| 00:01.47 | *** join/#asterisk schirpich (n=dvs@ip21.farheap.net) |
| 00:03.29 | schirpich | whats the best way to go about creating a standalone voicemail box in its own context? Do I have to define the context in sip.conf |
| 00:04.11 | schirpich | or just voincemail.conf and extensions.conf |
| 00:04.27 | Qwell[] | schirpich: just voicemail.conf |
| 00:04.38 | Qwell[] | (obviously, extensions.conf will need to know to use it) |
| 00:08.38 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:09.02 | *** join/#asterisk mtaht4 (n=m@reserve-64-79-114-30.wiline.com) |
| 00:11.14 | *** join/#asterisk dongs (n=HPUX@h193012.ppp.asahi-net.or.jp) |
| 00:11.39 | dongs | May 31 09:07:26 WARNING[8715]: chan_sip.c:9559 handle_response_invite: Forbidden - wrong password on authentication for INVITE to '"myfwd#" <sip:myfwd#@fwd.pulver.com>;tag=as79cc62c5' |
| 00:11.45 | dongs | what is this supposed to mean? |
| 00:12.02 | dongs | (obviously the psasword is correct as i can register=> just fine |
| 00:12.28 | schirpich | Qwell: i've done that much. I have a seperate context and the voicemail boxes created in voicemail.conf. I have done the stuff in extensions.conf also, but when i dial the vm and enter my vmbox and password it tells me the login is incorrect, like its looking in the wrong context or somthing. |
| 00:13.08 | schirpich | its like its looking in my default context |
| 00:13.16 | dongs | so? |
| 00:14.19 | *** join/#asterisk [hC] (i=root@donkey.voxter.ca) |
| 00:14.53 | [hC] | Anyone have experience putting together a sangoma a200 with one remora daughterboard? |
| 00:14.56 | [hC] | (or more?) |
| 00:15.51 | [hC] | i WAS using two a200's, but replaced it with a single + daughterboard, and now channel 5 repeatedly answers what it thinks is an incoming call, however there is none. |
| 00:16.06 | [hC] | also one of the fxo modules seems to have decided not to work, and im not sure if theres a trick to doing it this way or what. |
| 00:17.19 | *** join/#asterisk ctjctj (n=ctj@grapeshot.cyberpaladin.com) |
| 00:18.23 | JunK-Y | ~agi api |
| 00:18.24 | jbot | [agi api] at http://home.cogeco.ca/~camstuff/agi.html |
| 00:20.28 | ctjctj | Yesterday all four of my phones attached to adaptors (linksys Phone adapter and a sipra(sp)) both stopped giving dail tone. All four still ring and I've made no changes in any configuration file since march 1st, 2006 which was a change to meetme.conf (last change to sip.conf was dec20, extensions was sep22) Would somebody please give me a general pointer as to where to start researching how to get dial tone back (both adapters have been powerc |
| 00:20.28 | ctjctj | ycled) |
| 00:20.30 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 00:21.52 | dongs | May 31 09:07:26 WARNING[8715]: chan_sip.c:9559 handle_response_invite: Forbidden - wrong password on authentication for INVITE to '"myfwd#" <sip:myfwd#@fwd.pulver.com>;tag=as79cc62c5' < what is this supposed to mean? (obviously password is correct, sip registration succeeds. i get this error when dialing. |
| 00:23.33 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 00:24.28 | mitcheloc | zoa: you around? |
| 00:26.20 | *** join/#asterisk rva (n=rafa@200.210.51.130) |
| 00:27.05 | dongs | HELLO CAN PEOPLE GET HELP IN HERE OR WHAT |
| 00:27.47 | file | dongs: yes, but we aren't obligated to help you... and some people don't know how to help you... or are busy... |
| 00:28.09 | denon | dongs: file is actually the programmer of asterisk, he will help you |
| 00:28.12 | denon | oh, there he is now |
| 00:28.22 | file | ut roh |
| 00:28.27 | dongs | okay great. i am sure you can see the error line above. |
| 00:28.33 | denon | heheh |
| 00:28.42 | denon | file: hey, at least I didnt tell him to privmsg |
| 00:29.01 | ctjctj | denon: Do you have a pointer for a no dialtone situation? |
| 00:29.09 | denon | fix your zap configs |
| 00:29.17 | dongs | okay so? |
| 00:30.15 | file | dongs: read what I said. |
| 00:32.36 | *** join/#asterisk IronHelixz (n=irc@ool-45785cfe.dyn.optonline.net) |
| 00:32.50 | dongs | ookay. |
| 00:33.53 | rva | hi guys...could anyone please help with a TE110P? |
| 00:33.59 | rva | it just gives me red alarm |
| 00:34.03 | Qwell[] | ~ask |
| 00:34.04 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a quesiton first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily. See also http://catb.org/~esr/faqs/smart-questions.html |
| 00:34.19 | Qwell[] | quesiton? wtf added that? |
| 00:34.58 | JASON99 | If no dial plan is matched in my context how would I play a recording to say invalid number? |
| 00:35.00 | dongs | okay fuck this, i'm just going to grab a normal phone, this voip shit is a waste of time |
| 00:35.07 | Qwell[] | dongs: k, bye |
| 00:35.16 | Qwell[] | ~door |
| 00:35.17 | jbot | [door] don't let it hit you on your way out |
| 00:35.31 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
| 00:35.53 | dongs | Qwell[]: yea, you're so clever. ever thought that you can be replaced by some machine that just types ~randomcockshit every few minutes? |
| 00:35.59 | dongs | Qwell[]: because thats about your limit of contribution in here |
| 00:36.21 | Qwell[] | Well, I've done my part. |
| 00:36.21 | file | how rude |
| 00:36.22 | dongs | YOU are exactly the reason opensource is a fucking failure. |
| 00:36.25 | dongs | well, people like you anyway. |
| 00:36.32 | JASON99 | opensource is not a failure |
| 00:36.34 | Qwell[] | aww, thanks |
| 00:36.37 | Qwell[] | such a sweet thing to say |
| 00:36.53 | rva | ok..i'll try: first, i only find documentation on cross over E1/T1 cables. Could anyone point me how to make a cable to connect to a pstn E1 trunk? |
| 00:36.57 | ctjctj | *laughs* Dong's don't you get it, these guys are here on their own time helping people as best they can. You're being rude and demanding of volanteers. Take a chill pill and try again. |
| 00:37.12 | Qwell[] | rva: one sec |
| 00:37.15 | Qwell[] | unless.. |
| 00:37.20 | Qwell[] | ~cables |
| 00:37.49 | Qwell[] | http://www.jaredsmith.net/misc/cables/ |
| 00:37.51 | file | http://www.jaredsmith.net/misc/cables/ |
| 00:37.51 | JASON99 | ~nomatch |
| 00:37.58 | Qwell[] | file: slow |
| 00:38.03 | Qwell[] | jbot: cables is http://www.jaredsmith.net/misc/cables/ |
| 00:38.05 | jbot | okay, Qwell[] |
| 00:38.05 | file | Qwell[]: :( |
| 00:38.06 | ctjctj | Qwell, can you get the magic robot tell me how come my ata's are suddenly giving me no dial tone? |
| 00:38.33 | file | ctjctj: make sure they're registered on your asterisk, as depending on your settings... they won't give dialtone if not registered |
| 00:38.43 | file | make sure DNS is ok as well... |
| 00:39.09 | file | eeeeep |
| 00:39.13 | Qwell[] | :D |
| 00:39.14 | JASON99 | Is there sometime of catch all that you can put at the end of a context in case there are no matches? |
| 00:39.18 | ctjctj | file: sip show peers says they are registered. I.e. 2001/2001 192.x.x.x 255.255.255.255 5060 Unmonitored |
| 00:39.21 | JASON99 | oops.. sometime = sometype |
| 00:39.40 | file | ctjctj: and what do the web interfaces say? |
| 00:39.56 | Qwell[] | JASON99: like the 'i' exten? |
| 00:40.10 | *** mode/#asterisk [+o file] by russellb |
| 00:40.27 | ctjctj | file: Can you ask about a specific setting? I'm looking at the linksys pap and not seeing anything untoward and the IP is correct and DNS and gateway. |
| 00:40.43 | JASON99 | Qwell: Maybe? exten => i,1,HangUp Woudl that do it? |
| 00:40.50 | Qwell[] | JASON99: should |
| 00:40.55 | Qwell[] | try it :) |
| 00:40.57 | file | I don't have the web interface memorized |
| 00:41.03 | Qwell[] | file: WHY NOT?! |
| 00:41.09 | Qwell[] | oh wait, +o.. |
| 00:41.39 | file | I believe... the Line 1 Status section |
| 00:41.51 | file | Hook State, Registration State |
| 00:41.51 | rva | Qwell: the site you told me have an ethernet crossover, an T1 crossover and a straigh. Is the straithg an ethernet one or a T1 one? |
| 00:42.04 | Qwell[] | rva: It's a straight-through cable :) |
| 00:42.16 | Qwell[] | You want one end to be straight, and the other as t1 crossover |
| 00:42.23 | [hC] | oh wow. |
| 00:42.38 | [hC] | this is so much easier. |
| 00:42.39 | [hC] | :P |
| 00:42.41 | ctjctj | file: only the Call 1 tone: none looks strange to me |
| 00:42.57 | vooduhal | Ok, back to the reason I was on for tonight. Anyone have any idea how to disable the "Transfer" softkey on the polycom. I've already reprogrammed the hard Transfer key, but it doesn't follow the same rules as the soft key. |
| 00:42.59 | Qwell[] | ctjctj: Is that editable? |
| 00:43.09 | file | that's normal |
| 00:43.22 | *** join/#asterisk ramo (n=ramo@59.92.165.142) |
| 00:43.30 | ctjctj | Qwell, If need be I'll find the passwords again. Dang random sequence of letters. |
| 00:49.11 | mitcheloc | vooduhal: why do you want to disable it? |
| 00:50.13 | operat0r | so I am doing ipkalll > my asterisk box just for AGI scripts I dont need a sip.conf ? just extentions.conf ? |
| 00:50.28 | vooduhal | Long story, but here goes again. QueueMetrics can only record a transfer if done through * and not as a refer. So pressing # to initiate a transfer and pressing transfer are two entirely different things. I've already remapped the physical Transfer button as the # key to fix the physical button. |
| 00:53.36 | vooduhal | So anyone have any ideas short of crushing the LCD. :) |
| 00:53.58 | file | smashing the LCD sounds nicer |
| 00:54.17 | robl^ | sledgehammer.. NEXT!!! |
| 00:54.30 | file | exactly! |
| 00:55.27 | vooduhal | Lol. |
| 01:01.29 | operat0r | http://asterisk.rmccurdy.com/log1.txt what does this mean |
| 01:02.48 | operat0r | I tried with blank sip.conf |
| 01:03.44 | ctjctj | file: I restarted the gateway machine. I power cycled the linksys. I watched it register with asterisk. Any ideas or pointers? |
| 01:04.00 | file | nope... they should produce dialtone |
| 01:04.39 | operat0r | here is all my conf http://asterisk.rmccurdy.com |
| 01:04.59 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 01:05.19 | Dr-Linux | huh, i have bad connection :( |
| 01:05.27 | operat0r | Dr-Linux what you mean ? |
| 01:05.56 | Dr-Linux | operat0r: my internet connection is very bad. |
| 01:06.13 | operat0r | Dr-Linux define very bad ? bad copper slow ? |
| 01:06.13 | mishehu | hmm... this is weird. I have a dual proc pIII-800 running 1.2.7.1, and whenever I launch it via safe_asterisk, typing "show translation recalc 5" crashes it out. however, if I launch it by asterisk -gvvvvvvvvc, I can run "show translation recalc 5" and it gives a proper calculation. |
| 01:06.29 | mishehu | and it doesn't seem to happen on my dual core opteron... |
| 01:06.36 | mishehu | or my 2 mini itx's |
| 01:06.47 | denon | do a clean rebuild on the p3-800? |
| 01:07.00 | mishehu | denon: it was a clean install. |
| 01:07.03 | mishehu | from source. |
| 01:07.27 | denon | might not hurt to make clean and try again |
| 01:07.31 | ctjctj | file: thank you. It is a two problem situation. 1) a power failure and restart sequence caused the ATA's to go funky. (no dial tone). So when I verified all lines on all ATA's non-had dial tone. 2) Baby pulled phone of my desk and it is DEAD. So it gives no dial tone or keytones and is NOT a good test instrument. |
| 01:07.33 | denon | or even re-acquire src, just to be sure |
| 01:07.51 | file | ctjctj: eep |
| 01:07.58 | *** join/#asterisk FinboySlick (n=FinboySl@c207.134.125-161.clta.globetrotter.net) |
| 01:08.03 | FinboySlick | Yo. |
| 01:08.24 | mishehu | denon: source matches sha1sum on all machines. |
| 01:08.42 | Dr-Linux | mishehu: kinda same here, whenever i my server reboots, everything works fine, but i can't do "asterisk -r" to get into CLI |
| 01:08.53 | ctjctj | file: *just shakes his head* So my debuging test, lift the handset next to me was meaning less. Somewhere in the last 6 hours of power cycles, resets and other misc standard stuff. All the other lines got "fixed" |
| 01:09.00 | mishehu | Dr-Linux: what happens when you try to? |
| 01:09.19 | file | ctjctj: I'm not surprised |
| 01:09.25 | mishehu | denon: only other diffs between the machines is gcc version (this pIII-800 has 3.3.4 while the others have 3.4.4) and kernel version. |
| 01:09.34 | ctjctj | Thank you again for your help. And remember dongs are cheap, every man comes with one, few learn to use them politely. You all are doing a good job. |
| 01:09.52 | Dr-Linux | mishehu: i can't get into CLI, but everything works fine, so i kill the asterisk, then again i do "asterisk -gcvvvv" |
| 01:10.06 | operat0r | cat file > dog ? |
| 01:10.28 | mishehu | Dr-Linux: that's a bit different of a problem I'd think... maybe related somehow, but I don't know. |
| 01:10.37 | mishehu | I don't step in the minefield of source code. |
| 01:10.39 | mishehu | heh |
| 01:10.39 | Dr-Linux | operat0r: i'll be good if you use one more ">" lolz |
| 01:10.41 | ctjctj | cat file | sed -e s/bow/wow/ >dog? |
| 01:11.02 | operat0r | Dr-Linux sry I only have append rights |
| 01:11.02 | ctjctj | be well all. |
| 01:11.10 | *** join/#asterisk cybergyp1y (n=mark@APoitiers-152-1-90-230.w86-201.abo.wanadoo.fr) |
| 01:11.12 | *** part/#asterisk ctjctj (n=ctj@grapeshot.cyberpaladin.com) |
| 01:11.19 | mishehu | I'm already involved in 3 projects already, 2 of which are php and one which is c++... |
| 01:11.22 | *** join/#asterisk mog_home (n=achika54@68.62.237.103) |
| 01:11.44 | mishehu | operat0r: becareful, it might crash on you! |
| 01:12.17 | mishehu | this is strange as hell. why do I have an odd feeling that if I disable codec_speex.so from loading tha tit will not do this... |
| 01:12.18 | *** join/#asterisk cods (n=cods@tuxee.net) |
| 01:12.22 | operat0r | mishehu im such a noob I wamt to smack myself 2 conf files and I can't even get tt-monkeys |
| 01:12.34 | mishehu | though that would be strange again since all 4 machines have speex 1.1.12 |
| 01:12.57 | mishehu | operat0r: tt-weasels is more fun, the monkeys get too messy when they throw around their poop |
| 01:13.38 | FinboySlick | Best would be a monkey/weasel hybrid. |
| 01:14.04 | mishehu | ahah, yeah, I unload speex and it doesn't shit on that machine agian. |
| 01:14.09 | operat0r | so the tt-monkeys are the reason I get SIP/2.0 404 Not Found |
| 01:15.26 | mishehu | operat0r: it's the flying feces man |
| 01:15.51 | robl^ | operat0r: do this... change the context=1111 to context=default |
| 01:16.44 | robl^ | operat0r: next. change exten => homephoneline... to exten=> 1111,1,Answer |
| 01:17.11 | robl^ | and so on. get rid of homephoneline. |
| 01:17.19 | robl^ | then dial 1111 from the phone |
| 01:18.18 | operat0r | homephoneline is the extention ? and the [1111] is what ? or ? |
| 01:18.28 | robl^ | change all occurrances of "homephoneline" to 1111 |
| 01:18.47 | operat0r | robl^ testing |
| 01:19.02 | robl^ | [1111] is a context.. ig nore it for now.. ya don't need it.. first you want to get a BASIC set up.. make it answer and play the file |
| 01:19.30 | operat0r | robl^ ahhhhh ok |
| 01:19.56 | *** join/#asterisk marv (n=marv@12-219-145-181.client.mchsi.com) |
| 01:19.59 | operat0r | so you have context and extentions so a context and have other extentoins correct |
| 01:20.06 | robl^ | umm.. and you still need to define a phone in sip.conf |
| 01:20.06 | operat0r | robl^ wOOt it works |
| 01:20.32 | operat0r | robl^ I dont have a shoft phone setup I dont even want to I just want to write AGI scripts |
| 01:21.01 | robl^ | operat0r: it sorta works. I set ya on the right path.. your sip.conf definately needs work too. ;-) |
| 01:21.16 | operat0r | sounds like poop voip > voip maybe prob ? |
| 01:22.08 | operat0r | ohh im wathcing TV thats why |
| 01:26.03 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 01:27.33 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 01:28.49 | *** join/#asterisk ramo (n=ramo@59.92.165.142) |
| 01:31.40 | *** join/#asterisk jahani (n=k@41.250.49.207) |
| 01:37.56 | *** join/#asterisk demigod2k (n=joey@cpe-24-210-97-162.twmi.res.rr.com) |
| 01:37.58 | demigod2k | hi |
| 01:42.23 | Dr-Linux | Qwell: you around? |
| 01:42.58 | dlynes_home | ~seen qwell |
| 01:43.17 | jbot | qwell is currently on #asterisk (3h 45m 5s), last said: 'mitcheloc: most of the developers use Linux, and that's what's officially supported'. |
| 01:44.11 | Dr-Linux | dlynes_home: actually i wanted to ask him something, bcoz he suggested something |
| 01:44.28 | Dr-Linux | dlynes_home: let me ask you |
| 01:45.03 | Dr-Linux | dlynes_home: as Cisco phone doesn't work with PPPoE internet connection |
| 01:45.47 | Dr-Linux | bcoz cisco phone do have dialer |
| 01:46.33 | mitcheloc | cisco phone + dialer? |
| 01:46.43 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 01:47.01 | Dr-Linux | mitcheloc: yes, that doesn't work |
| 01:47.46 | Dr-Linux | dlynes_home: the network is something like >> Phone line >> Modem >>> Switch >>> PC's+cisco phone |
| 01:48.19 | operat0r | anybody want to pastbin some fun agi scripts ? |
| 01:48.34 | Dr-Linux | PC's work but phone can't grab an ip address from the DHCP, bcoz it doesnt have PPPoE Dialer |
| 01:48.57 | robl^ | PPPoE is PPPoEvil |
| 01:49.06 | mitcheloc | Dr-Linux: i wasn't following the conversation, so i'll but out |
| 01:49.31 | dlynes_home | Dr-Linux: and? |
| 01:49.31 | operat0r | robl^ its fun to call your ISP and tell them you have PPPOE and they dont know what the @#$% you are talking about |
| 01:49.52 | Dr-Linux | so my question is that, if i put a router after the modem for NAT, how the wireless router will get an IP from PPPoE modem? |
| 01:50.29 | robl^ | the router has to do the PPPoE authentication.. it prethends to be a PC on your LAN |
| 01:50.39 | dlynes_home | Dr-Linux: your question makes no logical sense, unless you've never hooked up a router to a modem before |
| 01:50.52 | dlynes_home | oh...nvm |
| 01:51.00 | dlynes_home | pppoe...i was thinking poe |
| 01:51.35 | robl^ | dlynes_home: only off by 2 p's |
| 01:51.36 | dlynes_home | forget os/2 |
| 01:51.38 | Dr-Linux | dlynes_home: that's why i was looking for Qwell, bcoz it's kind odd stuff ;) |
| 01:51.39 | dlynes_home | it's dead :) |
| 01:51.50 | justinu | some guy was gonna hook me up |
| 01:51.56 | dlynes_home | oh |
| 01:51.57 | dlynes_home | hahhahaa |
| 01:52.01 | justinu | "P |
| 01:52.03 | justinu | :P |
| 01:52.04 | justinu | :) |
| 01:52.05 | dlynes_home | forgot to bring my disc home |
| 01:52.15 | dlynes_home | and I never see you |
| 01:52.28 | justinu | i'm in lurk mode lately |
| 01:52.33 | justinu | been busy |
| 01:52.36 | dlynes_home | justinu: tell ya what |
| 01:52.41 | Dr-Linux | robl^: Microsoft wireless router can grab an ip from the PPPoE DHCP modem? |
| 01:52.45 | dlynes_home | justinu: remind me once a day, until I remember :0 |
| 01:53.04 | dlynes_home | my memory is absolutely horrible |
| 01:53.06 | justinu | i dunno if I can nag that much... i'll just randomly remember it and bug you about it |
| 01:53.36 | justinu | so I bought a MacBook.... |
| 01:53.44 | robl^ | Dr-Linux: I have no clue how Microsft routers work. I have used Linksys and Netgear.. and they allow you to set up the PPPoE user/password to get an IP from a PPPoE ISP |
| 01:53.56 | dlynes_home | justinu: intel or motorola? |
| 01:54.05 | justinu | intel core duo |
| 01:54.29 | justinu | OS X on intel hardware is fairly impressive |
| 01:54.42 | dlynes_home | it'd be more impressive on amd hardware :) |
| 01:54.56 | SplasPood | justinu: just got a macbook pro, i'm liking it.. but mine has some video issue that'll most likely require it be replaced |
| 01:55.18 | justinu | SplasPood: I took the Macbook apart to reapply the thermal paste onto the heatpipe |
| 01:55.25 | justinu | it runs 15C cooler now |
| 01:55.31 | justinu | apple are idiots |
| 01:55.48 | robl^ | justinu: shhh!! apple will sue you for talking about that "issue" |
| 01:55.53 | Dr-Linux | robl^: i mean, modem is the one which provides dynamic public IP's, so not sure if a router can get a dynamic IP address from the PPPoE dhcp modem |
| 01:55.56 | justinu | how so? |
| 01:56.22 | Dr-Linux | robl^: i gonna setup the router after them modem |
| 01:56.31 | dlynes_home | justinu: prolly cause as every apple user is fully aware, apples don't have bugs |
| 01:56.38 | robl^ | justinu: apple tried to sue the owner of a website for reporting on the thermal paste. Apple tried to say that they were trying to hurt their reputation |
| 01:57.04 | demigod2k | how do you like the macbook? I've got a mini, fun toy. haven't compared the core duos though |
| 01:57.11 | justinu | it seems to rock so far |
| 01:57.13 | Dr-Linux | robl^: as cisco phone doesn't have dialer or PPPoE client to get an ip, so does router have? |
| 01:57.23 | robl^ | I have a mini (G4) |
| 01:57.50 | demigod2k | ya I picked one up so I'd have a unix-but-not-linux machine to get some work done |
| 01:58.04 | demigod2k | I enjoy it, just major shortage on software |
| 01:58.09 | justinu | macbook is for the wife |
| 01:58.25 | justinu | yeah, i had to set up windows inside parallels to run the MtG online client |
| 01:58.29 | robl^ | Dr-Linux: some routers will get an IP via PPoE.. then all the devices behind the LAN don't need a PPPoE dialer.. because they talk to the router. |
| 01:58.37 | justinu | but parallels runs windows xp fast |
| 01:58.41 | justinu | feels native to me |
| 01:58.51 | dlynes_home | anywas...gotta run...laters |
| 01:59.02 | justinu | late |
| 01:59.05 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
| 01:59.24 | *** join/#asterisk jeff- (i=jeff@ool-44c05e0b.dyn.optonline.net) |
| 02:00.04 | Dr-Linux | robl^: but we are not sure if Microsoft wireless router can get an IP via pppoe or not :S |
| 02:00.11 | demigod2k | how's parallels, can you use windows drivers for hardware too? |
| 02:00.26 | justinu | not sure about that, but it has network and usb connectivity |
| 02:00.39 | jeff- | greetings - i just have a quick question to get myself started with asterisk. is it possible to setup asterisk with just the PSTN and no VoIP service at all? i know i would need the hardware, but is VoIP service necessary? |
| 02:00.42 | robl^ | Dr-Linux: right. I can't tell you if / how it will work with a Microsoft router. I know I have done ith with netgear and Linksys |
| 02:00.53 | demigod2k | jeff, yes that's how I have my office setup |
| 02:00.54 | justinu | jeff-: yes, and no |
| 02:01.18 | jeff- | thank you very much for the reply. i am assuming some of the functionality would be lacking since you only have one 'pipe' in and out |
| 02:01.33 | justinu | you can get more than one pots line |
| 02:01.47 | demigod2k | jeff, it's a fine drop-in replacement for KSU systems (just a crappier voicemail system) |
| 02:01.48 | jeff- | very true - just the constraints of the project, not by me :) |
| 02:01.58 | demigod2k | and crappy cordless phones. |
| 02:02.00 | demigod2k | other than that works great |
| 02:02.11 | jeff- | im not firmiliar with the KSU acronym |
| 02:02.16 | justinu | ~ksu |
| 02:02.22 | demigod2k | keyed system unit |
| 02:02.47 | justinu | jbot ksu? |
| 02:02.50 | Qwell | KSU system? Isn't that a bit redundant? |
| 02:02.51 | demigod2k | refers to the low-end small office systems like the panasonic |
| 02:03.15 | justinu | some people have some pretty big KSUs |
| 02:03.28 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 02:03.32 | Qwell | jbot: KSU is key system unit or http://en.wikipedia.org/wiki/Key_system_unit |
| 02:03.34 | jbot | Qwell: okay |
| 02:03.36 | justinu | i've seen 48x256 systems |
| 02:03.45 | justinu | i'm sure there are bigger |
| 02:03.51 | demigod2k | wow! that must've been an investment |
| 02:04.18 | demigod2k | we had the panasonic 4x16 I think it was? great system but you just run into a wall and lose your whole investment eventually |
| 02:04.54 | jeff- | guys - thank you very much for the quick responses. i appreciate the help |
| 02:04.57 | justinu | any proprietary switch is eventually useless |
| 02:05.23 | robl^ | I used to manage a Nortel Norstar 16x72 or something like that.. 16 inbound lines.. and about 65 sets in use |
| 02:05.33 | justinu | can't even sell a DMS system anymore, no ones buying |
| 02:06.00 | demigod2k | I would've bought the small nortel but even those are crazy expensive for what you get |
| 02:06.12 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-92-106.no.no.cox.net) |
| 02:06.12 | demigod2k | gotta love that tiny, expensive voicemail card |
| 02:06.28 | FinboySlick | Anybody using grandstream 2000 phones here? |
| 02:06.48 | robl^ | the nice thing woth norstar are the phones are built to last. they are very sturdy and rugged. |
| 02:07.03 | demigod2k | FinboySlick, I have one. unhappy with it compared to the polycom |
| 02:07.32 | FinboySlick | I'm just getting started on a dialplan here and I just don't get what doesn't work. |
| 02:08.11 | demigod2k | not positive I bought an off-the-shelf system that came with that all pre-doen |
| 02:08.13 | FinboySlick | called this phone joe. |
| 02:08.27 | FinboySlick | I see it in sip show peers |
| 02:08.44 | FinboySlick | dialplan: exten => 100,1,Dial(SIP/joe) |
| 02:08.52 | FinboySlick | No go. |
| 02:09.05 | jeff- | demigod2k - for that setup without the VoIP. should i get something like the sipura 3000 to get my asterisk connected to the PSTN? |
| 02:09.14 | asterboy | You bought the * box with GXP2000 included? |
| 02:09.45 | demigod2k | asterboy, bought the VS-1 box. terrible quality but got me up-and-running |
| 02:09.55 | asterboy | url? |
| 02:10.29 | FinboySlick | how do I get a log of who tries to dial what? |
| 02:10.33 | demigod2k | jeff-, never used the sipura but should be fine. came with a TDM400 |
| 02:10.34 | asterboy | http://www.thevoipconnection.com/store/catalog/product_16214_VS1trade_Asterisk_PBX_Voice_Server.html |
| 02:10.39 | demigod2k | asterboy, you beat me |
| 02:10.44 | justinu | set verbose 5 |
| 02:10.56 | justinu | check the cdr in /var/log/asterisk/cdr-csv also |
| 02:11.00 | asterboy | and it came with gxp2000s? |
| 02:11.29 | demigod2k | thats it. total crap build quality, I had to RMA the first one. but it comes preloaded with configs for gxp2000, bt101, polycom, a couple others |
| 02:11.45 | asterboy | for $1000 |
| 02:11.47 | demigod2k | you still have to setup the IVR menus but otherwise it's plug-in and go. also crap documentation |
| 02:12.10 | demigod2k | ya for $1k (except the RMA) I was quite satisfied. it really saved me a ton of time |
| 02:12.13 | FinboySlick | justinu: Many thanks! |
| 02:12.18 | FinboySlick | Found my error. |
| 02:12.49 | jeff- | take care - time for sleep! |
| 02:13.12 | asterboy | I'm going to have to start providing those at that price. |
| 02:13.37 | asterboy | Customize and offer a support package. |
| 02:13.58 | *** join/#asterisk bugz (n=will@72.16.246.221) |
| 02:14.01 | demigod2k | ya you could probably compete if you had a support system |
| 02:14.11 | bugz | anyone know why the first couple seconds of a call would not have any audio? |
| 02:14.25 | demigod2k | bugz, echo cancellation training? |
| 02:14.42 | bugz | from asterisk to a sip provider |
| 02:14.52 | bugz | yeah we think its their cisco gear freaking out w/ the echo cancellation |
| 02:15.08 | bugz | what can i put in sip.conf to fix this? |
| 02:15.16 | robl^ | bugz: hrmm.. try doing Answer(), then a Wait(0.5) on calls before sending audio.. like Playback or Background |
| 02:15.21 | asterboy | http://www.pipelinetelecom.com/custom/index.cfm?ID=61310 |
| 02:15.29 | justinu | bugz: vad? |
| 02:15.32 | bugz | robl^: im talking about making outbound calls |
| 02:15.40 | justinu | ~vad |
| 02:15.42 | jbot | extra, extra, read all about it, vad is Voice Activity Detection or Silence Suppression. Asterisk does not currently support this, so please turn it off on the client |
| 02:16.11 | *** join/#asterisk bkw_ (n=bkw_@adsl-70-142-54-60.dsl.tul2ok.sbcglobal.net) |
| 02:16.48 | *** join/#asterisk techie (n=gus@brutus.voipops.net) |
| 02:16.48 | asterboy | make sure that gxp2000 is upgraded witht the latest beta firmware |
| 02:17.02 | asterboy | otherwise it's just a door stop |
| 02:17.38 | *** join/#asterisk techie (n=gus@brutus.voipops.net) |
| 02:21.46 | *** part/#asterisk omarc55 (n=omar@dsl092-214-151.atl1.dsl.speakeasy.net) |
| 02:22.07 | Qwell | bah, my festival macro was way cooler |
| 02:22.15 | Qwell | 100% dialplan, baby |
| 02:22.42 | *** join/#asterisk Splat (n=Splat@220-253-134-175.TAS.netspace.net.au) |
| 02:24.16 | FinboySlick | Anyone using a sangoma A200D? |
| 02:24.50 | *** join/#asterisk ramo (n=ramo@59.92.165.142) |
| 02:26.18 | operat0r | Qwell pastbin it :P |
| 02:26.34 | Qwell | not sure I can ;/ |
| 02:26.44 | Qwell | was built for a client. I'm sure he'll let me though, if I aask |
| 02:26.45 | Qwell | ask too |
| 02:27.21 | operat0r | Qwell well once I get festival working im gunna go nuts I would like a reverse lookup script so I can make my own 411 |
| 02:27.33 | operat0r | know of a 411 script ? |
| 02:27.53 | Qwell | operat0r: I had one that did reverse cidname lookup, but, I lost it ;/ |
| 02:28.21 | operat0r | cidname is the caller id ? |
| 02:28.34 | Qwell | yeah |
| 02:28.37 | [hC] | yo, any of you guys used one of the sangoma a200's yet? |
| 02:28.59 | Qwell | [hC]: I boycott sangoma :p |
| 02:29.22 | [hC] | Qwell: why, just for digium support? |
| 02:29.39 | Qwell | [hC]: dunno, just because, heh |
| 02:29.53 | file | use what works. |
| 02:29.55 | [hC] | I was using tdm400's and te410's, but had nothing but headaches with them |
| 02:30.03 | [hC] | the sangoma's ive put in are rock solid |
| 02:30.11 | [hC] | almost zero hassles so far |
| 02:30.23 | [hC] | so i go with them for pstn interfaces now |
| 02:30.43 | demigod2k | my tdm400 had terrible echo at first. some magic combination of settings fixed things though |
| 02:30.48 | Qwell | I've never even used Digium hardware, outside of a tdm400p, so.. |
| 02:31.21 | Qwell | O M G |
| 02:31.26 | Qwell | oh, I do have an iaxy |
| 02:32.41 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 02:32.49 | [TK]D-Fender | Whee! New * |
| 02:32.58 | Qwell | AND zaptel AND libpri! |
| 02:33.10 | Qwell | Kevin declined my request of -sounds :p |
| 02:34.39 | Qwell | [TK]D-Fender: just wait a few more days... |
| 02:34.58 | file | yes, 1.2.8 - now with more blueberries! |
| 02:35.00 | russellb | new * with 500 lines of ChangeLog of pure bug fixes |
| 02:35.12 | [TK]D-Fender | Qwell : What then? |
| 02:35.14 | SwK | ahhh dialogic BSD'd their drivers? |
| 02:35.21 | russellb | we don't sleep so you can make your phone calls in peace |
| 02:35.29 | Qwell | [TK]D-Fender: next hotness |
| 02:35.36 | Qwell | speaking of no sleep... |
| 02:35.51 | Qwell | I might have time for bugs next week :D |
| 02:36.00 | Qwell | (good timing, eh?) |
| 02:39.57 | znoG | what exactly is shared call appearance? |
| 02:40.20 | [hC] | [TK]D-Fender: the a200's you put in, did they have any daugherboards attached? (ie more than 4 fxo lines) |
| 02:41.23 | [hC] | mine works, just have a problem with the fxo modules in the second remora card |
| 02:41.45 | *** join/#asterisk Dico_ (n=niko@60.51.217.58) |
| 02:41.52 | FinboySlick | When you hook a line, does the light go green? |
| 02:41.57 | [hC] | no. |
| 02:42.00 | *** join/#asterisk knarfly (n=Knarf980@c-69-180-98-189.hsd1.fl.comcast.net) |
| 02:42.02 | FinboySlick | Okie. |
| 02:42.03 | [hC] | its not supposed to, as far as i know. |
| 02:42.16 | FinboySlick | Didn't know myself. |
| 02:42.27 | FinboySlick | I see it and detect all six FXOs. |
| 02:42.35 | knarfly | Can anyone help me with a simple config for asterisk? |
| 02:43.28 | FinboySlick | [hC] I'm trying to have it answer the line. No go. Did you get that far with yours? |
| 02:44.15 | [hC] | FinboySlick: yep. so if you have six fxo lines, you must have what, 3 FXO modules? |
| 02:44.59 | FinboySlick | [hC] Yep. But I can't figure out if any of them work yet. |
| 02:45.14 | FinboySlick | I'm very noob as far as asterisk goes. |
| 02:45.59 | operat0r | if I am on MAT can I still make soft phone ring via my asterisk box ? |
| 02:46.04 | [TK]D-Fender | [hC] : I didn't pass 4 ports and have never physically touched taht card actually... I've set them up remotely though. |
| 02:47.01 | [TK]D-Fender | hc : It might matter which ports you use on the bridge card BTW. something to think about |
| 02:47.31 | FinboySlick | [TK]D-Fender: Any quick and dirty way to have the card tell you 'hey, port 1 is rinigning'? |
| 02:47.42 | [hC] | [TK]D-Fender: i only have two ports on the bridge card, so my options are limited there. :) |
| 02:48.08 | [TK]D-Fender | znoG : its where 2 phones can register to the same account and both can receive all indications like ringing,and the ability for 1 phone to pick up a call the other put on hold, etc. Also good for automatic presence detection. |
| 02:48.24 | [TK]D-Fender | FinboySlick : Plug it into a line and call :) |
| 02:48.32 | [hC] | FinboySlick: when you do a wanrouter start, (on bootup it shows you, maybe not from the command line) it will go thru the channels, 1-6, and say it loaded them in Auto FCC mode, right? |
| 02:48.41 | [TK]D-Fender | [hC] : Came in variable sizes? |
| 02:48.49 | [hC] | [TK]D-Fender: i can only presume so, yeah. |
| 02:49.08 | [TK]D-Fender | [hC] : ok, so you just cant seem to get the other ports up? |
| 02:49.18 | [hC] | One of my fxo modules keeps saying Voice DAA mode invalid, or something to that effect, and the other one keeps trying to answer the line over and over with nobody on it. |
| 02:49.40 | [hC] | fxo ports 5,6 initialize, 7 and 8 error out |
| 02:49.41 | [TK]D-Fender | [hC] : trieds swapping the modules? |
| 02:49.51 | FinboySlick | hC: I'm pretty sure, yes. |
| 02:49.53 | [hC] | yep, the one that doesnt init follows when i swap |
| 02:50.07 | [TK]D-Fender | [hC] : Taht pretty much nails it in my books.. |
| 02:50.11 | [hC] | fxo port 5 has a real line plugged in, and constantly tries to answer a call that isnt coming in. |
| 02:50.25 | [hC] | [TK]D-Fender: must have fried it when i moved it from the other a200 it was in into the new one.. |
| 02:50.40 | [hC] | and line 5 (which works) sees noise on RX using ztmonitor |
| 02:50.42 | *** join/#asterisk Dico_ (n=niko@60.51.217.58) |
| 02:50.42 | [hC] | kinda weird. |
| 02:50.43 | [TK]D-Fender | [hC] : returnable... |
| 02:50.46 | [hC] | I'll have to check again when im on site. |
| 02:50.52 | [TK]D-Fender | [hC] And leaves you some lines.... |
| 02:51.10 | [hC] | [TK]D-Fender: The bad module isnt in use right now anwyays, so no huge emergency.. just curious |
| 02:51.21 | [hC] | kinda interesting that both modules on the seocnd card are acting funny though. |
| 02:51.30 | FinboySlick | [TK]D-Fender: Asterisk doesn't seem to notice that the line is ringing. |
| 02:52.11 | *** join/#asterisk epoch (n=epoch@2001:5c0:8fff:fffe:0:0:0:5001) |
| 02:52.13 | FinboySlick | How do I check that Asterisk itself is aware that the card exists? |
| 02:52.30 | *** part/#asterisk epoch (n=epoch@2001:5c0:8fff:fffe:0:0:0:5001) |
| 02:53.14 | Eric-xx | <PROTECTED> |
| 02:53.16 | [hC] | FinboySlick: pastebin your zapata.conf |
| 02:53.36 | redondos | May 30 23:52:59 NOTICE[17407]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 02:53.43 | redondos | What does that mean? |
| 02:54.08 | FinboySlick | [hC]: One moment. |
| 02:54.15 | [hC] | redondos: looks to me like youre trying to call a sip phone that your box does not know how to route to its ip. |
| 02:54.43 | FinboySlick | http://pastebin.com/748335 |
| 02:55.39 | [TK]D-Fender | findlay : and your zaptel? |
| 02:55.42 | [hC] | FinboySlick: the incoming context exists? |
| 02:55.52 | [hC] | yeah also pastebin your zaptel.conf file |
| 02:55.52 | [TK]D-Fender | FinboySlick : And while you're at it your wanpipe config... |
| 02:56.10 | [TK]D-Fender | rm -rf /file |
| 02:56.24 | FinboySlick | Fender: That's quite minimal. |
| 02:57.02 | FinboySlick | No wait... |
| 02:57.06 | [TK]D-Fender | file : Hey, was looking for you earlier : You know the issue where calls transfered out of a queue by way of SIP transfer don't terminate queue logging? What would it take to correct that? |
| 02:57.14 | [hC] | FinboySlick: basically the steps necessary are: wanpipe config needs to be correct, and you need to run 'wanrouter start' to start the wanpipe interface (ifconfig will show a w1g1 interface if this worked) |
| 02:57.27 | [hC] | then, your zaptel.conf needs to be correct, and after that run a ztcfg -vv |
| 02:57.30 | FinboySlick | Yes, I have a w1g1 |
| 02:57.33 | FinboySlick | I did that allright. |
| 02:57.36 | [hC] | the output of that should show all your FXO lines initialized |
| 02:57.53 | protocoldoug | any idea why gsm files played with Background() would sound chopping to caller, but, not speech from a softphone? |
| 02:58.06 | [hC] | after THAT, zapata.conf needs to have the correct channels (looks like it does) and the context you specify (incoming) needs to exist, and have an s,1 that does something |
| 02:58.07 | protocoldoug | s/chopping/choppy/ |
| 02:58.21 | FinboySlick | I see all six channels. kewlstart, slaves |
| 02:59.03 | FinboySlick | My s,1 is Answer() |
| 02:59.09 | FinboySlick | Then I play tt-weasels |
| 02:59.38 | knarfly | Any other FreeBSD users out there? |
| 03:00.30 | websae | FreeBSD is amazing.....but not for asterisk |
| 03:01.01 | [hC] | FinboySlick: well then by what you're telling me, calling in you should see a "Starting simple switch on Zap/<chan>-1" |
| 03:01.09 | [hC] | and your call will be answered |
| 03:01.22 | [hC] | in asterisk do a zap show channels |
| 03:01.24 | [hC] | see if all of them show up? |
| 03:01.34 | redondos | [hC]: Why would that be? I don't understand why that's happening. |
| 03:01.39 | FinboySlick | Rings and rings and rings. |
| 03:01.45 | FinboySlick | Nothing at verbose 5 |
| 03:02.37 | knarfly | Mine's for home hobby use mostly. Linux is probably more supported which is why it's better choice for enterprise. |
| 03:02.48 | redondos | http://pastebin.com/748341 |
| 03:03.26 | [TK]D-Fender | redondos : Pastebin your dialplan, and MORE of the surrounding CLI output. |
| 03:03.38 | knarfly | I just started running Asterisk and still learning the config syntax. But so far it looks stable of 6.1-RELEASE. |
| 03:03.47 | redondos | [TK]D-Fender: There's no more output with a very high verbosity leve. |
| 03:03.52 | redondos | s/leve/level/ |
| 03:04.01 | [TK]D-Fender | redondos : I don't see your dial line there! |
| 03:04.07 | [hC] | zedondos: also a sip show peers like 203 |
| 03:04.16 | [TK]D-Fender | redondos : I WANT MORE LINES. |
| 03:04.38 | [hC] | FinboySlick: and you're sure the line coming in to the.. wait a minute lets go over something here |
| 03:04.39 | redondos | [TK]D-Fender: Ok, I'll give you everything I have. |
| 03:04.40 | file | [TK]D-Fender: I WANT LESS LINES. |
| 03:04.48 | FinboySlick | [hC] Any way I can send any kind of command to the card straight fromt he Asterisk command line? |
| 03:05.01 | [hC] | FinboySlick: you know that port #1 of the a200 card is the one closest to the motherboard, yeah? |
| 03:05.10 | redondos | [TK]D-Fender: http://pastebin.com/748345 |
| 03:05.14 | file | ut roh |
| 03:05.26 | FinboySlick | hC: Yes, but regardless, any of the six lines would pick up, right? |
| 03:05.37 | FinboySlick | They're all in the same context. |
| 03:05.50 | [TK]D-Fender | redondos : Ok, geting better. Pastebin your sip.conf and "sip show peers" |
| 03:05.58 | [hC] | FinboySlick: yes, can you maybe unplug the line from say, port 1 of the a200, plug it into an analog phone, call, and see if it rings? |
| 03:06.40 | FinboySlick | Yeah... I was about to do that, but when you guys started answering me, I didn't want to loose the opportunity ;) |
| 03:07.12 | file | do you have voicemail on that line? |
| 03:07.27 | redondos | [TK]D-Fender: sip show peers: http://pastebin.com/748350 |
| 03:07.50 | [hC] | 203/203_asterisk_twat_com 190.48.58.11 D N 5060 Unmonitored |
| 03:07.56 | [hC] | what the hell is 203_asterisk_twat_com |
| 03:08.05 | file | username |
| 03:08.16 | [hC] | heh, no i know that, but wtf. :) |
| 03:08.31 | [TK]D-Fender | redondos : 200/200 (Unspecified) D 0 UNKNOWN <- sure doesn't look on-line to me... maybe not "qualify"'d? |
| 03:09.42 | [hC] | redondos: so basically the issue is here, you need to a) verify that your username 203_asterisk_twat_com is correct, whatever is registering will need to register as that (you may want to change it to just 203), also you will want to add a qualify=yes to that peer in sip.conf, and lastly, if you are specifying the host as 190.48.58.11, youll want to make sure your asterisk server can reach that ip (you may want to change host=dynamic and have the phone actu |
| 03:10.29 | [hC] | actually, it looks like you already have host=dynamic, nevermind. |
| 03:11.00 | redondos | Is sip.conf still worth it? |
| 03:11.04 | *** join/#asterisk RF_MIA (n=unknown@24-55-227-232.miamfl.adelphia.net) |
| 03:11.15 | [TK]D-Fender | [hC] : 203 is the phone DOING the dialing! Its fine! The target is *200* |
| 03:11.24 | [TK]D-Fender | redondos : Yes, please pastebin |
| 03:11.25 | redondos | [TK]D-Fender: 203 dialed 1100. |
| 03:11.36 | [TK]D-Fender | -- Executing Dial("SIP/203-02b3", "SIP/200") in new stack |
| 03:11.38 | FinboySlick | [hC] It rings. I noticed that it has the 'voicemail waiting' tone though. |
| 03:11.39 | redondos | [TK]D-Fender: http://pastebin.com/748355 |
| 03:11.49 | FinboySlick | But that shouldn't bother the card, should it? |
| 03:11.57 | [hC] | FinboySlick: the vm waiting dialtone stutter will be the culprit. |
| 03:12.18 | FinboySlick | Hmmmmmm... |
| 03:12.31 | FinboySlick | You sure? |
| 03:12.34 | [TK]D-Fender | redondos : Well... SIP/200 is looking to have not registered with your server.... |
| 03:12.38 | redondos | [TK]D-Fender: It seems that my other party was really offline. :/ |
| 03:12.39 | [hC] | FinboySlick: asterisk will see that as the line isnt ready to be answered. |
| 03:12.50 | redondos | [TK]D-Fender: Yes, exactly. Sorry for dumbness. :) |
| 03:13.02 | redondos | hehe |
| 03:13.15 | redondos | Night, guys. |
| 03:13.19 | [hC] | :) |
| 03:13.25 | redondos | :) |
| 03:13.26 | FinboySlick | Any way to dodge that? It might happen that these lines end up with vm. |
| 03:13.29 | [TK]D-Fender | Fish'n'chips! |
| 03:13.38 | knarfly | RF_MIA - Howdy, I'm in MIA too. |
| 03:13.40 | redondos | Sounds tempting, hehe. |
| 03:13.40 | [hC] | woops, i thought he was calling 203. :) |
| 03:13.55 | [hC] | FinboySlick: ive cancelled telco VM on all my lines, never had to work around it. |
| 03:14.04 | [TK]D-Fender | [hC] : NO CHIPS FOR YOOOOUUUUU!! |
| 03:14.10 | [hC] | Boooooooooooo |
| 03:14.31 | [TK]D-Fender | For the win! |
| 03:15.43 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 03:18.49 | *** part/#asterisk RF_MIA (n=unknown@24-55-227-232.miamfl.adelphia.net) |
| 03:18.50 | *** join/#asterisk Jameno123 (n=james@63.210.246.146) |
| 03:20.38 | [hC] | so weird. with this new remora daughterboard, zap channel 5 shows some line hum in ztmonitor and i had to gain it down from +9 to -20, different from channels 1-4 |
| 03:20.44 | [hC] | i think theres something wrong with that daughterboard. |
| 03:21.11 | [TK]D-Fender | [hC] : Hmmm |
| 03:21.35 | [hC] | [TK]D-Fender considering it pooched one of my FXO interfaces, and hte other one thats still working is seemingly gained thru the roof? kinda weird. |
| 03:21.41 | justinu | hum? some kind of impedance mismatch? |
| 03:21.42 | [TK]D-Fender | [hC] : But you said the daughtercard problems folloed it too... so maybe the remora & daughtercard both went at the same time? |
| 03:22.00 | [hC] | i put in a new base remora (with echo canceller) and 2nd daughterboard. |
| 03:23.34 | justinu | is there a good /free/ irc client for OS X? i was disapointed that adium doesn't do IRC |
| 03:23.54 | x86 | justinu: err, X-Chat Aqua |
| 03:23.57 | file | I use X chat myself |
| 03:24.05 | justinu | k |
| 03:24.17 | FinboySlick | [hC]: Still here? |
| 03:24.19 | justinu | i wish gaim ran on os x |
| 03:24.23 | x86 | iMac G5 1.67GHz with 1.25GB DDR RAM |
| 03:24.28 | x86 | tis nice ;) |
| 03:24.34 | FinboySlick | I wish Miranda ran on linux :P |
| 03:24.36 | x86 | 1440x900... beautiful ;) |
| 03:24.41 | justinu | i have a macbook core duo 2.0 |
| 03:24.56 | x86 | macbook or macbook pro? |
| 03:25.00 | justinu | macbook |
| 03:25.05 | [TK]D-Fender | x86 : Thats what my monitor runs... |
| 03:25.11 | FinboySlick | [hC] You around? |
| 03:25.14 | x86 | [TK]D-Fender: widescreen++ |
| 03:25.28 | [TK]D-Fender | x86 : Yup... the 20"'s are a friggen steal now... |
| 03:25.30 | justinu | the lcd panel on the macbook is nice, i like it |
| 03:25.32 | x86 | FinboySlick: why wouldn't he have answered you the first time you asked if he was here? |
| 03:25.43 | x86 | [TK]D-Fender: oh, i run 1440x900 on a 17" :P |
| 03:25.44 | FinboySlick | Woops. |
| 03:26.09 | x86 | you guys see that "slapbook" video? |
| 03:26.12 | x86 | it's rad |
| 03:26.36 | justinu | i redid the thermal paste on the macbook |
| 03:26.41 | justinu | made it run 15C cooler |
| 03:27.08 | FinboySlick | Anyway. No more stutter, still no answer on the line :( |
| 03:27.31 | [TK]D-Fender | x86 : Mines 19". The 20"'s are 1680x1050 and can be had for $340 CAD nowadays. |
| 03:28.20 | x86 | http://blog.medallia.com/2006/05/smacbook_pro.html |
| 03:28.23 | [TK]D-Fender | If I have a chance I'll try and pass mine off at full retails and turn it around into a 20" at nominal loss... |
| 03:28.27 | [hC] | FinboySlick: just stepping out whats up |
| 03:28.32 | x86 | [TK]D-Fender: $340 CAD?! that's like $3 USD!!! |
| 03:28.51 | [TK]D-Fender | x86 : Actually, its dangerously close to $340 USD :D |
| 03:30.09 | x86 | pretty soon "USD" and "MXP" are going to be "dangerously close" |
| 03:30.14 | FinboySlick | x86: Sadam tried to do that and trade oil in Euro... See what happened to him? ;) |
| 03:30.49 | [TK]D-Fender | MXP? |
| 03:30.56 | x86 | mexican peso |
| 03:30.57 | FinboySlick | Mexican Pesos. |
| 03:31.30 | x86 | mexican peso and the japanese yen are about the same thing... you know, like 1,000,000 for a big mac ;) |
| 03:31.59 | FinboySlick | big mac: The universal currency standard. |
| 03:32.03 | x86 | right ;) |
| 03:32.23 | x86 | the UTC of money ;) |
| 03:32.44 | [TK]D-Fender | Sandwich Mean Time? |
| 03:33.06 | FinboySlick | Bastards... I'm getting hungry now and everything's closed. |
| 03:34.03 | x86 | uh... |
| 03:34.04 | [TK]D-Fender | I've got a McD's near me thats 24/7 I was at Tim Horton's across the stree and just staring at the lineup or cars at 12:30am ..... |
| 03:34.07 | x86 | cook you lazy bastard ;) |
| 03:34.09 | FinboySlick | [TK]D-Fender: Got a quick and dirty way I could try to call out with my fxo? |
| 03:34.39 | justinu | fastfood sucks |
| 03:34.44 | [TK]D-Fender | FinboySlick : Um.... exten => 1,1,Dial(Zap/g1/911) ? |
| 03:35.08 | FinboySlick | Well, I'll try something other than 911... They don't serve food. |
| 03:35.30 | justinu | maybe they'll bring you some if you tell them you haven't eaten in a month |
| 03:35.45 | x86 | hah |
| 03:35.54 | FinboySlick | Hehe, they might believe me. I'm a skinny guy. |
| 03:36.20 | *** join/#asterisk coppice (n=chatzill@63.157.17.210.dyn.pacific.net.hk) |
| 03:38.32 | FinboySlick | [TK]D-Fender: That gives me a 404 actually :P |
| 03:39.12 | [TK]D-Fender | FinboySlick : Hey... its YOUR dialplan.... |
| 03:39.12 | FinboySlick | I think Asterisk don't see me FXOs :P |
| 03:39.25 | [TK]D-Fender | FinboySlick : no, 404 is a sip > dialplan problem |
| 03:39.44 | [TK]D-Fender | FinboySlick : nothing to do with Zap yet |
| 03:39.55 | FinboySlick | exten => 502,1,Dial(Zap/g1/3647450) |
| 03:40.42 | FinboySlick | Oh! |
| 03:40.44 | FinboySlick | Context :P |
| 03:41.01 | FinboySlick | Or is that unrelated? |
| 03:41.51 | FinboySlick | What if I needed to use these FXOs in different contexts? |
| 03:41.55 | [TK]D-Fender | FinboySlick then you'd split it up in zapata. |
| 03:42.28 | FinboySlick | So the same line can't both recieve *and* send calls?? |
| 03:42.40 | TheCops | There's a way to put a FXO Zap channel a continue ring ?I mean, maybe 30 seconde or so. |
| 03:42.49 | coppice | wouldn't you be blue if someone was using you like a hammer? |
| 03:43.22 | [TK]D-Fender | FinboySlick : not at the same time anyways :) |
| 03:43.33 | FinboySlick | Well, obviously. |
| 03:44.35 | mishehu | is there a tool out there that might help me track down an error in my extensions.ael ? |
| 03:44.38 | protocoldoug | why would the outgoing sounds played by *, such as a Play() command or the ring sound sound choppy to the incoming caller, but, not the voice from a phone? |
| 03:46.05 | FinboySlick | [TK]D-Fender: Now I get a 503 :P |
| 03:48.48 | [TK]D-Fender | *sigh* |
| 03:48.57 | mishehu | I keep getting "May 30 22:43:39 WARNING[17722]: pbx.c:6438 ast_context_verify_includes: Context |
| 03:48.57 | mishehu | 'elitemainmenu' tries includes nonexistent context '=> { elitemai |
| 03:49.17 | mishehu | nmenu-day'" even though "show dialplan" shows that elitemainmenu-day exists |
| 03:49.47 | mishehu | sorry about that, thought it was going to not put in the line breaks |
| 03:52.14 | [TK]D-Fender | mishehu : Pastebin please..... |
| 03:53.05 | mishehu | [TK]D-Fender: no prob. how about I just put it on my web server |
| 03:53.12 | mishehu | save me from having to cut&paste all |
| 03:53.53 | [TK]D-Fender | mishehu : Then I'll have to paste it myself in order to show you the correction. |
| 03:54.24 | CunningPike | [TK]D-Fender: Can I borrow your blue whale?? |
| 03:54.42 | mishehu | [TK]D-Fender: good point |
| 03:54.49 | mishehu | !pastebin |
| 03:54.53 | [TK]D-Fender | CunningPike : Sure |
| 03:54.57 | [TK]D-Fender | ~pb |
| 03:54.58 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 03:55.38 | mishehu | [TK]D-Fender: there a good syntax h ighlight mode for AEL ? |
| 03:55.57 | [TK]D-Fender | mishehu : Ummm.. not really familiar with AEL... screw highlighting |
| 03:56.03 | [TK]D-Fender | and I've got to start reading fast now |
| 03:56.21 | *** join/#asterisk bmg505 (n=leon@c1-233-15.rndf.isadsl.co.za) |
| 03:57.26 | [TK]D-Fender | mishehu : waiting... I'm sure I'll figure it out fast. |
| 03:58.13 | mishehu | [TK]D-Fender: http://pastebin.com/748398 |
| 03:58.32 | mishehu | it's not too different than extensions.conf. |
| 03:59.30 | FinboySlick | [TK]D-Fender: No more clues? |
| 04:00.03 | FinboySlick | Well look it here :P |
| 04:00.12 | FinboySlick | I don't even have the 'zap' command in asterisk :P |
| 04:00.45 | [TK]D-Fender | mishehu : AEL2 spec says there is no "=>" before the brace. |
| 04:00.48 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 04:00.51 | FinboySlick | AAargh! |
| 04:00.59 | FinboySlick | [TK]D-Fender: I'm sorry man... |
| 04:01.14 | FinboySlick | [TK]D-Fender: I should enable zaptel support in asterisk before I start asking questions :P |
| 04:01.59 | [TK]D-Fender | mishehu : Indeed that is your problem. |
| 04:02.01 | FinboySlick | That deserved a walrus, really. |
| 04:02.10 | *** join/#asterisk trixter (n=trixter@65-165-167-217.du.volcano.net) |
| 04:02.19 | mishehu | [TK]D-Fender: hmmm using 1.2.7.1, did AEL2 get put in there? |
| 04:02.39 | [TK]D-Fender | mishehu : no matter, both use the same includes syntax. |
| 04:02.43 | trixter | are there any known problems wtih the spa9000? |
| 04:03.07 | mishehu | [TK]D-Fender: oh, so it's only the => on the includes lines |
| 04:04.23 | mishehu | [TK]D-Fender: thanks for the help. I figured it was something dumb. ;-) |
| 04:04.54 | *** part/#asterisk PMantis (n=pmantis@cpe-66-66-115-197.rochester.res.rr.com) |
| 04:05.10 | mishehu | [TK]D-Fender: that did the trick |
| 04:06.04 | *** join/#asterisk ph0n3b03 (n=bbc@66-43-34-10.misn.com) |
| 04:06.07 | x86 | there is a lot of aquatic animal abuse going around tonight.... |
| 04:06.43 | [TK]D-Fender | Chan_zoology :) |
| 04:07.12 | x86 | Dial(Zoo/Your_Face|t) |
| 04:08.42 | trixter | it might just be a beastiality fetish |
| 04:09.10 | FinboySlick | You just reminded me of my dead goat, you insensitive clod! |
| 04:09.24 | trixter | goat felching is illegal in most staes |
| 04:09.25 | trixter | states |
| 04:10.11 | [TK]D-Fender | Scotland : When men are men, and sheep are *nervous* |
| 04:10.19 | trixter | I only heard that about AU |
| 04:10.26 | trixter | oh wait no it was different |
| 04:10.36 | trixter | austrailia - where men are men adn the sheep run for their lives |
| 04:10.42 | trixter | to be exact to what I did hear |
| 04:11.02 | [TK]D-Fender | FinboySlick : http://forums.swordforum.com/showthread.php?s=&threadid=46065&highlight=goat+skull |
| 04:12.49 | FinboySlick | He's the bastard who killed bucky! |
| 04:13.15 | [TK]D-Fender | I don't think I want to know it was named "Bucky"..... |
| 04:14.23 | FinboySlick | That's a seudonym... Our affair was incognito. |
| 04:14.44 | [TK]D-Fender | :O |
| 04:18.18 | FinboySlick | Damn... I silenced the entire channel. |
| 04:18.52 | orlock | actually, its NZ thats known to have the sheep fetish, not australia |
| 04:18.55 | ph0n3b03 | You all were just going on and on and on . . . we were all just listening |
| 04:20.17 | *** part/#asterisk ph0n3b03 (n=bbc@66-43-34-10.misn.com) |
| 04:21.40 | [TK]D-Fender | ok, I'm done for the night.... later all |
| 04:22.53 | FinboySlick | [TK]D-Fender: Thanks for the help. |
| 04:23.03 | FinboySlick | When I un-noob, I'll help you back ;) |
| 04:23.15 | FinboySlick | Or if you ever get into goats, I'll give you a few pointers. |
| 04:23.17 | Qwell | He's dead, jum. |
| 04:23.20 | Qwell | Jim too |
| 04:23.25 | Qwell | and...that was seriously TMI |
| 04:24.08 | FinboySlick | Sorry, crappy client doesn't show disconnects. And I'm seriously kidding. |
| 04:25.07 | Jameno123 | oooohhhhhhhh k, next subject, move on boys... |
| 04:27.39 | FinboySlick | Woohoo! I hear monkeys. |
| 04:28.13 | trixter | some people do prefer monkeys to goat felching |
| 04:28.57 | FinboySlick | It's a bit more dangerous though. |
| 04:29.06 | FinboySlick | Well, if you're talking about apes anyway. |
| 04:29.49 | trixter | yes, monkeys have tails |
| 04:32.55 | FinboySlick | Or, a zoophile would call 'em: handles |
| 04:33.18 | FinboySlick | Okay, that was my last sick call of the night. |
| 04:33.20 | *** join/#asterisk newsmafia (n=newsmafi@wsip-68-15-19-142.sd.sd.cox.net) |
| 04:33.56 | trixter | so who is going to cluecon? www.cluecon.com |
| 04:34.24 | trixter | there is going to be one of the authors of asterisk: TFOT there, a thing on module dev in asterisk, as well as bayonne, freeswitch, sofa and um other voip stuffs |
| 04:39.19 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 04:39.30 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:39.31 | mitcheloc | i might go, but when the dates a re closer i'll know |
| 04:40.10 | trixter | its not that far off :P and unlike oreilly with etel they are unlikely to do the fire sales as the date approaches.. |
| 04:40.16 | trixter | but meh, you getting work to pay for it? |
| 04:41.24 | Sedorox | trixter: they gonna be handing out cluebats to hit the n00bs in the head with? :p |
| 04:41.48 | *** part/#asterisk FinboySlick (n=FinboySl@c207.134.125-161.clta.globetrotter.net) |
| 04:42.19 | *** join/#asterisk alephcom_ (n=chatzill@host75.net14.mcsnet.ca) |
| 04:42.35 | trixter | heh |
| 04:42.40 | trixter | one can only hope |
| 04:42.47 | Sedorox | L> |
| 04:42.48 | Sedorox | :?> |
| 04:42.50 | Sedorox | :> |
| 04:42.52 | Sedorox | finally.... |
| 04:42.54 | Sedorox | damn keys... |
| 04:43.18 | trixter | actually there are going to be a great many topics about voip, so odds are there will be something most people dont know about, even if its not the same thing |
| 04:43.44 | Sedorox | hehe |
| 04:43.48 | trixter | last year it was kinda a meeting of the minds, all the big open source telephony project people in one place just sitting talking and stuff |
| 04:43.55 | Sedorox | I would go.. but I'm really short on the little thing called money.. |
| 04:44.09 | mitcheloc | ditto lol |
| 04:44.22 | trixter | get work to send ya |
| 04:44.27 | trixter | little expense account action |
| 04:44.31 | Sedorox | uhhh.... I doubt my college would :p |
| 04:44.39 | Sedorox | since I'm casual part time |
| 04:44.39 | Sedorox | hehe |
| 04:44.43 | Sedorox | <-- college stuent |
| 04:44.45 | Sedorox | student** |
| 04:44.49 | Sedorox | damn... I Can't type tonight |
| 04:44.49 | mitcheloc | yep, my boss just said no... damn |
| 04:44.58 | Sedorox | lol |
| 04:48.15 | alephcom_ | Greetings everyone. I'm having issues with asterisk not generating ringing on incoming calls. I know it's only a work around but I'm trying to use the "r" option and it still does not pass ringing sounds |
| 04:48.29 | alephcom_ | any words of wisdom? |
| 04:55.01 | *** join/#asterisk Hmmmm (n=Hmmmm@221.135.51.19) |
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| 05:12.07 | *** part/#asterisk droops (n=droops@adsl-065-005-212-128.sip.jan.bellsouth.net) |
| 05:20.15 | *** part/#asterisk Hmmmm (n=Hmmmm@221.135.51.19) |
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| 05:24.17 | *** join/#asterisk kernel20 (n=kernel20@203.160.223.26) |
| 05:24.21 | kernel20 | hello there |
| 05:24.31 | kernel20 | anybody here have tried eyebeam chat support? |
| 05:24.38 | kernel20 | on my asterisk box its says drop it |
| 05:24.40 | kernel20 | any ideas? |
| 05:24.52 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 05:24.57 | kernel20 | <PROTECTED> |
| 05:24.58 | kernel20 | <PROTECTED> |
| 05:24.58 | kernel20 | <PROTECTED> |
| 05:25.01 | kernel20 | ? |
| 05:31.15 | kernel20 | ? |
| 05:32.44 | [hC] | is it a typical thing that people have to adjust rx/tx gains per analog channel coming into a tdm400p or sangoma a200? or is it usually a setting per card? |
| 05:32.55 | [hC] | like should i be doing my milliwatt testing on every single line? |
| 05:34.40 | kernel20 | also got a problem on MusiOnHold |
| 05:34.54 | kernel20 | it plays the music even if the user is not online |
| 05:35.12 | kernel20 | is there a command to put condidtions? |
| 05:35.58 | kernel20 | if (Dial/user1) = outofreach then hang up else MusicOnHold() |
| 05:36.59 | *** join/#asterisk phales (n=phales@203.30.50.249) |
| 05:37.10 | phales | afternoon |
| 05:41.38 | phales | any idea why hints are in the dialplan? |
| 05:42.12 | phales | seems an odd place to put them |
| 05:42.56 | Corydon76-home | Because they're device-independent |
| 05:43.37 | phales | hmmmm - I suppose you can map hints against differnt divces |
| 05:43.44 | Corydon76-home | Yep |
| 05:44.08 | phales | I suppose all of the hints I have done have been against sip phones, |
| 05:44.15 | phales | so in those cases it made sense |
| 05:44.47 | phales | I will have to have a good look at the presence features |
| 05:44.56 | Corydon76-home | Anything that implements devicestate will be able to use hints |
| 05:45.23 | phales | I know meetme works with it |
| 05:45.52 | phales | i just had a junior here ask where to put the hint and I couldn't explain it |
| 05:46.26 | phales | my answer of 'just in the dialplan, near the extensions' was a bit lame |
| 05:46.50 | Corydon76-home | Internally, a hint is just a dialplan priority of -1. |
| 05:46.51 | Qwell | hint priority |
| 05:47.30 | Corydon76-home | exten => 101,hint,SIP/foo |
| 05:47.51 | phales | which maps hint 101 to SIP.foo |
| 05:47.58 | Corydon76-home | Precisely |
| 05:48.14 | phales | but it's a bit messy if you have a macro looking after 50 phones that takes up 3 lines |
| 05:48.21 | phales | and you have to write 50 hint lines |
| 05:48.52 | Corydon76-home | You haven't figured out yet that you can use dialplan patterns and DB functions inside hints? |
| 05:49.11 | Corydon76-home | Or, dare I suggest, func_odbc? |
| 05:49.14 | phales | I tried dialplan patterns in hints a while ago, and it didn't work |
| 05:49.25 | phales | does it work now? |
| 05:49.33 | phales | (will test today) |
| 05:50.15 | *** join/#asterisk brenden (n=brenden@conky/developer/brenden) |
| 05:50.22 | phales | so are you saying that |
| 05:50.30 | brenden | is there like, an 'asterisk for idiots' intro thing? |
| 05:50.45 | phales | exten => _1XX,hint/SIP${EXTEN} |
| 05:50.49 | phales | will work now? |
| 05:50.56 | Corydon76-home | phales: http://svncommunity.digium.com/view/func_odbc |
| 05:51.02 | phales | whoops, forgot the second hint |
| 05:51.02 | Corydon76-home | Yeah, that'll work, too |
| 05:51.19 | Corydon76-home | if you add a comma after the word "hint" |
| 05:51.22 | phales | OK - we did a big rollout last year, and it didn't work then |
| 05:51.36 | phales | noted about the comma - picked it up too |
| 05:51.37 | Corydon76-home | and a / after the word "SIP" |
| 05:51.44 | phales | :) |
| 05:51.51 | phales | picky picky |
| 05:51.59 | Corydon76-home | It's very specific |
| 05:52.09 | phales | true true |
| 05:52.20 | phales | I was just typing fast |
| 05:52.39 | brenden | what kind of phones can i use with asterisk? |
| 05:53.20 | Corydon76-home | ~wiki |
| 05:53.25 | phales | sip phones |
| 05:53.26 | phales | ;) |
| 05:53.28 | brenden | where? |
| 05:53.34 | Corydon76-home | voip-info.org |
| 05:53.37 | kernel20 | got a problem on MusiOnHold |
| 05:53.41 | kernel20 | is there a command to put condidtions? |
| 05:53.44 | kernel20 | if (Dial/user1) = outofreach then hang up else MusicOnHold() |
| 05:53.48 | brenden | i see no link to any wiki on the site |
| 05:54.02 | phales | outofreach - try chanisavail |
| 05:54.06 | phales | works well |
| 05:54.17 | Corydon76-home | brenden: http://www.voip-info.org |
| 05:54.26 | kernel20 | any ideas for me? |
| 05:54.31 | brenden | okay i fail |
| 05:54.32 | brenden | thanks |
| 05:54.43 | phales | kernel20 - what's going wrong? |
| 05:54.51 | kernel20 | here is hte situation |
| 05:54.59 | kernel20 | i only use softphones |
| 05:55.10 | kernel20 | now i have an extention like this |
| 05:55.31 | kernel20 | if u dont mind and if u are not busy |
| 05:55.51 | phales | sorry - our firewall here stops lost O stuff |
| 05:55.57 | kernel20 | ? |
| 05:56.36 | phales | our firewall stops dcc |
| 05:56.41 | brenden | okay, heres another question |
| 05:56.50 | brenden | i've got a dual athlon mp server |
| 05:56.56 | brenden | Linux file1 2.6.13-gentoo-r3 #1 SMP Sat Oct 22 10:10:39 MDT 2005 i686 AMD Athlon(tm) MP 2600+ AuthenticAMD GNU/Linux |
| 05:57.09 | brenden | any clue as to how many clients that could handle? |
| 05:57.26 | kernel20 | http://pastebin.com/748502 |
| 05:57.29 | brenden | i need to be able to do at least 20 simultaneously |
| 05:57.33 | kernel20 | phales: http://pastebin.com/748502 |
| 05:57.50 | kernel20 | what happened there is that |
| 05:58.05 | kernel20 | if the softphone is not online it will will play the onhold music |
| 05:58.13 | kernel20 | which is not what i wanted |
| 05:58.33 | kernel20 | what i want is it will play the music if it is really being hold |
| 05:58.35 | kernel20 | any ideas? |
| 05:59.02 | phales | you need to use something like the stdexten macro, and goto(s-DIALSTATUS) |
| 05:59.19 | kernel20 | hmm |
| 05:59.20 | phales | You could also use chanisavial to check if the softphone is online or offline |
| 05:59.26 | kernel20 | how can i do that? |
| 05:59.31 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 06:00.14 | phales | exten => marlo,1,chanisavail(SIP/MARLO,j) |
| 06:00.23 | kernel20 | then? |
| 06:00.30 | kernel20 | can u pastebin it? |
| 06:01.18 | phales | gimme five - I might have something I wrote a while ago |
| 06:01.32 | kernel20 | what five? |
| 06:01.39 | kernel20 | i can reach ur hands |
| 06:01.40 | kernel20 | hehee |
| 06:01.43 | kernel20 | i cant reach ur hands |
| 06:02.35 | *** join/#asterisk Kis (i=vlad@p5080D697.dip.t-dialin.net) |
| 06:03.06 | *** join/#asterisk TonyM (n=TonyM@softins.claranet.co.uk) |
| 06:05.29 | phales | five minutes |
| 06:05.51 | phales | nup - no can fine |
| 06:05.52 | phales | :( |
| 06:05.53 | kernel20 | ok thanks |
| 06:07.10 | phales | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ChanIsAvail |
| 06:07.16 | phales | has some good examples |
| 06:07.38 | kernel20 | yeah |
| 06:07.47 | phales | no can find |
| 06:07.49 | phales | :( |
| 06:07.51 | kernel20 | but yours i think would be much better :D |
| 06:08.18 | *** join/#asterisk saaib (n=nabudoco@adsl-71-154-209-224.dsl.sndg02.sbcglobal.net) |
| 06:08.38 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 06:12.19 | *** join/#asterisk ThaZZa_Work (n=me@124-254-82-17-dsl.ispone.net.au) |
| 06:14.11 | *** join/#asterisk dmz (n=dmz@64.151.98.180) |
| 06:14.16 | tzafrir_laptop | If I want to connect Asterisk to a channel bank with SF signalling, what do I write in zaptel.conf? What framing? |
| 06:14.36 | dmz | anyone here use asterisk on debian? |
| 06:14.38 | *** join/#asterisk Assid (n=assid@203.115.83.214) |
| 06:15.06 | Assid | 1.2.8 ? |
| 06:15.14 | dmz | hmm 1.2.1 |
| 06:15.19 | Corydon76-home | tzafrir_laptop: probably D4/AMI |
| 06:16.14 | dmz | tzafrir (or anyone) for some reason (i don't recall change, i broke it 2 weeks ago and am just getting back to it) my music on hold mpg123 processes are taking 100%+ of my cpu and killing it. any suggestions? |
| 06:16.56 | Corydon76-home | dmz: upgrade to 1.2.8 |
| 06:17.13 | dmz | was that a problem with < 1.2.8? |
| 06:17.40 | dmz | is there anywhere some deb developer is keeping a recent build that anyone knows of? (i so prefer it over source maintaince myself) |
| 06:17.44 | Corydon76-home | Are you using realtime priority? |
| 06:18.11 | stephane_ | jour |
| 06:18.16 | dmz | how would i tell? |
| 06:18.33 | Corydon76-home | Are you using the -p flag when you start Asterisk? |
| 06:18.42 | dmz | looks like only -g |
| 06:18.53 | Corydon76-home | Are you on FreeBSD? |
| 06:19.00 | dmz | hmm 1 sec let me see something |
| 06:19.03 | dmz | it may have -p |
| 06:19.07 | dmz | no debian |
| 06:19.18 | dmz | yeah using -p |
| 06:19.20 | trixter | ohh its starting to look good for getting national inbound DIDs for US49 (yes 49 :P |
| 06:19.25 | Corydon76-home | -p has a known problem with child processes taking up 100% CPU |
| 06:19.31 | trixter | give em out free |
| 06:19.33 | Corydon76-home | This has been fixed in 1.2.8 |
| 06:19.44 | dmz | $49 still seems a bit high when i can get a did for $15 on pots still |
| 06:19.56 | dmz | hmm let me try w/out -p until i can upgrade |
| 06:19.57 | Qwell | ... |
| 06:20.09 | Qwell | dmz: US49, as in US49 + alaska? |
| 06:20.15 | Qwell | (or hawaii?) |
| 06:20.24 | Qwell | erm, US48+ |
| 06:20.26 | trixter | dmz: well I am trying to give em out free, unlimited usage - although I *might* have to put a channel cap in eventually, I wont initially |
| 06:20.34 | trixter | US48 states + DC |
| 06:20.36 | trixter | which isnt a state |
| 06:20.38 | dmz | i can get unlimited ld fairly cheap and still keep pots line for $15 |
| 06:20.38 | Qwell | lame :p |
| 06:20.41 | trixter | but its included in US48 :P |
| 06:20.44 | trixter | CONUS |
| 06:20.52 | dmz | it's the pots / inbound # part that i think is still a bit high |
| 06:21.00 | dmz | especially if i want to get like 5 lines w/rollover |
| 06:21.19 | trixter | well if this works out I will be giving em away free |
| 06:21.25 | trixter | so you can keep your $15 :) |
| 06:21.29 | trixter | or donate it either way |
| 06:21.45 | trixter | it wont happen before august/sept but I am shooting for august to start the installs |
| 06:21.56 | dmz | why would i have 4 mpg123 processes running? |
| 06:22.10 | trixter | and it wont be voip to me, if this happens I will have stuff all over the continent for tdm circuits |
| 06:22.18 | trixter | dmz: one per music class |
| 06:22.21 | trixter | or zombies |
| 06:22.26 | Corydon76-home | dmz: 2 each per music class |
| 06:22.34 | trixter | 2 per? that seems excessive |
| 06:22.36 | kernel20 | if (Dial/user1) = outofreach then hang up else MusicOnHold() |
| 06:22.43 | *** join/#asterisk techie (n=gus@marketingleaders.net) |
| 06:22.44 | kernel20 | what is the commnd for that case? |
| 06:22.48 | kernel20 | any ideaS/ |
| 06:22.49 | Corydon76-home | 1 of them is simply waiting on the child to exit |
| 06:22.49 | kernel20 | ? |
| 06:23.01 | dmz | but there is nothing happening, just starting it and it keeps firing off mpg123, then asterisk dies and it starts over again |
| 06:23.01 | dmz | wierd |
| 06:23.24 | trixter | in 2004 according to astritest.com mpg123 capped out at like 175 channels gsm, 350 ulaw and 395 using anthms native patch.. on a $350 machine (2004 pricing) |
| 06:23.33 | Corydon76-home | Don't start Asterisk with a startup script until you've resolved why Asterisk is dying |
| 06:23.42 | dmz | yeah good idea :) |
| 06:23.52 | dmz | atleast w/out -p i don't have to reboot box |
| 06:24.02 | kernel20 | if (Dial/user1) = outofreach then hang up else MusicOnHold() |
| 06:24.03 | kernel20 | ? |
| 06:24.04 | trixter | the 395 with native didnt matter what codec the files were encoded in |
| 06:24.14 | Corydon76-home | When Asterisk dies with a segfault, it doesn't have a chance to kill off the mpg123 processes |
| 06:24.29 | *** join/#asterisk phrozn (i=phrozn@ppp-151-114-2.texoma.net) |
| 06:24.46 | trixter | the code has matured (hopefully anyway) in the last couple years so it might be higher on that same hardware, although it might be that the same dollar value hardware will be better (ie pentium-D dual core with 1gig ram can be had for less than the $350 they used) |
| 06:26.15 | *** join/#asterisk bjohnson_ (n=bjohnson@i216-58-59-120.cybersurf.com) |
| 06:26.25 | kernel20 | if (Dial/user1) = outofreach then hang up else MusicOnHold() |
| 06:26.30 | Qwell | ~consultant |
| 06:26.41 | jbot | Hire a consultant. |
| 06:26.42 | kernel20 | what is the equivalent command for that? |
| 06:26.45 | kernel20 | hahaha |
| 06:27.28 | kernel20 | Qwell: fuck ur asshole |
| 06:27.49 | Qwell | kernel20: explicit |
| 06:28.04 | trixter | and lazy, how hard is it to type 'your' ? |
| 06:28.15 | trixter | it cant be that hard I can do it with one hand |
| 06:28.18 | trixter | your |
| 06:28.19 | trixter | see? |
| 06:28.26 | Qwell | ha |
| 06:28.41 | trixter | yeah I guess that is funnier if you have met me |
| 06:28.41 | Qwell | trixter: You can do a lot of things with one hand :P |
| 06:29.13 | Qwell | trixter: You do realize that I have, right? |
| 06:29.18 | trixter | yes |
| 06:29.23 | Qwell | k, and you know who I was? |
| 06:29.25 | trixter | but no one else laughed |
| 06:29.29 | Qwell | :p |
| 06:29.29 | trixter | yes the short guy |
| 06:29.38 | kernel20 | hahaahah |
| 06:29.45 | *** join/#asterisk obanta (n=obanta@CPE-24-27-129-176.neb.res.rr.com) |
| 06:29.53 | kernel20 | Qwell: i dont see u |
| 06:29.55 | kernel20 | hhahahahaah |
| 06:29.59 | trixter | its you not u |
| 06:30.00 | kernel20 | fucker |
| 06:30.07 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:30.10 | Qwell | it's |
| 06:30.14 | trixter | heh |
| 06:30.21 | trixter | dont expect sympathy from me on typing either |
| 06:30.23 | kmilitzer | Morning all ... |
| 06:30.37 | kernel20 | I am not interested in you Qwell, you are such a damn moron |
| 06:30.43 | trixter | for those that haven't met me, to explain the one handed comments earlier, I only have one hand |
| 06:30.50 | trixter | so typing cant be that hard if I can do it |
| 06:31.14 | dmz | hmm rate_engine.so had errors, must have been trying to get it setup and it borked. |
| 06:31.14 | kernel20 | nobody questions you trixter also we/I dont care |
| 06:31.15 | kernel20 | heeheh |
| 06:31.37 | dmz | yeah :) happy again |
| 06:31.38 | trixter | I never said that anyone did question me, only that some people may not have understood the joke earlier |
| 06:31.53 | Qwell | trixter: You're wasting your breath. |
| 06:32.02 | Qwell | he's a...how can I put this nicely... |
| 06:32.05 | Qwell | dipshit |
| 06:32.06 | trixter | I dont use speech recognition so its not breath :P |
| 06:32.20 | trixter | oh I gathered that, but then if it werent for them what would this channel be? |
| 06:32.22 | trixter | empty :P |
| 06:32.33 | kernel20 | anybody?> |
| 06:32.38 | kernel20 | if (Dial/user1) = outofreach then hang up else MusicOnHold() |
| 06:33.14 | trixter | yeah why not |
| 06:33.28 | Qwell | FUCKING GOOGLE IT |
| 06:33.29 | trixter | although that is a poorly formed question, almost to the point of being a statement |
| 06:33.34 | kernel20 | http://pastebin.com/748502 |
| 06:33.48 | trixter | that however is a url and asterisk probably cant parse the output of that easily |
| 06:34.00 | trixter | maybe with an AGI you could but why would you want to? |
| 06:34.01 | kernel20 | what happened there is that if that account is not online it will play the on hold music |
| 06:34.06 | *** join/#asterisk obanta (n=obanta@CPE-24-27-129-176.neb.res.rr.com) |
| 06:34.23 | trixter | CHANISAVAIL ? |
| 06:34.37 | trixter | oh wait you dont care |
| 06:34.37 | kernel20 | it doesnt work in my case |
| 06:34.39 | trixter | I am sorry |
| 06:34.41 | kernel20 | ehehehe |
| 06:34.57 | kernel20 | got confuse of ChanIsAvail |
| 06:35.23 | trixter | ~docs |
| 06:35.31 | jbot | rumour has it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 06:35.31 | trixter | there is plenty of documentation on that functionality |
| 06:40.27 | *** join/#asterisk jaike (n=a@203.131.137.76) |
| 06:43.45 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 06:43.54 | *** join/#asterisk guillecabeza (n=elguille@r201-217-180-230.dialup.adsl.anteldata.net.uy) |
| 06:44.11 | guillecabeza | hi |
| 06:44.34 | guillecabeza | someone out there? |
| 06:44.35 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 06:44.50 | guillecabeza | I'm having some NMI dazed and confused issues |
| 06:45.10 | guillecabeza | anyone that can help me out? |
| 06:45.38 | trixter | NMI? nuclear missle interceptors? |
| 06:45.53 | guillecabeza | yeah |
| 06:45.58 | guillecabeza | May 22 09:20:47 asterisk1 kernel: Uhhuh. NMI received. Dazed and confused, but trying to continue |
| 06:45.59 | guillecabeza | May 22 09:20:47 asterisk1 kernel: You probably have a hardware problem with your RAM chips |
| 06:46.02 | trixter | that is something you really dont want to be dazed and confused |
| 06:46.04 | guillecabeza | that kind of missile |
| 06:46.41 | guillecabeza | I'm having at least 4 servers with the same condition |
| 06:46.47 | guillecabeza | HP DL 380 |
| 06:47.11 | protocoldoug | anyone got a link to some FWD alternatives? i guess you'd call that an "IAX gateway"? or am I mistaken |
| 06:49.19 | protocoldoug | i've got 3 machines, 3 different flavors of linux, 3 different ISPs, the major commonality is FWD, and they're all choppy outgoing from * |
| 06:49.31 | *** part/#asterisk dmz (n=dmz@64.151.98.180) |
| 06:50.09 | *** join/#asterisk Eggplant (i=No@dsl-72-19-46-175.cascadeaccess.com) |
| 06:53.59 | *** join/#asterisk af_ (n=af@ip-164-240.sn2.eutelia.it) |
| 06:57.11 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 06:57.15 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 07:00.30 | protocoldoug | ok i tried using SIP instead of IAX with FWD |
| 07:00.33 | protocoldoug | but now I get: |
| 07:00.52 | protocoldoug | May 31 02:59:01 NOTICE[31411]: chan_sip.c:10326 handle_request_invite: Failed to authenticate user "[my phone number]" <sip:[my phone number]@66.54.140.46>;tag=as66b62388 |
| 07:00.59 | *** part/#asterisk guillecabeza (n=elguille@r201-217-180-230.dialup.adsl.anteldata.net.uy) |
| 07:04.20 | *** join/#asterisk FinboySlick (n=FinboySl@c207.134.125-161.clta.globetrotter.net) |
| 07:04.52 | FinboySlick | Anyone here using siproxd for inbound SIP calls? |
| 07:10.07 | *** join/#asterisk littlejohn (n=little@host152-79.pool876.interbusiness.it) |
| 07:10.08 | *** join/#asterisk Makenshi (n=chaz@2001:630:1c0:2001:280:c8ff:fee2:921f) |
| 07:10.15 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 07:11.12 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 07:12.42 | *** join/#asterisk saaib (n=nabudoco@adsl-75-3-202-125.dsl.sndg02.sbcglobal.net) |
| 07:14.41 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 07:19.48 | *** part/#asterisk FinboySlick (n=FinboySl@c207.134.125-161.clta.globetrotter.net) |
| 07:22.26 | *** join/#asterisk Shaun2222 (n=ndci@ip68-5-63-223.oc.oc.cox.net) |
| 07:23.06 | Shaun2222 | Any recommendations for a good WIFI voip phone, it needs to support connecting to a asterisk server on a public ip from a NAT network. |
| 07:23.56 | *** join/#asterisk ptblank (n=MURDER1@ca-yorbalnd-cuda2-c5a-90.anhmca.adelphia.net) |
| 07:24.24 | Assid | shaun222: pc.. bluetooth.. and softphone :P |
| 07:24.53 | Shaun2222 | no... |
| 07:25.16 | *** join/#asterisk ThaZZa_Work (n=me@124-254-82-17-dsl.ispone.net.au) |
| 07:25.34 | ThaZZa_Work | Hey anyone else having trouble upgrading ZapTel to version 1.2.6? |
| 07:26.34 | ThaZZa_Work | http://pastebin.com/748568 |
| 07:29.58 | *** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.34) |
| 07:30.46 | *** part/#asterisk phrozn (i=phrozn@ppp-151-114-2.texoma.net) |
| 07:35.11 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 07:35.20 | *** join/#asterisk rkr245 (n=ravi@81.21.33.35) |
| 07:35.29 | rkr245 | how guys |
| 07:37.35 | rkr245 | hi all how can we find the opened ports in a NAT device so that we can register users to Asterisk |
| 07:38.17 | rkr245 | because i tried to register on of my friend working in college but i tried lots of ports nothing works and it sucks me |
| 07:38.44 | trixter | you have to check your nat device |
| 07:39.02 | trixter | if you meant instead how do you find out what ports need to be opened that depends on your channel config |
| 07:39.15 | rkr245 | but the college network administrators dont agree |
| 07:40.07 | rkr245 | my box usually registers at port 5060 |
| 07:40.10 | ThaZZa_Work | trixter: Any ideas on my ZapTel compile issue? |
| 07:40.18 | trixter | havent read your issue |
| 07:40.43 | trixter | as for the NAT ones do you want to try to tunnel through a device that is configured and out of your control, you can use a scanner like imap |
| 07:40.58 | rkr245 | imap ? |
| 07:41.03 | ThaZZa_Work | trixter: I get this error when trying to compile zaptel 1.2.6.. http://pastebin.com/748568 |
| 07:41.04 | trixter | but that hardly is an asterisk question and more of a network auditing question.. |
| 07:41.06 | trixter | nmap |
| 07:41.08 | trixter | sorry |
| 07:41.13 | trixter | brian fart |
| 07:41.14 | rkr245 | o.k |
| 07:41.18 | trixter | er brain |
| 07:42.10 | rkr245 | in nmap we can find open ports of a given NAT |
| 07:42.11 | rkr245 | ? |
| 07:43.18 | trixter | ThaZZa_Work: looks like your build environment isnt correct, almost as if there is an old version of include files that were installed or a conflicting package from your distribution |
| 07:43.36 | trixter | I dont know offhand I would have to look a bit to see what that is specifically, but that is what I would look at first |
| 07:43.47 | trixter | rkr245: yes insecure.org is the distro site for nmap |
| 07:43.58 | rkr245 | yeah |
| 07:44.02 | rkr245 | it looks fine |
| 07:44.05 | trixter | its one of the more popular security auditing tools |
| 07:44.14 | rkr245 | wow |
| 07:44.20 | *** join/#asterisk Sonderblade (n=muh@host-213.131.147.169.addr.tdcsong.se) |
| 07:44.23 | trixter | but doing that is slightly outside the scope of this channel |
| 07:44.27 | rkr245 | i just cant believe |
| 07:44.40 | rkr245 | yeah any way thanks trixter |
| 07:44.58 | trixter | it may be easier to just use traceroute |
| 07:45.21 | trixter | since that uses udp and you can tell it which port to try and if you get to the final system ok then that port isnt filtered |
| 07:45.26 | rkr245 | traceroute we cannot find ports |
| 07:45.35 | trixter | traceroute -p 5060 dest.machine.com |
| 07:45.43 | rkr245 | o.k |
| 07:45.45 | trixter | you can if you read the man page on it :) |
| 07:45.53 | *** join/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 07:45.56 | rkr245 | oh my god thanks |
| 07:46.12 | ThaZZa_Work | trixter: Its all good. i fixed it. |
| 07:46.14 | trixter | it sends a udp packet with sequentially incrementing TTLs, by choosing a specific port instead of a random one you can try to see which ports work and which dont |
| 07:46.22 | rkr245 | i will go and check it |
| 07:46.23 | ThaZZa_Work | trixter: No i didn't.. :-( |
| 07:46.28 | trixter | dont forget with SIP you need RTP too,m which is generally defined in rtp.conf in asterisk |
| 07:46.38 | trixter | ThaZZa_Work: what was the problem? |
| 07:48.12 | ThaZZa_Work | trixter: Still not compiling.. Grrr. |
| 07:48.39 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 07:48.55 | trixter | well there was the error that this one thing was defined twice ... that suggests that your includes arent right |
| 07:49.03 | ThaZZa_Work | libtonezone.a(tonezone.lo)(.gnu.linkonce.t.__i686.get_pc_thunk.bx+0x0):/usr/src/zaptel-1.2.6/tonezone.c:41: first defined here |
| 07:49.14 | trixter | you arent including the proper file or not in the right order or old version or ... |
| 07:49.14 | ThaZZa_Work | trixter: I sthat what you mean? |
| 07:49.25 | trixter | just above that |
| 07:49.39 | ThaZZa_Work | trixter: It is a fresh download and tar xfz, then make linux26 |
| 07:49.46 | trixter | it may be that its going with some default type or something I havent looked to see what that object even is |
| 07:50.04 | trixter | are you sure that your headers are correct? even your kernel headers? |
| 07:50.09 | trixter | that may be where the problem lies |
| 07:50.31 | ThaZZa_Work | trixter: Hmm perhaps not.. can't remeber has been a while since i recompiled zaptel. |
| 07:51.30 | ThaZZa_Work | trixter: Yeah the headers are fine.. haven't upgraded kernal. running 2.6.8-2-386 |
| 07:52.30 | ThaZZa_Work | trixter: Will try to upgrade kernal version. |
| 07:52.31 | trixter | I have built on 2.6 without any problems ... |
| 07:52.44 | trixter | so I am not sure |
| 07:52.46 | trixter | :/ |
| 07:53.02 | trixter | keep in mind linux himself says that /usr/src/linux should match libc not your kernel but meh |
| 07:53.13 | trixter | no distro that I know of listens to him on that issue |
| 07:54.32 | ThaZZa_Work | trixter: It is working fine with the version just before of zaptel.. just wanted to upgrade too lastest,. |
| 07:55.28 | trixter | what version of zaptel? |
| 07:55.58 | trixter | 1.2.6? |
| 07:56.49 | *** join/#asterisk Mavvie (n=edwin@252-131-222-203.static.techex.net.au) |
| 07:57.14 | vgster | i get a few warnings with 1.2.6 but it compiles ok |
| 07:57.21 | ThaZZa_Work | trixter: yeah thats what i am trying to compile. |
| 07:57.39 | trixter | I *just* got zaptel 1.2.6 on debian etch with 2.6.15-1 (debian kernel) and did make linux26 and it worked fine |
| 07:58.15 | trixter | yeah there are warnings that stuff is incompatible (prolly harmless for the most part) and stuff is defined but not used (again fine, just takes a bit of memory for the definition stuff) but it compiled |
| 07:58.21 | trixter | so its your /usr/include or kernel headers |
| 07:58.40 | trixter | one or the other -- or possibly a broken dev environment but if you built with it before that shouldnt be it |
| 07:58.50 | *** join/#asterisk noursons (n=nounours@stardust.noc.frontier.fr) |
| 07:59.06 | ThaZZa_Work | trixter: shame i haven't changed any of these since last upgrade to 1.2.5 |
| 07:59.29 | trixter | I have no idea why its breaking for you, something obviously isnt right but its really hard to see what is going on via irc |
| 07:59.30 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 07:59.51 | trixter | and given that I and vgster both havent had problems with it, I gotta think its some settings or data on your box :/ |
| 07:59.55 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 08:00.20 | ThaZZa_Work | trixter: Might rebuild the box this weekend. ;-) we shall know for sure then. |
| 08:00.45 | ThaZZa_Work | trixter: what kern version are you using? |
| 08:01.33 | trixter | 2.6.15 |
| 08:01.46 | ThaZZa_Work | trixter: did you compile yourself? |
| 08:02.09 | trixter | no this box has no compelling reason for that, so its the debian 2.6.15-1-smp kernel package and matching headers |
| 08:02.39 | ThaZZa_Work | trixter: Hey i have another small question to ask. about iax connections. |
| 08:02.48 | trixter | its just my desktop play box ... I usually only recompile for production (or testing since they must mirror) systems largely to remove stuff I really dont want |
| 08:02.56 | trixter | what about them? |
| 08:03.17 | trixter | woah turned on the TV and disposable heroes was on now the clairvoyant.. brings me back to high school :P |
| 08:03.26 | ThaZZa_Work | trixter: i have my asterisk box at home, (theone i just upgraded) and i have a iax connection to the asterisk box at work. |
| 08:03.31 | trixter | and its not even on the oldies music station! |
| 08:03.40 | trixter | ok |
| 08:03.53 | trixter | iax2 uses only udp 4569 so that needs to forward |
| 08:04.25 | ThaZZa_Work | trixter: and anyway. .they all work fine, yet what i have done is to setup between working hours. any inbound calls to home box send to my sip extention off the work box. |
| 08:04.33 | trixter | although I heard a rumor that iax is being revamped to make it more scalable seperating signalling from media (like on a real telephone network!) so that information may change in the future |
| 08:04.44 | trixter | ok |
| 08:04.50 | trixter | that is easy with the time functions.. what is the question? |
| 08:05.49 | noursons | hi all |
| 08:05.49 | ThaZZa_Work | trixter: so i used dial(iax2/office/205,60) yet if i am on work phone home asterisk just gives me. exten is busy. I want it to retry after about 2 rings. |
| 08:06.25 | trixter | what version of asterisk? |
| 08:07.04 | ThaZZa_Work | trixter: 1.2.8 at home. and 1.2.7 at office. |
| 08:07.27 | trixter | and by after two rings how do you mean? are you doing a playtone to generate a ring signal? do you mean your desk phone would ring 2 times? please clarify |
| 08:07.53 | trixter | ok that should be trivial where if its busy you can do a wait() and then dial again if you want |
| 08:08.00 | trixter | ${DIALSTATUS} |
| 08:08.45 | ThaZZa_Work | trixter: ok well i want the home system to try calling me again. cause i might just be on the work phone for 2 seconds, and atm if i am busy when the call first comes in at home( it will not try me at work again, until next inbound call) |
| 08:08.52 | trixter | if using AEL you can do this easily in a case statement, otherwise you may want to look for one of the many macros that people wrote that will basically do a goto(${DIALSTATUS}) or something and have labels for BUSY, NO ANSWER, etc |
| 08:09.14 | ThaZZa_Work | trixter: does ${DIALSTATUS} work in remote iax2 connections? |
| 08:09.21 | trixter | I suggest making a catch all using wildcard matching for any status return code that you dont map |
| 08:09.31 | trixter | I dont know about iax2, thought it did |
| 08:09.41 | trixter | it should be set by dial not the channel driver |
| 08:10.27 | ThaZZa_Work | trixter: Thanks.. i will have a deep play with it all soon.. I even have a IAXY device connected to my home box.. I can recieve a call almost anywhere that i am. lol |
| 08:10.42 | trixter | but there is kinda a blurring between apps and channels and all that.. why app_voicemail wants res_adsi and stuff.. because obviously voicemail needs to know the details of all channels that are connecting instead of some abstraction layer for the channels so voicemail can generically say 'you have a new message' and the channel driver speaks the channel specific protocols |
| 08:10.43 | trixter | but meh |
| 08:10.46 | trixter | I digress |
| 08:10.57 | ThaZZa_Work | trixter: huh? |
| 08:11.19 | *** join/#asterisk cfh (n=luca@host194-20.pool21757.interbusiness.it) |
| 08:11.36 | *** join/#asterisk satlan32 (n=pargit@212.150.142.211) |
| 08:11.41 | trixter | asterisk design blurs core, apps, channels, etc.. app_voicemail.so requires res_adsi.so to load becuase it has some adsi stuff in it, when it really shouldnt, adsi should be for the analog channel driver |
| 08:11.46 | satlan32 | hello |
| 08:12.06 | satlan32 | i need help with a mysql query passed from the dial plan.. |
| 08:12.22 | ThaZZa_Work | trixter: oh okie.. thanks for all your advice and help.. i better finish up work and go home, so i can play some more. ;-) |
| 08:12.24 | trixter | its just a design philophosy, there should be an abstraction layer that isolates apps from knowing anything about channel particulars which makes the apps easier to write, and generally less code - especially potentially duplicated code |
| 08:12.42 | trixter | satlan32: what asterisk version? |
| 08:13.15 | satlan32 | 1.2.7.1 |
| 08:13.41 | satlan32 | http://pastebin.ca/60228 |
| 08:13.42 | trixter | in asterisk-addons iirc (it may be part of the main branch now I dont know) there is a mysql resource so you can do this very easily |
| 08:13.58 | satlan32 | where? |
| 08:14.25 | trixter | what error do you get when that is run? |
| 08:14.44 | trixter | and you do know that 'userpass' is really a username and a password right? they need a space or the tokenizer wont know they are together |
| 08:15.18 | cfh | hi all, i have an asterisk v1.2.4 server in NT mode (with misdn driver and chan_misdn-0.3.1-rc0) |
| 08:15.18 | cfh | <PROTECTED> |
| 08:15.23 | satlan32 | yes...i just removed my values :) |
| 08:15.44 | satlan32 | May 31 11:08:53 VERBOSE[2255] logger.c: -- Executing MYSQL("SIP/504-ff56", "Query resultid 1 INSERT INTO CDR ( FID , Start , End , From , To , Length , File ) VALUES (UUID(), 20060531-110829, 20060531-110853, 504, 0525992504, 21, /tmp/1149062900.96)") in new stack |
| 08:15.44 | satlan32 | May 31 11:08:53 DEBUG[2255] pbx.c: Launching 'MYSQL' |
| 08:15.51 | trixter | ok, you may also want to do a MYSQL(clear ${resultid}) when you are done to be sure that memory is gone |
| 08:16.01 | satlan32 | ok |
| 08:16.18 | trixter | ok, is that all asterisk says? it looked like it worked |
| 08:16.27 | trixter | I dont see where there is a problem yet |
| 08:16.30 | satlan32 | no error after this section in the log, but also no row is entered to the db |
| 08:16.43 | satlan32 | -- Executing MYSQL("SIP/504-ff56", "Query resultid 1 INSERT INTO CDR ( FID , Start , End , From , To , Length , File ) VALUES (UUID(), 20060531-110829, 20060531-110853, 504, 0525992504, 21, /tmp/1149062900.96)") in new stack |
| 08:16.44 | trixter | are you sure that your mysql query is correct? |
| 08:17.26 | satlan32 | i added a line in the phpmyadmin, copied the query and used it, according to the asterisk demands (\) |
| 08:17.56 | satlan32 | i don't see why this is not working... |
| 08:18.45 | trixter | you dont escape () how does the tokenizer know that the ) you type is not the end of the query? |
| 08:19.00 | trixter | lemme clean it up a bit, cause you dont need to escape space so ... hold |
| 08:19.51 | trixter | try that |
| 08:20.00 | trixter | http://pastebin.ca/60229 |
| 08:20.18 | trixter | oops missed one |
| 08:20.27 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-251-57.telkomadsl.co.za) |
| 08:20.40 | trixter | http://pastebin.ca/60230 try that one |
| 08:20.41 | X-Gen | heya freaks |
| 08:22.07 | trixter | satlan32: did that do it? |
| 08:22.16 | satlan32 | i'm checking ... :) |
| 08:22.22 | trixter | ok be back in 5 |
| 08:22.32 | satlan32 | ok |
| 08:27.21 | satlan32 | not working... |
| 08:27.47 | trixter | lemme know if that is still broken, if it is then I am thinknig that its your query, so I would check the data base schema and make sure ... although I didnt really looke at the query itself to see if your args are the same as your variable stuff, nor did I even look at quotes for non numeric input which iwll be required |
| 08:28.06 | trixter | my guess ist hat is quotes then ... |
| 08:28.41 | satlan32 | i tried this also |
| 08:29.02 | trixter | http://pastebin.ca/60232 see how I put in some quotes? |
| 08:29.30 | trixter | you may need to do that for more (I can almost guarantee that some of those need some) just do it the same way for any non numeric only input |
| 08:30.39 | trixter | http://pastebin.ca/60233 oops I put some in the wrong places |
| 08:31.14 | trixter | also check to make sure that your timestamps are the correct format for mysql |
| 08:31.23 | trixter | if infact you are using the datetime object in mysql |
| 08:31.30 | trixter | if its just a text string then quote it |
| 08:31.59 | satlan32 | ok |
| 08:32.00 | trixter | but most if not all of that will need to be quoted |
| 08:32.03 | satlan32 | let me try |
| 08:32.47 | trixter | the query should work everywhere else, perl, php, mysql cli, etc.. if it doesnt work there then it wont work from asterisk |
| 08:33.13 | trixter | so you may try to do a verbose that will expand the variables and all that so you can cut and paste what is getting passed to mysql() and see if that works for you in the mysql cli |
| 08:33.26 | satlan32 | ok |
| 08:33.28 | satlan32 | i'll try |
| 08:33.30 | satlan32 | thanks |
| 08:33.32 | trixter | once you get that working, you can then just escape any quotes, commas and parens |
| 08:33.43 | satlan32 | ok |
| 08:33.48 | *** join/#asterisk SheriF_WorK (n=sherif@212.103.170.135) |
| 08:37.59 | trixter | I think that sirius radio should have irish punk as a station... but no one listens to me |
| 08:38.09 | *** join/#asterisk Diablit0 (n=diablito@80.254.37.205) |
| 08:39.07 | *** join/#asterisk Shoragan (n=shoragan@134.169.175.72) |
| 08:41.52 | *** join/#asterisk key2 (n=ashdown@sd-420.dedibox.fr) |
| 08:41.58 | *** join/#asterisk AltnTab (n=ecs@nrjsoft13.networx-bg.com) |
| 08:57.58 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 08:59.18 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 08:59.37 | RoyK | tag |
| 09:03.39 | trixter | tag you're it |
| 09:04.41 | dlynes_home | that's not very sirius though |
| 09:08.53 | key2 | someone familiar with Queues ? |
| 09:09.14 | dlynes_home | Yeah...they're something like pipes, only different |
| 09:12.01 | RoyK | ~lart dlynes_home |
| 09:12.07 | dlynes_home | heh |
| 09:12.29 | dlynes_home | I guess you didn't like my programmer humor :) |
| 09:15.33 | motu | im using kewlstart, but can't detect the current feed disconnection when remote caller hangs up, i've checked this, its not a polarity switch i'm getting, but a short break in current feed |
| 09:16.08 | RoyK | dlynes_home: peace pipes are nice |
| 09:16.17 | RoyK | key2: just fill them up with people and start calling..... |
| 09:17.34 | trixter | when I liuved in scotland I got familiar with queues |
| 09:18.06 | trixter | had to stand in a queue a few times, at the grocery store, at food places, movie theaters, all sortsa stuff |
| 09:18.36 | RoyK | is there a place without queues in stores? |
| 09:20.16 | *** join/#asterisk ToTo (n=ToTo@81.174.33.2) |
| 09:20.22 | trixter | um lessee |
| 09:20.43 | trixter | walking in there was no queue |
| 09:20.56 | trixter | so yes, the entrance has no queue in the stores :P |
| 09:22.37 | *** join/#asterisk Kis (i=vlad@p5080E671.dip.t-dialin.net) |
| 09:25.33 | RoyK | hehe |
| 09:26.04 | RoyK | trixter: but what's so special about scotland? i never found more queues there than elsewhere..... |
| 09:27.49 | trixter | its a brittish thing |
| 09:28.07 | trixter | like in the hitchikers guide that recently came out ... "Leave this to me I am brittish I know how to queue" |
| 09:28.14 | trixter | or maybe he said english I dont recall now |
| 09:28.23 | trixter | but all england is is southern scotland |
| 09:28.25 | trixter | :D |
| 09:29.26 | trixter | so royk are you going to cluecon.com iun august? or do you have other plans? |
| 09:30.05 | RoyK | where is it? |
| 09:30.09 | trixter | chicago |
| 09:30.13 | trixter | aug 1-3 iirc |
| 09:30.38 | RoyK | are there anything like that in europe? |
| 09:30.47 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 09:31.04 | trixter | asterisk related there will be a thing on modules and one of the authors of asterisk TFOT will be speaking, but there will be stuff on bayonne, sipx, freeswitch, sofaswitch, googletalk, opal, a few other things |
| 09:31.28 | key2 | Someone knows why AddQueueMember(myqueue|SIP/something@somedomain.com) doesn't work if "something" is not specified in sip.conf ? |
| 09:31.44 | trixter | no the closest is astricon in 3 weeks or so but that wont cover the depth of topics that cluecon is going for, since astericon is well, astrisk centric as opposed to just telephony centric |
| 09:33.09 | RoyK | I somehow doubt I'll go... |
| 09:33.52 | trixter | cluecon is talking about doing an asia and european one but that wont be this year |
| 09:34.23 | trixter | last year they had I think 100 people, and that isnt quite big enough yet or something.. I am not sure of what all the decisions are becuase well I am not a decision maker, infact I am not affiliated with it at all :P |
| 09:34.36 | RoyK | key2: i'll see if I can ffind out. just a sec |
| 09:40.46 | key2 | RoyK: would be very nice :) but I really think it's an asterisk pb |
| 09:41.13 | RoyK | key2: does asterisk accept SIP/blah in queues.conf if SIP/blah doesn't exist in sip.conf? |
| 09:41.23 | key2 | no |
| 09:41.24 | key2 | that's the pb |
| 09:41.32 | key2 | RoyK: it takes it |
| 09:41.32 | faber3 | morning, good |
| 09:41.33 | key2 | yeah |
| 09:41.38 | trixter | its not just sip/blah he said sip/blah@somewhere implying somewhere else but that wasnt clarified |
| 09:41.55 | trixter | so I am thinking its a parser issue but I am not sure |
| 09:42.03 | trixter | nor have I looked at it |
| 09:42.27 | RoyK | ops. gotta go. servers to setup... server farm upcoming |
| 09:42.29 | key2 | RoyK: the other thing is that if someone else in asterisk has the same host or something, it would take his info |
| 09:42.50 | faber3 | i am running a CAPI/AVM Fritz (passive) installation with chan_capi.so. it works fine, but sometimes the system freezes hard. so i was wondering if you guys have experiences with chan_misdn, and if that works stable ? |
| 09:42.54 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 09:43.16 | key2 | trixter: do you have an idea ? |
| 09:43.51 | trixter | nope, as I said I havent looked at it |
| 09:43.58 | key2 | :( |
| 09:44.53 | *** join/#asterisk azeteg (n=azeteg@c115.brewhouse.se) |
| 09:45.41 | azeteg | does latest asterisk have t38 suppport? |
| 09:45.52 | key2 | t38 |
| 09:45.53 | key2 | ? |
| 09:47.38 | faber3 | is anyone running a BRI-ISDN gateway (outbound ISDN, inbound Asterisk/VO-IP) together with a passive Fritz/AVM? Whats your setup? |
| 09:51.22 | SheriF_WorK | anyone played with rxgain txgain ? |
| 09:52.15 | chapeaurouge | if i have a FXO/FXS card, and plug a fax in it, getting out via my own ISDN-BRI line, and receiving it thru ISDN as well, I shouldn't have any issues correct? Fax is only an issue if I go onto IP? |
| 09:52.19 | *** join/#asterisk groogs (n=greg@d221-73-237.commercial.cgocable.net) |
| 09:53.53 | puzzled | morning |
| 10:00.19 | *** join/#asterisk Thazza (n=me@229.9.233.220.exetel.com.au) |
| 10:07.14 | motu | is there any way to access contents in sip messages, eg sip info from a client? |
| 10:08.04 | trixter | like sip debug? |
| 10:08.08 | motu | ie, access sip info sent from a client |
| 10:08.20 | motu | yea, something like that |
| 10:08.38 | trixter | no there is no way |
| 10:08.54 | motu | hmm, theres no plugin that can do it? |
| 10:09.14 | trixter | um I thought we agreed that sip debug was what you wanted |
| 10:09.25 | trixter | what do you want that it doesnt do? |
| 10:09.32 | azeteg | how can I get asterisk / txfax+rxfax to work with Cisco ATA-186 |
| 10:09.33 | azeteg | ? |
| 10:09.40 | azeteg | I get error message when trying |
| 10:10.05 | azeteg | rtp.c: Unknown RTP codec 100 received |
| 10:10.13 | trixter | you probably want that to be 101 |
| 10:10.28 | trixter | > 96 isnt defined by the spec and telephony events are mapped in asterisk generally to 101 |
| 10:10.30 | motu | i have a sip client that uses sip info to access pbx features in another ip pbx, i want asterisk to be compatible |
| 10:11.05 | trixter | unless its trying something totally different like a codec that isnt on the 'official' list which would explain why its > 96 |
| 10:11.24 | trixter | motu: did you enable sip-info in the auth declaration? |
| 10:11.29 | trixter | where you defined that account |
| 10:11.42 | *** join/#asterisk downunder33 (n=robert@219.95.154.69) |
| 10:12.10 | motu | u mean dtmf via sip info? |
| 10:12.40 | trixter | isnt that what you said you wanted? |
| 10:12.55 | chapeaurouge | I get "No translator path exists for channel type Zap (native 76) to 256" errors, when trying to pass calls from Zap to SIP and vice-versa... am i missing some configuration values? |
| 10:13.04 | motu | no, this client send data in xml format via sip info |
| 10:13.17 | motu | so i need to access the whole info message |
| 10:15.59 | *** join/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr) |
| 10:18.46 | tparcina | group, i don't understand how hint-ing works. i have grandstream gxp2000 and i don't know how, from grandstream on 125 extension, hint 137 extension. |
| 10:19.01 | chapeaurouge | would that because I would need to only allow alaw, and not g729? If it knows it can't do it, why doesn't it switch over? |
| 10:19.13 | tparcina | it could be that i don't understand what hint does att all... |
| 10:21.45 | tparcina | do i have to dial (or do anything else) to se the status of 137 extension, or the phone should allways light me status of that extension? |
| 10:23.07 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 10:25.33 | *** join/#asterisk RoyK (n=roy@213.160.242.91) |
| 10:26.32 | azeteg | anyone here that uses cisco ata-186? |
| 10:27.31 | azeteg | I'm wondering howto set the AudioMode parameter to get Asterisk to receive faxes from the ATA |
| 10:29.35 | motu | is there some way i can pipe the sip debug output into stdout? |
| 10:30.18 | motu | or even better, to access the debug output from within asterisk |
| 10:30.41 | *** part/#asterisk jaike (n=a@203.131.137.76) |
| 10:32.02 | RoyK | <PROTECTED> |
| 10:32.25 | trixter | <PROTECTED> |
| 10:32.27 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 10:34.12 | *** join/#asterisk mr_horsepower (n=igor@82.102.1.42) |
| 10:39.14 | mut | anyone here any good at cracking? i'm trying to unpack a prog and having some difficulties, could use some guidance |
| 10:44.22 | RoyK | unpack? doesn't seem to have anything to do with neither asterisk nor cracking... |
| 10:44.58 | chapeaurouge | exten => _X.,1,SetVar(SIP_CODEC=alaw) |
| 10:44.59 | chapeaurouge | exten => _X.,n,Macro(internalext,101,SIP/plcm300-01) |
| 10:45.07 | chapeaurouge | would that effectively force the codec to be used? |
| 10:45.15 | mut | has everything to do with cracking.. |
| 10:45.17 | chapeaurouge | im having issues to get call Zap through to SIP... |
| 10:45.18 | mut | but with asterisk.. no |
| 10:45.19 | Alex | It's stupid to hilight *alex* in a channel with 'internalext' being mentioned quite a lot :) |
| 10:45.24 | mut | you are correct |
| 10:45.46 | chapeaurouge | mut, I ? |
| 10:45.56 | mut | no robl^ |
| 10:45.59 | chapeaurouge | ok |
| 10:45.59 | mut | RoyK rather |
| 10:46.16 | tparcina | azeteg, you should use AudioMode: 0x00140014 on cisco ATA |
| 10:46.30 | tparcina | azteg, that one works for me |
| 10:47.48 | mut | ditto |
| 10:49.36 | RoyK | mut: is it encrypted? |
| 10:49.45 | RoyK | mut: tar xzf ..... |
| 10:50.32 | mut | heh |
| 10:59.03 | azeteg | tparcina: I have that |
| 10:59.26 | azeteg | I still get that rtp.c: Unknown RTP codec 100 received |
| 10:59.36 | azeteg | is it trying to set some weird codec? |
| 11:01.03 | motu | how do i access dtmf tones during a call between 2 zap devices? |
| 11:01.36 | azeteg | tparcina: you only have txfax/rxfax? |
| 11:01.40 | azeteg | or some other thing? |
| 11:02.29 | *** join/#asterisk LokeshIndian (n=lokesh_k@estrela.nortenet.pt) |
| 11:03.06 | tparcina | can somebody chack my sip.conf and extensions.conf and tell me why my hint-ing doesn't work? - http://pastebin.ca/60255 |
| 11:03.52 | tparcina | azteg, no, i have used that one with hardware fax machine |
| 11:07.58 | azeteg | hmmm |
| 11:08.43 | tparcina | can anybody send me his working sip and extensions.conf file for hint on grandsteam gpx2000, please |
| 11:18.14 | RoyK | tparcina: rtfm :) |
| 11:20.04 | tparcina | RoyK, i would, please send it to me! :) |
| 11:20.16 | tparcina | i think i know where could be the problem |
| 11:20.21 | RoyK | tparcina: don't have it. sorry. don't know about hints at all |
| 11:21.01 | azeteg | tparcina: I have only got ATA's in this setup for faxing |
| 11:21.33 | tparcina | can anybody tell me how i need to define lines on grandstream gpx2000 for hint-ing? because i'm preaty sure that is all fine configured in sip and extensions conf files |
| 11:21.52 | RoyK | azeteg: t.38? |
| 11:22.13 | RoyK | ~foip |
| 11:22.18 | jbot | [foip] Fax over IP. This requires funtionality standardised by T.38, realtime fax over IP, or T.37, store-and-forward via email. See http://soft-switch.org/foip.html for more detailed info about the subject |
| 11:22.46 | tparcina | aztek, sory but i can't help you. i have used cisco ata 186 for pluging hardware fax machine and i have used AudioMode that i have sent you. maybe somady else know more about that one |
| 11:24.04 | tparcina | RoyK, i now that you don't have hint-ing, you are seccond league asterisk player :PP |
| 11:29.29 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 11:31.04 | *** part/#asterisk rleyba (n=root@60-241-132-21.tpgi.com.au) |
| 11:32.08 | *** join/#asterisk coppice (n=chatzill@95.206.17.210.dyn.pacific.net.hk) |
| 11:33.43 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 11:33.44 | RoyK | tparcina: bah |
| 11:34.01 | tparcina | i know why hinting doesn't work reason is here - http://pastebin.ca/60257 so how to make gpx2000 to register as hinter? |
| 11:34.26 | *** join/#asterisk Egonis (n=Egonis@207.245.14.10) |
| 11:34.56 | Egonis | I have a TDM400 with two FXO Cards, I am using KB1 Echo Cancel, is there a utility I should use? Where would I find a howto for morons? |
| 11:35.12 | Strom_C | a utility for what, exactly? |
| 11:35.28 | azeteg | RoyK: t38 works? |
| 11:35.40 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
| 11:35.41 | Egonis | echo cancel, i.e. adjusting rx/tx gains, etc. |
| 11:35.51 | azeteg | RoyK: I heard it was like in pre-early-alpha stage |
| 11:36.33 | Strom_C | Egonis: you can use fxotune, or you can tune it manually |
| 11:36.52 | Egonis | fxotune is a utility that comes with zaptel? |
| 11:36.56 | Strom_C | yes |
| 11:37.04 | Strom_C | i prefer to do the tuning manually |
| 11:37.22 | Egonis | how do I use it? I found fxotune.h in my zaptel includes, but no binaries |
| 11:37.39 | Strom_C | /usr/src/zaptel/fxotune |
| 11:38.08 | uwe | um, sometimes some sip phones here stop reciving calls, and they reply as if they were busy ... restarting asterisk solves the problem ... doesn anyone know why this could happen? |
| 11:38.27 | Strom_C | uwe: what kind of sip phones? |
| 11:38.38 | uwe | cisco |
| 11:38.49 | azeteg | model? |
| 11:39.02 | Strom_C | which phone, which firmware version, and which version of asterisk? |
| 11:39.02 | azeteg | what does sip debug tell you? |
| 11:39.13 | uwe | 7940 |
| 11:39.47 | azeteg | I didn't experience that |
| 11:39.51 | Egonis | Strom_C: Where do I get fxotune.conf? it's asking for /etc/fxotune.conf |
| 11:39.54 | azeteg | might be a network problem? |
| 11:40.02 | RoyK | azeteg: some of t might work |
| 11:40.03 | Strom_C | uwe: which firmware version, and which version of asterisk |
| 11:40.12 | azeteg | RoyK: I'm just installing |
| 11:40.17 | uwe | Asterisk 1.0.10-BRIstuffed-0.2.0-RC8q |
| 11:40.22 | Strom_C | Egonis: it should make fxotune.conf - which version of asterisk are you using? |
| 11:40.28 | azeteg | 1.0.10? |
| 11:40.28 | RoyK | uwe: ouch |
| 11:40.34 | Strom_C | uwe: 1.0.10 is ANCIENT |
| 11:40.43 | azeteg | thats like from 1934 or so |
| 11:40.50 | Strom_C | no, older |
| 11:40.50 | Egonis | Strom_C 1.2.7.1 |
| 11:40.51 | uwe | :) |
| 11:40.56 | RoyK | azeteg: 1.2.8 |
| 11:41.10 | Strom_C | thats the version that Bell just barely beat to the patent office by 1.3 hours |
| 11:41.31 | Egonis | ***RoyK: Whaaaaat? :) |
| 11:41.31 | uwe | i suppose something like 1010 AD ;) |
| 11:42.30 | Strom_C | "Watson! Come here! I've got jitter!" |
| 11:42.58 | Egonis | So I'm guessing I should upgrade to 1.2.8? |
| 11:43.05 | Egonis | with regards to fxotune? |
| 11:43.18 | azeteg | RoyK: 1.2.8? |
| 11:43.23 | azeteg | I have 1.2.7 installed |
| 11:43.29 | Strom_C | Egonis: fxotune without any arguments should spit out instructions |
| 11:43.41 | Egonis | Strom_C: it did, and so I specified -s |
| 11:44.00 | Egonis | Strom_C: -i specifies a specific channel and a number to dial (which I don't understand) |
| 11:44.16 | Egonis | Strom_C: Does it create an fxotune.conf as per the -i instruction, and then -s sets it to the device? |
| 11:44.30 | Strom_C | oh christ, you're going to make me ssh into my box |
| 11:44.32 | Strom_C | one moment |
| 11:44.32 | coppice | RoyK: is the jitter buffer making it into 1.4? |
| 11:45.02 | azeteg | RoyK: I have a problem applying the t38 patch |
| 11:45.16 | Strom_C | Egonis: ok |
| 11:45.25 | Strom_C | just run fxotune -i 6 |
| 11:46.00 | Strom_C | then AFTER it tunes your card, run fxotune -s |
| 11:46.00 | Strom_C | the number to dial is just to break dial tone |
| 11:46.09 | RoyK | coppice: it might seem so |
| 11:46.27 | RoyK | azeteg: I think there's a svn tree with it, right, coppice? |
| 11:46.58 | Egonis | Strom_C: Ah, cool! Thanks.. why 6? just out of curiosity |
| 11:47.03 | coppice | the SVN tree containing T.38 is seriously broken. the T.38 is fine, but nothing else in that branch seems to work |
| 11:47.18 | Strom_C | Egonis: because I like the number 6 |
| 11:47.23 | Egonis | Strom_C: Good answer |
| 11:47.46 | tzanger | coppice: ugh |
| 11:48.46 | azeteg | RoyK: what would you recommend to get proper faxing with asterisk then? |
| 11:48.51 | coppice | tzanger: I think its part of a plot to keep anything useful out of 1.4 |
| 11:49.00 | RoyK | azeteg: patience |
| 11:49.17 | tzanger | coppice: it sure feels that way sometimes |
| 11:49.36 | tzanger | I don't know enough of the details to intelligently comment, but I really don't know what the new loader brought us |
| 11:49.47 | coppice | bugger all |
| 11:49.58 | tzanger | yet that was done in -trunk, whereas jitter buffer, t38... all relegated to branches |
| 11:50.21 | coppice | like all the changes to make PLC fit more smoothly, omitted the jitter buffer changes that actualyl make it of any use :-) |
| 11:50.36 | tzanger | branches are good if they're used properly, but I don't think we use them properly |
| 11:50.58 | coppice | the branches are graveyards for good ideas |
| 11:51.07 | tzanger | IMO (for what it's worth, which is not much admittedly) branches should be used to bring new functionality in and make it mostly clean, then merge it into trunk for the heavy beatings |
| 11:51.36 | coppice | in the real world nothing gets tested until it is in the trunk |
| 11:51.38 | tzanger | menuselect was done properly as a branch, so I know this is at least in theory possible |
| 11:51.42 | tzanger | coppice: exactly |
| 11:52.01 | tzanger | which is why the branch should be used as a proof of concept and "first pass" testing, then let us mere mortals beat it up in trunk |
| 11:52.56 | tzanger | was there a clean rtp jitter buffer branch for would-be trunk/ |
| 11:54.38 | coppice | half the stuff is clean the day it is submitted to bugs.digium.com. The rot occurs after that |
| 11:54.48 | tzanger | I understand |
| 11:55.11 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
| 11:55.26 | RoyK | I wonder why people don't spend more time with openpbx |
| 11:55.32 | RoyK | or freeswitch |
| 11:55.38 | tzanger | coppice: were you able to collect on that g726 bounty? |
| 11:55.48 | tzanger | RoyK: the same reason we don't spend more time wiht yate |
| 11:55.55 | coppice | his problem hasn't been fixed yet |
| 11:56.07 | *** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.34) |
| 11:56.38 | tzanger | oh -- I thought his last post was more showing he was looking for ways to make it sound odd |
| 11:56.44 | RoyK | tzanger: working on opbx != sleeping with diana |
| 11:56.52 | tzanger | ? |
| 11:57.17 | tzanger | I was just implying that the reason people don't work on opbx is the same reason they don't work with yate -- the inertia just isn't there for most of us |
| 11:57.23 | tzanger | it has nothing to do with diana |
| 11:58.46 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 11:58.55 | chapeaurouge | http://pastebin.ca/60262 |
| 11:58.59 | *** join/#asterisk MGSsancho (n=user@adsl-67-126-140-26.dsl.irvnca.pacbell.net) |
| 11:58.59 | chapeaurouge | someone could help me on that? |
| 11:59.01 | X-Gen | *yawn* |
| 11:59.20 | chapeaurouge | trying to make outbound calls using ISSN-BRI line |
| 11:59.25 | chapeaurouge | with an IP hardphone |
| 11:59.32 | chapeaurouge | but i seem to have issues... |
| 11:59.49 | chapeaurouge | first was bc of codec. then somehow busy... |
| 12:00.00 | RoyK | chapeaurouge: isdn bri zaptel card? didn't know those even existed :) |
| 12:00.09 | chapeaurouge | yea... junghanns.net |
| 12:00.16 | chapeaurouge | quadBRI.. with BRIstuff drivers |
| 12:00.52 | [TK]D-Fender | chapeaurouge : Your phone is trying to use G.729 and it looks like you have no licenses for transcoding |
| 12:00.52 | RoyK | ah. that one... |
| 12:01.17 | chapeaurouge | i do... [TK]D-Fender ok.. my polycom phones come with a license i think |
| 12:01.21 | chapeaurouge | asterisk server needs one as well? |
| 12:01.22 | [TK]D-Fender | May 31 13:56:00 WARNING[8410]: channel.c:2541 ast_request: No translator path exists for channel type Zap (native 76) to 256 |
| 12:01.41 | [TK]D-Fender | chapeaurouge : Yes, but * has to do the translation to a Zap compatible format. |
| 12:01.45 | mr_horsepower | chapeaurouge: use g711 |
| 12:01.47 | chapeaurouge | [TK]D-Fender, any idea about the second message, when using alaw? |
| 12:02.08 | [TK]D-Fender | chapeaurouge : That's what you should be using : G711a (ALAW) |
| 12:02.09 | mr_horsepower | chapeaurouge: are you portuguese? |
| 12:02.21 | chapeaurouge | alaw is 90k bandwitdth... too much. |
| 12:02.28 | chapeaurouge | g729 is 30k |
| 12:02.32 | chapeaurouge | ran some tests here |
| 12:02.39 | [TK]D-Fender | chapeaurouge : Then go buy some licenses |
| 12:02.52 | chapeaurouge | ok.. any best sellers? |
| 12:02.57 | mr_horsepower | digium? |
| 12:02.58 | mr_horsepower | :o |
| 12:02.59 | tzanger | actually ulaw/alaw is closer to 80kbps on the wire with all overheads unless you're doing something different than the rest of us |
| 12:03.01 | chapeaurouge | lol ok |
| 12:03.13 | chapeaurouge | mine was 90k with iptraf |
| 12:03.17 | chapeaurouge | dunno |
| 12:03.22 | mr_horsepower | chapeaurouge: are you portuguese? |
| 12:03.27 | chapeaurouge | mr_horsepower, no why |
| 12:03.29 | [TK]D-Fender | chapeaurouge : http://www.digium.com/en/products/voice/g729codec.php |
| 12:03.33 | chapeaurouge | thx |
| 12:03.36 | mr_horsepower | i have a friend with that nick. |
| 12:03.50 | chapeaurouge | must not be registered with it then ;) |
| 12:04.32 | chapeaurouge | why isn't my SIP_CODEC variable taking effect when g729 is in sip.conf? |
| 12:04.34 | mr_horsepower | not in freenode. |
| 12:04.37 | chapeaurouge | and it does when it's not? |
| 12:05.14 | Ironhand | hello, I'm trying to do the following: when someone dials an extension, have asterisk dial a fixed extension in the background, but do not connect this outgoing call to the incoming one yet... send a string of DTMF tones into the outgoing call, wait for a bit, and only *then* connect the incoming call to the outgoing one. How would I do something like that? |
| 12:05.52 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 12:06.12 | [TK]D-Fender | chapeaurouge : Just force it in sip.conf and on the phone. |
| 12:06.19 | chapeaurouge | i dont want to do that. |
| 12:06.29 | chapeaurouge | bc my outgoing calls are using g729 |
| 12:07.39 | tzanger | chapeaurouge: then use separate sip peers, dial out through one that only has g729 enabled |
| 12:07.48 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 12:07.53 | chapeaurouge | ok |
| 12:07.58 | chapeaurouge | http://pastebin.ca/60264 |
| 12:08.01 | chapeaurouge | hehe .. :) |
| 12:08.04 | tzanger | I use that for my IAX2 faxing |
| 12:08.07 | chapeaurouge | got that pb with zap now |
| 12:08.12 | chapeaurouge | tzanger, ok |
| 12:08.21 | tzanger | the iax2 user/peer for faxing only has ulaw enabled. the iax2 user/peer ofr voice has only gsm enabled |
| 12:08.32 | tzanger | perhaps not the most elegant solution, but it has worked for over 18 months |
| 12:09.55 | tzanger | anyway... back in a bit |
| 12:11.35 | tparcina | yeesssssss, ihave make it work :)) here are complete instructions how to enable hinting on grandstream gpx2000 - http://www.jackenhack.com/blog/archives/2005/11/22/setting-up-subscribenotify-blf-in-asteriskhome-for-grandstream-gxp-2000-phones/ |
| 12:11.54 | Strom_C | tinyurl.com is your friend |
| 12:12.09 | Ironhand | ok, let me try to reduce my question into two simple basic questions: 1) how do I mute the sound from the outgoing call towards the dialing user for a certain period; 2) how do I send DTMF tones into the outgoing call rather than to the calling party? |
| 12:13.25 | tparcina | Storm_C, thank you ;)) here is the link http://tinyurl.com/dpod8 |
| 12:13.45 | rkr245 | tzanger,Iam using Sip Express Router as sip proxy and its woking fine and its also have the media proxy it can with stand rtp&rtcp streams up to 80 at a time and now i want to use asterisk as gateway to pstn .. can you suggest any wikis which are providing some sample configs to asterisk |
| 12:14.17 | chapeaurouge | thx all |
| 12:14.44 | rkr245 | and here i have a doubt how the rtp stream handled by asterisk? |
| 12:14.47 | Ironhand | or for that matter, how do I let asterisk create an outgoing call that is initially not linked to any incoming call, and then later "merge" the incoming call with it? |
| 12:16.46 | *** join/#asterisk fjean (n=fjean@201.29.140.206) |
| 12:17.18 | fjean | good morning |
| 12:17.42 | *** part/#asterisk Egonis (n=Egonis@207.245.14.10) |
| 12:18.12 | motu | how do i add two numbers together in the dial plan? |
| 12:18.28 | fjean | motu - like a prefix ? |
| 12:18.37 | motu | yea |
| 12:18.39 | Strom_C | or are you talking actual addition |
| 12:18.51 | motu | both actually |
| 12:19.07 | fjean | motu: when using Dial ? or anywhere else ? |
| 12:19.11 | Strom_C | for mathematical addition, use the math function |
| 12:19.26 | Strom_C | to make an amalgam of numbers, use variables next to each other |
| 12:19.37 | motu | not just dial |
| 12:19.37 | motu | ok |
| 12:19.39 | motu | thnx |
| 12:21.15 | fjean | it's funny, I get "Allow unknown access = Yes" on sip show settings...but allowguest is commented out in sip.conf... |
| 12:21.28 | fjean | are they related ? |
| 12:23.08 | Eric-xx | <PROTECTED> |
| 12:23.39 | key2 | In asterisk Realtime, if I do AddQueueMember, does it add it to the mysql database ? |
| 12:23.43 | key2 | or it keeps it local |
| 12:24.02 | azeteg | $50 hah |
| 12:26.56 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:32.28 | *** join/#asterisk aze (n=aze@ACayenne-101-1-11-194.w81-248.abo.wanadoo.fr) |
| 12:35.23 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 12:37.54 | motu | can i catch dtmf tones while in dial? |
| 12:38.35 | [TK]D-Fender | motu : You can only capture DTMF while it is ringing, and only for the purpose of aborting the dial. |
| 12:39.07 | *** join/#asterisk ToTo (n=ToTo@81.174.33.2) |
| 12:40.16 | motu | so its impossible to make a call transfer/handover while in a conversation? |
| 12:40.29 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 12:40.33 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 12:40.40 | tzanger | what is the polycom-acd-functions branch? |
| 12:41.19 | fjean | how do I confirm 100% that the server's SIP port is being blocked by the ISP ? |
| 12:41.19 | tzanger | the svn repo shows no new docs or anything, as was expected :-) |
| 12:41.44 | tzanger | fjean: shutdown the server, put netcat listening there and try to hit it with netcat on the other side |
| 12:42.01 | Ariel_ | Eric-xx, I can help you |
| 12:42.01 | Dr-Linux | fjean: do like me, i'm using 2 different ports for SIP |
| 12:42.09 | fjean | tzanger: i can;t shutdown the server.. |
| 12:42.20 | Dr-Linux | in pakistan all ISP's have blocked 5060 port |
| 12:42.37 | tzanger | fjean: then tcpdump and watch |
| 12:42.40 | fjean | dr-linux: that's an idea...you specify them in sip.,conf ? |
| 12:43.12 | fjean | tzanger: what about your users, that gave you a lot of trouble ? |
| 12:43.29 | tzanger | fjean: my ISP doesn't block me. and if your ISP was blocking you, you owuldn't have any users now would you? |
| 12:43.32 | Dr-Linux | fjean: nope, use any port on client site like 8080 and in your server just redirect it to 5060 |
| 12:43.38 | Dr-Linux | both will work |
| 12:43.49 | Ariel_ | good morning everyone |
| 12:43.54 | Dr-Linux | fjean: you also can change your sip port in sip.conf |
| 12:44.10 | Dr-Linux | but 2 ports for sip is a good idea |
| 12:44.19 | fjean | tzanger: there are several different ISPs on the country..the server is in the states |
| 12:44.37 | Dr-Linux | fjean: same here :) |
| 12:44.39 | fjean | tzanger: but we are in brazil |
| 12:44.41 | Dr-Linux | fjean: where from you? |
| 12:45.43 | Dr-Linux | fjean: just simple use 8080 port in barazil clients and redirect this port to 5060 UDP at your US server |
| 12:45.58 | *** join/#asterisk parag7732 (n=root@83.110.255.112) |
| 12:46.03 | *** part/#asterisk parag7732 (n=root@83.110.255.112) |
| 12:46.10 | *** join/#asterisk rhowe (i=rhowe@87.127.14.179) |
| 12:46.13 | fjean | dr-linux: great idea, I'll see that |
| 12:46.48 | [TK]D-Fender | Ariel_ : I've seen the call-flow for his request. I'll be shocked if you take it on.... |
| 12:46.49 | Dr-Linux | fjean: chaning sip port is not a good idea, bcoz some of your state users maybe already using it, so don't bother SIP port |
| 12:47.05 | Dr-Linux | just use iptables one line and enjoy |
| 12:47.16 | Ariel_ | [TK]D-Fender, ok reading it right now. |
| 12:47.53 | fjean | dr-linux: sure |
| 12:50.04 | *** join/#asterisk HolyGod (i=nobody@got.securebinary.com) |
| 12:51.43 | fjean | doing telnet to port 5060 would give me a hint , right ? |
| 12:51.43 | Nugget | telnet is eeeeeeevil! |
| 12:53.11 | RoyK | fjean: hehe |
| 12:53.21 | RoyK | fjean: telnet uses tcp, sip uses udp..... |
| 12:53.25 | RoyK | nmap -sU -p 5060 |
| 12:53.34 | fjean | royk, right |
| 12:53.40 | fjean | thanks |
| 12:53.42 | fjean | hehe |
| 12:53.45 | CoaxD | its all about that pesky udp. |
| 12:53.55 | RoyK | CoaxD: would you like rtp over tcp? |
| 12:54.07 | CoaxD | RoyK: Over a LAN, sure! *lol* |
| 12:54.16 | RoyK | be my guest... |
| 12:54.33 | CoaxD | RoyK: Over a WAN of any kind? Hell no. |
| 12:54.40 | CoaxD | RoyK: Too much overhead. |
| 12:54.47 | MrChimpy | holy crap. i dial from my mobile to our inbound asterisk box, it routes over IAX2 to outbound asterisk box, it dials my desk phone via E1 line. caller ID works! that never happens! |
| 12:55.19 | CoaxD | MrChimpy: hahaha. Sounds like a slightly overcomplexified solution. |
| 12:55.54 | MrChimpy | nah, not really. it's the architecture for a large IVR system |
| 12:56.01 | CoaxD | MrChimpy: Ok |
| 12:56.09 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
| 12:57.21 | RoyK | CoaxD: the overhead isn't important, but the latency is |
| 12:57.29 | RoyK | CoaxD: retransmits and so on on rtp? |
| 12:57.40 | hwt | what does this mean? May 31 14:57:31 NOTICE[7771]: sched.c:234 ast_sched_add_variable: Scheduled event in 0 ms? |
| 12:57.49 | CoaxD | RoyK: Under screwed up network conditions.. yea. |
| 12:57.57 | hwt | i'm only doing a setlanguage. |
| 12:58.04 | hwt | um, a set() actually. |
| 12:59.02 | RoyK | CoaxD: under _normal_ network connections, not screwed |
| 12:59.22 | RoyK | CoaxD: anyway, you don't want retransmits for rtp, you want plc |
| 12:59.26 | RoyK | and jitterbuffers |
| 12:59.57 | *** join/#asterisk UlbabraB (n=UlbabraB@host-84-222-44-6.cust-adsl.tiscali.it) |
| 13:03.11 | SheriF_WorK | i have strang problem. the sip works fine from softphones but the voice qualti is really bad in hardphones "GipTel" what migh be the problem ? |
| 13:04.31 | hwt | SheriF_WorK: what codec are you using? |
| 13:06.11 | SheriF_WorK | hwt: ulaw |
| 13:06.51 | SheriF_WorK | 20 mx RTP |
| 13:06.57 | SheriF_WorK | 20 ms RTP |
| 13:08.54 | hwt | SheriF_WorK: are you sure the hardphones are actually using ulaw? |
| 13:09.00 | hwt | SheriF_WorK: not gsm or anything. |
| 13:10.35 | SheriF_WorK | hwt: yes i just made sure ;-) it's the 1st priority on the phone |
| 13:10.53 | SheriF_WorK | hwt: in softphones " x-lite " and sjphone it's perfect ! |
| 13:11.39 | MikeJ[Laptop] | SheriF_WorK, if you actually watch the verbose on a call.. is it actually using it.. |
| 13:11.47 | MikeJ[Laptop] | priorities are not allways used. |
| 13:12.07 | MikeJ[Laptop] | the only way to be sure is to force only one codec on the peer |
| 13:12.55 | SheriF_WorK | MikeJ[Laptop]: so how do i make sure ? |
| 13:13.34 | hwt | SheriF_WorK: disallow=all, allow=ulaw |
| 13:13.41 | hwt | SheriF_WorK: in the clients. |
| 13:14.03 | SheriF_WorK | hwt: i done that in the main SIP.conf section . |
| 13:14.19 | SheriF_WorK | but how do i verbos the call to make sure of what codec is used? |
| 13:14.23 | MikeJ[Laptop] | well.. that's one way to be sure ;) |
| 13:14.39 | MikeJ[Laptop] | if you have disallow=all, allow=ulaw then it's using ulaw |
| 13:14.58 | SheriF_WorK | MikeJ[Laptop]: i can disable all the the codecs in side the phone ;-) i don't wann mess asterisk at the moment.. :P |
| 13:15.01 | SheriF_WorK | but ok let me try |
| 13:15.28 | *** part/#asterisk dongs (n=HPUX@h193012.ppp.asahi-net.or.jp) |
| 13:15.28 | MikeJ[Laptop] | disabling the codecs in the phone would work too.. |
| 13:20.38 | *** join/#asterisk skeffling (n=andrew@andrew.1ec.aaisp.net.uk) |
| 13:22.12 | sevard | Wow, this is a weird problem... |
| 13:22.33 | sevard | has anyone ever added a voicemail box in voicemail.conf but yet it doesn't work? |
| 13:22.46 | sevard | it might be too early in the morning but i'm failing atm |
| 13:23.59 | *** join/#asterisk Dovid (n=none@bzq-84-108-212-3.cablep.bezeqint.net) |
| 13:24.11 | Dovid | Hi, anyone know how to set ztdummy in asterisk@home ? |
| 13:24.59 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:25.12 | tparcina | grup, question. calls thrue zap interface (e1 - 10 channels), are they loged anywhere except cdr (and database)? what i would like to do is to check what was maximum usage of zap channels in the past. |
| 13:25.58 | tparcina | it would be perdect if it can connect to my existing mysql database and drow a graph of usage of zap lines... anybody know for something similar? |
| 13:26.04 | tdi | hej |
| 13:27.27 | RoyK | hej hej |
| 13:27.51 | sevard | hej hej hej |
| 13:28.54 | RoyK | hemskt mycket hej |
| 13:31.53 | Dovid | Anyone know ast@home here ? |
| 13:32.59 | tparcina | SheriF_WorK, when you establish call do sip show channels - it will show you all active chennals, then do sip show channel channel_id ant it will show you informations about that call (including codec) |
| 13:33.28 | [TK]D-Fender | Dovid : Please read the channel topic |
| 13:34.15 | Ariel_ | Dovid, ztdummy comes pre-done in asterisk@home setup |
| 13:34.26 | *** part/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 13:35.41 | SheriF_WorK | thx tparcina |
| 13:35.54 | SheriF_WorK | but the sound qualti still the same even with ulaw only |
| 13:36.03 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:36.03 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:36.36 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 13:38.47 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 13:42.00 | tparcina | so guys, how do you check what is your maximum usage of zap lines in past few months? |
| 13:44.41 | [TK]D-Fender | tparcina : use your CDR's to calculate overlap |
| 13:44.56 | sevard | tparcina: I suppose a couple of months ago you put in a system that calculates usage of zap lines |
| 13:45.28 | *** join/#asterisk esculapio__ (n=ESCulapi@200.88.44.66) |
| 13:45.31 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 13:45.33 | Dr-Linux | -- Remote UNIX connection |
| 13:45.33 | Dr-Linux | <PROTECTED> |
| 13:45.33 | Dr-Linux | <PROTECTED> |
| 13:45.35 | tparcina | fender, do you know for any working php script? |
| 13:45.53 | [TK]D-Fender | tparcina : ?! |
| 13:45.55 | Dr-Linux | why i'm getting this on CLI after every 2 seconds? |
| 13:45.56 | tparcina | swvard, i have cdr in mysql database. that should be enough |
| 13:46.00 | MikeJ[Laptop] | Dr-Linux, manager connections |
| 13:46.43 | Dr-Linux | MikeJ[Laptop]: thanks |
| 13:46.54 | tparcina | Fender, i have all records in mysql database. for php experts, it shouldn't be hard to pull out data and make graph and show it on web page... |
| 13:47.59 | [TK]D-Fender | tparcina : That would not be me. |
| 13:48.42 | tparcina | Fender, me nither :) or beter :( anyway, i'm searching has anybody make something similar |
| 13:48.51 | *** join/#asterisk schuylerdigium (n=schuyler@gateway.digium.com) |
| 13:49.38 | tparcina | why RoyK isn't returning on chat... i made joke about him and i don't know has he take it sirisuly... :( |
| 13:50.39 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 13:50.48 | *** join/#asterisk mercestes (n=merceste@69.15.174.114) |
| 13:50.55 | buzzyd | Hi All, I am getting a "Forbidden - wrong password on authentication for INVITE to" on a snom and mp102 on the snom I think it a phone setting anyone got an idea? |
| 13:51.15 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 13:51.50 | *** join/#asterisk NouRsOnSFR (n=nounours@stardust.noc.frontier.fr) |
| 13:51.50 | NouRsOnSFR | Hio |
| 13:52.13 | *** join/#asterisk jahani (n=k@41.250.37.232) |
| 13:52.29 | NouRsOnSFR | do you have a test for reboot GXP2000 witch asterisk |
| 13:52.31 | NouRsOnSFR | . |
| 13:52.34 | NouRsOnSFR | ? |
| 13:55.28 | Splat | anyone able to tell me if I need to install libpri is I'm setting up asterisk from source and only going to be using a BRI interface (an Eicon Diva Server BRI 2M) and IAX and SIP connections? |
| 13:56.09 | [TK]D-Fender | SplasPood : I believe so. |
| 13:56.28 | [TK]D-Fender | SplasPood : Thats where the ISDN signalling work is done IIRC |
| 13:57.07 | Splat | ok, thanks |
| 13:58.35 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 13:59.44 | *** part/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr) |
| 14:01.07 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:02.23 | *** join/#asterisk javar (n=javar@Dynamic-IP-cr20011868204.cable.net.co) |
| 14:02.32 | Juggie | ~centosbug |
| 14:02.33 | jbot | i heard centosbug is a problem with the latest Centos kernel (4.2 and 4.3). To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. This is part of the 'kernel-devel' package. |
| 14:02.37 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 14:04.32 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 14:08.15 | key2 | pif ? |
| 14:09.12 | pif | yes? |
| 14:10.57 | NouRsOnSFR | do you have test for reboot gxp2000 with asterisk ? |
| 14:12.50 | *** join/#asterisk cfh (n=luca@82.193.23.6) |
| 14:13.22 | *** join/#asterisk Tagor (n=Tagor@s55928c6d.adsl.wanadoo.nl) |
| 14:13.24 | Tagor | Hi |
| 14:13.29 | Tagor | I've the following problem: |
| 14:13.39 | Tagor | Trying to call from outside to Asterisk |
| 14:13.49 | Tagor | And then I use the Dial command to forward the call to a SIP phone |
| 14:13.56 | Tagor | But the ring is not really clear |
| 14:14.01 | Tagor | Sometimes I even don't hear a ring |
| 14:14.04 | Tagor | Is there a way to fix this? |
| 14:17.02 | Tagor | Nobody? |
| 14:17.04 | [TK]D-Fender | Tagor : When you answer the call is the sound weak or choppy? |
| 14:17.16 | Tagor | No, it isn't |
| 14:18.01 | Tagor | Only the ring is, [TK]D-Fender |
| 14:18.57 | [TK]D-Fender | Tagor : pastebin your dialplan related to that call. |
| 14:18.59 | [TK]D-Fender | ~pb |
| 14:19.01 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 14:19.38 | Tagor | It's not very big: Just 1,Answer() ... 2,Dial(...some number...) |
| 14:19.39 | key2 | in dialplan, how do I drop the 3 first char from a string ? |
| 14:19.55 | key2 | ${myvar:3} ? |
| 14:21.32 | Tagor | Oh, I forgot to mention this, [TK]D-Fender, if I call the menu internally it works fine |
| 14:21.42 | wunderkin | key2, yes... |
| 14:22.03 | *** join/#asterisk mq_ (n=mquinton@cug31-1-82-234-60-151.fbx.proxad.net) |
| 14:22.46 | [TK]D-Fender | Tagor : Show me the actual dialplan bits. Why should I assume you did everything right when its not working? |
| 14:23.19 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 14:25.50 | *** join/#asterisk C4T3l (n=rcall01@216.54.143.2) |
| 14:26.06 | *** part/#asterisk javar (n=javar@Dynamic-IP-cr20011868204.cable.net.co) |
| 14:32.04 | iq | yo |
| 14:32.08 | Tagor | [TK]D-Fender >> http://pastebin.ca/60303 |
| 14:32.39 | *** part/#asterisk fjean (n=fjean@201.29.140.206) |
| 14:34.05 | [TK]D-Fender | Tagor : Ok, that really doesn't say much as to why... what codec is in use? |
| 14:34.32 | Tagor | u711 |
| 14:34.46 | *** join/#asterisk fugitivo (n=ajf@190.48.171.172) |
| 14:34.47 | fugitivo | hello |
| 14:34.48 | [TK]D-Fender | Tagor : Ok, definately no idea then... |
| 14:35.15 | fugitivo | any ATA with auto-answer/paging support? :) |
| 14:35.25 | *** join/#asterisk southtel (n=slester@c-67-191-211-17.hsd1.ga.comcast.net) |
| 14:35.31 | [TK]D-Fender | fugitivo : No. |
| 14:35.56 | [TK]D-Fender | fugitivo : How can analog line MAKE a phone answer?! |
| 14:37.04 | fugitivo | it can't :) |
| 14:37.49 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 14:38.16 | [TK]D-Fender | fugitivo : Trying thinking just a bit more before you start to feel a little too silly :) |
| 14:38.35 | *** join/#asterisk acehunky (n=chat_jok@59.184.55.52) |
| 14:39.52 | fugitivo | [TK]D-Fender: well, it can be done with an ata an adapter for paging, i don't know what's the translation of "portero electrico" for english |
| 14:40.43 | [TK]D-Fender | fugitivo : That's because that PAGING DEVICE plugged into it is DESIGNED to answer every call! Show me a PHONE that does that... |
| 14:41.10 | [TK]D-Fender | fugitivo : So in effect it isn't being told to answer, it just DOES. |
| 14:41.45 | Hmmhesays | so I got this bluetooth headset for this phone |
| 14:43.55 | flujan | guys, I'm having problems with the Unicall signalling... |
| 14:44.12 | flujan | Do you have experience working with this? |
| 14:48.29 | fugitivo | flujan: i do |
| 14:49.36 | flujan | fugitivo, I'm having problems connecting asterisk to a legacy pbx |
| 14:49.41 | flujan | fugitivo, using a E1 link |
| 14:50.03 | flujan | fugitivo, the legacy pbx sends the first digit and asterisk does not ask for the next one. |
| 14:50.15 | flujan | fugitivo, I have the log from the legacy pbx... |
| 14:50.50 | flujan | fugitivo, http://pastebin.com/749101 |
| 14:51.47 | fugitivo | flujan: i had the same problem, what version of asterisk and unicall are you using? |
| 14:52.12 | fugitivo | flujan: that happened to me when using the wrong unicall version with asterisk 1.2.x |
| 14:52.29 | Juggie | fugitivo, if you fixed your problem, please sign up with the asterisk wiki (www.voip-info.org) and post some information about unicall |
| 14:52.34 | Juggie | there seems to be almost none available |
| 14:52.48 | motu | is there some way for asterisk to detect 10 seconds of silence, and hangup? |
| 14:53.13 | Juggie | yes |
| 14:53.34 | Juggie | app_BackgroundDetect |
| 14:53.41 | fugitivo | Juggie: i gave up and changed everything to PRI, R2 was working for me but not for a production system |
| 14:53.57 | Juggie | still using unicall? |
| 14:54.02 | fugitivo | no |
| 14:54.28 | Juggie | i never heard of unicall till about a week ago |
| 14:56.09 | yxa | i heard digium is coming out with BRI cards? |
| 14:56.19 | file | B410P |
| 14:56.49 | flujan | fugitivo, asterisk version : 1.2.7 |
| 14:56.55 | flujan | fugitivo, unicall version 0.0.3 |
| 14:57.03 | flujan | fugitivo, the last version of unicall. |
| 14:57.13 | flujan | fugitivo, could you help me please? |
| 14:57.39 | *** join/#asterisk clive- (n=pirch@dsl-145-43-219.telkomadsl.co.za) |
| 14:58.00 | *** join/#asterisk kuku5 (n=kuku5@c-71-201-217-245.hsd1.il.comcast.net) |
| 14:58.22 | fugitivo | flujan: hold on |
| 14:58.24 | kuku5 | I get May 31 09:47:10 NOTICE[2144]: chan_zap.c:8171 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 error, I have no IRQ conflicts |
| 14:58.25 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 14:59.43 | fugitivo | flujan: download this versions http://www.soft-switch.org/downloads/snapshots/unicall/ |
| 15:00.53 | *** join/#asterisk BugKham (i=CKGLOB@125.24.12.213) |
| 15:01.17 | *** part/#asterisk southtel (n=slester@c-67-191-211-17.hsd1.ga.comcast.net) |
| 15:01.39 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 15:01.53 | fugitivo | kuku5: what's the output of zttest? run it for 5 minutes |
| 15:02.18 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 15:02.37 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.79.188) |
| 15:03.12 | *** join/#asterisk L|NUX (n=linux@203.101.160.4) |
| 15:03.14 | L|NUX | hello |
| 15:03.22 | L|NUX | can some one help me with enum |
| 15:03.26 | L|NUX | i did tried docs |
| 15:03.43 | flujan | which version of asterisk are you using? |
| 15:03.48 | *** join/#asterisk adker (n=adker@74-33-201-18.br1.glv.ny.frontiernet.net) |
| 15:04.16 | L|NUX | Asterisk CVS HEAD built by root@server.e-maili.com on a i686 running Linux on 2005-10-07 13:02:45 UTC |
| 15:06.03 | kuku5 | fugitivo: 99.987793 so far |
| 15:07.31 | kuku5 | fugitivo: sometimes 100 |
| 15:09.24 | fugitivo | what motherboard? |
| 15:10.07 | kuku5 | eh, its an hp server |
| 15:10.12 | kuku5 | 350 |
| 15:10.26 | *** join/#asterisk IOscanner (n=IOscanne@c-67-164-154-209.hsd1.tx.comcast.net) |
| 15:10.53 | fugitivo | kuku5: read this http://www.asteriskguru.com/tutorials/hdlc_bad_fcs.html |
| 15:11.14 | fugitivo | kuku5: i have the same problem but none of that solved my problem, so i think i have bad hardware |
| 15:12.06 | kuku5 | but ppl actually have problems dialing out |
| 15:13.37 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 15:16.41 | Hmmhesays | ahh crap i didn't realize nochex was UK only |
| 15:17.09 | *** join/#asterisk _Sam-- (n=sam@fresco.kneedraggers.com) |
| 15:21.53 | *** join/#asterisk hfb (n=hfb@pool-71-116-252-188.lsanca.dsl-w.verizon.net) |
| 15:22.33 | *** join/#asterisk cyn0n (n=cyn0n@n223-99-152-206.tranquility.net) |
| 15:23.17 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 15:23.47 | cyn0n | so I have simul. ring for incoming like 'exten => ZAP/1,1,Dial(ZAP/1&ZAP/2&ZAP/3&ZAP/4)' but the phones keep ringing after one is picked up... can someone point me in the right direction for the extensions to hang up once one is answered? |
| 15:23.55 | BugKham | oh, doesn't Dial(SIP/${EXTEN},20,rL(70000:61000:30000)) work in 1.2.8 still? |
| 15:24.18 | BugKham | I wish it'd been fixed |
| 15:24.30 | BugKham | can anyone try? |
| 15:25.57 | *** part/#asterisk cfh (n=luca@82.193.23.6) |
| 15:29.27 | *** join/#asterisk zmb (n=chatzill@miracle.cc.columbia.edu) |
| 15:30.38 | zmb | hello all. I'm having some issues using asterisk realtime voicemail. when calling voicemailmain(@mycontext) there is dead silence |
| 15:31.11 | zmb | if i place that call on hold and reconnect, voicemail seems to start prompting for extension/pass |
| 15:31.17 | zmb | any suggestions? |
| 15:33.57 | zmb | hello hello ello lo lo |
| 15:38.46 | C4T3l | jbot: hello |
| 15:38.53 | jbot | Howdy Bub |
| 15:39.59 | zmb | i need help with asterisk realtime and voicemail... when calling voicemailmain(@mycontext) there is dead silence |
| 15:42.43 | [TK]D-Fender | zmb : Please don't spam your requests like that. Very few people here use what you are describing. |
| 15:43.56 | [TK]D-Fender | zmb : And if putting the call on hold and takeing it back gets you audio then its a VOICE issue, not Voicemail, Realtime, or any of the other components of your call that is to blame. |
| 15:44.13 | [TK]D-Fender | zmb : Check your phone configs |
| 15:44.57 | zmb | TK: actually i noticed that before hold the sql query is made with username/number only |
| 15:45.34 | zmb | tk: where as the after removing from hold, sql query also includes the context |
| 15:45.39 | fugitivo | well, for what i've seen, sql queries are not well written in asterisk :) |
| 15:46.26 | *** join/#asterisk wunderkin (i=kev@69.26.192.234) |
| 15:49.02 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:49.13 | [TK]D-Fender | zmb : Maybe you could try pastebining some high-verbose CLI output to back it up. |
| 15:49.16 | [TK]D-Fender | ~pb |
| 15:49.18 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 15:51.38 | *** join/#asterisk Tier_1 (n=Tier@c-24-9-75-234.hsd1.co.comcast.net) |
| 15:55.45 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 15:56.36 | C4T3l | ~seen jbot |
| 15:56.51 | jbot | jbot is currently on #asterisk-doc (4d 16h 12m 8s) ##t42 (4d 16h 12m 8s) #asterisk (4d 16h 12m 8s) #byumug (4d 16h 12m 8s) ##ducleague (4d 16h 12m 8s) #aegis (4d 16h 12m 8s) ##stouteriks (4d 16h 12m 8s) ##pxe (4d 16h 12m 8s) ##essy (4d 16h 12m 8s), last said: 'rumour has it, sarge is ... |
| 15:57.06 | *** join/#asterisk Wowzers10 (n=pbaker@nnat-gw.adeptra.com) |
| 15:57.10 | *** join/#asterisk MstlyHrmls (n=mh@66.195.193.151) |
| 15:57.32 | *** join/#asterisk naquada (n=FreePBX3@terlouw01.adsl.nl.inmeso.net) |
| 15:57.49 | Wowzers10 | Hi all, I'm trying to cut the last 4 digits off a caller id and I'm having trouble - could someone look at exten=>s,1,Set(last=${CUT(CALLERID(num),,4)}) and tell me what I'm doing wrong |
| 15:58.10 | sevard | <PROTECTED> |
| 15:58.24 | naquada | hi people i have a question : My Asterisk does not see when a caller hangs up, this is very annoying .. anything i can do to fix this ?? |
| 15:59.14 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 15:59.47 | [TK]D-Fender | Wowzers10 : ${CALLERID(num):0:$[${LEN(${CALLERID(num)})}-4]} |
| 16:01.24 | BugKham | [TK]D-Fender: r u running Dial with the L flag at all? |
| 16:01.57 | L|NUX | can some one help me with enum |
| 16:08.23 | fugitivo | naquada: zap calls? |
| 16:08.45 | key2 | In a dialplan, how do I drop the last char from a string ? like when I want to drop the first car, I do ${string:1} , what is it for dropping the last one ? |
| 16:10.52 | key2 | [TK]D-Fender: u know ? |
| 16:12.19 | *** join/#asterisk arcy (n=arcanum@195.74.248.37) |
| 16:12.20 | [TK]D-Fender | key2 : Looko up.... |
| 16:12.35 | blitzrage | wow... Roger's internet sux0rz |
| 16:12.35 | *** join/#asterisk guillecabeza (n=elguille@r201-217-191-147.dialup.adsl.anteldata.net.uy) |
| 16:13.03 | key2 | [TK]D-Fender: I just cleaned the screan :( |
| 16:13.20 | *** join/#asterisk KranZ (n=user@imail.bestline.net) |
| 16:13.22 | guillecabeza | Hi |
| 16:13.42 | guillecabeza | I woner if someone can help me out with some hardware issues? |
| 16:13.50 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 16:14.17 | xheliox | Stupid question. Can you have a single SIP account in the sip.cfg but have a multi line phone log into that same account more than once? And if so, what is the result? |
| 16:14.20 | BugKham | key2: http://www.voip-info.org/wiki/index.php?page=Asterisk+variables |
| 16:14.24 | fugitivo | key2: <[TK]D-Fender> Wowzers10 : ${CALLERID(num):0:$[${LEN(${CALLERID(num)})}-4]} |
| 16:14.46 | fugitivo | xheliox: nonsense? |
| 16:14.52 | xheliox | :) |
| 16:15.03 | xheliox | Sorta figured. |
| 16:15.26 | xheliox | So there's no way around defining another sip profile for each line, eh? |
| 16:15.37 | CunningPike | xheliox: The result is that the registration ends up with the last line appearance to register |
| 16:15.52 | CunningPike | xheliox: Each grabs the registration from the other |
| 16:16.05 | guillecabeza | Uhhuh. NMI received. Dazed and confused, but trying to continue |
| 16:16.05 | guillecabeza | You probably have a hardware problem with your RAM chips |
| 16:16.08 | xheliox | CunningPike: Hmm, that might not be so awful.. but not ideal. |
| 16:16.11 | guillecabeza | have you ever seen those messages? |
| 16:16.31 | fugitivo | guillecabeza: i did |
| 16:16.36 | fugitivo | guillecabeza: change the card to another slot |
| 16:16.37 | guillecabeza | happening at least on 4 production servers |
| 16:16.46 | CunningPike | guillecabeza: Yes - and we've never figured out what causes them. It doesn't seem to cause us any problems..... |
| 16:16.47 | fugitivo | guillecabeza: cat /proc/interrupts to see irq sharing |
| 16:16.48 | guillecabeza | is it an irq prblem? |
| 16:16.58 | guillecabeza | 0: 883490 16510579 IO-APIC-edge timer |
| 16:16.59 | guillecabeza | <PROTECTED> |
| 16:16.59 | guillecabeza | <PROTECTED> |
| 16:16.59 | guillecabeza | <PROTECTED> |
| 16:16.59 | guillecabeza | <PROTECTED> |
| 16:16.59 | guillecabeza | <PROTECTED> |
| 16:17.01 | guillecabeza | 169: 0 100000 IO-APIC-level uhci_hcd, uhci_hcd |
| 16:17.03 | guillecabeza | 177: 0 0 IO-APIC-level uhci_hcd |
| 16:17.05 | guillecabeza | 185: 0 0 IO-APIC-level uhci_hcd |
| 16:17.07 | guillecabeza | 193: 0 0 IO-APIC-level ehci_hcd |
| 16:17.08 | fugitivo | ~pb |
| 16:17.15 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 16:17.15 | guillecabeza | 201: 47246522 423 IO-APIC-level eth0 |
| 16:17.15 | guillecabeza | 217: 15808 955992 IO-APIC-level cciss0 |
| 16:17.15 | guillecabeza | 225: 1489034 15869693 IO-APIC-level wct4xxp |
| 16:17.15 | CunningPike | lol |
| 16:17.15 | guillecabeza | 233: 15869665 1489048 IO-APIC-level wct4xxp |
| 16:17.16 | [TK]D-Fender | xheliox : Why would you try to register multiple times from the same phone |
| 16:17.34 | guillecabeza | sorry |
| 16:18.09 | BugKham | [TK]D-Fender: are you using cmd Dial with the "L" flag at all? |
| 16:18.15 | CunningPike | guillecabeza: We tried every combination of APIC, IRQ settings, the works |
| 16:18.21 | BugKham | or anyone else? |
| 16:18.22 | [TK]D-Fender | BugKham : nope |
| 16:18.28 | xheliox | [TK]D-Fender: Because I'm lazy and I didn't want to define multiple sip accounts for each phone. :) |
| 16:18.29 | fugitivo | guillecabeza: try changing the card to another slot |
| 16:18.44 | [TK]D-Fender | xheliox : Why would you do EITHER? |
| 16:18.53 | [TK]D-Fender | xheliox : What kind of phone are you using? |
| 16:19.02 | xheliox | Because it has multiple lines? Polycom 501.. |
| 16:19.13 | guillecabeza | CunningPike: you've also solved your problems changing slots? |
| 16:19.27 | [TK]D-Fender | xheliox : But you basically want to be on more than 1 call at a time on a sigle registration, right? |
| 16:19.28 | fugitivo | guillecabeza: what does the NMI line says when you do cat /proc/interrupts? |
| 16:19.32 | CunningPike | guillecabeza: If I had another slot to change to.......... maybe :) |
| 16:19.42 | fugitivo | guillecabeza: is it != 0 ? |
| 16:19.56 | guillecabeza | yeah |
| 16:19.58 | xheliox | [TK]D-Fender: Yeah, idealy, I'd like all lines to be the same. |
| 16:20.00 | guillecabeza | 169 |
| 16:20.02 | sevard | How can a phone be registerd just fine and have the exact same syntax as another working phone but yet only have the ability to recieve calls not make them |
| 16:20.12 | fugitivo | guillecabeza: read this http://kb.digium.com/entry/1/64/ |
| 16:20.17 | CunningPike | fugitivo: Our is always 1 |
| 16:20.25 | sevard | May 31 11:19:21 NOTICE[24014]: chan_sip.c:10326 handle_request_invite: Failed to authenticate user 02 <sip:02@192.168.1.78:5060> |
| 16:20.29 | sevard | what the hell |
| 16:20.34 | fugitivo | CunningPike: read this too :) http://kb.digium.com/entry/1/64/ |
| 16:20.44 | CunningPike | fugitivo: I am ;) |
| 16:20.49 | KranZ | CunningPike: i had issues with those cards. Ended up disabled onboard sata and buying the equivalent pci controller card |
| 16:20.49 | guillecabeza | oh |
| 16:20.51 | guillecabeza | thanks! |
| 16:20.51 | [TK]D-Fender | xheliox : then set it to linekeys="3" and callsperlinekey="1" |
| 16:20.54 | KranZ | CunningPike: fixed |
| 16:21.10 | [TK]D-Fender | xheliox : and you only need that 1 reg |
| 16:21.15 | *** join/#asterisk slobberknocker (n=ckwall@63.149.122.94) |
| 16:21.24 | xheliox | [TK]D-Fender: So in my .cfg file, do I not register each line? |
| 16:21.45 | *** join/#asterisk burizaa (n=freeee@cm73.omega100.maxonline.com.sg) |
| 16:21.47 | [TK]D-Fender | xheliox : No, only the first |
| 16:21.50 | CunningPike | KranZ: Ah - that might help us........ our drives are all SCSI, so I'll check and see if SATA is enabled or not.... |
| 16:21.57 | xheliox | Hmm, sweet. :) |
| 16:22.03 | KranZ | CunningPike: you getting HDLC 6 aborts? |
| 16:22.38 | slobberknocker | I am looking for something to the effect of where to set my ip gateway in either my [macaddress].cfg file or my sip.cfg file but am not finding it. is it named something else on the polycom phones or is it just not there? |
| 16:22.39 | xheliox | [TK]D-Fender: That's exactly what I wanted to hear. |
| 16:22.44 | xheliox | [TK]D-Fender: Thanks. |
| 16:22.45 | guillecabeza | Kranz: I had HDLC aborts also |
| 16:22.47 | burizaa | Any one got example how to setup asterisk to work with SIP provider whose doesn't supply username and password, but just allow our traffic trough? |
| 16:23.07 | KranZ | guillecabeza: its probably because your disk controller is on the same bus as your pri card |
| 16:23.13 | *** join/#asterisk xMOe (n=Blade@62.149.93.73) |
| 16:23.17 | KranZ | guillecabeza: has nothing to do with irqs |
| 16:23.18 | fugitivo | burizaa: IPCOMMS? |
| 16:23.19 | [TK]D-Fender | xheliox : Seperate these concepts : a Line is an IDENTITY. Any identity can support up to 24 calls per line key, and span as many Linekeys as you have available. |
| 16:23.35 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 16:23.43 | burizaa | fugitivo: it's actually belong to my friend, and i just want to try using them |
| 16:23.52 | xheliox | TK: So can I presume that when a second call comes in, it'll present itself on the next line? |
| 16:24.00 | burizaa | fugitivo: it's suvfone ... no website tough |
| 16:24.01 | CunningPike | KranZ: No - no HDLC problems |
| 16:24.03 | KranZ | guillecabeza: try "hdparm -tT /dev/sda (for scsi or sata) or /dev/hda (ide)" |
| 16:24.04 | fugitivo | KranZ: NMI means Non Maskable Interrupt |
| 16:24.14 | KranZ | ahh |
| 16:24.15 | naquada | no SIP calls |
| 16:24.16 | flujan | fugitivo, I installed the snapshot libs and still have problems using unicall. |
| 16:24.31 | KranZ | guess i came into the conversation a bit late |
| 16:24.58 | CunningPike | KranZ: We don't really have a problem - I was just sharing experiences with guillecabeza |
| 16:25.13 | slobberknocker | i have set up provisioning with my polycom 301s and 501s, but they do not contact the sntp server. before provisioning i had to set the ip gateway to our default gateway address. where can i set that from the files for provisioning? |
| 16:25.18 | CunningPike | We get a single NMI when the card loads and that's it |
| 16:25.25 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 16:25.34 | slobberknocker | dhcp server does not send default gateway |
| 16:25.42 | fugitivo | flujan: pastebin your unicall.conf |
| 16:25.45 | CunningPike | slobberknocker: I was just about to say DHCP |
| 16:25.49 | CunningPike | :) |
| 16:25.55 | fugitivo | flujan: and zaptel.conf |
| 16:26.06 | slobberknocker | if not dhcp is there an option? |
| 16:26.09 | [TK]D-Fender | xheliox : Using the specs I gave you, yes. |
| 16:26.13 | guillecabeza | CunningPike: besides the tunning for NMI, is it really known why is so pervasive the NMI issue with Digium boards? I mean, it's not an isolated case... |
| 16:26.45 | CunningPike | slobberknocker: I don't believe so |
| 16:26.47 | [TK]D-Fender | xheliox : each line key will be tied to that first reg and subsequent calls will automatically flow to the next key. |
| 16:27.01 | fugitivo | guillecabeza: well... I gave up with digium boards, I'll go with sangoma |
| 16:27.01 | CunningPike | guillecabeza: I wish I knew |
| 16:27.13 | xheliox | [TK]D-Fender: And one more question -- if I use that setting, what will the display name on the phone be for the second line? Same as the first line? |
| 16:27.43 | [TK]D-Fender | xheliox : your phone will have ONE line, but support multiple calls. Same identity |
| 16:27.52 | guillecabeza | CPK : I've sent an email to support@digium but no response yet... |
| 16:28.05 | *** join/#asterisk sb_mx (n=sb_mx@200.94.154.226) |
| 16:28.21 | [TK]D-Fender | slobberknocker : how is it you DHCP server isn't setting a default gateway? |
| 16:28.40 | xheliox | [TK]D-Fender: Okay. Thanks. You're always a great help. :) |
| 16:29.23 | fugitivo | guillecabeza: one advice, call them, you won't receive any answer by email |
| 16:29.29 | [TK]D-Fender | slobberknocker : And you CAN hard-code just about everything and anything you want in the provisioning files, and MUCH of it in DHCP (extended options) |
| 16:31.57 | guillecabeza | ok guys, thank you all, I'll try your advices, see what happens |
| 16:32.08 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 16:32.18 | key2 | [TK]D-Fender: Is there a nice way to add something in a SQL table from the extension.conf ? or have to use AGI ? |
| 16:32.36 | *** part/#asterisk guillecabeza (n=elguille@r201-217-191-147.dialup.adsl.anteldata.net.uy) |
| 16:32.45 | CunningPike | slobberknocker: According to the SIP Admin Guide, IP gateway is not an option in the config files - you need to use DHCP options or the phone menu |
| 16:32.48 | naquada | I have a question.. when asterisks receives calls (sip) it does not notice it when they hang up...how can i solve this? |
| 16:33.03 | Wowzers10 | [TK]D-Fender: with ${CALLERID(num):0:$[${LEN(${CALLERID(num)})}-4]} is there a way to have it subtract from the end to the begining because that will cut out the last 4 digits |
| 16:33.33 | slobberknocker | CunningPike: thank you |
| 16:33.35 | key2 | Wowzers10: ${foo:4} |
| 16:33.38 | [TK]D-Fender | Wowzers10 : Taht chops off the last 4. Period. |
| 16:33.54 | key2 | [TK]D-Fender :) |
| 16:34.04 | *** join/#asterisk copland (n=stonecol@209.216.65.10) |
| 16:34.08 | Wowzers10 | I know, but I need the last 4 digits |
| 16:34.20 | [TK]D-Fender | key2 : your method removes LEADING chars, mine removes TRAILING chars... |
| 16:34.26 | flujan | fugitivo, http://pastebin.com/749317 |
| 16:34.29 | [TK]D-Fender | Wowzers10 : OH. 1 sec |
| 16:34.34 | copland | Is anyone using Quantum voice service I can make outbound calls fine inbound are noather issue |
| 16:34.38 | key2 | [TK]D-Fender: what the hell deos he want then |
| 16:34.43 | fugitivo | naquada: please be more specific with your question |
| 16:35.06 | [TK]D-Fender | Wowzers10 : ${CALLERID(num):$[${LEN(${CALLERID(num)})}-4]} |
| 16:35.12 | key2 | naquada: is there anything in the middle ? Cuz aparently you're loosing the BYE |
| 16:35.13 | naquada | fugitivo, the calls just dont disconnect after the caller has hung up... |
| 16:35.30 | key2 | naquada: make sure u get the BYE |
| 16:35.52 | fugitivo | flujan: you have span 1 configured twice in zaptel.conf |
| 16:35.52 | key2 | [TK]D-Fender: so for doing an Insert into a mysql, do I have to use AGI ? |
| 16:35.57 | naquada | how do i do that? :s |
| 16:36.11 | key2 | naquada: who hangs up ? |
| 16:36.16 | key2 | naquada: asterisk ? |
| 16:36.18 | [TK]D-Fender | key2 : MySQL() |
| 16:36.21 | naquada | the caller hangs up |
| 16:36.28 | key2 | [TK]D-Fender: is it a function ? |
| 16:36.31 | naquada | and asterisk does not notice that... |
| 16:36.37 | flujan | fugitivo, igore it please |
| 16:36.38 | [TK]D-Fender | key2 : There is a dedicated application for it already... so depending what you need to pass it... NO. |
| 16:36.51 | flujan | it was a error in the copy and paste process... I edited removing the comment lines. |
| 16:36.54 | flujan | fugitivo, sorry |
| 16:37.08 | fugitivo | flujan: config looks ok, did you stop asterisk and unload zaptel after the installation of snapshots? |
| 16:37.56 | flujan | fugitivo, Yes I reboot the machine |
| 16:38.22 | flujan | fugitivo, the legacy pbx works well if I plug a E1 direct in it |
| 16:38.51 | flujan | fugitivo, asterisk should works like a E1 to the legacy pbx. |
| 16:38.54 | fugitivo | flujan: what if you plug the E1 line to asterisk? |
| 16:39.10 | flujan | I didn't try it yet |
| 16:39.21 | fugitivo | can you try it? |
| 16:39.21 | key2 | naquada: run SIP DEBUG |
| 16:39.31 | key2 | naquada: and make sure u see the BYE comming |
| 16:39.36 | flujan | fugitivo, I will try it today at 8:00PM |
| 16:39.41 | sevard | [TK]D-Fender: do you know why? :| |
| 16:39.51 | flujan | fugitivo, I'm just making sure I did things well |
| 16:39.55 | fugitivo | flujan: i had your problem and solved it with the snapshots |
| 16:40.11 | sevard | [TK]D-Fender how can a phone be registerd just fine and have the exact same syntax as another working phone but yet only have the ability to recieve calls not make them |
| 16:41.11 | key2 | sevard: Pastebin your channel conf |
| 16:41.14 | fugitivo | flujan: did you try calling from the pbx to asterisk? does it work? |
| 16:41.25 | sevard | sorry, channel.conf ? |
| 16:41.36 | key2 | sevard: sip.conf ? |
| 16:41.39 | sevard | ah. |
| 16:41.39 | jlewis | anyone know if 2N voiceblue GSM gateways are sold in the US? |
| 16:41.41 | key2 | sevard: iax.conf ? |
| 16:41.44 | flujan | fugitivo, please, take a look in my extensions.conf http://pastebin.com/749325 |
| 16:41.46 | sevard | it is sip, gotcha. |
| 16:42.27 | fugitivo | flujan: that doesn't look right |
| 16:42.27 | flujan | fugitivo, is it ok? |
| 16:42.38 | flujan | fugitivo, what? |
| 16:42.51 | [TK]D-Fender | sevard : ... because? Whats the question... I missed it somehow... |
| 16:43.24 | sevard | I'm pastebining my sip.conf |
| 16:43.25 | L|NUX | hello |
| 16:43.32 | L|NUX | can some one please help me with enum :( |
| 16:43.33 | sevard | [TK]D-Fender how can a phone be registerd just fine and have the exact same syntax as another working phone but yet only have the ability to recieve calls not make them |
| 16:43.36 | sevard | May 31 11:19:21 NOTICE[24014]: chan_sip.c:10326 handle_request_invite: Failed to authenticate user 02 <sip:02@192.168.1.78:5060> |
| 16:43.39 | [TK]D-Fender | sevard : Oh, there it is. Because your registration is with the RIGHT user/pass so it will RECEIVE calls, but the auth is different. |
| 16:43.39 | L|NUX | i have read the docs |
| 16:43.43 | L|NUX | and implement it |
| 16:43.47 | L|NUX | but its not working |
| 16:43.48 | L|NUX | :( |
| 16:43.57 | [TK]D-Fender | sevard : pastebin your phoneXX.cfg and sip.conf |
| 16:44.06 | flujan | fugitivo, what is wrong in the extesnons.conf? |
| 16:44.26 | sevard | [TK]D-Fender: what do you mean by a phoneXX.cfg |
| 16:44.39 | jlewis | L|NUX: how are you implementing enum? |
| 16:44.54 | L|NUX | i edited my enum.conf and extensions.conf |
| 16:45.06 | L|NUX | i registered my DID at e164.org |
| 16:45.08 | *** join/#asterisk Seggy (i=rbutler@tsss.org) |
| 16:45.08 | jlewis | L|NUX: we've been doing enum using our own AGI for about a year |
| 16:45.33 | L|NUX | jlewis : if normal will work for me then AGI will be not a problem |
| 16:47.06 | fugitivo | flujan: you are sending calls comming from span 1 to context incoming, but your extensions.conf doesn't have any context |
| 16:47.18 | sevard | [TK]D-Fender: key2: sip.conf== http://pastebin.com/749336 |
| 16:47.34 | flujan | fugitivo, sorry again... |
| 16:48.05 | flujan | I have the line [incoming] just below line 11 in the pastebin |
| 16:48.19 | [TK]D-Fender | sevard : ok, need the phone conf now. |
| 16:48.20 | flujan | fugitivo, a problem editing the file with vi. :) |
| 16:48.29 | fugitivo | :) |
| 16:48.32 | *** part/#asterisk slobberknocker (n=ckwall@63.149.122.94) |
| 16:48.40 | flujan | fugitivo, does the dialplan look reasonable to you? |
| 16:48.41 | sevard | [TK]D-Fender: do you mean the cfg that the Aastra gets via TFTP? or is this an asterisk configuration file that I am unaware of |
| 16:48.56 | flujan | fugitivo, I just want to answer and playback hello-world. :) |
| 16:49.11 | fugitivo | flujan: well, could you pastebin it again? |
| 16:49.20 | [TK]D-Fender | sevard : Aastra? |
| 16:49.37 | [TK]D-Fender | sevard : Lemme see it anyways... |
| 16:49.42 | sevard | [TK]D-Fender: that's the phone that is trying to make calls, line 1 works, the rest of the lines can't call out |
| 16:49.47 | sevard | [TK]D-Fender: what exactly do you mean by phoneXX.conf |
| 16:50.21 | sevard | [TK]D-Fender: :P http://pastebin.com/749342 |
| 16:50.23 | [TK]D-Fender | sevard : Sorry, for some reason I assumed Polycom... show me what you've got. |
| 16:51.23 | [TK]D-Fender | sevard : So line2 is no good? And the rest are? |
| 16:51.25 | flujan | fugitivo, here it goes: http://pastebin.com/749348 |
| 16:51.45 | *** join/#asterisk Persilon (n=ajolodov@200.123.112.115) |
| 16:52.06 | sevard | [TK]D-Fender: line one works great, that's 0, the rest are bunk |
| 16:52.19 | *** join/#asterisk FuRR_ (i=[U2FsdGV@adsl-69-108-94-232.dsl.irvnca.pacbell.net) |
| 16:52.33 | Persilon | Hey, I need some help setting up inbound calls, may anyone help me ? |
| 16:52.45 | fugitivo | flujan: add loglevel=1023 to unicall.conf |
| 16:52.59 | CunningPike | Persilon: If you provide some info, maybe ;) |
| 16:53.21 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 16:53.47 | fugitivo | flujan: what test are you doing? dialing from the pbx to asterisk or asterisk to pbx? |
| 16:54.08 | Persilon | CunningPike: I have a voip telephone number, but when I call from a pstn line to it, I get Failed to authenticate user "My Telephone" <sip:My Telephone@My VoIP Provider> |
| 16:54.21 | [TK]D-Fender | sevard : WELL... only your first reg has a "screen Name" value... perhaps Synch that up? |
| 16:54.23 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 16:54.40 | [TK]D-Fender | sevard : Hmm.. its there.. just lower... scratch that maybe |
| 16:54.53 | [TK]D-Fender | sevard : Whats that "mode" imply? |
| 16:55.39 | CunningPike | Persilon: That exact message? If so, sounds like to need to insert your information in place of the defaults. What is your setup? |
| 16:55.57 | flujan | fugitivo, I'm dialing using a extension connect to the legacy pbx... the legacy pbx place the call in the E1 where I put asterisk.... |
| 16:56.09 | Persilon | CunningPike: I get the real values (the actual pstn line number) |
| 16:56.19 | sevard | [TK]D-Fender: used for special phones, 0 for generic |
| 16:56.39 | sevard | like special modes for Nortel stuff, or something. |
| 16:56.41 | CunningPike | Persilon: How are you connected to the PSTN?? |
| 16:57.00 | CunningPike | Persilon: Apologies for the double ? |
| 16:57.18 | Persilon | CunningPike: I'm not connected to the pstn line via asterisk... I'm just calling from a normal telephone from my office |
| 16:57.19 | Persilon | CunningPike: np :) |
| 16:57.23 | fugitivo | flujan: what number are you dialing? |
| 16:57.44 | flujan | fugitivo, take a look at: http://pastebin.com/749355 |
| 16:57.52 | CunningPike | Persilon: You'll need to describe your call path then..... |
| 16:58.21 | flujan | fugitivo, 44882120 The pbx sends the number 4. Asterisk send a seize ack... but the pbx never receive the seize ack to send the other digit. |
| 16:58.25 | [TK]D-Fender | sevard : Ok, I'm at a loss... don't see anything obvious to tip me off and don't psossess any real-world knowledge of Aastra's to contribe more. Sorry |
| 16:58.30 | mocker | Has anyone ever had the problem of cell phones not letting their callerID be overwritten? |
| 16:58.37 | Persilon | CunningPike: my extensions.conf looks like this: [inbound] exten => $NUMBER,1,Answer() |
| 16:58.52 | sevard | [TK]D-Fender: does asterisk have problems with extensions starting with a 0? |
| 16:58.53 | mocker | When someone calls my queue from a landline, the info gets rewritten just fine. Cell phones don't get overwritten at all. :( |
| 16:59.06 | Persilon | CunningPike: I tried doint Set(CALLERID(name)=myusername) before that, but it didn't work either |
| 16:59.16 | flujan | fugitivo, at least... this is what appear to be happening. |
| 16:59.17 | fugitivo | flujan: you need to setup that number in extensions.conf |
| 16:59.19 | [TK]D-Fender | sevard : nope |
| 16:59.23 | sevard | [TK]D-Fender: ojspojge. |
| 16:59.33 | flujan | fugitivo, the full number? 44882120 for instance? |
| 16:59.42 | sevard | [TK]D-Fender: I -do- know that this phone was working flawlessly on the old serve.r |
| 16:59.52 | fugitivo | flujan: try with an extension first, for example 1234 |
| 17:00.05 | fugitivo | flujan: add it to your incoming context |
| 17:00.08 | [TK]D-Fender | sevard : You seemed pretty clear its an auth issue.... maybe the phone isn't actually RUNNING that config? Can you confirm that its actually picking it up? |
| 17:00.10 | flujan | fugitivo, if i put something like _XXXXXXXX.. |
| 17:00.11 | sevard | i most definatly believe it to be an asterisk configuration option i have overlooked |
| 17:00.25 | sevard | [TK]D-Fender: yes, i've confirmed that it is running that config |
| 17:00.26 | CunningPike | Persilon: Yes, yes - but I don't have a picture of your overall setup - 50,000 ft level - how is your inbound PSTN call arriving at your asterisk server? |
| 17:00.35 | [TK]D-Fender | sevard : Maybe your * config needs a "sip reload"? |
| 17:00.38 | fugitivo | flujan: it should work, but for tests I recommend starting with the basics |
| 17:00.48 | sevard | [TK]D-Fender: done that with full reloads / stop / restarts |
| 17:00.58 | fugitivo | flujan: exten => 1234,1,Answer() exten => 1234,1,Playback(hello-world) |
| 17:01.08 | Persilon | CunningPike: telephone line -> my voip provider -> asterisk |
| 17:01.23 | flujan | fugitivo, doesn't work... some problme. |
| 17:01.33 | Persilon | CunningPike: outbound calls work just fine |
| 17:01.37 | flujan | asterisk send the seize ack and the pbx doesn't receive it |
| 17:01.52 | [TK]D-Fender | fugitivo : Please try to avoid having 2 priority "1" :) |
| 17:01.57 | CunningPike | Persilon: OK, I see |
| 17:02.00 | flujan | fugitivo, i'm using a cross T1 cable. to connect then |
| 17:02.03 | flujan | ops |
| 17:02.05 | flujan | them |
| 17:02.15 | fugitivo | flujan: do you get green light? |
| 17:02.22 | *** join/#asterisk salviadud (n=ralfalfa@201.133.207.93) |
| 17:02.31 | flujan | fugitivo, yeap no alarms |
| 17:02.31 | CunningPike | Persilon: Can you pastebin your sip.conf? |
| 17:02.32 | Persilon | CunningPike: It's like it's trying to validate my telephonenumber@myvoipprovider |
| 17:02.46 | Persilon | CunningPike: yes, give me a minute |
| 17:02.49 | fugitivo | flujan: run ztcfg -vvvvvvvv and try again |
| 17:02.56 | russellb | fyi, the rtp jitterbuffer just got merged into the trunk |
| 17:03.10 | salviadud | fugitivo, thats a lot of verbose, que chingados |
| 17:03.43 | fugitivo | salviadud: :) |
| 17:03.49 | flujan | fugitivo, the problem persists... I'm not receiving all digits |
| 17:04.16 | fugitivo | flujan: ok, i remember having your problem, and upgrading unicall with snapshots... |
| 17:04.36 | flujan | fugitivo, the legacy pbx says it... it neither send the digits nor end the call. |
| 17:04.36 | fugitivo | flujan: did you install spandsp snapshot? |
| 17:04.43 | flujan | fugitivo, yes |
| 17:04.44 | flujan | :) |
| 17:05.08 | flujan | fugitivo, I will try to speak with coppice (Steves) who wrote unicall and spandsp. |
| 17:05.15 | flujan | fugitivo, maybe he have some clue... :) |
| 17:05.49 | fugitivo | flujan: change protocolvariant from 20 to 4 and restart asterisk |
| 17:05.56 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 17:07.05 | Persilon | CunningPike: http://pastebin.com/749370 |
| 17:07.18 | copland | is there a way to tell if which codecs are currently being used in a call |
| 17:07.27 | flujan | flujan, done... the problem still occurs... :( |
| 17:08.10 | flujan | fugitivo, I think the problem is in the seize ack which asterisk sends... The legacy pbx never get it or never understand it . |
| 17:08.46 | fugitivo | flujan: you should try connecting the E1 line directly to the asterisk box |
| 17:08.54 | flujan | flujan, yeap |
| 17:09.03 | flujan | today I will have a answer to this question... :D |
| 17:09.46 | flujan | fugitivo, thanks for the help!!! |
| 17:09.47 | CunningPike | Persilon: I assume your 'fromuser' and 'fromdomain' are set to whatever they should be? |
| 17:09.52 | flujan | fugitivo, see yas |
| 17:10.02 | Persilon | CunningPike: yes yes, the settings are right |
| 17:10.18 | Persilon | CunningPike: outbound calls works with those settings |
| 17:11.34 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 17:14.25 | CunningPike | Persilon: Hmmm - there's nothing obviously wrong, assuming that all the 'mypass' etc is actual values in your real file |
| 17:14.54 | Persilon | CunningPike: I think they are, otherwise oubound calls would not work either |
| 17:15.31 | Persilon | CunningPike: the problem is that when I get a call it tries to authenticate as telephone_number@voip_provider, and obviously that's wrong |
| 17:16.27 | CunningPike | Persilon: I'm not sure that it's wrong per se - maybe someone else can help |
| 17:16.30 | *** join/#asterisk amdtech (n=amd011@ab1-1-246.shsu.edu) |
| 17:17.39 | Persilon | CunningPike: I think that's wrong cause it tries to authenticate... if I call from my cell phone I'd get sip:01115mycell@ip_provider |
| 17:18.00 | Persilon | CunningPike: it treates the telephone number as an actual user |
| 17:18.24 | *** join/#asterisk slobberknocker (n=ckwall@63.149.122.94) |
| 17:19.49 | *** join/#asterisk Wowzers10 (n=pbaker@nnat-gw.adeptra.com) |
| 17:19.51 | slobberknocker | ok, so i have formatted the file system on my polycom 501 and have set the ntp address in my sip.cfg. But my boot.log says that it is not finding the ntp server at 0.0.0.0, which is the settings on the phone it's self. how do i get it to search the sip.cfg file for ntp? |
| 17:20.14 | Persilon | CunningPike: I also don't know if my extensions.conf is right, I have: [inbound] _myvoipbumber,1,Answer() |
| 17:20.21 | Wowzers10 | Hi all, is it possible to use includes in the voicemail.conf, IE I have several vm contexts but want to combine them for the use of the Directory application |
| 17:20.26 | CunningPike | Persilon: Hmmm - but isn't that how you would get the equivalent of CID? There must be a way to set it up so that all calls from @voipprovider are accepted - unfortunately, I have never done this myself |
| 17:20.43 | MstlyHrmls | slobberknocker: the bootrom doesn't use the ntp setting from sip.cfg, only the SIP app does |
| 17:21.12 | Persilon | CunningPike: I tried to add the sip_proxy context in sip.conf but it doesn't work either |
| 17:21.21 | CunningPike | Wowzers10: In theory, yes - but you run into problems with password changes....... |
| 17:21.28 | slobberknocker | ok, so if i want to set ntp globally how would i do that? |
| 17:21.29 | MstlyHrmls | slobberknocker: is the phone's time set correctly after it completes its boot? |
| 17:21.34 | slobberknocker | no |
| 17:21.56 | slobberknocker | it actually give me an error in the log file that says it did not find the time server... hang on, i will copy it |
| 17:22.03 | MstlyHrmls | slobberknocker: DHCP, or in the phone. But it sounds like something else is wrong in your config if the App's time isn't correct |
| 17:22.25 | slobberknocker | 0101000053|app1 |4|00|Could not load time from 0.0.0.0(0.0.0.0). |
| 17:22.38 | *** part/#asterisk BugKham (i=CKGLOB@125.24.12.213) |
| 17:23.19 | slobberknocker | if i set the time on the phone, it works. but i dont want to have to do that on every phone, i want it to be global. |
| 17:23.25 | CunningPike | slobberknocker: What are the ntp settings from your sip.cfg? |
| 17:23.36 | slobberknocker | hang tight... |
| 17:23.42 | MstlyHrmls | slobberknocker: does it pick up the rest of the config? does it register against the server correctly? |
| 17:23.48 | slobberknocker | yes |
| 17:24.31 | MstlyHrmls | slobberknocker: what versions are you using? |
| 17:24.53 | CunningPike | slobberknocker: Here is ours: |
| 17:24.56 | slobberknocker | 1_6_5 |
| 17:25.10 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 17:25.10 | CunningPike | <PROTECTED> |
| 17:25.31 | slobberknocker | and that sets your phones right? |
| 17:25.33 | [TK]D-Fender | slobberknocker : OverrideDHCP="1" |
| 17:25.43 | CunningPike | slobberknocker: Yes |
| 17:25.46 | slobberknocker | <SNTP tcpIpApp.sntp.resyncPeriod="86400" |
| 17:25.46 | slobberknocker | tcpIpApp.sntp.address="10.0.1.188" tcpIpApp.sntp.address.overrideDHCP="1" tcpIpApp.sntp.gmtOffset="" |
| 17:25.46 | slobberknocker | tcpIpApp.sntp.gmtOffset.overrideDHCP="1" |
| 17:25.52 | CunningPike | slobberknocker: Eventually ;) |
| 17:25.57 | slobberknocker | i just took out the gmtoffset to test... |
| 17:26.02 | slobberknocker | but that is not the issue |
| 17:26.03 | Persilon | CunningPike: If I add the insecure=very to the voip declaration in sip.conf I don't get that error anymore (but I still can't get the call :P) |
| 17:26.32 | [TK]D-Fender | slobberknocker : So what is serving up SNTP at that address? |
| 17:26.38 | MstlyHrmls | slobberknocker: what snmp server are you using? |
| 17:26.50 | slobberknocker | eventually may be the key phrase.... let me try that. cuz now it is set, all but offset |
| 17:26.55 | slobberknocker | internal sntp server |
| 17:27.00 | CunningPike | Persilon: Hmmm - what message do you get then? Anything in the CLI |
| 17:27.05 | slobberknocker | asterisk server has ntp server running on it |
| 17:27.27 | [TK]D-Fender | slobberknocker : I might suggest just using "pool.ntp.org" for this and leaving it in your DHCP settings. |
| 17:27.38 | Persilon | CunningPike: nope, I get nothing in the CLI (like the call is not being made)... but if I change it to insecure=yes or erase it, I get the same old error |
| 17:28.43 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 17:28.43 | CunningPike | Persilon: You might try a 'sip debug' with insecure=very and see what you et |
| 17:28.48 | CunningPike | s/et/get/ |
| 17:31.12 | Persilon | CunningPike: Mmmhh... I'm getting Transmitting (NAT) to 200.69.195.49:5060: SIP/2.0 404 Not Found |
| 17:32.45 | CunningPike | Persilon: What's the SIP destination for that message |
| 17:33.23 | [TK]D-Fender | Persilon : # isn't valid. |
| 17:33.35 | Persilon | CunningPike: I'm using a macro that works locally |
| 17:33.44 | L|NUX | [TK]D-Fender : bro can you help me with enum :( |
| 17:33.53 | [TK]D-Fender | L|NUX : Nope, never touched |
| 17:35.34 | Persilon | CunningPike: I changed it to MusicOnHold() but the calls gots cut after 4 rings |
| 17:35.43 | Persilon | [TK]D-Fender: changed it to MusicOnHold() but the calls gots cut after 4 rings |
| 17:36.11 | [TK]D-Fender | Persilon : Please pastebin your sip.conf and extensions.conf. |
| 17:36.13 | [TK]D-Fender | ~pb |
| 17:36.15 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 17:36.40 | Persilon | [TK]D-Fender: sip.conf: http://pastebin.com/749370 |
| 17:36.48 | MikeJ[Laptop] | ~Hmmhesays |
| 17:37.42 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 17:38.01 | Persilon | [TK]D-Fender: In [inbound] in extensions.conf the only things that I have (about this) are exten => _number,1,Answer() \n exten=> _number,2,MusicOnHold() |
| 17:41.29 | *** part/#asterisk Tier_1 (n=Tier@c-24-9-75-234.hsd1.co.comcast.net) |
| 17:46.20 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 17:48.45 | Persilon | [TK]D-Fender: I'm trying with different combinations, but I get |
| 17:48.46 | Persilon | Reliably Transmitting (NAT) to 200.69.195.49:5060: |
| 17:48.46 | Persilon | SIP/2.0 404 Not Found |
| 17:49.06 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 17:51.26 | *** join/#asterisk kodok (n=freeee@cm73.omega100.maxonline.com.sg) |
| 17:51.41 | kodok | hi all, does asterisk support H323 over G723 codec ? |
| 17:51.49 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 17:52.15 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 17:53.08 | Vorondil | hi all, quick question: is it possible to have asterisk delete a voicemail message via email? i.e. - reply to the voicemail notification with the word "delete" or something. |
| 17:53.08 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 17:54.11 | file | Vorondil: if you code it... |
| 17:54.28 | Nugget | sounds like a five line procmail recipe to me. |
| 17:55.01 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:56.10 | Hmmhesays | ja |
| 17:56.16 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.79.188) |
| 17:56.22 | file | ja? |
| 17:56.23 | file | ja, |
| 17:56.25 | file | ja! |
| 17:57.07 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
| 17:59.56 | Vorondil | file: hmm, define "code" ;) |
| 18:00.02 | L|NUX | ja |
| 18:00.04 | L|NUX | O_o |
| 18:00.10 | file | code, like... code |
| 18:00.14 | file | not exactly code - but like it |
| 18:00.37 | L|NUX | file : can you help me with enum |
| 18:00.37 | L|NUX | :$ |
| 18:00.37 | file | ...no |
| 18:00.38 | L|NUX | okay |
| 18:00.59 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 18:01.02 | L|NUX | bye every one |
| 18:02.10 | Vorondil | file: well, i suppose i'm asking if it's "code" as in "code a patch for asterisk" vs "code" as in "code a script for an imap server somewhere to do it" |
| 18:03.37 | salviadud | geez, what a slow da |
| 18:03.38 | salviadud | day |
| 18:03.46 | kodok | is there any "complete" guide to install H323 ? |
| 18:07.37 | *** join/#asterisk aze_ (n=aze@ACayenne-101-1-4-244.w81-248.abo.wanadoo.fr) |
| 18:07.39 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 18:07.49 | bkw_ | kodok, install woomera |
| 18:07.53 | [TK]D-Fender | file : Hey, in * 1.2.8 they fixed that DTMF Queue issue I mentioned to you :) |
| 18:08.01 | file | sweet |
| 18:08.16 | [TK]D-Fender | Vorondil : It'd be a mail server script. |
| 18:09.07 | Dr-Linux | awww my doctor just said, i can't go to work, i need 1 week rest :S |
| 18:09.41 | Vorondil | [TK]D-Fender: i gotcha, i figured it shouldn't be too involved a set up |
| 18:09.42 | mitcheloc | Dr-Linux: you are a doctor.... |
| 18:09.47 | Vorondil | [TK]D-Fender: thanks. :D |
| 18:09.59 | [TK]D-Fender | mitcheloc : Physician, heal thyself? |
| 18:10.39 | Dr-Linux | mitcheloc: but my this leg issue is out of my mind :) |
| 18:10.48 | *** join/#asterisk SimmerMoarn (n=SimmerMo@www3.datarack.nl) |
| 18:11.21 | mitcheloc | if i was a doctor i'd right notes to get myself off of work =)...then again I'd be working for myself... |
| 18:11.32 | [TK]D-Fender | Vorondil : Ummm.. yeah, it can be tricky... hard to pin down which VM file matches. Lets say its MSG0001. WELL HMM!!! Guy goes into the VM box by phone and wipes it out. Then another one comes in and takes its place numerically, then you reply to the message, it will delete the NEWLY ARRIVED one now. Thats the problem with msgs's not being serial #'d |
| 18:11.48 | [TK]D-Fender | mitcheloc : Thats called "conflict of interest" |
| 18:12.42 | mitcheloc | [TK]D-Fender: all i have to say is "Alto Mail" |
| 18:12.46 | mitcheloc | it's a prediction... |
| 18:13.14 | Vorondil | [TK]D-Fender: interesting, i didn't know they weren't |
| 18:13.26 | [TK]D-Fender | Vorondil : Go look in your voicemail folder.... |
| 18:14.01 | [TK]D-Fender | Vorondil : its very basic implementation. It works for normal stuff, but was not designed for real flexibility |
| 18:14.21 | [TK]D-Fender | Vorondil : Otherwise they'd serial # each and you could hard-link stuff. |
| 18:14.43 | mitcheloc | [TK]D-Fender: you can use "origtime" and "duration" to find it uniquely... |
| 18:15.41 | [TK]D-Fender | mitcheloc : An idea.... not solid, butperhaps realistically functional... though the complexity goes up. |
| 18:16.18 | *** join/#asterisk ToTo (n=ToTo@host216-145.pool874.interbusiness.it) |
| 18:16.18 | mitcheloc | bleh, it's good enough heh |
| 18:16.25 | mitcheloc | Vorondil: what is your e-mail client? |
| 18:17.23 | Vorondil | [TK]D-Fender: ah, i see now |
| 18:17.54 | Vorondil | mitcheloc: why do you ask? |
| 18:18.21 | *** join/#asterisk lorinc (n=ang@caracas-1582.adsl.interware.hu) |
| 18:18.21 | mitcheloc | just curious ;) |
| 18:19.23 | Vorondil | well, i use mostly thunderbird at home and outlook at work |
| 18:20.04 | mitcheloc | okay, cool |
| 18:24.06 | mitcheloc | i'll figure this out.... |
| 18:26.03 | mitcheloc | [TK]D-Fender: okay...how about an MD5 hash? |
| 18:27.00 | *** join/#asterisk southtel (n=slester@c-67-191-211-17.hsd1.ga.comcast.net) |
| 18:28.13 | *** join/#asterisk japerry (n=japerry@216.231.51.208) |
| 18:28.27 | japerry | heya-- does anyone have experience here with the GXP-2000?? |
| 18:28.52 | japerry | It seems every time I reboot the phone, it looses custom settings--and the custom settings are commented out in the provisioning configuration file |
| 18:30.31 | *** join/#asterisk ajohnson (i=EternalK@VDSL-130-13-154-157.PHNX.QWEST.NET) |
| 18:30.46 | mitcheloc | [TK]D-Fender/Vorondil: an md5 sum looks like the right way to go, it works |
| 18:30.51 | *** part/#asterisk ajohnson (i=EternalK@VDSL-130-13-154-157.PHNX.QWEST.NET) |
| 18:34.52 | key2 | [TK]D-Fender: if I have a proxy, and I want asterisk to dial foo@bar.com trhough the proxy, how could I do that ? |
| 18:35.20 | key2 | [TK]D-Fender: since Dial(SIP/proxy/foo@bar.com) doesnt work |
| 18:40.01 | brimstone | s/doesnt/doesn't/ |
| 18:40.12 | docelmo | WHADUP!? |
| 18:42.27 | CunningPike | Vorondil: We posted a bounty for that functionality.......... |
| 18:43.38 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 18:43.54 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 18:45.26 | Vorondil | CunningPike: yeah, i found that a minute ago. it's at like $3K |
| 18:45.40 | Vorondil | but the latest message/link is from november of last year =/ |
| 18:45.41 | mitcheloc | CunningPike: which one are you on the bounty list? |
| 18:46.13 | CunningPike | Yup - we had a programmer lined up, but kpfleming has said that there is another project afoot to do it and he won't accept any commits for it |
| 18:46.22 | CunningPike | That was months ago and no progress |
| 18:46.32 | CunningPike | mitcheloc: District of North Vancouver |
| 18:46.48 | mitcheloc | CunningPike: i'll pm you |
| 18:46.50 | file | progress on what? |
| 18:46.58 | Vorondil | CunningPike: that's too bad |
| 18:47.04 | CunningPike | file: IMAP integration with voicemail |
| 18:47.14 | *** join/#asterisk CoffeeIV (i=rgr@cpe-70-112-100-20.austin.res.rr.com) |
| 18:47.18 | file | oh you mean the thing that has a branch on the SVN server and is being brought up to date with trunk? |
| 18:47.54 | *** join/#asterisk adker (n=adker@74-33-212-148.br1.glv.ny.frontiernet.net) |
| 18:48.09 | CunningPike | file: The thing that has a branch that hasn't been touched..... :) |
| 18:48.16 | trelane_ | file++ |
| 18:48.22 | *** part/#asterisk amdtech (n=amd011@ab1-1-246.shsu.edu) |
| 18:48.27 | CoffeeIV | I have a SIP channel that is stuck as "busy" for somereason. How can I force it to reset or hangup or whatever from the *CLI> ? |
| 18:48.31 | file | that's because it's based on 1.2, but it's being brought up to trunk as we speak |
| 18:48.54 | Dr-Linux | CoffeeIV: soft hangup <channel> |
| 18:49.00 | CoffeeIV | thanks |
| 18:49.28 | Dr-Linux | CoffeeIV: the reason was network disconnection. |
| 18:49.28 | CoffeeIV | at |
| 18:49.40 | russellb | that $3k is owed to the university of pennsylvania, heh |
| 18:49.54 | key2 | russellb: what $3k |
| 18:50.25 | Dr-Linux | pennsylvania << is not bad word in english? |
| 18:50.35 | CunningPike | file: This apparently is the branch: http://lists.digium.com/pipermail/asterisk-commits/2006-April/002996.html |
| 18:50.37 | russellb | no |
| 18:50.38 | Dr-Linux | ~pennsylvania |
| 18:50.46 | KranZ | that must have been for channel #moneyowedtotheuniversity |
| 18:51.03 | russellb | no, they are the ones that funded the IMAP-voicemail code |
| 18:51.10 | mishehu | hmm... with ael, don't included sections take precedence? I have a context named foo with 95551212 => Congestion(); which is included in another context. When a user bound to the context bar that includes foo calls 95551212, the _9NXXXXXX => { ... }; is actually called up instead of the exact match. |
| 18:51.14 | Dr-Linux | ~dict pennsylvania |
| 18:51.28 | CunningPike | russellb: Where is that at? |
| 18:51.43 | russellb | CunningPike: in a branch right now, getting updated to trunk as we speak to be included for the 1.4 release |
| 18:51.46 | KranZ | ~texas |
| 18:51.48 | jbot | hmm... texas is a fine place for y'all to have a big ol' time |
| 18:51.59 | Strom_C | ~california |
| 18:52.01 | jbot | somebody said california was Psychic spies from China try to steal your mind's elation |
| 18:52.08 | CunningPike | russellb: Good news |
| 18:52.09 | mishehu | the same goes for _91NXXNXXXXXX => { ... }; taking precendence over th eincluded _91NXX5551212 => Congestion(); one. |
| 18:52.16 | Strom_C | that definition must have been written in Berkeley |
| 18:52.18 | russellb | in other news, we got the generic jitterbuffer merged today, i'm excited about that |
| 18:52.22 | file | indeed |
| 18:52.28 | file | so no more complaining about it not being in there |
| 18:52.33 | russellb | no more! |
| 18:53.27 | key2 | russellb: while you're here, would you tell me how I could Dial "foo@bar.com" using a SIP proxy ? since Dial(SIP/proxy/foo@bar.com) doesnt work |
| 18:53.28 | key2 | ? |
| 18:54.06 | CunningPike | russellb: Will there be a 1.2 patch for the IMAP feature then? Or do we need to wait for 1.4? |
| 18:54.22 | russellb | 1.4 |
| 18:54.28 | key2 | CunningPike: 1.4 comes out tomo |
| 18:54.33 | russellb | unless you want to apply the patch yourself |
| 18:54.40 | russellb | which you could pull out of svn |
| 18:54.45 | CunningPike | russellb: We just might........ |
| 18:54.48 | key2 | russellb: no answers ? |
| 18:54.55 | CunningPike | russellb: Patch ID? |
| 18:55.05 | russellb | it's not in the bug tracker |
| 18:55.21 | mitcheloc | russellb: how does the imap server work? what username/password do you login with? is it a seperate e-mail "account"? |
| 18:55.24 | distortion | ~1.4 |
| 18:55.31 | russellb | i have no idea how it works. |
| 18:55.32 | distortion | doh |
| 18:55.41 | russellb | ask mog_work |
| 18:55.43 | russellb | :-p |
| 18:55.44 | key2 | mog_work :) |
| 18:55.50 | Dr-Linux | if 1.4 beta has no new features, then what's special in it? :S |
| 18:55.57 | CunningPike | russellb: How can I find it in SVN? |
| 18:56.12 | russellb | i don't know :) |
| 18:56.21 | russellb | it's time for lunch, that's all i knwo |
| 18:56.37 | file | who said 1.4 beta had no new features? |
| 18:56.44 | russellb | file: nubs |
| 18:56.47 | key2 | file: it has jabber |
| 18:56.51 | Strom_C | nub++ |
| 18:56.59 | key2 | key2++ |
| 18:56.59 | mitcheloc | jabber who? |
| 18:57.07 | mitcheloc | key2: links!!! |
| 18:57.16 | key2 | svn |
| 18:57.17 | russellb | jingle what? |
| 18:57.22 | *** join/#asterisk adker_ (n=adker@67.136.214.38) |
| 18:57.26 | Dr-Linux | file: what's main new features in 1.4 beta? |
| 18:57.32 | russellb | it now makes breakfast |
| 18:57.42 | key2 | Dr-Linux: changelog |
| 18:57.44 | file | we don't have a list made yet, that'll be done for the release... |
| 18:57.49 | file | so look at the commit list if you really want to know |
| 18:58.03 | mitcheloc | file: can you give me a teaser at least as to what you mean by jabber? |
| 18:58.13 | russellb | diff -u asterisk/1.2 asterisk/trunk |
| 18:58.16 | CunningPike | How do I locate a specific patch in SVN? |
| 18:58.27 | key2 | file: would you tell me how I could Dial "foo@bar.com" using a SIP proxy ? since Dial(SIP/proxy/foo@bar.com) doesnt work ? |
| 18:58.38 | key2 | russellb: thx :) |
| 18:59.04 | file | I'm under attttttttack |
| 18:59.21 | key2 | file: gimme the answer before u die then :) |
| 18:59.23 | mitcheloc | file: come on! |
| 18:59.46 | file | hey you two - I'm working |
| 18:59.53 | key2 | mitcheloc: sending msg to jabber |
| 19:00.31 | CunningPike | russellb: Do you know where I can find the IMAP patch in SVN? |
| 19:00.41 | file | patch? it's a branch |
| 19:01.38 | key2 | file: could you gimme the answer of my previous question plz |
| 19:01.39 | CunningPike | file: Patch, branch, whatever |
| 19:01.59 | file | key2: ...no |
| 19:02.09 | file | not right now... |
| 19:02.20 | file | CunningPike: http://svn.digium.com/svn/asterisk/team/jrothenberger/asterisk-imap |
| 19:02.49 | CunningPike | file: Thanks - that's the one I'm looking at, but there's been no updates ;( |
| 19:03.13 | file | what do you mean, no updates... |
| 19:04.23 | mitcheloc | file: i looked at the jabber.h and chan_jingle.c code...jabber features are *only* for jingle right? |
| 19:04.35 | file | ask mog, he wrote it! |
| 19:04.35 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
| 19:05.46 | CunningPike | file: All the files say 'Original copy' |
| 19:06.01 | CunningPike | s/Original copy/Initial copy/ |
| 19:06.15 | mitcheloc | CunningPike: i'm asking mog right now, when he lets me know, i'll tell you |
| 19:06.40 | CunningPike | mitcheloc: Great - I'd appreciate it :) |
| 19:07.05 | file | CunningPike: eh? no they don't :) it was just brought up to date with 1.2 by mog, and the app_voicemail.c contains the IMAP stuff |
| 19:07.41 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 19:07.44 | key2 | file: why not ? |
| 19:08.16 | file | because I'm holding 5 conversations, tracking down a memory leak, looking over some code, and configuring something |
| 19:08.27 | mitcheloc | i have no idea how mark has so much patience.... app_voicemail.c is several thousand lines long... |
| 19:08.38 | sevard | dlynes is rewriting that |
| 19:08.49 | sevard | app_voicemail2.c |
| 19:08.55 | sevard | voicemail without the bloat |
| 19:09.26 | CunningPike | file: It helps if I strip the version specific part of the URL off |
| 19:09.34 | file | :D |
| 19:10.07 | *** join/#asterisk TripleFFFF (n=Miranda@147-102.mc.cite.net) |
| 19:10.19 | mitcheloc | the *only* part about the imap stuff that sounds silly to me is that the client has to have imap supprot |
| 19:10.19 | *** join/#asterisk reza_ (n=reza@64-142-78-129.dsl.static.sonic.net) |
| 19:10.20 | mitcheloc | * support |
| 19:10.24 | TripleFFFF | any extensions.conf to realtime converter ? |
| 19:11.07 | reza_ | quick questin - how do i add an extension to the context? |
| 19:11.09 | TripleFFFF | mitcheloc i got a PC wireless security module and writing a DND function for asterisk |
| 19:11.25 | mitcheloc | TripleFFFF: you could use this... trig_hm: you c |
| 19:11.29 | mitcheloc | ** http://www.columbia.edu/kermit/vt320-keyboard-700.jpg |
| 19:11.40 | mitcheloc | ---^ that will convert it really well |
| 19:11.41 | TripleFFFF | meaning.. you move further then 5 foot form computer. screen save locks computer and sens a DND to asterisk@homme asterisk to says phone is not avail |
| 19:11.47 | reza_ | as in "Don't forget to include the apps extension in your context. " |
| 19:12.02 | TripleFFFF | wow thats old |
| 19:12.20 | TripleFFFF | any extensions.conf to realtime converter ? |
| 19:12.23 | TripleFFFF | or i need to write one |
| 19:13.26 | TripleFFFF | inverse of retrieve_sip_conf_from_mysql.pl |
| 19:13.33 | TripleFFFF | i mean |
| 19:13.36 | TripleFFFF | inverse of retrieve_extensions_from_mysql.pl |
| 19:14.29 | *** join/#asterisk techie (n=gus@marketingleaders.net) |
| 19:16.51 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 19:22.05 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 19:23.00 | *** join/#asterisk adker (n=adker@67-136-222-125.dsl1.glv.ny.frontiernet.net) |
| 19:23.06 | C4T3l | ~seen c4t3l |
| 19:23.23 | jbot | c4t3l is currently on #asterisk (4h 57m 33s). Has said a total of 3 messages. Is idling for 17s, last said: '~seen c4t3l'. |
| 19:23.23 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 19:23.23 | tzafrir_home | reza_, you can add extentions temporarily from the CLI . write 'add' and press some tabs |
| 19:24.01 | CunningPike | C4T3l: Talking to yourself again? |
| 19:24.03 | tzafrir_home | But this is not for a permanent addition. Basically just edit the relevant part in extensions.conf and run 'extensions reload' |
| 19:24.06 | CunningPike | :) |
| 19:24.13 | C4T3l | :) |
| 19:24.31 | reza_ | tzafrir_home thanks, but i think i figured it out, i added "include => apps" to the beginning of my default context -- is that what i was supposed to do? |
| 19:24.47 | reza_ | i'm trying to get call forwarding setup |
| 19:24.50 | reza_ | but i'm failing |
| 19:25.12 | Dr-Linux | tzafrir_home: what do you mean temporarily? this extension will be there till next reload or till asterisk restart? |
| 19:25.14 | tzafrir_home | reza_, I suggest some basic reading regarding extensions.conf |
| 19:25.26 | Dr-Linux | or untill manually dell |
| 19:25.29 | Dr-Linux | del* |
| 19:25.30 | tzafrir_home | they will be there until reload, I believe |
| 19:25.58 | Dr-Linux | tzafrir_home: :S , hhm.. it's mean i was told wrong :S |
| 19:26.14 | tzafrir_home | Dr-Linux, actually, I never tried it |
| 19:26.20 | C4T3l | how difficult is it to do one touch recording really?? |
| 19:26.38 | reza_ | strange, i had to reboot for the music on hold to work |
| 19:26.43 | Dr-Linux | tzafrir_home: someone told me here, that will there in DB untill you donot del it from CLI |
| 19:26.49 | Dr-Linux | remove* |
| 19:27.16 | reza_ | tzafrir - is it not include=> ? |
| 19:28.14 | tzafrir_home | reza_, have you read some basic docs about extensions.conf, e.g: http://voip-info.org/wiki/view/Asterisk+config+extensions.conf ? |
| 19:28.28 | reza_ | woah |
| 19:28.30 | reza_ | crazy loop |
| 19:28.53 | tzafrir_home | reza_, 'include' adds the content of another context into the current context, basically |
| 19:28.54 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 19:31.00 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 19:31.03 | reza_ | which is what i wanted, thanks |
| 19:32.05 | reza_ | WARNING[17036]: pbx.c:1556 pbx_extension_helper: No application 'StripLSD' for extension ... <- huh |
| 19:36.27 | reza_ | can someone tell me what this command does : exten => _*21*X.,2,GotoIf($${EXTEN:-1} = #?3:4) |
| 19:37.17 | reza_ | something like if the last digit of the extension is a variable then to go to 3 else 4 |
| 19:37.20 | reza_ | oh |
| 19:37.24 | reza_ | if it's equal to a pund |
| 19:37.26 | reza_ | pound |
| 19:37.26 | TripleFFFF | wow |
| 19:37.40 | TripleFFFF | so you can test with add on the cli and make them go away on a extension reload |
| 19:37.41 | TripleFFFF | lol |
| 19:37.53 | TripleFFFF | why is my pound sign not doing company directory |
| 19:38.01 | reza_ | it doesnt like you ;) |
| 19:38.36 | Dr-Linux | TripleFFFF: replace the pound sign or your finger :) |
| 19:39.16 | diLLec | hey guys. |
| 19:39.20 | reza_ | can i simulate a number dialed on the cli? |
| 19:39.20 | TripleFFFF | lol |
| 19:39.32 | diLLec | is there any way to get app_queue working in real roundrobin Method ? |
| 19:39.41 | mog_work | anyone know easy way to find grandstream atas on a busy network |
| 19:39.51 | Hmmhesays | someone make an iso of the ps2 game flatout and send it to me |
| 19:40.00 | diLLec | so that all calls will be distributed in the way 1st agent-> 2nd agent -> 3rd agent |
| 19:40.22 | reza_ | Hmmhesays - sure, send me your mailing address and phone number |
| 19:40.52 | Hmmhesays | heh, you've probably never even played it |
| 19:41.42 | reza_ | dude, why is it so hard to get call forwarding working |
| 19:41.46 | reza_ | now DBput isn't working |
| 19:41.47 | Hmmhesays | its not |
| 19:41.48 | Dr-Linux | diLLec: so what's your question about queues? roundrobin works fine though |
| 19:42.26 | Hmmhesays | DBput is deprecated |
| 19:42.31 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 19:42.35 | *** join/#asterisk \lart (i=nobody@c-69-249-148-237.hsd1.nj.comcast.net) |
| 19:42.42 | \lart | good day all.. |
| 19:43.07 | reza_ | what's replaced it? |
| 19:43.07 | diLLec | Dr-Linux |
| 19:43.10 | [TK]D-Fender | [15:36] <reza_> can someone tell me what this command does : exten => _*21*X.,2,GotoIf($${EXTEN:-1} = #?3:4) |
| 19:43.23 | reza_ | TK - i figured it out |
| 19:43.26 | [TK]D-Fender | reza_ : that line is very incorrectly formatted |
| 19:43.26 | Dr-Linux | file: why there is no option "longest idle" in the queue? |
| 19:43.32 | diLLec | it works find but distributes first call 1->2->3 then 2->3->1 then 3->1->2 |
| 19:43.35 | reza_ | i didnt write it, it's in a faq |
| 19:43.42 | *** join/#asterisk TonyM (n=TonyM@softins.claranet.co.uk) |
| 19:43.48 | diLLec | i want it always to distribute it like 1->2-> |
| 19:43.49 | diLLec | 3 |
| 19:43.55 | [TK]D-Fender | Dr-Linux : There is, its called LEASTRECENT |
| 19:43.58 | \lart | occasionally, I get an automated incoming call that requires I push the # key. Of course, my *@home (2.8 if it matters) installation thinks it should initiate a call xfer. Any thoughts? |
| 19:44.10 | reza_ | it's to strip of an extra # sign, i didn need it so i removed it |
| 19:44.20 | [TK]D-Fender | \lart : please read the channel topic |
| 19:44.24 | reza_ | i'm looking at the DBput page on the wiki -- no mention of what function has replaced it |
| 19:44.40 | file | why do birds suddenly appear |
| 19:44.43 | [TK]D-Fender | \lart : Second thats because you enabled # as a transfer key on the PHONE side of that call.... |
| 19:44.52 | Strom_C | every time chocolate cake is near |
| 19:45.01 | [TK]D-Fender | reza_ : Lookup "functions" on the Wiki. |
| 19:45.20 | \lart | I would have asked in #freepbx, but it seemed more an asterisk question, than an amp question. |
| 19:45.31 | \lart | i'll check the phone settings, but i don't recall that one. |
| 19:45.36 | *** part/#asterisk TonyM (n=TonyM@softins.claranet.co.uk) |
| 19:45.39 | [TK]D-Fender | \lart : Very wrong since AMP controls your * dialplan. |
| 19:46.03 | [TK]D-Fender | \lart : and that part is deeply imbedded in the diaparties.agi |
| 19:46.12 | \lart | then i seem to have been mistaken. |
| 19:46.19 | *** part/#asterisk \lart (i=nobody@c-69-249-148-237.hsd1.nj.comcast.net) |
| 19:47.06 | reza_ | i give up for now. all the examples i look at seem to be depreciated. grr. |
| 19:47.18 | [TK]D-Fender | reza_ : pastebin what you've created so far |
| 19:47.21 | [TK]D-Fender | ~pb |
| 19:47.23 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 19:47.48 | CunningPike | reza_: I don't believe so - which is why I have a softphone on my computer........ |
| 19:48.34 | *** join/#asterisk MGSsancho (n=user@adsl-68-120-229-58.dsl.irvnca.pacbell.net) |
| 19:52.04 | *** join/#asterisk fourcheeze (n=rich@82.153.23.79) |
| 19:52.31 | fourcheeze | What could stop ringing tone being passed back through local channels? |
| 19:54.32 | *** join/#asterisk tsurk0 (n=tsurko@digsys226-159.pip.digsys.bg) |
| 19:54.44 | *** join/#asterisk brif8 (n=Administ@lazyjtrainingcenter.com) |
| 19:54.53 | fourcheeze | I have a few contexts I use for routing different calls and I use local channels to pass around |
| 19:55.06 | fourcheeze | and it seems by the end of it I get no ringing tone in some circumstances |
| 19:55.24 | fourcheeze | trying to pin down what that can be |
| 19:55.36 | fourcheeze | does /n on a local channel make a difference with this? |
| 19:55.55 | fourcheeze | this is * 1.2.4 |
| 19:56.01 | brif8 | I have * running but trying to connect an IP phone using SIP that is not on my LAN but is to use my * server. I have it in sip.conf with host = ip.address but registration is timing out and failing ? |
| 19:56.45 | fourcheeze | brif8: is your * behind a NAT? |
| 19:57.02 | CunningPike | diLLec: Check out roundrobin with memory |
| 19:57.47 | brif8 | fourcheeze: yes but the firewall is open and directs directly to the * box for ports 5060 and 10000:20000 |
| 19:58.30 | brif8 | directs -> forwards |
| 19:58.36 | key2 | [TK]D-Fender: would you tell me how I could Dial "foo@bar.com" using a SIP proxy ? since Dial(SIP/proxy/foo@bar.com) doesnt work ? |
| 19:58.58 | fourcheeze | brif8: you want host=dynamic I think |
| 19:59.00 | `Sauron | Anyone have any idea why ld would try to link against a static library instead of against a shared library when -static is _not_ used... ? |
| 19:59.08 | fourcheeze | brif8: otherwise you wouldn't need to register |
| 19:59.39 | brif8 | fourcheeze: for I need a register => also for the phone |
| 20:00.05 | fourcheeze | you can't normally register with a phone |
| 20:01.20 | brif8 | ok so I have [8000] in sip.conf and host dynamic |
| 20:01.34 | *** join/#asterisk BZBW (n=wlwzhang@ip67-153-142-109.z142-153-67.customer.algx.net) |
| 20:03.13 | BZBW | emm, it's quiet... |
| 20:03.48 | fourcheeze | brif8: yeah get the put a username and secret in there |
| 20:03.54 | fourcheeze | brif8: get the phone to register |
| 20:04.48 | brif8 | I have I'm gettting this http://pastebin.com/749752 |
| 20:04.57 | BZBW | anyone know why we need features.conf if we can put whatever in features.conf into extensions.conf? |
| 20:05.38 | mercestes | Same reason we need the "include" statement in C++ when we can put everything we need in one monster ass sourecode file. |
| 20:05.46 | brif8 | http://pastebin.com/749760 has the sip.conf for 8000 also |
| 20:06.10 | fourcheeze | brif8: is the other phone behind nat? |
| 20:06.28 | TripleFFFF | 46 register_verify: No registration for peer 'register => blah' |
| 20:06.30 | BZBW | mercestes: so whatever I put in features.conf should work in extensions.conf as well? |
| 20:06.34 | TripleFFFF | hmm bad config i asume |
| 20:06.44 | brif8 | yes and also the firewall port forward 5060 and 10000:20000 directly to the phone |
| 20:06.46 | TripleFFFF | else would show .. No registration for peer 'blah' |
| 20:06.47 | TripleFFFF | ? |
| 20:07.23 | fourcheeze | brif8: you don't need to do that |
| 20:07.31 | fourcheeze | brif8: you just want to call the phone, right? |
| 20:07.39 | *** join/#asterisk fholmes (n=fholmes@rrcs-24-227-237-197.sw.biz.rr.com) |
| 20:07.50 | fourcheeze | brif8: what's in your dial command? |
| 20:07.54 | brif8 | fourcheeze: call the phone from * yes and have the phone make calls via * |
| 20:08.12 | *** join/#asterisk CodyC (n=cody@207.200.23.194) |
| 20:08.32 | fourcheeze | brif8: just give the phone an ID let it register and Dial(id) |
| 20:08.54 | brif8 | exten => 88,1,dial(SIP/8000,90,r) |
| 20:09.11 | fourcheeze | and 8000 is registered? |
| 20:09.11 | TripleFFFF | aint the register string this ? register => blah:pass@host ? |
| 20:09.38 | brif8 | 8000/8000 PHONE.PUBLIC.IP D 5060 Unmonitored |
| 20:09.57 | brif8 | but when I dial 8000 is get the person is unavailable |
| 20:10.03 | brif8 | message |
| 20:10.12 | fourcheeze | brif8: add qualify=yes to the user |
| 20:10.19 | TripleFFFF | and i get alot odf res_features.c:844 builtin_atxfer: Did not read data. |
| 20:10.31 | brif8 | it does already have qualify =yes |
| 20:10.48 | fourcheeze | sholdn't be unmonitored then |
| 20:11.14 | brif8 | http://pastebin.com/749760 line 35 |
| 20:11.29 | reza_ | how do you delete a key from the DB in v1.2? |
| 20:11.31 | brif8 | right which tells me even though it has the IP it is not connected. |
| 20:11.42 | fourcheeze | brif8: typo |
| 20:11.50 | fourcheeze | s/quality/qualify |
| 20:12.05 | brif8 | ooops :( :( |
| 20:12.31 | brif8 | 8000/8000 PHONE.PUBLIC.IP D 5060 OK (51 ms) |
| 20:12.35 | fourcheeze | aha |
| 20:12.36 | fourcheeze | ok |
| 20:12.44 | brif8 | thanks better now why when I dial 8000 the phone doesn't ring ? |
| 20:12.53 | fourcheeze | dunno :( |
| 20:13.14 | fourcheeze | brif8: make the console very verbose |
| 20:13.18 | *** join/#asterisk lorinc (n=ang@caracas-0125.adsl.interware.hu) |
| 20:13.27 | mercestes | set verbose 99 |
| 20:13.30 | mercestes | >.> |
| 20:14.01 | brif8 | Called 8000 |
| 20:14.13 | brif8 | Got SIP response 410 "Gone" back from PHONE.PUBLIC.IP |
| 20:14.25 | fourcheeze | aha |
| 20:14.30 | fourcheeze | it's the phone's fault |
| 20:14.35 | TripleFFFF | guys ? |
| 20:14.44 | brif8 | what do I look for or change on the phone then ? |
| 20:14.59 | fourcheeze | what sort of phone? |
| 20:15.05 | brif8 | siptura ATA |
| 20:15.24 | fourcheeze | hmm |
| 20:18.15 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 20:18.55 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 20:20.36 | *** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 20:20.40 | fourcheeze | brif8: not sure why that's happening |
| 20:23.09 | CodyC | an agent will be on a call, and (I'm assuming because he's a member of multiple queues) will receive and incoming call while on the phone |
| 20:23.18 | CodyC | anyone know how to prevent this? |
| 20:25.12 | brif8 | fourcheeze: the phone siptura web console shows that it is registered |
| 20:25.29 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) [NETSPLIT VICTIM] |
| 20:26.45 | reza_ | can i do this : http://pastebin.com/749827 |
| 20:27.02 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
| 20:27.23 | [TK]D-Fender | CodyC : Reduce the number of lines and CW ability on the phone to 1. Either that or use the Local channel to dial an exten that will check before dialing so as not to interrupt them. |
| 20:28.00 | [TK]D-Fender | reza_ : Close..... your first line is pulling a FIXED famil/key however |
| 20:28.16 | reza_ | i want it to be global |
| 20:28.19 | fourcheeze | brif8: can the phone make a call? |
| 20:28.21 | [TK]D-Fender | reza_ : Is that really your intent? |
| 20:28.26 | reza_ | i just tried dialing the number adn got this : -- Executing GotoIf("SIP/213.246.216.81-08941b80", "0 ? 5:3") in new stack |
| 20:28.31 | [TK]D-Fender | reza_ : Ok, I"m looking at the rest now then |
| 20:28.47 | bmg505 | HI ALL |
| 20:28.47 | bmg505 | soz caps was on |
| 20:28.48 | bmg505 | does anybody here use freepbx to make outgoing calls, I have a customer that wants to use freepbx, and it was an ordeal to get sip in calls to work |
| 20:28.53 | bmg505 | outgoing sip calls fro soft phones |
| 20:28.55 | reza_ | tk - thanks |
| 20:28.55 | bmg505 | s/fro/from/ |
| 20:29.38 | reza_ | i would rather be able to delete CF/NUM and test to see if that key exists or not in the db, though i can't find the docs on how to delete a key fromthe db |
| 20:29.51 | brif8 | fourcheeze: trying and reading debug |
| 20:30.36 | [TK]D-Fender | reza_ : Here's the fixed version : http://pastebin.com/749841 |
| 20:30.50 | brif8 | fourcheeze: yes obisouly needed time to find each other correctly :) Thanks you |
| 20:30.57 | *** part/#asterisk FuRR_ (i=[U2FsdGV@adsl-69-108-94-232.dsl.irvnca.pacbell.net) |
| 20:31.30 | [TK]D-Fender | reza_ : DBDel is not deprecated like DBPut and DBGet. You'd also want to use EXIST_DB() (or whatever its called to verify for sure |
| 20:31.50 | CodyC | fender: they commonly need to put a person on hold, then call a support number and get things fixed before returning to the caller |
| 20:32.08 | reza_ | thanks, will try |
| 20:32.09 | [TK]D-Fender | reza_ : I removed the "t" exten from that context because nothingin there can cause it to be called... |
| 20:33.14 | TripleFFFF | why do i get han_iax2.c:5046 register_verify: No registration for peer 'register => popo' |
| 20:33.20 | TripleFFFF | hes in the realtim |
| 20:33.24 | TripleFFFF | i got other clients there all o |
| 20:33.25 | TripleFFFF | kk |
| 20:33.33 | [TK]D-Fender | bmg505 : Please read the channel topic.... |
| 20:33.38 | dlynes_office | Is there a way to specify a username and a password in a call to Dial(), when Dialing a sip device? |
| 20:33.53 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-112-179.telkomadsl.co.za) |
| 20:33.56 | TripleFFFF | dial(sip/TEC/numbner) ? |
| 20:34.09 | dlynes_office | tec? |
| 20:34.11 | [TK]D-Fender | dlynes_home : Dial(SIPuser:pass@host) |
| 20:34.13 | TripleFFFF | so dial(SIP/USER:PASS@HOST/${EXTEN} |
| 20:34.19 | dlynes_office | [TK]D-Fender: thanks |
| 20:34.29 | TripleFFFF | or exten@blah |
| 20:34.31 | TripleFFFF | i think |
| 20:34.45 | [TK]D-Fender | dlynes_home : Dial(SIPuser:pass@host/exten) <- slightly better for assuming you have a target # in mind |
| 20:34.56 | dlynes_office | [TK]D-Fender: no spaces or anything in between SIP and user? |
| 20:35.04 | [TK]D-Fender | dlynes_home : nope |
| 20:35.08 | dlynes_office | [TK]D-Fender: ok |
| 20:35.16 | dlynes_office | TripleFFFF: the format you used is for iax, not sip |
| 20:35.27 | [TK]D-Fender | reza_ : Test fast, I'm heading home in 2 mins :) Should be on-line then |
| 20:36.18 | *** join/#asterisk mooey (n=chris@85-210-5-108.dsl.pipex.com) |
| 20:36.32 | reza_ | i can't find the db_exist entry |
| 20:36.37 | reza_ | there is an exists tag |
| 20:36.38 | reza_ | hmm |
| 20:36.40 | reza_ | er, function |
| 20:36.44 | TripleFFFF | yeah SIP is .. dial(SIP/${EXTEN}@USER:PASS@HOST) ? |
| 20:36.49 | mooey | hi guys, sometimes (over iax) asterisk dosen't notice when a user presses a button (dtmf), where should i look to diagnose this? |
| 20:37.10 | dlynes_office | TripleFFFF: that might be correct...dunno...but your earlier one was for iax |
| 20:37.14 | [TK]D-Fender | ok, heading home.... BBIAB |
| 20:37.16 | TripleFFFF | oh |
| 20:37.20 | TripleFFFF | well works anyhow |
| 20:37.43 | dlynes_office | TripleFFFF: i just know for sip Dial(SIP/${EXTEN}@host) works |
| 20:37.53 | dlynes_office | TripleFFFF: the exten has to be after the '/' |
| 20:38.01 | dlynes_office | :q |
| 20:38.15 | dlynes_office | bleh...not in vi :0 |
| 20:38.46 | dlynes_office | hrm...tk's solution doesnt' work, either |
| 20:38.52 | dlynes_office | i'll try your second suggestion |
| 20:40.11 | dlynes_office | nah...that doesn't work, either, but I do know why |
| 20:40.23 | *** part/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 20:41.27 | Dr-Linux | anybody using RHEL 4 ? |
| 20:41.31 | *** join/#asterisk loveman (n=ricklove@rrcs-67-79-4-58.sw.biz.rr.com) |
| 20:41.59 | brif8 | fourcheeze: cool thanks man :) |
| 20:42.51 | brif8 | how does one enable call forwarding per extension. say I have exten 200 but when I set the phone to DND it calls forwards to another number ? |
| 20:43.28 | TripleFFFF | scary im upgrading bios of laptop via a win biosupdte... darn cpu says 72 celc |
| 20:44.47 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-112-179.telkomadsl.co.za) |
| 20:45.44 | Dr-Linux | dlynes_home: today we are going to install new Dell PE 2850 dual core machine for asterisk. |
| 20:46.50 | Dr-Linux | dlynes_home: actually we were using RHEL AS 3 with with 2.4 kernel so far for all our servers. so i'm thinking to install RHEL AS 4 , 2.6 kernel in this machine, what you think? |
| 20:47.30 | *** join/#asterisk Katty (n=angela@64.82.232.54) |
| 20:47.33 | Katty | hi lads. |
| 20:47.45 | file | Katttttttttty |
| 20:49.37 | TripleFFFF | chan_iax2.c:5046 register_verify: No registration for peer 'register => bla' (from 1.2.3.4) |
| 20:49.42 | TripleFFFF | afain.. anyreason ? |
| 20:49.43 | Ariel_ | hello Katty long time |
| 20:49.48 | TripleFFFF | sip realtime and iax realtime all ok |
| 20:49.51 | TripleFFFF | others are ok |
| 20:49.53 | *** join/#asterisk arguile (n=arguile@66.38.201.234) |
| 20:49.54 | TripleFFFF | could it be password ? |
| 20:50.04 | bmg505 | when I dial out with kpsi to sip:8521@sip.lrlabs.com kpsi comes back with "Call Failed: Not Found" |
| 20:50.04 | bmg505 | I have tested with fwdnet and I can call and receive fwd calls |
| 20:50.05 | bmg505 | s/call/make/ |
| 20:50.09 | bmg505 | from fwd communicator I can call the sip addy which means my incoming sip path is ok |
| 20:51.32 | TripleFFFF | i see why USERNAME : register =>blah |
| 20:51.36 | TripleFFFF | in debug |
| 20:51.48 | TripleFFFF | means hes not using = but using => or -> ? |
| 20:52.10 | TripleFFFF | what format for iax.conf regiuster it changes alwyas... is it register = or register => or register -> |
| 20:52.50 | *** join/#asterisk bcnl (n=mike@S010600131078957c.vc.shawcable.net) |
| 20:53.45 | *** join/#asterisk pruonckk (n=pruonckk@wl4.starweb.com.br) |
| 20:53.50 | pruonckk | Hello |
| 20:53.56 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-112-179.telkomadsl.co.za) |
| 20:54.54 | pruonckk | im interested on learn about asterisk, somebody can say me a good doc to initiate |
| 20:55.40 | Ariel_ | ~docs |
| 20:55.42 | jbot | docs is, like, probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 20:56.47 | *** join/#asterisk MoutaPT (n=MoutaPT@85.139.196.147) |
| 20:56.56 | pruonckk | Ariel_, sorry, thanks |
| 20:57.51 | *** join/#asterisk bmg505 (n=leon@c1-233-15.rndf.isadsl.co.za) |
| 20:57.51 | *** join/#asterisk {zombie} (i=zombie@soulasylum.penguincare.com.au) |
| 20:57.53 | MoutaPT | hi, any one with experience with chan_misdn.so ? |
| 20:58.24 | *** join/#asterisk adorah (n=Asterjet@87.69.72.228) |
| 20:58.42 | TripleFFFF | never mind.. i found.. |
| 20:58.49 | TripleFFFF | user using freepbx crap |
| 20:59.13 | TripleFFFF | ~providers |
| 20:59.42 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 21:00.29 | *** part/#asterisk brif8 (n=Administ@lazyjtrainingcenter.com) |
| 21:00.45 | *** part/#asterisk CodyC (n=cody@207.200.23.194) |
| 21:02.52 | BZBW | hi, in my features.conf, I have "context => parkedcalls", what do I need to do in extensions.conf to link features.conf and extensions.conf? |
| 21:03.56 | BZBW | It does not seem to work even if I enable "xfersound = beep". |
| 21:08.07 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 21:11.26 | bkw_ | where did the cli originate command go? |
| 21:11.28 | bkw_ | in 1.2.8? |
| 21:12.02 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 21:12.48 | Corydon-w | AFAIK, the cli originate command is only in trunk |
| 21:18.20 | RoyK | <PROTECTED> |
| 21:19.06 | *** part/#asterisk BadPacket (n=root@unaffiliated/badpacket) |
| 21:23.15 | terrapen | is there a way |
| 21:23.21 | terrapen | err |
| 21:23.30 | terrapen | is there a way to kill a channel from the CLI |
| 21:23.45 | CunningPike | terrapen: soft hangup? |
| 21:24.16 | terrapen | yup |
| 21:25.22 | trixter | stop now will also kill the channel |
| 21:25.24 | trixter | :D |
| 21:25.53 | [TK]D-Fender | "shutdown -h now" |
| 21:25.58 | trixter | that works too |
| 21:26.12 | jsaunders | 3 finger salute |
| 21:26.23 | trixter | pull the power cord |
| 21:26.29 | trixter | ha |
| 21:26.32 | jsaunders | sledgehammer |
| 21:26.37 | CunningPike | terrapen: Is it a zap channel? |
| 21:26.58 | trixter | tere was an ibm mainframe years ago that when you hit the big red button it would physically sever the power cord - forcing you to call ibm any time you hit that button (which normally shouldnt be used unless someone is getting fried or a fire starts |
| 21:30.58 | *** join/#asterisk Delta239 (n=paparapa@201.226.130.55) |
| 21:35.09 | BZBW | anyone knows where I can find an example file for using "ParkAndAnnounce" in features.conf? WIKI does not help and features.conf.sample do not have such a feature:(. |
| 21:36.07 | CunningPike | BZBW: Ask me in about half an hour - I can look up what we do for you.... |
| 21:36.14 | CunningPike | BZBW: In a meeting right now |
| 21:37.21 | BZBW | CunningPike: thx. I'll wait till then, but thought someone has a quick answer. |
| 21:37.21 | Katty | Ariel_: yeah, it's been busy |
| 21:38.05 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 21:38.08 | CunningPike | BZBW: We're using Polycoms - are you? |
| 21:38.48 | [TK]D-Fender | Katty: Long time no see.... |
| 21:38.50 | BZBW | CunningPike: no, Grandstream |
| 21:39.15 | CunningPike | BZBW: Hmm - our solution is likely Polycom-specific |
| 21:39.36 | [TK]D-Fender | CunningPike : What do you do? |
| 21:39.48 | BZBW | CunningPike: this part of implementation should not be vendor specific. |
| 21:39.59 | CunningPike | [TK]D-Fender: In general? |
| 21:40.01 | *** join/#asterisk trelane` (i=trelane@everest.sosdg.org) |
| 21:40.19 | [TK]D-Fender | CunningPike : specifics work as I center on Polycoms myself |
| 21:40.32 | CunningPike | [TK]D-Fender: Local govt in BC, Canada |
| 21:40.44 | blitzrage | when a call is hungup, is it possible to run separate apps on the 'h' application for each leg of the call? |
| 21:40.48 | trelane` | I've got a 24xxp with 8 fxo's provisioned. I have a group set up using 1-7 for outbound dialing. When I dial Zap/g1, I get a return of congestion immediately, has anyone seen this issue? |
| 21:40.53 | BZBW | my GS phone is working well:) |
| 21:40.56 | [TK]D-Fender | CunningPike : OH, I meant for ParkAndAnnounce :) |
| 21:41.41 | CunningPike | [TK]D-Fender: lol - we have a special extension (call-park or something) that calls the parker back with an announcement of the stall number |
| 21:42.09 | BZBW | Here is what I don't understand, if ParkAndAnnounce is an feature that can be applied in extensions.conf, it should be in features.conf as well, right? |
| 21:42.14 | [TK]D-Fender | CunningPike : I have yet to ever test it myself.... |
| 21:42.40 | CunningPike | I'll post the config when I get back to my desk |
| 21:44.20 | [TK]D-Fender | cool... always good to learn quick... there are a number of aspects of * I have never needed to touch before that its best I get caught up on. |
| 21:45.05 | Katty | hey mister fender (= |
| 21:45.06 | CunningPike | [TK]D-Fender: There's so much to it that it's hard to have done everything |
| 21:45.14 | Katty | [TK]D-Fender: yeah, i've been....insanely busy. |
| 21:45.16 | Katty | also! |
| 21:45.26 | Katty | i'm thinking about trying out some polycom video conferencing stuff |
| 21:45.31 | Katty | is it a pain to work with? |
| 21:45.37 | [TK]D-Fender | blitzrage : What 2 legs? As far as * is concerned there is only 1 unless you maybe bridge through Local. |
| 21:46.18 | [TK]D-Fender | Katty : Never tried any. My reseller doesn't do their video line and our needs are small in that regard (translation: Nobody cares ATM) |
| 21:46.38 | *** join/#asterisk Bullseye_Network (n=Kyle@216.143.192.69) |
| 21:46.38 | [TK]D-Fender | Katty : Tried eyeBeam? Or are you looking to spend real $? |
| 21:46.38 | file | you can't prove I blew up that ATM! NOOOOOO |
| 21:47.02 | [TK]D-Fender | me starts stuffing bags with the cash |
| 21:47.11 | *** join/#asterisk dracco (i=nunya@ip68-109-219-17.lu.dl.cox.net) |
| 21:47.48 | Bullseye_Network | In the queuing system is there a way to view who the next call will goto next. |
| 21:47.57 | blitzrage | [TK]D-Fender: here is the issue... when someone places a call, a dialplan function is executed that increases the number of calls being used by 1 in the DB (postgres) -- then if the call is determined that it needs to go to another local phone, then the call usage number is increased by 1 for that end point. Now.. when either end hangs up, I need to decrease the value for each side of the call. |
| 21:47.58 | Katty | [TK]D-Fender: i'm not really sure yet... |
| 21:48.11 | Katty | [TK]D-Fender: i know they're going to have a really nice conference room upstairs and they want polycom video phones |
| 21:49.07 | Katty | [TK]D-Fender: is that server controled? |
| 21:49.20 | Katty | i require video conferencing edjimications |
| 21:49.30 | Katty | i didn't get the memo! |
| 21:49.31 | Strom_C | blitzrage: I tried doing something like that, but it ended up growing out of control and still not working right |
| 21:49.33 | trixter | hi katty |
| 21:49.44 | Katty | trixter: herro. |
| 21:49.44 | Strom_C | blitzrage: your best bet is probably to use chanisavail() |
| 21:50.09 | *** join/#asterisk Hmmhesays (i=negative@66.173.103.110) |
| 21:50.12 | Hmmhesays | [TK]D-Fender you around? |
| 21:50.36 | BZBW | speaking of video, anyone tried GXV-3000? |
| 21:50.36 | Katty | Hmmhesays: get in line |
| 21:50.38 | blitzrage | Strom_C: can't do that because it needs to be kept off the local server since it's in a cluster |
| 21:50.41 | Hmmhesays | haha |
| 21:50.42 | Katty | Hmmhesays: also, hi ;) |
| 21:50.52 | Hmmhesays | I need to make the bluetooth headset work with this moto e815 |
| 21:50.52 | BZBW | does 1.2.8 support H.264? |
| 21:51.03 | *** part/#asterisk dracco (i=nunya@ip68-109-219-17.lu.dl.cox.net) |
| 21:53.01 | Hmmhesays | Hi Katty |
| 21:54.34 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 21:55.03 | BZBW | emm, looks like no one knows what I'm talking about:(. |
| 21:56.27 | *** join/#asterisk rmayorga (n=churro@168.243.89.17) |
| 21:56.50 | [TK]D-Fender | Katty : A lot of their stuff is rather proprietary and won't even pass through *. |
| 21:56.53 | sevard | Has anyone ever gotten |
| 21:56.55 | [TK]D-Fender | Hmmhesays : Barely. |
| 21:56.55 | sevard | Tuning module 1................Done! |
| 21:56.55 | sevard | Could not fill input buffer |
| 21:56.55 | sevard | Tuning module 2Failure! |
| 21:56.58 | sevard | with fxotune |
| 21:57.28 | SpaceBass | hey folks |
| 21:57.34 | Katty | [TK]D-Fender: hmm, so does asterisk need a module specifically for video? |
| 21:57.44 | Katty | [TK]D-Fender: i really have no idea how this works ;) |
| 21:58.08 | Hmmhesays | nm i got it running |
| 21:58.10 | [TK]D-Fender | Bullseye_Network : Now way to see who'll be next before the dial, because as they call is holding agents can become available who weren't before and their priority isn't evaluated until the next distribution begins |
| 21:58.11 | CunningPike | [TK]D-Fender, BZBW: I have this line in extensions.conf |
| 21:58.16 | CunningPike | exten => callpark,1,ParkAndAnnounce(pbx-transfer:PARKED|120|SIP/${DIALEDPEERNUMBER}|internal,${DIALEDPEERNUMBER},1) |
| 21:58.34 | MoutaPT | any one can help me how do i make script for chan_misdn to run /etc/init.d/misdn-init scan /etc/init.d/misdn-init config /etc/misdn-init start all this before asterisk boot with safe_Asterisk ? |
| 21:58.36 | CunningPike | And I think that's all I need |
| 21:58.37 | Katty | Hmmhesays: any new mews? |
| 21:58.39 | BZBW | where is this line in? extensions.conf or features.conf? |
| 21:58.46 | [TK]D-Fender | Katty : No, its allows H.263 in pass-through. Thing is most of Polycom's stuff doesn't do video via SIP at all. |
| 21:58.49 | CunningPike | BZBW: Extensions.conf |
| 21:59.02 | SpaceBass | Hey Hmmhesays - what's shaking |
| 21:59.12 | BZBW | CunningPike: Thx! |
| 21:59.17 | Hmmhesays | nm just leaving |
| 22:00.03 | BZBW | what's in features.conf? don't you need a context like "context => parkedcalls" in extensions.conf for link features.conf and extensions.conf together? |
| 22:00.10 | CunningPike | BZBW: No problem - when you use the Park softkey on a Polycom, it places a call to a SIP extension called 'callpark' - ymmv with Grandstream |
| 22:00.18 | Bullseye_Network | thanks Fenter.. :( |
| 22:00.24 | Bullseye_Network | ops. Fender |
| 22:00.27 | CunningPike | BZBW: Maybe - lemme check |
| 22:01.08 | Dr-Linux | CunningPike: we are going to install new Asterisk server. |
| 22:01.12 | [TK]D-Fender | CunningPike : I've never even seen it... you scroll to get it I'm sure, right? |
| 22:01.51 | CunningPike | BZBW: http://pastebin.ca/60425 |
| 22:01.57 | *** join/#asterisk Ludo_ (n=Ludo@obelix.zoxx.net) |
| 22:02.05 | CunningPike | BZBW: That's for features.conf |
| 22:02.38 | BZBW | CunningPike: thx, let me take a look |
| 22:02.55 | CunningPike | [TK]D-Fender: During active calls, the park button is on the 'More' screen |
| 22:02.59 | *** part/#asterisk rmayorga (n=churro@168.243.89.17) |
| 22:04.02 | [TK]D-Fender | CunningPike : I'll have to activate it first obviously :) Much reading to do..... |
| 22:04.08 | BZBW | CunningPike: I don't seem to see ParkAndAnnounce:( |
| 22:04.14 | [TK]D-Fender | CunningPike : Though I'm sure I'll master it in short order. |
| 22:04.28 | CunningPike | [TK]D-Fender: It's one of the features listed in the Admin Guide |
| 22:04.41 | CunningPike | BZBW: Make sure you have that module loaded |
| 22:04.54 | [TK]D-Fender | CunningPike : I know whre it is.. just never having found a need to just left well enougha alone :) |
| 22:05.07 | BZBW | CunningPike: good suggestion. |
| 22:05.13 | CunningPike | [TK]D-Fender: Good policy ;) |
| 22:05.21 | CunningPike | [TK]D-Fender: IIABDFI |
| 22:06.01 | BZBW | CunningPike: what's the module name for features.conf or parking? |
| 22:06.04 | [TK]D-Fender | CunningPike : Ok, thats a new one... translation? |
| 22:06.19 | CunningPike | If it ain't broke....... |
| 22:06.20 | [TK]D-Fender | CunningPike : I think I just got the last 3 :) |
| 22:07.03 | [TK]D-Fender | CunningPike : My configs are never "broken", just not always taking full advantage.... this WILL be a good plus later on.... |
| 22:13.52 | BZBW | CunningPike: can you pastebin the section for context "parkedcalls" in your extensions.conf(if possible)? |
| 22:14.02 | *** join/#asterisk gcarrillog (n=gcarrill@201.155.92.48) |
| 22:14.07 | gcarrillog | hi |
| 22:14.23 | gcarrillog | i have a digitnetworks X100P |
| 22:14.25 | gcarrillog | but |
| 22:14.37 | gcarrillog | th'as not run |
| 22:14.41 | BZBW | Fender: does 1.2.8 support H.264 passthrough? |
| 22:14.42 | gcarrillog | i have FreeBSD 6.0 |
| 22:14.59 | CunningPike | BZBW: It's just one line |
| 22:15.15 | CunningPike | BZBW: The one I posted earlier |
| 22:15.42 | BZBW | so following is under "parkedcalls"? |
| 22:15.43 | BZBW | exten => callpark,1,ParkAndAnnounce(pbx-transfer:PARKED|120|SIP/${DIALEDPEERNUMBER}|internal,${DIALEDPEERNUMBER},1) |
| 22:16.36 | CunningPike | BZBW: It's in our [internal] context. which is what we use for all calls originating inside our network |
| 22:16.55 | [TK]D-Fender | BZBW : yup |
| 22:19.40 | BZBW | Fender: so I just enable H.264 so I can use my GXV-3000? |
| 22:20.25 | BZBW | CunningPike: how does "parkedcalls" referred by extensions? |
| 22:21.02 | CunningPike | BZBW: I'm just wondering that myself - I need to go back a few versions of my extensions.conf and see where they went...... |
| 22:23.02 | [TK]D-Fender | BZBW : yup |
| 22:24.46 | BZBW | cool, do u know how to configure * to leave a videomail and retrieve it? |
| 22:25.31 | BZBW | Fender: where to see the complete list of release note for 1.2.8? |
| 22:26.48 | Jameno123 | BZBW, my complaints too |
| 22:26.54 | Jameno123 | where do you find release notes? |
| 22:27.06 | BZBW | no where:( |
| 22:27.10 | Jameno123 | the notes on asterisk.org are so "vauge", it doesnt really inform me if i should upgrade |
| 22:29.24 | [TK]D-Fender | BZBW : * does not actually TOUCH video. Its just allows passthrough between 2 capable peers. |
| 22:29.30 | mitcheloc | hmm.... you all know how aah is a big deal b/c everything is pre-installed right? |
| 22:29.38 | [TK]D-Fender | BZBW : And go to asterisk.org for the changelog |
| 22:29.49 | mitcheloc | when do you think that distros will have a package group for asterisk.... that would make sense... they have dns/ftp/mail/web..etc |
| 22:30.02 | [TK]D-Fender | mitcheloc : We all know about the kind of chumps who end up here anyways ;) |
| 22:30.08 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
| 22:30.23 | [TK]D-Fender | mitcheloc : * sure... AMP <- NO |
| 22:30.35 | mitcheloc | [TK]D-Fender: well chumps pay our bills |
| 22:31.01 | [TK]D-Fender | mitcheloc : Who is "our"? Your company gives A@H support? |
| 22:31.04 | mitcheloc | [TK]D-Fender: yea, * sure, but when??!!! not that it's hard to install... nevermind.. |
| 22:31.15 | mitcheloc | HELL NO |
| 22:31.25 | mitcheloc | just a figure of speech |
| 22:31.43 | [TK]D-Fender | mitcheloc : Thems fightin' words! |
| 22:31.48 | BZBW | Fender: thx. I know that, just that I remember markster did something so that videomail can be retrieved. |
| 22:33.42 | [TK]D-Fender | BZBW : If you say so.... thats news to me.... |
| 22:34.12 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 22:36.08 | StyleWarz | on voip-info.org it says "The number of messages for each voicemailbox is limited to 99 messages." <- can i raise that somewhere? |
| 22:37.01 | russellb | sounds bogus to me |
| 22:37.01 | mitcheloc | 99 seems reasonable... |
| 22:37.27 | Jameno123 | its 100 |
| 22:37.38 | Jameno123 | used to be hardcoded, but i think voicemail.conf has an option now? |
| 22:37.49 | Ariel_ | StyleWarz, yes you can increase it by you have to edit the code |
| 22:40.47 | BZBW | Fender: if 1.2.8 support H.264, can I specify "allow = h264" in sip.conf? |
| 22:44.03 | BZBW | jameno123: http://ftp.digium.com/pub/asterisk/ChangeLog-1.2.8 |
| 22:44.14 | [TK]D-Fender | BZBW : check out the codecs page on the wiki for valid values. |
| 22:44.58 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 22:45.05 | bcnl | I know 1.2.8 just came out, but for the past few weeks I've been getting complaints from users about audio stuttering |
| 22:45.24 | bcnl | like a second and a bit of audio, then a equal amount of time of silence |
| 22:45.29 | Jameno123 | BZBW, ;) nice thx. |
| 22:45.57 | bcnl | I'm at a loss of where to look on this one since it's not every call |
| 22:47.03 | Alex | bcnl: Codecs changed at all? |
| 22:47.41 | bcnl | Alex: no |
| 22:47.48 | bcnl | using g729 from digium |
| 22:48.40 | Alex | Anything changed on the box as a whole that might cause it? |
| 22:49.42 | bcnl | not that I can tell |
| 22:49.51 | bcnl | latency is good |
| 22:49.52 | SplasPood | StyleWarz: yea you can increase the message limit |
| 22:49.55 | bcnl | <100ms for all parties |
| 22:50.05 | StyleWarz | SplasPood: in the code or where? |
| 22:50.07 | bcnl | my outbound IAX trunks are 5-10ms away |
| 22:50.15 | *** join/#asterisk adker (n=adker@70-100-239-157.br1.glv.ny.frontiernet.net) |
| 22:50.24 | StyleWarz | Jameno123: a limit is unneeded there :) if i want a limi there, i can configure it ;) |
| 22:50.27 | bcnl | but this happens with sip -> * -> outbound IAX, and with sip -> * -> sip |
| 22:50.32 | SplasPood | StyleWarz: voicemail.conf, there should be an example in the sample conf, or on voip-info.org.. I believe it's maxmsgs= tho |
| 22:52.06 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 22:52.20 | StyleWarz | SplasPood: thats where i looked, but there is none :/ |
| 22:52.21 | *** join/#asterisk NDT (n=nunya@cpe-72-228-10-145.nycap.res.rr.com) |
| 22:52.34 | CunningPike | BZBW: We don't have anything else in our config - it just works |
| 22:52.34 | SplasPood | StyleWarz: Its there... |
| 22:52.40 | SplasPood | StyleWarz: you checked voip-info? |
| 22:52.55 | StyleWarz | ah i didn't check voicemail.conf, i just checked voicemail |
| 22:53.02 | StyleWarz | sorry :) |
| 22:53.12 | SplasPood | ; Maximum number of messages per folder. If not specified, a default value |
| 22:53.12 | SplasPood | ; (100) is used. Maximum value for this option is 9999. |
| 22:53.12 | SplasPood | ;maxmsg=100 |
| 22:54.24 | NDT | Anyone know if ${RDNIS} has changed since 1.2.1? As in...I have to servers setup exactly the same way but ${RDNIS} isn't giving me anything on the second server. 1st server is running 1.2.1 |
| 22:54.48 | bcnl | NDT: what format are the calls comming in at? |
| 22:55.01 | bcnl | I went from getting calls inbound via SIP to IAX and lost RDNIS |
| 22:55.05 | bcnl | :( |
| 22:55.38 | NDT | What happens is we have numbers pointed to a toll free...I then have to use RDNIS to see what number was actually dialed to route the call |
| 22:56.01 | NDT | all TDM |
| 22:56.12 | BZBW | damn, I don't see from WIKI mentioning about video codec support, for "allow=":(. |
| 22:56.26 | *** part/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 22:56.28 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 22:56.33 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 22:56.58 | BZBW | CunningPike: thx. It's a bit mysterious:). |
| 22:58.08 | MoutaPT | any one could advice me how do I Add misdn-init to my run level and be sure that is run before safe_asterisk script ? |
| 22:58.35 | MoutaPT | misdn-init is in my /etc/init.d/misdn-init |
| 22:58.59 | Dr-Linux | MoutaPT: then it should run |
| 22:59.25 | Dr-Linux | MoutaPT: or put it in your /etc/rc.local |
| 22:59.33 | Dr-Linux | then sleep 5 |
| 22:59.40 | Dr-Linux | then safe_asterisk |
| 22:59.46 | *** join/#asterisk iq|mobile (n=iq@71-215-34-237.omah.qwest.net) |
| 23:00.41 | MoutaPT | Dr-Linux i've red this for debian: update-rc.d misdn-init start 60 2 3 4 5 . stop 60 0 1 6 . |
| 23:00.46 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 23:00.56 | MoutaPT | what would be similar for Centos? |
| 23:00.59 | NDT | bcnl: Did you figure out how to get RDNIS back on yours? |
| 23:01.05 | MoutaPT | can u help me? |
| 23:01.13 | Dr-Linux | i don't know debian |
| 23:01.27 | MoutaPT | perfect i'm using centos |
| 23:01.28 | MoutaPT | :) |
| 23:01.35 | Dr-Linux | but in centOS , you put stuff in /etc/rc.local |
| 23:01.44 | Dr-Linux | i believe centos and RHEL are same |
| 23:02.11 | MoutaPT | problem is that i need to run with arguments |
| 23:02.16 | MoutaPT | misdn-init start |
| 23:02.41 | MoutaPT | do i need to make a script? |
| 23:04.35 | CunningPike | BZBW: It made me think :) |
| 23:05.01 | Dr-Linux | CunningPike: look at this .. heh i just .. |
| 23:05.06 | Dr-Linux | Adventure: On June 6, 2006 at 6 min & 6 sec after 6 AM Day, Time date will be 06:06:06:06:/06/06/06 www.syednetworks.com |
| 23:05.46 | CunningPike | Dr-Linux: Neat - there was the 4/5/06 thing earlier, too |
| 23:05.50 | Dr-Linux | This happens once in 100000 years. |
| 23:06.15 | Qwell[] | 06:06:06:06? |
| 23:06.17 | CunningPike | Dr-Linux: What about July 7 next year? |
| 23:06.19 | Qwell[] | wtf |
| 23:07.35 | *** join/#asterisk Sp3ciaL_K (n=alex@d141-136-176.home.cgocable.net) |
| 23:08.01 | SplasPood | CunningPike: Well the 3 sixes hold some meaning for some people :P |
| 23:08.10 | *** join/#asterisk pjchilds (n=pjchilds@pdpc/supporter/student/pjchilds) |
| 23:08.57 | Sp3ciaL_K | im trying to use the nvfaxdetct and it seems to be detecting a fax tone but it gives me an error saying fax detected but no fax extension...eventhough i do have the fax extension |
| 23:09.12 | Sp3ciaL_K | any ideas? |
| 23:10.38 | [TK]D-Fender | Dr-Linux : Tha'd be every 1000 years..... |
| 23:10.41 | *** join/#asterisk fjean (n=fjean@201.29.140.206) |
| 23:11.19 | SplasPood | someone call me when it's 06/06/6666 |
| 23:11.52 | fjean | hi, I get this error while trying to compile libmfcr2: UC_EVENT_DIALEDNUMBER undeclared...anybody got that? |
| 23:11.53 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 23:12.15 | Dr-Linux | [TK]D-Fender: i'm thinking about next year in july :S |
| 23:16.24 | RoyK | ~lart wasim |
| 23:19.26 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 23:20.31 | *** join/#asterisk [hC] (i=turnerd@donkey.voxter.ca) |
| 23:20.44 | *** join/#asterisk JASON99 (n=jason@jason.unitz.ca) |
| 23:20.47 | *** join/#asterisk [hC] (i=turnerd@donkey.voxter.ca) |
| 23:20.52 | JASON99 | Hello, i've created a context for inbound calls and I am playing a tone and I need to listen for the following dial patterns.. How would I do this? [1NXXNXXXXXX|NXXNXXXXXX|011.T|XXXXXXX#] |
| 23:22.10 | [hC] | Hey guys, for FXO line tuning, is this something that is typically done on each line (and rx/tx gains are set per line in zapata.conf)? or is it usually a per-card thing? |
| 23:22.35 | JASON99 | as far as I understand the read command only allows you to specify a max digit length but I need to be able to enter different lengths without the # |
| 23:23.02 | [TK]D-Fender | JASON99 : You don't need #. you can just time out. |
| 23:23.05 | *** join/#asterisk websae_ (n=websae@CPE-24-167-206-22.wi.res.rr.com) |
| 23:23.47 | iq | hi |
| 23:23.47 | JASON99 | [TK]D-Fender: that is a good point.. what is the timeout length? |
| 23:24.00 | JASON99 | I will check ;) Thanks for the tip |
| 23:24.33 | [TK]D-Fender | JASON99 "show application read" |
| 23:25.10 | JASON99 | perfect |
| 23:26.52 | NDT | 1.2.1 on one server I can get ${RDNIS}, 1.2.4, 1.2.7.1, 1.2.8 I can't heh...2 servers exactly the same other then the builds...all TDM...any ideas? |
| 23:27.41 | NDT | All 419 Quad cards too |
| 23:27.45 | NDT | err 410 |
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| 23:30.08 | bcnl | 16:03 < NDT> bcnl: Did you figure out how to get RDNIS back on yours? |
| 23:30.11 | bcnl | nadda |
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| 23:36.16 | *** part/#asterisk websae_ (n=websae@CPE-24-167-206-22.wi.res.rr.com) |
| 23:38.09 | *** join/#asterisk cfassoni (n=cfassoni@c911444e.rjo.virtua.com.br) |
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| 23:38.27 | NDT | it's driving me crazy heh |
| 23:40.41 | *** part/#asterisk pcm (n=pcm@65.4.1.115) |
| 23:41.42 | key2 | could someone tell me how I could Dial "foo@bar.com" using a SIP proxy ? since Dial(SIP/proxy/foo@bar.com) doesnt work ? |
| 23:46.22 | key2 | nobody ? |
| 23:46.34 | dlynes_home | key2: you mean an outbound proxy? |
| 23:46.51 | *** join/#asterisk Splat (n=Splat@220-253-134-175.TAS.netspace.net.au) |
| 23:46.56 | key2 | dlynes_home: for example |
| 23:48.01 | key2 | but I want my proxy do call "foo@bar.com" and not asterisk resolving foo@bar.com and sending it straight |
| 23:48.01 | dlynes_home | for example what? |
| 23:48.01 | dlynes_home | you forgot something after that |
| 23:48.10 | key2 | dlynes_home: i mean "yes, for example." |
| 23:48.24 | dlynes_home | but what's the example? |
| 23:48.36 | key2 | an outbound proxy :) |
| 23:48.55 | dlynes_home | ah...i guess your english just isn't strong |
| 23:48.56 | dlynes_home | sorry |
| 23:49.08 | key2 | all I mean is that if i do a Dial(SIP/foo@bar.com) asterisk is gonna resolve bar.com and send it straight |
| 23:49.36 | dlynes_home | I thought you were missing something, because of the way you stated your question (i.e. forgot to type something), but i guess it's just you didn't know how to word it |
| 23:49.52 | dlynes_home | Yeah, I understand waht you want |
| 23:50.07 | key2 | lol |
| 23:50.14 | dlynes_home | key2: iow, you're wondering if asterisk supports sip domains |
| 23:50.16 | [hC] | Hey guys, for FXO line tuning, is this something that is typically done on each line (and rx/tx gains are set per line in zapata.conf)? or is it usually a per-card thing? |
| 23:50.18 | dlynes_home | key2: no, it doesn't |
| 23:50.34 | Dr-Linux | dlynes_home: his engish is better than mine though :P |
| 23:50.36 | dlynes_home | key2: it's planned for a later version, and there is work on it in oej's test-this-branch |
| 23:50.38 | key2 | dlynes_home: what does it have to see with sip domain |
| 23:50.48 | dlynes_home | key2: but nothing in trunk atm |
| 23:51.06 | dlynes_home | key2: @bar.com is the domain in question |
| 23:51.14 | key2 | right |
| 23:51.42 | dlynes_home | key2: whether or not the domain support in oej's test-this-branch covers what you need or not, is another issue |
| 23:51.47 | key2 | dlynes_home: and for the moment, asterisk can only call bar.com straight ? it can't send the INVITE to a proxy first and let the proxy do the routing ? |
| 23:52.14 | dlynes_home | key2: i believe people have proxying working with asterisk installs |
| 23:52.23 | dlynes_home | key2: but i believe they're doing it with SER, not asterisk |
| 23:52.37 | key2 | dlynes_home: well that's what I do |
| 23:53.14 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
| 23:53.23 | dlynes_home | Dr-Linux: his english is just fine, for the most part...it's just the way he worded his question, I understood it, but I thought He typed something in after it that never showed up in the channel |
| 23:53.24 | key2 | I got a SIP express router before my asterisk, it sends me a call, and I want asterisk to Dial foo@bar.com but not directly, just through the SER |
| 23:53.27 | *** join/#asterisk Kokey (n=jramirez@201.133.218.194) |
| 23:53.47 | dlynes_home | key2: yeah, i think there's people doing that in this channel |
| 23:53.53 | dlynes_home | key2: but I don't know how, personally |
| 23:54.36 | key2 | dlynes_home: I guess with Set(SIP_HEADER), they remake the SIP header |
| 23:54.41 | key2 | probably something like that |
| 23:54.51 | dlynes_home | key2: maybe, or maybe using an agi script |
| 23:55.30 | key2 | dlynes_home: now that oej got back to Sweden, i guess he's asleep at this time :) |
| 23:55.38 | dlynes_home | key2: lol |
| 23:55.47 | key2 | dlynes_home: how could an agi solve this probleme |
| 23:56.06 | dlynes_home | key2: i have no idea...never worked with it yet, myself |
| 23:56.16 | key2 | mog_home: there ? |
| 23:56.19 | dlynes_home | key2: i just ran into a problem today where i'm going ot ahve to use agi to solve it |
| 23:56.26 | dlynes_home | key2: never ran into such a situation before |
| 23:56.47 | dlynes_home | key2: so, I'm not entirely sure what agi can solve and what it can't solve |
| 23:58.05 | nextime | <ironic mode> with an EAGI in python and using the shtoom sip stack maybe </ironic mode> |
| 23:58.28 | dlynes_home | I take it shtoom is buggy as all hell? |
| 23:58.55 | nextime | dlynes_home : no idea, never used it |
| 23:59.19 | dlynes_home | nextime: so it's non-existent then? why ironic? |
