properly |
06:07.03 | _Eagle_ | but maybe i'm misunderstanding transfer() completely |
06:07.12 | Dovid | anyone here use sjphone ? |
06:07.28 | denon | _Eagle_: use an agi to fire off a spool file :) |
06:07.41 | _Eagle_ | denon: spool files dont give call progress |
06:07.47 | *** join/#asterisk psycodad (~obiwan@2001:4060:4419:b1:0:0:0:2) |
06:07.48 | denon | nor do they give headaches ;) |
06:07.49 | _Eagle_ | how do i know if it failed? busy? no answer? |
06:08.01 | Dovid | anyone here use sjphone ? |
06:08.24 | dec | _Eagle_: thats not how Transfer() works, from my understanding. But I can't actually work out what Transfer is meant to do, and how... |
06:09.05 | psycodad | anybody know why I get 'CAPI[contr1/8]/0' ast extension from capi instead of '8', I can't match against incoming capi calls |
06:09.07 | *** part/#asterisk AlexCeli (~Alex@200.37.85.95) |
06:09.41 | _Eagle_ | well, is there any other way to bring a third party into the system then? |
06:10.08 | _Eagle_ | what i need, is kinda like a Dial() that fork()'s into its own channel |
06:11.40 | foobos | psycodad, that is just the channel name. you can still get the extension if you have incomingmsn and context properly set in capi.conf |
06:12.06 | *** join/#asterisk clive- (~pirch@rndf-146-44-91.telkomadsl.co.za) |
06:12.24 | _Eagle_ | from your reactions, it looks like i'm gonna get stuck writing my own application to do this :( |
06:13.05 | psycodad | foobos: I have msn=3,4,6,7,8 and incomingmsn=* |
06:13.14 | psycodad | I also tried incomingmsn=3,4,6,7,8 |
06:13.40 | foobos | psycodad, your phonenumber can't be 8 can it? |
06:13.53 | _Eagle_ | anyone know which source file the /var/spool/outgoing stuff is in? |
06:14.44 | psycodad | no, I am behind a pbx with s0 bus and I only get the last digit I guess... the internal number would be 43,44,46 etc.. |
06:15.11 | *** part/#asterisk darkskiez (~mhb@host-84-9-102-21.bulldogdsl.com) |
06:15.44 | elric | has anyone had much success with app_machinedetect.c ? |
06:16.03 | foobos | well then if you have lets say context=capi-in in the capi.conf the [capi-in] 8,1,Answer() should work in extensions.conf |
06:16.10 | _Eagle_ | ahh.. found it.... pbx subdirectory |
06:16.19 | _Eagle_ | thanks people... bbl |
06:16.37 | *** join/#asterisk Koshatul (~evangelio@inf-203-132-65-157.bne.ipnetworks.net.au) |
06:16.48 | psycodad | foobos: do you mean since I don't have it defined in the capi-context it falls back to default and s ? |
06:17.04 | foobos | psycodad, that's how it works |
06:19.48 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
06:20.28 | psycodad | foobos: I removed the extensions from default and moved them to local-capibus which is my capi context and I get: |
06:20.32 | psycodad | <PROTECTED> |
06:24.48 | *** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu) |
06:24.48 | *** join/#asterisk gres (~serg@81.222.48.242) |
06:26.25 | gres | Hi all. Have anybody success whith reciving fax by asterisk? |
06:26.46 | *** join/#asterisk ra|n_man (~mIRC@210.213.173.53) |
06:27.25 | ra|n_man | is it possible to use non-digium products with asterisk? |
06:27.40 | gres | Sending fax from pstn fax: fax(pstn)<->asterisk(te100p)<->cisco ata 186 <-> fax work well. |
06:27.41 | ra|n_man | we have here a quintum tenor asg 400 |
06:28.06 | gres | Mmm... |
06:28.21 | remmo | gres: i have fax receiving working |
06:28.33 | remmo | gres: but its not compat with all faxes |
06:28.41 | dec | can i receive fax via an IAX connection? :P |
06:28.52 | remmo | no |
06:28.56 | dec | damn. |
06:28.57 | dec | thanks |
06:29.01 | remmo | well depends on the codec |
06:29.04 | remmo | and latency |
06:29.13 | ra|n_man | is it possible to use non-digium products with asterisk? |
06:29.16 | dec | gsm and 40ms pings |
06:29.21 | remmo | but no is a very safe answer |
06:29.27 | dec | okay, thanks remmo. |
06:29.29 | gres | remmo: you receive fax by RxFax by Steave Underwood? |
06:29.35 | remmo | gres: yes |
06:30.05 | remmo | gres: but i'm only seeing 85% compatiability with fax machines, there are just some faxes that can not connect |
06:30.12 | *** join/#asterisk |Vulture| (~Vulture@64.234.204.68.cfl.res.rr.com) |
06:30.15 | ra|n_man | ??? |
06:30.22 | *** join/#asterisk afrosheen (~afro@c-67-166-172-141.hsd1.tx.comcast.net) |
06:31.01 | remmo | ra|n_man: yes |
06:31.36 | gres | remmo: What hardware do you use? is it Digium's hardware? |
06:31.52 | remmo | i have used an e100p, a clone x100p and nothing |
06:31.53 | *** join/#asterisk mcnobody (~laaksola@server.kopteri.net) |
06:32.14 | heison | |Vulture|: did you find a fix for the voicemail password problem? |
06:32.21 | ra|n_man | remmo: we have here a quintum tenor asg 400 gateway. can asterisk be configured to use this? |
06:32.42 | odie_flocon | Damn, why won't my FXS pickup my DTMF |
06:32.48 | remmo | dont know if it supports standards like sip and h323 there should be no reason, just time and tweaking |
06:33.13 | gres | What seting in you zapata.conf? Is Echocancelation off? |
06:33.26 | odie_flocon | let me see |
06:34.21 | odie_flocon | no |
06:34.26 | *** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu) |
06:34.43 | ra|n_man | remmo: Quintum supports sip & h323 |
06:35.09 | odie_flocon | gres no it is not |
06:35.17 | ra|n_man | remmo: the thing is the quintum also has its own configuration software that looks like asterisk |
06:35.23 | remmo | ra|n_man: then should be fine |
06:35.39 | remmo | ra|n_man: that product should do everything you need |
06:35.56 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@d01m-1-164.d4.club-internet.fr) |
06:36.29 | |Vulture| | heison: yes it was DTMF problems with 1.0.5-->1.0.7 not voicemail |
06:37.05 | ra|n_man | remmo: but can i tie up asterisk with the quintum? |
06:37.06 | |Vulture| | prolly just bad config on my side and they tightened it up on their side and it messed up my config |
06:37.27 | gres | odie_flocon, remmo: Ok. thks. I'll try... |
06:37.28 | ra|n_man | remmo: am using asteriskwin32 |
06:38.01 | odie_flocon | ???? |
06:38.18 | BoRiS | odie: Have you tried a different phone? |
06:38.21 | remmo | ra|n_man: probably not |
06:38.27 | odie_flocon | yes 2 different phones |
06:39.49 | *** join/#asterisk oej (~oej@apollo.webway.se) |
06:39.59 | afrosheen | so who knows how to use disa from the ivr to dump an outside caller into a meetme room |
06:40.00 | ra|n_man | remmo: i was hoping that i could use the fxs of the quintum with asterisk |
06:40.24 | ra|n_man | remmo: thanks anyways |
06:42.28 | *** part/#asterisk ra|n_man (~mIRC@210.213.173.53) |
06:46.03 | *** join/#asterisk Mazda-MX5 (~root@220-130-142-43.HINET-IP.hinet.net) |
06:46.19 | Mazda-MX5 | hi , all |
06:46.33 | remmo | mazda 6's are better |
06:46.41 | Mazda-MX5 | ha ~ |
06:46.53 | Mazda-MX5 | I love MX5 |
06:47.16 | remmo | i used to love 6. mazdas just dont have much BALLS |
06:47.18 | Mazda-MX5 | but , no money |
06:48.41 | Mazda-MX5 | I have a question , why the asterisk report "Unable to find a path from alaw to g729" when I accept a call ?? |
06:49.16 | Mazda-MX5 | caller is LP-201 , be caller is Cisco 7940 |
06:51.19 | Mazda-MX5 | remmo , I love drift racing |
06:51.34 | Zgarbi | I have mazda 323, old but works |
06:52.06 | Zgarbi | in real |
06:52.15 | Mazda-MX5 | I agree |
06:52.50 | *** join/#asterisk wiseguy_ (~chivilis@vadyba.vtu.lt) |
06:52.55 | wiseguy_ | hellow |
06:53.01 | remmo | Mazda-MX5: hard to drift race in a front wheel drive |
06:53.02 | Mazda-MX5 | hi |
06:53.04 | wiseguy_ | anybody has ata186 cisco sip ISO? |
06:53.09 | wiseguy_ | IOS im |
06:53.10 | wiseguy_ | :) |
06:53.19 | remmo | that would be illegal |
06:53.35 | wiseguy_ | what?:) |
06:53.40 | Mazda-MX5 | How body know Why the asterisk report "Unable to find a path from alaw to g729" when I accept a call ?? |
06:53.43 | remmo | i didnt see anything |
06:53.57 | remmo | cause the call authed ? |
06:54.15 | remmo | you can accept a call but where any audio is passed thats another question |
06:55.37 | Mazda-MX5 | I accept a call , then asterisk report it , and I can not listen any voice |
06:55.42 | wiseguy_ | anybody |
06:55.48 | wiseguy_ | using cisco ata186? |
06:55.49 | Zgarbi | can somebody provide me - where is it possible to buy linksys pap2, as I'm international buyer and want to buy via internet, as cheap as possible. at this case 10 units. |
06:55.49 | wiseguy_ | :) |
06:56.27 | Zgarbi | Mazda-MX5 did u in sip.conf did: canreinvite=no ? |
06:56.28 | Mazda-MX5 | No , I never use ata186 , sorry |
06:56.42 | Mazda-MX5 | wait , I check |
06:57.12 | Mazda-MX5 | my setting is canreinvite=yes |
06:57.21 | Zgarbi | change to no |
06:57.29 | Mazda-MX5 | the keypoint is canreinvite value ? |
06:57.42 | *** part/#asterisk tylorflys (~tylorflys@ip68-104-178-155.ph.ph.cox.net) |
06:57.53 | Mazda-MX5 | thank you , I try it , Now |
06:59.16 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
07:00.19 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
07:01.18 | Mazda-MX5 | Zgarbi , Thank you ,but it do not work , I still asterisk report |
07:01.42 | Mazda-MX5 | still see asterisk report it |
07:01.43 | |Vulture| | anyone use ChanSpy? |
07:02.51 | Zgarbi | wht hardware do u have? |
07:02.53 | Zgarbi | what |
07:03.34 | Mazda-MX5 | caller is a LP-201 , be caller is CISCO 7940 |
07:04.27 | Mazda-MX5 | I think CISCO only support g729... |
07:06.02 | shepherd | cisco should support more than just g729 |
07:06.07 | Zgarbi | is your asterisk supports g729? |
07:06.21 | shepherd | asterisk doesn't have to support g729 for g729 to work |
07:06.43 | Mazda-MX5 | asterisk do not support g729 ??!! |
07:06.43 | shepherd | as long as it doesn't have to transcode |
07:07.00 | shepherd | asterisk supports g729, but you have to buy a license |
07:07.03 | Zgarbi | as I remembers with addons |
07:07.25 | shepherd | but, if both phones support g729 |
07:07.29 | shepherd | you don't need a license |
07:07.53 | Mazda-MX5 | my /usr/lib/asterisk/modules have format_g729.so , but not have codec_g729.so |
07:07.55 | *** part/#asterisk oej (~oej@apollo.webway.se) |
07:08.19 | Silik0n | thats cuase you gotta buy g729 cause its patented |
07:08.27 | remmo | if you search hi and low on the web you will find such a beast but its not called codec_g729.so |
07:08.36 | remmo | the only codec i'm missing is speex |
07:08.41 | Mazda-MX5 | so , I see, thank you ,Silik0n |
07:08.44 | Silik0n | remmosure it is |
07:08.45 | *** join/#asterisk pbxjunkie (~Stormtroo@ppp14-adsl-159.ath.forthnet.gr) |
07:08.50 | Silik0n | there are 2 different one |
07:08.51 | Silik0n | s |
07:08.55 | Mazda-MX5 | thank you , Zgarbi |
07:09.01 | shepherd | http://www.digium.com/index.php?menu=asterisk_g729 |
07:09.11 | shepherd | you can get the codec, but no license |
07:09.27 | Silik0n | that license is only$10 |
07:09.31 | pbxjunkie | mornin' :) Has anybody got any idea WHY my asterisk fails to load chan_zap.so ? with error message: ast_load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_retrieve_call_to_death |
07:09.35 | shepherd | true :) |
07:09.35 | Mazda-MX5 | but , CISCO sould support more than g729 , such as ulaw , alaw |
07:09.44 | shepherd | g711 |
07:09.46 | shepherd | gsm |
07:09.57 | shepherd | gsm #1 |
07:09.57 | shepherd | :) |
07:10.31 | Silik0n | pbx its prolly configured wrong |
07:10.36 | Mazda-MX5 | so , If my 3 IP phone all support gsm , my sip.conf only need allow=gsm ? |
07:10.43 | shepherd | yes |
07:10.52 | shepherd | disallow=all |
07:10.53 | Silik0n | Mazda-MX5 only if you just want to use GSM |
07:10.54 | shepherd | allow=gsm |
07:10.55 | shepherd | :) |
07:11.06 | Silik0n | on a lan use g711 |
07:11.21 | *** join/#asterisk pooh_ (user78@cust.15.241.adsl.cistron.nl) |
07:11.28 | Silik0n | hell 711 works just fine over the internet too... (unless you're on dialup or something) |
07:11.33 | Mazda-MX5 | Thnk you , all~~~ Thank you very much!! |
07:11.54 | shepherd | and you're not downloading pr0n |
07:11.59 | Mazda-MX5 | I try now~ |
07:12.07 | pooh_ | What is the international access code when dialing FROM the US pls ? |
07:12.12 | Silik0n | 1 |
07:12.15 | Silik0n | 011 |
07:12.15 | shepherd | 11 |
07:12.17 | Silik0n | rather |
07:12.31 | pooh_ | ok, thx |
07:12.33 | Silik0n | 011+number |
07:12.36 | pooh_ | thx |
07:13.32 | Zgarbi | can somebody provide me - where is it possible to buy linksys pap2, as I'm international buyer and want to buy via internet, as cheap as possible. at this case 10 units. |
07:13.43 | Mazda-MX5 | Now I get "chan_sip.c:2773 process_sdp: No compatible codecs!" |
07:13.48 | shepherd | it would have been funny if you gave him some 24 number number |
07:13.58 | shepherd | 24 digit |
07:13.59 | shepherd | also |
07:13.59 | shepherd | hehe |
07:14.00 | Mazda-MX5 | my sip.conf only have allow=gsm |
07:14.17 | Silik0n | Mazda-MX5 that happens when theres not a common codec or transcoder available |
07:14.18 | pbxjunkie | if I keep 'load => chan_zap.so' in my modules.conf then asterisk fails, if I remove it.. then it loads fine.. and it also 'sees' zap channels (i.e. give me access to zap show channels command) |
07:14.26 | shepherd | you could try allow=all |
07:14.45 | |Vulture| | zapata.conf/zaptel.conf is configured wrong pbxjunkie |
07:14.55 | Silik0n | pbxjunkie yu dont have to tell it to load it... asterisk will autoload it when needed by default |
07:15.11 | Silik0n | if you are using zap devices you probably have your zap configs wrong |
07:15.28 | Silik0n | ztcfg -vvvv will tell you if you have the drivers configured right or not |
07:15.43 | |Vulture| | wow... its that time again boys and girls... its time to redesign my dialplan LOL |
07:15.48 | Silik0n | * will seg like bitch with bad zap configs |
07:16.18 | wiseguy_ | hellow |
07:16.26 | Silik0n | my dialplan is exten => _X.,1,Dial(Zap/g1/911) |
07:16.27 | wiseguy_ | anybody using cisco ata186? |
07:16.28 | wiseguy_ | :) |
07:16.39 | |Vulture| | lol |
07:16.47 | |Vulture| | nice Silik0n |
07:16.53 | Silik0n | aight i go bed now |
07:16.57 | Silik0n | peice out |
07:17.00 | |Vulture| | lata |
07:17.06 | |Vulture| | haha lata.... |
07:17.10 | |Vulture| | nvm :P |
07:17.34 | Mazda-MX5 | I can not write "allow=all" that will become "Unable to find a path from g729 to ulaw" |
07:17.54 | pbxjunkie | can somebody quickly take a peek at my zaptel / zapata files? http://pastebin.ca/9522 |
07:18.00 | shepherd | allow=all |
07:18.05 | shepherd | disallow=g729 |
07:18.16 | |Vulture| | pbxjunkie: post your ztcfg -vv please too |
07:18.26 | |Vulture| | oh wow |
07:18.27 | Mazda-MX5 | thank you , I am trying |
07:18.27 | |Vulture| | you did |
07:18.28 | |Vulture| | hahaha |
07:19.21 | |Vulture| | pbxjunkie: thats quite the setup... looks correct to me |
07:19.26 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
07:19.37 | |Vulture| | pbxjunkie: Id recommend backing it off and trying to get it to run with just 1 card configured |
07:20.43 | pbxjunkie | |Vulture| i only have 1 card :) |
07:21.14 | |Vulture| | pbxjunkie: I didn't mean ripping the card out I meant just configure 1 module at a time |
07:21.18 | |Vulture| | the the TE400P? |
07:21.25 | pbxjunkie | quadBri |
07:21.49 | |Vulture| | most Ive used is the TE110P |
07:21.53 | pbxjunkie | oh so I should try configuring just 1 module at a time |
07:22.03 | Mazda-MX5 | shepherd , Thank you , I have been not see "Unable to find a path from g729 to ulaw" , but , when Iaccepted a call , I can not listen any voice ~><~ |
07:22.12 | |Vulture| | yea... issolate your problem |
07:22.19 | |Vulture| | but the config looked fine |
07:22.29 | |Vulture| | * may not be liking the config of one of them though |
07:25.18 | *** join/#asterisk zoa (~zoa@pirus.securax.be) |
07:28.29 | *** join/#asterisk brc__ (~brian@brc.base.supporter.pdpc) |
07:28.41 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
07:29.18 | Mazda-MX5 | Oh~ I will crazy |
07:32.22 | Mazda-MX5 | .. |
07:33.35 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
07:35.21 | *** join/#asterisk UPMeduardo (~UPMeduard@tauro2.dit.upm.es) |
07:38.48 | *** join/#asterisk ta[i]nted (~tainted@adsl-69-108-114-226.dsl.irvnca.pacbell.net) |
07:40.24 | Mazda-MX5 | .. |
07:40.58 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
07:41.01 | Delvar | Mazda-MX5: do you have g729 licances installed on your asterisk server? |
07:42.00 | Delvar | Mazda-MX5: if not either 1. dont use g729 2. use clients that both suport g729 3. buy licances from digium |
07:51.03 | *** join/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com) |
07:51.14 | *** join/#asterisk oden (~oden@194-237-146-22.customer.telia.com) |
07:52.21 | ta[i]nted | is it possible to get the asterisk-addons 1.0.6 package from cvs? |
07:52.22 | |Vulture| | I need to get an electric shcoker for peoples chairs for when they twist the phone cord... |
07:52.43 | ta[i]nted | |Vulture| why |
07:52.53 | |Vulture| | ta[i]nted: http://www.asterisk.org/html/downloads/asterisk-addons-1.0.6.tar.gz |
07:52.54 | darkskiez | is it normal for the milliwatt test to pop on a lan ? |
07:53.19 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
07:53.28 | |Vulture| | ta[i]nted: they keep twisting the chords on these IP500s into a coiled bunch on the table where you can't even life the handset |
07:54.05 | ta[i]nted | get one of those cord untwisters |
07:54.14 | ta[i]nted | it's like 0.99 |
07:54.31 | Delvar | supperglue the handset down... that will teach them! |
07:54.50 | |Vulture| | ta[i]nted: yea we are going to do that... but still... is it that hard? |
07:54.59 | |Vulture| | they have to be like sitting and spinning on the chair |
07:55.23 | |Vulture| | they got plantronics headsets now though... |
07:55.25 | Delvar | you would be supprised what users actualy do instead of working... |
07:55.30 | ta[i]nted | those are nice |
07:55.56 | |Vulture| | Delvar: its pretty hard for them to not work.. we have a pretty tight lock on our offices |
07:56.15 | Delvar | lol |
07:56.48 | *** join/#asterisk ra|n_man (~mirc@203.87.133.32) |
07:56.59 | |Vulture| | yea... we are pretty big brotherish |
07:57.24 | ra|n_man | can an asterisk server be configured to be a gatekeeper? |
07:57.59 | ra|n_man | i already have a gateway that i need to register to a gatekeeper |
07:59.22 | ra|n_man | ??? |
07:59.22 | |Vulture| | gatekeeper and keyholder... nevermind I have had lack of sleep |
07:59.22 | |Vulture| | just ignore me |
07:59.30 | Delvar | wondered why it sounded familiar :) |
07:59.41 | kajtzu | wasn't it the gatekeeper and keymaster? |
07:59.46 | Delvar | watched it not so long ago... |
07:59.52 | |Vulture| | keymaster thats it |
08:00.03 | ra|n_man | seriously guys |
08:00.07 | *** join/#asterisk inspired (mikael@213.197.167.61) |
08:00.18 | Delvar | serius? omg this is #asterisk! |
08:00.20 | ra|n_man | is it possible to configure asterisk as a gatekeeper? |
08:00.57 | Delvar | tbh i dont know, i have no idea what a gatekeeper is |
08:01.02 | remmo | no |
08:01.41 | |Vulture| | damn its 4am... |
08:01.47 | kajtzu | Delvar: it's a thingy that keeps track of h.323 zones |
08:01.53 | kajtzu | and registrants within them |
08:02.02 | darkskiez | the mozilla xul xml namespace is http://www.mozilla.org/keymaster/gatekeeper/there.is.only.xul |
08:02.21 | darkskiez | that has to be in every firefox plugin, it makes me laugh... god i'm sad. |
08:03.52 | remmo | xul rox |
08:04.16 | remmo | anyway home tmie |
08:04.28 | ta[i]nted | should i put the contents of the asterisk-addons into /usr/src/asterisk-1.0.6? |
08:05.02 | *** join/#asterisk RES2 (~res-1@gateway1.nemox.net) |
08:05.03 | RES2 | hi |
08:05.04 | ta[i]nted | when i try to make install, i get 'asterisk.h: No such file or directory' |
08:05.46 | RES2 | Can anyone help me with a callerID-problem (sorry for my bad english)? |
08:07.06 | RES2 | I make a call-redirection in a SIP-phone. So the outgoing callerID ist the callerID of the caller. I want to send the callerID of the redirectin phone. |
08:07.33 | *** join/#asterisk ra|n_man (~mirc@203.87.133.32) |
08:07.39 | ta[i]nted | hep me hep me peas |
08:07.53 | ra|n_man | am back |
08:08.14 | ra|n_man | so? is it possible for asterisk to be a gatekeeper for my gateway? |
08:09.07 | ta[i]nted | what is gatekeeper |
08:09.26 | ta[i]nted | define gatekeeper roles |
08:13.06 | ra|n_man | gatekeepers give access to gateways and border elements to interconnect and do voip |
08:13.21 | *** join/#asterisk Moc_ (~Moc@modemcable165.109-70-69.mc.videotron.ca) |
08:13.59 | ta[i]nted | like a proxy? |
08:14.16 | ra|n_man | kinda like that |
08:14.26 | ra|n_man | like a security guard of a building |
08:15.01 | ta[i]nted | yea like a proxy |
08:15.07 | ta[i]nted | u could use asterisk |
08:15.10 | Delvar | then yes asterisk can do that |
08:15.15 | ra|n_man | because a gateway needs to register to it before it can have access |
08:15.23 | ra|n_man | how is it done? |
08:15.33 | *** join/#asterisk nrc (~username@zeus.eurotux.com) |
08:15.34 | ta[i]nted | what kind of gateway is it |
08:15.43 | Delvar | www.voip-info.org - look for sip.conf - register => |
08:15.50 | Delvar | its prety simple |
08:16.02 | Delvar | hahaha |
08:16.26 | *** join/#asterisk ra|n_man (~mirc@203.87.133.32) |
08:16.46 | ra|n_man | my gateway is a quintum tenor asg 400 |
08:16.47 | RES2 | Delvar ... do you mean me? |
08:18.21 | ra|n_man | how do i configure asterisk to be a gatekeeper |
08:18.23 | ra|n_man | ??? |
08:21.17 | cypromis | you don't |
08:21.19 | *** join/#asterisk basta (~kqj@62-101-126-233.fastres.net) |
08:21.23 | cypromis | asterisk has no gatekeeper functionality |
08:21.56 | basta | hallo, anyone using cisco 7912 and/or 7960 out there ? |
08:22.02 | *** join/#asterisk tainted_ (~tainted@adsl-69-108-114-226.dsl.irvnca.pacbell.net) |
08:23.12 | darkskiez | loads of folks |
08:23.13 | *** join/#asterisk terracon (~tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com) |
08:24.19 | basta | seems music on hold/transfer stopped working with 1.0.7, can anyone confirm this ? |
08:28.22 | basta | sorry, it was 1.0.6 |
08:28.22 | inspired | can a member in queues.conf be a IAX2 friend? |
08:28.37 | basta | and sip protocol |
08:31.10 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.161.115.68.195.rev.coltfrance.com) |
08:35.30 | psycodad | I am still fidling with CAPI channel: I get the correct MSN in ${DNID} but in ${EXTEN} I get the full channel string not just the MSN... I guess I can do it with gotoif based on ${DNID} but that should work with ${EXTEN} too, right ? |
08:35.54 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
08:38.17 | *** join/#asterisk vagwin (~vagwin@mk-ns500-1.uk.tiscali.com) |
08:39.52 | jalsot | hi |
08:40.36 | jalsot | does anybody know how much consumes one recorded call in sln format? is it 2x64kbps=128kbit/s ? |
08:41.29 | jalsot | [without mixing] |
08:42.40 | PoWeRKiLL | who use call fowarding ? |
08:44.00 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
08:44.58 | RES2 | Can anyone help me with a callerID-problem (sorry for my bad english)? |
08:45.00 | RES2 | I make a call-redirection in a SIP-phone. So the outgoing callerID ist the callerID of the caller. I want to send the callerID of the redirectin phone. |
08:51.53 | *** part/#asterisk RES2 (~res-1@gateway1.nemox.net) |
08:54.31 | *** join/#asterisk DrJolo (~chatzilla@cerber.ftj.agh.edu.pl) |
09:01.06 | psycodad | anybody working with CAPI channels at all ? |
09:02.19 | pooh_ | I do |
09:04.03 | pgpkeys | goddamn, this is one hell of an active channel. i like, i like :) |
09:04.24 | pgpkeys | been lurking for the past few days, good stuff. |
09:07.15 | pooh_ | psycodad: Fiddle with ${EXTEN:x} , where x is the number of digits you want to cut off of the exten from the start. e.g. exten=1234, ${EXTEN:2} = 34 |
09:09.47 | *** join/#asterisk Betu| (~betul@62.244.193.101) |
09:10.17 | *** part/#asterisk Betu| (~betul@62.244.193.101) |
09:10.50 | *** join/#asterisk zoa (~zoa@pirus.securax.be) |
09:11.19 | *** join/#asterisk Betu| (~betul@62.244.193.101) |
09:11.23 | *** part/#asterisk Betu| (~betul@62.244.193.101) |
09:11.33 | *** join/#asterisk Betu| (~betul@62.244.193.101) |
09:11.40 | foobos | psycodad, i'm guessing your MSN is too short so chan_capi stuffs the whole channel in exten |
09:11.55 | foobos | cause i get the exten just fine |
09:17.34 | *** join/#asterisk ddum (~spamfilte@argus.nwl.se) |
09:21.16 | *** join/#asterisk sympad (~Misha@195.138.127.98) |
09:21.47 | sympad | !list |
09:23.31 | *** join/#asterisk Alexi1 (~alexis@www.trim.it) |
09:23.59 | *** join/#asterisk psiderov (~pi@83.228.0.212) |
09:24.11 | *** join/#asterisk shmooz (~nobody@host6411912762.biz.tor.fcibroadband.com) |
09:25.04 | psiderov | anyone can help me how to forward codec capabilities of the UAC to the pstn gw, not the codecs set in sip.conf |
09:27.39 | psiderov | or is there such scenario at all ? |
09:28.25 | zoa | yes there is |
09:29.35 | psiderov | thanks zoa, any idea how to do that ? |
09:31.57 | psiderov | or where shoul I read that / who I have to ask ? |
09:32.06 | *** part/#asterisk Alexi1 (~alexis@www.trim.it) |
09:33.04 | psiderov | I'm trying to edit chan_sip.c but still with no success |
09:35.27 | zoa | you dont need to edit anything |
09:35.37 | zoa | there are some vars / patches for it normaly |
09:35.41 | zoa | dont know them by head |
09:35.48 | zoa | and dont have the time to look for it |
09:35.49 | zoa | sorry |
09:36.14 | psiderov | just something that can push me on ... name or something ? |
09:38.50 | *** join/#asterisk ikey1 (ikey@220.226.16.30) |
09:39.03 | psiderov | name of the patch ? |
09:39.12 | zoa | codec preference maybe |
09:39.30 | psiderov | ok, thanks zoa :) |
09:43.37 | *** join/#asterisk anonymous12345 (~anonymous@60.48.111.128) |
09:45.41 | wiseguy_ | anybody using cisco ata186? |
09:46.12 | psiderov | I have few clients using it ? |
09:46.17 | *** join/#asterisk teq- (~p@xdsla026.osnanet.de) |
09:46.32 | newl | We don't know, do you? |
09:46.33 | basta | wiseguy, me |
09:48.37 | *** join/#asterisk yaboo (~jsirucka@220.245.131.131) |
09:48.54 | *** join/#asterisk Betu| (~betul@62.244.193.101) |
09:49.12 | anonymous12345 | hi i m interested to do further development on asterisk... since i m new i know the place to start if from bug level - i m from win background |
09:49.50 | anonymous12345 | but i found out the cvs thing is rather confusing... anyone mind to explain... |
09:50.05 | *** join/#asterisk Donuil (~cini_lab@217.9.64.213) |
09:50.45 | basta | yes, there's a guy named google who can perfectly explain you how cvs works ;) |
09:51.15 | anonymous12345 | wow tat good... but the tree and branches is the one confuse me... |
09:51.17 | Donuil | hi to all... can someone give me a suggestion about configuration of zaptel.conf and zapata.conf? |
09:51.43 | anonymous12345 | there is thing called cvs-head and a lot of other version stable version... |
09:54.50 | *** join/#asterisk Betu| (~betul@62.244.193.101) |
09:55.21 | anonymous12345 | is there any tools (web) that can view the cvs tree?? |
09:57.03 | *** join/#asterisk Yellow_Fuzzy (yellow@c211-31-41-9.wavrl1.nsw.optusnet.com.au) |
09:57.06 | Yellow_Fuzzy | hi |
09:57.34 | Yellow_Fuzzy | Im trying to setup my account on a new SIP provider with Asterisk@home but am having a few problems |
09:59.32 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
10:00.56 | *** join/#asterisk kapanol (~edit@203-118-140-154.pppoe.ihug.co.nz) |
10:01.05 | *** join/#asterisk RoyK (~roy@143.80-202-166.nextgentel.com) |
10:03.42 | *** join/#asterisk emitrax (~emitrax@ingnatdyn33.unime.it) |
10:03.45 | emitrax | hi |
10:04.50 | emitrax | WARNING[16658]: chan_sip.c:8838 reload_config: Section 'xlite1' lacks type |
10:05.32 | emitrax | never mind... I'm an idiot :) |
10:06.35 | emitrax | NOTICE[16658]: chan_sip.c:7681 handle_request: Registration from 'sip:line1@X.17.32.4' failed for 'X.17.129.167' |
10:07.05 | emitrax | I just changed the firmware on a 7940 from SCCP to SIP 7.0 and now Im trying to register it with asterisk |
10:07.11 | emitrax | but I get that error |
10:07.41 | emitrax | I think I edited correctly either sip.conf and extensions.conf |
10:08.07 | emitrax | so I don't know how to debug it... can someone give me a hint please? |
10:08.24 | *** join/#asterisk Donuil (~cini_lab@217.9.64.213) |
10:09.41 | Donuil | I try to call my zap phone but unsuccessfully... can someone help me with the zapta.conf file? |
10:09.57 | *** join/#asterisk Jax (~tsche@adsl-62-167-77-76.adslplus.ch) |
10:10.03 | Jax | hello :) |
10:10.49 | Jax | i'm realyl new to VoIP.. not sure what kind of hardware i need, if i need to get an additional service @ my ISP, etc.. maybe somebody can fill me in? |
10:12.36 | *** join/#asterisk pbxjunkie (~stormtroo@videocomputer.gr) |
10:13.29 | *** join/#asterisk Betu| (~betul@62.244.193.101) |
10:14.30 | *** part/#asterisk Betu| (~betul@62.244.193.101) |
10:18.16 | *** join/#asterisk Donuil (~cini_lab@217.9.64.213) |
10:18.43 | pbxjunkie | all this time I've been having trouble making calls through my quadbri when only now I noticed that all 4 leds on the back are black |
10:19.08 | pbxjunkie | eeer.. red :D |
10:20.24 | Donuil | sorry I'm falling sometime... so someone have experience with the configuration of the zapata.conf file? |
10:21.32 | RoyK | http://www-128.ibm.com/developerworks/library/pa-nl3-marenostrum.html |
10:22.09 | Donuil | I try to call unsuccessful my zap phone... I sure that the card is correctly installed... |
10:22.54 | RoyK | why do you try to be unsuccessful? |
10:23.55 | Donuil | because it doesn't ring and start voicemail as by setting |
10:25.28 | FaithX | do you have to run ztcfg with those cards/ |
10:25.37 | FaithX | ? |
10:26.00 | Donuil | I have a digium tdm400p installed in configuration 22b, with 2 fxs and 2 fxo |
10:28.01 | Donuil | i configured my zaptel.conf with fxo=1-2 fxs=3-4 and when I launch ztcfg -vvv all is right |
10:30.24 | sympad | put smth. in extension.conf ? |
10:31.03 | basta | donuil, since the voicemail starts, isn't maybe an extensions misconfiguration ? |
10:31.05 | Donuil | the I configured my zapata.conf 'signaling=fxo_ks callerid="mycallerid" <1003> channel => to call my zap phone on Zap/3... is correct according to you? |
10:32.07 | sympad | where from do you try to place the call ? |
10:33.24 | Donuil | sympad from asterisk console... basta according to you is a card problem?however I have no error logs about it |
10:33.43 | facek_ | can someone help me with PAP2 ? |
10:33.57 | newl | your provider can. :) |
10:36.26 | Donuil | maybe a signaling problem?It is correct my configuration in zapata.conf matching with zaptel.conf? |
10:36.51 | sympad | 1-2 are FXS or FXO ports ? |
10:38.40 | Donuil | in my zaptel.conf fxo=1-2 and fxs=3-4... Does means it that 1-2 are fxs port? |
10:46.35 | elric | does anyone have an example / snip of extensions.conf with MachineDetect() used in it? theres naught on the wiki about it |
10:49.47 | elric | or take a look at this and tell me what i am doing wrong |
10:49.48 | elric | http://pastebin.ca/9527 |
10:50.29 | Jax | so will anybody tell me what i need to do VoIP ? |
10:50.58 | elric | wait i pasted the wrong section |
10:51.31 | *** join/#asterisk pbxjunkie (~stormtroo@213.5.44.113) |
10:54.34 | malbech | Anyone konws a SoftSwitch Solution for a low cost ? |
10:54.50 | kajtzu | cost is relative |
10:54.50 | kajtzu | :) |
10:55.05 | malbech | of course ... :) |
11:01.48 | pbxjunkie | guys, if I want to connect the telco's S0 lines onto my PCI QuadBri card, which mode is that? NT right? |
11:02.18 | pbxjunkie | TE is if you want to connect ISDN telephones on the port, right?:) |
11:03.05 | Jax | nobody wants to tell me what i need for VoIP ;( |
11:04.29 | emitrax | how do I konw whether a phone is registered with asterisk or not? |
11:04.37 | Mavvie | sip show peers |
11:04.42 | Mavvie | well, if it is a sip phone |
11:04.46 | emitrax | yeah |
11:04.52 | emitrax | but the status is unmonitored |
11:04.58 | emitrax | what does it mean? |
11:05.09 | Mavvie | unmonitored means: not monitored (i.e. can be up, can be down, asterisk doesn't care) |
11:05.22 | emitrax | may I paste 3 lines in the chan? |
11:05.35 | Mavvie | means yuo haven't set the host= statement in the phones sip configuration on asterisk. |
11:06.08 | emitrax | i did |
11:06.13 | emitrax | host=dynamic |
11:06.31 | Mavvie | well, not a real hostname then. |
11:06.39 | cypromis | TE is for connecting to the NTBA |
11:06.45 | cypromis | NT is for emulating an NTBA |
11:06.54 | emitrax | so? ... I'm sorry but I don't understand... |
11:06.59 | cypromis | that is for pbxjunkie |
11:07.04 | Mavvie | try it with a real hostname |
11:07.09 | cypromis | egocentrism is useless in a irc channel |
11:07.11 | cypromis | :P |
11:07.24 | pbxjunkie | cypromis:D |
11:07.27 | emitrax | how do I put the hostname if they are phone ? |
11:07.57 | pbxjunkie | aand.. NTBA is? :D |
11:08.07 | cypromis | NTBA is the silly box you get from the telco |
11:08.54 | pbxjunkie | so to connect my quadbri to the silly box ..I neeed.. TE mode for the ports? NOT NT like I have now? |
11:09.08 | pbxjunkie | is that why all the leds are red?:) |
11:10.12 | pbxjunkie | brb , off to re-jumper my quadBRI :D |
11:11.16 | *** join/#asterisk wildgoose (~edward@xunil.mail.wildgooses.com) |
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11:18.08 | *** part/#asterisk Donuil (~cini_lab@217.9.64.213) |
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12:03.52 | *** join/#asterisk Jackthe (MyIdent@prac002.ewi.utwente.nl) |
12:03.52 | *** join/#asterisk ccfiel (ccfiel@210.5.72.36) |
12:03.52 | ccfiel | hello ppl.. |
12:06.05 | ccfiel | can somebody help me with my problem.. a remote iax connection connect to my * box ...and communicate with a local iax connection.. before a few second at least 10 seconds there is a lag in commnunication...after a while lag will disapper...what's would be the problem..can somebody help me with this? :( |
12:07.06 | tzanger | ccfiel: what version of asterisk |
12:07.29 | Jackthe | ccfiel: can you make a packettrace of the setup of the call? |
12:07.34 | *** join/#asterisk wildgoose (~edward@xunil.mail.wildgooses.com) |
12:07.42 | Jackthe | then look at the voicetimestamps |
12:08.25 | Jackthe | I think there will be a jump in the timestamp somewhere in the 1 second |
12:08.44 | tzanger | Jackthe: stevek and I hae been working on this problem |
12:08.57 | ccfiel | jackthe: what would be the solution for this? |
12:09.10 | tzanger | ccfiel: what version of asterisk? |
12:09.26 | ccfiel | wait..let me see.. |
12:09.53 | Jackthe | tzanger: great |
12:09.56 | Yellow_Fuzzy | hi |
12:10.02 | ccfiel | sorry the dump question..how can know the version of my asterik? |
12:10.15 | ccfiel | asterik/asterisk |
12:10.23 | tzanger | ccfiel: show version |
12:10.31 | Yellow_Fuzzy | any one here happen to have any experience with OzTell? |
12:10.31 | ccfiel | ok..thanks...wait..:) |
12:11.09 | *** join/#asterisk tzafrir (~tzafrir@62.90.10.53) |
12:11.20 | tzanger | Yellow_Fuzzy: no, they made me send back the ruby slippers |
12:12.51 | pgpkeys | hah! |
12:13.10 | Yellow_Fuzzy | lol |
12:13.47 | tzanger | you know, it amazes me just how well safety scissors cut paper and nothing else |
12:14.03 | *** join/#asterisk _THEEND_ (~DrEaM@80.18.184.226) |
12:14.13 | pgpkeys | oh now that's bad |
12:15.26 | *** join/#asterisk dizzydiffi (dizzydiffi@adsl-70-240-211-145.dsl.hstntx.swbell.net) |
12:15.30 | dizzydiffi | hello |
12:16.09 | dizzydiffi | has anyone compiled Asterisk 1.0.7 with OH323 |
12:16.28 | dizzydiffi | anyone alive in here |
12:16.43 | Mavvie | no, we just have been killed by a band of ninja turtles. |
12:16.54 | dizzydiffi | wow |
12:16.59 | dizzydiffi | turtles huh |
12:17.02 | pgpkeys | roll 2D20 for Ninja Master save |
12:17.07 | dizzydiffi | thats crazy |
12:17.15 | ManxPower | I really hate mornings |
12:17.18 | ManxPower | ~docs |
12:17.26 | 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 |
12:17.26 | tzanger | ninja master gets 2d20? |
12:17.26 | bugbot | docs is assigned nothing and reported nothing. |
12:17.26 | tzanger | damn |
12:17.26 | ManxPower | ~mailinglist |
12:17.27 | jbot | mailinglist is probably Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
12:17.27 | bugbot | mailinglist is assigned nothing and reported nothing. |
12:17.43 | tzanger | damn that reminds me of my AD&D days |
12:18.03 | pgpkeys | tzanger: i haven't gotten to the Magic Rat Tail adjustment yet |
12:18.09 | tzanger | hahaha |
12:18.16 | tzanger | rat tails... what a fashion statement |
12:18.22 | pgpkeys | hehe |
12:18.39 | tzanger | rat tails were the pre-mullet mullet |
12:18.57 | pgpkeys | the rodent version of afros? |
12:19.09 | tzanger | no that is what happens when you christmas tree a cat |
12:19.18 | pgpkeys | hjahahahahaha |
12:19.32 | pgpkeys | thank you very much. i almost spit up my meatballs when i read that |
12:19.40 | tzanger | meatballs? hwere are you at |
12:19.50 | tzanger | it's 8:20am here (EDT) |
12:19.55 | pgpkeys | buffalo, NY USA. my rendition of breakfast |
12:19.59 | tzanger | ahh okay |
12:20.01 | pgpkeys | last night's spagetti |
12:20.03 | tzanger | you're about 2.5h from me |
12:20.07 | Godsey | this is OT but might anyone use Illustrator? :) |
12:20.12 | tzanger | Godsey: not me |
12:20.21 | Godsey | I can't remember how to change the starting point of text to path tool once text is on it |
12:20.22 | pgpkeys | rochester? |
12:20.25 | tzanger | I love making a huge pot of spaghetti and then eating it cold for breakfast |
12:20.36 | tzanger | nah, Kitchener/Waterloo, ON |
12:20.40 | *** join/#asterisk Mimmus (~viggiani@ext.pitagora.it) |
12:21.15 | pgpkeys | ahh other side of the mud splatter |
12:21.16 | Mimmus | hi, I need a double E1 card to introduce Asterisk in my company, located in Italy |
12:21.17 | tzanger | and buffalo is *not* 2.5h from rochester |
12:21.39 | tzanger | Mimmus: I'd suggest a TE405P, not a pair of TE110Ps... lowre interrupt load and room to expand |
12:21.40 | pgpkeys | it's about that depending on if you go scenic or not ;) |
12:21.43 | Mimmus | I see that Digium offers only quad card. Is it only chance? |
12:21.50 | tzanger | Mimmus: pretty much, yeah |
12:22.05 | tzanger | sangoma offers a 2-port card but I've no experience with it |
12:22.07 | Mimmus | tzanger: and are themy chaper? |
12:22.42 | tzanger | well a 1-port Sangoma or Digium is $500. A 2-port Sangoma I think is $1000, and a 4-port Digium or Sangoma is $1500 |
12:23.08 | tzanger | but as I said with the 4 port you ahve room to grow iwthout buying additional hardware |
12:23.30 | Mimmus | tzanger: well, and now a more difficult question: if I use Digium ,will I have more advantages (support, etc) |
12:23.59 | Mimmus | I mean help on this channel, on the mailing list, etc |
12:24.16 | *** join/#asterisk Betu| (~betul@62.244.193.101) |
12:24.21 | tzanger | Mimmus: well everyone here tries to help, and several of us are quite good at it. Digium has official support included in the price of their cards, plus there's the warm fuzzy feeling of helping out the people who wrote Asterisk |
12:24.22 | blitzrage | Mimmus: IRC and mailing list are not official Digium support channels |
12:24.26 | tzanger | morning Betu| |
12:24.40 | blitzrage | Mimmus: just like tzanger said :) |
12:24.44 | tzanger | blitzrage: what'd your name go for? |
12:24.49 | blitzrage | tzanger: $7.00! |
12:24.53 | tzanger | blitzrage: ha |
12:24.55 | blitzrage | hehehe |
12:24.57 | tzanger | hardly worth the auction |
12:24.59 | elric | does anyone have an example / snip of extensions.conf with MachineDetect() used in it? theres naught on the wiki about it |
12:25.04 | tzanger | blitzrage: you like my name this morning? |
12:25.06 | Mimmus | blitzrage: I know but a good 'open' support is better sometime :-) |
12:25.07 | blitzrage | tzanger: the auction only cost like .45 or something |
12:25.19 | tzanger | elric: where's MachineDetect? |
12:25.25 | blitzrage | tzanger: yah, I laughed when I saw it :) |
12:25.32 | blitzrage | tzanger: but alas, I have no prune juice |
12:25.48 | blitzrage | tzanger: I'll be biking all over Mississauga today... probably like 30 km's or osmething like that |
12:25.53 | elric | tzanger, http://www.thenetbrain.com/files/app_machinedetect.c |
12:26.03 | tzanger | blitzrage: nice. I used to do that |
12:26.10 | blitzrage | tzanger: yah, its for work too :) |
12:26.11 | tzanger | blitzrage: I biked from listowel to kitchener and back |
12:26.22 | blitzrage | tzanger: yah, thats a good little trek :) |
12:26.33 | elric | i got the link off www.voip-info.org |
12:26.38 | blitzrage | anyways, gotta shower and get breakfast, then head down to the GO |
12:26.40 | blitzrage | lates |
12:26.51 | tzanger | blitzrage: bike courier? |
12:27.13 | tzanger | anyway I should get into work too |
12:27.14 | blitzrage | tzanger: working on the phone systems at several CIBC branches |
12:27.18 | tzanger | talk to y'all in a bit |
12:27.22 | tzanger | blitzrage: asterisk-related? |
12:27.29 | blitzrage | tzanger: unfortunately not |
12:27.32 | tzanger | blitzrage: :-) |
12:27.34 | blitzrage | just alarm system's |
12:27.38 | tzanger | ahh oaky |
12:27.39 | tzanger | ttyl |
12:27.41 | blitzrage | lates |
12:27.50 | _THEEND_ | b |
12:27.55 | elric | http://pastebin.ca/9527 <--- is a snip of my extensions.conf |
12:28.14 | pgpkeys | cibc central intelligent bueracracy channel? |
12:30.09 | *** join/#asterisk zotz (~zotz@24.231.32.109) |
12:31.15 | ccfiel | tzanger,jackthe: my * version is : localhost*CLI> Asterisk CVS-HEAD-12/16/04-07:57:25 built by root@localhost.localdomain on a i686 running Linux |
12:31.28 | ccfiel | is this the one? |
12:31.48 | ManxPower | I just announced 2 bounties |
12:32.33 | Mimmus | I'd like to integrate Asterisk with an existing Alcatel PBX... I'm planning to put * in front of Alcatel. Is it a goo didea? |
12:32.48 | Mavvie | Mimmus: yes |
12:32.55 | Mavvie | Mimmus: that's what we have here, in front of an A4400 |
12:33.28 | Mimmus | Mavvie: the schema should be: PSTN --- PRI --- Asterisk --- PRI --- ALcatel |
12:33.33 | Mavvie | yes |
12:33.57 | Mimmus | Mavvie: analog phones will remain connected to Alcatel |
12:34.07 | cypromis | ManxPower: announcing bounties is for sure more positive than bitching at newbies |
12:34.10 | cypromis | ;) |
12:34.46 | Mimmus | Mavvie: will it be difficult to configure Asterisk to route all calls to *? |
12:34.53 | ManxPower | cypromis, but much less rewarding. |
12:34.58 | cypromis | hehe |
12:35.00 | cypromis | true |
12:35.01 | cypromis | ;) |
12:35.12 | ccfiel | jackthe: still there? |
12:35.15 | ManxPower | Bounty 1: generate an errror if you try loading BOTH digium card modules and ztdummy |
12:35.21 | Mavvie | Mimmus: exten => _.,1,Dial(Zap/${TRUNK_A4400}/${EXTEN}) |
12:35.38 | ManxPower | Bounty 2: generare an error if Hangup is run in exten => |
12:35.54 | ManxPower | Maveric, Why do you want to dial all calls twice? That's what _. will do. |
12:36.04 | ccfiel | can somebody help me with my problem.. a remote iax connection connect to my * box ...and communicate with a local iax connection.. before a few second at least 10 seconds there is a lag in commnunication...after a while lag will disapper...what's would be the problem..can somebody help me with this? :( |
12:36.16 | Mavvie | ManxPower: to forward it to a different PRI. |
12:36.23 | elric | can someone look at http://pastebin.ca/9527 and tell me if I am doing something wrong? |
12:36.42 | Mimmus | Mavvie: ah, ok! And if I want to use a set of extensions (starting with 6, for istance) to dial SIP phones, directly connected to Asterisk? |
12:36.44 | ManxPower | Maveric, So as soon as the call ends it will dial the exact same Dial line twice. |
12:37.07 | ManxPower | DON'T USE _.!!!!!!!!! |
12:37.11 | Mavvie | ManxPower: I don't see that behaviour here. |
12:37.39 | ManxPower | Mavvie, Maybe you have an exten => h also. That would cause it not to loop. |
12:38.01 | Mavvie | true. |
12:38.17 | ManxPower | It's still a bad idea. |
12:39.12 | ManxPower | Anyone want to comment on the two bounties? |
12:39.52 | bjohnson | arhg .. we be hunting bounties today mate |
12:42.57 | Mimmus | Can I use Asterisk to setup a dialup server? |
12:43.04 | ManxPower | Mimmus, no |
12:43.23 | ManxPower | if by "dialup" you mean "analog modem dialup" |
12:43.41 | Mimmus | ManxPower: ok, it was only an idea |
12:44.31 | cypromis | you could do an isdn dialup |
12:44.34 | cypromis | but not a analog dialup |
12:44.56 | azid | voicemail-question: if a caller hangup during the vm welcome-message i always end up with a 0-legth message in my VM, anyone know why? (PSTN call over SIP) |
12:46.17 | ManxPower | azid, Report it as a bug |
12:46.36 | basta | anyone using cisco 7912/7960 with sip ? music on hold/forward stopped working and I can't understand why |
12:46.41 | ManxPower | There should already be a bug report about this, but I don't know the bug # |
12:47.00 | azid | manxpower, ok. i was hoping i did something wrong in my config-files :( |
12:47.26 | ManxPower | basta, I have seen MoH during a transfer not working, but not music on hold during a forward. |
12:48.10 | *** join/#asterisk bmg505 (~leon@rndf-146-59-117.telkomadsl.co.za) |
12:48.20 | basta | yes, i mean forward |
12:48.49 | *** join/#asterisk jbAU (~johnblade@c210-49-42-214.rochd2.qld.optusnet.com.au) |
12:48.56 | ManxPower | then I've not seen it. What kind of forward? Dialplan forward, or a 302 Moved SIP forward? |
12:49.27 | basta | no, sorry, i mean i mean a transfer (need a holiday) |
12:49.42 | ManxPower | basta, Yes, I've experienced that. |
12:49.48 | basta | any idea ? |
12:49.57 | ManxPower | basta, no, I never fixed it. |
12:50.21 | basta | x| |
12:50.22 | ManxPower | MoH works when the call is on hold, but not when the calls has been transfered and is runing. |
12:50.33 | ManxPower | ringing |
12:51.44 | basta | if you do a direct transfer transferred person hears ringing, which is good since she has feedback on the call beying transferred |
12:52.23 | basta | but on a hold she hears nothing, and sometimes they hangup, thinking the connection went lost |
12:52.37 | basta | manx, what version of * ? |
12:53.07 | basta | maybe it stopped working after an upgrade to 1.0.6 for me |
12:57.26 | jakepdev | anyone working with * outcalling? |
12:58.02 | jakepdev | does * succesfully detect disconnected/busy/no answer, etc? |
13:02.30 | *** join/#asterisk jeffik (jefik@69.158.30.24) |
13:02.45 | *** join/#asterisk delYsid (~user@delYsid.developer.debian) |
13:02.59 | delYsid | yay, my asterisk setup now works! |
13:03.47 | tzanger | excellent |
13:03.56 | *** join/#asterisk ckruetze (ckruetze@cpc3-cmbg7-5-0-cust100.cmbg.cable.ntl.com) |
13:04.54 | delYsid | sip proxy directly calls in to my 6970g |
13:05.06 | delYsid | I wonder though, how can I implement remote mailbox access now? |
13:05.37 | delYsid | er, 7960g |
13:07.25 | *** join/#asterisk iq (~iq@70-59-161-91.omah.qwest.net) |
13:08.10 | ManxPower | jakepdev, Analog ports do not support detecting disconnected/busy/no answer |
13:08.29 | ManxPower | well, it can have a timeout, of course. |
13:10.52 | tzanger | ManxPower: well they can detect disconnect but not busy/no answer |
13:11.50 | ManxPower | tzanger, Huh? |
13:12.01 | ManxPower | ANALOG FXO? |
13:12.06 | jakepdev | http://www.voip-info.org/wiki-NVLineDetect ??? |
13:12.09 | ManxPower | Nuh uh! |
13:12.34 | ManxPower | jakepdev, That's a 3rd party module and not part of Asterisk. I do not know how well it works. |
13:12.46 | jakepdev | k |
13:13.05 | jakepdev | tnx |
13:14.11 | ManxPower | just get a non-analog connection like VoIP or PRI |
13:14.36 | jakepdev | that may be a possibility - it's new client |
13:14.51 | jakepdev | through PRI - * can detect intercept tones? |
13:15.06 | jakepdev | busy, no answer etc? |
13:15.19 | jakepdev | without NVLineDetect? |
13:16.28 | ManxPower | jakepdev, yes. |
13:17.00 | jakepdev | great! I'll push er. guide them in that direction |
13:18.13 | Gand_DJ | What's the best codec to use for doing voip over 28.8k dialup? My fiance still has dialup in her area and I want to setup a softphone on her pc to link to my * server. |
13:18.18 | tzanger | ManxPower: yes |
13:18.42 | tzanger | ManxPower: CPD is certainly available IF YOUR TELCO OFFERS IT. Bell Canada does battery reversal, and the FXO modules can see it |
13:18.43 | vaewynAFK | Gand_DJ: g.729 or ilbc or sometimes GSM |
13:18.46 | ManxPower | Gand_DJ, none |
13:18.58 | vaewyn | Gand_DJ: and don't expect it to work |
13:19.15 | vaewyn | 56k you have a prayer... 28.8 good luck |
13:19.20 | Gand_DJ | heh |
13:19.32 | ManxPower | tzanger, CPD is not the same as "disconnected number" |
13:19.54 | tzanger | ManxPower: oh I thought you meant disconnected as in hung up |
13:19.57 | tzanger | my apologies |
13:19.58 | Gand_DJ | doesn't g729 use 8kbps? that's only 1/3 of 28.8k dialup. |
13:20.09 | tzanger | those are all inband audio and the callprogress just doesn't cut it at this point |
13:20.16 | ManxPower | Gand_DJ, You then have UDP overhead. |
13:20.27 | tzanger | ManxPower: why should you not call hangup from h exten? |
13:20.33 | tzanger | (looking at your bounty post) |
13:20.42 | ManxPower | tzanger, Mostly because it's stupid. |
13:20.47 | tzanger | ManxPower: so? |
13:20.48 | vaewyn | ok... the Norhell is on crack... hooked the TE405P into our 5300 that is configured to talk to the Norhell... and voila... I have a connection... plug it into the Norhell... and nothing |
13:20.52 | tzanger | ManxPower: there are lots of stupid things people do |
13:21.13 | ManxPower | But also it will help generate errors when you accidently have _. that calls hangup |
13:21.21 | tzanger | I think it's far more of an issue that asterisk doesn't generate errors if you say "FunkyConfigParamThatDoesntExist=10e-98" |
13:21.21 | ManxPower | tzanger, Notice it's only a $10 bounty |
13:21.45 | ManxPower | tzanger, post a bounty for that |
13:21.56 | tzanger | I think that a better bounty would be whenever _. is used it posts "ARE YOU FUCKING INSANE?" |
13:22.33 | tzanger | or better, whenever _. matches a "special" exten like s,h,i or t |
13:22.37 | vaewyn | tzanger: amen! |
13:22.38 | jakepdev | but has an override |
13:22.42 | jakepdev | like windows |
13:22.46 | jakepdev | "Are you sure..." |
13:22.51 | jakepdev | Are you really sure???" |
13:23.04 | vaewyn | "You do realize you are stupid?!?" |
13:23.05 | tzanger | actually _. would match o and a too |
13:23.36 | vaewyn | hm... what's 'a' hadn't heard of that one |
13:23.36 | tzanger | personally I like having Hangup in h |
13:23.46 | tzanger | 'a' is for voicemail when you hit '*' |
13:23.50 | vaewyn | 'o' I use way to much |
13:23.51 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
13:23.56 | vaewyn | ahhhh |
13:23.58 | vaewyn | asterisk |
13:23.59 | vaewyn | hehehe |
13:24.17 | tzanger | you use Hangup when you want to make deadly sure that that's as far as that extension gets. And if I have post-hangup stuff to do, I still end it with Hangup |
13:24.37 | *** join/#asterisk moy (~kvirc@201.135.105.124) |
13:25.14 | *** join/#asterisk blop (~blop@2001:6f8:204:33:bbbb:bbbb:bbbb:bbbb) |
13:25.16 | blop | hi :) |
13:26.19 | Gand_DJ | Non * related question.. anyone know what kind of codec msn & yahoo messenger use for audio connections? |
13:28.34 | *** join/#asterisk funxion (~chatzilla@mtnuser.icgws.com) |
13:28.51 | funxion | |vulture| are you here? |
13:29.24 | funxion | anyone |
13:29.27 | funxion | anyone |
13:29.31 | funxion | bueller? |
13:29.31 | mistral | i have astrisk running here connected with a sip provider i seem to be able to dail out to voice mail and a talking clock but cannot dail full numbers. Also when i get an incoming call astrisk says something about no compatible codecs anyone know why this is ? |
13:30.14 | funxion | what codec are they connecting with? |
13:30.26 | mistral | in which connection ? |
13:30.31 | funxion | inbound |
13:30.32 | inspired | can asterisk 1.0.7 play gsm files in musiconhold? |
13:30.36 | inspired | if so, how? |
13:30.38 | mistral | dont know |
13:30.57 | mistral | its actually an incoming call from a land line |
13:31.10 | funxion | how is it hitting asterisk |
13:31.15 | funxion | fxo? |
13:31.38 | mistral | no there is a internet sip provider that allows incoming calls to a land number |
13:31.55 | mistral | as in the internet sip provider gives your a land line number |
13:32.13 | *** join/#asterisk phpboy (~sj@tpr-165-239-114.telkomadsl.co.za) |
13:32.18 | phpboy | hello my loves |
13:32.39 | mistral | i can get it to work if i get x-lite soft phone to talk directly to the sip provider and it uses codec g711u |
13:33.06 | foobos | mistral, well force it ulaw then in asterisk. it prolly tries g729 and you don't have licesnse |
13:33.17 | funxion | do you have g711u allowed |
13:33.17 | mistral | how do i force it ? |
13:33.32 | funxion | allow it in sip.conf |
13:33.45 | funxion | make it a higherpriority then g729 |
13:33.49 | phpboy | least cost call routing |
13:33.53 | foobos | mistral, in sip.conf disallow=all allow=ulaw |
13:33.56 | mistral | disallow=all |
13:33.56 | mistral | allow=ulaw |
13:33.56 | mistral | allow=alaw |
13:33.57 | phpboy | is this a possibility with asterisk |
13:33.57 | phpboy | ? |
13:33.59 | mistral | i have that |
13:34.19 | foobos | mistral, well then you can try "sip debug" in the commandline |
13:34.34 | Hmmhesays | phpboy: why wouldn't it be? |
13:34.48 | Hmmhesays | looks as though you are familiar with php, write an agi |
13:34.51 | funxion | anyone in here have a te110p installed? |
13:35.01 | phpboy | Hmmhesays: agi |
13:35.02 | phpboy | ? |
13:35.04 | jakepdev | funxion - 2 of them |
13:35.10 | Hmmhesays | ~agi |
13:35.12 | jbot | it has been said that agi is the Asterisk Gateway Interface... similar to CGI for web applications AGI lets you script call control and access databases using your favorite language. AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages |
13:35.12 | bugbot | agi is assigned nothing and reported nothing. |
13:35.13 | funxion | did you have any problems installing them |
13:35.24 | jakepdev | actually 3 |
13:35.28 | jakepdev | nope - no problems |
13:35.31 | funxion | hmm |
13:35.33 | funxion | what os? |
13:35.33 | jakepdev | fit right in |
13:35.46 | jakepdev | CentOS 4 |
13:35.49 | funxion | hmm |
13:35.57 | funxion | i have centos 4.3 |
13:36.07 | jakepdev | shouldbe fine |
13:36.12 | jakepdev | what errors? |
13:36.12 | funxion | getting modprobe error when I try to modprobe wcte11xp |
13:36.38 | funxion | I get /lib/modules/2.4.21-27.0.1.ELsmp/misc/wcte11xp.o: init_module: No such device |
13:36.46 | funxion | following by some other errors |
13:36.59 | jakepdev | no such device... hmm... |
13:37.07 | jakepdev | irq sharing? |
13:37.11 | funxion | |vulture| tried to help me fix it yesterday |
13:37.15 | funxion | not that I know of |
13:37.25 | jakepdev | try digium |
13:37.28 | funxion | I updated zaptel source |
13:37.32 | jakepdev | they offer free hardware install support |
13:37.39 | jakepdev | 877-linux-me |
13:37.42 | funxion | recompiled then tried to load modules again |
13:37.44 | funxion | yeah |
13:37.45 | funxion | kewl |
13:37.48 | funxion | didnt know about that |
13:37.50 | funxion | thnx |
13:37.52 | jakepdev | np |
13:38.10 | ManxPower | funxion, does lspci show the card? |
13:38.14 | funxion | yes |
13:38.45 | funxion | 03:01.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
13:40.05 | Gand_DJ | Having trouble decide on a voip provider to handle my * outbound. I'd prefer someone who has access to cdn DID, but not needed (I have a pots line ppl can call me on) |
13:40.15 | jakepdev | Gand - try a few |
13:40.19 | jakepdev | they're cheap |
13:40.26 | delYsid | I have exten => 666,2,Voicemail(u666) in my extensions.conf and the mailbox works if no one answers, but how can I do remote mailbox access? |
13:41.21 | jakepdev | they're is no one size fits all VOIP provider |
13:41.27 | jakepdev | therer |
13:41.31 | jakepdev | ah f it |
13:41.34 | RestLessGemini | does anyone knows about any modem other Intel Ambiant, which can be used as fxo/fxs |
13:41.56 | vaewyn | no "modem" can be used as FXS |
13:41.56 | jakepdev | ya cheap bastard :) Just get a real FXO/FXS |
13:42.10 | jakepdev | the SPA 300 goes for $100 |
13:42.13 | jakepdev | 3000 |
13:43.10 | RestLessGemini | jakepdev: i am sitting in pakistan and I'm using Intel Ambiant, and the bad thing is, they are short in the market :) |
13:43.24 | jakepdev | i hear ya |
13:43.25 | ManxPower | Some of the other SPAs can be $65 or less |
13:43.42 | ManxPower | RestLessGemini, The ambient chipset is no longer made. |
13:43.59 | *** join/#asterisk ccfiel (ccfiel@210.5.72.36) |
13:44.06 | ccfiel | hello ppl.. |
13:44.22 | ccfiel | can somebody help me with my problem.. a remote iax connection connect to my * box ...and communicate with a local iax connection.. before a few second at least 10 seconds there is a lag in commnunication...after a while lag will disapper...what's would be the problem..can somebody help me with this? :( |
13:44.29 | RestLessGemini | ManxPower: Yes i know, this is why I asked if any of you knows about any other compatable card like IA |
13:44.39 | ManxPower | RestLessGemini, No. there are none. |
13:44.51 | RestLessGemini | ManxPower: thanks |
13:45.48 | ccfiel | hello can somebody help me.. :( |
13:46.43 | ManxPower | ccfiel, It sounds like a router or ISP problem. |
13:48.26 | ccfiel | manxpower..i thinks it's not...because when a iax remote connection connect to my * and call a sip local extension..there is no lag at all.. |
13:48.38 | Gand_DJ | The main ones that have my attention are broadvoice, link2voip, and teliax. |
13:48.55 | ManxPower | ccfiel, Your problem is very strange and I've never heard of such a problem that was not caused by a network problem |
13:49.31 | ccfiel | this happen with iax remote connection --> * server ----> iax local extension |
13:49.52 | ccfiel | but when i do ... iax remote connection --> * server ----> sip local extension it works well.. |
13:50.46 | ManxPower | then the only difference is the sip local connection |
13:51.04 | ManxPower | Softphones are well known for causing latency problems |
13:51.21 | ManxPower | ~google site:lists.digium.com softphone latency |
13:51.22 | bugbot | google site:lists.digium.com softphone latency is assigned nothing and reported nothing. |
13:51.52 | jakepdev | Gand - I tried 4 before I came to a conclusion - maybe cost me abut $20 |
13:52.03 | Gand_DJ | Who you currently use? |
13:52.06 | jakepdev | but was worth it |
13:52.34 | jakepdev | in my area, voicepulse provided the best results |
13:52.43 | jakepdev | but it could be different for you |
13:52.57 | jakepdev | voicepulse just happens to have a POP close to here |
13:53.15 | Gand_DJ | I live in canada, so most will probably be similiar..lol |
13:53.22 | Gand_DJ | I found one called voipforcanada.com |
13:53.29 | Gand_DJ | but the site layout makes me wonder about it |
13:53.37 | tzanger | I use nufone almost exclusively (I'm canadian) |
13:53.44 | tzanger | push about 5000 min/mo through them |
13:53.45 | ManxPower | I use Teliax now. |
13:53.50 | jakepdev | easiest way to find out is to try it |
13:54.00 | jakepdev | make a few test calls |
13:54.18 | Gand_DJ | I've thought of looking at nufone but they are still doing upgrades.. heh |
13:54.21 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
13:54.34 | tzanger | Gand_DJ: msg shido6 and he can set you up |
13:54.36 | jakepdev | if you pay via PayPal, they should accept you as a new usr |
13:54.44 | vaewyn | They rock if you can stand the wait |
13:55.00 | tzanger | jakepdev: unfortunately a few test calls will only reveal if they're poor at that time... mind you the poor ones, if they are truly poor will be crap all the time |
13:55.33 | jakepdev | right - but if you space the test calls apart... |
13:55.47 | jakepdev | over a few days, different times, etc |
13:55.48 | Gand_DJ | well, before I signup with someone (such as nufone), I'd want to read their rates & fees... and none of that is on their site currently. |
13:56.14 | *** join/#asterisk zoa (~zoa@pirus.securax.be) |
13:56.21 | tzanger | Gand_DJ: their rates are available on the main site |
13:56.28 | jakepdev | http://www.nufone.net/rates.csv |
13:56.35 | tzanger | fees are just their rates; there is no signup fee, there is no strange shit |
13:56.43 | vaewyn | 2cents/min USA and such |
13:56.45 | tzanger | push as many concurrent calls through them as you can handle, it's all the same per-minute |
13:56.58 | tzanger | vaewyn: yeah but cdn term through nufone is slightly cheaper :-) |
13:57.13 | vaewyn | yep |
13:58.01 | Gand_DJ | What about connection fees, taxes, etc. (read teliax charges 2c connect fee per pots call) |
13:58.19 | [TK]D-Fender | http://www.voipforcanada.com < $.01 min CDN for outbound |
13:58.26 | Gand_DJ | Seems some of the voip providers want to charge you state taxes, even though it's illegal. |
13:58.37 | nvrswork | has anyone successfully gotten asterisk 1.0.7 gentoo port compiled on 2005.0? |
13:59.29 | Gand_DJ | [TK]D-Fender, yeah I like that part.... but the site makes me wonder how trustworthy they are. alot of stuff doesn't work for links. |
13:59.56 | [TK]D-Fender | Gand_DJ : I noticed a lack of finish as well, but for the price, looks worth it. |
13:59.57 | Gand_DJ | I have a free / test acct through them... also with simpletelecom right now |
14:00.03 | [TK]D-Fender | esp for a trial |
14:00.42 | Gand_DJ | haven't setup the v4c acct yet as they only emailed me softphone settings.. nothing for * setup |
14:00.59 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
14:04.29 | Gand_DJ | hrm... what's the difference between getting a DID and leasing a DID? |
14:04.37 | Gand_DJ | either way you don't own it. |
14:04.41 | Gand_DJ | I think |
14:05.14 | *** join/#asterisk malbech (Phils@m199.net81-66-243.noos.fr) |
14:06.07 | delYsid | ahh, 'a' was what I needed |
14:09.41 | *** join/#asterisk zaptel (~just@216.194.173.2) |
14:12.14 | *** join/#asterisk wazquis (~akv@lnxbx.dk) |
14:12.28 | *** join/#asterisk roamer323 (~sing@toronto-HSE-ppp4074896.sympatico.ca) |
14:13.47 | wazquis | hey, is it possible to send a text message from one asterisk to an other? |
14:16.11 | moy | does anybody knows how can i make that IAX2 use other port than 4569?? |
14:16.29 | moy | i tryied modifying iax.conf port parameter, but does not work |
14:18.04 | *** join/#asterisk bannerman (~bannerman@209.216.176.42) |
14:19.32 | *** join/#asterisk fugitivo (~ajf@201.255.105.220) |
14:19.35 | fugitivo | hello |
14:21.25 | *** join/#asterisk gmcinnes (~gmcinnes@Toronto-HSE-ppp3681363.sympatico.ca) |
14:22.07 | ManxPower | moy, you have to modify the asterisk source to change the port from 4569 |
14:22.17 | bjohnson | Gand_DJ: some voip providers allow you to transfer out your DID .. others do not |
14:22.32 | bannerman | shido6: Did you guys make some changes that allowed G.729 to start working? Because, this morning, it's working fine... |
14:22.56 | Gand_DJ | bannerman, you using firefly for g729? |
14:22.59 | bannerman | no |
14:23.10 | ManxPower | I didn't think Nufone supported G729 |
14:23.17 | bannerman | They do. |
14:23.23 | ManxPower | officially? |
14:23.31 | Gand_DJ | They might not support transcoding.. but maybe passthrough |
14:23.45 | ManxPower | Gand_DJ, You can't pass thru to the PSTN |
14:24.52 | gmcinnes | Anyone used asterisk with a Nortel Meridian Option 61-C switch? |
14:24.55 | Gand_DJ | Right, but does nufone directly link to pstn, or they link to another backend voip that goes direct to pstn? |
14:28.38 | *** join/#asterisk carlos-d-man (~carlos@201.135.87.60) |
14:28.41 | malbech | I search a softswitch for a good price but it's very diificult to find one ... no ? |
14:28.43 | carlos-d-man | hi guys |
14:30.18 | *** join/#asterisk girabraz (~christian@200.121.129.178) |
14:31.15 | *** join/#asterisk olivier_ (~olivier_@obs92-4-82-239-116-113.fbx.proxad.net) |
14:32.07 | *** join/#asterisk Dovid (~hirisk@pool-138-89-169-188.mad.east.verizon.net) |
14:33.01 | *** join/#asterisk twisted[work] (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
14:33.01 | *** mode/#asterisk [+o twisted[work]] by ChanServ |
14:33.11 | jalsot | hi |
14:33.49 | jalsot | does anybody know what can be the problem with IAX2->*->*->PSTN while SIP->*->*->PSTN works fine? |
14:34.42 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
14:34.42 | *** mode/#asterisk [+o bkw_] by ChanServ |
14:35.26 | bannerman | I think they officially do. Jeremy and Greg both said that they have digium G.729 licenses. |
14:36.01 | Dovid | can anyone tell me the diffrence in version 1.0.6 to 1.0.7 ? |
14:37.23 | bannerman | 0.0.1 |
14:37.27 | bannerman | ;-) |
14:38.05 | Dovid | hehe |
14:38.07 | Dovid | seriousley |
14:38.17 | bannerman | Gand_DJ: Pretty sure Nufone does their own termination. |
14:38.41 | robl^ | I've used g729 with Nufon |
14:38.54 | bannerman | robl:^: Have you been using it this last week? |
14:38.55 | Gand_DJ | hrm.. ne1 here have asterisk@home working with fwd for incoming? I keep getting busy signal when calling into by * box. |
14:39.02 | carlos-d-man | how may I connect dialx or any other sip phone to asterisk? |
14:39.11 | Gand_DJ | I can finally call out using fwd. |
14:39.44 | robl^ | bannerman, no. I've switched to gsm for Nufone. I only use g729 now for one remote extension on a low bandwidth connection |
14:39.52 | inspired | is it safe to use mpg123 for music on hold? |
14:40.04 | inspired | I can't get the fake mpg123 script to work |
14:40.58 | robl^ | inspired, I've used mpg123 for more than a year now. I have little trouble with it. Asterisk used to leave it running when you stopped, but not anymore |
14:41.23 | inspired | ok |
14:41.34 | ManxPower | Dovid, It's all listed in the asterisk-cvs mailing list (only messages with Tag 1-0 apply to 1.0.x) . There is also a changelog included in the tarball |
14:42.02 | ManxPower | ~mailinglist |
14:42.03 | jbot | hmm... mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
14:42.03 | bugbot | mailinglist is assigned nothing and reported nothing. |
14:42.14 | ManxPower | bugbot needs to be melted down. |
14:42.14 | inspired | is it possible to convert a .raw file back to .mp3? |
14:42.27 | Dovid | is it stable ? |
14:42.27 | ManxPower | inspired, That is not an Asterisk question. |
14:42.36 | robl^ | inspired, the worst thing with mpg123 is that it uses a bit of memory and cpu.. but not bad unless you have a large number of calls on a server |
14:42.41 | ManxPower | Dovid, Is what stable? |
14:43.05 | Dovid | 1.0.7 |
14:43.28 | ManxPower | Dovid, There have been some reports of SIP issues with 1.0.7 but I've not seen any major issues with 1.0.7. |
14:43.35 | johnnyb | Is there a timeline for releasing 1.1? |
14:43.35 | ManxPower | Dovid, You don't read the mailing lists, do you? |
14:43.48 | Dovid | i get them, dont have time to read em |
14:43.51 | robl^ | Dovid, version 1.0.x is ALL considered stable. |
14:43.55 | ManxPower | johnnyb, sometime before the sun burns out and the universe implodes. |
14:44.07 | johnnyb | ManxPower: so, like, this year? |
14:44.19 | ManxPower | Dovid, but you have time to ask OTHER people to read them for you and then distil the information for you. |
14:45.07 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfj1c.dialup.mindspring.com) |
14:45.26 | pgpkeys | OH NO! THE UNIVERSE IS IMPLODING?? Man! now they tell me, and i just bought a new planet for the kids too. |
14:45.28 | pgpkeys | damn it! |
14:46.03 | ManxPower | I SUSPECT 1.1 will be released in 6 - 9 months, but I have no proof. |
14:46.05 | *** join/#asterisk IronHelix (~irc@ool-182c3fe9.dyn.optonline.net) |
14:46.07 | robl^ | pgpkeys, they have a 30 day return policy. just a 15% restocking fee |
14:46.23 | file[laptop] | ManxPower: proof pfft, who needs that?!? |
14:46.26 | pgpkeys | well then that's fine. |
14:46.37 | inspired | robl^: is there a way to do music on hold without using so much cpu? |
14:46.40 | fugitivo | anyone is using cisco ip phones? |
14:46.43 | ManxPower | file[laptop], Well, I have plenty of poof, but no proof. |
14:46.45 | pgpkeys | that's a measley $1,500,000,000 for restock. i can deal with that |
14:48.22 | robl^ | inspired, really, the CPU hit is not bad. I have never had any performance problems. one tip is to use only one moh class. if you have it playing multiple playlists concurrently, you use more CPU and memory |
14:50.39 | *** join/#asterisk _THEEND_ (~DrEaM@80.18.184.226) |
14:51.00 | _THEEND_ | which fax software is better with asterisk? |
14:51.40 | pgpkeys | the one you write |
14:51.53 | _THEEND_ | ? |
14:51.56 | _THEEND_ | sorry? |
14:51.56 | vaewyn | hylafax or spandsp |
14:52.07 | _THEEND_ | i'm looking for hylafax |
14:52.10 | inspired | robl^: ah, I see |
14:52.18 | pgpkeys | _THEEND_: my fault. I read taht as better THAN not better WITH |
14:52.54 | _THEEND_ | :) |
14:53.29 | _THEEND_ | but Hylafax interfacing with asterisk it's less documentated |
14:55.34 | ManxPower | hylafax requires a modem |
14:56.19 | _THEEND_ | also only for receving fax? |
14:56.50 | *** join/#asterisk veto (~upirc@015-823-051.area5.spcsdns.net) |
14:58.08 | *** part/#asterisk veto (~upirc@015-823-051.area5.spcsdns.net) |
14:58.40 | *** join/#asterisk vooduhal (~christoph@67.19.25.178) |
15:00.11 | vooduhal | Hey, can anyone answer a few res_perl questions? Specifically, where are the channel status constants and what is the proper way of replacing ast_readstring since I can seem to find it in res_perl? |
15:01.15 | _THEEND_ | hmm |
15:01.38 | ManxPower | I want a nap. |
15:01.49 | _THEEND_ | but if i have an hfc-s chip based isdn card for calling with isdn can i use that card as fax modem? |
15:02.09 | vooduhal | NM, the channel state problem I just found where they are defined. |
15:05.19 | *** join/#asterisk eivindtr (~eivindtr@062016241059.customer.alfanett.no) |
15:06.23 | gmcinnes | Hi, all you helpful little elves ;) |
15:06.41 | gmcinnes | does anyone know anything about Meridian 1 systems? |
15:06.52 | gmcinnes | *ouch |
15:07.27 | gmcinnes | I know its blasphemous to ask about such things on an asterisk channel :) |
15:08.06 | *** join/#asterisk pointer (pointer@aj.catt.com) |
15:08.07 | Moonwick | only if it's not in the context of "how do I connect asterisk to it in order for asterisk to eventually replace it" :) |
15:08.23 | gmcinnes | Moonwick: that's it exactly :) |
15:09.51 | gmcinnes | I need to know what kind of signalling it can do, and whether I can connect a TDM400P card to it without melting the card. |
15:11.50 | gmcinnes | ah. the silence is deafening :) |
15:12.49 | Moonwick | PBX knowledge is hard to come by in here unless you're lucky |
15:13.02 | Moonwick | fwiw, I doubt you'd melt the card |
15:13.17 | tzanger | mcnobody: what are you trying to do? |
15:13.22 | Moonwick | but beyond that, I'm not sure how much cooperation you'll be able to coerce out of the two systems |
15:13.30 | ManxPower | I don't give a flying rat about the brand of PBX. If you tell me what port types you have available, I can tell you what might work. |
15:13.34 | tzanger | you can't use a TDM card with a Norstar system unless you're going through an ATA |
15:13.48 | jakepdev | what's the easist way to do a simple lookup table in the * dialplan? i.e. if IN=1, set OUT =9 |
15:13.56 | ManxPower | tzanger, unless you use FXS ports on the TDM card and connect them to the CO ports on the PBX |
15:14.06 | tzanger | ManxPower: well yes you could do that too :-) |
15:14.09 | ManxPower | jakepdev, gotoif |
15:14.16 | *** part/#asterisk pointer (pointer@aj.catt.com) |
15:14.27 | gmcinnes | tzanger: what's an ATA ? |
15:14.47 | tzanger | norstar ATA or ATA2 |
15:14.57 | ManxPower | gmcinnes, it sort of converts the nortel phone ports into an analog port. notice the "sort of" |
15:14.57 | tzanger | lets you connect a regular phone/fax as an extension |
15:15.04 | *** part/#asterisk sympad (~Misha@195.138.127.98) |
15:15.17 | ManxPower | I was never able to have nortel digital phones to send DTMF out an ATA |
15:15.19 | tzanger | I have asterisk between our Norstar MICS and the PRI |
15:15.22 | tzanger | works well |
15:15.33 | ManxPower | tzanger, Yeah, well that's the RIGHT way to do it. 8-) |
15:15.33 | jakepdev | ManxPower - that creates a ton of branches. I'm looking for something I can do in a single line if possible |
15:15.37 | tzanger | ManxPower: intersting; I've never had to try that. it does work the opposite way :-) |
15:15.42 | ManxPower | jakepdev, can't be done. |
15:16.04 | ManxPower | tzanger, yes, but the ata ALSO doesn't seem to provide CPD |
15:16.10 | tzanger | ManxPower: no it doesn't |
15:16.19 | gmcinnes | I built an ivr on asterisk. It only needs to recieve incoming dtmf. It was based on being connected to POTS lines, but now they want it on a DID from their Option 61 C switch |
15:16.32 | tzanger | I've heard rumour that you should be able to detect a 'D' tone on hangup but have not played with it |
15:16.40 | ManxPower | gmcinnes, use a PRI port on the Nortel if you can |
15:17.07 | vooduhal | So anyone with ideas on the proper replacement for ast_readstring in res_perl? |
15:17.14 | tzanger | ManxPower: will a dialogic card not work? Ithought that was an option (an expensive one) |
15:17.18 | gmcinnes | ManxPower: And something like a TE410 on the asterisk box? |
15:17.27 | tzanger | gmcinnes: well a TE110P would work just fine |
15:17.30 | ManxPower | gmcinnes, yes. |
15:17.56 | gmcinnes | ManxPower: but there's no easy way to connect a DID to a TDM400P ? |
15:18.05 | ManxPower | The best way is Telco(T-1/PRI)<-><T-1/PRI)PBX(T-1/PRI)<->(T-1/PRI)Asterisk |
15:18.13 | tzanger | gmcinnes: with an ATA or ATA2, but you won't know if it works properly without testing |
15:18.24 | gmcinnes | tzanger: Forgive my dense-ness. This pbx stuff is all new to me. |
15:18.38 | tzanger | ManxPower: well not necessarily |
15:18.38 | ManxPower | gmcinnes, yes, but you will have all sorts of problems unless you are a telco, asterisk, and Nortel expert. |
15:18.42 | pgpkeys | hehe i'm still learning to set mine up |
15:18.45 | gmcinnes | ManxPower: I am none of the above. |
15:18.46 | pgpkeys | never set up a pbx before |
15:18.47 | ManxPower | tzanger, Well for some values of "best" |
15:18.51 | ManxPower | 8-0 |
15:19.01 | tzanger | ManxPower: the Norstar PRIs will not work as extensions without an expensive MCDN license key... and there is no support for MCDN or SL1 in libpri |
15:19.28 | tzanger | you can create a 3-digit or 4-digit extension and try ot route it out (I do that now) but it's hacky |
15:19.28 | ManxPower | An alternative is Telco(T-1/PRI)<->(T-1/PRI)Asterisk(T-1/PRI)<->(T-1/PRI)PBX |
15:19.36 | *** join/#asterisk UK_Mister (~fred@host81-137-167-89.in-addr.btopenworld.com) |
15:19.45 | UK_Mister | hello all |
15:20.15 | ManxPower | An alternative is Telco(T-1/PRI)<->(T-1/PRI)Asterisk(T-1/PRI)<->(T-1/PRI)ChannelBank(ANALOG)<->(ANALOG)PBX |
15:20.21 | *** join/#asterisk Rick_Hunter (~rhunter@05-136.008.popsite.net) |
15:21.03 | ManxPower | Reworded: In a Perfect World he best way is Telco(T-1/PRI)<-><T-1/PRI)PBX(T-1/PRI)<->(T-1/PRI)Asterisk |
15:21.06 | inspired | robl^: what bitrate should I use? |
15:21.12 | inspired | and sample rate? |
15:21.40 | ManxPower | Nortels do not exist in a Perfect World 8-) |
15:21.44 | gmcinnes | tzanger: does it matter that I don't ever need to route a call out from the asterisk box? |
15:21.44 | UK_Mister | i've been playing around with asterisk for a few weeks, i'm about to install it at one of our small sites, anyone recommend some good SIP handsets? |
15:22.11 | ManxPower | UK_Mister, Polycom IP 500 or SIPura SPA-841 |
15:22.27 | UK_Mister | thnx ManxPower |
15:22.44 | ManxPower | The Polycom IP 300 is you MUST have PoE. Otherwise use IP500's for people that need speakerphone, SPA-841 for people that don't need speakerphone. |
15:23.18 | UK_Mister | do they all have programable soft keys? |
15:23.32 | gmcinnes | ManxPower: no, they certainly don't. |
15:23.44 | *** join/#asterisk dasuberdavid (~david@207.111.174.1) |
15:24.28 | Gand_DJ | anyone here used Simpletelecom? so far 1 person doesn't like them.. |
15:24.50 | UK_Mister | i'm looking for some cheap and cheerfull single line display and a couple of executive with soft keys etc |
15:25.56 | ManxPower | UK_Mister, The polycoms do, but I never managed to make them work. Didn't try all THAT hard. |
15:26.19 | ManxPower | UK_Mister, buy one of each. Test. |
15:26.43 | *** join/#asterisk Nix (~Nix@dsl81-214-65337.adsl.ttnet.net.tr) |
15:26.55 | UK_Mister | will do, thnx again |
15:26.56 | malbech | I search a softswitch for a good price but it's very diificult to find one ... no ? |
15:27.57 | UK_Mister | i've got a grandstream handset, it works perfectly, just a bit to simple |
15:32.09 | ManxPower | Have y'all heard that lighters will be banned on USA airplanes soon? |
15:32.14 | tzanger | ManxPower: I prefer Telco - PRI - * - PRI - PBX |
15:32.59 | ManxPower | I wonder what they do with all the lighters that are taken. I wonder if I could just go down to the airport anytihng I run out of lighers and get a bag of them from airport security. |
15:33.09 | ManxPower | anything = anytime |
15:34.49 | robl^ | ManxPower, they give them away for every used fingernail clipper you buy :) |
15:35.04 | _THEEND_ | none has interfaced hylafax with asterisk? |
15:35.11 | Moonwick | it'd be a fun prank to run some sort of "lighter exchange" program |
15:35.19 | Moonwick | coordinate with friends in a few cities |
15:35.34 | Moonwick | and have them set up shabby little stands in front of the entrance to security at a few airports |
15:36.11 | Moonwick | where people getting ready to go somewhere can trade lighters and nail clippers for a coupon to receive same at their destination airport |
15:36.18 | ManxPower | _THEEND_, Sure! Plug the modem Hylafax uses into an FXS port of Asterisk. Done! |
15:36.40 | ManxPower | robl^, I think the "h" bounty has generated more messages than any other bounty I've ever posted. |
15:36.42 | _THEEND_ | uhm... |
15:37.06 | ManxPower | _THEEND_, Other than that stop complaining and write support. you'll find out it's VERY VERY tough. |
15:37.34 | ManxPower | You'll have to basically write a softmodem. You could use the DSP from spamdsp. |
15:37.46 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
15:37.46 | *** mode/#asterisk [+o bkw_] by ChanServ |
15:37.48 | Gand_DJ | kewl.... voipforcanada supports G729 |
15:38.13 | ManxPower | In fact you might only have to write an AT command set emulator and serial port emulator. |
15:38.32 | tzanger | ManxPower: :-) |
15:38.44 | tzanger | actually I was going ot write a serila port telnet server |
15:38.53 | ManxPower | That should be EASY compared to coppiece's software DSP library |
15:38.56 | tzanger | because I have a Lucent Max that you can telnet to port 9000 and get a modem |
15:39.01 | tzanger | plug THAT in to HylaFax |
15:39.24 | ManxPower | tzafrir, there are a couple of things available to do that with Linux |
15:39.52 | tzanger | ManxPower: I have only found the reverse |
15:40.25 | tzanger | telnet servers that connect to serial ports, not serial port emulators that really telnet |
15:40.53 | ManxPower | tzanger, Oh! sorry. wasn't reading carefully enough |
15:41.05 | ManxPower | tzafrir, serredir? |
15:41.09 | ManxPower | modempoold |
15:41.13 | ManxPower | mpoold |
15:41.15 | tzanger | that's the idea yeah |
15:41.18 | ManxPower | there's an RFC for it |
15:41.19 | tzanger | but it wouldn't be a daemon |
15:41.21 | *** join/#asterisk jcollie (~jcollie@lt16586.campus.dmacc.edu) |
15:41.29 | tzanger | or I guess it would be |
15:41.40 | jcollie | ~seen anthm |
15:41.41 | jbot | anthm <~anthmct@CPE-69-76-83-52.wi.res.rr.com> was last seen on IRC in channel #asterisk, 8d 22h 28m 58s ago, saying: 'at cluecon!'. |
15:41.42 | bugbot | seen anthm is assigned nothing and reported nothing. |
15:41.50 | tzanger | basically opening /dev/ttyT0->T23 would telnet to w.x.y.z port 9000 |
15:42.00 | ManxPower | *nod* |
15:42.27 | *** part/#asterisk jcollie (~jcollie@lt16586.campus.dmacc.edu) |
15:42.32 | tzanger | well if nothing else, ManxPower, your h,1,Hangup bounty has created discussion :-) |
15:42.33 | psycodad | how do i initiate a supervised call transfer (which keys) ? |
15:42.44 | ManxPower | tzanger, Yup! |
15:42.53 | ManxPower | psycodad, WHAT PHONE???? |
15:43.36 | psycodad | ManxPower: i.e. a sip phone, i can do #extension and get blind transfer, but how to talk to the target party first ? |
15:44.50 | tzanger | psycodad: which phone |
15:44.59 | tzanger | specifics man, speicifics |
15:45.02 | ManxPower | psycodad, I didn't ask you what type of phone. What BRAND of phone. |
15:45.03 | cypromis | how do you know the target party wants to talk to you ? |
15:45.09 | psycodad | 2$ analog phone on ATA |
15:45.15 | ManxPower | psycodad, #transfer is an ugly evil hack and you will go to hell for using it. |
15:45.21 | ManxPower | psycodad, FLASH |
15:45.23 | psycodad | ;-) |
15:45.28 | tzanger | ManxPower: hahaha |
15:45.35 | ManxPower | psycodad, or read the damn documentation for the ATA. That's what it's there for. |
15:45.40 | jlewis | If I have a couple of POTS ports on a TDM400P, * isn't smart enough to find a port (in a group) with working dialtone for making outgoing calls is it? |
15:45.41 | tzanger | yeah I am a huge fan of hookflash for zap... but I odn't know how well that works on an ATA |
15:45.59 | tzanger | jlewis: Asterisk is a PBX, not an aswering machine |
15:46.04 | ManxPower | tzanger, Since all this stuff is handled totally by the ATA it really depends on the brand of ATA. |
15:46.07 | tzanger | put it in charge of the phone lines and it will do its job properly |
15:46.26 | ManxPower | but since psycodad seems incapable of telling us the brand of ATA I think he needs to be put on my /ignore list. |
15:46.27 | jlewis | tzanger: don't think you understood the question...let me restate |
15:46.36 | tzanger | jlewis: I understand the question |
15:46.37 | cypromis | so now that you know that SIP ATA's suck |
15:46.41 | cypromis | what will you do ? |
15:46.43 | cypromis | *gg |
15:46.48 | tzanger | you have a number of POTS lines that * is sharing with phones outside of asterisk |
15:46.49 | ManxPower | jlewis, Asterisk cannot detect dialtone on a line. |
15:47.10 | tzanger | in effect, you want * to act as an aswering machine does, subservient to everything else on the line |
15:47.13 | bjohnson | Gand_DJ: I'd like to here how you find the voipforcanada service. They seem to have started up recently (domain registered in Feb 2005) |
15:47.17 | mrunix | is ftp.asterisk.org being wonky? it always times out on me from work and home |
15:47.27 | Wonka | .oO( mmh? ) |
15:47.30 | jlewis | so if I have a Zap group and 1 line goes dead, outgoing calls using Zap/g1/number will break until the bad channel is unconfigured from the group? |
15:47.35 | psycodad | ManxPower: I have a few Zyxel 2002, a Zap IF and Zyxel 2000W as well as some softphones |
15:47.44 | tzanger | jlewis: asteirsk is a PBX; it does not play nicely "in parallel" with other equipment on the FXO lines |
15:47.46 | ManxPower | tzanger, You go to the 17.25th level of hell. That's where you are surounded by Polycom IP 600's, Cisco 7960G's and high end SNOM phones. All JUST out of reach. |
15:47.49 | tzanger | jlewis: and in the case of a line going dead... no |
15:47.52 | Gand_DJ | bjohnson, you can call toll-free # for free |
15:48.02 | Gand_DJ | just signup for a an acct... |
15:48.04 | tzanger | ManxPower: hahaha wouldn't be bad for me... I don't do SIP |
15:48.08 | jlewis | ok...that's the way it appeared...just wanted to be sure I wasn't missing something |
15:48.18 | ManxPower | psycodad, The correct way to do a transfer is to read the documentation for the device you are using. |
15:48.20 | cypromis | ManxPower: fine, I'll take my swissvoice with me |
15:48.26 | Gand_DJ | I signed up & got my sip info. |
15:48.31 | jlewis | it'll just use the first available channel...whether there's a good line there or not |
15:48.35 | Gand_DJ | also a v4c extension # |
15:48.37 | *** join/#asterisk RoyK (~roy@host-81-191-165-149.bluecom.no) |
15:48.38 | RoyK | v |
15:48.50 | cypromis | w |
15:48.50 | tzanger | jlewis: now "go dead" meaning "not plugged in" or "no battery voltage" -- * does see that |
15:49.11 | psycodad | ManxPower: ok, I'll try that, thnx so far... but anyway, is there a way for supervised transfer via gets-me-to-hell-pound-hack ? |
15:49.25 | cypromis | depends |
15:49.27 | Gand_DJ | they seem to accept Ulaw, Alaw, GSM, and G729 |
15:49.29 | cypromis | there where some apps for that |
15:49.31 | cypromis | and some patches |
15:49.31 | ManxPower | psycodad, only in CVS-HEAD, but I don't think it works (maybe its been fixed) |
15:49.33 | Gand_DJ | nothing else |
15:49.37 | tzanger | psycodad: park it, call the 2nd exten and tell them it's on 701. :-) |
15:49.38 | cypromis | and they where allways broken 2-3 days later |
15:49.39 | cypromis | :) |
15:49.39 | jlewis | in the case I was testing, we only had one line plugged in, and it happened to be the last one...outdial via Zap/g1 would pick channels with no lines attached |
15:49.45 | bjohnson | tzanger: actually, my SPA running parallel to my answering machine copperates nicely .. but that is a feature of that ATA .. not asterisk |
15:49.56 | tzanger | jlewis: odd, what card? TDM400P? It should throw red alarm on nonconnected lines |
15:49.57 | bjohnson | jlewis: ^^ |
15:50.03 | tzanger | bjohnson: correct |
15:50.13 | jlewis | it is a TDM400P |
15:50.18 | ManxPower | tzanger, It should, but I don't think I've ever actually seen a TDM400P go into red alarm |
15:50.18 | bjohnson | exactly |
15:50.24 | tzanger | ManxPower: ahh... |
15:50.43 | ManxPower | jlewis, Asterisk SHOULD detect if there's no line plugged in (i.e. no voltage), but I have never actually TESTED that. |
15:51.04 | Gand_DJ | bjohnson, I tried calling a toll-free number.. and audio seems to cut out after 6 seconds..lol |
15:51.06 | psycodad | tzanger: okay, that sounds not too bad for the start ;-) Anyway the Userguide for Zyxel 2002 sucks big time, no mention of transfering anything, I guess they don't think you ever get a call on this crap ;-) |
15:51.07 | Gand_DJ | trying another one |
15:51.32 | Gand_DJ | bjohnson, if you signup on them.. then we can try calling eachother for quality test |
15:51.48 | psycodad | tzanger: how do I park the call...sorry for sounding stupid ... I have a printed asterisk doc in front of me and spidered voip-info.org for the last 3 days ;-) |
15:51.51 | jlewis | zttool shows no alarms on the TDM400Ps (there are 2 4-port boards installed) even though only the last channel on second card has a line |
15:51.58 | Dovid | hi |
15:51.59 | bjohnson | I've already got about 6 voip accounts .. I'm not signing up unless I think they offer good service |
15:52.03 | Dovid | i am new to asterisk |
15:52.04 | Gand_DJ | hehehe |
15:52.06 | Gand_DJ | ok |
15:52.15 | tzanger | psycodad: find yourself valetparking or supervaletparking, that's what you want |
15:52.18 | Dovid | what is fxo for and what is fxs for ? |
15:52.23 | shido6 | good, Dovid |
15:52.28 | tzanger | Dovid: FXO means you plug the Central Office into it |
15:52.35 | tzanger | FXS means you plug a telephone SET into it |
15:52.35 | Dovid | fxo = pstn |
15:52.36 | Dovid | ? |
15:52.40 | Dovid | ah |
15:52.42 | bjohnson | ~fxofxs |
15:52.43 | jbot | methinks fxofxs is An FXO port expects to receive dialtone and receive ring voltage. An FXS port expects to provide dialtone and provide ring voltage. |
15:52.43 | bugbot | fxofxs is assigned nothing and reported nothing. |
15:52.46 | tzanger | FXO means goes to PSTN (O)ffice |
15:52.52 | jlewis | http://www.digium.com/index.php?menu=fxsvfxo |
15:52.53 | tzanger | FXS means goes to tleephone (S)tation |
15:52.54 | bjohnson | well .. not always |
15:52.56 | Dovid | so fxs is to a fax machine or pts phonr |
15:53.00 | bjohnson | well .. not always |
15:53.11 | tzanger | bjohnson: for all intents and purposes, yes |
15:53.25 | cypromis | most itents and purposes |
15:53.35 | bjohnson | hehe .. yes most |
15:53.37 | cypromis | my phones mostly go to NT mode BRI ports |
15:53.43 | tzanger | bjohnson: yes you can conjure up some oddball scenarios but don't confuse the poor guy, at least wait until he's discovering why exten _. is bad |
15:53.48 | bjohnson | gets confusing when intercting with another pbx |
15:53.51 | *** join/#asterisk kant (~bernd@metrored-gw.tropicohn.com) |
15:53.54 | psycodad | tzanger: thnx found it...will try that way... and if anybody has Zyxel 2002 ATAs : How do you transfer calls (except with #) |
15:54.02 | *** join/#asterisk bonez41 (~aint@c-67-166-77-14.hsd1.ut.comcast.net) |
15:54.03 | *** join/#asterisk CoolAcid (~jk@216.99.98.39) |
15:54.34 | *** join/#asterisk tessier (~treed@210.245.102.159) |
15:54.56 | jakepdev | is there a way in 1.0.7 stable to send DTMF digits? |
15:55.04 | jakepdev | to the same trunk |
15:55.10 | tzanger | jakepdev: SendDTMF()? |
15:55.24 | jakepdev | in HEAD only I think |
15:55.45 | file[mac] | nope it's in stable too... |
15:55.47 | Gand_DJ | I appear to be the 26th person to signup for voipforcanada... based on the owners ex # being 8000 and I am 8026 :) |
15:55.48 | tzanger | jakepdev: ahh |
15:56.07 | Gand_DJ | I checked their paypal log acct, and so far no payments were made to them |
15:56.44 | ManxPower | jakepdev, HUH? I do that ALL THE TIME. |
15:56.53 | jakepdev | i call flash - work |
15:56.55 | jakepdev | works |
15:57.06 | jakepdev | then call SendDTMF - and it doesn't come up |
15:57.24 | ManxPower | jakepdev, stop being so lazy. "send dtmf digits" can mean about 450 different things. Make sure people understand what you mean. |
15:57.34 | jakepdev | what are you talking about |
15:57.34 | jakepdev | ? |
15:57.41 | jakepdev | SendDTMF is what I'm looking for |
15:58.00 | `Sauron | Hehn |
15:58.05 | `Sauron | Fresh from NANOG: |
15:58.11 | `Sauron | you wanna see bad second grade playground behavior, try the |
15:58.12 | `Sauron | asterisk mailing list. |
15:58.27 | jakepdev | just isn't running the cmd |
15:58.42 | ManxPower | I can send DTMF digits out a Zap FXO to dial a number. I can send DTMF using D() in Dial. I can SendDTMF to the caller on an inbound call that's in an IVR, I can send DTMF while on a call to interact with an outside IVR. |
15:58.45 | ManxPower | Shall I continue? |
15:58.49 | jakepdev | nope |
15:59.02 | jakepdev | cause I'm the one with the error :) |
15:59.09 | ManxPower | Oh, I can send dtmf to the callee using the M() option and a macro during a dial |
15:59.24 | file[laptop] | ManxPower: How sexy! |
15:59.34 | Gand_DJ | hrm... ne1 using *@home and have fwd setup for incoming? I can call out (finally), but get busy signal for incoming. |
15:59.37 | ManxPower | jakepdev, what makes you think SendDTMF is the correct way to do what you want? |
16:00.42 | jakepdev | just want to Flash the Send didts to pass back to the host switch |
16:00.42 | tzanger | ManxPower: well you could use playtones but that's pretty hardcore :-) |
16:00.42 | ManxPower | Oh! Yes. You can use Playtones to send dtmf too! |
16:00.43 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
16:00.43 | jakepdev | was hoping not to go that route |
16:00.43 | *** join/#asterisk jf_ (~jeanfranc@HSE-Montreal-ppp332742.sympatico.ca) |
16:00.43 | ManxPower | OK, maybe not 450 ways, but more ways than you can ever imagine, Grasshopper |
16:00.57 | jakepdev | tnx - 1 working way is enough |
16:01.09 | jf_ | I've just installed * on a new system, i get these error, who can help me |
16:01.09 | jf_ | Apr 14 12:06:49 WARNING[9080]: chan_iax2.c:7478 load_module: Unable to open IAX timing interface: No such file or directory |
16:01.10 | *** join/#asterisk Marlow (~martin@cerberus.bluetree.ie) |
16:01.16 | jakepdev | maybe it's failing on Flash |
16:01.34 | jf_ | <PROTECTED> |
16:01.42 | tzanger | jf_: read the zaptel README |
16:01.55 | jf_ | read it already |
16:02.03 | ManxPower | jakepdev, Explain what you want to ACCOMPLISH. |
16:02.07 | *** join/#asterisk sean (~sean@iconoclast.caedmon.net) |
16:02.27 | jakepdev | just want to Flash then Send DTMF to pass back the call to the host switch |
16:02.31 | gmcinnes | jf_: what distro are you using? |
16:02.38 | ManxPower | You are asking "how do I chop down a tree for firewood?" when you really mean "I want to stay warm in winter. What is the best way to do that?" |
16:02.41 | jf_ | gentoo |
16:02.56 | ManxPower | jf_, then read it again. |
16:02.58 | *** join/#asterisk Blackthorn (blackthorn@ws-10.smyth.net) |
16:03.16 | jakepdev | i'mm spending too much time trying to decipher that last comment |
16:03.30 | jakepdev | the dialplan is easier |
16:03.35 | BuckRogers | good morning |
16:03.41 | *** join/#asterisk RChadwell (~rob@rrcs-24-227-48-86.se.biz.rr.com) |
16:03.58 | RChadwell | Am I right that you can only register one sip connection? |
16:04.06 | jakepdev | rc - no |
16:04.14 | jakepdev | you can register as many as you please |
16:04.27 | RChadwell | how about for incoming calls |
16:04.45 | tzanger | jf_: if you read it, read it again |
16:05.14 | L|NUX | is there any Presence server for * |
16:05.15 | tzanger | jf_: you're not listening to it |
16:05.21 | tzanger | jf_: if it's a 2.6 kernel, read README.udev |
16:05.31 | jf_ | ok |
16:05.34 | tzanger | jf_: FIRST RULE of open source: what little documentation is provided, READ IT ALL :-) |
16:05.49 | tzanger | jf_: it's a lot easier to read it than the ocde itself, which is also available |
16:05.55 | cypromis | tzanger: ever met someone following that rule ? |
16:05.56 | jf_ | thank i will in the future |
16:06.05 | tzanger | which is a major plus over closed source, where the documentation's equally shitty, but you can't see the code when you need it :-) |
16:06.10 | RChadwell | scenario I am lookging for is 2 numbers each ringing to the s extension for a different context - seems that the incoming calls all go to the 2nd sip registered - is it because of the /s on the end? |
16:06.14 | tzanger | cypromis: I can try to change the world, can't I? :-) |
16:06.29 | ManxPower | The second rule is: TEST TEST TEST, PROTOTYPE PROTOTYPE PROTOTYPE |
16:06.36 | tzanger | RChadwell: don't put both SIP clients in the same context= |
16:06.38 | tzanger | in sip.ocnf |
16:06.51 | ManxPower | And the 3rd rule: Users are lieing weasels and should never be believed. |
16:06.52 | RChadwell | they aren't - that is the weird thing |
16:07.01 | tzanger | ManxPower: that one is very true |
16:07.08 | tzanger | RChadwell: what is the context line for each |
16:07.11 | RChadwell | I think it is 1) Collect Underpants 2) ??? 3) PROFIT! |
16:07.24 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
16:07.33 | tzanger | RChadwell: no that business model's proven to be ineffective due to the sheer volume of underpants gnomes |
16:07.35 | RChadwell | context=cmgincoming for first ---- context=chadwellhomesincoming for second |
16:07.43 | RChadwell | HA |
16:07.45 | ManxPower | For the install I'm doing: User said: We need 4 fax machines and 1 modem. What we discovered: They need 15 analog ports for analog devices |
16:07.49 | Blackthorn | How do you check the version of * that I am running. I did a show version but it just says head and then gives a date. I noticed on the * download it's at 1.07 |
16:08.03 | tzanger | RChadwell: ok, now "show dialplan cmgincoming and show dialplan chadwellhomesincoming |
16:08.06 | tzanger | and make sure they're not the same |
16:08.20 | tzanger | in extensions.conf you may have included a context with an s exten |
16:08.23 | tzanger | and it's getting called |
16:08.30 | *** join/#asterisk jdg (~jdg@CA03F897.adsl.mana.pf) |
16:08.31 | jlewis | this is weird...X100P in the same system does give me an alarm / no alarm when the line is missing / plugged in...but the TDM400Ps are always happy |
16:08.43 | tzanger | jlewis: yeah the TDM400P doesn't seem to have that bit of code |
16:08.56 | tzanger | jlewis: if you're eager, it shouldn't be too terribly hard to add to wctdm |
16:08.58 | RChadwell | gotcha - I will check that - I read in the asterisk FAQ that it can't register more than one sip - I guess it was old |
16:09.01 | jlewis | that's surprising since they're "higher end" cards |
16:09.04 | *** join/#asterisk egon_ (~egon@pc-10-19-104-200.cm.vtr.net) |
16:09.07 | L|NUX | is there any Presence server for * |
16:09.12 | tzanger | L|NUX: no |
16:09.22 | tzanger | there's a jabber server, you could probably integrate with that |
16:09.31 | jlewis | my C / driver writing foo would not likely cut it |
16:09.41 | tzanger | jlewis: there's no better way to learn |
16:09.45 | vaewyn | that @#$@#$@#$ putz... had me plugged into the wrong port on the Norhell.... plug it in the correct port and VOILA! I have a norhell <-> asterisk connection.... zero config |
16:09.55 | tzanger | who doesn't want strong fo |
16:09.55 | tzanger | er foo |
16:10.06 | vaewyn | I like a strong bar |
16:10.06 | tzanger | vaewyn: :-) |
16:10.07 | vaewyn | :P |
16:10.24 | jlewis | is this really the way its supposed to be?...wondering if a post to the list is worth it |
16:10.49 | tzanger | jlewis: it just needs to be done is all |
16:11.14 | ManxPower | jlewis, Post a bounty to the mailing lists. |
16:11.23 | ManxPower | I'll bet you could get it added for less than $100 |
16:11.23 | *** join/#asterisk ikey1 (ikey@220.226.42.220) |
16:11.29 | tzanger | yeah the lists are hot for bounties right now |
16:11.33 | tzanger | we just like btiching abotu the implementations |
16:12.07 | *** join/#asterisk gnmraju (gnmraju@220.226.41.204) |
16:12.17 | jf_ | tzanger: do i run modprobe aftter adding the lines |
16:12.26 | L|NUX | tzanger : hmm |
16:12.38 | tzanger | yeah remove and reinstall the wctdm and zaptel drivers |
16:12.39 | tzanger | it should just work then |
16:13.18 | jf_ | remove it physically or the bin |
16:14.05 | drumkilla | ManxPower: why do you want an error on exten => h,1,hangup ? |
16:14.22 | drumkilla | woah ... or I could just read all of the replies |
16:14.25 | tzanger | :-) |
16:14.32 | tzanger | jf_: no no just rmmod wctdm zaptel |
16:14.36 | malbech | I search a softswitch for a good price but it's very diificult to find one ... no ??? |
16:14.39 | tzanger | (or whatever wcxxx card you're using) |
16:14.46 | *** join/#asterisk cpatry (~grepmoo@65.39.228.5) |
16:15.17 | jf_ | ok and then modprobe |
16:15.52 | ManxPower | drumkilla, because people call hangup from within _. and don't realize what's happening. |
16:15.57 | ManxPower | see the discussions on -dev |
16:16.31 | ManxPower | drumkilla, For my next trick watch me start a riot on the mailing list! *grin* |
16:16.47 | drumkilla | haha, yeah, I'm reading now |
16:16.56 | drumkilla | sorry, didn't think about stupid people ;) |
16:17.09 | ManxPower | drumkilla, you know that I help a lot of stupid people. |
16:18.02 | drumkilla | yeah, and it's really cool that you have the patience to do so |
16:18.13 | jf_ | got that now WARNING[8898]: chan_zap.c:771 zt_open: Unable to specify channel 1: No such device or address |
16:18.39 | JunK-U | jf_: cause ur /dev/zap/1 isnt there. |
16:18.48 | ManxPower | jf_, maybe the module on your TDM400P is really on port 4 and not port 1 |
16:19.04 | jf_ | i only have a wcfxo |
16:19.32 | tzanger | jf_: them rmmod wcfxo zaptel and modprobe wcfxo |
16:19.57 | ManxPower | drumkilla, someday I'll go to geek heaven for this. |
16:20.08 | ManxPower | but I'd rather go to supermodel heaven |
16:20.41 | bannerman | ManxPower: If a geek dies and doesn't see supermodels, he went to the other palce. |
16:20.59 | gmcinnes | ManxPower: You help me a lot, and I'm pretty thick :) |
16:21.02 | jalsot | is it possible to make a call transfer when the number is dialed through AGI? |
16:21.08 | gmcinnes | ManxPower: Thank you. |
16:21.20 | ManxPower | gmcinnes, you can thank me by finding me a job in europe. |
16:21.20 | gmcinnes | tzanger: You too! |
16:21.39 | tzanger | time for lunch |
16:21.42 | ManxPower | gmcinnes, you can thank tzanger by finding me a job in europe. 8-) |
16:21.43 | tzanger | no proble gmcinnes |
16:21.45 | tzanger | hahaha |
16:21.48 | tzanger | yeah pay it forward to manx |
16:21.54 | gmcinnes | ManxPower: europe == supermodel heaven? :) |
16:22.01 | ManxPower | gmcinnes, nope. |
16:22.14 | eKo1 | wwwwhat? |
16:24.34 | Marlow | drumkilla: wouldn't it be better to use that ticket for the flight, rather than letting the ticket fly on it's own ? |
16:24.49 | Marlow | :o) |
16:24.59 | drumkilla | heh, I'll still be using it |
16:25.17 | *** join/#asterisk egon_b (~egon@pc-10-19-104-200.cm.vtr.net) |
16:25.22 | drumkilla | von europe |
16:25.27 | vaewyn | ok... that DS3 card is calling my name ;P |
16:25.34 | ManxPower | drumkilla, You going to Von Europe? |
16:25.41 | ManxPower | I'll be good to meet you |
16:25.46 | RoyK | I'll try to go ther |
16:25.47 | RoyK | e |
16:25.54 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-201-68.dsl.scarlet.be) |
16:25.54 | vaewyn | drumkilla is a cool guy IRL |
16:26.02 | RoyK | to see stockholm during the weekend and then the von |
16:26.04 | ManxPower | vaewyn, I'm an asshole IRL. 8-) |
16:26.05 | vaewyn | he can't say the same for me :P |
16:26.06 | file[laptop] | drumkilla rocks |
16:26.21 | vaewyn | ManxPower: that's different how?!? ;P |
16:26.23 | Marlow | royk: there isn't much you'll see in stockholm .. |
16:26.26 | vaewyn | ManxPower: had to... :P |
16:26.30 | ManxPower | vaewyn, LOL! |
16:26.35 | ManxPower | vaewyn, Someone had to. |
16:26.42 | Marlow | royk: usually it's just getting drunk, and then you won't see anything after that . |
16:26.44 | RoyK | Marlow: what? I've been there several times - I like the city :) |
16:26.49 | RoyK | hehehe |
16:27.02 | RoyK | that's part of the seeing the city :) |
16:27.06 | Marlow | royk: i've been living there for over 1 1/2 years :) |
16:27.13 | RoyK | okk |
16:27.16 | ManxPower | Whoo! Whoo! The TA 750 is "out for delivery" |
16:27.21 | RoyK | I just live in oslo ....... |
16:27.28 | RoyK | ta750? |
16:27.31 | Marlow | royk: me galway |
16:27.41 | ManxPower | RoyK, Adtran channel bank |
16:27.46 | RoyK | k |
16:27.52 | drumkilla | ManxPower: yep, I'll be there :) |
16:27.53 | RoyK | marlow where is galway? |
16:28.03 | Marlow | royk: irish westcoast :) |
16:28.19 | drumkilla | ManxPower: should I be scared to meet you? :p |
16:28.20 | RoyK | ahki. north of limerick somewhere? |
16:28.37 | Marlow | royk: yeah .. about 1 1/2 hours drive north of Limerick |
16:28.44 | BuckRogers | good morning |
16:28.49 | RoyK | "stab city" |
16:28.50 | RoyK | :P |
16:28.59 | RoyK | marlow talar du svenska? |
16:29.19 | Marlow | royk: jeg er dansk, men jeg pratar också svenska, om det skal vära |
16:29.27 | RoyK | ok |
16:29.33 | vaewyn | drumkilla: meeting anyone on here IRL is a crap shoot... emphasis on crap ;P |
16:29.42 | Marlow | royk: det var jeg godt klar over :) |
16:29.46 | RoyK | ahki |
16:30.17 | RoyK | Marlow: du får komme inn på asterisk-no .. |
16:30.18 | RoyK | <PROTECTED> |
16:30.26 | *** join/#asterisk jtodd (~jtodd@dsl027-191-178.sfo1.dsl.speakeasy.net) |
16:30.30 | vaewyn | actually I am surprised how well everyone on here gets along IRL... usually better than on here :P |
16:30.31 | Marlow | royk: bedre ? |
16:30.36 | RoyK | æøå |
16:30.44 | Marlow | royk: æøå |
16:30.47 | RoyK | Marlow: ah |
16:30.48 | RoyK | :) |
16:30.57 | Marlow | royk: default er UTF-8 i gaim |
16:31.03 | Marlow | royk: havde ikke fikset det |
16:31.14 | RoyK | oki |
16:33.08 | RoyK | Marlow: er du fra København? |
16:33.17 | Marlow | royk: nej .. sønderjylland |
16:33.23 | RoyK | ok |
16:33.48 | Marlow | royk: men jeg har boet lidt af hvert sted både indenfor og udenfor DK |
16:34.04 | RoyK | skjønte det. |
16:34.06 | Marlow | royk: er lige flyttet fra Dublin til Galway |
16:34.16 | RoyK | hvor stort er galway? |
16:34.18 | file[laptop] | vaewyn: indeed, we're just a great big bunch of geeks IRL |
16:34.29 | Marlow | royk: 66k i byen, 150k i countien |
16:34.39 | RoyK | file[laptop]: correct |
16:34.42 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
16:34.48 | ManxPower | drumkilla, Prolly no need to be scared. |
16:34.54 | ManxPower | I'm MUCH more scary online than in person |
16:34.55 | RoyK | Marlow: ok, som en middels stor by i .no, med andre ord :) |
16:35.02 | RoyK | s/middles/ganske/ |
16:35.07 | file[laptop] | in person he's a pussy cat I bet :p |
16:35.12 | Marlow | royk: og områdemæssigt en af de største counties i irland |
16:35.20 | RoyK | ok |
16:35.20 | Marlow | royk: + den mest venlige :) |
16:35.24 | RoyK | fint :) |
16:35.30 | RoyK | ikke som limerick? |
16:35.35 | RoyK | aka stab city? |
16:35.37 | Marlow | royk: venlige = social |
16:35.46 | Marlow | royk: limerick = dell .. that's it .. |
16:35.49 | RoyK | vet |
16:35.52 | Marlow | royk: der er ikke andet i Limerick |
16:36.07 | Marlow | royk: Galway er en by, hvor man lever |
16:36.14 | bannerman | How do the pro's do QoS? By owning the router on both sides? |
16:36.15 | RoyK | jeg har vært der én gang, og det var på fabrikken til dell for å teste ut en SAN-løsning som sugde noe helt jævlig |
16:36.33 | ScythelX | norsk? |
16:36.33 | Marlow | royk: eh .. som kunde ? |
16:36.37 | RoyK | ja |
16:36.41 | RoyK | jobba i fast.no på den tida |
16:36.50 | Marlow | scytheIX: royk = norsk, marlow=dansk |
16:37.15 | Marlow | royk: jeg har arbejdet for Dell indtil sidste uge |
16:37.18 | RoyK | etter å ha påpekt det for dell at løsninga deres ikke holdt mål fikk jeg masse pepper fra sjefen om at jeg var "uprofesjonell" mot våre leverandører...... |
16:37.21 | ScythelX | hehe i understand very little i have a lot of friends from denmark and norway |
16:37.23 | Marlow | royk: gad ikke mere |
16:37.29 | vaewyn | vaewyn = stupidamerican |
16:37.31 | vaewyn | :} |
16:37.34 | RoyK | det her var før dell begynte å brande SMC |
16:37.42 | RoyK | eh. ikke smc, men .. ? |
16:37.51 | Marlow | EMC |
16:37.52 | RoyK | ScythelX: where from? |
16:37.57 | RoyK | Marlow: yeah |
16:38.10 | ScythelX | I live in the US now, but I used to live in Nice, France |
16:38.20 | Marlow | royk: 650F er ok, 660F = wouldn't touch it .. |
16:38.22 | jero | and me in marseilles :) |
16:38.35 | RoyK | de hadde en hjemmesnekra løsning før det uten LUN masking eller noen ting. alt som fantes av LUN masking måtte gjøres i software |
16:38.37 | Blackthorn | Following the * web site instructions on how to login to cvs and update your * source files, as it was scrolling across the screendoing the update i saw "xxx file no longer in repository" does that it mean that it deletes it from the current source directory? Since i guess it's no longer needed? |
16:38.54 | vaewyn | Blackthorn: yep |
16:38.59 | RoyK | ...og når vi da skulle ha opp freebsd og linux uten annet enn vanlig fibrechannel-drivere, gikk det til helvete... |
16:39.00 | Blackthorn | kewl |
16:39.03 | Marlow | royk: jup ... men der er ikke mange af dem mere ... |
16:39.08 | RoyK | det er godt |
16:39.11 | RoyK | veldig godt |
16:39.31 | jero | obenstrü |
16:39.33 | RChadwell | I think the problem is in my sip.conf file - mind if I put a portion here? |
16:39.34 | Marlow | royk: de bliver ikke solgt mere ... i flere år |
16:39.47 | eKo1 | RChadwell: use pastebin |
16:39.51 | Blackthorn | now that i've updated my * source I guess i just enter each directory and do the make clean;make install correct? |
16:39.57 | *** join/#asterisk didz_ (didz_@200.218.192.52) |
16:40.03 | file[laptop] | I'm afraid |
16:40.07 | file[laptop] | my step-dad knows what bluetooth is |
16:40.21 | denon | file[laptop]: he just got back from the dentist? |
16:40.22 | eKo1 | Blackthorn: eh, 'make update && make clean && make install' |
16:40.28 | RChadwell | Never heard of pastebin - any instrux online? |
16:40.35 | eKo1 | ~pastebin |
16:40.36 | jbot | pastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca |
16:40.36 | bugbot | pastebin is assigned nothing and reported nothing. |
16:40.36 | file[laptop] | he was unpacking a plantronics wireless headset |
16:40.36 | [TK]D-Fender | www.pastebin.ca |
16:40.45 | file[laptop] | denon: but anyway, what did you want me for? |
16:40.52 | RoyK | marlow jeg vet, de tok til vettet :) |
16:41.01 | denon | nothin .. had a Q last night, figured it out |
16:41.06 | file[laptop] | now now otl |
16:41.46 | ManxPower | jbot, in my experience rope works better. |
16:42.06 | didz_ | anyone knows how to pause a queue member from console? |
16:42.09 | RChadwell | http://pastebin.ca/9545 |
16:42.10 | RoyK | you'll need all those hooks in the floor |
16:42.15 | RChadwell | btw - that is very cool |
16:42.25 | ManxPower | RoyK, You don't already have hooks in the floor??? |
16:42.35 | RoyK | ManxPower: no, sorry |
16:42.36 | vaewyn | kinky |
16:42.38 | RoyK | :) |
16:42.50 | denon | ManxPower: bedposts |
16:43.07 | vaewyn | ack... jbot goes metric on us ;P |
16:43.12 | RoyK | :) |
16:43.34 | jf_ | is there any to know the ring length to set distinctive ring |
16:43.38 | vaewyn | jbot: 1 mile per hour in fulrongs per fortnight |
16:43.45 | vaewyn | ack... bad spelling |
16:43.52 | RoyK | fulrong.. |
16:43.53 | vaewyn | jbot: 1 mile per hour in furlongs per fortnight |
16:44.13 | RoyK | wtf is a furlong? I've heard the word, but.... |
16:44.24 | vaewyn | distance measurement |
16:44.28 | RoyK | yeah |
16:44.31 | RoyK | i knopw |
16:44.32 | RoyK | know |
16:44.35 | RoyK | but how long? |
16:44.43 | robl^ | its longer than a cubit |
16:44.48 | RoyK | .no and .se has something called Mil, being 10km :P |
16:44.49 | RChadwell | The problem is that every incoming call uses the [sip2.broadvoice.com] registration |
16:45.09 | vaewyn | 1 furlong = 660 feet |
16:45.20 | RoyK | I should have thought of that... |
16:45.22 | RChadwell | any ideas? |
16:45.26 | RoyK | pretty obvious |
16:45.36 | Gand_DJ | bjohnson, I just noticed on voipforcanada website they added "We are currently in Beta stages of development". lol |
16:45.59 | Gand_DJ | Explains why audio cuts out after 5 seconds.. which sucks. emailed them on it |
16:47.29 | RoyK | hm. so one furlong is ten chains, one mile eight furlongs.. |
16:47.34 | didz_ | anyone knows how to pause a queue member from console? |
16:47.38 | vaewyn | Is there a way to get callerid name passed over the PRI to my Norhell? I seem to remember that isn't an option but... |
16:47.38 | RoyK | stupid english measurement |
16:48.28 | Marlow | royk: 1 scandinavian mile is 10km .. the only ones that aren't using that anymore are us danes :) |
16:48.28 | jero | :) |
16:48.39 | vaewyn | I'm surpised the metric nazis havn't come up with a way to make time divisible by 10 |
16:48.45 | ManxPower | Gand_DJ, When you researched voipforcanada by asking on the mailinglists nobody mentioned problems? |
16:48.47 | RoyK | Marlow: poor buggers |
16:48.58 | Marlow | royk: take ireland .. that's the complete mess .. |
16:49.11 | ManxPower | vaewyn, It's called Internet Time. I think Swatch has info on their web site. |
16:49.12 | RoyK | Marlow: it used to be slightly more than 10km, being the distance you could walk before having to rest |
16:49.16 | Marlow | royk: when i came over here, speeds where in mph, but distances in km |
16:49.30 | vaewyn | ManxPower: yeah... but even scientists think iut is crap :P |
16:49.33 | BuckRogers | in canada voip service providers have 90 days to comply with 911 calling regulations or shut down |
16:49.33 | RoyK | http://en.wikipedia.org/wiki/Mil |
16:49.34 | *** join/#asterisk boch (~as24@200.59.172.98) |
16:49.36 | Gand_DJ | ManxPower, I haven't done any google searchs for mailing lists that would have voipforcanada in it |
16:50.05 | ManxPower | http://www.timeanddate.com/time/internettime.html |
16:50.06 | RoyK | speed measurement should perhaps be chains per teabreak |
16:50.07 | RoyK | :P |
16:50.10 | Marlow | royk: now it's "go metric", meaning speeds were also changed from mph to km/h, but you have to be careful, if there is a "km/h" in the bottom of the speed-limitation or not .. |
16:50.19 | jf_ | is there any to know the ring length to set distinctive ring |
16:50.23 | Marlow | royk: you can from time to time still get some, that are in mph |
16:50.55 | RoyK | well wtf. just treat all signs as km/h and hope they're mph and no cops around :P |
16:50.55 | Marlow | royk: same as that you still find signs on the country-side, that show the distance in miles, instead of kms |
16:51.02 | ManxPower | jf_, where would you set distinctive ring? |
16:51.21 | |Vulture| | is there a way to set a variable to the reciever of a current call? |
16:51.28 | RoyK | can someone please tell me wtf "distinctive ring" really is? |
16:51.28 | jf_ | on my line: there is 2 number, with to different ringtone, i want * to handle only 1 number |
16:51.29 | Marlow | royk: cops aren't really a problem around here, and irish people can't drive .. it's worse than stockholm or paris |
16:51.53 | Marlow | royk: you can have different ringtones for different numbers on the same pstn line |
16:51.53 | |Vulture| | say x101 picks up a call that rings into 101/102/103, and you want to set userfield to 101 |
16:52.08 | ManxPower | look in zapata.conf.sample |
16:52.12 | ManxPower | I'm outta here |
16:52.23 | vaewyn | RoyK: is a good way to make a phone sound sick ;P |
16:52.29 | RoyK | hehe |
16:52.37 | RoyK | is it that fast ringing tone? |
16:52.48 | RoyK | as in 500ms on 500ms off or something? |
16:52.52 | vaewyn | there are 4 patterns |
16:53.00 | RoyK | ok |
16:55.11 | *** join/#asterisk Ridgeback (~Ridgeback@ppp130-78.lns1.adl2.internode.on.net) |
16:55.20 | Ridgeback | good morning! |
16:55.21 | jf_ | manxpower: i knew that i cant just find which length, can it be 900,0,0 |
16:55.25 | RChadwell | http://pastebin.ca/9545 - Would someone be interested in taking a quick peek at the sip.conf file to tell me why all calls are going to the second context? |
16:55.59 | *** join/#asterisk florz (nobody@2001:1a50:503c:0:0:0:0:1) |
16:56.12 | RChadwell | Can you register a sip connection within a context or does it have to be in the general context? |
16:56.32 | Gand_DJ | Dumb question.. if you can use iax & sip for for asterisk to a voip provider... would iax be the better choice in general? |
16:56.48 | vaewyn | Yes |
16:56.51 | RChadwell | less overhead with iax2 |
16:56.55 | vaewyn | you can trunk then |
16:57.03 | Gand_DJ | k |
16:57.04 | Ridgeback | RChadwell, does your second register command need to be sip2.broadvoice.com??? |
16:57.13 | Gand_DJ | Signing up at link2voip.com |
16:57.22 | RChadwell | not really - just wanted a way to differentiate between the 2 |
16:57.26 | Ridgeback | woohoo got Dundi working between 4 switches!!!!!!! |
16:57.36 | Gand_DJ | ne1 tried them? They seem to have ok rates, and are from canada, but don't mention if fees are cdn or urd |
16:57.37 | Gand_DJ | usd |
16:57.51 | RChadwell | oh - sorry, misunderstood - broadvoice requires that register command |
16:58.09 | Ridgeback | hey when is nuphone going to accept new customers again? I need a 1800 did :) |
16:58.27 | RChadwell | register - then create a context for it that drives to an extensions.conf context - right? |
16:58.43 | Wazb | when patteren will asterisk receive after establishment of call from Regular phone in order to select other option |
16:59.59 | RChadwell | I mean, it shouldn't be hard to take two phone numbers (both sip) and route them to 2 different extensions.conf contexts... |
17:00.14 | Ridgeback | RChadwell, yes the register statement will have a matching block with a context labeled. in this context one would then put thier exten statements |
17:02.45 | gnmraju | is there any sppech processing module for Asterisk available anywhere |
17:03.10 | Ridgeback | gnmraju, festival can talk, sphinx can listen |
17:04.11 | robl^ | ..and Alison can "mooo" |
17:04.22 | *** part/#asterisk Marlow (~martin@cerberus.bluetree.ie) |
17:04.34 | Ridgeback | yeah she can :) |
17:04.47 | denon | has anyone implemented sphinx with asterisk though? |
17:04.54 | Gand_DJ | appears link4voip is USD... says so after you logged into the site |
17:05.33 | *** join/#asterisk myrkraverk (~user@myrkraverk.user) |
17:05.44 | myrkraverk | hello |
17:06.02 | gnmraju | i heard about Sphinx, but Ridge do u know any guy who can integrate everything for us on digium quad wild card |
17:06.22 | myrkraverk | does it make sense to use asterisk for voip backend in a real-time collaboration tool? |
17:06.37 | RoyK | gm |
17:06.39 | Ridgeback | gnmraju, not native to asterisk, it would all be a custom setup you desing |
17:06.39 | RoyK | hmmmm |
17:06.42 | myrkraverk | or am I researching the wrong tool? |
17:06.48 | Ridgeback | *design |
17:06.58 | RoyK | running 1.0.6 it seems asterisk leaks quite a lot of memory |
17:07.01 | Ridgeback | myrkraverk, it would woork well |
17:07.11 | Ridgeback | RoyK, uses the latest CVS, much better |
17:07.27 | myrkraverk | Ridgeback: k, then I'll continue researching ;) |
17:07.31 | RoyK | Ridgeback: what do you mean? cvs head???? |
17:07.49 | Ridgeback | RoyK, yes the latest works great with my shiny new Polycom IP600 :) |
17:08.02 | gnmraju | yes Ridge i know, but do u know any person providing paid consultency on this or any person who can integrate, like some guys who configure vc dial etc |
17:08.23 | Ridgeback | myrkraverk, yes meetme conferences are great for collarboration. easy to setup too |
17:08.50 | RoyK | Ridgeback: you don't want to use cvs head for production! |
17:08.55 | Ridgeback | gnmraju, consultants? hmmm dont know any one who does, but the http://www.voip-info.org has lists to Asterisk consultants |
17:09.00 | *** part/#asterisk JunK-U (~grepmoo@65.39.228.5) |
17:09.03 | myrkraverk | Ridgeback: cool stuff |
17:09.27 | bkw_ | where the hell is manx |
17:09.33 | Ridgeback | RoyK, i agree, but I've used the latest CVS head for a year now on 4 switches. they all work fine |
17:10.02 | vaewyn | bkw_: left about 45 mins ago |
17:10.13 | Ridgeback | RoyK, maybe valgrind memory leak detecto could help determine which module asterisk is leaking out of? |
17:10.17 | RoyK | Ridgeback: also, this system uses sipfriends, the old stuff from 1.0 |
17:10.26 | robl^ | bkw_, his internet went down |
17:10.31 | Ridgeback | RoyK, good grief! |
17:10.42 | RoyK | Ridgeback: what else is there that is 'stable'? |
17:10.48 | *** join/#asterisk jf_ (~jeanfranc@HSE-Montreal-ppp332742.sympatico.ca) |
17:10.50 | Ridgeback | rob, oh the humanity! |
17:11.17 | Ridgeback | RoyK, hmmm the latest stable version of asterisk is probably umm stable ;) |
17:11.31 | RoyK | this is the latest 'stable' asterisk |
17:11.33 | Ridgeback | RoyK, sorry didnt mean to sound like ajerk ;) |
17:11.34 | RoyK | or pretty close |
17:11.38 | RoyK | you don't |
17:11.43 | jf_ | any way to call on zap channel without have to press 9 before, for now it trunk the forst umber of the telephone number |
17:11.44 | RoyK | but asterisk jerks |
17:11.52 | Ridgeback | RoyK, lol |
17:11.56 | *** part/#asterisk Nix (~Nix@dsl81-214-65337.adsl.ttnet.net.tr) |
17:12.19 | vaewyn | jf_: remove the :1 from ${EXTEN:1} |
17:12.20 | Wazb | what patteren will asterisk receive if option is selected after establishment of call from Regular phone? |
17:12.51 | jf_ | k |
17:12.51 | Ridgeback | RoyK, one thing you could try is one of those new asterisk on CD. i jsut read an article and it said this one particular variant was the most stable he has ever seen |
17:13.10 | bkw_ | robl^, haha bet he did exten => h,1,Hangup |
17:13.20 | RoyK | where are the test results? what tests have been run? where are the patches that makes the diff from 1.0.7? |
17:13.55 | Ridgeback | RoyK, dont know. I dont use the stable versions... |
17:14.15 | *** part/#asterisk myrkraverk (~user@myrkraverk.user) |
17:16.01 | jf_ | any know the ring lenght of bell canada for long ring |
17:16.02 | didz_ | anyone knows how to pause a queue member from console? |
17:16.04 | robl^ | bkw_, of course! he needed to hang up after hanging-up! |
17:16.30 | bkw_ | haha |
17:17.26 | Ridgeback | anyone here use dundi yet? |
17:17.32 | robl^ | why does "sudo rm -fr /*" delete everything? gotta love wild cards |
17:20.25 | foobos | rob, * is not required there even.. since -r implies recursive |
17:20.48 | Ridgeback | hey is there a way to use the SendText() application to send text to another users phone? |
17:20.53 | *** join/#asterisk Holos (~asdf@207.164.188.10) |
17:21.56 | Ridgeback | all SendText() offers is SendText(text) no extension number to send text toward (other than the calling extension) |
17:22.14 | robl^ | foobos, shh!! I was making a point. you gone and messed it all up. :) |
17:22.28 | Holos | Can anyone give me some extra tips on reducing echo on SIP to PSTN with TDM400? I have tweaked my RXgain to 5,10, and 12 and TXgain to 0,-5,10 and I am not noticing much difference, but there is no echo on sip-sip or on far side sip to pstn. |
17:23.28 | robl^ | Holos, sometimes echo comes from cheap phone / speaker phone on the pstn side. |
17:23.48 | Holos | robl^: I get echo while listening to ringtone. |
17:24.42 | Holos | My Sip client is x-lite and is using a headphone/mic combo |
17:24.43 | robl^ | Holos, I haven't seen/heard that one before. |
17:24.54 | *** join/#asterisk [Outcast] (~bill@c-24-218-94-11.hsd1.ma.comcast.net) |
17:25.35 | Holos | What are most peoples RX/TX gains set at for TM400 cards? |
17:26.22 | Ridgeback | Holos, im not sure gain has much to do with echo. thats usually a latency issue with your ISP |
17:27.08 | robl^ | right. gain is more for sound level |
17:27.59 | jakepdev | Manx - that SendDTMF turned out to be a bug in the Flash cmd |
17:28.14 | Ridgeback | time for bed for me... later guys! |
17:28.22 | zoa | cheers |
17:28.35 | zoa | bbl |
17:28.49 | *** join/#asterisk agent_sx (~agent_@12-220-171-226.client.insightBB.com) |
17:28.56 | jakepdev | if you attempt Flash on an IAX trunk, instead of just returning -1, it quits |
17:31.08 | *** join/#asterisk zoa (~zoa@pirus.securax.be) |
17:33.26 | *** join/#asterisk illek (~Mike@ip68-13-238-168.ok.ok.cox.net) |
17:33.44 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com) |
17:34.25 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
17:35.33 | *** join/#asterisk Bentley (~Bentley@S01060080c8135e6a.cg.shawcable.net) |
17:37.17 | Holos | robl^: In this case there is less then 1ms to my server, and then it goes over copper pstn. |
17:39.08 | gnmraju | does voicexml work with asterisk |
17:39.25 | file[laptop] | no, NEXT!!! |
17:39.35 | cypromis | but you are welcome to add voicexml support |
17:39.45 | Bentley | hi all, in queues.conf, setting "monitor-format=wav" will save the recordings using Monitor() defaults. Is there a way to also pass options (like basename & flags)? (v1-0) |
17:39.53 | *** part/#asterisk illek (~Mike@ip68-13-238-168.ok.ok.cox.net) |
17:40.58 | *** join/#asterisk jets (~brian@guardian.pmt.org) |
17:41.32 | slePP | what joy :> |
17:41.57 | Sedorox | hmmmmm |
17:41.57 | jf_ | why * does not transfert when i press # key |
17:41.58 | file[laptop] | what'cha up to? |
17:42.06 | tzanger | jf_: show application dial() |
17:42.29 | jf_ | tzanger: what you wanna see |
17:42.41 | foobos | jf_, you need to give certain flag to Dial() to enable #-transfer |
17:42.48 | jf_ | oh ok |
17:43.01 | tzanger | jf_: it's not what I want to see, it's what you need to see. :-) |
17:43.39 | jf_ | so i just create a context or put dial() what in the context i want |
17:44.19 | slePP | what'm i up to? |
17:44.21 | tzanger | A 911 dispatcher was reprimanded for responding to a mother's plea for help with an unruly child by saying: "OK. Do you want us to come over to shoot her?" |
17:44.27 | slePP | making SER go zoom |
17:44.36 | foobos | jf_, no.. you put t or T flag to dial-string.. but i've forgotten which one is it |
17:44.40 | tzanger | jf_: you need to read the asterisk handbook and get a good handle on the basics |
17:45.00 | file[laptop] | slePP: Yay |
17:45.07 | *** join/#asterisk juiceib269 (~juiceib26@24.236.130.31.bay.mi.chartermi.net) |
17:45.28 | robl^ | slePP, zoom? or BOOM!! ? |
17:45.34 | gnmraju | does voicexml work with asterisk |
17:46.11 | Sedorox | [13:39] <gnmraju> does voicexml work with asterisk |
17:46.12 | Sedorox | [13:39] <file[laptop]> no, NEXT!!! |
17:46.26 | Sedorox | so to repeat.. No.. it doesn't |
17:46.54 | fugitivo | will it work? :) |
17:47.33 | robl^ | Sedorox [13:39] <gnmraju> does voicexml work with asterisk |
17:47.34 | robl^ | Sedorox [13:39] <file[laptop]> no, NEXT!!! |
17:47.40 | *** join/#asterisk rene- (~root@200.106.49.195) |
17:47.59 | file[laptop] | maybe this is better |
17:48.03 | file[laptop] | gnmraju: no, NEXT!!! |
17:48.10 | Sedorox | lol |
17:48.31 | Wazb | once call established with Asterisk , after greeting if user press 1 then what format of extension will asterisk get |
17:48.51 | file[laptop] | Wazb: what? |
17:50.10 | rene- | hey |
17:50.57 | Wazb | i need to know when person call to asterisk server through DID , after hearing greeting , if he press 1 then will comes to * ? |
17:51.23 | file[laptop] | Wazb: DTMF tones go to asterisk yes, there is an application called Background which will allow you to make an IVR (a digit based phone menu)... |
17:51.34 | tzanger | Wazb look at the example dialplan, there's an entire IVR in there. |
17:51.39 | slePP | robl^: zooooooooooooom |
17:51.44 | *** part/#asterisk nitram (nitram@superblob.com) |
17:52.37 | *** join/#asterisk Bola_King (~john@62.175.14.244) |
17:53.18 | Sedorox | hmmmm |
17:53.39 | Holos | robl^: SOLVED: You need to define your channels after the echocancel=yes otherwise it just sets it to no. zap show channel 1 reported that there was no echo canceling enabled. |
17:54.01 | Bola_King | I'm new to this forum, is it ok to post a question related to td400 ? |
17:54.23 | rene- | i have a tdm400, and im seeing the trunks blocked every now and then, zap destroy channel does not seem to free the trunk, only restarting asterisk solves the problem, what could be causing my trunks getting blocked? |
17:55.28 | [Outcast] | has anyone seen anthem? |
17:55.34 | cypromis | he is in away |
17:56.01 | [Outcast] | has there been a fix for res_perl for 1.0.7? |
17:56.37 | RChadwell | Has anyone got two phone numbers to work with Broadvoice? I am still working on it and all calls go to the second sip registration - which sucks. |
17:57.21 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
17:57.57 | RChadwell | It is fun to search the web for comments on Broadvoice from like 2003 - I was hoping someone here has actually setup 2 sip connections |
17:58.04 | RChadwell | :) |
17:59.01 | robl^ | RChadwell, I have 2 BV accounts. works without a problem |
17:59.16 | RChadwell | will you post your sip.conf to pastebin? |
17:59.31 | RChadwell | or just the relevant register and initial contexts? |
17:59.42 | robl^ | RChadwell, just a sec |
17:59.45 | rene- | i also have another question, im running voice over a vtun vpn with IAX, sometimes voice just stops but the iax channel keeps itself up, the wan links are not heavily used but i understand that the data is travelling over the internet, and that data loss is likely, i was loooking at the drop count parameter, ihave set it up at 2 and well im seeing less of this issue, what is a good value for this param? |
17:59.48 | RChadwell | Thanks - lifesaver |
18:03.20 | Wazb | tzanger , where i can find that exaples |
18:03.27 | tzanger | /etc/extensions.conf |
18:03.31 | tzanger | er /etc/asterisk/extensions.conf |
18:04.25 | *** join/#asterisk ikey1 (ikey@220.226.29.74) |
18:05.14 | Wazb | tzanger , i am pointing my DID to * and it gives me greeting |
18:06.23 | file[laptop] | dejavu |
18:06.58 | rene- | dejavu you say |
18:07.16 | rene- | that only happens when they change something |
18:08.00 | Wazb | tzanger , but when i press 1 then it wont execute extern => 1,Dial(sip/1....) |
18:08.05 | file[laptop] | well they certainly didn't increase the average IQ of people in here |
18:08.25 | Sedorox | lol |
18:08.30 | rene- | i can see that |
18:08.38 | Sedorox | no.. but jbot turned into a agent... |
18:08.42 | rene- | nor did they improve response times |
18:08.47 | file[laptop] | indeed |
18:08.51 | vaewyn | jbot always was an agent :P |
18:08.54 | Sedorox | true... |
18:09.30 | robl^ | RChadwell, http://pastebin.ca/9556 that's the relevent bits |
18:09.38 | Wazb | tzanger , but when i press 1 then it wont execute extern => 1,Dial(sip/1....) |
18:10.20 | tzanger | Wazb: you need to read the handbook and work through some examples |
18:10.25 | tzanger | ~handbook |
18:10.26 | jbot | i guess handbook is http://www.digium.com/handbook-draft.pdf |
18:10.26 | bugbot | handbook is assigned nothing and reported nothing. |
18:11.08 | *** join/#asterisk egon_l (~egon@pc-10-19-104-200.cm.vtr.net) |
18:11.33 | *** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it) |
18:11.36 | vaewyn | we don't need 2 $#@%#$% bots |
18:12.15 | robl^ | me need 3 bots! |
18:12.30 | robl^ | there are always plenty of bots! :) |
18:13.58 | RChadwell | robl^, At the end of your register line you have a 10-digit extension. Does that mean that you route directly to that extension in extensions.conf? |
18:14.34 | Wazb | tzanger , this is what i have used in my extension file |
18:14.35 | Wazb | [test] |
18:14.35 | Wazb | exten => 416123456,1,Goto(test,s,1) |
18:14.35 | Wazb | exten => s,1,Answer |
18:14.35 | Wazb | exten => s,2,Ringing |
18:14.35 | Wazb | exten => s,3,Background(thnaks) |
18:14.36 | Wazb | exten => s,4,Wait(2) |
18:14.38 | Wazb | exten => 4455,1,Dial(SIP/4455,20,rt) |
18:14.41 | tzanger | Wazb: do not flood |
18:14.51 | Wazb | sorry |
18:14.53 | robl^ | RChadwell, all incoming SIP calls go to a single context for me. incoming-sip. then that context looks at the extension and directs it to a different context |
18:15.30 | tzanger | Wazb: don't use Wait() it doesn't listen for digits |
18:15.38 | tzanger | Background(silence/2) or Read() |
18:15.54 | tzanger | Wazb: have you gone through the handbook? have you gone through the default extensions.conf? |
18:16.39 | Wazb | yes i did |
18:17.38 | tzanger | Wazb: there are specific examples in the handbook IIRC |
18:18.33 | zoa | hey ho |
18:21.16 | Wazb | tzanger , in Asterisk Handbbok Version 2 , right? |
18:21.32 | tzanger | correct |
18:21.33 | tzanger | ~handbook |
18:21.34 | jbot | handbook is, like, http://www.digium.com/handbook-draft.pdf |
18:21.45 | bugbot | handbook is assigned nothing and reported nothing. |
18:25.22 | Wazb | thanks tzanger |
18:29.18 | RChadwell | Thanks robl^, your solution made my life a lot simpler. That is awesome! |
18:31.30 | tzanger | Wazb: chapter 4 specifically deals with the dialplan |
18:31.39 | *** part/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
18:33.30 | *** join/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
18:33.42 | *** join/#asterisk afrosheen (~afro@txprotoa22.august.net) |
18:35.05 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
18:35.14 | afrosheen | so what's going on these days |
18:35.17 | *** join/#asterisk NewSole (~david@i216-58-44-245.avalonworks.net) |
18:35.18 | afrosheen | channel = slow |
18:35.27 | *** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.res.rr.com) |
18:36.07 | Wazb | thanks again tzanger |
18:36.15 | NewSole | question... anyone here have a PRI with DID's |
18:36.31 | *** join/#asterisk _Brian (brian@unix01.voicenet.com) |
18:36.47 | afrosheen | has anyone done a robust meetme interface, like having people about to join a meetme record their name, then * announces their arrival in the meetme room? |
18:37.46 | _Brian | does anyone know if * has any time of audio detection? I have an application that needs to Flashhook a call to put them on hold and then dial another extension utilizing SendDTMF. The problem I am having is that * will continue to the next step even before the remote party answers. If I utilize a Dial string, then i use another channel...... |
18:37.54 | _Brian | um..time should be type :) |
18:38.29 | _Brian | a Wait would not be applicable either, since it could be several rings before the remote party answers |
18:38.59 | Juggie | how do i fix, chan_iax2.c:5067 socket_read: meta trunk cmd 1 received, I only understand 0 (perhaps the remote side is sending trunk timestamps?) |
18:39.42 | afrosheen | _Brian: a wait would work, because you'll want a finite time before the dial gives up and finishes your dial string |
18:40.53 | _Brian | afrosheen: that is fine, but if the person answers the call on the first ring, then they will be stuck waiting for the next prompt to be played out |
18:41.17 | _Brian | afrosheen: if they answer it after the 5th ring, then it might play fine, but the early answerer's would be penalized |
18:41.22 | afrosheen | _Brian: yeah true :( |
18:41.55 | _Brian | a Dial would work great for this, but I am forced to flash hook the extension and then send the DTMF digits... |
18:42.11 | _Brian | at that time, i am connected........so it is not typically a dial, since I am utilizing the same channel |
18:42.30 | *** join/#asterisk DrJolo (~chatzilla@217.153.194.10) |
18:43.02 | NewSole | When a call comes in off PRI group is there a wat to know what DID number they called |
18:43.29 | bjohnson | for anyone who cares .. just got asterisk running on a wrt54g (as reported by other users). Haven't had time to test quality or capacity yet |
18:43.34 | tzanger | NewSole: ${EXTEN} or ${DNIS} |
18:43.41 | _Brian | bjohnson: cool... |
18:43.46 | tzanger | or if you want the number that redirected them to the DID, ${RDNIS} |
18:43.53 | Sedorox | bjohnson: I thought you've always had one running? or was that someone else? |
18:43.55 | tzanger | bjohnson: nice |
18:43.59 | tzanger | Sedorox: that was jerjer I think |
18:44.04 | Sedorox | ah ok |
18:44.05 | Sedorox | maybe |
18:44.06 | bjohnson | bunch of others |
18:44.06 | tzanger | bjohnson: any major hurdles? |
18:44.19 | bjohnson | I was trying to get it to run but kept banging my head against space limitations |
18:44.29 | _Brian | afrosheen: any other ideas? |
18:44.29 | afrosheen | bjohnson: cool |
18:44.30 | Sedorox | lol |
18:44.35 | bjohnson | no major hurdles if you do it right the first time :) |
18:44.37 | afrosheen | _Brian: naw, I'm not that smart yet |
18:44.45 | _Brian | afrosheen: rofl |
18:44.48 | *** join/#asterisk Bonbon (~bonbon@83.146.53.34) |
18:45.10 | Bonbon | i haven't tried flash operator panel, but does anyone know if you can transfer calls with it? |
18:45.15 | bjohnson | I still have to set up a nfs share to use sound files and store voicemail messages |
18:45.22 | afrosheen | Bonbon: yeah you can |
18:45.26 | _Brian | anyone else? |
18:45.42 | bjohnson | but the working part of the system is stored on the flash of the wrt itself |
18:47.15 | Sedorox | Bonbon: I tried it.. I actually didn't like it too much... but thats just me... |
18:47.58 | The_Ape | hmm, i have a question.. when i type "sip show peers" in the CLI, in the name/username field everyone shows up as 1000/1000 and 1001/1001 and so on .. extept one that shows up as 1004/s. everyone has the same config. Any ideas? |
18:50.20 | afrosheen | The_Ape: monkey with the database command, see if you can catch a bad entry in it for that extension |
18:51.19 | _Brian | i guess that would be a no ....oh well..i will keep plugging away.. |
18:52.04 | Hogie | Im running Whitebox Linux 4... I just installed a TDM03 (3fxo), I have in rc.local to /sbin/modprobe wctdm then to /sbin/ztcfg -vvv, and when I do boot up, I get line 0: Unable to open master device '/dev/zap/ctl', but it works when I login as root. What can I do to fix it? |
18:52.06 | *** join/#asterisk riksta (~rick@81-178-248-131.dsl.pipex.com) |
18:52.52 | *** join/#asterisk brainchil (~daver@bigbrother.kdsi.net) |
18:53.00 | cypromis | Hogie: check README.udev in the zaptel source dir |
18:53.39 | brainchil | could some one point me to a faq/info/article on setting up a button on a polycom phone to do unattended transfers like a tdm pbx? |
18:53.40 | The_Ape | afrosheen: sip/registry/1004 IP-address:5060:120:s:sip:s@IP-address. The s is there.. where is he doing it wrong? he's still getting registerd? |
18:53.44 | brainchil | or tell me if it's possible |
18:54.09 | Hogie | cypromis: i did that, this is before login that it fails |
18:54.49 | brainchil | I know they can do it with software but I want to be able to have a transfer -> dial extension from the phone itself |
18:55.03 | brainchil | and searching google and voip-info I came up short |
18:55.33 | afrosheen | The_Ape: have a look at your sip.config and your extensions.conf, it's getting it from somewhere |
18:55.43 | afrosheen | The_Ape: woops I meant sip.conf |
18:58.26 | The_Ape | afrosheen: no "s" in either file. :/ |
18:58.28 | brainchil | anyone? |
19:01.20 | johnnyb | hehehehe -- one of my users just plugged BOTH the PC and LAN connection of the phone into the same hub |
19:01.31 | johnnyb | Now THAT will bring down a network. |
19:04.08 | *** join/#asterisk syle (~blah@wnpgmb02dc1-176-155.dynamic.mts.net) |
19:05.04 | The_Ape | afrosheen: The s was the remote asterisk's local incomming extensionname for my box. Thanks for helping :) |
19:05.14 | Blackthorn | For those that might be interested I am running an 802.11b network around my town. Currently 3 towers, 43 users with 27 of them using sipura phones connected to * here in my office. So far been working great. |
19:07.20 | vaewyn | Blackthorn: my home runs off a 6.1 mile 802.11b link :P SIP and IAX2 calls work great :P |
19:08.51 | bkw_ | haha |
19:08.59 | bkw_ | why do people really fail to understand what _. can do |
19:09.20 | Hogie | does digium still do free installation support? |
19:09.26 | Hogie | of their hardware? |
19:09.32 | bkw_ | you need help? |
19:09.59 | Hogie | yes, 1 stupid problem with a tdm card |
19:10.22 | funxion | yeah me too |
19:10.53 | funxion | I finally got the modules for my te110p to load |
19:11.35 | funxion | can someone help me with zapata.conf |
19:11.50 | *** join/#asterisk hohum (corbe@snoop.burghcom.com) |
19:12.12 | hohum | anyone who knows the asterisk codebase well enough point me in the right direction? |
19:12.18 | hohum | I want to disable loop detection |
19:12.26 | bkw_ | loop detection on what? |
19:12.26 | hohum | not sure what file(s) deal with that |
19:12.34 | hohum | bkw: SIP channels |
19:12.39 | bkw_ | chan_sip.c |
19:12.47 | file[laptop] | hahaha.... |
19:12.51 | bkw_ | and It shoudln't do the loop detected thing anymore |
19:12.58 | hohum | no? |
19:13.02 | bkw_ | let me look at the code |
19:13.03 | hohum | why? |
19:13.29 | Nugget | no! we won't let you! |
19:13.29 | bkw_ | well I recall this |
19:13.40 | bkw_ | you shouldn't call yourself |
19:13.45 | file[laptop] | it'll do a Loop Detected if a literal loop occurs... ie: asterisk sends out an invite, and it goes back to itself |
19:13.47 | bkw_ | you deal with local stuff in the local dialplan |
19:13.47 | bkw_ | duh |
19:13.47 | hohum | I'm not calling myself |
19:14.08 | bkw_ | you're calling another extension in your asterisk box |
19:14.11 | hohum | I'm calling a SIP UA that's registered to my SER box and has call forwarding enabled |
19:14.12 | bkw_ | its still a loop |
19:14.24 | bkw_ | show me CLI output |
19:14.28 | hohum | so it starts another phone call with the same call ID |
19:14.41 | bkw_ | OH |
19:14.43 | file[laptop] | what? |
19:14.47 | file[laptop] | what odd behavior |
19:14.47 | bkw_ | munge the callid in ser |
19:15.07 | hohum | according to my interpretation of the RFC loop detection is optional anyways |
19:15.32 | bkw_ | then look in chan_sip.c and comment out that part |
19:16.21 | hohum | guess I'll find out tonight when I go to upgrade this asterisk box :) |
19:16.40 | hohum | I'm going to move it from 1.0.7 to CVS-HEAD |
19:17.00 | Sedorox | which is newer, head or release? |
19:17.04 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
19:17.05 | hohum | head |
19:17.11 | Sedorox | whats more stable? lol |
19:17.13 | shido6 | head |
19:17.15 | *** part/#asterisk didz_ (didz_@200.218.192.52) |
19:17.15 | shido6 | and mo head |
19:17.20 | Sedorox | hmmmm |
19:17.31 | Sedorox | maybe I'll try out head... |
19:17.40 | _Brian | afro: looks like you can use BackgroungDetect, but it appears that does not work correctly :( |
19:17.48 | hohum | release would be more stable |
19:18.05 | hohum | I only want head</pun> |
19:18.24 | hohum | cos some changes were made to the way SIP Codecs are handled |
19:19.12 | MajestiK | I'm having some troubles with my x100p card, it doesn't seem to drop the line when someone hangs up, and I'm getting "hang up your phone now" voicemails |
19:19.26 | shido6 | hehee |
19:19.27 | vaewyn | So... how long until head becomes 1.1? ;P |
19:19.31 | bkw_ | hohum, those changes are in cvs-stable |
19:19.32 | bkw_ | btw |
19:19.39 | hohum | oh? |
19:19.40 | hohum | neat |
19:19.42 | bkw_ | yes |
19:19.46 | bkw_ | backported because it was a major issue |
19:19.49 | bkw_ | that needed to be fixed |
19:19.51 | funxion | I just set up a te110p t1 card and got to the point of configuring it in zapata.conf and zaptel.conf when I do zap show channels it shows me 23 channels for PRI but the card still has red light |
19:20.01 | funxion | can anyone point me in the right direction |
19:20.13 | bkw_ | funxion, show me your spanline |
19:20.51 | tzanger | bkw_: that's personal |
19:21.11 | _Brian | does anyone know if * has any time of audio detection? I have an application that needs to Flashhook a call to put them on hold and then dial another extension utilizing SendDTMF. The problem I am having is that * will continue to the next step even before the remote party answers. If I utilize a Dial string, then i use another channel...... |
19:21.58 | juiceib269 | funxion have you trieds the command ztcfg? |
19:22.49 | juiceib269 | opps is it a red light or is it blinking red? |
19:24.02 | vaewyn | Hmm... why is my span saying 'internal timing' when it should be getting it fromthe norhell? |
19:24.19 | vaewyn | It is working ok... so I won't argue but... seems odd |
19:27.16 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@d01m-1-164.d4.club-internet.fr) |
19:27.59 | hohum | bkw: thanks |
19:28.56 | Bonbon | afrosheen: so we can use a touchscreen tft monitor and use flash operator panel with no need to use a mouse |
19:28.56 | Bonbon | ? |
19:29.25 | rrk | question about did's? |
19:29.47 | vaewyn | FOP is cool... but wth flash and not just dhtml? |
19:29.57 | vaewyn | but anyways :P |
19:30.50 | funxion | shido6 you still there? |
19:30.54 | *** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net) |
19:31.05 | Bonbon | vaewyn: would my idea to use with a tft touchscreen mean that the operator doesn't have to touch a mouse? |
19:31.14 | nestAr | hrmm. monitor-join doesn't seem to wrok. |
19:31.27 | Bonbon | vaewyn: in order to transfer a call |
19:31.48 | _Brian | funxion: i think he disappeared |
19:31.55 | funxion | kewl |
19:32.04 | fearnor | <Bonbon> vaewyn: in order to transfer a call |
19:32.05 | fearnor | err |
19:32.53 | *** join/#asterisk P-Chan (~jpfingstm@68.142.66.200) |
19:33.23 | P-Chan | Hello. I'm using AMP and in my log I have ast_yyerror(): syntax error: syntax error; Input: |
19:33.24 | P-Chan | <PROTECTED> |
19:33.30 | P-Chan | I think it comes from: |
19:33.39 | *** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl) |
19:33.41 | P-Chan | extensions_additional.conf:exten => s,1,GotoIf($[${DIALSTATUS} = ANSWER]?4) ; |
19:33.51 | agent_sx | daper dan god damit |
19:33.53 | agent_sx | lol |
19:34.08 | vaewyn | Bonbon: correct |
19:34.09 | agent_sx | great moviee |
19:34.21 | P-Chan | Any idea what I should do about this? I hate "ignoring" errors - I can't seem to see it causing a problem tho. |
19:38.02 | *** join/#asterisk easimon (~easimon@baghira.kawo2.RWTH-Aachen.DE) |
19:39.39 | syle | can asterisk tell difference between an answering machine and a real person? |
19:39.56 | tzanger | syle: no |
19:42.14 | vaewyn | can tell a fax or modem... but not answering machine |
19:42.32 | tzanger | there is an app_machine() but it's not part of the standard set and I have no idea how well it actually works |
19:43.09 | Sedorox | I think we should make a app_whops that runs whenever something isn't configured right.. and output something like "Check your config, stupid" |
19:44.13 | tzanger | Sedorox: nah I am a huge fan of having everything default to the default context, and the default context be FORCED to have nothing but exten => _.,1,Wait(1), 2,Playback(please-configure-asterisk) 3,Goto(s,1) |
19:44.17 | hohum | answering machine detection is very difficult to do |
19:44.36 | hohum | the old school train of thought is to measure how much background noise there is on the call |
19:44.51 | hohum | because answering machines typically generate alot of white noise |
19:44.53 | Sedorox | ahaha |
19:44.56 | hohum | but that doesn't work too well anymore |
19:45.04 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-206-168-218-206.rockynet.com) |
19:45.05 | hohum | Cell phones do the same thing |
19:45.08 | vaewyn | tzanger: but then more people know about _. and that is evil ;P |
19:45.14 | tzanger | vaewyn: hmm this si true. |
19:45.16 | tzanger | fine then |
19:45.27 | Sedorox | hmmm |
19:45.37 | tzanger | s,1,all that and _X.,1,Goto(s,1) |
19:45.37 | *** part/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
19:46.54 | hohum | and I come from a telemarketing background :) so |
19:48.57 | rene- | what type of hardware could possibly service the DS3 card from digium? |
19:50.16 | vaewyn | quad opeteron :P |
19:50.19 | vaewyn | opteron even |
19:50.36 | vaewyn | just enough to split the traffic up for other servers :P |
19:50.51 | vaewyn | don't even think of landing G.729 on it ;P |
19:51.12 | *** join/#asterisk eKo1 (~bernd@207.42.191.67) |
19:51.44 | *** join/#asterisk WGFreewill (~chatzilla@24-75-221-174.miamfl.adelphia.net) |
19:52.32 | hohum | asterisk TDM :( |
19:52.49 | P-Chan | Is it possibly to forward a call from an asterisk server with a PRI to another via IAX2 trunking and have spandsp on the otherside of the trunk do fax detection? |
19:53.22 | *** join/#asterisk mgth (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
19:54.53 | dca[laptop] | anyone ever seen this error using res_config/realtime: |
19:54.54 | dca[laptop] | Apr 14 13:40:38 WARNING[8281]: res_config_odbc.c:277 realtime_multi_odbc: SQL Fetch error! |
19:54.54 | dca[laptop] | [SELECT * FROM extensions WHERE exten LIKE ? AND context = ? AND priority = ? ORDER BY exten] |
19:55.16 | dca[laptop] | it's like something snaps and this error just spews |
19:55.58 | *** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl) |
19:56.10 | slePP | does the extensions table exist and have fields 'exten', 'context' and 'priority'? |
19:56.14 | slePP | does the user accessing it have permission? |
19:56.28 | dca[laptop] | yes, and i think so |
19:56.39 | slePP | test the same query from isql |
19:56.40 | outtolunc | what version of mysql? |
19:56.45 | WGFreewill | iax jitterbuffer broken in CVS? |
19:56.53 | dca[laptop] | it works like a charm for a ton of calls, then, for not apparant reason, whamo! |
19:57.03 | slePP | which db backend is it using? |
19:57.13 | *** join/#asterisk darby_t (mua@dno210.neoplus.adsl.tpnet.pl) |
19:57.27 | dca[laptop] | slePP: res_odbc with mysql 5 |
19:57.38 | slePP | is mysql allowing enough connections from the host? |
19:57.48 | dca[laptop] | how can check? |
19:57.53 | slePP | dunno :> |
19:57.56 | dca[laptop] | hehe |
19:58.13 | file[laptop] | muahahahaha |
19:58.33 | dca[laptop] | slePP: and, lets say it isn't, would would that cause a bottle neck? or to break, like it currently is? |
19:58.39 | slePP | for a bill for 130.75 hours. they're arguing 124.25 hours of it |
19:58.57 | slePP | dca[laptop]: i could see if it tried to open another connection, and it was refused, it'd blow up |
19:59.06 | dca[laptop] | hmm |
19:59.19 | dca[laptop] | time to /join #mysql |
19:59.19 | Wazb | tzanger , i looked at handbook and exaples but i am confuse in on thing. |
19:59.29 | tzanger | slePP: that is why I almost universally prefer to quote for a specific block of work and get a P.O. |
20:00.27 | rene- | vaewyn: nice, so any codec translation or agi should be done on other machines, right? would you use TDMoE to forward the calls to the other machines? |
20:00.28 | file[laptop] | slePP: How are they arguing? |
20:00.51 | Wazb | tzanger , in SIp.config at top i changed context = demo , i got nothing when i dial my DID no. |
20:01.18 | tzanger | Wazb: because that is not how it works with PRI |
20:01.30 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
20:01.37 | vaewyn | rene-: TDMoE? just IAX2 it off :P |
20:02.30 | vaewyn | rene-: and yes... do EVERYTHING you can elsewhere |
20:02.30 | FuriousGeorge | hi all |
20:02.30 | *** join/#asterisk jcollie (~jcollie@lt16586.campus.dmacc.edu) |
20:02.32 | Wazb | tzanger , so i which config file i need to configure |
20:02.36 | rene- | that is simpler and nicer |
20:02.40 | jcollie | ~seen bkw_ |
20:02.52 | jbot | bkw_ is currently on #asterisk (4h 25m 6s). Has said a total of 26 messages. Is idling for 42m 39s |
20:02.52 | bugbot | seen bkw_ is assigned nothing and reported nothing. |
20:03.01 | tzanger | Wazb: you have a PRI. Quick quiz. What interface do PRIs come in on? |
20:03.26 | FuriousGeorge | whats preferable for a small * install. using a residential gateway "router" device, or using the asterisk box for the NAT, dhcp server and firewall? |
20:03.56 | WGFreewill | Anybody know how to troubleshoot DTMF across H323 |
20:04.18 | FuriousGeorge | i ask because i hear QoS to the outside world is mostly a myth, but i wonder about incomming QoS for voice streams |
20:04.20 | Wazb | tzanger , actually i have Cisco in which PRI is terminating and from there through i am forwarding call to * usign SIP |
20:05.34 | *** join/#asterisk stevej (~stevej@67.97.36.243) |
20:06.05 | rene- | what do you people think is the recommended sub 200 business phone? |
20:06.16 | jcollie | rene-: SPA-841 |
20:06.26 | nvrswork | if I have a X100P clone do I need to have ztdummy in the kernel? |
20:06.36 | hohum | clone? |
20:06.38 | nvrswork | yes |
20:06.42 | tzanger | Wazb: oh so you're really a glutton for punishment |
20:06.43 | hohum | people are cloning digium cards now? |
20:06.54 | jcollie | nvrswork: no, you shouldn't |
20:06.59 | nvrswork | thanks |
20:07.15 | hohum | what kind of horse shit is that? |
20:08.53 | brainchil | how do you do an unattended transfer from a sip phone on asterisk? |
20:08.57 | Wazb | tzanger , anything wrong? |
20:09.05 | TomL | ~seen manxpower |
20:09.11 | jbot | manxpower <~eric@adsl-35-236-60.msy.bellsouth.net> was last seen on IRC in channel #asterisk, 3h 16m 59s ago, saying: 'I'm outta here'. |
20:09.11 | bugbot | seen manxpower is assigned nothing and reported nothing. |
20:09.11 | brainchil | or rather is it possible |
20:09.15 | dca[laptop] | any mysql guru's here? need to know if there is a bottleneck in mysql my.cnf that is causing asterisk to spew errors... |
20:09.18 | hohum | I wish someone would make a card that does hardware TDM and works with PRIs in Asterisk |
20:09.26 | tzanger | Wazb: my first guess is that * is not seeing DTMF from the Cisco |
20:09.34 | FuriousGeorge | let me ask this way: does anyone use their * server as their NAT and firewawll and pppoe gw etc |
20:09.43 | FuriousGeorge | is that a bad idea? |
20:09.44 | tzanger | don't use inband unless you can't help it, and if you can't help it, you must only use ulaw between the cisco and asterisk |
20:09.50 | rene- | brainchil: sometimes the phone implements those things for you |
20:10.03 | brainchil | rene-: How and what phones? |
20:10.20 | rene- | siemens optipoint 400s can do it |
20:10.29 | rene- | but you could also do it in asterisk |
20:10.29 | P-Chan | Is it possibly to forward a call from an asterisk server with a PRI to another via IAX2 trunking and have spandsp on the otherside of the trunk do fax detection? |
20:10.35 | rene- | with a parking extension |
20:10.46 | *** join/#asterisk ManxPower (~eric@stirprop-S0-0-0-26.ndcr2.datasync.net) |
20:10.59 | tzanger | P-Chan: yes but you must be very careful... spandsp on the far side (i.e. an IAX2 hop away from the PRI) will be spotty |
20:11.00 | rene- | but you should only do # EXTEN |
20:11.08 | rene- | and it should be taken care of |
20:11.11 | ManxPower | OK eveyone I'm calling in some of my Good Asterisk Karma |
20:11.29 | tzanger | ManxPower: after that exten -> h bounty you have no karma |
20:11.35 | tzanger | or rather it's dangerously low |
20:11.37 | *** join/#asterisk ritesh (~ritesh@natint3.juniper.net) |
20:11.45 | hohum | hey |
20:11.49 | brainchil | rene-:I'll look at that .. is there a faq/howto someplace that you could point me ... I have been unable to locate one |
20:11.51 | hohum | a juniper employee :) |
20:11.56 | hohum | can I get a free M20? |
20:12.14 | ManxPower | I'm having a problem with Call Parking. Call comes into an extension which runs a macro. The user transfers to the call parking extension. If nobody picks up the call goes to extension "s", not the correct extension that parked it. |
20:12.19 | ManxPower | tzanger, Ha! |
20:12.25 | P-Chan | tzanger: so it would be better to have the asterisk server @ the PRI receive the faxes (perhaps on a dedicated DID? |
20:12.33 | tzanger | ManxPower: without the macro does it work? |
20:12.34 | *** part/#asterisk ritesh (~ritesh@natint3.juniper.net) |
20:12.37 | *** part/#asterisk jcollie (~jcollie@lt16586.campus.dmacc.edu) |
20:12.41 | tzanger | P-Chan: very much so, yes |
20:12.43 | tzanger | but be careful |
20:12.48 | hohum | I guess that was a big fat "no" |
20:12.49 | hohum | :( |
20:12.51 | tzanger | I have had spandsp crash out asterisk the odd time |
20:13.31 | P-Chan | tzanger: Oh...so it might be a problem where the asterisk server is mission critical. I may not want to do it like that... hmm... |
20:13.42 | rene- | how does one go about blocked analog zap channels |
20:13.56 | rene- | i was having issues with spandsp and rxfax |
20:14.09 | ManxPower | tzanger, you DID have to ask a useful question, huh? I assume it does work, but I'll check |
20:14.15 | rene- | i got rid of all of it (rxfax) |
20:14.26 | rene- | but some calls still block my outbound zap trunks |
20:14.32 | tzanger | ManxPower: :-) |
20:14.38 | tzanger | P-Chan: correct. |
20:15.12 | P-Chan | tzanger: Thanks. |
20:15.36 | ManxPower | tzanger, checking now |
20:15.37 | rene- | whats the deal with zap destroy channel. i t doesnt really do anything for me |
20:15.54 | gnmraju | exit |
20:18.10 | n4y | Does polycom ip 300 phone work with Asterisk? |
20:18.25 | ManxPower | tzanger, Nope! Doesn't work! |
20:18.30 | ManxPower | CVS 1.0.x |
20:18.40 | Wazb | tzanger , i addedd dtmfmode = inbad in SIP.conf file and it works |
20:19.14 | FengShui | is there any easy way to signal a flash to a zap channel that's briged to a cisco 7960? |
20:19.16 | tzanger | rene-: what do you expect it to do, shoot smoke out the TDM card? |
20:19.22 | FengShui | I want to flash the zap channel from the 7960l. |
20:19.24 | rene- | heh |
20:19.36 | *** join/#asterisk bah (048830696@ACA1C854.ipt.aol.com) |
20:19.41 | tzanger | Wazb: make sure you do not use anything other than ulaw in that case |
20:20.11 | rene- | tzanger: well unblocking the trunks would be nice |
20:20.12 | *** join/#asterisk riksta (~rick@81-178-248-131.dsl.pipex.com) |
20:20.19 | fugitivo | one question, I need asterisk to answer a call (zap channel), and play the IVR, then, I need to redirect the call to another pbx, i was trying to use flash and senddtmf, but that way i lose the control of the call, is any way to do that? |
20:20.39 | rene- | you do dial fugitivo |
20:20.47 | rene- | like dial(C_Party) |
20:20.51 | fugitivo | can't do that |
20:20.54 | rene- | asterisk will bridge the call |
20:20.56 | ManxPower | tzanger, I find it hard to believe that call parking is really broken in 1.0.x. Let me investigate more |
20:20.57 | fugitivo | the zap channel is in use |
20:21.00 | _Brian | rene- if you use dial, you will connect two channels |
20:21.09 | rene- | he said redirect |
20:21.11 | _Brian | rene- he wants to use one channel for this..... |
20:21.16 | rene- | my mistake |
20:21.18 | _Brian | sounds like my problems..somewhat |
20:21.26 | _Brian | does anyone know if * has any time of audio detection? I have an application that needs to Flashhook a call to put them on hold and then dial another extension utilizing SendDTMF. The problem I am having is that * will continue to the next step even before the remote party answers. If I utilize a Dial string, then i use another channel...... |
20:21.37 | bjohnson | what is res_crypto.so used for? |
20:22.02 | Wazb | thnaks tzanger |
20:22.15 | fugitivo | _Brian: it's a similar problem |
20:22.20 | Corydon-w | bjohnson: for rsa authentication |
20:22.35 | _Brian | fugitivo: yup..the problem i am seeing is that the * system does not handle call progress when doing a FlashHook... |
20:23.11 | _Brian | fugitivo: all call progress is built into the Dial Command, but this utilizes another channel..... |
20:23.48 | fugitivo | yes, i don't want to use another channel |
20:24.00 | _Brian | fugitivo: yup..i hear ya... |
20:24.24 | _Brian | what do you want to do with the call once you flashhook to send to the other pbx |
20:24.36 | ManxPower | tzanger, http://pastebin.ca/9566 |
20:24.55 | *** join/#asterisk Tili (~Tili@202-133-65-206-dialup.sat.net.pk) |
20:25.07 | fugitivo | i only want to detect if it's busy, and return to the ivr |
20:25.18 | fugitivo | if the call is answered, then hungup the channel |
20:25.36 | bjohnson | I need to delete some files on this wrt54g to make room for the nfs package so I can use the fileserver |
20:25.53 | ManxPower | <PROTECTED> |
20:25.55 | bjohnson | how about res_agi.so? is that needed? |
20:25.59 | ManxPower | read that carefully |
20:26.06 | FuriousGeorge | whats preferable for a small * install. using a residential gateway "router" device, or using the asterisk box for the NAT, dhcp server and firewall? |
20:26.16 | _Brian | fugitivo: sounds like you are looking for the same as I am...call progress detection with a Flashook |
20:26.20 | FuriousGeorge | i ask because i hear QoS to the outside world is mostly a myth, but i wonder about incomming QoS for voice streams |
20:26.27 | rene- | pointers on dealing with blocked analog zap trunks? |
20:26.35 | tzanger | ManxPower: 005 == Parked SIP/0004f200cf85-a-7187 on 3516. Will timeout back to toll-access,s,1 in 30 seconds |
20:26.36 | ManxPower | you can't do QoS on incoming packets, only transmitted packets |
20:26.41 | tzanger | it looks like it's specifically doing that |
20:26.50 | fugitivo | _Brian: it seems there's no solution... |
20:26.50 | ManxPower | tzanger, Yeah. why? 8-) |
20:27.00 | tzanger | Park says it's being called with empty parameters |
20:27.04 | dca[laptop] | anyone ever seen this error: |
20:27.05 | tzanger | does it default to s? |
20:27.06 | dca[laptop] | Apr 14 13:40:38 WARNING[8281]: res_config_odbc.c:277 realtime_multi_odbc: SQL Fetch error! |
20:27.06 | dca[laptop] | [SELECT * FROM extensions WHERE exten LIKE ? AND context = ? AND priority = ? ORDER BY exten] |
20:27.10 | bjohnson | what about app_adsiprog.so? what does that do? |
20:27.25 | [TK]D-Fender | FuriousGeorge : VoIP through NAT is a PITA. Save yourself some suffering and use your * box as your router |
20:27.29 | FuriousGeorge | manxpower, so i could use a residential "router" gw and there is no reason it wouldnt work as well as using a nix machine, or the server itself to do the NAT, etc |
20:27.33 | _Brian | fugitivo: there is always a solution......just gotta find it :) |
20:27.49 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
20:28.10 | ManxPower | FuriousGeorge, There are some ways to help with packet priority, but it's complex |
20:28.13 | FuriousGeorge | [TK]D-Fender: going to wikipedia to see what PITA is. you would do it on the same box as your * server (small install) or is that generally a bad idae |
20:28.23 | [TK]D-Fender | PainInTheAss |
20:28.33 | FuriousGeorge | gotcha |
20:28.35 | fugitivo | anyone knows the solution? :) |
20:29.02 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
20:29.07 | [TK]D-Fender | FuriousGeorge : Yeah, use your * box as a router (NAT on iptables). I do it and everything works great with 3 shell lines to set it up. |
20:29.32 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
20:29.32 | *** mode/#asterisk [+o bkw_] by ChanServ |
20:29.36 | FuriousGeorge | interesting b/c linux people always say that a server should be specialized |
20:29.55 | FuriousGeorge | one of my friends said "your firewall takes all the crap" so you dont want to put critical stuff on it |
20:29.56 | [TK]D-Fender | oF COURSE i COULD PROBABLY BE syn'D OUT OF EXISTANCE, BUT i LIKE TO LIVE ON THE EDGE :d |
20:30.10 | _Brian | fugitivo: if i find it...i will let you know :) |
20:30.14 | stevej | D-Fender: Can you pass on the shell lines? |
20:30.21 | shmaltz | fugitivo, what you trying to do? |
20:30.29 | fugitivo | FuriousGeorge: it's true, but if the crap affects the firewall, then the crap will affect all you'r network ;) |
20:30.32 | [TK]D-Fender | FuriousGeorge : I use mine as my HTPC server too as well as X-10 :D Don't worry about eggs in a basket |
20:30.55 | _Brian | shmaltz: he is looking for call supervision on a Zap channel when a Hookflash is sent... |
20:30.55 | [TK]D-Fender | stevej : Nope, I'm outta here unfortunately.... I'm here often though. |
20:30.58 | rene- | FuriousGeorge: we use it as vpn and router also |
20:31.06 | [TK]D-Fender | Later people! |
20:31.27 | shmaltz | like chanspy? |
20:31.38 | FuriousGeorge | lol, im a big fan of the one box does everything approach. i love hearing that you use it as an htpc too |
20:31.51 | fugitivo | shmaltz: asterisk is answering the calls, playing the IVR, and then it transfers the call to another PBX, i'm using flash and senddtmf for that, but the problem with that, is the busy signal, once i transfer the call, i lost the control of it, so i can't do anything if the extension of the other pbx is busy |
20:32.13 | FuriousGeorge | [TK]D-Fender: X-10, as in security cameras |
20:32.14 | fugitivo | shmaltz: i want to check if the other pbx is busy, if it's busy, then play again the IVR |
20:32.51 | shmaltz | what is the make of the other PBX? |
20:33.32 | shmaltz | b/c on some of them you could have asterisk set up as a foreign PBX, or VM system, and program it to always use DTMF, even for busy |
20:33.40 | fugitivo | shmaltz: it's just like that, i can't use asterisk for everything yet |
20:33.59 | shmaltz | fugitivo, that I understand, by question however is what PBX is it? |
20:34.17 | shmaltz | can you program the PBX to use an external VoiceMail system? |
20:34.41 | fugitivo | no :) |
20:34.43 | ManxPower | tzanger: This is a little simplier and clearer: http://pastebin.ca/9568 |
20:34.55 | fugitivo | its a panasonic 616 |
20:35.10 | shmaltz | fugitivo, you could with pana |
20:35.20 | shmaltz | just let me hang up on this phone call |
20:35.25 | fugitivo | okey |
20:35.49 | tzanger | 023 == Starting SIP/0004f201e463-a-7650 at toll-access,s,1 failed so falling back to exten 's' |
20:35.52 | tzanger | <PROTECTED> |
20:35.55 | tzanger | I don't like that |
20:36.01 | tzanger | and Park is still saying "I *will* go to s" |
20:36.04 | tzanger | so that is where it's broken |
20:36.17 | *** join/#asterisk jcollie (~jcollie@lt16586.campus.dmacc.edu) |
20:36.39 | elric | http://pastebin.ca/9569 <---- can someone please take a look at this and tell me if I am doing anything wrong? |
20:37.52 | blitzrage | tzanger: get back to work |
20:38.22 | ManxPower | tzanger, Yeah. Looks like it actually is broken. This is bad. |
20:38.45 | elric | basically the macro in that link I posted never gets executed once the call is connected |
20:39.06 | *** join/#asterisk Juxt (~Juxt@adsl-068-213-216-087.sip.bct.bellsouth.net) |
20:39.12 | Juxt | good afternoon |
20:39.17 | *** join/#asterisk swente (faOsXUs6pl@hal.infinitumb.de) |
20:39.26 | swente | 'lo |
20:39.32 | Juxt | can someone suggest a linux or bsd based firewall distro? |
20:39.49 | elric | Juxt yes www.m0n0.ch |
20:39.53 | shmaltz | elric, whats that machinedetect command |
20:40.29 | elric | shmaltz, http://www.thenetbrain.com/files/app_machinedetect.c <-- that is an answering machine detect |
20:40.35 | elric | ion app |
20:41.20 | fugitivo | Juxt: openbsd |
20:41.51 | swente | before going any deeper into voip-matter, i'd like to know what bandwith a voip-connection needs for say .. isdn telephony quality, and what ip-latency is considered as the minimal requirement. |
20:41.56 | Juxt | <PROTECTED> |
20:42.00 | shmaltz | elric, why u using transfer and not dial? |
20:42.20 | fugitivo | Juxt: then what are you asking for? livecd? |
20:42.33 | Juxt | monowall probably will work just fine |
20:42.42 | dca[laptop] | anyone from digium around? |
20:42.46 | Juxt | i tried smoothwall but it doesn't have traffic shaping |
20:42.47 | Juxt | monowall does |
20:42.49 | Juxt | so i think i am set |
20:42.51 | dca[laptop] | or from voipsupply? |
20:42.52 | *** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl) |
20:43.15 | elric | shmaltz, thats an outgoing call made from the manager interface, if a machine is detected the call is to be transferred to an extension |
20:43.36 | elric | but that will be eventually |
20:43.51 | Poincare | where do I define the sounds dir? |
20:44.02 | elric | right now i just want machine detect to woek and it doesnt. the macro never gets executed. |
20:44.59 | shmaltz | what do you mean never gets executed, you dont see it doing step 1 ? |
20:45.41 | elric | nope shmaltz , it is supoosed to display a message on the CLI, it never does. |
20:46.11 | elric | if i just do machinedetect it shows output on the CLI |
20:46.11 | shmaltz | then you are not executing it right from the manager interface, try using a direct exten |
20:46.32 | elric | that is a direct exten shmaltz |
20:47.24 | elric | if it works here, i will use it in my manager script. |
20:47.27 | shmaltz | elric you have something like this: |
20:47.29 | shmaltz | exten => 1234,1,Macro(detect) |
20:50.16 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-199-204.dsl.scarlet.be) |
20:50.17 | elric | hrm, MachineDetect actually needs to work on a connected outgoing call but I guess i could make that work somehow |
20:50.17 | _Brian | dont you just love when someone says "Yeah, I can help you with your issue, just email me"....you email the specifics, and then you get a email back telling how the rates and how much they want to charge you for the help the offered for free.... |
20:50.37 | shido6 | I never offered help for free, _Brian |
20:50.40 | _Brian | how sad.. |
20:50.46 | shido6 | it is |
20:51.29 | _Brian | you never indicated that you were looking to consult either.. |
20:52.17 | _Brian | i am glad i included some configs...for your later use....bleh |
20:52.46 | tzanger | _Brian: relax, shido6's a regular here |
20:53.04 | _Brian | tzanger: oh i am relaxed...:) |
20:53.13 | tzanger | I very seriously doubt he's interested in stealing your tech |
20:53.18 | _Brian | i just hate misrepresentations |
20:53.33 | Sedorox | hrm |
20:53.37 | _Brian | whatever.. |
20:54.10 | *** part/#asterisk Juxt (~Juxt@adsl-068-213-216-087.sip.bct.bellsouth.net) |
20:54.38 | smiley- | is there any problems with softphones and #NN# extensions? I just can't get them to work... |
20:55.17 | shido6 | assumption is the mother of all fuckups |
20:55.21 | elric | ah I wish techie were around |
20:55.43 | smiley- | with my regular phone connected to a SIP-box there is no problem.. but from all the softclient I have tested it fails.. Sjphone, X-lite and SIPS |
20:55.44 | _Brian | i will have to quote you on that one shido6 |
20:55.54 | Sedorox | hrm |
20:56.25 | _Brian | rofl..now i will get a email saying I have to royalty fees for quoting... |
20:57.31 | sean | hello, all. I'm definitely a newbie when it comes to asterisk, but can generally find my way around. I'm trying to connect xlite to my asterisk. I get "Apr 14 16:57:01 NOTICE[6150]: chan_sip.c:7532 handle_request: Registration from 'sean <sip:sean@caedmon.net>' failed for '10.20.30.100'" Can someone please point me in the right direction? |
20:57.33 | elric | _Brian, i fail to see how someone could charge you royalty for colloquial forms of speech? |
20:58.00 | _Brian | elric: it was a joke.... :) |
20:58.15 | elric | _Brian, ah |
20:58.16 | Nugget | sean: make sure sip.conf is well-formed and make sure you're getting the password right. |
20:58.21 | _Brian | maybe not a good one :) |
20:58.23 | syle | Apr 14 13:58:04 WARNING[23628]: res_musiconhold.c:565 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. |
20:58.26 | syle | what is this |
20:58.29 | *** join/#asterisk izo (~izo@izo.warpl.ipxxi.pl) |
20:58.33 | shido6 | sean, are you using a type=friend in your sip.conf? |
20:58.35 | shmaltz | sean, you are doing something wrong with the registration |
20:58.36 | shido6 | pastebin.ca your config |
20:58.45 | shido6 | and reply here with the pastebin link they provide you |
20:58.51 | sean | sec |
20:59.06 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
20:59.26 | sean | yes, type=friend for [xlite] |
20:59.30 | sean | I'll paste |
20:59.32 | elric | ah techie i was looking for you, if you have a minute or two to spare? |
20:59.33 | Nugget | for [xlite]? |
20:59.38 | Nugget | you're trying to log in as [sean] |
20:59.38 | shido6 | break it out to a peer and user for the xlite user |
20:59.43 | techie | elric: sure. |
20:59.45 | shido6 | users dont need hosts and peers dont need contexts |
20:59.57 | shido6 | and reload chan_sip.so at the CLI |
21:00.27 | syle | anyone running fedora core 3 with no problems with asterisk? |
21:00.55 | RChadwell | I am |
21:01.00 | Nugget | it's impossible to run linux with no problems. :) |
21:01.04 | sean | ok, I'll try that. Thanks |
21:01.54 | |Vulture| | fc3 is fine |
21:02.06 | |Vulture| | fc3+te110p == problems but I worked around them |
21:02.23 | |Vulture| | but its also because its new hardware from digium |
21:02.53 | *** part/#asterisk RChadwell (~rob@rrcs-24-227-48-86.se.biz.rr.com) |
21:03.03 | bannerman | I don't like the hardware that I'm using for my asterisk box, and another server became available, and I want to load a new system from scratch. Any advice for Linux distro? |
21:03.27 | Sedorox | Gentoo.... |
21:03.41 | bannerman | Sedorox: seriously? |
21:03.48 | Sedorox | I love gentoo |
21:04.09 | Sedorox | Gentoo.. Debian... |
21:04.13 | Nugget | asterisk doesn't care, so use whatever pleases you. anything beyond that is a religious decision. |
21:04.14 | fugitivo | Gentoo! |
21:04.15 | Sedorox | my top choices |
21:04.19 | Sedorox | or FBsd |
21:04.31 | elric | i use FreeBSD with * |
21:04.34 | funxion | hey |Vulture| |
21:04.40 | Sedorox | same here... |
21:04.49 | Sedorox | my * boxes are actually fbsd |
21:04.51 | *** join/#asterisk RoyK (~roy@143.80-202-166.nextgentel.com) |
21:04.53 | RoyK | hm |
21:05.00 | funxion | got past the module problem |
21:05.05 | funxion | got another problem now |
21:05.17 | RoyK | seems to me a single xeon 3.0 can easily run two te410p cards |
21:05.29 | funxion | trying to configure PRI to connect to meridian opt 81c |
21:05.35 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfnji.dialup.mindspring.com) |
21:05.39 | funxion | anyone ever done this before |
21:07.25 | sivana | is Digium going to have a single span card with built-in echo can? |
21:07.42 | RoyK | is jesus going to return as a muslim? |
21:07.57 | sean | potentially stupid question: how do I decrease verbosity at the CLI |
21:08.12 | drumkilla | set verbose 0 |
21:08.21 | sean | thanks.. sorry for the silly question |
21:08.24 | RoyK | :) |
21:08.26 | drumkilla | no problem, it's not silly |
21:08.38 | RoyK | stupid is as stupid does |
21:08.39 | RoyK | :P |
21:09.16 | drumkilla | <PROTECTED> |
21:09.17 | drumkilla | oops! |
21:09.19 | drumkilla | :p |
21:09.28 | syle | that ztdummy.o doesn;t even compile by default for rh9 |
21:09.43 | elric | ok i am beyond my wits end |
21:09.49 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
21:10.04 | drumkilla | syle: it doesn't compile by default, period |
21:10.29 | sivana | does any mfg sell a single T1 echo can? |
21:10.36 | *** join/#asterisk Blackvel (~blackvel@dsl-082-083-172-135.arcor-ip.net) |
21:11.21 | Blackvel | lo, have no connected my ata (zap pbx crashed) to * and changed extensions.conf for using now a macro. anyone wants to give me a test call? fwd, sipgate or nikotel is fine |
21:12.03 | dca[laptop] | most boxes are 3.3v, yes? |
21:12.36 | shido6 | next |
21:12.47 | shido6 | brb |
21:14.13 | *** join/#asterisk Syncros (~sysop@noc.routermonkey.net) |
21:14.23 | *** join/#asterisk dave_mwi_ (~dave_mwi@adsl-11-102-74.mia.bellsouth.net) |
21:14.46 | dave_mwi_ | anyone know off-hand how many auto atendants asterisk can handle at once? |
21:15.36 | *** join/#asterisk Fabrizioxxx (1002@ip-138-106.sn1.eutelia.it) |
21:15.49 | drumkilla | 021980293840234823!!!!!! |
21:15.50 | *** join/#asterisk jf_ (~jeanfranc@modemcable077.187-80-70.mc.videotron.ca) |
21:15.54 | Blackvel | oh cool |
21:15.56 | *** part/#asterisk Fabrizioxxx (1002@ip-138-106.sn1.eutelia.it) |
21:16.05 | Blackvel | called my pstn voip number and it seems to work :) |
21:16.16 | *** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
21:16.39 | |Vulture| | Blackvel: thats just the beggining |
21:16.42 | harryvv | Has anyone sucessfully tested astrisk on a ide flash card? |
21:16.44 | Blackvel | nope |
21:16.44 | *** part/#asterisk swente (faOsXUs6pl@hal.infinitumb.de) |
21:16.48 | Blackvel | my pbx crashed |
21:16.55 | Blackvel | had to reconfigure my complete * |
21:17.02 | P-Chan | Ok, instead of trying to figure ways to do it and then ask questions, let just throw this out there: 1 asterisk server w/ 2 PRIs, IAX2 Trunk to other remote asterisk servers, we want to have fax capabilities @ the sites with the remote asterisk servers, what's the best way to do this? |
21:17.11 | harryvv | did not backup your config files? |
21:17.17 | jf_ | Someone know why i can transfert from sip to sip or iax to sip but i can't transfert to another sip while talking on zap channel (pstn) |
21:17.19 | Blackvel | |Vulture|: i am no newbie :) |
21:17.34 | Blackvel | not my asterisk pbx, but my telco pbx |
21:17.35 | fearnor | sivana: not that i know of |
21:17.47 | fearnor | but i wish asterisk had a non-gay echo canceller so it wouldn't be necessary |
21:17.49 | *** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) |
21:17.50 | Poincare | where do I define the sounds dir? |
21:17.51 | fearnor | g.168 uber alles |
21:18.08 | sivana | fearnor: yes, I agree |
21:18.09 | Blackvel | i can not use my phones anymore to call into * (call thru). too bad ;) |
21:18.16 | harryvv | fearnor, there are third party echo cancelers on the market |
21:18.22 | drumkilla | Poincare: don't think it's configurable ... /var/lib/asterisk/sounds/ |
21:18.37 | Blackvel | okies, cu tomorrow |
21:18.42 | |Vulture| | Blackvel: oh nice... how much did that run you? |
21:18.52 | Poincare | drumkilla: it's located elsewhere in the debian packagers |
21:18.57 | sivana | harryvv: is there one that does a single t1 |
21:19.14 | Poincare | is the location available as a predefined variable? |
21:19.18 | harryvv | sil, I think so ask paulc I think he told me one one once. |
21:19.30 | fearnor | harry: those are usually pretty big things |
21:19.30 | *** part/#asterisk jcollie (~jcollie@lt16586.campus.dmacc.edu) |
21:19.38 | fearnor | and cancel 20 t1s at a time ;) |
21:19.44 | |Vulture| | fearnor: option AGGRESSIVE_SUPPRESSOR solved most my problems |
21:19.45 | drumkilla | Poincare: oh. :) |
21:20.03 | fearnor | asterisk echo canceller is *broken*, period end of story. |
21:20.16 | fearnor | aggressive or not |
21:20.21 | jf_ | Someone know why i can transfert from sip to sip or iax to sip but i can't transfert to another sip while talking on zap channel (pstn) |
21:20.31 | |Vulture| | fearnor: broken implies it worked at one time |
21:20.39 | harryvv | fearnor, my echo is totally gone on my x100p |
21:20.44 | harryvv | none zero. |
21:20.45 | *** join/#asterisk tessier_ (~treed@210.245.99.64) |
21:21.57 | P-Chan | Anyone know how to do "SIP Trunking"? I can't find any info in voip-info.org wiki, maybe its called something else?? |
21:22.06 | *** join/#asterisk smurfix (~smurf@smurfix.developer.debian) |
21:22.39 | syle | Apr 14 13:58:04 WARNING[23628]: res_musiconhold.c:565 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. |
21:22.42 | syle | this a bad error? |
21:22.59 | file[mac] | it's a warning |
21:23.04 | fearnor | syle: i don't know. do you mind choppy sound? |
21:23.07 | syle | this server is just used for sip/aix only stuff no cards |
21:23.23 | Corydon-w | ~ztdummy |
21:23.24 | jbot | ztdummy is probably zaptel timing source which uses a usb-ohci compatible usb controller as source. (part of zaptel cvs) |
21:23.25 | bugbot | ztdummy is assigned nothing and reported nothing. |
21:23.38 | syle | i tried loading it and got problems |
21:23.44 | Corydon-w | Ooops... that should be usb-uhci |
21:24.21 | harryvv | probebly? it is |
21:24.32 | syle | going to paste 4 lines sorry if it is not allowed |
21:24.34 | syle | ice:~/voip/zaptel# insmod ./ztdummy.o |
21:24.34 | syle | ./ztdummy.o: unresolved symbol zt_unregister |
21:24.34 | syle | ./ztdummy.o: unresolved symbol zt_transmit |
21:24.34 | syle | ./ztdummy.o: unresolved symbol zt_receive |
21:24.34 | syle | ./ztdummy.o: unresolved symbol zt_register |
21:24.35 | syle | ice:~/voip/zaptel# |
21:24.39 | Sedorox | ~pastebin |
21:24.40 | jbot | pastebin is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca |
21:24.44 | *** join/#asterisk Rick_Hunter (~rhunter@01-098.008.popsite.net) |
21:24.45 | bugbot | pastebin is assigned nothing and reported nothing. |
21:25.10 | Sedorox | hehe |
21:25.23 | Sedorox | I _think_ you might need another module loaded |
21:25.25 | |Vulture| | isnt he suppose to be in -dev? |
21:25.43 | file[mac] | syle: try modprobe ztdummy |
21:26.23 | syle | modprobe: Can't locate module ztdummy |
21:26.23 | Sedorox | modprobe ./ztdummy |
21:26.41 | syle | same thing |
21:26.49 | file[mac] | syle: did you, install the modules? |
21:27.00 | bkw_ | depmod -a |
21:27.04 | bkw_ | modprobe ztdummy |
21:27.27 | syle | i thought the make install would have |
21:27.29 | syle | but it didn;t |
21:27.31 | |Vulture| | bkw_: will there be a release of ChanSpy for v1-0? |
21:27.44 | syle | bad enough i had to do a make ztdummy just to get the .o file |
21:28.13 | Sedorox | .... |
21:28.22 | bkw_ | |Vulture|, NO |
21:28.34 | syle | so can i manually copy it somewhere maybe |
21:28.39 | bkw_ | ztdummy builds by default on 2.6 |
21:28.48 | syle | not on redhat 9 |
21:28.59 | |Vulture| | bkw_: hehee okay :P |
21:29.24 | dca[laptop] | bkw_: got a sec? |
21:29.31 | dca[laptop] | having realtime w/ odbc errors |
21:29.38 | dca[laptop] | wondering if my mysql5 is too blame |
21:29.49 | *** join/#asterisk PTG123 (~PTG123@66.213.239.122) |
21:29.57 | bkw_ | no time |
21:29.58 | syle | hmmm |
21:30.07 | syle | apparentely it install no modules anywhere on make install |
21:30.18 | *** join/#asterisk Egonis (~chultay@69.194.211.129) |
21:30.19 | jf_ | to transfert a call i any situation do i have to put T in all dial() |
21:31.47 | syle | originally i wanted to go with fedora core 2 when i first installed this but it didn;t support the DELL 2850 |
21:31.57 | syle | so i went redhat 9.0 |
21:32.19 | syle | i bet fedora core 3 would work now but that means a whole reinstall remotely blah |
21:32.26 | brainchil | redhat 9 is EOL almost a year ago |
21:32.42 | brainchil | no security updates unless you do it yourself |
21:33.31 | syle | well security updates don;t bug me i compile everything by hand anyways, but updated package releases is a must |
21:33.49 | syle | like cvs etc, standard crap |
21:34.00 | syle | so yeah i got to throw out rh9 eventually i guess |
21:34.22 | brainchil | Use centos if you want a redhat like OS |
21:34.27 | *** join/#asterisk _mountie (~mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com) |
21:34.39 | syle | i read up on centos and whitelinux or whatever |
21:34.41 | brainchil | it's a binary compile of the redhat enterprise distro |
21:34.47 | syle | whitelinux doesn;t even want to do it anymore |
21:34.51 | brainchil | centos is also what asterisk@home uses |
21:34.55 | syle | fedora there is a core team of developers |
21:35.23 | brainchil | yes but it's as unstable as charles manson by design |
21:35.23 | syle | i heard asterisk is now using windows as their devel environment |
21:35.32 | brainchil | it's meant to be bleading edge |
21:35.36 | *** join/#asterisk zotz (~zotz@24.231.32.109) |
21:36.23 | *** join/#asterisk topping (~topping@cpe-24-210-82-196.columbus.res.rr.com) |
21:36.36 | syle | i don;t know about unstable, i ran fedora core 2 on a couple other servers and it worked out fine |
21:36.50 | *** part/#asterisk dave_mwi_ (~dave_mwi@adsl-11-102-74.mia.bellsouth.net) |
21:37.15 | syle | i know its redhat's development branch for their enterprise releases but still pretty stable |
21:37.20 | brainchil | I honestly gave up on redhat between 8-9 when they imported new posix threading into 2.4 initially and broke a bunch of stuff |
21:37.36 | brainchil | I don't even think Fedora is stable enough for desktop use |
21:38.13 | bannerman | brainchil: my asterisk box ran for several weeks, at the time I had gnome and was using it as my workstation as well |
21:38.14 | syle | idk i don;t use fedora for desktop use, i use it as a server and xp with securecrt as a client |
21:38.15 | Bentley | hello all. I've got a system running fxos on 2 tdm400ps. On occasion, an outbound call and an inbound call will get answered by the same Zap channel (ie: dial out with sip client and immediately find an unrelated caller on the horn with you). Anyone else experience this? |
21:38.21 | brainchil | all of my servers are running slackware or debian now becasue I was tired of redhat breaking things |
21:38.27 | *** join/#asterisk _mountie (~mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com) |
21:38.30 | bannerman | brainchil: using FC3 was an improvement from WinXP for sure |
21:38.52 | bannerman | Bentley: I haven't, but it sounds hilarius :) |
21:38.59 | brainchil | using anythins is an improvement from XP |
21:39.24 | syle | they did break alot of things in redhat9 but from what i have noticed everybody is going that way, so if you don;t patch your programs they won;t work on anything new regardless |
21:39.28 | Bentley | bannerman: heh - not really |
21:39.33 | brainchil | 8 new critical security updates this month alone :-) |
21:39.49 | *** join/#asterisk PTG12345 (PTG123@66.213.239.122) |
21:39.54 | harryvv | brainchil, it is. i use fc3x86_64 with kde |
21:39.57 | PTG12345 | anyone know the command to include another conf file in a conf file? :) |
21:39.59 | bannerman | Bentley: yeah, sorry, I do empathize with you. Still.. |
21:40.13 | file[mac] | #include "blah.conf" |
21:40.14 | brainchil | my desktops are all ubuntu and I PREFER slackware on the servers |
21:40.29 | blitzrage | file[mac]: without the quotes right? |
21:40.30 | syle | harryvv how is that working out for you? |
21:40.32 | brainchil | because it's so simplistic |
21:41.22 | harryvv | syle, I have some gripes about x86_64 since its more orianted to the server market there is alot of apps that are not compiled for it. cannot see flash plugins for firefox is a example. |
21:41.24 | syle | i use to use slackware, then so many companies i worked for over last 10 years always used redhat, so i guess i got use to it |
21:41.30 | bannerman | brainchil: I can't stand slackware because after all these years their installer is still horrid and frustrating. You mess up on one option, have to go through 5 minutes worth of crap, go go back, redo it, hope you don't mess up again... |
21:41.41 | CoaxD | what is the place i should be going to - to do my taxes online? |
21:41.48 | brainchil | my only 64 gripe is that it's not really that much faster at much of anything :-) |
21:41.53 | harryvv | so, I should not have install x86_64 if it would have been a graphics work station. |
21:42.01 | harryvv | brain you are probebly right ;) |
21:42.20 | Sedorox | hmmmmm |
21:42.21 | syle | personally i run redhat for work crap, but i still have my freebsd at home :) |
21:42.29 | brainchil | bannerman: actually all of my servers are identical I do one slackware install and mirror |
21:42.31 | blitzrage | syle: I like CentOS |
21:42.57 | file[mac] | blitzrage: you're soooo cute my dear! |
21:43.00 | harryvv | brainchil, is there a wiki on mirroring and how fast does it take to mirror to another system? |
21:43.14 | brainchil | heheh ... freebsd is fun if you have a good ups and generator if the power outage exceeds that .. otherwise 5+_ hour fsck ... no thank you |
21:43.14 | bannerman | brainchil: there's just no reason they shouldn't take the time to make an installer that isn't designed to be frustrating. |
21:43.24 | harryvv | I keep hearing things about CentOS what makes it that much greater over Fedora? |
21:43.28 | harryvv | or redhat? |
21:43.54 | syle | brainshild can;t say i ever had that experience, i;ve always ran latest 4.x stable branch with no problems |
21:44.15 | brainchil | bannerman: the point is that they have an installer that is very simple ... not frustrating ... but it was meant to be unixlike and simple |
21:44.27 | brainchil | simple doesn't mean easy ... just simple |
21:44.37 | harryvv | or clean |
21:44.50 | bannerman | brainchil: you can do simple without making it hard. |
21:45.06 | syle | i fell in love with freebsd a long time ago when i did make install mrtg and it installed all dependancies with it for me lol |
21:45.06 | brainchil | syle: got a disk bigger than 40Gs? start a heavy write-read operation and hit the power button or unplug it |
21:45.28 | brainchil | it's meant to actually teach you linux/unix |
21:45.29 | *** join/#asterisk mark_wales (~Mark@cpc3-swan1-4-0-cust224.swan.cable.ntl.com) |
21:45.44 | brainchil | redhat doesn't teach you linux it teaches you redhat |
21:45.48 | *** join/#asterisk ManxPower (~eric@stirprop-S0-0-0-26.ndcr2.datasync.net) |
21:45.52 | harryvv | yup |
21:45.58 | syle | naw freebsd is meant to be for more experienced unix users |
21:46.04 | brainchil | because like many other distros they have their own F%$ked up way of doing a lot of things |
21:46.06 | syle | after your done with linux |
21:46.11 | syle | move on to freebsd |
21:46.14 | syle | then solaris etc |
21:46.14 | harryvv | I have done freebsd |
21:46.24 | brainchil | no freebsd is meant for people that are into S and M |
21:46.39 | brainchil | I like freebsd ... the filesystem is just a piece of shit |
21:46.43 | fugitivo | "after your done with linux" ? |
21:46.48 | syle | well |
21:46.50 | syle | depends |
21:46.51 | fugitivo | when are you done with linux? |
21:46.54 | brainchil | just like openbsd |
21:46.57 | syle | depends on what you are doing |
21:47.07 | syle | i use freebsd on single cpu machines |
21:47.08 | brainchil | and don't even start talking about soft updates and background fsck |
21:47.09 | elric | background fsck |
21:47.10 | elric | :( |
21:47.12 | *** join/#asterisk MatsK (~matsk@107.80-202-57.nextgentel.com) |
21:47.18 | elric | only peev with freebsd |
21:47.21 | syle | i use linux on dual cpu machines as SMP sucks on bsd |
21:47.22 | brainchil | that's a half ass solution to the problem |
21:47.53 | brainchil | ever use a machine with a newer reiser/jfs/xfs? |
21:47.55 | elric | I like Solaris 10, performs alright on x86, |
21:48.00 | mark_wales | i wonder if anyone could help with a config/hw prov I have.... I am unable to make outbound calls from a SIP client but I am getting 'No Channel type registered for 'Zap' and 'unable to open /dev/dsp: No such device - i have used modprobe zaptel, modprobe wcfxo and ztcfg before launching asterisk. I am using an X100P card connected to a standard PSTN line. Any help would be greatfully appreciated |
21:48.01 | Nugget | here comes the religion. |
21:48.07 | fugitivo | elric: isn't it slow? |
21:48.12 | syle | naw, with ext3 out i stopped caring about installing reiser |
21:48.12 | brainchil | hard down to up and recovered with a 400 gig array in 10 minutes |
21:48.26 | P-Chan | So, SIP trunking for ulaw support for fax transmission? No FAQ on sip trunking on voip-info.org - is it even possible with Asterisk? (Or am I only asking because I don't fully understand asterisk and need to read between the lines?) |
21:48.39 | brainchil | ext3 is just ext2+journal ... it's really not very advanced |
21:48.45 | elric | fugitivo, a bit. But generally better than all previous releases of Solaris on x86 |
21:48.56 | harryvv | mark, have asterisk running on it? |
21:48.56 | brainchil | it's just what redhat stuck with because it was an easy migration bath for ext2 |
21:49.03 | file[laptop] | P-Chan: SIP, trunking, no |
21:49.08 | fugitivo | elric: did you try asterisk on that? :) |
21:49.13 | syle | are yopu running latest 2.6.x kernels on your slackware boxes? |
21:49.18 | *** join/#asterisk riksta (~rick@81-178-248-131.dsl.pipex.com) |
21:49.34 | harryvv | mark_wales, is asterisk running on that solaris system |
21:49.41 | elric | fugitivo, not yet, but I will soon. Right now I am smitten with FreeBSD |
21:49.46 | P-Chan | file[laptop] - bah.... so for fax signals to "sanely" traverse a Asterisk to Asterisk trunk, what do I use? |
21:49.58 | syle | solaris sucks on x86 you really want sparcs |
21:49.58 | file[laptop] | a regular ULAW channel? |
21:50.13 | brainchil | solaris is faster on x86 |
21:50.15 | elric | syle, have you tried the new Solaris 10? |
21:50.18 | mark_wales | no SUSE9.1 |
21:50.24 | mark_wales | 2.6 kernel |
21:50.33 | brainchil | yes some 2.6.x kernels |
21:50.38 | P-Chan | file[laptop] - ? You mean don't use trunking, just a standard sip forwarding (for lack of a better term)? |
21:50.39 | elric | plus OpenSolaris should be interesting |
21:50.45 | syle | naw i gave up on solaris after i stopped working for this company, who knows when they will start charging for it again |
21:50.57 | brainchil | but the most open solaris will still be encumbered |
21:51.08 | brainchil | though they at least do journaling :-) |
21:51.23 | brainchil | sorry ... filesystem logging |
21:51.26 | brainchil | :-) |
21:51.45 | file[laptop] | P-Chan: SIP trunking doesn't exist in asterisk |
21:51.48 | syle | if a company can afford sparcs and solaris and veritas, oracle, fiber channel arrays then sure go for it, otherwise a nice load balancer and multiple linux boxes is a much cheaper solution |
21:52.02 | brainchil | amen |
21:52.06 | elric | my old work had a 14 processor Ultra Sparc3 machine |
21:52.11 | elric | it was a beast |
21:52.18 | file[laptop] | P-Chan: what I'm saying is just have it as a regular ULAW phone call.. |
21:52.18 | P-Chan | file[laptop] - Ok, I gathered that much from your first statement, but you say a "ULAW Channel" - can you clarify that a bit? |
21:52.35 | brc_ | MARVIN[laptop]! |
21:53.04 | P-Chan | fileplaptop] - Ok, so add a sip client to sip.conf on the originating server and authenticate the receiving server that way and use "ulaw" as the compression? |
21:53.24 | *** join/#asterisk MatsK (~matsk@107.80-202-57.nextgentel.com) |
21:53.25 | file[laptop] | P-Chan: sure... there's nothing special about the fax, it's just another phone call passed through uncompressed |
21:53.43 | syle | holy crap elric |
21:53.52 | syle | idon;t even want to ask how much that thing cost lol |
21:54.00 | *** join/#asterisk R3DB0x (nobody@66.142.28.36) |
21:54.12 | P-Chan | file[laptop] Ok, thanks. This helps get me on the right path. ;) I'm still kinda wet behind the ears - learning from an existing system as I need to. /sigh |
21:54.16 | Egonis | I just did a successful install and a sip phone is now connected... the demo worked, but dialing '#' or '2' do not cause any response.. |
21:54.45 | juiceib269 | sssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssss, .vjdfii`sssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssxljdsj |
21:54.54 | blitzrage | Egonis: sounds like a dtmf problem. Try dtmfmode=rfc2833 in sip.conf |
21:55.18 | elric | lol syle, a shitload i tell you |
21:55.34 | Egonis | blitzrage: will try that, ty! |
21:55.51 | elric | it had more ram than my hdd had gbs |
21:58.14 | Egonis | blitzrage: doesn't seem to have helped, I tried dtmfmode=info as well |
21:58.35 | blitzrage | Egonis: and you did a reload chan_sip.so ? |
21:58.40 | blitzrage | after making the changes |
21:58.50 | Egonis | blitzrage: I did /etc/init.d/asterisk restart |
21:59.11 | jf_ | when transfering a call, it seem to only take 1 digit, how can i set the length |
21:59.14 | blitzrage | Egonis: paste the dialplan and your phone definition in sip.conf |
21:59.24 | blitzrage | Egonis: to pastebin.com |
21:59.33 | tzanger | ~handbook |
21:59.34 | jbot | hmm... handbook is http://www.digium.com/handbook-draft.pdf |
21:59.34 | bugbot | handbook is assigned nothing and reported nothing. |
22:01.00 | file[laptop] | blitzrage: what's up? |
22:02.26 | blitzrage | file[laptop]: not too much... trying to install gstreamer |
22:02.44 | blitzrage | file[laptop]: keep getting a PKG_CONFIG_PATH error :) |
22:03.04 | file[laptop] | bad blitzrage bad |
22:03.14 | blitzrage | the handbook made sense after I understood Asterisk |
22:03.25 | Egonis | blitzrage: http://www.pastebin.com/271379 |
22:04.31 | pgpkeys | i haven't even started a setup yet. i just printed the manual, installed asterisk via ports, looked through a couple *.conf files to see what everything looked like, read the first couple sections of the book,and waiting til i feel slightly better before diving in |
22:04.33 | Egonis | blitzrage: dialplan? |
22:04.55 | pgpkeys | damn cold is kickin my ass. not much energy for my usual 24h+ runs i usually do to learn something |
22:07.42 | *** join/#asterisk nvrs (RUR@toronto-HSE-ppp4255113.sympatico.ca) |
22:09.30 | blitzrage | dialplan = extensions.conf |
22:11.17 | syle | anyone use osoft |
22:11.50 | syle | i bought that asterisk ebook and that is the reader for it, won;t get me copy anything to paste |
22:11.55 | tzanger | sounds like toilet paper |
22:12.19 | tzanger | you just HAVE to try this new toilet paper! What's it called? Osoft! |
22:12.31 | syle | osoft thout reader |
22:12.38 | Egonis | blitzrage: http://www.pastebin.com/271385 |
22:12.59 | syle | anyone in here actually do development on asterisk? |
22:13.25 | syle | you do the c coding for it? |
22:13.29 | tzanger | bkw, blitzrage, manxpower, mikej, kpfleming, pcadach, anthm, lots of people |
22:13.39 | tzanger | I have submitted a few patches |
22:13.55 | blitzrage | wow, I'm honored to be included in that list :) I'm no developer, I'm a solutions guy :) |
22:14.03 | blitzrage | I "get shit done". |
22:14.14 | tzanger | yeah me too but that sometimes involves coding |
22:14.25 | Egonis | blitzrage: fyi: I haven't touched extensions.conf |
22:14.37 | syle | how do you feel about the windows environment they are working on now |
22:14.59 | blitzrage | tzanger: true... I don't know C unfortunately. One of these days when I have some money I'm going to take a fast paced C course to get myself some fundamentals |
22:15.10 | tzanger | blitzrage: don't do that |
22:15.15 | blitzrage | I know of phone systems running on NT4 and they suck |
22:15.19 | tzanger | I can teach you the fundamentals of C easily |
22:15.35 | tzanger | I even have a fun doubly linked list course |
22:15.40 | syle | you don;t learn c to learn fundamentals you learn it to code it everyday hehe |
22:15.41 | blitzrage | tzanger: really? I need a crash course for a few hours |
22:15.50 | Egonis | My server has no sound card... what do I need to config for alsa? |
22:16.03 | tzanger | Egonis: ... a sound card! |
22:16.03 | blitzrage | I know how to script, just don't know C syntax and pointers :) |
22:16.12 | Egonis | tzanger: Do I *NEED* one for asterisk? |
22:16.14 | *** join/#asterisk riksta (~rick@81-178-248-131.dsl.pipex.com) |
22:16.20 | syle | socket programming is the most fun |
22:16.33 | blitzrage | sound cards are NOT needed except for CONSOLE channel |
22:16.52 | Egonis | blitzrage: So do I leave alsa alone? because in the 'messages' log, it says /dev/dsp not found |
22:16.59 | blitzrage | if no sound card, don't load the modules. In modules.conf set: noload => chan_alsa.so |
22:17.02 | syle | but i don;t consider myself a leet programmer by any means since i have never coded a device driver in c or asm lol |
22:17.18 | tzanger | Egonis: no, you need one for alsa |
22:17.27 | tzanger | you asked how to config for alsa |
22:17.30 | tzanger | and you need a sound card for that |
22:17.37 | blitzrage | syle: I just want to be able to read and understand C a bit better and be able to implement ideas |
22:17.39 | Egonis | blitzrage: that is already set |
22:18.12 | blitzrage | tzanger: I'm serious, I want to learn C. Let me know if you are serious about teaching some fundamentals (you know my IM :)) |
22:18.26 | tzanger | I am serious |
22:18.29 | syle | lots of free tutorials online |
22:18.40 | blitzrage | syle: thats not going to work. I don't have the time to learn it on my own |
22:18.43 | syle | for basic variable, arrays , pointers stuff |
22:18.51 | *** join/#asterisk Rez (lorez@lorez.staff.freenode) |
22:19.14 | blitzrage | syle: in terms of all the reading etc. A crash course would do me a world of good, then I can go and refine it all on my own time. |
22:19.29 | blitzrage | I hate being limited in Asterisk simply because I don't know C |
22:19.35 | blitzrage | its the only reason :) |
22:20.35 | *** join/#asterisk Tuplink (~dsfsk@68-232-92-239.chvlva.adelphia.net) |
22:21.16 | Tuplink | im having a problum... |
22:21.31 | syle | if it makes you feel any better i don;t have time for c either, i just code perl applications to run onto of base c things usually |
22:21.32 | marlowe | Tuplink - Ohreallllly? |
22:21.53 | Tuplink | i have FWD set to forward all calls to my PBX.... asterisk recives them but not user input afrer that |
22:22.06 | Tuplink | like i hear it... |
22:22.16 | Tuplink | but nothing works the other way. |
22:22.32 | Tuplink | could it have somthing to do with NAT? |
22:22.32 | blitzrage | syle: I have weak programming unfortunatly... and if I'm going to spend the time learning one particular language well, by damn its going to be C :) |
22:22.48 | *** join/#asterisk jf_ (~jeanfranc@modemcable077.187-80-70.mc.videotron.ca) |
22:23.00 | syle | if you learn c you can learn any language easily , just sytax differences |
22:23.00 | jf_ | can someone help me out with call transert |
22:23.21 | harryvv | blitzrage, same here. I would start to learn it but dont see the end of the tunnel for something I would start to work on. |
22:23.22 | syle | i prefer perl usually since i had to in 2 lines of perl what it would take me 10 lines or more in c to do |
22:23.31 | Hydroxide | syle: I think scheme is a bit more different than that, not that it's relevant for asterisk purposes |
22:23.46 | drumkilla | there is no end of the tunner :) |
22:23.50 | drumkilla | tunnel* |
22:23.57 | harryvv | typo |
22:24.09 | syle | asterisk has to stay c though for speed , i like fact its all c |
22:24.12 | tzanger | I *love* the way that bbq'd hot dogs grow to like 3x their original size |
22:24.23 | drumkilla | I love C. :) |
22:24.28 | *** join/#asterisk CoolCat_ (~god@200.162.252.66.user.ajato.com.br) |
22:24.34 | harryvv | tzanger, try putting one in a microwave for 40 min :) |
22:24.45 | CoolCat_ | hi again! =o) |
22:24.57 | tzanger | harryvv: forty minutes?? JESUS man I put them in for like 1 |
22:25.02 | harryvv | hehehe |
22:25.17 | harryvv | it was a mistake once. It looks like what a dog would leave behind |
22:25.21 | CoolCat_ | ...more doubts emerge from my studies! =o/ |
22:25.25 | Tuplink | what ports need to be routed for IAX to work corectly? |
22:25.28 | drumkilla | 4569 |
22:25.49 | CoolCat_ | does anyone here knows the hw zoom v3? |
22:25.49 | Tuplink | forwarded threw a router? |
22:25.53 | Tuplink | or just able to pass |
22:26.03 | Hogie | why doesn't a tdm03B detect red alarms on the line like my x101p does? When I unplug the co line from the ports, it still tries to dial out of them.... |
22:26.42 | blitzrage | drumkilla: damn those un-ending tunnels |
22:26.49 | syle | well i am stil unsure of that ztdummy problem, the unresolved symbols usually mean its looking for dynamically linked libraries, any solutions from you guru c programmers much appreciated |
22:26.50 | drumkilla | it's so much fun, though |
22:26.53 | jf_ | can someone help me out with call transert |
22:27.13 | drumkilla | transert isn't even a word :) |
22:27.25 | jf_ | what is the word then :) |
22:27.52 | blitzrage | transfer? |
22:28.00 | blitzrage | nahh.... too easy |
22:28.01 | file[laptop] | Transfer in progress. |
22:28.06 | file[laptop] | Please hold two centuries while it is completed. |
22:28.09 | syle | might be an idea to compile these zaptel modules statically not sure |
22:28.15 | file[laptop] | Thank you for choosing dial-up internet service. |
22:28.25 | drumkilla | i'm about to work on a pretty silly bug |
22:28.31 | file[laptop] | silly you say? |
22:28.33 | drumkilla | yep |
22:28.37 | drumkilla | look at 4020 |
22:28.39 | file[laptop] | as silly as... YOU?!? |
22:28.43 | CoolCat_ | http://www.zoom.com/products/voip_products.html ... how good asterisk would be with it? can i gatewat global village SIP <-> skype using asterisk? |
22:28.43 | jf_ | ya if i want to send the call to another extensoion like call parking |
22:28.43 | drumkilla | not quite :) |
22:28.50 | file[laptop] | otay |
22:28.57 | file[laptop] | oh that one |
22:29.10 | blitzrage | drumkilla: 4020 you say? |
22:29.12 | file[laptop] | yeah, it's uh interesting |
22:29.17 | drumkilla | blitzrage: I know, haha ... |
22:29.21 | blitzrage | drumkilla: lol |
22:30.52 | pigpen | Am I correct that I can force the outgoing caller id info by: |
22:30.53 | pigpen | exten => _7NXXNXXXXXX,1,SetCallerID(5555555) |
22:30.53 | pigpen | exten => _7NXXNXXXXXX,2,SetCIDName() |
22:31.02 | pigpen | in my outgoing plan... |
22:31.31 | pigpen | Everytime I change it...it seems my line provider overrides it... |
22:31.37 | Egonis | blitzrage: so what should I do about my dtmf? |
22:32.39 | *** join/#asterisk eKo1 (~bernd@metrored-gw.tropicohn.com) |
22:33.02 | smiley- | pigpen: sounds logical.. so you can't spoof your number |
22:33.55 | easimon | on analog lines you can't set it at all |
22:34.11 | pigpen | I really want the ability to not send caller id info at all... |
22:34.16 | pigpen | we have a pri. |
22:34.28 | smiley- | ah |
22:34.32 | Tuplink | <--- needs help.... i can make a call from asterisk to a FWD user and all goes fine.... i can recive a call from FWD can hear me but i cant hear them whats the prob? |
22:34.41 | smiley- | in .se you can dial #31# as prefix to hide the number... |
22:35.00 | pigpen | can I have * prepend this automatically? |
22:35.03 | easimon | pigpen: did you try to clear it? |
22:35.13 | pigpen | try to clear it? |
22:35.28 | easimon | pigpen: instead of giving a bogus number. |
22:35.31 | *** join/#asterisk Rez (lorez@lorez.staff.freenode) |
22:35.42 | pigpen | like make it blank....yes. |
22:35.51 | pigpen | still sends the real number. |
22:36.00 | Tuplink | <--- needs help.... i can make a call from asterisk to a FWD user and all goes fine.... i can recive a call from FWD can hear me but i cant hear them whats the prob? |
22:36.24 | pigpen | hmm...I will try one of my other numbers... |
22:36.30 | pigpen | I have 100...why not. |
22:36.51 | easimon | pigpen: setting any other *valid* number should work at least... |
22:37.09 | pigpen | yeah..that is what I was thinking... |
22:37.25 | easimon | pigpen: not sure about how to send no id... |
22:38.05 | pigpen | yeah..that worked.. |
22:38.09 | pigpen | I will call the telco then... |
22:38.15 | pigpen | thanks... |
22:38.37 | PatrickDK | pigpen, have you tried *70,number |
22:38.53 | PatrickDK | or does that not work on pri |
22:39.50 | Tuplink | any one here have a FWD station i can try to test my FWD stuff? |
22:40.36 | PatrickDK | tuplink, no, fwd has their own test numbers for you to use |
22:40.49 | Tuplink | yes... what would that be? |
22:41.07 | Tuplink | its jsut that i like a real person for trouble shooting |
22:41.12 | PatrickDK | I don't know, ask fwd |
22:41.13 | CoolCat_ | someone saw my zoomtel v3 ask? |
22:41.55 | jf_ | can someone tell me why i am able to park any call from my cell phone and im not from sip |
22:42.12 | PatrickDK | jf_, cause you supplied T or t |
22:42.19 | harryvv | Tuplink, is FWD sip or iax? |
22:42.26 | Tuplink | IAX |
22:42.28 | PatrickDK | harryvv, both |
22:42.34 | FengShui | anyone here conversant with the channel cloning in ast_do_masquerade? |
22:42.35 | harryvv | which one is he using |
22:42.50 | jf_ | Patrick: i set both |
22:42.58 | PatrickDK | jf, that is the problem |
22:43.13 | PatrickDK | one is for allowing the person making the call to, the other is for the person recieving the call to |
22:43.29 | harryvv | Tuplink, wakup |
22:43.51 | jf_ | i know |
22:44.12 | PatrickDK | well, if you specify both, then everyone can do anything to the call |
22:44.23 | jf_ | but for the one making the call (sip) i cant, i can from receiving the call |
22:44.33 | jf_ | i know im just testing |
22:45.32 | jf_ | can i allow both |
22:45.41 | PatrickDK | jf, sounds like you don't have dtmf signalling set right |
22:45.51 | PatrickDK | inband or outofband, or sip info |
22:45.52 | jf_ | on sip side |
22:46.00 | PatrickDK | on sip phone and asterisk |
22:46.04 | PatrickDK | they have to match |
22:46.06 | jf_ | ok |
22:47.04 | jf_ | maybe it's why |
22:47.11 | PatrickDK | well, options are rfc2833, inband, and info |
22:47.25 | PatrickDK | I use rfc2833 always |
22:47.30 | jf_ | ok |
22:47.30 | PatrickDK | grandstream likes info |
22:47.43 | jf_ | what do i use in xlite |
22:47.55 | PatrickDK | rfc2833, and tell xlite outofband |
22:48.03 | PatrickDK | atleast that is how I had mine setup |
22:48.21 | shmaltz | finally my goverment is doing something usefull with the money I sent in yesterday to the Treasury department |
22:48.23 | shmaltz | http://news.yahoo.com/news?tmpl=story&e=3&u=/ap/20050414/ap_on_go_ca_st_pe/fugitive_roundup&sid=84439559 |
22:49.25 | *** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc) |
22:49.35 | *** join/#asterisk ManxPower (~eric@stirprop-S0-0-0-26.ndcr2.datasync.net) |
22:49.36 | harryvv | shmaltz, and how many of those are illegal mexians entering our borders? |
22:49.42 | harryvv | mexicans ;) |
22:50.00 | shmaltz | who knows, the article doesn't mention that |
22:50.20 | shmaltz | since the DHS didn't take part in this, they might not know for another week or so |
22:50.40 | harryvv | I served in albuquerque nm while in the service and a collage girl was being followed by somone, she did a uturn and thay shot and killed her..the fugitives drove back into mexico. |
22:50.54 | jf_ | Patrick: can u transfert |
22:51.01 | PatrickDK | ya |
22:51.17 | PatrickDK | worst case, use inband, and use ulaw |
22:51.19 | jf_ | what u suggest for dmtf |
22:51.31 | harryvv | Terrorist can easly slip into the states from mexico or canada. So many open holes. |
22:51.33 | shmaltz | there was another story last weekend about some mexican that just crossed the border after killing an official in CA |
22:51.42 | PatrickDK | any other codec, you need rfc2833, and outofband |
22:51.50 | shmaltz | canada is actualy better closed off then ppl think |
22:51.52 | harryvv | California? |
22:51.52 | jf_ | k |
22:52.07 | shmaltz | the only known hole is thru the inidian reservations |
22:52.09 | shmaltz | yep |
22:52.13 | shmaltz | california |
22:52.40 | harryvv | shmaltz, there are actually 50 open holes on the us/canada border people can drive into the states unimpeeded. |
22:52.59 | shmaltz | oh really, with out any border patrol? |
22:53.03 | harryvv | yup |
22:53.11 | shmaltz | look at this one: |
22:53.13 | shmaltz | http://story.news.yahoo.com/news?tmpl=story&ncid=1212&e=1&u=/ap/20050414/ap_on_hi_te/comcast_internet_problems&sid=95573501 |
22:53.15 | newl | or more if you've got a boat. :) |
22:53.18 | shmaltz | looks like a DOS on their DNS |
22:53.36 | harryvv | yea |
22:53.38 | harryvv | mabey |
22:53.42 | shmaltz | whats the official answer the that the DOT has for this? |
22:53.49 | jf_ | patrick: im inband both side, does not work |
22:54.00 | PatrickDK | you using ulaw right? |
22:54.00 | harryvv | I dont have that much confidence in voip if things like that happen. |
22:54.19 | harryvv | on a local network thats fine. |
22:54.26 | jf_ | ya |
22:54.29 | shmaltz | harryvv, you are right about that, but if you really think about it this is what it comes down to |
22:54.49 | shmaltz | DOS on PSTN is easier than you think |
22:54.56 | harryvv | the only way around it is route the calls though a router with two wan links |
22:55.05 | harryvv | with two seperate providers |
22:55.16 | harryvv | two seperate backbones |
22:55.28 | shmaltz | you could just take 2 VoIPs and make 50 simultaneous phone calls to a PSTN and that person is DOSed |
22:55.44 | harryvv | true |
22:55.58 | jf_ | Patrick: should i use something else |
22:56.00 | jf_ | alaw |
22:56.07 | *** join/#asterisk Rez (lorez@lorez.staff.freenode) |
22:56.08 | PatrickDK | dunno, you sure it's using ulaw? |
22:56.13 | jf_ | ya |
22:56.15 | shmaltz | using what you are saying we actualy have better protection (if you use BGP you are even better protected) for VoIP than for PSTN |
22:56.16 | PatrickDK | it highlights ulaw when you make a call? |
22:56.52 | harryvv | I started to learn Border gateway protocol in cisco. thats a pretty high level protocol to learn. |
22:57.16 | harryvv | I think the major backbones based cisco routers use that right? |
22:57.16 | newl | out of 50 calls, only one is going to go through though. The remaining 49 will receive BUSY, get diverted to voicemail, or perhaps a second call will ring on CW until it isn't answered and the possibly diverted to voicemail on no anwer. :) |
22:57.43 | jf_ | can someone tell me what is that ARNING[12481]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 86B65EE2-AD38-11D9-9881-000D93C516B2@ |
22:57.45 | jf_ | .... |
22:57.56 | shmaltz | so it comes down to when you have a DOS (most providers nowadays block spoofed IPs, so DDOS is realy not so common anymore) attack you might not have any connectivity, but the difference in price will make the decision for ppl |
22:58.16 | shmaltz | of course they *all* use BGP |
22:58.19 | harryvv | yea |
22:58.21 | shmaltz | read this: |
22:58.25 | shmaltz | ~dos |
22:58.27 | jbot | from memory, dos is (Disk Operating System) This OS is what got it all started for PCs. Denial of Service... |
22:58.28 | bugbot | dos is assigned nothing and reported nothing. |
22:58.32 | shmaltz | ~ddos |
22:58.33 | jbot | i guess ddos is Distributed Denial of Service, or http://grc.com/dos/grcdos.htm, or see slashdot.org |
22:58.33 | bugbot | ddos is assigned nothing and reported nothing. |
22:58.38 | CoolCat_ | what is the easy way of gateway skype and sip? |
22:58.49 | shmaltz | follow that link, and read the one about drdos |
22:59.08 | harryvv | i already know what ddos is :) |
22:59.11 | *** part/#asterisk Rez (lorez@lorez.staff.freenode) |
22:59.18 | shmaltz | the DrDOS attack he had was exploiting the fact that most core routers use BGP |
22:59.38 | shmaltz | but that link has something about DrDOS |
22:59.52 | shmaltz | Distributed Reflected Denial Of Service |
23:00.01 | shmaltz | the reflected part is new |
23:00.19 | shmaltz | I think he was the only one that was a victim of this attack |
23:00.33 | shmaltz | I never heard anybody else that was a victim of this attack |
23:04.53 | ManxPower | *whine* Nobody responded to my message |
23:05.02 | marlowe | eh? |
23:05.58 | harryvv | what message |
23:06.18 | shmaltz | ManxPower, looks like you posted the message to a different channel :/ |
23:06.27 | harryvv | yea |
23:06.35 | *** join/#asterisk bjohnson (~bjohnson@66.11.188.213) |
23:06.36 | harryvv | I looked back on this channel :) |
23:08.17 | *** join/#asterisk Micc (~mic@c-24-18-35-120.hsd1.wa.comcast.net) |
23:08.35 | Micc | quick question. |
23:08.42 | Micc | Does asterisk work with broadvoice? |
23:08.59 | shmaltz | Micc, why not |
23:09.01 | MikeJ[Jayden] | micc, yees |
23:09.11 | shmaltz | stupid article: |
23:09.12 | shmaltz | http://story.news.yahoo.com/news?tmpl=story&ncid=1212&e=10&u=/ap/20050414/ap_on_hi_te/voip_regulations&sid=95573501 |
23:09.21 | MikeJ[Jayden] | there are some known qwirks with multiple broavoice accounts |
23:09.41 | Micc | Ok, so I've built and installed asterisk on my linux box. Is there some docs on how to set up asterisk with broadvoice or other sip? |
23:10.32 | shmaltz | nice move from bush: |
23:10.33 | shmaltz | http://story.news.yahoo.com/news?tmpl=story&ncid=1211&e=4&u=/nm/20050414/tc_nm/tech_government_amd_dc&sid=95573372 |
23:10.54 | shmaltz | this means asterisk can be installed in the white house ;) |
23:11.46 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
23:11.47 | ManxPower | shmaltz, no my mailing list message |
23:11.57 | shmaltz | oh, |
23:12.02 | shmaltz | whats the question about? |
23:12.14 | shmaltz | subject or otherwise |
23:12.17 | *** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net) |
23:12.32 | PBXtech | how do you get over putting in the 'username' in the DIAL string for IAX2? (AMP not likey) |
23:12.39 | harryvv | wow |
23:12.53 | harryvv | thats good new for amd |
23:12.54 | harryvv | ;) |
23:13.07 | harryvv | And relief for smaller companies. |
23:13.23 | shido6 | :) |
23:13.56 | Tuplink | what is Congestion in extentions |
23:14.05 | Tuplink | bussytone |
23:14.12 | Tuplink | ? |
23:14.51 | shido6 | yeah |
23:14.57 | CoolCat_ | is there any free sip provider out there? |
23:15.11 | shido6 | right |
23:15.12 | pgpkeys | well free for the number but you still pay for calls. |
23:15.23 | reallost1 | How do I unset a variable? |
23:15.25 | pgpkeys | nufone charges 0.02 a minute in the US for US48 and CAn calls |
23:15.40 | shido6 | same for inbound 8xx |
23:15.42 | shmaltz | harryvv, yep |
23:15.49 | pgpkeys | right |
23:15.52 | shmaltz | ManxPower, what is the subject line? |
23:15.55 | pgpkeys | 0.02 regardless of in or out |
23:16.02 | pgpkeys | you make it, they make it.. 0.02/m |
23:16.10 | pgpkeys | 0.08 to alaska and hawaii |
23:16.13 | Tuplink | FWD has free 1800 |
23:16.24 | harryvv | Whats the smallest motherboard that can take two pci slots and a embeded linux flash card? |
23:16.48 | pgpkeys | nufone gives a free toll free. 800, 866, etc etc. though their site is now saying that they have no numbers left. must have used up their allocation. |
23:17.20 | reallost1 | ipkall has free incoming numbers. |
23:17.25 | ManxPower | [Asterisk-Users] Call Parking timming out to the wrong extension |
23:18.05 | harryvv | I want to get * off my work station and into something small. |
23:18.26 | *** join/#asterisk fugitivo (~ajf@201.255.103.99) |
23:19.55 | CoolCat_ | this sound free, but im not sure, some terms talk about charges! =o/ http://www.earthlink.net/membercenter/benefits/onlinecalling/ |
23:20.59 | harryvv | yea..its a highly compeditive biz |
23:22.50 | shmaltz | ManxPower, what app you using for parking? |
23:23.19 | ManxPower | shmaltz, you don't have to use an app. |
23:23.33 | shmaltz | I know |
23:23.37 | shmaltz | but thats what I'm asking |
23:23.40 | ManxPower | BTW, any telco geek out there. Amphenol connector's gender is confusing. Can anyone help? |
23:23.43 | shmaltz | you just using xfer? |
23:23.46 | CoolCat_ | fwd sounds free! |
23:23.46 | ManxPower | shmaltz, features.conf |
23:23.50 | shmaltz | ok |
23:24.05 | ManxPower | yes, transfer to 3515 which is defined in features.conf. |
23:24.15 | shmaltz | Amphenol that has the 3 mm strip sticking out and 25 lines of copper on each side is male |
23:24.18 | MikeJ[Jayden] | parking:http://www.pbxclue.com/asterisk_apps/ |
23:24.20 | ManxPower | See http://bugs.digium.com/bug_view_page.php?bug_id=0004036 |
23:24.34 | ManxPower | shmaltz, You are SURE? |
23:24.50 | MikeJ[Jayden] | stivking out is male, going in is female :) |
23:25.10 | *** join/#asterisk darwin35 (~darwin35@24.3.226.147) |
23:25.29 | shmaltz | yep |
23:25.38 | shmaltz | usualy the 66 block comes with the male one |
23:25.52 | harryvv | Manx, I have worked with amp connectors what are you working with |
23:25.53 | shmaltz | and the equipment with the female connector |
23:26.27 | harryvv | Ohh telco mmm amp has diagrams and specs on there connectors. |
23:26.50 | ManxPower | shmaltz, Do I need AMP or Avaya style? |
23:26.50 | tzanger | Amphenol male has the connectors in a row in the middle |
23:26.52 | tzanger | Amphenol female has the connectors around the inside with nothing in the middle |
23:26.56 | ManxPower | So I need a female connector to plug into the Adtran Channel Bank |
23:27.14 | harryvv | manx whats the part number |
23:27.25 | shmaltz | I don't know about the Adrans but Adit needs a male |
23:28.12 | ManxPower | The adtran sticks out |
23:28.44 | PBXtech | how do you get over putting in the 'username' in the DIAL string for IAX2? (AMP not likey) |
23:29.08 | harryvv | Manx, generally what is that connector called? |
23:30.10 | ManxPower | harryvv, Amphenol |
23:30.23 | ManxPower | gotta run |
23:30.25 | harryvv | I know is the infinity style of connector |
23:31.09 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
23:33.18 | harryvv | Amphenol T-1 for the Total Access 750 Chassis T1 Channel bank |
23:33.22 | harryvv | My guess |
23:35.08 | *** join/#asterisk jf_ (~jeanfranc@modemcable077.187-80-70.mc.videotron.ca) |
23:35.24 | harryvv | tzanger, was he working with the 50 pin connector |
23:35.55 | jf_ | someone can tell me why each time i reboot my linux kernel 2.6 i have to rerun make install on zaptel and then modprobe zaptel wcfxo otherwise it does not load module |
23:36.00 | *** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) |
23:36.16 | tzanger | you asked for the amp connector on a channel bank |
23:36.58 | shido6 | thats sounds terrible, jf |
23:37.19 | shido6 | make config makes the init scripts you need |
23:43.26 | *** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net) |
23:45.06 | harryvv | Basicly COs mostly use channel banks for home users. |
23:45.15 | harryvv | Am I correct on this |
23:49.18 | *** join/#asterisk logarno (~logarno@80.125.208.234) |
23:49.44 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
23:50.13 | elric | how would you tell a macro to wait until the call has actually been answered by the party being called? |
23:51.55 | jf_ | shido6: what do u suggest |
23:52.01 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
23:53.05 | shmaltz | can I plug in an rj45 into a jack that is made for an rj48? |
23:53.14 | Tuplink | is their some program to make the womans voice say what you want? |
23:53.29 | tzanger | I never knew the difference between RJ45 and RJ48 aside from that 'tit' out the side |
23:53.46 | harryvv | :) |
23:54.49 | PatrickDK | tzanger, you mean DEC style |
23:54.51 | harryvv | tzanger, Manx is setting up a channel bank for a local business? |
23:55.03 | Tuplink | how do i make the asterisk womans voice say what i want? |
23:55.12 | tzanger | harryvv: don't know what he's doing |
23:55.23 | elric | is there a variable that tells you if the number you were calling has answered the phone? |
23:55.45 | tzanger | elric: you mean after the Dial()'s complete? |
23:56.38 | elric | tzanger, i am doing Dial(${EXTEN}|60|M(macro)) |
23:56.47 | harryvv | okay. I can understand a CO using channel banks to convert there high speed digital network into analog and push that to the residential homes in the area but what is the purpous in the astrisk realm? Why would a bussiness want to use analog lines instead of digital for its phones? or would it be used for fax? |
23:56.53 | elric | and the macro executes as soon as the number is dialed |
23:56.56 | tzanger | elric: add a g option and afterward look at HANGUPCAUSE and DIALSTATUS |
23:57.01 | *** join/#asterisk Legend (~Legend@24.244.142.134) |
23:57.09 | elric | i want it to wait till the call is answered |
23:57.18 | tzanger | harryvv: I use channel banks |
23:57.18 | elric | thanks tzanger |
23:57.28 | harryvv | for what application |
23:57.35 | tzanger | high quality hybrids, "high" port density |
23:57.39 | Tuplink | how do people make all of theyr menu sounds? |
23:57.42 | tzanger | a T100P+CB can handle 24 channels with only one zap card |
23:57.51 | tzanger | Tuplink: I use Record() |
23:57.51 | harryvv | but only for anlog phones or say fax machines |
23:57.56 | tzanger | harryvv: depends |
23:57.57 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
23:57.57 | *** mode/#asterisk [+o bkw_] by ChanServ |
23:58.00 | *** part/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
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23:58.20 | harryvv | cannot work with digital phones that would normally use a pbx then. |
23:58.26 | tzanger | harryvv: sure it can |
23:58.28 | Tuplink | will Record make it that womans voice |
23:58.37 | harryvv | okay |
23:58.37 | bkw_ | haha |
23:58.38 | tzanger | Tuplink: no that is alison |
23:58.55 | Tuplink | alison how do i get her to say what i want? |
23:58.58 | MichaelCat | Does anyone want to help me try to fix caller ID inbound to my X100P clone |
23:58.59 | tzanger | harryvv: before I did a PRI connection to the Norstar MICS I did it with analog trunks and FXS CB modules |
23:59.11 | harryvv | okay |
23:59.23 | bkw_ | Tuplink, you pay her.. thevoice.digium.com |
23:59.33 | bkw_ | beware.. she will not say the word "cunt" |
23:59.38 | tzanger | bkw_: hahaha |
23:59.38 | harryvv | so say then the pbx was off site you would have the digital phones plug into the channel bank route the t-1 digital traffic to the remore asterisk system then. |
23:59.46 | harryvv | one application? |
23:59.48 | tzanger | I still think you get her to say "call hunt group" and edit it |