| 00:00.28 | *** join/#asterisk coppice (n=chatzill@68.166.17.210.dyn.pacific.net.hk) |
| 00:04.05 | ManxPower | ~fxofxs |
| 00:04.06 | infobot | it has been said that fxofxs is An FXO port (red Digium module) expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port (green Digium module) expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 00:09.59 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:15.11 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 00:16.02 | *** join/#asterisk Xetrov` (n=xetrov@unaffiliated/xetrov/x-827361) |
| 00:21.44 | *** join/#asterisk neurosys (n=vinix@71.196.19.254) |
| 00:26.53 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 00:29.22 | *** join/#asterisk xchaos (n=none@CPE001d7ef0ba9d-CM001ceab63f9a.cpe.net.cable.rogers.com) |
| 00:31.59 | box2 | man, where can i find this Holy Grail |
| 00:32.11 | xchaos | hey guys |
| 00:32.20 | box2 | hello xchaos |
| 00:32.26 | xchaos | question what type of upload speed do i need to maintain an asterisk server |
| 00:32.33 | xchaos | how much upload and download speed? |
| 00:32.45 | box2 | depends how many calls you plan on supporting at once |
| 00:33.08 | xchaos | just a few |
| 00:33.08 | box2 | and what codecs you are using for your audio stream |
| 00:33.14 | drmessano | box2: In the Castle Aaargh |
| 00:33.14 | xchaos | i dont know i suck |
| 00:33.20 | xchaos | i hired d-fender to set it up |
| 00:33.22 | xchaos | he knows my system |
| 00:34.15 | box2 | drmessano: heh <3 |
| 00:36.52 | *** join/#asterisk ltd_wk (i=z@patwk.transact.net.au) |
| 00:38.21 | drmessano | This is the Castle of my master, Guy De Loimbard! |
| 00:40.08 | box2 | go and tell your master that we have been charged by God with a sacred quest |
| 00:40.42 | drmessano | I don't think he'll be very keen.. you seen, he's already got one |
| 00:40.55 | drmessano | s/seen/see/ |
| 00:41.03 | thehar | dear papa johns: hurry up |
| 00:41.10 | box2 | heh |
| 00:41.24 | thehar | I just clicked submit and they aren't here. =) |
| 00:41.27 | drmessano | Are you sure? Oh yes, it's very nice :) |
| 00:42.23 | jaytee | drmessano! |
| 00:42.42 | drmessano | Your mother was a Cisco and your father smelt of grandstream! |
| 00:42.48 | jaytee | lol |
| 00:43.03 | thehar | ew! |
| 00:43.48 | _ShrikE | Have at you! |
| 00:44.15 | jaytee | drmessano, which 1.6 release do you recommend for SIP/TCP to Exchange UM? 1.6.0 or 1.6.1? |
| 00:45.19 | drmessano | Late 1.6.0.x, not sure about 1.6.1.x and TCP |
| 00:46.05 | jaytee | cool |
| 00:46.07 | jaytee | thanks |
| 00:46.29 | drmessano | I am an enchanter.. There are some who call me "Tim" |
| 00:47.05 | *** join/#asterisk MaliutaLap (n=biteme@203.171.192.224) |
| 00:48.22 | jaytee | bring us a shrubbery! |
| 00:49.30 | drmessano | HA |
| 00:49.52 | drmessano | Cut down the mightiest tree in the forest with.. |
| 00:49.56 | drmessano | A herring! |
| 00:50.18 | thehar | puts the vaporizer in the middle of the room for use. |
| 00:50.48 | jaytee | "Look, in order to maintain air-speed velocity, a swallow needs to beat its wings forty-three times every second, right? It's a simple question of weight ratios. A five ounce bird could not carry a one pound coconut. " |
| 00:52.40 | *** join/#asterisk mumtazah (n=mumtazah@203.82.91.104) |
| 00:52.56 | drmessano | What is your name? What is your quest? What is your favorite color? Blue, no I mean.. AHHH |
| 00:53.19 | drmessano | That sounds like a newb answering TK |
| 00:55.06 | drmessano | WHICH VERSION OF ASTERISK? ARE YOU BEHIND A NAT? NO, I MEAN.. AHHHH |
| 00:55.15 | jaytee | hahahaa |
| 00:56.20 | *** join/#asterisk KingOfDos (n=irssi@ht1-ix.kingofdos.net) |
| 00:57.02 | *** join/#asterisk geneticx (n=geneticx@adsl-2-59-65.mia.bellsouth.net) |
| 00:57.16 | MaliutaLap | he's not the [TK]D-Fender, he's a naughty boy! |
| 00:58.27 | drmessano | lol |
| 00:58.29 | jaytee | "From now on I want you all to call me Loretta" |
| 00:59.08 | drmessano | Is that TK being lured into #asteriskwin32 by naughty VoIP virgins? |
| 00:59.23 | _ShrikE | I'm not old. Im 37. |
| 01:00.43 | drmessano | Come and see the violence inherent in the system |
| 01:01.21 | drmessano | COME AND SEE THE VIOLENCE INHERENT IN THE SYSTEM.. HELP, HELP, I'M BEING REPRESSED! |
| 01:01.30 | drmessano | Bloody peasant! |
| 01:03.53 | drmessano | Surely they have a clapping coconut iPhone app |
| 01:03.55 | drmessano | They MUST |
| 01:04.03 | thehar | shh shh |
| 01:04.05 | *** join/#asterisk nighty^ (n=nighty@210.188.173.245) |
| 01:04.08 | thehar | there there |
| 01:06.26 | drmessano | huh? |
| 01:07.59 | Sandheaver | ooh yeah exchange UM |
| 01:08.10 | Sandheaver | has exchange and asterisk. |
| 01:08.11 | Sandheaver | hrm |
| 01:08.12 | thehar | 1.6.x can do exchange UM? |
| 01:09.31 | drmessano | I had it working with 1.6.0.8+ |
| 01:10.27 | jaytee | my boss wants me to try to get 1.6 working with UM so we can dump using sipX as a udp/tcp proxy |
| 01:11.34 | hesco | although I am still able to make an outbound call, an attempt to reload my iax configuration (which has not changed) is yielding: "NOTICE[12946]: iax2-provision.c:518 iax_provision_reload: No IAX provisioning configuration found, IAX provisioning disabled.". Any clues what that might be about? I have been editing a two extension dial plan included in /etc/asterisk/extesnions.conf, but I can't imagine what that would have to do with this. |
| 01:12.18 | *** join/#asterisk ariel_ (n=ariel_@c-71-196-99-26.hsd1.fl.comcast.net) |
| 01:12.18 | hesco | rather a two extension file #include'ed in extensions.conf |
| 01:12.37 | ariel_ | hello everyone |
| 01:13.52 | Sandheaver | hola ariel |
| 01:15.33 | drmessano | SUP BRO |
| 01:16.01 | ariel_ | actually testing a java irc web clinet, I need to set this up when I go on the ships as they don't have allot of ports opened. |
| 01:16.11 | KyleK | hesco: are you sure thats not a normal warning? |
| 01:17.11 | KyleK | ariel_: if its a java client that connects to the irc server by itself, then it has to access the server over port 6667 or another open port |
| 01:19.23 | Sandheaver | exchange UM looks easy in 1.6... |
| 01:19.55 | ariel_ | not really it's located on http://java.freenode.net |
| 01:20.13 | Sandheaver | exchange just acts like any other sip device, if extension X doesn't answer, forward to extension Y, which is exchange, and it basically records the voicemail |
| 01:20.13 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 01:20.16 | ariel_ | I am testing right now with all ports execpt 80/81 closed |
| 01:20.27 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 01:21.09 | *** join/#asterisk leon01 (n=leon01@ool-457c12b2.dyn.optonline.net) |
| 01:21.09 | drmessano | Sandheaver: Pretty much |
| 01:21.17 | hesco | KyleK: no I am not, though I have never noticed it before. |
| 01:21.56 | drmessano | ariel_: If it's java, it's not connecting from the server |
| 01:22.09 | hesco | congrats ariel_ you are coming through l/c with your ports locked down |
| 01:22.20 | ariel_ | drmessano, it's using a http proxy |
| 01:22.25 | hesco | not sure how you pull that off though |
| 01:23.04 | Sandheaver | hesco: outgoing ports aren't usually blocked |
| 01:23.07 | drmessano | It's PJIRC, and no it's not |
| 01:23.24 | Sandheaver | not a proxy; direct TCP connection from the JVM it's running on |
| 01:23.42 | Sandheaver | ... unless it's going through a proxy |
| 01:23.44 | ariel_ | well it's so far one of the few that works off the ship's network which is what I need. |
| 01:23.47 | Sandheaver | which it won't by default |
| 01:24.02 | Sandheaver | ariel_: the ship's network? where the heck are you |
| 01:24.41 | ariel_ | Out on the ocean, Holand America Maasdam.... |
| 01:24.54 | box2 | ON A BOAT MF'RRRRR |
| 01:25.04 | hesco | though I noticed it when reloading iax.conf after my tests on my latest application gave me these errors: "channel.c:3477 ast_request: No channel type registered for 'IAX'" and "app_dial.c:1502 dial_exec_full: Unable to create channel of type 'IAX' (cause 66 - Channel not implemented)" |
| 01:25.15 | hesco | which aren't making much sense to me. |
| 01:25.44 | ariel_ | hesco, what is your dial string? |
| 01:25.55 | Sandheaver | box2: love that song. what a peaceful little ditty |
| 01:25.57 | hesco | Sandheaver: wish you could have told that to the network engineer on the installation I was doing last week. |
| 01:26.10 | Sandheaver | box2: F trees I climb bouy's mother F'er! |
| 01:26.18 | Sandheaver | box2: boat engine make noise mother F'er |
| 01:26.21 | Sandheaver | lol |
| 01:26.22 | *** join/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:26.23 | box2 | Sandheaver: haha yea <3 |
| 01:27.03 | ariel_ | is working on setting up some new DECT phones from Polycom on the ship... |
| 01:27.28 | hesco | my shells just froze, have to answer that one in just a moment |
| 01:28.00 | drmessano | http://www.youtube.com/watch?v=InBXu-iY7cw |
| 01:28.29 | hesco | ariel_: exten => 992,n,Dial(${TRUNK}/${RECIPIENT},20,g) |
| 01:28.59 | drmessano | http://www.youtube.com/watch?v=8hYtUYiuzkw <-- More asterisk appropriate |
| 01:29.05 | ariel_ | argh in the navy....lol |
| 01:29.26 | drmessano | jaytee: http://www.youtube.com/watch?v=8hYtUYiuzkw |
| 01:29.33 | thehar | sings In the Navy |
| 01:29.37 | drmessano | New Asterisk promotional film |
| 01:29.40 | ariel_ | hesco, and what do you have setup for ${TRUNK} |
| 01:30.16 | ariel_ | turn the youtube off, too much disco |
| 01:30.40 | jaytee | drmessano, this one's even better!!! http://www.youtube.com/watch?v=Yu_moia-oVI |
| 01:32.01 | drmessano | You rolled me, bitch |
| 01:32.04 | drmessano | :( |
| 01:32.52 | drmessano | picks up the shattered pieces of his own and walks off, pouting |
| 01:32.56 | Nivex | isn't that punishable by death these days? |
| 01:33.13 | eppigy | allo |
| 01:33.23 | drmessano | MACHO MACHO MAN |
| 01:33.23 | hesco | TRUNK=IAX2/dc_cf_com |
| 01:33.28 | jaytee | hey! considering the "In the Navy" vid I'd say it was justifiable |
| 01:33.33 | jaytee | DAVE!!!! |
| 01:33.35 | hesco | sorry for the delay, distracted by VillagePeople |
| 01:33.36 | *** join/#asterisk OrNix (n=ornix@l151-249-47.static.cn.ru) |
| 01:34.10 | drmessano | --> http://www.youtube.com/watch?v=PJQVlVHsFF8 |
| 01:35.20 | jaytee | oh dear fucking god! |
| 01:35.27 | ariel_ | you have a peer in iax.conf for dc_cf_com |
| 01:35.49 | hesco | I do indeed |
| 01:35.59 | ariel_ | hulu does not work from the ship....thinks I am some where in the canada |
| 01:36.08 | Nivex | saw "Hasselhoff" and closed the window before the video could spool |
| 01:36.22 | Nivex | ship? where the? |
| 01:36.59 | hesco | its a diamondcard account |
| 01:37.09 | ariel_ | Nivex, I am doing a installation aboard a ship |
| 01:37.27 | hesco | drmessano is being a bad influence, feeding me bad 80's music |
| 01:37.43 | ariel_ | hesco, do: iax2 show peers |
| 01:38.55 | *** join/#asterisk blackest_mamba (n=blackest@c-71-239-160-143.hsd1.il.comcast.net) |
| 01:38.59 | drmessano | We haven't hit the 80s yet |
| 01:39.02 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 01:39.21 | drmessano | But if you insist: http://www.youtube.com/watch?v=uUjIA3Rt7gk |
| 01:39.46 | kb3ien | hrm anyone in .ca wants to do a 1u colo swap with me? |
| 01:39.52 | ariel_ | can't really see those much, there far too slow and laggy |
| 01:40.10 | kb3ien | not ship board... |
| 01:40.12 | hesco | uh-oh, now I'm seeing: Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 01:40.39 | ariel_ | lol |
| 01:41.04 | *** join/#asterisk markgreene (n=markgree@130.160.45.32) |
| 01:41.14 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.69) |
| 01:42.52 | hesco | this is getting stranger and stranger |
| 01:44.34 | ariel_ | kb3ien, most of my servers on the main land are in the Nap behind some OC3's and OC12's |
| 01:44.34 | hesco | I've used the /ect/init.d script several times attempting to stop and restart the * server. It says: Starting Asterisk PBX: asterisk., then asterisk -r gives me: Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 01:45.13 | ariel_ | hesco see what it's failing on, with asterisk -vvvgc |
| 01:45.20 | ariel_ | from the linux prompt |
| 01:46.44 | eppigy | jaytee: HI |
| 01:46.46 | ariel_ | argh this connection I am on is lagging |
| 01:47.06 | jaytee | how are ya, Dave? |
| 01:48.23 | eppigy | I am pretty grood |
| 01:48.32 | eppigy | besides coming back in to work |
| 01:48.33 | eppigy | for bs |
| 01:48.50 | jaytee | aw that sucks |
| 01:49.09 | eppigy | its all good |
| 01:49.40 | ariel_ | need to switch back to my normal network, bbl I hope..... |
| 01:51.27 | *** join/#asterisk tjz (n=tjz@bb121-7-20-94.singnet.com.sg) |
| 01:52.04 | *** join/#asterisk propellerhead (n=yogurt2u@host42.190-231-29.telecom.net.ar) |
| 01:57.37 | *** join/#asterisk luckyaba (n=lucky@ip72-194-215-55.sb.sd.cox.net) |
| 01:58.01 | *** join/#asterisk moy (n=moy@mail.e-contact.cl) |
| 02:05.25 | *** join/#asterisk Firass-z0r (n=asadf@c-67-201-205-34.reshall.wwu.edu) [NETSPLIT VICTIM] |
| 02:05.25 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) [NETSPLIT VICTIM] |
| 02:05.30 | *** join/#asterisk ricko73 (n=dhartman@wilug/newlug/ricko73) [NETSPLIT VICTIM] |
| 02:05.31 | *** join/#asterisk IBC_jkenney (n=jkenney@ip65-44-169-66.z169-44-65.customer.algx.net) [NETSPLIT VICTIM] |
| 02:05.32 | *** join/#asterisk RonDutt (n=aiontsu@69.7.43.20) [NETSPLIT VICTIM] |
| 02:05.39 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) [NETSPLIT VICTIM] |
| 02:05.39 | *** join/#asterisk qdk (n=qdk@0x573d8d8b.bynqu1.dynamic.dsl.tele.dk) |
| 02:05.44 | *** join/#asterisk Chinorro (n=Chino@19.226.117.91.dynamic.mundo-r.com) [NETSPLIT VICTIM] |
| 02:11.51 | hesco | somehow my application created an endless loop feeding callfiles into into the outbound spool. ran the load average up close to 12. |
| 02:13.36 | hesco | now when I try to invoked the concole I head the banner, followed by: Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?); though file says that is a socket. |
| 02:17.38 | hesco | I killed the server. but the socket is still there. |
| 02:17.45 | hesco | is it safe to rm it? |
| 02:18.17 | hesco | rm /var/run/asterisk.ctl ; will it get recreated when I restart asterisk? |
| 02:18.17 | thehar | yup |
| 02:18.31 | hesco | thanks, will do then |
| 02:30.43 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 02:32.18 | *** join/#asterisk errotan (n=errotan@5403E408.catv.pool.telekom.hu) |
| 02:37.26 | *** join/#asterisk box2 (n=box2@DCC.SEND.startkeylogger.000.telephreak.org) |
| 02:40.34 | neurosys | is there a way to limit voicemail options? |
| 02:45.27 | Corydon76-dig | You can limit various options by turning off the associated configuration option in voicemail.conf, but no, there isn't any generic way to turn off specific options |
| 02:46.14 | *** join/#asterisk Naikrovek (n=jjohnson@98.214.112.102) |
| 02:47.52 | neurosys | Im doing a box for a hotel. Theyd like voicemail for all the rooms, but im trying to figure out how that would work. How could i make it where they can only check the voicemail from the room, and remove their option to set a password. |
| 02:48.27 | Naikrovek | neurosys: don't allow them to check anyone's mailbox but their own, then, don't ask for password |
| 02:48.58 | Naikrovek | easy to do with freepbx, i don't know how to do it with vanilla asterisk |
| 02:49.08 | ManxPower | neurosys: 1.6 has the "minivoicemail" set of features. i.e. you should have the tools available to build a simple voicemail system in the dialplan. |
| 03:01.24 | KyleK | neurosys: voicemailmain({some variable here},s) is how you pick up the voicemail from the room |
| 03:03.15 | *** join/#asterisk evil_gordita (n=evilgord@ip24-254-160-77.rn.hr.cox.net) |
| 03:04.12 | KyleK | neurosys: at worst possible case just comment out the part that lets someone choose to change the password, but im sure theres an option for that |
| 03:04.27 | neurosys | KyleK: I havent found it. |
| 03:05.40 | *** join/#asterisk blkry (n=chatzill@96.37.27.72) |
| 03:07.09 | neurosys | the other issue is if a caller hits *, hell get right into the voivemail box :P |
| 03:08.47 | KyleK | not really clear on what you're saying |
| 03:08.55 | KyleK | its set up that all calls immediately go to voicemail? |
| 03:09.45 | Nugget | It's tricky to rock a rhyme that's right on time. It's tricky. |
| 03:09.47 | neurosys | KyleK: If the caller rings the ext, it goes to voicemail after X seconds. If i have no passcode set, then an outside caller can hit * and get right into the box |
| 03:11.29 | KyleK | huh either set passcodes when making the boxes or turn that off? |
| 03:12.34 | neurosys | This is for a hotel. it would be very cumbersome to generate passcodes for each guest checking in and out. |
| 03:13.58 | KyleK | well then turn off the * feature |
| 03:14.48 | neurosys | KyleK: YOu can do that? I havent seen that. |
| 03:14.49 | KyleK | when am I supposed to press *, while its saying to leave a message? |
| 03:14.58 | neurosys | yes |
| 03:15.05 | KyleK | tries again |
| 03:15.09 | *** join/#asterisk Naikrovek (n=jjohnson@98.214.112.102) |
| 03:15.27 | KyleK | i think mines turned off :( |
| 03:16.08 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 03:18.19 | neurosys | I think Manxpower has the only viable solution. unless there are undocumented triggers i dont know about |
| 03:21.04 | KyleK | are you running a pre setup asterisk? |
| 03:21.20 | *** join/#asterisk Nivex (i=ac1f0304@user-0c2h3cu.cable.mindspring.com) |
| 03:22.47 | KyleK | yea a person has to explicitly set what pressing * does in the dialplan, I just made mine go to Musiconhold |
| 03:23.24 | KyleK | exten => a,1,Musiconhold() |
| 03:24.50 | neurosys | Hmm.> My asterisk box, * works. |
| 03:25.38 | KyleK | check for an a extension in the dialplan |
| 03:26.05 | neurosys | KyleK: There is none. |
| 03:26.19 | KyleK | odd, what version of asterisk |
| 03:26.57 | neurosys | 1.4.2 |
| 03:27.12 | KyleK | oh I'm on 1.6.1 lol |
| 03:28.01 | russellb | 1.4.2, ooh, that's old :-) |
| 03:28.11 | neurosys | my 1.6.0.6 box does the same thing. |
| 03:28.47 | KyleK | odd, its not working as expected for me |
| 03:29.54 | *** join/#asterisk ketema_ (n=ketema@ketema.net) |
| 03:32.47 | neurosys | Ok. I found how to deativate the * to jump into voicemail. |
| 03:33.19 | neurosys | I can rename the voicemail sound files that give the advanced options :P |
| 03:34.00 | KyleK | advanced options? like press # for more options/ |
| 03:37.02 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 03:37.29 | KyleK | Please leave your message after the tone, when done hang up or press the pound key is all I get out of it |
| 03:38.10 | neurosys | once you log into the mailbox. I dont want the guests to be able to set a password. |
| 03:41.32 | KyleK | ah |
| 03:41.42 | KyleK | try setting the password to -1 |
| 03:42.42 | Naikrovek | who was i talking to earlier about building asterisk 1.6 on ubuntu 9? |
| 03:43.04 | KyleK | wasn't me |
| 03:43.17 | drmessano | wonders why people choose to abuse asterisk so |
| 03:43.24 | neurosys | ick |
| 03:43.33 | KyleK | do like apt-get install build-essentials libxml2-dev libncurses5-dev |
| 03:43.37 | Naikrovek | explain abuse, drmessano |
| 03:43.38 | neurosys | Although Tomorrow i think ill try asterisk on fbsd :) |
| 03:43.57 | Naikrovek | don't even try to sell me some rpm distro |
| 03:43.59 | KyleK | Naikrovek: ubuntu isn't known as a server distro |
| 03:44.14 | KyleK | <somebody> lfs or bust! |
| 03:44.15 | russellb | Ubuntu Server LTS ftw! |
| 03:44.21 | Naikrovek | says who? works flawlessly for me, and has for years |
| 03:44.26 | drmessano | KyleK: NUH UH, they have an ISO called "SERVER", so NUH UH |
| 03:44.38 | Naikrovek | NEVER had a problem with ubuntu, ALWAYS have a problem with rpm distros |
| 03:45.03 | KyleK | I've never had a problem with an rpm distro that wasn't me mixing rpms around |
| 03:45.05 | KyleK | :) |
| 03:45.06 | drmessano | wonders why most run Asterisk on RPM based distros |
| 03:45.07 | Naikrovek | ubuntu desktop i have issues with, but never server |
| 03:45.30 | Naikrovek | i get circular rpm dependencies virtually every time i try to upgrade an rpm distro |
| 03:46.19 | drmessano | Funny, never had that problem |
| 03:46.20 | KyleK | really? |
| 03:46.28 | KyleK | just specify both on the same like |
| 03:46.42 | KyleK | rpm -whatever thing1.rpm thing2.rpm |
| 03:46.44 | Naikrovek | E-bola: http://www.ideaglu.net/?p=1150 |
| 03:46.49 | KyleK | *line |
| 03:47.00 | drmessano | Of course, I build a PBX to be a PBX, not a clown car |
| 03:47.26 | drmessano | Don't try to shove random RPMs from google searched in it, trying to make it into a toy |
| 03:47.39 | neurosys | You guys ever heard asterisk on fbsd in production? |
| 03:47.41 | KyleK | what distro do you use btw |
| 03:47.58 | neurosys | Ive been usimng CentOS |
| 03:47.58 | Naikrovek | kylek: this is doing something like yum update |
| 03:48.02 | drmessano | CentOS FTFW4EVA |
| 03:48.21 | drmessano | Naikrovek: You're doing it wrong then |
| 03:48.27 | Naikrovek | <scottish accent>centos is CRAP!</scottish> |
| 03:48.31 | Naikrovek | drmessano: fresh install brah |
| 03:48.34 | KyleK | hehe |
| 03:48.39 | KyleK | CentOS: not scottish |
| 03:48.51 | Naikrovek | i know, but i can say it in a crappy scottish accent if i wanna |
| 03:49.15 | Naikrovek | apt-get is superior to your silly rpm tools |
| 03:49.26 | KyleK | IRL most of the people I know have heard the phrase "if its not scottish its CRAP" but dont pick up on me calling something not scottish |
| 03:49.37 | drmessano | I've got a 5.0 > 5.1 > 5.2 > 5.3 box that've run Asterisk 1.2, 1.4, 1.6, and god knows what else on.. and it Yum's just fine |
| 03:49.58 | KyleK | they're changing versions too fast |
| 03:50.10 | KyleK | I've got root on a 5.1 box |
| 03:50.16 | Naikrovek | drmessano: good for you. quality has fallen then because i tried to yum update a server today and the thing never came back up after a reboot. oops it forgot to install a new kernel |
| 03:50.26 | Naikrovek | at that point i had officially had it with rpm |
| 03:50.38 | neurosys | fbsd ports rock :) |
| 03:50.50 | Naikrovek | a new memo was drafted: no more rpms on my network or you'll find yourself with a missing server |
| 03:50.56 | Naikrovek | overreaction: perhaps |
| 03:51.12 | drmessano | Naikrovek: i doubt that's ever happened with another distro, and it's probably inherent to CentOS, seeing as how highly reported that sort of thing is |
| 03:51.56 | KyleK | no failsafe? |
| 03:52.09 | Naikrovek | well for years i used redhat, and i always had the same issues. by the time i installed redhat 5.2 (this is a decade ago) i installed EVERYTHING via source, post-install. i could never get rpms to work, and i never fiddled with anything |
| 03:52.36 | drmessano | and what was this box today running? |
| 03:52.53 | *** join/#asterisk davidandgoliath (n=David@out.clearnet.com) |
| 03:54.42 | *** join/#asterisk Linuturk (n=linuturk@unaffiliated/linuturk) |
| 03:55.40 | Naikrovek | drmessano: centos |
| 03:55.47 | drmessano | Centos what? |
| 03:55.57 | drmessano | 2, 3, 4, 5? |
| 03:56.02 | Naikrovek | oh 5 |
| 03:56.39 | Naikrovek | maybe my experiences are unique |
| 03:56.48 | Naikrovek | but i'm not going to mess with any rpms ever again |
| 03:57.14 | Naikrovek | even if it could mean compiling an entire distro from source if debian & ubuntu go away |
| 03:57.29 | KyleK | what about slackware |
| 03:57.37 | Naikrovek | never used it |
| 03:58.02 | Naikrovek | but the suse servers at work and the centos & redhat servers I've set up are all unupgradeable because of circular dependencies |
| 03:58.25 | KyleK | shudders |
| 03:58.30 | Naikrovek | most are pretty close to default install, no upgraded kernels or anything |
| 03:58.44 | KyleK | suse is scary |
| 03:58.48 | Naikrovek | just apache or tomcat or mysql or whatever |
| 03:58.49 | Naikrovek | yes it is |
| 03:58.51 | Naikrovek | i hate it |
| 03:58.55 | Naikrovek | HATE |
| 04:00.17 | drmessano | So everything you install comes from the CentOS repo? |
| 04:00.21 | drmessano | No outside RPMs? |
| 04:01.01 | KyleK | I have access to a susebox that got moved hardware and doesn't have DMA mode on the IDE hard drive |
| 04:01.25 | drmessano | No executive decisions to upgrade PHP here or MySQL there to an RPM you downloaded elsewhere that bites you in the ass later? |
| 04:02.10 | KyleK | been trying to get the sysadmin to get around to moving stuff away from suse but the guy wants to build a new universe first or something :( |
| 04:04.50 | Naikrovek | drmessano: no all repository rpms |
| 04:05.23 | Naikrovek | my luck is either astronomically bad or it happens to everyone |
| 04:05.29 | Naikrovek | either way i'm done with rpm :) |
| 04:05.51 | Naikrovek | if it works for you, please don't change your opinion because of me |
| 04:06.05 | Naikrovek | but i'm full debian/ubuntu from here on in |
| 04:06.23 | Naikrovek | which (thankfully) means no asterisknow or trixbox |
| 04:07.01 | KyleK | lol i dont think drmessano would change his opinion due to any human |
| 04:07.30 | drmessano | Especially because I am convinced it's entirely operator error on his part? |
| 04:07.49 | KyleK | i meant besides that |
| 04:08.18 | drmessano | I'm not much of a follower. Ubuntu sucks.. Wanna fight? |
| 04:08.25 | Naikrovek | i don't know how fresh install + yum update is operator error |
| 04:08.43 | Naikrovek | perhaps I forgot to sacrifice enough windows CDs |
| 04:09.23 | Naikrovek | or fresh install + yum install apache + yum update is operator error |
| 04:09.47 | drmessano | So you install base, install apache, yum update, and it croaks? |
| 04:09.57 | Naikrovek | yeah that happened to me today in a virtual machine |
| 04:10.29 | drmessano | O.o |
| 04:10.31 | Naikrovek | i was so pissed off i just deleted the vm and swore off rpm |
| 04:10.36 | Naikrovek | drmessano: my thoughts exactly |
| 04:10.52 | Naikrovek | i'm not stupid |
| 04:11.01 | Naikrovek | i've been administrating linux for a very long time |
| 04:11.14 | Naikrovek | since 1995 or 1996 |
| 04:11.32 | Naikrovek | daily, excluding vacations |
| 04:12.06 | KyleK | I've been admistrating windows since 1994 ;) |
| 04:12.32 | Naikrovek | yeah i got my winnt4 mcse soon after winnt4 came out |
| 04:12.39 | Naikrovek | which was 1 august 1996 |
| 04:12.55 | Naikrovek | so, i've been doing this for a while |
| 04:12.58 | KyleK | the wink is because i was 9 in 1994 |
| 04:13.03 | drmessano | I just ran a yum update on my home PBX and it didnt die.. |
| 04:13.04 | Naikrovek | hah |
| 04:13.10 | Naikrovek | drmessano: you're not cursed |
| 04:13.22 | Naikrovek | drmessano: that matters, apparently |
| 04:13.28 | drmessano | pats his Trixbox |
| 04:13.30 | drmessano | j.k |
| 04:14.05 | Naikrovek | it's weird; some people have similar experiences to me, and some don't. |
| 04:14.32 | Naikrovek | but i know i'm not the only one who's had issues like this |
| 04:14.38 | Naikrovek | or maybe they're lying to me |
| 04:14.45 | drmessano | I just googled, and you actually are |
| 04:14.49 | Naikrovek | maybe i'm on the Truman show and the producers are just royally fucking with me |
| 04:15.15 | drmessano | Apparently the VGC is IP based |
| 04:15.29 | drmessano | (Vast Global Conspiracy) |
| 04:17.16 | ReDNeQ | morning |
| 04:17.22 | Naikrovek | evening |
| 04:17.25 | Naikrovek | but hello |
| 04:17.27 | ReDNeQ | oh yah |
| 04:17.30 | ReDNeQ | still evening |
| 04:19.02 | KyleK | I'm surprised that the voicemail app doesn't save chan->cdr->uniqueid to its .txt for easy linking together with a list of calls |
| 04:19.58 | Naikrovek | my god virtual machines are wonderful |
| 04:20.14 | Naikrovek | yay vmware/hyper-v/virtualbox/xen |
| 04:21.03 | drmessano | I'm surprised chicks dont dig me more when I throw hot tea at them.. The world is twisted, bro |
| 04:21.34 | Naikrovek | yeah funny how that works...... |
| 04:21.52 | KyleK | I wonder if I can run linux in Windows Virtual PC |
| 04:21.59 | Naikrovek | took me a long damn time to learn that females do not WANT to be worshipped |
| 04:22.08 | Naikrovek | KyleK: yes, but the network driver is a bit wonky |
| 04:22.15 | KyleK | :-/ |
| 04:22.32 | KyleK | I was going to install centos and try updating it |
| 04:22.35 | Naikrovek | KyleK: you can do it in hyper-v easy, it offers a generic network card that linux detects fine |
| 04:22.48 | Naikrovek | KyleK: maybe it works now, with vpc2007 |
| 04:22.54 | Naikrovek | or you could download virtualbox: free |
| 04:23.05 | KyleK | thats worth a shot |
| 04:23.14 | Naikrovek | http://virtualbox.org/ |
| 04:23.28 | KyleK | i tried vmware with the windows 7 rc and it added a user to the login screen which kinda pissed me off |
| 04:23.42 | Naikrovek | added a user to the login screen? |
| 04:24.18 | KyleK | __vmware_sometihng showed up in the graphical login screen |
| 04:24.22 | Naikrovek | really |
| 04:24.28 | Naikrovek | that's never happened to me |
| 04:24.33 | Naikrovek | i use vmware all the time |
| 04:24.52 | Naikrovek | wonder how it happened |
| 04:24.57 | drmessano | On Windows 7? |
| 04:25.01 | Naikrovek | yeah win7 |
| 04:25.15 | Naikrovek | in fact(!) i have a win7 vm running in vmware right now |
| 04:25.27 | drmessano | You have an answer for everything |
| 04:25.36 | drmessano | Almost.. botlike |
| 04:25.36 | Naikrovek | not really |
| 04:26.00 | Naikrovek | i do have opinions which i attempt to pass off as facts from time to time |
| 04:26.30 | Naikrovek | but i have spent a lot of time fiddling with virtual machines. had to set up a VM server at work and I spent many, many hours researching and trying things out |
| 04:26.59 | Naikrovek | omg i love the might boosh |
| 04:27.11 | Naikrovek | s/might/mighty/ |
| 04:28.08 | *** join/#asterisk jql (n=jql@12.9a.344a.static.theplanet.com) |
| 04:30.26 | *** join/#asterisk mintos (n=mvaliyav@115.240.120.110) |
| 04:32.17 | Naikrovek | builds asterisk. |
| 04:35.16 | *** join/#asterisk littlerock (n=littlero@58.30.184.78) |
| 04:36.10 | littlerock | which sip softphone support G.729, I just want to check whether asterisk support G.729 module |
| 04:36.56 | littlerock | so if a trial-version sip softphone is ok |
| 04:37.02 | littlerock | s/if/ |
| 04:37.24 | *** part/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 04:37.38 | drmessano | No such thing, bro |
| 04:38.34 | *** join/#asterisk Linuturk (n=linuturk@unaffiliated/linuturk) |
| 04:39.29 | littlerock | bad |
| 04:42.36 | littlerock | so which free codec has the equivalent effect as g.729 for WAN network? |
| 04:43.25 | drmessano | What effect are you looking for? |
| 04:43.41 | drmessano | Wammby bar? Wah-Wah peddle? |
| 04:43.44 | drmessano | -b |
| 04:43.45 | littlerock | bandwidth consuming |
| 04:43.54 | drmessano | GSM is free and light |
| 04:43.58 | littlerock | and voice quality |
| 04:45.08 | Naikrovek | gsm can't transmit DTMF reliably in my experience |
| 04:45.17 | Naikrovek | so no entering voicemail passwords or any useful stuff like that |
| 04:45.39 | Naikrovek | G726 maybe? |
| 04:46.11 | littlerock | I found that speex has 13kbps payload |
| 04:46.13 | Naikrovek | i'm not entirely familiar with the codecs |
| 04:47.00 | drmessano | He wants a softphone, so he needs to take what he can get with asterisk + softphone limitations.. Speex probably would be a good choice if he gets it running |
| 04:47.15 | *** join/#asterisk RonDutt (i=4c1606e2@gateway/web/freenode/x-easvjbqzmiilcywx) |
| 04:47.22 | Naikrovek | yeah |
| 04:47.27 | Naikrovek | i missed the softphone part |
| 04:47.31 | RonDutt | Is it possible to lock extensions to a specific trunk? For instance, I want extension 100 to only use trunka (sip trunk), but extension 200 to use trunkb (iax trunk). right now everything goes out whichever trunk is free |
| 04:47.32 | Naikrovek | speex would work |
| 04:48.29 | Naikrovek | RonDutt: i think you can specify by destination, but i dunno about source |
| 04:48.48 | Naikrovek | RonDutt: but maybe you can. i just don't know how |
| 04:48.58 | Naikrovek | (see that, drmessano - i don't have an answer for everything) |
| 04:49.13 | Naikrovek | I went to a dinner party for all the planets. And uh Uranus -- he knocked up a spaghetti bolognese. But what happened was there wasn't any parmesan, and I said "I'm the moon, I'm made of cheese, why don't you grate off a portion of me head?" And they did, and it tasted of baby sick. |
| 04:49.14 | RonDutt | Naikrovek: damn...dont want to get two seperate boxes to handle home phone and work/other phone |
| 04:49.26 | Naikrovek | oops hah that wasn't supposed to go here |
| 04:49.30 | kaldemar | context is the answer |
| 04:49.39 | Naikrovek | it's from the might boosh, if you find it funny |
| 04:49.43 | Naikrovek | kaldemar: thanks |
| 04:50.00 | Naikrovek | RonDutt: context is the answer :) |
| 04:50.16 | RonDutt | Naikrovek: humm? |
| 04:50.38 | Naikrovek | you can grab the context (extension) and dial based on that |
| 04:50.50 | Naikrovek | i wonder... |
| 04:50.54 | Naikrovek | ~context |
| 04:50.55 | infobot | well, context is like LaTeX but less messy and more oriented to DTP instead of academics. |
| 04:51.09 | Naikrovek | not what i was talking about, infobot, but nice try nonetheless |
| 04:51.21 | RonDutt | lol |
| 04:51.56 | RonDutt | Naikrovek: so right now its at from-internal, I would change that to the trunk name? |
| 04:52.43 | Naikrovek | are you using trixbox |
| 04:53.35 | RonDutt | Naikrovek: just to simplify things/get the gist of. Moving to asterisk standalone once I figure out how. |
| 04:54.03 | kaldemar | RonDutt: you can change context to whatever you want |
| 04:54.06 | Naikrovek | that's fine, i don't care, was just curious. "from-internal" seems to be a trixbox thing |
| 04:54.37 | RonDutt | kaldemar: so the context in my trunk right now is from-pstn, I would change that to like trunka and set the context for the extension to trunka? |
| 04:54.51 | kaldemar | RonDutt: and you can put whatever extensions to contexts. and remember that extensions are the "exten =>" lines in asterisk, phones are devices. |
| 04:55.54 | kaldemar | RonDutt: no, the context set for a device/channel is only for incoming calls, from asterisk's point of view. |
| 04:56.44 | kaldemar | RonDutt: if you want some phone to only use a certain trunk, you include the certain trunk in its context, not all. |
| 04:57.23 | kaldemar | read about dialplans, you need to understand the basics better. |
| 04:57.26 | kaldemar | ~book |
| 04:57.27 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 04:58.41 | RonDutt | Wow, didn't know the book was free |
| 05:00.26 | justdave | hmm, if I have a multi-homed box with asterisk running on it, is there a way to get asterisk to use a different sourceIP in the SIP packets depending on which interface it needs to use to reach the destination? |
| 05:00.53 | justdave | like can I statically assign the source IP on each trunk in sip.conf or something? |
| 05:01.00 | RonDutt | kaldemar: ah, pretty simple :) thanks! Going to test now |
| 05:03.51 | justdave | the docs don't seem to indicate that bindaddr or externip can be used in a user/peer/friend definition... |
| 05:04.57 | *** join/#asterisk jgoo (n=floops__@ppp66-112.adsl.forthnet.gr) |
| 05:06.00 | KyleK | externip is used for nat, like you have eth0 set to 192.168.1.12 and asterisk can look at that and realize its public ip is like 4.2.2.2 |
| 05:08.40 | justdave | right, and it's NATed on the firewall, so it needs that. |
| 05:08.50 | justdave | it's already set, for the first interface |
| 05:08.57 | justdave | the new one will be different though :( |
| 05:10.02 | justdave | oh... but the new interface is going to be directly outside instead of NATed... does externip get ignored if the interface the route goes through has an IP that's not in localnet? |
| 05:10.49 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 05:10.59 | Naikrovek | justdave: won't just basic linux routing handle this? it's not going to send out a packet meant for a 192.168 address out the public ip interface |
| 05:11.02 | KyleK | well its the interface you need to change |
| 05:11.21 | KyleK | multihomed as in multiple default routes? |
| 05:11.23 | justdave | Naikrovek: basic linux routing doesn't control the internal contents of SIP packets. |
| 05:11.50 | justdave | KyleK: no, the second interface will be static routes for specific IP addresses |
| 05:12.18 | KyleK | hrm |
| 05:12.28 | Naikrovek | i don't understand why you want to fiddle with the packets on the way out |
| 05:12.38 | KyleK | justdave: you dont need to futz with astersik |
| 05:12.44 | justdave | Naikrovek: I don't. I want asterisk to set them correctly to begin with |
| 05:12.47 | Naikrovek | this is a routing thing |
| 05:12.48 | KyleK | its in the routing and you need something on the outgoing packets |
| 05:13.01 | KyleK | p-somethng routing |
| 05:13.34 | Naikrovek | if a UDP packet is destined for public internet linux will put it there; if destined for the lan, linux will put it out that interface. |
| 05:13.39 | Naikrovek | all by itself, no configuration |
| 05:13.59 | justdave | Naikrovek: I'm not talking UDP, I'm talking SIP. |
| 05:14.04 | Naikrovek | sip = udp |
| 05:14.04 | KyleK | well with multihoming you need to have the outgoing packets going out on the right interface assuming its different ips on the interface |
| 05:14.04 | justdave | next layer up |
| 05:14.17 | justdave | udp is a lower layer in the network stack. |
| 05:14.25 | Naikrovek | sip is not aware of what interface it is going out, nor is it aware of the ip address it's going ot |
| 05:14.25 | Naikrovek | to |
| 05:14.29 | KyleK | justdave: is your udp layer traffic being routed properly though? |
| 05:14.54 | justdave | you guys need to learn how SIP works. It's the reason people have problems with NAT on asterisk all the time. |
| 05:15.08 | justdave | why do you think it needs to have externip defined in sip.conf if you're NATed? |
| 05:15.22 | drmessano | ~sipnat |
| 05:15.23 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 05:15.26 | justdave | because the *content* of the SIP packet (not the routing envelope) contains the IP the remote end is supposed to send RTP traffic to. |
| 05:15.31 | drmessano | Takes 5 mins |
| 05:15.56 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 05:15.58 | justdave | that IP address in the *content* of the packet needs to be set correctly. |
| 05:15.59 | Naikrovek | justdave: people have problems with NAT because SIP by default doesn't go through the asterisk server, Asterisk connects the two phones to each other then gets out of the way. NAT turns this off, and funnels everything through asterisk |
| 05:17.05 | KyleK | anyways |
| 05:17.23 | Naikrovek | seriously, is it us or him who doesn't understand sip well enough |
| 05:17.44 | KyleK | whats the term im looking for to set up multihoming properly |
| 05:17.50 | justdave | so to restate my situation, I have an interface with internal address space that's NATed to a public IP. externip is set to the public IP. Asterisk puts externip as the address in the packet content when it sends packets to external addresses. |
| 05:18.01 | justdave | that's all set up fine and working |
| 05:18.10 | KyleK | good |
| 05:18.12 | justdave | I'm adding a new interface that will have static routes for specific IPs |
| 05:18.20 | KyleK | whats the damn thing called so I can note it down? :) |
| 05:18.33 | justdave | which means asterisk needs to set a *different* source address in the packets that go out that interface. |
| 05:19.04 | justdave | basically, I need to tell it a different externip for that sip trunk |
| 05:19.57 | KyleK | oh |
| 05:20.00 | KyleK | fromdomain=thatip |
| 05:20.29 | KyleK | well, i guess thats not quite enough |
| 05:21.57 | Naikrovek | i have the exact same situation as justdave and i didn't have to do anything. two interfaces, one is NAT, one is public. it works fine. |
| 05:22.09 | Naikrovek | no custom routes |
| 05:22.13 | justdave | although that's where I was realizing... the second interface is actually not NATed, so that's when I was musing about whether asterisk would ignore externip if the interface it's sending on isn't listed in localnet |
| 05:22.27 | Naikrovek | i'm too tired to grok that further |
| 05:22.39 | KyleK | justdave: see if you can localnet the destination network |
| 05:23.40 | justdave | KyleK: that's a thought. I'll keep that in mind for something to try when we get it set up (I'm still researching right now and want to make sure it works before we waste energy on it. :) |
| 05:24.40 | drmessano | Asterisk will send externip to everyone but localnet= |
| 05:25.05 | justdave | and is it smart about which interface is in use when sending to stuff that is in localnet? |
| 05:25.16 | Naikrovek | justdave: linux handles that, and yes |
| 05:25.28 | Naikrovek | justdave: linux knows not to send lan stuff out to the internet |
| 05:25.30 | justdave | i.e. if linux routes it over the second interface, asterisk won't put the first interface's IP address in it? |
| 05:25.32 | drmessano | Of course.. |
| 05:25.41 | KyleK | justdave: your multihoming setup will handle which IP the packet goes out to the internet from |
| 05:25.57 | justdave | it requires asterisk to know which interface is actually going to get used to send it |
| 05:26.01 | justdave | before it gets to linux |
| 05:26.58 | *** join/#asterisk AeroCloud (n=aero@ip98-165-113-125.ph.ph.cox.net) |
| 05:27.04 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
| 05:27.11 | KyleK | or just use iptables to change the source IP |
| 05:27.21 | KyleK | (if needed) |
| 05:27.37 | justdave | KyleK: I'm sure the UDP packet will be sourced correctly |
| 05:27.46 | justdave | I'm talking about the RTP destination address inside the packet |
| 05:28.10 | Naikrovek | i think you just know too much about sip to trust it. download asterisknow or trixbox, install on a virtual machine, add your network interfaces, and try |
| 05:28.14 | Naikrovek | will take 30 minutes |
| 05:28.20 | justdave | it doesn't have to match, because as you pointed out earlier, you could well be sending them the IP address of the phone on the other end, rather than the asterisk server. |
| 05:28.57 | justdave | linux isn't going to care what it is, or try to change it on you, it's all in the data asterisk puts in the packet. |
| 05:28.57 | kaldemar | externip is unfortunately a global parameter |
| 05:29.46 | Naikrovek | i have an asterisk server, two interfaces. one is 192.168... one is public internet. asterisk has zero issues. both interfaces can reach the internet |
| 05:30.15 | AeroCloud | I have a question, When you hear 4 chimes after a ring from a dial command, does that mean the trunk has issues? Trunk was working earlier, then I changed dialplan a little, stopped working, changed dialplan back... still not working. |
| 05:31.01 | Naikrovek | AeroCloud: will need to see a log to figure that one out |
| 05:33.40 | Naikrovek | goes to sleep |
| 05:34.10 | AeroCloud | nevermind I got it working... network issue :) |
| 05:34.55 | TJNII | I spent the afternoon fighting one of those. |
| 05:35.11 | *** join/#asterisk fiddur (n=fiddur@dhcp08.textalk.com) |
| 05:35.20 | TJNII | Turns out my brilliant idea to virtualize my servers ... wasn't. |
| 05:35.42 | Naikrovek | wakes up |
| 05:35.42 | Naikrovek | why not |
| 05:36.06 | TJNII | Morning |
| 05:36.20 | Katty | grumbles |
| 05:36.37 | Katty | you know, it's not very often that i get so agitated that i can't sleep. |
| 05:36.49 | Naikrovek | TJNII: why were virtual machines a non-brilliant idea |
| 05:36.49 | Katty | but dagnabbit, i am irritated. |
| 05:36.55 | Naikrovek | Katty: what's up |
| 05:37.24 | Katty | Naikrovek: explain to me, why a teacher would refuse to show a presidental speech to a child. |
| 05:37.25 | KyleK | didn't do it |
| 05:37.34 | Katty | Naikrovek: explain this. |
| 05:37.39 | Katty | Naikrovek: it's somehow beyond me |
| 05:37.43 | Naikrovek | Katty: i can't. pisses me off too. |
| 05:37.45 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 05:38.03 | KyleK | like obama on the air during math class? |
| 05:38.08 | Katty | i could understand if there were like...flying body parts. |
| 05:38.10 | Naikrovek | Katty: racism, agenda, lie detection, ignorance come to mind immediately |
| 05:38.28 | Katty | but i am pretty damn sure that some of the things these kids watch is far worse than a presidental speech. |
| 05:38.35 | Naikrovek | like "oh man my mom has been lying to me about obama" is what i mean by lie detection |
| 05:38.39 | TJNII | Naikrovek: For whatever reason the virtual box couldn't handle IPtables properly. Sometimes I'd start it and have no connection to the net, but mostly it would randomly drop packets, decide certain calls didn't need RTP, and similar nonsense. |
| 05:38.40 | Katty | secondly. what is this REALLY telling the kid? |
| 05:38.59 | Katty | it's not telling the kid oh this guy has bad morals!! |
| 05:39.06 | Naikrovek | TJNII: what type of network did you use? NAT, bridged? |
| 05:39.08 | Katty | It's telling the kid that the presidency is a JOKE. |
| 05:39.16 | Naikrovek | Katty: we agree |
| 05:39.19 | TJNII | Naikrovek: Bridged. The box had a public IP |
| 05:39.27 | Katty | which, granted, sometimes it is. but it is still deserving of respect. |
| 05:39.33 | Katty | Naikrovek: yes, just ranting here. |
| 05:39.39 | Naikrovek | TJNII: really. |
| 05:39.39 | Katty | and THIRDLY. |
| 05:39.39 | KyleK | Katty: so what are you talking about? |
| 05:39.51 | KyleK | (I'm in canada and dont watch news) |
| 05:39.52 | Katty | let's think about who runs the public school system. |
| 05:39.57 | Katty | oh hmm. |
| 05:39.58 | TJNII | Naikrovek: The internal nic worked fine. The firewalled external nic had bad juju. |
| 05:39.59 | Katty | let me think |
| 05:39.59 | Katty | hmm |
| 05:40.02 | Katty | THE GOVERNMENT |
| 05:40.33 | Naikrovek | Katty: wow you're upset about this. don't waste too much energy getting tweaked, and vow to treat your children better. |
| 05:40.39 | Katty | KyleK: Obama wanted to give a quick speech to the children of the united states about staying in school...staying away from drugs, so on and so forth. |
| 05:40.39 | Naikrovek | if/when you have any |
| 05:41.02 | Katty | KyleK: instead, parents and teachers come unglued, because Obama is addressing the youth of our nation without their permission. |
| 05:41.20 | Naikrovek | Katty: yeah, and they said nothing when reagan or bush sr. did the same |
| 05:41.21 | Katty | Naikrovek: yes. i am extremely upset. |
| 05:41.36 | Katty | Naikrovek: because i know the message that it sends to children. |
| 05:41.40 | trnzmeta | go the repubs!!! |
| 05:41.45 | Katty | Naikrovek: and children are so innocent, and believe so blindly. |
| 05:41.48 | Naikrovek | Katty: [TK]D-Fender and i went on about this for a while this morning |
| 05:41.50 | TJNII | I'm convinced that a very vocal majority see the president as nothing but *ahem* an uppity nigger. That makes me sad. |
| 05:41.52 | Naikrovek | Katty: i know |
| 05:41.52 | trnzmeta | down with lefty crap! |
| 05:41.56 | drmessano | Katty: Too bad he told kids to stay in school.. With all that build-up, I would have gone ahmadinejad on them told them out to gut the kid next to them with a spork |
| 05:42.06 | Naikrovek | TJNII: i think so too |
| 05:42.26 | Katty | Naikrovek: and if their own PARENTS can't even teach them to think freely and make decisions on their own... |
| 05:42.30 | TJNII | Heh, I meant to say minority, I hope it isn't a majority. |
| 05:42.37 | Corydon76-dig | Could we please take this over to #politics and leave this channel to Asterisk? |
| 05:42.45 | Katty | Corydon76-dig: you're right. |
| 05:42.50 | Naikrovek | everyone knows that the truth has a hard liberal bias, and some people just can't handle the truth |
| 05:42.59 | *** join/#asterisk maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 05:43.00 | Katty | this conversation doesn't belong in here (= |
| 05:43.03 | Katty | I'm just ranting. |
| 05:43.05 | Naikrovek | wow Corydon76-dig came out of nowhere |
| 05:43.17 | TJNII | Mods tend to do that. |
| 05:43.20 | Naikrovek | i guess |
| 05:43.38 | Naikrovek | TJNII: so what virtual machine did you use? vmware, virtualbox? other? |
| 05:43.40 | AeroCloud | I have another question, I have trixbox setup with asterisk 1.6.0.9, I am creating a custom AGI script that I want to default to on EVERY incoming call on ANY did. What would the dialplan, context etc be in a custom file? |
| 05:43.49 | Katty | but i'm still grumpy. |
| 05:43.53 | Katty | sits, grumps. |
| 05:43.55 | drmessano | ~trixbox |
| 05:43.56 | infobot | it has been said that trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/. We do not recommend using it. |
| 05:44.12 | Corydon76-dig | hugs Katty |
| 05:44.16 | AeroCloud | I dont want to support trixbox |
| 05:44.20 | Katty | hugs Corydon76-dig |
| 05:44.21 | AeroCloud | I want to bypass it |
| 05:44.30 | Katty | Corydon76-dig: it makes me all sad inside :< |
| 05:44.37 | Naikrovek | AeroCloud: he's saying that we don't know enough about it to help you with it |
| 05:44.38 | AeroCloud | use trixbox for adding did's etc, but bypass it |
| 05:44.45 | drmessano | Your argument is nonsenical.. |
| 05:44.55 | TJNII | Naikrovek: VirtualBox. My opinion is that VMWare 2.0 is a steaming pile. (Or at least its web interface) |
| 05:45.07 | KyleK | AeroCloud: someone elses bright idea to be running trixbox in the first place? |
| 05:45.15 | AeroCloud | yes, I'm just a developer |
| 05:45.22 | drmessano | If you want to bypass Trixbox, reformat |
| 05:45.35 | AeroCloud | they like the interface to add did's, trunks etc |
| 05:45.40 | drmessano | Otherwise, you're asking for Trixbox centric help |
| 05:45.48 | AeroCloud | I need to figure out a way to bypass all the crappy dialplans they use |
| 05:45.50 | drmessano | Sheeple, please |
| 05:45.55 | TJNII | Naikrovek: Other than the asterisk issues, it worked fine. The Apache, Qmail, and DHCP boxes I made all seemd okay. However this problem and other assorted startup / shutdown oddities made me decide not to go this route. |
| 05:46.38 | AeroCloud | Well I will figure out how to bypass it, and I'll let you guys know if someone else needs to know. |
| 05:46.54 | kaldemar | AeroCloud: that requires you to stab trixbox or use something that overruns it's configurations. either way, you're asking in the wrong place. |
| 05:46.58 | AeroCloud | just was trying to save a few hours :) |
| 05:47.07 | Naikrovek | TJNII: i'm going to have to test this now. i've tested asterisk under virtual machines, placed calls and all, not had issues. |
| 05:47.15 | Naikrovek | TJNII: but i used vmware workstation for those tests |
| 05:47.25 | AeroCloud | kaldemar: I understand, just wondered if anyone has done it before thats all :) |
| 05:47.27 | KyleK | AeroCloud: grep extensions.conf for "include" and look at those files and extensions.conf :) |
| 05:47.31 | TJNII | Naikrovek: Like I sad, the only issue was firewall related. Otherwise, * ran fine. |
| 05:47.40 | Naikrovek | ah firewall |
| 05:47.43 | Naikrovek | that i've not done |
| 05:47.47 | Naikrovek | i despise iptables |
| 05:47.53 | TJNII | Naikrovek: Yea. I need it with the public IP and all. |
| 05:48.15 | TJNII | I use shorewall, but that is nothing but a wrapper for iptables. |
| 05:48.38 | AeroCloud | currently I use [from-pstn-custom] |
| 05:49.00 | Naikrovek | TJNII: thankfully i don't need to worry about that too much. i have an ASA and I can firewall everything there AND give my machines public addresses |
| 05:49.02 | AeroCloud | but when the did exists, it doesnt care about my s,1 |
| 05:49.24 | TJNII | Naikrovek: Calls to phones behind NAT (so no firewall): fine. Calls outside (so through firewall): crapshoot. |
| 05:49.28 | drmessano | AeroCloud: If you want to bypass the dialplan, the best place to ask are the Trixbox forums |
| 05:49.28 | TJNII | Naikrovek: I don't have those toys yet :) |
| 05:49.28 | AeroCloud | if I comment out the did's that trixbox adds in the extensions_additional.conf |
| 05:49.32 | AeroCloud | then it works just fine |
| 05:49.34 | Naikrovek | TJNII: did you google for it at all? maybe someone found a fix |
| 05:50.00 | Naikrovek | TJNII: you'll have great luck with ESXi server, I guarantee it |
| 05:50.30 | Naikrovek | i have inherited some machines with iptables running, I converted them to virtual machines, and they work fine |
| 05:50.36 | Naikrovek | but i've not done what you just tried |
| 05:51.07 | TJNII | Naikrovek: No. The main reason I did it was because I wanted to, essentially, whitelist the * box to only known IPs. At the time, I thought that would be system wide so I had to break * from Apache and Qmail. That turned out not to be true, so I just turned * back on on my old server with new firewall rules. |
| 05:51.34 | TJNII | Naikrovek: Plus, the server I was using for VirtualBox was loud as hell so I was kind of glad to turn it back off. |
| 05:52.01 | Naikrovek | ah |
| 05:52.03 | Naikrovek | heh |
| 05:52.36 | Naikrovek | okay |
| 05:52.45 | Naikrovek | virtualization talk done. /me returns to sleep |
| 05:52.46 | TJNII | Naikrovek: Another nail in the coffin was the server is quad core but doesn't support virtualization extensions, so all my guests had to be single core. That kind of screwed up my plans and left that box way underutilized. |
| 05:53.12 | *** join/#asterisk ltd_wk (i=z@patwk.transact.net.au) |
| 05:53.19 | Naikrovek | whoa wait a second |
| 05:53.25 | Naikrovek | TJNII: what's the processor |
| 05:53.33 | TJNII | Intel Xeon |
| 05:53.38 | Naikrovek | speed? |
| 05:53.43 | TJNII | 2.4Ghz |
| 05:53.54 | Naikrovek | okay so a recent one |
| 05:54.18 | Naikrovek | it supports VT, it must be the motherboard that doesn't. or maybe it's turned off in the bios |
| 05:54.45 | TJNII | There is no virtualization options in the BIOS, cpuflags show VME but not VMX, and VirtualBox complains loudly about no VT-x |
| 05:54.51 | TJNII | I'm guessing mobo. |
| 05:54.54 | Naikrovek | yup |
| 05:55.00 | Naikrovek | it's the motherboard |
| 05:55.30 | Naikrovek | that's gotta sting a little |
| 05:55.40 | TJNII | Yep |
| 05:55.56 | Naikrovek | esxi server 3.5 on that server would do fine. wouldn't be able to run 64-bit vms but 32 would work fine |
| 05:56.08 | TJNII | I'd use it as my desktop if it wasn't so gosh-darned loud. It is a IBM 1U server, so the fans really roar. |
| 05:56.14 | Naikrovek | aah yeah |
| 05:56.26 | Naikrovek | i have an ibm bladecenter i use for all my VMs |
| 05:56.29 | Naikrovek | well, most of them |
| 05:56.33 | TJNII | I might squirrel it off somewhere if I can find a spot and use X over SSH, though. |
| 05:56.35 | Naikrovek | dual quad core, 32GB RAM |
| 05:56.41 | TJNII | Noice. |
| 05:56.54 | Naikrovek | yeah i have .. 10? maybe 12 VMs on it |
| 05:57.04 | Naikrovek | told my CEO this and he about did a backflip |
| 05:57.21 | Naikrovek | "you have 12 servers running on one server??" yup |
| 05:57.37 | TJNII | Heheh. |
| 05:57.40 | Naikrovek | "saved you about $60k in hardware." |
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| 06:08.00 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 06:08.09 | Gokee2 | Well Naikrovek and everyone else earlier thanks for your help, I am gonna go ahead and order the TDM410P on Ebay (with two FXO's and two FXS's) three of Polycom Soundpoint IP 321's from telephonydepot and the TRENDnet PoE switch from newegg. Hopefully I will be able to make everything work when it arrives |
| 06:08.22 | trnzmeta | Naikrovek: do you use any for parrallelisation on the vm? |
| 06:09.15 | Qwell | Gokee2: DO NOT buy off ebay. It will almost certainly be a crap clone. |
| 06:09.34 | [TK]D-Fender | Qwell: I believe we confirmed that already |
| 06:10.30 | TJNII | Gokee2: My experience with VoIP stuff and eBay: If it's not sold in a lot, don't buy it. It is probably busted. |
| 06:10.41 | Gokee2 | Qwell, Hmm I am getting the http://cgi.ebay.com/TDM410P-Asterisk-card-2FXO-2FXS-port-FXO-card-TDM400P_W0QQitemZ150365890385QQcmdZ its a LOT cheaper then the Digium |
| 06:10.52 | TJNII | has had about a 50% success rate with eBay and phones. |
| 06:10.54 | Qwell | It's also a clone. |
| 06:10.57 | Qwell | A cheap one at that. |
| 06:11.02 | Qwell | Wonder why it's so cheap. |
| 06:11.05 | Qwell | rolls his eyes |
| 06:11.37 | [TK]D-Fender | Qwell: We chatted about these guys. (tm) infringing WIKI spamming fuck-offs |
| 06:11.48 | [TK]D-Fender | Qwell: Based on the old PCI design as well |
| 06:11.51 | Qwell | I know this seller in particular too. |
| 06:11.57 | [TK]D-Fender | Qwell: yup |
| 06:12.15 | Gokee2 | Qwell, Hmm? Have you tried one of there cards? |
| 06:12.17 | Qwell | They are cheap clones. |
| 06:12.21 | Qwell | I don't *have* to try them. |
| 06:12.44 | [TK]D-Fender | Gokee2: They are known (offenders), and if things go wrong, expect that you won't get much if any vendor support |
| 06:12.45 | Katty | feels better. |
| 06:12.58 | Qwell | and ebay will not help you either. *shrug* |
| 06:12.59 | Qwell | GL |
| 06:13.16 | Qwell | (also, neither will we - just throwing that out there) |
| 06:13.30 | Katty | pokes Qwell |
| 06:13.53 | [TK]D-Fender | Gokee2: Well may try and hlp with the basic configs, but once things start going wrong you're likely to find yourself up a creek |
| 06:14.35 | Gokee2 | Hmm, well I don't like the idea of paying 200+ for a card, are there any other options in the 100ish price range? |
| 06:14.58 | Qwell | Get a more realistic budget. |
| 06:15.03 | jgoo | Hi people, it is that time again, I am looking for the best way to model a human requirement in the system - they want one phone to ring only, but flash caller id up on all phones - is that possible / supported / easy with any combinations? |
| 06:15.13 | Qwell | Gokee2: Go price a Cisco. Ask for more. |
| 06:15.16 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 06:15.39 | [TK]D-Fender | Gokee2: These things have a price. I want a Ferrari for $10, but when I get the one I ordered from eBay I'll be betting it even has a FAKE "Matchbox" sticker on it |
| 06:16.03 | [TK]D-Fender | jgoo: Depends on the phones |
| 06:16.24 | TJNII | [TK]D-Fender: Nice analogy. |
| 06:16.28 | jgoo | [TK]D-Fender, "is that possible / supported / easy with any combinations?" |
| 06:16.37 | [TK]D-Fender | jgoo: You can silent-ring some phoens in a multi-dial |
| 06:16.44 | [TK]D-Fender | jgoo: those will get teh CID |
| 06:16.54 | [TK]D-Fender | jgoo: and have certain ones ring audible |
| 06:16.58 | Gokee2 | Hmm |
| 06:17.02 | jgoo | [TK]D-Fender, any combinations - I've thought about silent ring options for 'internal calls' and set them in the ring group.. |
| 06:17.16 | [TK]D-Fender | jgoo: Snom, Polycom, etc should all be capable of this |
| 06:17.30 | jgoo | [TK]D-Fender, that was my thinking, but I wondered, I read about a text-protocol for these phones too, but seems like hassle |
| 06:17.57 | [TK]D-Fender | jgoo: Cisco's you can "push" XML to. Not sure about how east any other is. |
| 06:17.59 | jgoo | [TK]D-Fender, is there an open source hardphone? |
| 06:18.11 | jgoo | Polycoms have this too |
| 06:18.37 | jgoo | well, a few years ago when I looked, I saw I could write up something |
| 06:18.45 | [TK]D-Fender | jplank: I've never touched Polycom's (new) push-capable stuff. |
| 06:18.57 | Gokee2 | So I would be better off with something like http://cgi.ebay.com/Digium-TDM410P-with-2-FXO-ports_W0QQitemZ300346351007QQcmdZViewItemQQptZLH_DefaultDomain_0?hash=item45ee099d9f&_trksid=p3286.c0.m14? |
| 06:19.01 | [TK]D-Fender | jgoo: They have PULL stuff sure, but no push |
| 06:19.10 | jgoo | really, hardphones suck. none of them are good not one. from what I have seen. some may be better than others, but take the best of the best, and it is really crap, and not 10% of what it could be. |
| 06:19.38 | [TK]D-Fender | Gokee2: Looks legit |
| 06:20.36 | Gokee2 | is slightly worried that after over $500 he is gonna get the system up and there will be sound quality issues |
| 06:21.12 | [TK]D-Fender | Gokee2: Of course its under $260 new |
| 06:21.27 | jgoo | [TK]D-Fender, why does this industry attract such attrocious programmers, tools, hardware and other things? |
| 06:21.38 | [TK]D-Fender | Gokee2: Actually you are running up those chances in the LOWER bracket |
| 06:21.51 | [TK]D-Fender | jgoo: Let me put this gently. |
| 06:21.53 | Gokee2 | [TK]D-Fender, Not once I get the phones |
| 06:22.03 | [TK]D-Fender | jgoo: IT'S A FUCKING PHONE! |
| 06:22.18 | jgoo | [TK]D-Fender, that doesn't quite make sense in the context of what I am saying at all |
| 06:22.29 | jgoo | [TK]D-Fender, why don't you put it clearly, and let me worry about if you are being harsh |
| 06:22.35 | [TK]D-Fender | jgoo: it isn't a web-browser, coffe machine, and automatic stapler |
| 06:23.00 | jgoo | [TK]D-Fender, ??? still, your point is lost on me, are you trying to say it is simpler / more complex / more or less important? |
| 06:23.06 | [TK]D-Fender | jgoo: And this industry has all sorts of great programmers. Look at Qwell here! |
| 06:23.12 | Qwell | ha |
| 06:23.16 | Qwell | oh, you were serious |
| 06:23.42 | [TK]D-Fender | "borrows' Qwell's chan_skinny bot-net for some fun.... |
| 06:23.53 | [TK]D-Fender | Qwell: Never forgotten! |
| 06:23.56 | Qwell | I don't think any of those were my fault :p |
| 06:24.00 | jgoo | [TK]D-Fender, you've lost me - I still don't get what you are trying to say, are you saying the phones are unimportant? |
| 06:24.35 | [TK]D-Fender | jgoo: No, I'm saying expecting phones to be so uber flexible is unnatural. They are computers in a sense, but only so loosely |
| 06:24.56 | [TK]D-Fender | jgoo: Expecting them to offer the kitchen sink will lead to all sort of instability & disappointme |
| 06:25.00 | [TK]D-Fender | tnt |
| 06:25.12 | jgoo | [TK]D-Fender, so far, I've not found one single asterisk setup that can actually replace an almost default install of an old dusty PBX |
| 06:25.24 | jgoo | Just because of call pickup handling |
| 06:25.29 | [TK]D-Fender | jgoo: Polycom's are already rather fewature-rich and pretty darn stable for it. |
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| 06:25.47 | [TK]D-Fender | jgoo: Guess we'd have to see the list of complaints. |
| 06:25.56 | jgoo | [TK]D-Fender, I think they don't need features as such, as asterisk support for things to do with... handling calls |
| 06:26.17 | [TK]D-Fender | jgoo: * is not a Key system so if thats the point of comparison I supposed a Ferrari is shit if being a 4-door is a requirement |
| 06:26.29 | [TK]D-Fender | jgoo: What shortcomings? |
| 06:26.45 | jgoo | I am writing them |
| 06:27.12 | [TK]D-Fender | jgoo: with * we have hold, transfers, conferencing (n-way + Meetme). |
| 06:27.40 | [TK]D-Fender | jgoo: multiple pickups, parking, presence... |
| 06:28.01 | jgoo | [TK]D-Fender, can't have one phone ring, but notify all phones of a call / second call - no default audio notification for second incoming call, (listing hold, transfer, conferencing isn't really helping, I know, I am talking about... how to pickup a phone and answer a call) |
| 06:28.17 | jgoo | There are two more scenarios that I cannot accurately define right now, my brain is foggy |
| 06:28.44 | [TK]D-Fender | jgoo: Deafult audio? This is up to the phone for CW beep, ext ring, etc. |
| 06:28.56 | [TK]D-Fender | jgoo: Polycom offers both & more. |
| 06:29.10 | [TK]D-Fender | jgoo: and you CAN "notify" by way of presence <- |
| 06:29.14 | jgoo | [TK]D-Fender, some things are going to be on the pbx or phone side, I am being general |
| 06:29.19 | [TK]D-Fender | jgoo: without actually sending the call. |
| 06:29.46 | [TK]D-Fender | jgoo: Yes, well if you are trying to compare * you need to be specific so the points can be validated |
| 06:30.47 | jgoo | [TK]D-Fender, I was in here the other day, asking how I can make a system work in a certain way. I was explicitly clear (about picking up a line - the terminology was confusing, but it was a combination of BLF, incoming call and sending the call to all phones) |
| 06:31.02 | [TK]D-Fender | jgoo: "[TK]D-Fender, so far, I've not found one single asterisk setup that can actually replace an almost default install of an old dusty PBX" <- well for this not to be a random shelling we'd need some real points to debate |
| 06:31.05 | jgoo | But I was shot down with 'no, you don't want to do that'. which was weird, because I did... |
| 06:31.31 | jgoo | can't have one phone ring, but notify all phones of a call / second call |
| 06:31.45 | jgoo | That one ^ the thing I was trying now, I guess I need to setup BLF and have one extension ring, and then pickup by pressing that extension BLF |
| 06:32.06 | [TK]D-Fender | jgoo: I can't say if I'd agree with their answer. Perhaps they felt it was sloppy or wasteful. Perhaps it was and you want it like that anyway. Some people just want things a certain way even if no other person could love it |
| 06:32.22 | jgoo | [TK]D-Fender, true, and these people were stuck in their ways |
| 06:32.42 | [TK]D-Fender | jgoo: The ones commenting against you? Or the ones asking this of you? |
| 06:32.56 | jgoo | But, I still feel things are not quite as they should be. The ones asking this of me |
| 06:33.32 | [TK]D-Fender | jgoo: Yes, customers can be a retarded bunch. Once some dead-end toaster offers them one way of thinking they can't think of any other way. |
| 06:33.49 | Gokee2 | Would caller id work with the Digum card and the phones I was talking about? |
| 06:33.50 | jgoo | and perhaps those commenting against me - because you cannot argue against a user requirement. It was something as basic as 'we need to hear a ringing noise, and be able to push a button, and then something else' and this amounted to not quite fitting with ring groups or queues or something else (and I was talking about muted rings on all phones) |
| 06:34.02 | jgoo | [TK]D-Fender, I see that in my real work |
| 06:34.05 | [TK]D-Fender | jgoo: While they are indeed the customer it doesn't mean you can't privately consider them back-water fuck-offfs :) |
| 06:34.36 | AeroCloud | If I used _.,1 it bypasses trixbox.. so dont use s,1 in the custom... :) |
| 06:34.46 | [TK]D-Fender | jgoo: There is a question of your capacity to steer their choices. You sell ideas as much as hardware, etc... |
| 06:34.49 | jgoo | Systems development - I've had DGs at the EU come up with the most insane batshit ideas, and... it gets stuck into the 2inch fat proposal... and causes us to flip the whole architecture on its ass |
| 06:35.21 | [TK]D-Fender | jgoo: "shit happens" <- official appendix |
| 06:35.47 | jgoo | [TK]D-Fender, no choices - they have one system, it has wires hanging out of every hole. They want the exact same thing, but without all the wiring (and add some extensions etc, which is now impossible, because nobody knows how to even open this ancient pbx thing...) |
| 06:36.55 | jgoo | [TK]D-Fender, yeah. OK. So, I am trying a new setup today - just curious about a few things, I am trying PIAB, *Now and Elastix today, on some cards, testing, and I wonder, if I setup Elastix,for instance - at which point wil I setup the loadzone? |
| 06:36.55 | jgoo | I'll see how it goes. |
| 06:37.16 | [TK]D-Fender | jgoo: Yup, life sucks that way sometimes. A way to get better responses from people here is to agree to their lamentations as much as you actually do and reinforce that the client demands it regardless of this and in resistance to other options you know are viable, and even possibly more functional/efficient |
| 06:37.48 | [TK]D-Fender | jgoo: loadzone is a basic system.conf thing. thats manual |
| 06:38.24 | [TK]D-Fender | jgoo: And for customers wanting things to work their specific way, FreePBX crams you down Dead-End-Alley |
| 06:38.29 | jgoo | [TK]D-Fender, that said, I've seen people take a customer demand, develop it without any real thought to redesign, have implemented layer upon layer of crap to maintain this immensely bad design and the work arounds required to make it usable in terms of processing, whereas it could all be simpler with a few lazy calls from the client... insane stuff |
| 06:39.05 | [TK]D-Fender | jgoo: Yup, but spare yourself the fight in here by clarifying your own disgust with your request :) |
| 06:39.07 | jgoo | Which is why you end up with machines with over 100GB of ram (which is fine, if you want that, not fine if you don't actually need it) |
| 06:39.32 | *** join/#asterisk |Cybex| (n=John@atwork-26.r-212.178.82.atwork.nl) |
| 06:39.36 | [TK]D-Fender | ok, far to late... off to bed. |
| 06:39.41 | [TK]D-Fender | later all. |
| 06:40.00 | jgoo | [TK]D-Fender, is loadzone just in zaptel.conf? I've googled that as part of my groking session, and I didn't find reference elsewhere |
| 06:40.05 | jgoo | [TK]D-Fender, nite nite, thanks |
| 06:40.16 | [TK]D-Fender | jgoo: yes, zaptel.conf / system.conf (DAHDI) |
| 06:40.20 | [TK]D-Fender | jgoo: ONLY there. |
| 06:40.24 | [TK]D-Fender | 'nite |
| 06:43.01 | Gokee2 | Anyone happen to know how the Polycom Soundpoint 320 is different from the 321? |
| 06:54.45 | Gokee2 | Well gonna put off the order till tomorrow I guess I might just go with the telephonydepot guys for the card as well |
| 06:55.58 | *** join/#asterisk Tim_Toady (n=moi@adsl285-115.kln.forthnet.gr) |
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| 08:04.51 | fiddur | Hmm, my asterisk 1.6.1.6 with new opsound moh, tries loading "CHANGES-asterisk-moh-opsound-2" and of course complains about no available formats... I have autoload on modules, and the sample conf for musiconhold.conf |
| 08:17.09 | *** join/#asterisk oej (n=olle@ns.webway.se) |
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| 08:25.12 | *** mode/#asterisk [+o mog] by ChanServ |
| 08:25.23 | *** join/#asterisk mattboll (n=mattboll@78.238.188.24) |
| 08:25.25 | mattboll | hi |
| 08:25.40 | mattboll | si there a way to sip debug only one softphone ? |
| 08:25.46 | mattboll | is* |
| 08:26.17 | mattboll | not a softphone in fact, a real phone |
| 08:26.46 | *** part/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
| 08:27.47 | kaldemar | mattboll: sip set debug peer <peer> |
| 08:28.13 | mattboll | thx kaldemar i didn't find it with the help command |
| 08:28.48 | kaldemar | help sip set debug will show the options. tab completion is also very helpful with cli commands. |
| 08:41.29 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 08:43.15 | *** join/#asterisk af_ (n=getsmart@88-149-240-140.dynamic.ngi.it) |
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| 08:58.13 | *** join/#asterisk iksik (i=xk@livedata.pl) |
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| 09:00.15 | *** join/#asterisk iksik (i=xk@livedata.pl) |
| 09:01.50 | mattboll | I've got a problem with voicemails, I can' leave a message, it hangs up |
| 09:01.59 | mattboll | there is my cli here : http://pastebin.com/m4f60b21d |
| 09:02.00 | *** join/#asterisk iksik (i=xk@livedata.pl) |
| 09:02.14 | mattboll | is it a known bug from asterisk ? |
| 09:02.44 | mattboll | it append with every phone |
| 09:04.00 | *** join/#asterisk iksik (i=xk@livedata.pl) |
| 09:05.05 | mattboll | appendS* |
| 09:05.45 | *** join/#asterisk iksik (i=xk@livedata.pl) |
| 09:06.21 | mattboll | if i search for voicemail and asterisk, i find a lot of things but for voicemail in asterisk, not when I call :p |
| 09:06.57 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 09:07.46 | *** join/#asterisk iksik (i=xk@livedata.pl) |
| 09:09.45 | *** join/#asterisk iksik (i=xk@livedata.pl) |
| 09:13.27 | *** join/#asterisk Faustov (i=user@gentoo/user/faustov) |
| 09:14.12 | Faustov | hello, what SIP provider for the UK (Wales) would you recommend? I'm mostly after stability & lack of extra restrictions |
| 09:14.21 | Faustov | and hopefully SIP rather than IAX2 |
| 09:16.07 | *** join/#asterisk dandate2 (n=gtejkgjk@112.202.109.63) |
| 09:16.13 | dandate2 | howto: make queue failover to a seperate pbx |
| 09:16.16 | *** join/#asterisk grEvenX (n=even@apb99b.ip.ssc.net) |
| 09:16.52 | dandate2 | in freepbx ? =) |
| 09:17.04 | kaldemar | mattboll: you need to add voicemail to your dialplan |
| 09:18.22 | *** join/#asterisk garymc (n=garymc@81.134.0.102) |
| 09:19.33 | dandate2 | is the only way to do this have fail over be an extention that forwards to a did on the seperate pbx and thus pay termination? |
| 09:20.20 | mattboll | i've got a problem when I call from the asterisk, not to asterisk |
| 09:25.14 | *** join/#asterisk [netman] (n=netman@175.Red-81-36-135.dynamicIP.rima-tde.net) |
| 09:27.43 | gr0mit | Faustov, i can help |
| 09:28.52 | gr0mit | origination or termination? |
| 09:30.58 | Faustov | brb |
| 09:32.05 | *** join/#asterisk SparFux (n=raoul@e182018161.adsl.alicedsl.de) |
| 09:35.17 | SparFux | Hi all. I have a problem using the chan_alsa.so. The module is loaded and logs give no errors, but the dial command is not available. |
| 09:35.38 | Tim_Toady | SparFux its "console dial" |
| 09:38.53 | SparFux | oh, thx. |
| 09:39.33 | SparFux | ah, all the new commands in asterisk 1.6 :-) |
| 09:39.41 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 09:39.50 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 09:41.10 | Tim_Toady | i think its console dial in 1.4 too |
| 09:46.17 | SparFux | Tim: Yes, but dial worked still. |
| 09:53.44 | *** join/#asterisk basty (n=basty@212.218.65.240) |
| 09:53.50 | basty | Good Day |
| 09:55.24 | basty | I have a little problem. I have setup a queue with 3 agents in it. Is it possible to route the call to an external extensions, as soon as the three agents are busy (already talking) ? |
| 09:56.26 | kaldemar | mattboll: you can't control what happens somewhere else. looks like you may have NAT setting problems though. |
| 10:02.49 | *** join/#asterisk Kevin` (n=kevin@rrcs-67-52-47-69.west.biz.rr.com) |
| 10:04.20 | *** join/#asterisk mahiti-irc (n=mahiti@122.166.50.88) |
| 10:04.57 | mahiti-irc | hi guys, asterisk now really cool at my place |
| 10:05.03 | mahiti-irc | i just want ont hing |
| 10:05.09 | mahiti-irc | one thing |
| 10:05.40 | mahiti-irc | i need a AGI script that will accept input and return back the input in speech |
| 10:05.48 | mahiti-irc | i have festival installed |
| 10:08.19 | mahiti-irc | is anyone awake here? |
| 10:08.49 | mattboll | klademar: the problem is not somewhere else, I've tried to call several phones, when I call from a "not asterisk" phone it's normal, I go to the voicemail |
| 10:09.50 | *** part/#asterisk SparFux (n=raoul@e182018161.adsl.alicedsl.de) |
| 10:11.33 | kaldemar | mattboll: ah, you have a 30 second timeout in the Dial command: -- Executing [0952044141@nereide:3] Dial("SIP/BNobody_116-092d03d0", "SIP/0952044141@ovh-3612|30|rtT") |
| 10:12.05 | mattboll | oh ! gg ! |
| 10:12.12 | kaldemar | mattboll: |30| <-- make this long enough for the remote party to drop you in the voicemail. |
| 10:12.31 | gr0mit | Faustov, ping |
| 10:15.26 | mahiti-irc | i need a AGI script that will accept input and return back the input in speech |
| 10:15.34 | mahiti-irc | can someon help me? |
| 10:19.00 | lipek | mahiti-irc: it is quite easy even without AGI |
| 10:19.20 | lipek | mahiti-irc: search for SayDigits |
| 10:20.16 | mahiti-irc | lipek, thanks for that but how would you pass the input to saydigits? |
| 10:22.27 | Faustov | gr0mit: hi, I think I already got a couple of numbers from you via IAX2 |
| 10:22.31 | Faustov | do you also have SIP? |
| 10:22.47 | gr0mit | aah yes! |
| 10:22.53 | gr0mit | yup, can do sip |
| 10:23.03 | Faustov | ok I'll deal with the formalities right now |
| 10:23.12 | Faustov | I'll talk to you in a moment |
| 10:23.15 | Faustov | hell of a day here |
| 10:23.51 | gr0mit | np.. |
| 10:26.52 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
| 10:27.13 | lipek | mahiti-irc: http://www.voip-info.org/wiki/view/Asterisk+cmd+Read |
| 10:27.27 | lipek | The Following Example uses Read and Say digits to read 3 digits from user, saying each one as user types. |
| 10:34.16 | *** join/#asterisk VEc (n=Vector@84.12.253.146) |
| 10:36.59 | mahiti-irc | lipek, thanks man |
| 10:37.06 | mahiti-irc | but didnt workout |
| 10:37.18 | mahiti-irc | i dont get a dtmf input |
| 10:37.50 | mahiti-irc | in debug it says user entered nothing even though i pressed the keys |
| 10:38.18 | mahiti-irc | is there something wrong with config? |
| 10:44.09 | lipek | what phone are you using ? |
| 10:44.24 | lipek | the phone generates dtmf |
| 10:47.05 | mahiti-irc | i am using a PRI line |
| 10:47.40 | mahiti-irc | asterisk connects to it via TE122 PRI card |
| 10:47.42 | mahiti-irc | and there is no device connected to the asterisk |
| 10:47.56 | mahiti-irc | i mean phone device |
| 10:52.38 | mahiti-irc | \quit i will e back |
| 11:03.25 | *** join/#asterisk Kchehab (n=kchehab@212.98.141.199) |
| 11:03.32 | Kchehab | evening all |
| 11:06.34 | Kchehab | how can i modify asterisk, if it detects two early media to stop OR MUTE the first RTP early media AND let the user hear the second early media |
| 11:09.26 | *** join/#asterisk mahiti-irc (n=mahiti@122.166.50.88) |
| 11:09.46 | mahiti-irc | lipek, any suggestions for me? |
| 11:12.58 | basty | if an agent in a queue is busy..i want to redirect the call to another extension..is it possible ? For right now the call hangs in the queue untill the agent is free again..and that sucks ;) |
| 11:16.05 | *** join/#asterisk ideogon (n=ideogon@pool-71-178-191-38.washdc.east.verizon.net) |
| 11:17.13 | *** part/#asterisk ideogon (n=ideogon@pool-71-178-191-38.washdc.east.verizon.net) |
| 11:19.20 | lipek | mahiti-irc: try connect any phone directly to the asterisk |
| 11:19.32 | lipek | you can download softphone (x-lite) and connect it |
| 11:19.36 | lipek | and check if it is working |
| 11:20.08 | mahiti-irc | lipek, the thing is the port 5060 is blocked by my ISP |
| 11:20.19 | mahiti-irc | i cant use x-lite |
| 11:20.26 | Kchehab | any update for my case ,any idea ! |
| 11:21.05 | kaldemar | mahiti-irc: use another port |
| 11:22.58 | kaldemar | mahiti-irc: but basically, if you have debug enabled in logger conf, the cli should show you incoming DTMF. if it doesn't, chances are that your provider isn't sending it to you. |
| 11:24.36 | kaldemar | core set debug channel <channel> might also be useful |
| 11:30.14 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 11:31.50 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:31.50 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:34.39 | *** join/#asterisk iksik (i=xk@livedata.pl) |
| 11:36.35 | iksik | morning |
| 11:36.55 | iksik | Who wants to help me with realtime? :) I'm stuck with extensions... I'm watching postgresql queries log file, and there is nothing about select from extensions table :/ |
| 11:37.34 | Druken | i use realtime for everything EXCEPT the dialplan :) |
| 11:38.23 | Druken | my dialplan stays flatfile and will always be, unless they change that... i like being able to visually traverse it myself... |
| 11:38.54 | iksik | ;) |
| 11:39.54 | Druken | not only that, my dialplan don't change that much :) |
| 11:39.58 | iksik | hm, I was configure it exactly like they said here http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions |
| 11:41.30 | Druken | can ya pastebin the appropiate files? |
| 11:41.53 | iksik | http://pastebin.com/d404bd2ee |
| 11:41.54 | kaldemar | iksik: the book has an example with postgresql. have you looked at it? |
| 11:44.50 | iksik | kaldemar, yeap... but there is no examples with dialplans - unless I'm missing something |
| 11:47.36 | kaldemar | what does your extconfig.conf look like? |
| 11:47.58 | iksik | http://pastebin.com/d404bd2ee |
| 11:48.30 | kaldemar | extconfig.conf, not extensions.conf |
| 11:49.11 | iksik | oh |
| 11:49.13 | iksik | sorry |
| 11:50.04 | iksik | http://pastebin.com/m475ea1bc |
| 11:51.47 | kaldemar | if i'm not completely mistaken, you should have switch => Realtime/DPRP_Outgoing@extensions instead of switch => Realtime/DPRP_Outgoing@vextensions in extensions.conf |
| 11:52.02 | iksik | hm |
| 11:52.16 | iksik | but there is no extensions table |
| 11:52.16 | iksik | there is only vextensions |
| 11:52.24 | iksik | but.. I will try |
| 11:52.37 | kaldemar | that's not the table, it's the family |
| 11:52.41 | iksik | ok |
| 11:52.54 | iksik | testing ;) |
| 11:53.23 | kaldemar | the table and family are defined in extconfig.conf, and a context and a family is called from extensions.conf |
| 11:53.33 | iksik | still nothing |
| 11:53.46 | iksik | he doesn't even try to get some data from database :/ |
| 11:54.14 | iksik | accounts_sip works... and I see select for user... but for extensions, nope ;/ |
| 11:54.47 | kaldemar | do you see asterisk connecting to the database via TCP? |
| 11:55.05 | iksik | yeap... and I see ie. this: LOG: statement: SELECT * FROM accounts_sip WHERE name = '7012' AND host = 'dynamic' |
| 11:55.43 | iksik | or this: LOG: statement: UPDATE accounts_sip SET ipaddr = '198.162.0.1', port = '5060', regseconds = '1252670103', defaultuser = '7012', useragent = '', lastms = '0', fullcontact = 'sip:7012@198.162.0.1:5060;transport=udp' WHERE name = '7012' |
| 11:55.52 | iksik | this means, that connection works, right? |
| 11:56.47 | kaldemar | probably |
| 11:56.52 | *** join/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
| 11:57.29 | Druken | in extensions you don't supply a family, you supply a context |
| 11:58.41 | iksik | right.. Alt. Switch => 'Realtime/DP_Outgoing@extensions' [pbx_config] |
| 11:59.24 | kaldemar | Druken: either of those or both |
| 11:59.54 | Druken | the DP_outgoing is the extension |
| 12:00.05 | Druken | or, the context, my bad |
| 12:00.08 | iksik | yes |
| 12:00.51 | Druken | if you don't specify a family, it defaults to extensions |
| 12:01.09 | iksik | so... how should I set it up? :S |
| 12:01.49 | iksik | I'm confused now :( |
| 12:01.53 | Druken | if you want to use the current context and are using extensions family, just use switch => realtime/@ |
| 12:03.28 | Druken | are you sure you know what your using the realtime dialplan for? |
| 12:03.32 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:03.45 | iksik | hah, It works now ;) |
| 12:03.48 | kaldemar | try to change vextensions to extensions in extconfig.conf and extensions.conf |
| 12:03.55 | kaldemar | what did you do? |
| 12:04.14 | iksik | Like Druken said... I've change this DP_Outgoing/extensions into @ |
| 12:04.50 | Druken | hears the angels sing.... |
| 12:04.54 | iksik | hehe |
| 12:04.55 | Druken | rotflmao |
| 12:05.07 | iksik | but this extensions query is weird: SELECT * FROM vextensions WHERE exten LIKE '\\_%' AND context = 'DP_Outgoing' AND priority = '1' ORDER BY exten |
| 12:05.18 | iksik | that means, there is no way to use "s" in D |
| 12:05.19 | iksik | DP* |
| 12:05.36 | [TK]D-Fender | Druken: Walk towards the light!!! |
| 12:06.18 | [TK]D-Fender | iksik: just prefix with "_" |
| 12:06.38 | iksik | yeah, but what should I do withi this \\ ? |
| 12:06.39 | iksik | :| |
| 12:07.19 | [TK]D-Fender | iksik: Go run it yourself and see what you get |
| 12:07.29 | iksik | with \\ nothing |
| 12:07.31 | *** join/#asterisk garymc (n=garymc@host81-134-0-102.in-addr.btopenworld.com) |
| 12:07.35 | iksik | without \\ 1 result |
| 12:07.37 | *** join/#asterisk msetim (n=setim@200.195.162.210) |
| 12:07.41 | msetim | hello |
| 12:08.51 | msetim | I've been read about autofallthrough however I don't understand what kind of changes it can to bring to my extensions.conf when I change it from "no" to "yes". |
| 12:09.11 | msetim | by the way today it is setted to "no" |
| 12:09.31 | iksik | [TK]D-Fender, is that "\\" thing required? :| |
| 12:10.11 | [TK]D-Fender | msetim: no = when pretty much any exten runs out of priorities it will hang like an IVR waiting for input unless you have explicitly hung up |
| 12:10.29 | [TK]D-Fender | iksik: Show your records |
| 12:10.51 | iksik | there is only one record... with exten = "_7050" |
| 12:11.12 | [TK]D-Fender | iksik: I'm a little gray on why that's there... |
| 12:12.27 | iksik | ok, dunno why, but it works now ;) |
| 12:12.31 | iksik | thanks for help |
| 12:12.50 | leifmadsen | I don't want to meet your Mom! I just want... |
| 12:13.58 | *** join/#asterisk coppice (n=chatzill@68.166.17.210.dyn.pacific.net.hk) |
| 12:14.16 | iksik | oh, and that would be awesome: [*] switch => Realtime/@ - is it possible? :D |
| 12:14.20 | iksik | some kind of wildcard |
| 12:14.34 | leifmadsen | people still use switch statements? |
| 12:14.37 | leifmadsen | wow! |
| 12:16.32 | [TK]D-Fender | c'mon c'mon c'mon c'mon c'mon BABY! SWITH AND SHOUT! |
| 12:16.38 | [TK]D-Fender | switch* |
| 12:16.40 | [TK]D-Fender | gah |
| 12:16.51 | [TK]D-Fender | 1 beatle is rolling in his grave.... |
| 12:16.52 | lipek | you've ruined the moment |
| 12:17.34 | iksik | leifmadsen, what is wrong with switch? |
| 12:18.04 | leifmadsen | I don't like loading dialplans from databases -- there are better ways |
| 12:18.15 | msetim | [TK]D-Fender, thanks :) so it's better let the asterisk choise when is better hangup the call \o/ |
| 12:23.17 | [TK]D-Fender | msetim: IMO its best when you specify everything explicitly |
| 12:23.55 | *** join/#asterisk wonderworld (n=w@ip-62-143-22-226.unitymediagroup.de) |
| 12:27.11 | msetim | [TK]D-Fender, but i'm having troubles because I'm using manager originate and I get the OriginateResponse, when I use "no" the response always get reason 1 (NO ANSWER) ignoring status like BUSY and CONGESTION |
| 12:30.46 | iksik | why I see in logs No such extension/context 050@default while calling Local channel - I'm callling extension 7050 not 050 |
| 12:36.32 | *** part/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
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| 12:38.46 | msetim | [TK]D-Fender, Wow, I found this variable ${HANGUPCAUSE}, I will try to use it to set the correct hangupcause to OriginateResponse Hangup(${HANGUPCAUSE}) |
| 12:39.11 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 12:39.11 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 12:47.51 | [TK]D-Fender | msetim: that should be a readonly variable and hanging up with the same cause shouldn't change anything |
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| 13:23.22 | phix | how can I list all devices that java picked up? |
| 13:23.29 | phix | hi |
| 13:23.47 | *** part/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 13:24.03 | phix | sorry wrong chan |
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| 13:43.10 | grandpapadot | Greetings mighty baud warriors! Level I tech support is in da house, all you level ii guys can go get some coffee and fud ... |
| 13:43.35 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
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| 13:46.50 | viraptor | does anyone know a way to figure out which side of a sip->sip call did a blind transfer? I need to set some information in order to get a proper callerid / billing, but how... ? |
| 13:48.42 | [TK]D-Fender | viraptor: ${BLINDTRANSFER} <- look for it |
| 13:49.30 | *** join/#asterisk Skeeter- (n=wil_c_wi@c216.218.2-65.clta.globetrotter.net) |
| 13:49.54 | Skeeter- | 2 servers trunkin works just great |
| 13:50.11 | Skeeter- | 3 servers trunking gives me restriction |
| 13:50.25 | Skeeter- | wanna make all of em act like one server |
| 13:50.48 | [TK]D-Fender | Skeeter-: ? |
| 13:50.56 | Skeeter- | here is the situation |
| 13:51.39 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 13:51.39 | viraptor | [TK]D-Fender: yeah... that's helpful - thanks |
| 13:51.57 | Skeeter- | server 1 and server 2 are trunking just fine, while i trunk server 2 with server 3, server 1 can no longer call server, but server2 can still call server 1 |
| 13:52.29 | Skeeter- | if i trunk server 3 with server 1, server1 and 2 works just fine, but server 3 wont connect to server 2 |
| 13:52.32 | [TK]D-Fender | Skeeter-: makes no sense |
| 13:52.51 | Skeeter- | i know, sounds like the server1 is acting like a host or something |
| 13:53.28 | [TK]D-Fender | Skeeter-: Show the failure |
| 13:54.23 | *** join/#asterisk ethicx (n=chatzill@208.90.127.141) |
| 13:56.49 | ethicx | hello everyone. I'm having trouble with my asterisk box and my service provider Im getting this error 'got SIP response 440 "Device not Authenticated" back from IP.... what could be wrong? |
| 13:57.54 | [TK]D-Fender | ethicx: could be you need to be registered to place a call to that host |
| 13:58.45 | ethicx | [TK]D-Fender: yeah but it was working till yesterday night.. |
| 13:58.59 | seanbright | so you know the usual pattern of Answer(), Wait(2) |
| 13:59.02 | [TK]D-Fender | ethicx: But maybe you're not registered NOW, and you're definitely not showing much. |
| 13:59.08 | seanbright | (or Answer(2000)) |
| 13:59.12 | seanbright | is that necessary with VoIP? |
| 13:59.22 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:59.31 | [TK]D-Fender | seanbright: Still suggested due to RTP setup delay |
| 13:59.38 | seanbright | alright |
| 13:59.42 | seanbright | 2 seconds just seems long |
| 13:59.42 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 13:59.57 | Skeeter- | Fender: i simply need to trunk server 2 and 3 thats it, i think it will solve the problem |
| 14:00.01 | Skeeter- | 3 way trunking |
| 14:00.39 | Skeeter- | btw, server 3 is a voip provider, iax number |
| 14:01.09 | [TK]D-Fender | Skeeter-: Care to actually redescribe this setup since you made it seems as if YOU had 3 servers |
| 14:03.29 | Skeeter- | server 1 trunks with server 2, and vice versa(no problem here), server 1 trunks with server3(the IAX provider), still no problem here, now i need to trunk the server 3 with server 2 |
| 14:03.51 | *** join/#asterisk MRH2 (n=chatzill@62.49.242.3) |
| 14:03.53 | Skeeter- | server 2 is acting like a slave somehow, and server 1 like the main/host server |
| 14:04.37 | Skeeter- | i might be wrong about the last statement |
| 14:05.10 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 14:06.55 | *** join/#asterisk neurosys (n=vinix@173.9.159.182) |
| 14:07.13 | Skeeter- | maybe if i set some EXTEN config, so when i dial an server 2 ext from server 3, that the call pass from server 3 to server1 to server 2 |
| 14:07.56 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:07.56 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:10.10 | MRH2 | hi asterisk 1.2.x externip in sip.conf no longer seems to be working for me 100% of the time. (has been working fine for ages). I know there is a multitude of possible issues but anything common I should check. |
| 14:10.11 | *** join/#asterisk AeroCloud (n=aero@ip98-165-113-125.ph.ph.cox.net) |
| 14:11.37 | *** join/#asterisk moy (n=moy@74.12.131.104) |
| 14:11.38 | [TK]D-Fender | Skeeter-: Stop referring to your PROVIDER as SERVER 3 |
| 14:11.59 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 14:12.24 | [TK]D-Fender | Skeeter-: MRH2 No reason it shouldn't. Show us |
| 14:12.29 | [TK]D-Fender | MRH2 : No reason it shouldn't. Show us |
| 14:12.42 | [TK]D-Fender | Skeeter-: and show us the failure |
| 14:13.14 | [TK]D-Fender | Skeeter-: And there is the option "transfer=no" which will prevent IAX2 reinvite |
| 14:13.49 | MRH2 | so i check the SIP messages have the correct value and if not blame the remote machine? |
| 14:14.04 | MRH2 | i mean if so |
| 14:14.42 | tgunr | I think I have finally nailed down where the 'Unable to lookup ''` error is coming from but don't know how to fix it. When I try to dial outbound after having defined the outgoing rule and dialplan and users in the GUI, the call fails because trunk_2 is not found. Looking thru the startup logs I could not find trunk_2 being defined |
| 14:14.45 | dandre | hello, |
| 14:14.54 | tgunr | Instead I found this http://paste.lisp.org/display/86927 (the calling error followed by the restart log). My users.conf looks like this http://paste.lisp.org/display/86928 What is wrong in here? I just add the type = user and same result with or without that line. |
| 14:15.05 | dandre | where can I find this patch for 1.4 branch: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg201588.html |
| 14:15.32 | [TK]D-Fender | MRH2: Of course you look at the SIP message |
| 14:16.03 | *** join/#asterisk neurosys (n=vinix@173-9-159-182-miami.txt.hfc.comcastbusiness.net) |
| 14:17.06 | ethicx | what's the right way to stop asterisk? from the CLI? |
| 14:17.07 | *** join/#asterisk manxpower (n=EWieling@50.sub-70-222-21.myvzw.com) |
| 14:17.17 | tgunr | stop gracefully |
| 14:17.22 | tgunr | or stop now |
| 14:18.42 | ronr | hi, callerid on outgoing isn't working, I figured out using NoOp that asterisk has the right callerid, I just found out that running ztcfg causes an error: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 in the asterisk console, I guess this could be the problem, any clues on how to fix this? |
| 14:19.05 | ethicx | and starting it just by invoking the daemon process right? |
| 14:19.41 | [TK]D-Fender | dandre: https://issues.asterisk.org/view.php?id=13135 |
| 14:19.52 | manxpower | ronr: HDLC abort means "error from PRI". It would drop calls, not mess up callerid |
| 14:20.31 | MRH2 | ok i'll have to keep it on IAX2 and check it later. Thnks |
| 14:20.38 | ronr | manxpower: hmm, ok, I only get it once on ztcfg and calls aren't dropped; any clues on what the callerid issue could be? |
| 14:20.43 | ronr | or things I can check |
| 14:21.29 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 14:22.00 | vuc_spammer | good afternoon, Asterisk people |
| 14:22.06 | manxpower | ronr: are your sure your carrier alllows you to set outgoing callerid number? |
| 14:22.11 | dandre | ok |
| 14:22.12 | [TK]D-Fender | dandre: And you're not going to find it. |
| 14:22.37 | vuc_spammer | good morning TK] |
| 14:22.39 | ronr | manxpower: they used to, bug I guess they could have changed that.. it's worth checking out |
| 14:22.43 | dandre | but is this for 1.4? |
| 14:22.57 | dandre | I have seen 1.6 and trunk in the issue |
| 14:23.02 | manxpower | ronr: pastebin the CLI output showing the failed call including the Noop showing the callerid info |
| 14:23.06 | *** join/#asterisk KavanS (n=KavanS@71.117.242.28) |
| 14:23.39 | asterwiki | ronr: Show the CLI output when you make a call so we can see what happens... |
| 14:23.57 | *** join/#asterisk boomerang (n=boomeran@unaffiliated/boomerang) |
| 14:24.05 | ronr | with pri debug data? |
| 14:24.37 | manxpower | ronr: gads no, |
| 14:24.55 | [TK]D-Fender | dandre: This is a NEW feature. 1.4 doesn't GET new features |
| 14:24.58 | dandre | I am looking for SIPPEER(peer,chanvar[]) function for 1.4 |
| 14:25.29 | [TK]D-Fender | dandre: Apparently you have failed to grasp how code gets added to * |
| 14:26.10 | manxpower | I always wondered why otherwise (mostly) intelligent people think that new features get added to 1.4, 1.2, etc |
| 14:26.11 | dandre | ok but in the mail there is: |
| 14:26.14 | dandre | The branch is based on 1.4 and the patch will soon be included in the 1.6 trunk. |
| 14:26.22 | ronr | http://pastebin.com/m169ef78a |
| 14:26.22 | ronr | the NoOp is on ${CALLERID(num)} |
| 14:26.32 | manxpower | dandre: that does not say it is part of 1.4 |
| 14:27.03 | dandre | yes but there may be a patch against 1.4 |
| 14:27.28 | manxpower | dandre: good luck finding it |
| 14:27.31 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 14:27.49 | jaytee | SIPPEER is in 1.4 but chanvar[] is not a valid item |
| 14:28.30 | [TK]D-Fender | dandre: And they may yet find Jimmy Hoffa... |
| 14:28.37 | manxpower | ronr: 0104365050 is the number. what part of that is the phone number and what part is country/city code? |
| 14:29.16 | [TK]D-Fender | dandre: People who are convinced they will find things that there is no valid reason to suspect they will are typically crazy |
| 14:29.23 | ronr | 010 is city, country is not included (in this test), the incoming callerid on the receiving end is 0107070081 (which is another one of our numbers btw) |
| 14:29.47 | manxpower | ronr: I thought most countries used the leading "0" as a toll access code? |
| 14:29.54 | dandre | What I am looking for is: |
| 14:29.54 | dandre | I have a sip peer declared with setver in it. I want those var to be set if I issue an originate command from the manager |
| 14:30.10 | ronr | manxpower: around here every city code starts with a 0 |
| 14:30.29 | [TK]D-Fender | dandre: Not going to happen. |
| 14:30.31 | manxpower | ronr: try removing the leading 0 |
| 14:30.45 | dandre | what do you mean? |
| 14:30.57 | [TK]D-Fender | dandre: Originate calls the first channel. It has no point of origin and no vars to declare |
| 14:31.10 | *** join/#asterisk jtodd (i=ddyt1otf@ns.fox-den.com) |
| 14:31.10 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 14:31.13 | [TK]D-Fender | dandre: If you want vars set you're going to have to do them yourself. |
| 14:31.36 | ronr | manxpower: same result |
| 14:31.39 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 14:31.41 | ethicx | [TK]D-Fender: This is the error that im getting http://pastebin.com/d769fc854 |
| 14:31.52 | manxpower | ronr: contact your carrier |
| 14:32.04 | ronr | ok, will do, thx |
| 14:32.31 | [TK]D-Fender | ethicx: Worthless output. Show the ENTIRE conversation, with SIP DEBUG, the peer dump, and the reg dump |
| 14:32.45 | dandre | the originate command is sent by a host that don't know how variables are declared in the sip peer |
| 14:33.14 | [TK]D-Fender | dandre: A. Too bad. or B. Upgrade. Take your pick |
| 14:33.34 | asterwiki | ronr:if that doesn't help we may need to see the ISDN q.931 traces |
| 14:34.20 | ronr | asterwiki: ok, first I'll contact the telco, than I'll try to find the docs on how to get those :D (but weekend'll be likely to be here before that) |
| 14:34.24 | manxpower | dandre: the whole point of Originate is so you can bypass the sip/iax/whatever peer |
| 14:34.48 | neurosys | [TK]D-Fender: Do you happen to know of a way i can customize teh voicemail system? Manxpower mentioned the new "minivoicemail" class in the latest version, but this is a 1.4 box. any suggestions? |
| 14:34.56 | manxpower | ronr: it looks like your dialplan stuff is correct. I suspect some other issue |
| 14:35.08 | dandre | manxpower: I don't understand |
| 14:35.17 | manxpower | neurosys: for 1.4 your option is to rewrite app_voicemail.c |
| 14:35.25 | [TK]D-Fender | neurosys: Upgrade |
| 14:35.36 | neurosys | eww :P ok |
| 14:35.53 | manxpower | dandre: Originate: is there so you can make a call without having to pick up a phone and dial something from a device. it bypasses all the device stuff like peer variables |
| 14:37.02 | dandre | ok but the phone rings before the call is established |
| 14:37.16 | manxpower | dandre: no, that is the DESTINATION device. not the SOURCE device. |
| 14:37.25 | [TK]D-Fender | dandre: the call is TO the device, not FROM it. the vars do not get set. TOO BAD |
| 14:37.38 | manxpower | I have actual work to do. ttfn |
| 14:37.40 | *** part/#asterisk manxpower (n=EWieling@50.sub-70-222-21.myvzw.com) |
| 14:37.50 | dandre | and when picked up, if the call ends to voice mail it is not in the correct language for instance |
| 14:38.30 | [TK]D-Fender | dandre: How many times do we have to say it? |
| 14:38.33 | [TK]D-Fender | dandre: the call is TO the device, not FROM it. the vars do not get set. TOO BAD <---------- |
| 14:38.42 | [TK]D-Fender | dandre: Peer settings do not get used |
| 14:38.52 | dandre | ok |
| 14:39.18 | [TK]D-Fender | dandre: the call takes charateristics of the device that ORIGINATED the call. That would the the CONSOLE which has precisely JACK SHIT set. |
| 14:39.24 | [TK]D-Fender | dandre: Clear now? :) |
| 14:40.19 | dandre | ok |
| 14:40.45 | [TK]D-Fender | be* |
| 14:43.41 | Zeeek | damn |
| 14:43.51 | Zeeek | the nick JackShit is taken |
| 14:47.54 | *** join/#asterisk L2Networks (n=nospam@38.101.207.250) |
| 14:48.02 | *** join/#asterisk Weasel[DK] (n=eht@93.164.121.150) |
| 14:48.14 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 14:48.44 | *** join/#asterisk retentiveboy (n=pdugas@74.223.226.30.nw.nuvox.net) |
| 14:49.59 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 14:50.35 | Weasel[DK] | im missing Congestion tone on my SIP phones. On the console i see the message "Busy here" but in the phone is total silence ? |
| 14:50.42 | Weasel[DK] | what have i forgotten ? |
| 14:51.11 | [TK]D-Fender | Weasel[DK]: to ANSWER first |
| 14:51.30 | martst | I hope someone can give me some help with wctdm24xxp from dahdi-linux-2.2.0.2 |
| 14:51.54 | Zeeek | jtodd: you should be sure to be around today, you might win the g722 phone, therefore getting us all out of an embarassing situation |
| 14:52.12 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:52.26 | L2Networks | Good morning, i'm not certain of IRC channel etiquette.. where would one present a question regarding early media as it effects mobile ringbacktones, toll free IVRs, etc.. ? |
| 14:52.27 | martst | All compiles fine but when I modprobe it, it says it is looking for zaptel.conf and it |
| 14:52.48 | *** join/#asterisk pugachevcobra (n=fighter@189.107.81.179) |
| 14:52.55 | martst | can't open /dev/zap/ctl or /etc/zaptel.conf |
| 14:53.12 | martst | I have checked that there are no other versions of the module about so I am confused |
| 14:53.31 | [TK]D-Fender | martst: Shouldn't be using zaptel.conf. You must have a compatibility mode flag set which isn't really recommended |
| 14:55.21 | pugachevcobra | HI. I'm having a frequent problem with my incoming calls from a sip trunk registered at a voip provider. Many times when a call arrives, asterisk doesn't answers. It seems as it answers and hangups instantly, as the caller hears nothing and the call drops. Absolutely nothing appears on full logs, console and sip debug. It only happens with asterisk, if a softphone registers with the provider, no calls are dropped. |
| 14:55.30 | martst | Is compatibility mode set for the compile? |
| 14:55.34 | *** join/#asterisk flujan (n=flujan@189.111.254.251) |
| 14:57.28 | flujan | hello guys, I am looking for a hardphone that supports xml configuration. This is my scenario: A user log on the hardphone and work from 08AM to 02PM. At 02PM another user log and work until 08PM. I need to always log on the web interface and configure the hardphone. I would like to just send a xml and reboot the phone. |
| 14:57.34 | flujan | someone with this scenario? |
| 14:57.34 | [TK]D-Fender | martst: Check your system.conf |
| 14:58.02 | Chainsaw | flujan: Cisco 7960 has some degree of XML configuration. It's rudimentary and poorly documented though. |
| 14:58.13 | Chainsaw | flujan: Polycom is likely your best bet. |
| 14:58.36 | grandpapadot | "poorly documented" is an understatement |
| 14:58.44 | [TK]D-Fender | flujan: SIP phones don't tend to allow config changes on the fly. How they get ID'd is up to the server normally. What you seem to be looking for is the kind of thing I'd only expect to see in a proprietary solution |
| 14:59.03 | MRH2 | I use an abstraction layer for users (like channel agent in the older versions) |
| 14:59.14 | *** join/#asterisk stimpie (n=stimpie@84-104-5-142.cable.quicknet.nl) |
| 14:59.29 | [TK]D-Fender | flujan: With many phones you can update provisioning and reboot and they'll take changes, but thats kinda fugly. |
| 14:59.40 | [TK]D-Fender | flujan: And what I presume you don't really want to do. |
| 15:00.07 | flujan | [TK]D-Fender: yeah, but on old telco system you have this features. *1 "type your extension" "type your password" you are logged. *0 your are logged off from the system... |
| 15:00.41 | flujan | on a IP based solution, the name/password stays on the phone... It is hard to put that into my boss head. :( |
| 15:01.04 | flujan | he used to work this way, and still wanna work this. |
| 15:01.09 | martst | D-Fender: my system.conf was generated by dahdi_genconf and there is no mention of ZAP in there |
| 15:01.10 | ethicx | [TK]D-Fender: sorry but I'm new to asterisk and I'm still learning how to get all the different system dumps.. here is what I got so far setting the debugger on: http://pastebin.ca/1562189 |
| 15:01.11 | [TK]D-Fender | flujan: Sorry, try #neurosurgery |
| 15:01.19 | flujan | [TK]D-Fender: lol |
| 15:04.20 | *** join/#asterisk meesterarend (n=frans@vc-41-31-107-103.umts.vodacom.co.za) |
| 15:04.21 | *** join/#asterisk jtodd (i=ywqx9g7l@ns.fox-den.com) |
| 15:04.21 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 15:05.56 | Skeeter- | trying to connect an analog phone via a mediatrix to my asterisk server, http://pastebin.com/m4a09e536, when i call from 300(polcyom) to 400(mediatrix extension) |
| 15:07.21 | *** join/#asterisk asif (n=chatzill@122.166.40.72) |
| 15:07.32 | jjshoe_ | mm freepbx |
| 15:07.48 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 15:08.23 | jjshoe_ | Skeeter- look at line 64 |
| 15:08.50 | jjshoe_ | looks like you'll need to sip debug |
| 15:09.21 | Kobaz | hmmm |
| 15:10.33 | [TK]D-Fender | Skeeter-: where is the peer dump? |
| 15:10.56 | *** join/#asterisk wcselby (n=wcselby@216.110.88.254) |
| 15:11.33 | Skeeter- | i thought the error was from line 50 |
| 15:11.45 | Skeeter- | 400 cf is disabled |
| 15:11.55 | *** join/#asterisk dlynes (n=daniel@bas5-hamilton14-1242444747.dsl.bell.ca) |
| 15:11.57 | *** join/#asterisk wcselby (n=wcselby@216.110.88.254) |
| 15:11.58 | [TK]D-Fender | Skeeter-: Meaningless crap |
| 15:12.03 | [TK]D-Fender | Skeeter-: PEER DUMP |
| 15:12.03 | Skeeter- | kk |
| 15:12.24 | Skeeter- | maybe the mediatrix is not registered |
| 15:12.25 | wcselby | hahaha, what a way to enter the channel - <[TK]D-Fender> Skeeter-: Meaningless crap |
| 15:12.40 | dlynes | How do you get Asterisk to use /var/run/asterisk/ to dump the .pid and .ctl files? |
| 15:12.51 | Skeeter- | ~Skeeter- |
| 15:13.24 | Skeeter- | mediatrix is not registered, *97(voicemail) fails |
| 15:13.47 | dlynes | Skeeter-: You need to put 'asterisk' in for your realm....mediatrix is pretty picky about that |
| 15:14.05 | dlynes | Skeeter-: which particular mediatrix boxes are you trying to use? |
| 15:14.11 | jaytee | "Welcome to Meaningless Crap Incorporated! We've been making add-in products for Microsoft Office applications for over 10 years! For a free demo click here and enjoy our BSOD presentation." |
| 15:14.15 | *** join/#asterisk L2Logic (n=nospam@38.101.207.250) |
| 15:14.15 | Zeeek | Insert shameless plug for the VoIP Users Conference here - http://VUC.me - a bunch of people just like you talking trash about VoIP - go to #voip-users-conference for a drink and you'll stay for the floor show |
| 15:14.22 | Skeeter- | 2101 |
| 15:14.22 | dlynes | Skeeter-: the 1124's are quite problematic |
| 15:14.27 | Skeeter- | 2102 |
| 15:14.40 | dlynes | Skeeter-: yeah...you shouldn't have a problem with the 2102, but you'll have tons of problems with the 2101 |
| 15:14.46 | dlynes | Skeeter-: because the 2102 doesn't exist |
| 15:14.54 | dlynes | erm 2101 doesn't exist |
| 15:15.06 | Skeeter- | i was like |
| 15:15.24 | Skeeter- | 2102, doesnt exist, wt... i got here if the sticker says 2102 |
| 15:15.34 | dlynes | Skeeter-: and fwiw, i've never gotten the t.38 faxing to work on them |
| 15:15.46 | dlynes | Skeeter-: I ended up saying piss on it, and I use Linksys pap2's for that |
| 15:15.51 | Skeeter- | realm: serverip, username: extension password : extension sercret |
| 15:15.55 | dlynes | Skeeter-: they work flawlessly |
| 15:16.05 | dlynes | Skeeter-: no |
| 15:16.10 | dlynes | Skeeter-: realm: asterisk |
| 15:16.24 | jaytee | 2102? if we're talking Linksys SPA I've got 14 of them around the zoo of that exact model |
| 15:16.26 | dlynes | Skeeter-: It will never work with any value except asterisk |
| 15:16.56 | dlynes | Skeeter-: if you want, i can email you the pdf document for setting up mediatrix boxes with asterisk |
| 15:17.10 | Skeeter- | wow |
| 15:17.12 | Skeeter- | that |
| 15:17.14 | dlynes | Skeeter-: i deal with mediatrix boxes all day |
| 15:17.21 | Skeeter- | dlynes: that would be very nice |
| 15:17.37 | Skeeter- | dlynes: can i pm you |
| 15:17.40 | dlynes | Ok...it's not going to happen this morning, but when I get back from running around town, there's a strong chance |
| 15:17.44 | dlynes | Yes, go ahead |
| 15:18.48 | L2Logic | does anyone have any experience enabling early media with asterisk and a media gateway ? |
| 15:18.57 | *** join/#asterisk propellerhead (n=yogurt2u@host26.190-137-6.telecom.net.ar) |
| 15:22.13 | wcselby | ~phones |
| 15:22.14 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 15:22.27 | [TK]D-Fender | Skeeter->mediatrix is not registered, *97(voicemail) fails <- Registration has nothing to do with calls failing to auth |
| 15:22.43 | [TK]D-Fender | Skeeter-: the fact its not registered is clearly why you can't call it |
| 15:23.17 | [TK]D-Fender | L2Logic: * supports this on digital interfaces IIRC, but I don't think over SIP..... |
| 15:23.23 | dlynes | [TK]D-Fender: he should be able to get it working with the document I just sent him (registration issues, that is) |
| 15:23.39 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 15:23.39 | dlynes | [TK]D-Fender: mediatrixes are quite particular with how they need to be configured to talk to asterisk |
| 15:24.31 | [TK]D-Fender | dlynes: I've used an 1124 once. pretty dead-easy. Lot easier thant he AudioCodes Mediant I demo'd right next to it |
| 15:24.38 | *** join/#asterisk BlargMaN00 (n=blarg@12.234.16.130) |
| 15:24.46 | dlynes | [TK]D-Fender: yeah...the audiocodes are a nightmare |
| 15:25.03 | dlynes | [TK]D-Fender: i've got two of those as well...and their tech support is the worst I've ever come across |
| 15:25.46 | *** join/#asterisk Maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 15:25.48 | dlynes | [TK]D-Fender: the audiocodes is more configurable, but mediatrix is more configurable, where it counts |
| 15:25.50 | *** join/#asterisk BlargMaN00 (n=blarg@12.234.16.130) |
| 15:25.53 | *** join/#asterisk csmyth (n=csmyth@ext-52.sagetelecom.net) |
| 15:26.05 | [TK]D-Fender | dlynes: I wouldn't say nightmare, but inexplicably more complex for little visible reason. I did like how you could export all the settings which makes mass provisioning once you have 1 setup up right so unbelieveably easy |
| 15:26.49 | *** join/#asterisk docelmo (n=chatzill@65.114.160.138) |
| 15:26.59 | L2Logic | <PROTECTED> |
| 15:27.21 | docelmo | What did I miss? |
| 15:27.26 | dlynes | [TK]D-Fender: the mediatrix allows you to do that, also |
| 15:27.28 | docelmo | DS3's are fun to play with |
| 15:27.37 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 15:27.38 | dlynes | [TK]D-Fender: you just need their special software installed |
| 15:27.42 | [TK]D-Fender | dlynes: the smaller Mediatrix are really powerful boxes and do all sorts of realyl sick things like transparent proxying, etc. You can have a PPPoE modem on one side and shove this in front of your router and it'll never know the difference. Even have it spit out DHCP relaying the WAN IP. |
| 15:28.14 | [TK]D-Fender | dlynes: there is so much in the SW to do that it is kinda confusing at the start |
| 15:28.16 | coppice | the mediatrix software has some of the industry's most creative bugs |
| 15:28.22 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
| 15:28.37 | dlynes | [TK]D-Fender: ah...never knew that...but then again...I haven't used the 2102's that much |
| 15:28.41 | [TK]D-Fender | coppice: Good, I wouldn't want to be behind the curve ;) |
| 15:28.47 | dlynes | [TK]D-Fender: i've mainly been using the 4108's and 4124's |
| 15:29.42 | [TK]D-Fender | dlynes: Been a long while since I've used any of their products, and only brief demos at that... |
| 15:30.30 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 15:30.40 | dlynes | [TK]D-Fender: we're quite happy with the 4124's |
| 15:31.22 | dlynes | [TK]D-Fender: just not so happy that a cheap $50 PAP2 works flawlessly with t.38 and then mediatrix tells us that their expensive boxes are more configurable and that that's why they don't work so well with t.38 |
| 15:31.48 | *** join/#asterisk garymc (n=garymc@host81-134-0-102.in-addr.btopenworld.com) |
| 15:32.17 | L2Logic | with that siad, the other challenge we're experiening is finding a sip redirect server, or some type of SER software that would be friendly to non-tech people.. so that it doesn't take an engineer to move a DID from one * box to another, etc.. anyone know if such a beast exists... ? |
| 15:32.18 | dlynes | [TK]D-Fender: and the tftp config is quite flaky on the mediatrix boxes |
| 15:32.36 | [TK]D-Fender | dlynes: Care to share your T.38 experiences on the PAP2 with coppice here... no-one has seemed to find any evidence they actually support it |
| 15:32.56 | dlynes | [TK]D-Fender: it works flawlessly for me |
| 15:33.00 | [TK]D-Fender | L2Logic: Highly doubt it... |
| 15:33.01 | coppice | the PAP2 does not support T.38 |
| 15:33.12 | dlynes | coppice: so why does it work for me? |
| 15:33.37 | [TK]D-Fender | dlynes: See? He's convinced. trry giving him reason to reconsider that so we can have some solid info on this! |
| 15:33.46 | L2Logic | we use T.38 on PAP2s day in day out...behind a Audiocodes mediagateway |
| 15:34.41 | wcselby | how about a PAP2T? |
| 15:34.41 | wcselby | or is that the same thing? |
| 15:34.42 | coppice | PAP2T doesn't support T.38 either |
| 15:34.43 | dlynes | hyeh |
| 15:36.44 | coppice | the SPA2102 supports T.38 |
| 15:36.53 | coppice | and the SPA3102 |
| 15:37.07 | coppice | but not very well |
| 15:37.28 | [TK]D-Fender | coppice: Was the "not very well" specifically for the 3102 or the entire line? |
| 15:37.37 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 15:37.38 | L2Logic | the MP202s support t.38 and are relatively comparitively priced to a PAP2T... |
| 15:37.48 | L2Logic | "relatively comparitively" |
| 15:38.03 | coppice | most of the testing I did was with the SPA3102. Its very quirky. |
| 15:38.23 | [TK]D-Fender | coppice: Also I recall sotries that t.38 usage wore out the processors on the SPA and lead to premature device deaths |
| 15:38.39 | coppice | change the lead from the SPA3102 to the FAX machine by a few metres, and the performance changes dramatically |
| 15:39.09 | wcselby | i guess I'm not up to snuff on t.38....isn't it just a protocol for transmiting fax over VoIP? and if your provider supports it, does it really matter if your ATA does? I need to do some reading.......... |
| 15:39.20 | [TK]D-Fender | coppice: is further better, or worse? |
| 15:39.20 | Zeeek | connect pin 3 to pin 16 and it will work a lot better |
| 15:39.38 | coppice | short leads seem to give the most trouble |
| 15:39.49 | L2Logic | both the provider and the ata has to support it... |
| 15:40.03 | wcselby | (like I said, I need to do some reading) |
| 15:40.23 | *** join/#asterisk Skeeter- (n=wil_c_wi@190-141.cgocable.ca) |
| 15:40.31 | coppice | interesting idea that a protocol might work if only one end supports it :-\ |
| 15:40.49 | wcselby | coppice - lol, sorry. it's still early for me |
| 15:41.23 | L2Logic | hmmm, perhaps someday we can have a rj-11 to rj-45 adapter.. plug the ethernet into the phone line, and fax away !!! |
| 15:41.27 | L2Logic | LOL |
| 15:42.18 | bmoraca | or how bout a fax machine with a built-in SIP client |
| 15:42.19 | L2Logic | totally passive.. |
| 15:42.29 | Skeeter- | mediatrix are not cool |
| 15:42.33 | L2Logic | that exists... but it didn't work right... |
| 15:42.40 | L2Logic | at least in our testing... |
| 15:43.18 | L2Logic | i think it was a Ricoh scanner/copier/fax/print server/document server/ ata / coffee maker... |
| 15:43.37 | L2Logic | as I recall, it didn't do anything really well |
| 15:43.56 | coppice | a few expensive do everything machines have T.38 option cards, but I only know one inexpensive T.38 FAX machine |
| 15:44.16 | garymc | anyone in here know how I setup zoiper to connect to my asterisk. Ive opened ports UDP 5060 and 10001-20000 on my router. Do i need to do anything else like add a trunk. Bearing in mind i just want this softphone to act like an extension? |
| 15:44.35 | *** join/#asterisk JimDickenson (n=dickenso@98.232.185.121) |
| 15:44.42 | *** join/#asterisk cuco (n=elcuco@bzq-79-181-138-129.red.bezeqint.net) |
| 15:44.56 | L2Logic | care to share the inexpensive t.38 fax machine? i'd like to test it... |
| 15:45.19 | coppice | its made by sagem. I can't remember the model name |
| 15:46.18 | *** join/#asterisk elguero (n=elguero@12.187.84.162) |
| 15:50.32 | coppice | they make a GSM FAX machine, too. I think its the only one of those |
| 15:54.05 | L2Logic | it's sad.. that you can send a fax over GSM.. but we've spent tens of thousands making it work over ip reliably... |
| 15:54.56 | L2Logic | even in a managed end to end fiber environment with <2ms packet delay with effecitvely zero jitter... |
| 15:55.23 | L2Logic | someone should build a t.37 ata |
| 15:55.39 | coppice | you can only send a FAX over GSM because there are a mechanism in the GSM protocol specifically for this |
| 15:57.14 | *** join/#asterisk blkry (n=chatzill@64.147.222.130) |
| 15:57.59 | coppice | [TK]D-Fender: I suspect that thing about T.38 wearing out PAP2s is an urban myth, since Linksys is definitive that PAP2s and PAP2Ts do not support T.38, there are no T.38 config options in the menus, and they refuse to accept T.38 calls. |
| 16:00.21 | *** join/#asterisk spck (n=spck@unioncab.com) |
| 16:00.30 | bmoraca | i've never had a problem faxing to PAP2Ts using g711. it's never dropped calls or anything. |
| 16:00.58 | *** join/#asterisk pugachevcobra (n=fighter@189.107.81.179) |
| 16:01.01 | Zeeek | join us, live now #voip-users-conference and call in in wideband: http://VUC.me |
| 16:01.16 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 16:01.18 | coppice | bmoraca: then you are fortunate to have a good path. few people are so lucky |
| 16:01.30 | *** join/#asterisk martyndev (i=be18869a@gateway/web/freenode/x-vsnoffahpetkcffo) |
| 16:01.36 | martyndev | hi asterisk people :D |
| 16:03.02 | L2Logic | I just logged into a Pap2 (Firmware Version: 5.1.6(LS)) and it had the fax options on the advanced configuration page... |
| 16:03.04 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 16:03.10 | pugachevcobra | hi people of asteriskya |
| 16:03.22 | coppice | L2Logic: right |
| 16:03.28 | martyndev | I have a question. I have an asterisk where when i do a meetme from a sip user the dtmf is ok and recive ok, but when i recive a call from Zap channel the dtmf failed :(.. do you know why happend this ? |
| 16:03.29 | L2Logic | doesn't say anything about T.38.. but I'm almost sure it the T.38 protocol in it.. advertised or not |
| 16:03.42 | coppice | L2Logic: wrong |
| 16:04.00 | *** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 16:04.02 | L2Logic | i'm gonna have to go wireshark a fax and look at it now.. |
| 16:04.04 | pugachevcobra | My sip trunk provider recommends a disallow=all and allow 3 codecs. THe problem is that if I do as said, my announcement is never played. Is there a way to fix this? Only removing the disallow=all makes the announcement play. |
| 16:04.05 | L2Logic | lol |
| 16:04.16 | docelmo | PAP2's only support FAX via ulaw.. Sipura's support t.38 |
| 16:04.33 | coppice | L2Logic: lots of people swear those things do T.38. When I followed up with people it always turned out to be untrue |
| 16:05.11 | coppice | those FAX config options in the PAP2 are to select fixed buffering when a FAX tone is detected |
| 16:05.16 | L2Logic | for them not to support T.38 (not saying that they do) they sure do work reliably in comparison to A LOT of other vendors... |
| 16:05.54 | coppice | well, their T.38 capable ATAs (SPA2102, SPA3102) don't work too well in T.38 mode :-) |
| 16:06.29 | L2Logic | i'm having excellent results with the SPA8000... but all in all the audiocodes hardware has proven to be the most reliable for us... |
| 16:06.40 | L2Logic | in t.38 |
| 16:06.46 | coppice | I think the SPA8000 supports T.38 |
| 16:06.51 | L2Logic | it does |
| 16:06.56 | coppice | and probably the SPA400 |
| 16:07.14 | L2Logic | SPA400 ? never tried one of those... |
| 16:07.33 | bmoraca | i like the SPA8000...i like that it's got the harmonica connector on it...although I don't like the fact that it has a fan in it. i don't like moving parts in my telephony equipment |
| 16:07.39 | coppice | a 4 port ATA with asterisk inside, doing voice mail :-) |
| 16:07.57 | Kobaz | harmonica? |
| 16:07.58 | L2Logic | we've tried grandstream, linksys, audiocodes, patton, mediatrix, .... and the audiocodes has been the best... |
| 16:08.05 | Kobaz | grandstream is tehsux for phone |
| 16:08.06 | Kobaz | s |
| 16:08.11 | bmoraca | Kobaz: 25-pair amphenol connector |
| 16:08.12 | L2Logic | the amphenol connector |
| 16:08.21 | coppice | granstream ATAs are OK |
| 16:08.26 | Kobaz | bmoraca: yeah i call those amphenol connectors |
| 16:08.27 | Kobaz | heh |
| 16:08.36 | dwery | http://www.chan-mobile.org/ seems gone.... any clue? |
| 16:08.42 | *** join/#asterisk jtodd (i=py13c8ta@ns.fox-den.com) |
| 16:08.42 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 16:08.51 | L2Logic | had to rma a few spa8000s.. you hear hum from the fans in the audio |
| 16:09.04 | bmoraca | Kobaz: harmonica's an old name for them. back when key systems were around...meh |
| 16:09.18 | Kobaz | heh |
| 16:09.26 | L2Logic | fan singular |
| 16:09.56 | coppice | its more likely to be a poor power supply causing that hum |
| 16:10.40 | L2Logic | i swapped the power supply and it was still there.. (power supply being defined as the transformer between the SPA and the well).. could it possibly be bad filter caps on the board? |
| 16:10.58 | L2Logic | wall |
| 16:11.14 | coppice | there is probably further regulation on the board |
| 16:11.30 | bmoraca | that's not encouraging |
| 16:11.51 | bmoraca | i was hoping to use the SPA8000 as a cheaper alternative to the TA908 for providing analog trunks to people |
| 16:12.20 | L2Logic | they are sooo much cheaper than the TA908s... and they work as well.. but occasionally we get a bad one.. |
| 16:12.31 | L2Logic | I would encourage you to test them... |
| 16:12.57 | L2Logic | 4-5 SPA8000s to the 1 TA908 |
| 16:13.13 | bmoraca | i have one, and have been testing it...but have been reticent to install it anywhere...thankfully, i've only had need to provide PRIs to people, so I've been using the TA908e and TA904s for that... |
| 16:13.41 | bmoraca | about 3...i get the 908s for about $800 and the SPA8000s for about $250 |
| 16:14.00 | Kobaz | i've heard the linksys gateways are pretty decent |
| 16:14.03 | bmoraca | i could get them for $212 if Techdata would ever get them in |
| 16:14.07 | L2Logic | I guess I was thinking about the 924s... |
| 16:14.07 | Kobaz | i played with a single port one, it had a decent interface |
| 16:14.18 | bmoraca | 924s are way expensive, lol |
| 16:14.21 | L2Logic | have you had any luck creating a PRI from SIP with the 924s? |
| 16:14.36 | bmoraca | oh yeah...it works wonderfully |
| 16:14.43 | L2Logic | does fax work? |
| 16:15.12 | bmoraca | depends on the connection, and i've found it works better if the TA is NOT using the internal timing, but rather taking it from the T1 port |
| 16:15.47 | bmoraca | of course, that was trying to have NVBackroundDetect detect the fax tones...if I'd been spitting it out to a PAP2T, i probably would have been fine |
| 16:15.53 | L2Logic | that was my question.. I was wondering where the time source would come from.. i've found that the clocks in most PBXs aren't that great either... |
| 16:16.51 | bmoraca | i generally recommend that my customers not take faxes over our trunks anyway, and the major CLEC out here recommends the same |
| 16:17.14 | bmoraca | AT&T measured rate lines are too cheap to deal with the potential problems caused by faxing over our trunks |
| 16:17.31 | L2Logic | well, my service offering is FTTx and the customers seem to want all or nothing.. |
| 16:17.48 | bmoraca | i get 908s for $747 and 908es for $1498 |
| 16:17.54 | L2Logic | wow!! |
| 16:17.56 | L2Logic | not bad... |
| 16:18.25 | bmoraca | never had a need for a 924 yet |
| 16:18.26 | L2Logic | what does the ATT measured rate line run? |
| 16:18.44 | bmoraca | around here, roughly $8/mo |
| 16:18.50 | L2Logic | wow!!! |
| 16:18.52 | viraptor | is there any good way to get information from some other channel? I've got the BLINDTRANSFER variable after some phone transfers, but in the dialplan itself it seems a bit useless... is there a way to get to (for example) callerid of a channel usign that var? |
| 16:19.29 | L2Logic | bomoraca: do you guys pick that up or does the customer get the att bill? |
| 16:19.31 | bmoraca | most of my customers don't deal too much with faxes anymore, and most of the faxes are inbound anyway |
| 16:19.40 | bmoraca | customer gets the AT&T bill |
| 16:19.46 | L2Logic | o |
| 16:19.51 | bmoraca | i've looked in to resale lines from AT&T, but it's just too much work |
| 16:20.07 | bmoraca | most of our customers are coming from AT&T, so I just port over all lines except the fax |
| 16:20.18 | L2Logic | we're heavily debating buying out a clec to overcome the paperwork process... |
| 16:20.28 | IBC_jkenney | ok i have a real world asterisk question |
| 16:20.38 | IBC_jkenney | i want let the sales manager spy on and whisper to the sales reps |
| 16:20.58 | IBC_jkenney | but i only want him to spy and whisper to the people in that department |
| 16:21.06 | IBC_jkenney | has anyone done this |
| 16:21.09 | IBC_jkenney | and how hard is it |
| 16:22.48 | L2Logic | bmoraca: you're in Cali?? |
| 16:22.52 | bmoraca | yes |
| 16:23.18 | L2Logic | i sent you a chat request... (i think) |
| 16:23.48 | L2Logic | it's been a very long time since i've done anything with mIRC.. today is day one.. after about 10 years... |
| 16:25.00 | martyndev | asd |
| 16:25.01 | martyndev | _:D |
| 16:25.02 | martyndev | :D |
| 16:26.17 | wcselby | IBC_jkenney - ChanSpy -> http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy |
| 16:26.21 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
| 16:26.51 | wcselby | also, core show application ChanSpy |
| 16:28.47 | spck | anyone have any advice on dividing trunks into groups and what not? |
| 16:29.29 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
| 16:32.33 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
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| 16:40.30 | pugachevcobra | after clicking a bunch of times on an empty conference in fop, the green circle turned to yellow and i cant get it back to green. what does it mean with the yellow, and how to turn it back? |
| 16:41.57 | eppigy | YOUR FACE |
| 16:43.23 | jaytee | TRABAJO! |
| 16:49.03 | *** join/#asterisk wcselby (n=wcselby@216.110.88.254) |
| 16:49.36 | *** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 16:51.01 | eppigy | DONDE ESTA |
| 16:51.09 | wcselby | your face |
| 16:51.18 | wcselby | laughs quietly at himself |
| 16:51.31 | eppigy | BURN |
| 16:58.58 | wcselby | bummer, flowroute doesn't currently officially support t38 |
| 16:59.07 | [TK]D-Fender | looks at wcselby and just bursts out laughing publicly |
| 16:59.13 | wcselby | so no testing for me |
| 17:00.59 | hardwire | anybody know of a cdma phone that supports voice modem modes? |
| 17:07.52 | *** join/#asterisk raden_work (n=tanning@69-179-99-17.stat.centurytel.net) |
| 17:08.51 | *** join/#asterisk ayeso (n=chatzill@ext-52.sagetelecom.net) |
| 17:09.23 | raden_work | [TK]D-Fender, vitelity wants me to set my host as static ip's is there a good practice ? |
| 17:09.46 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 17:09.51 | [TK]D-Fender | raden_work: which host? |
| 17:10.01 | raden_work | lemee paste what they sent |
| 17:10.26 | raden_work | http://pastebin.com/d31e67de3 |
| 17:10.31 | raden_work | from vitelity.net |
| 17:10.50 | raden_work | having many DNS issues with them |
| 17:11.44 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 17:12.32 | *** join/#asterisk kerchunk (n=kerchunk@173.49.10.152) |
| 17:14.05 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 17:15.03 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 17:15.46 | [TK]D-Fender | raden_work: So do it |
| 17:16.34 | raden_work | you originally told me not to do host=ip |
| 17:16.49 | [TK]D-Fender | raden_work: I most certainly did not. |
| 17:17.21 | spck | does anyone have any advice on splitting trunk groups and using them effectively in the dial plan? |
| 17:17.22 | raden_work | ok guess you didnt |
| 17:18.32 | [TK]D-Fender | spck: WTF is a "trunk group"? |
| 17:18.55 | *** join/#asterisk Skeeter- (n=wil_c_wi@c216.218.2-65.clta.globetrotter.net) |
| 17:18.56 | spck | in chan_dahdi.conf you define groups of channels |
| 17:19.04 | spck | i believe they are called trunk groups |
| 17:19.13 | Skeeter- | dlynes: you still here buddy?? |
| 17:19.21 | spck | so when you are dialing out you specify which group you use |
| 17:19.43 | raden_work | <PROTECTED> |
| 17:20.08 | spck | my understanding is it is a way to guarantee you have available lines for outgoing but not let the incoming take them all |
| 17:20.24 | iksik | how can I include another file in extensions.conf ? |
| 17:20.36 | spck | tk occasionally makes up for his grouchiness by being helpful |
| 17:21.12 | spck | and he's one of the few people who are here consistently and attempt to answer questions |
| 17:21.24 | raden_work | spck, you want to limit the amount of channels available for inbound calls ? |
| 17:21.33 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
| 17:21.43 | spck | well this phone system is for a cab company |
| 17:22.00 | [TK]D-Fender | spck: incoming lands wherever it wants. YOU don't get that choice |
| 17:22.03 | spck | kind of an odd situation, we have normal business lines, and then there's the dipsatch office |
| 17:22.27 | spck | meaning i can't direct a DID to a specific group? |
| 17:22.36 | Chesther | Not within asterisk |
| 17:22.46 | spck | hrmm |
| 17:22.49 | Chesther | The switch at the other end decides which channel to send it down. |
| 17:23.10 | Chesther | You can work with your upstream provider to mark off specific channels for inbound. |
| 17:23.11 | spck | makes sense |
| 17:23.16 | Chesther | For them it will, in fact, be a trunk group. |
| 17:23.26 | Chesther | But it's configured in their switch. |
| 17:23.42 | spck | ok think i can manage that then |
| 17:24.01 | spck | what's the point of the groups in asterisk then? to delimit outgoing? |
| 17:24.31 | Chesther | In part, yes. |
| 17:24.41 | spck | do they have another utility? |
| 17:25.06 | *** join/#asterisk gardo (n=gardo@121.97.215.101) |
| 17:25.09 | Chesther | If your T1 was connected to a channel bank that had half FXS ports and half FXO ports, you'd want to handle htem differently. |
| 17:25.23 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:25.25 | Chesther | So it depends a lot on the setup. |
| 17:25.33 | spck | i don't think that's the case here |
| 17:25.47 | spck | i'm using a redfone fonebridge to send the traffic to the servers |
| 17:26.12 | iksik | ok, #include works fine |
| 17:26.20 | spck | i have to admit i don't know about this end of it |
| 17:28.16 | Chesther | Not particularly familiar with fonebridge. |
| 17:28.22 | *** join/#asterisk came0 (n=came0@rrcs-71-42-53-211.se.biz.rr.com) |
| 17:28.26 | spck | ya it's not popular i'm discovering |
| 17:29.11 | Chesther | The blessing and the curse of opensource telephony is that there are lots and lots of different ways to do it. |
| 17:29.23 | spck | that i am also discovering |
| 17:29.42 | spck | been one hell of a trip so far |
| 17:30.10 | Chesther | I'm currently figuring out how to architect a system that could handle up to 35,000 users. |
| 17:30.23 | spck | all i have to about that is holy fuck |
| 17:30.26 | spck | say* |
| 17:30.29 | Chesther | The big question being what the user/active call ratio would be, as I have no idea. |
| 17:30.36 | docelmo | Chesther: simple |
| 17:30.53 | Chesther | oh? |
| 17:30.54 | docelmo | Chesther: I built one system that is scalable to 50k |
| 17:30.54 | generalhan | can some one help me understand why i cannot press the '#' key once in a call using my linksys 2101 with my polycom soundstation ? same call on any other phone # works just fine. just not with this 2102/soundstation combo |
| 17:31.12 | spck | prolly the digitmap in the phone |
| 17:31.38 | generalhan | so you would assume it is the soundstation, not the linksys device ?! |
| 17:31.43 | docelmo | Chesther: PM me if you want to talk out of the channel about it. |
| 17:31.50 | KyleK | whats the # key supposed to do |
| 17:31.59 | [TK]D-Fender | [13:23]<spck>what's the point of the groups in asterisk then? to delimit outgoing? <- entirely |
| 17:32.20 | generalhan | well for this exact purpose its a conference call. and their system requires a password followed by # |
| 17:32.21 | spck | if you have it set in features.conf you can transfer people with it while you are on the phone with them |
| 17:32.26 | [TK]D-Fender | generalhan: It is the ATA dialplan |
| 17:32.43 | [TK]D-Fender | DTMF transfers = SUCK |
| 17:33.16 | spck | i agree, i've had nothing but trouble with them |
| 17:33.17 | generalhan | [TK]D-Fender: where can i get to that dialplan information? i am in the web config under voice, but cant seem to find the dialplan info |
| 17:33.46 | [TK]D-Fender | generalhan: Look harder. there are only 4 scrrens |
| 17:36.06 | generalhan | [TK]D-Fender: the only screen that really lets me change anything is under the 'User 1' User 2' section and i dont see much there. Call Forward Settings, Selective Call Forward Settings, Speed Dial Settings, Supplementary Service Settings, Distinctive Ring Settings, and Ring Settings |
| 17:36.49 | *** join/#asterisk garymc (n=garymc@host86-173-16-209.range86-173.btcentralplus.com) |
| 17:36.51 | [TK]D-Fender | generalhan: And you aren't looking at USER vs ADMIN, BASIC vs ADVANCED |
| 17:37.15 | wcselby | lunch |
| 17:37.43 | generalhan | [TK]D-Fender: damnit, my fault. i set this thing more than a year ago and havent touched it since. forgot there was an admin section seperate from the other settins |
| 17:38.01 | KyleK | lol |
| 17:38.57 | dustybin | Do i really need a text-to-speech weather and traffic service on my box? |
| 17:39.24 | KyleK | you tell me |
| 17:39.32 | dustybin | i dont know.. |
| 17:40.36 | KyleK | i can see it being handy if you drive and dont like listening to radio for that |
| 17:40.45 | dustybin | yep |
| 17:40.57 | dustybin | but i dont drive, i walk to work |
| 17:41.11 | dustybin | and i dont care about the weather |
| 17:41.23 | Chesther | Then you can call in and find out how screwed your coworkers are. :-) |
| 17:41.30 | dustybin | LOL |
| 17:41.37 | dustybin | i could offer other people the service |
| 17:41.39 | dwery | anyone can suggest a web based application that shows nifty statistics on asterisk usage, like calls per day et al. ? |
| 17:41.52 | dustybin | and buy a special VOIP number what charges £5 per min |
| 17:42.18 | *** join/#asterisk louben (n=lou@212-70-216-131.ath.static.tee.gr) |
| 17:42.37 | dustybin | dwery: i like the sound of that, something very light weight |
| 17:42.54 | dustybin | you cannot always ssh into a shell when your away |
| 17:43.06 | dwery | dustybin: I can, but the customer cannot :D |
| 17:43.11 | dustybin | yep |
| 17:43.27 | dwery | ( I won't let he in anyway) |
| 17:44.54 | dustybin | i think i will setup my private menu tonight |
| 17:45.01 | dustybin | that i _do_ need |
| 17:45.01 | garymc | using zoiper in the office to connect to the asterisk pbxthrough my windows laptop wireless it all worked great with sound etc. Now I get home connect ok. Zoiper registers the extension rings fine but the phones cant hear each other |
| 17:45.16 | garymc | why? |
| 17:45.27 | garymc | Do i need to also open ports in my home router? |
| 17:45.46 | dustybin | garymc: sip or iax? |
| 17:45.51 | garymc | sip |
| 17:45.55 | garymc | as far as im aware |
| 17:46.07 | dustybin | sip requires special settings to work behind NAT |
| 17:46.24 | garymc | it all worked great back in the office, but at home I can call the office extensions but we cant hear each other |
| 17:46.32 | garymc | Also the office get a busy tone when calling me |
| 17:46.33 | dustybin | garymc: http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:47.01 | garymc | do you think i need to open any ports on my home broaband router? |
| 17:47.33 | dustybin | garymc: you need to forward port 5060 UDP + 10000-20000 UDP |
| 17:47.44 | garymc | at home? |
| 17:47.48 | dustybin | those are essential for SIP |
| 17:47.50 | garymc | I did this back at the office |
| 17:47.53 | dustybin | no, only on the server |
| 17:47.57 | garymc | on the office router |
| 17:48.07 | dustybin | but you need to make sure outbound ports are open |
| 17:48.14 | dustybin | at home |
| 17:48.15 | hesco | when voip-info shows this example: SoftHangup(SIP/1234|a), is that '1234' an extension number? or a channel? The syntax reads: SoftHangup(Technology/resource[|a]). So what exactly is a 'resource' ??? |
| 17:48.54 | garymc | ok so i just open the same ports on my home router? |
| 17:50.08 | KyleK | home routers typically dont restrict outgoing ports |
| 17:50.22 | garymc | so why am i getting no sound when i get home? |
| 17:51.15 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 17:51.23 | hesco | I just experienced something similar here, was able to make a polycom 501 ring my ht-486, but no audio across the connection. |
| 17:51.39 | hesco | cli*> complained of inability to register, |
| 17:51.54 | hesco | no matching peer |
| 17:52.15 | hesco | does the softphone's config agree with your sip.conf? |
| 17:52.49 | *** join/#asterisk seanmh (n=johndoe@207.114.199.107) |
| 17:53.26 | garymc | as i say it all worked fine when i was connecting through my office internet. but I was aiming back at the same connection so maybe thats why it worked. Now when i get home, zioper will ring a connection but no voice and the extensions cant hear me |
| 17:53.43 | garymc | [TK]D-Fender can you shed some light here? |
| 17:54.19 | [TK]D-Fender | garymc: All of the SIP & NAT settings n3glv told you HOURS ago and you never followed |
| 17:54.29 | garymc | I did follow |
| 17:54.34 | garymc | The worked great |
| 17:54.37 | [TK]D-Fender | garymc: PASteBIN <- |
| 17:54.43 | garymc | im not in the office |
| 17:54.45 | garymc | now |
| 17:54.48 | garymc | im at home |
| 17:54.50 | garymc | :( |
| 17:54.57 | [TK]D-Fender | garymc: Don't ask for an autopsy when you can't show me the body |
| 17:55.16 | garymc | it worked great in the office, so was wondering do i need to set ports at home router? |
| 17:55.23 | [TK]D-Fender | garymc: No |
| 17:55.23 | generalhan | LOVES [TK]D-Fender's analogies !!!! |
| 17:55.44 | TJNII | Yea, [TK]D-Fender Has been on a roll with the analogies lately. |
| 17:55.45 | *** join/#asterisk Skeeter- (n=wil_c_wi@c216.218.2-65.clta.globetrotter.net) |
| 17:56.04 | garymc | ok so i need someone out of the office while im in the office to diagnose? |
| 17:56.15 | garymc | sighs |
| 17:56.17 | TJNII | garymc: Or, you know, ssh |
| 17:56.40 | garymc | asterisk aint setup for ssh yet |
| 17:57.16 | TJNII | So install it. It takes what, 2 minutes? |
| 17:57.36 | [TK]D-Fender | gareven SMARTER. |
| 17:57.42 | [TK]D-Fender | garymc: even SMARTER. |
| 17:58.02 | garymc | ok thanks for the barracking |
| 17:58.14 | garymc | right ill sort ssh out this weekend |
| 17:58.14 | TJNII | That's what we're here for. :) |
| 17:58.24 | TJNII | garymc: You'll be better off for it. |
| 17:58.28 | garymc | i know |
| 17:58.44 | garymc | thanks again, just it worked in the office a treat, thought it would from home too |
| 17:59.37 | garymc | The reason it worked in the office, would it be cos I was on the same internet connection? or does that not matter? |
| 18:00.19 | [TK]D-Fender | garymc: guessing is a awaste of time. |
| 18:00.59 | garymc | yeah... i realise that, but is it reasonable to think that could be why it worked? |
| 18:06.01 | [TK]D-Fender | garymc: Guessing is not reasonable, and my trust is attempting to go negative... |
| 18:08.17 | jplank | anyone every integrate * with Microsoft Dynamic CRM? |
| 18:09.43 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 18:13.21 | *** join/#asterisk superc (n=superc@87.234.216.82) |
| 18:13.39 | *** part/#asterisk superc (n=superc@87.234.216.82) |
| 18:14.31 | *** join/#asterisk Naikrovek (n=jjohnson@98.214.112.102) |
| 18:24.55 | *** join/#asterisk ethicx (n=chatzill@host-208-88-126-198.biznesshosting.net) |
| 18:25.12 | spck | how did he manage to setup a box without ssh? |
| 18:27.49 | Qwell | spck: don't ask |
| 18:27.54 | *** join/#asterisk pzn (n=pzn@187.23.90.124) |
| 18:28.22 | pzn | how can I specify that if someone dials to extension 204 and is it busy, I'd like it to try automatically ext 208? |
| 18:29.56 | docelmo | use gotoif and check for ${DIALSTATUS} == BUSY |
| 18:30.33 | pzn | docelmo: how could I do this (asterisk novice)? |
| 18:31.03 | pzn | docelmo: just a simple example or some url for me to search would be enough for a quickstart... |
| 18:31.32 | docelmo | www.voip-info.org is your friend |
| 18:31.35 | *** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 18:31.46 | docelmo | live it love it like it then come here and ask if you cant find it. |
| 18:38.31 | garymc | spck: I have got ssh when im in the office |
| 18:38.50 | garymc | just dont know how to connect to it from home |
| 18:38.50 | leifmadsen | anyone use OpenSIPS as a registrar and Asterisk with peers in realtime (SIP) and got MWI to work? |
| 18:42.02 | *** join/#asterisk Failrar (n=Failrar@2001:470:1f15:316:2a0:d1ff:fe4e:e802) |
| 18:47.22 | *** join/#asterisk l2cache (n=l2cache@adsl-75-21-103-128.dsl.rcfril.sbcglobal.net) |
| 18:48.30 | *** join/#asterisk Skeeter- (n=wil_c_wi@c216.218.2-65.clta.globetrotter.net) |
| 18:48.32 | [TK]D-Fender | garymc: Ever heard of TCP/IP? I hear it even works over the internets. Yes, BOTH of them... |
| 18:48.55 | Skeeter- | How to resgister a mediatrix with asterisk??? |
| 18:49.01 | hesco | when voip-info shows this example: SoftHangup(SIP/1234|a), is that '1234' an extension number? or a channel? The syntax reads: SoftHangup(Technology/resource[|a]). So what exactly is a 'resource' ??? |
| 18:49.56 | leifmadsen | hesco: the [name] in sip.conf |
| 18:50.05 | [TK]D-Fender | Skeeter-: I'm sure it has some VErY FINE MANUALS... |
| 18:50.22 | leifmadsen | just like you would do Dial(SIP/1234) if the peer was defined as [1234] in sip.conf |
| 18:50.31 | [TK]D-Fender | hesco: peer, etc. Like what you would DIAL |
| 18:50.37 | leifmadsen | (although using extension numbers for peers is not a great methodology) |
| 18:50.41 | Skeeter- | Fender: i follow the last 3 manual edition and it didnt work |
| 18:50.44 | leifmadsen | [TK]D-Fender: jinx! more beers! |
| 18:51.08 | Naikrovek | garymc: you need to open port 22 on the firewall at your work. that's the SSH connection port. You can open it so you can access it from home only or the entire internet |
| 18:51.37 | *** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be) |
| 18:52.31 | ethicx | Hello everyone. Does anyone have any feedback with this provider 'Telasip;? |
| 18:52.38 | ethicx | Telasip |
| 18:54.01 | Naikrovek | spongebob is kinda underrated |
| 18:54.08 | Naikrovek | never heard of telasip |
| 18:54.15 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 18:54.17 | KyleK | ditto |
| 18:54.57 | *** join/#asterisk afink (n=afink@204.26.87.226) |
| 18:55.26 | ethicx | humm..just curious to know who you guys recommend as a voip provider for a business, the one we currently have really sucks. |
| 18:55.45 | KavanS | ethicx, I use voicepulse and vitelity |
| 18:57.10 | ethicx | KavanS: oh ok, any feedback on those two? reliable? |
| 18:57.28 | KavanS | ethicx, I've had pretty good results....some quirks here and there, but overall quite reliable |
| 18:58.57 | ethicx | KavanS: Oh ok, yeah I have etherspeak but they are having problems since the morning and we can't make outbound calls since then.. |
| 19:00.12 | ethicx | KavanS: and with so many VoIP providers out there it's hard to pick one without some sort of feedback from others =D |
| 19:00.12 | leifmadsen | I've had good luck with Unlimitel (Canada) and Bandwidth.com (USA) |
| 19:00.42 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:02.09 | louben | 8a elega kati twra |
| 19:02.11 | Defraz | In a dial plan can I do a _X. to catch anything? |
| 19:02.25 | Defraz | exten => _X.,1,Playback(your-account) |
| 19:03.13 | Defraz | I use vitelity and voipstreet and I seem to have some good luck. |
| 19:03.18 | KyleK | ooo 3.3 cents a min incoming on toll free numbers at unlimitel |
| 19:03.39 | [TK]D-Fender | Defraz: that will catch anything dialed that starts with a digits and is at least 2 chars long |
| 19:07.36 | Naikrovek | ethicx: where are you physically? what state |
| 19:08.15 | Naikrovek | i use voicespringvoip.com and have never had a single issue with them whatsoever |
| 19:08.40 | ethicx | Naikrovek: FL |
| 19:08.58 | Naikrovek | i don't think they have a POP near you though |
| 19:09.11 | Naikrovek | NY, IL, CA and I think TX |
| 19:09.30 | timeshell | [TK]D-Fender Greetings. Still no joy killing the buddies option |
| 19:09.44 | Naikrovek | ah i was wrong |
| 19:09.50 | Naikrovek | ethicx: they have a POP in GA |
| 19:10.07 | Defraz | Slick that worked nicely. |
| 19:10.15 | wcselby | argh! |
| 19:10.38 | Naikrovek | wcselby: argh? |
| 19:10.46 | wcselby | polycom soundstation ip 7000 - boots up, gives error "Config File Error; Error is 0x0" |
| 19:11.00 | wcselby | this is straight out of the box |
| 19:11.05 | wcselby | well, not quite |
| 19:11.05 | iksik | huh, realtime asterisk dynamic + static rox ;) |
| 19:11.09 | ethicx | Naikrovek: Ah Ok, I will check them out |
| 19:11.20 | Naikrovek | wcselby: had the phone been previously configured? |
| 19:12.03 | wcselby | Naikrovek - i misstated. I added the server address to the config, and it downloaded a new firmware off of our server - i think. i'm working through this, it's just annoying. |
| 19:12.33 | Naikrovek | wcselby: i have lots of experience with polycom so ask if you have Qs |
| 19:12.49 | Naikrovek | check the (t)ftp log to make sure it's downloading what you think it's downloading |
| 19:19.17 | *** join/#asterisk propellerhead (n=yogurt2u@host186.190-30-30.telecom.net.ar) |
| 19:19.19 | wcselby | Naikrovek - do you have any soundstation IP 7000's ? |
| 19:19.38 | *** part/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 19:19.54 | *** join/#asterisk stimpie (n=stimpie@84-104-5-142.cable.quicknet.nl) |
| 19:23.29 | Naikrovek | wcselby: have a 6000 |
| 19:24.59 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 19:25.05 | Naikrovek | man it's slow in here today |
| 19:29.02 | p3nguin | I have a single SIP trunk from a VoIP provider to my main asterisk box in the main office. I have a second location that needs phones. Should I have each of the phones in the remote office make peer connections over the internet back to the main asterisk box, or should I set up a second asterisk box in the remote office, letting the phones make peer connections to it (locally), and subsequently have the second asterisk box ... |
| 19:29.05 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
| 19:29.08 | p3nguin | ... form a trunk back to the main asterisk box? |
| 19:29.16 | *** join/#asterisk jtodd (i=nah0kyfd@ns.fox-den.com) |
| 19:29.16 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 19:31.46 | wcselby | p3nguin - depends on if you have an extra server available that can act as a proxy for you |
| 19:32.18 | p3nguin | If I do have a box available to set up the second switch on, is that the correct way to set things up? |
| 19:32.34 | p3nguin | I'm not sure if I'm creating extra work for myself without gaining any advantage. |
| 19:33.09 | p3nguin | It's not a lot of work, so any advantage at all is probably worth it. |
| 19:33.45 | [TK]D-Fender | p3nguin: Box on site means if the link goes down at least they can talk amonsgt themselves. Also outbout call s need not waste double the BW at central |
| 19:34.06 | wcselby | there you go |
| 19:34.15 | wcselby | those are better advantages than what I was going to say |
| 19:35.01 | p3nguin | Okay, so a switch at each location sounds like the best way to handle it. |
| 19:35.01 | *** part/#asterisk asterwiki (n=asterwik@69.77.169.14) |
| 19:36.26 | KyleK | peer to peer between phones sounds like a pain in the ass |
| 19:36.54 | KyleK | whoops you didn't say anything about that |
| 19:38.51 | *** join/#asterisk tripps (n=tripps@nmd.sbx06320.housttx.wayport.net) |
| 19:39.09 | tripps | how do I force log an agent off where I used agentcallbacklogin()? |
| 19:40.43 | *** join/#asterisk mahlon (i=mahlon@martini.nu) |
| 19:41.15 | wcselby | tripps - exten => 2002,2,System(asterisk -rx "agent logoff Agent/${AGENTBYCALLERID_${CALLERID(number)}}") |
| 19:41.43 | wcselby | that's how we do it |
| 19:41.57 | *** join/#asterisk shriven (n=shriven@64.105.202.242) |
| 19:42.17 | [TK]D-Fender | tripps: AgentCallbackLogin() <- |
| 19:43.05 | tripps | wcselby, I already have the logoff extension, etc., set up for them. this is a 1 time I need to kick him off from the console request |
| 19:43.17 | shriven | Hello. Sorry if this is a common question.... Does anyone know if it is possible to get asterisk to transcode voicemail and send the transcription as an email with the voicemail recording? I already have imap voicemail going, but I am trying to figure out if I can add a transcription to that. |
| 19:43.32 | wcselby | tripps - agent logoff Agent/AGENT# |
| 19:43.54 | tripps | wcselby, ah sweet. duh :) |
| 19:44.02 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 19:46.55 | KyleK | shriven: I haven't come across any howtos on that yet :-/ |
| 19:47.33 | KyleK | I'm hoping I'll somehow be able use google voice in the future for that |
| 19:49.45 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 19:50.12 | shriven | kylek: indeed, that is looking like the most reasonable option... But I want to keep my voicemail management on my servers... : ( |
| 19:50.27 | shriven | Not exactly sure how it would all work, just trying to dig into it for the first time |
| 19:56.03 | *** join/#asterisk evil_gordita (n=evilgord@24.254.160.77) |
| 19:57.34 | *** join/#asterisk simcop2387 (n=simcop23@p3m/member/simcop2387) |
| 20:01.37 | AeroCloud | I have a question: Basically I have multiple did's incoming to dial out like calling cards, where they dial a phone number and it connects them. Anyone know an easy way to Block multiple calls from same caller id? Or a way to disconnect the previous call that was made by the caller? |
| 20:02.45 | timeshell | [TK]D-Fender AgentCallbackLogin is deprecated. |
| 20:03.17 | timeshell | RemoveQueueMember |
| 20:03.43 | [TK]D-Fender | timeshell: And he expressly mentioned how he ADDED them. That dictates the removal method |
| 20:03.51 | timeshell | meh |
| 20:03.57 | timeshell | Ok, so about buddies |
| 20:04.20 | [TK]D-Fender | AeroCloud: GROUP_COUNT() + CAlleriD() |
| 20:04.23 | AeroCloud | I run everything through an AGI script, do you know how I would find the persons call, access it and hang it up ? |
| 20:04.44 | *** join/#asterisk asif (n=chatzill@61.17.225.68) |
| 20:04.58 | KyleK | you might have to use the management interface from the agi |
| 20:05.12 | [TK]D-Fender | AeroCloud: AGI... management interface? HUH? |
| 20:05.22 | [TK]D-Fender | KyleK: ^ |
| 20:05.32 | AeroCloud | AGI processes the calls |
| 20:05.44 | KyleK | well can an AGI look around at other calls? |
| 20:05.54 | AeroCloud | thats my question ^ |
| 20:06.07 | AeroCloud | and if it can, how would it access and hangup the first one. |
| 20:06.08 | [TK]D-Fender | AeroCloud: Yes, you should probably not kill off an existing call, but rather abort the new one |
| 20:06.18 | AeroCloud | ok aborting new one is ok too |
| 20:06.25 | [TK]D-Fender | AeroCloud: soft hangup [channel] |
| 20:06.26 | AeroCloud | but I was thinking if there was a hung call |
| 20:06.33 | AeroCloud | to kill off first one. |
| 20:06.38 | [TK]D-Fender | AeroCloud: do you have a porblem with hung calls? |
| 20:06.48 | AeroCloud | none, at this time |
| 20:06.59 | AeroCloud | just trying to think of issues before they happen |
| 20:07.36 | [TK]D-Fender | AeroCloud: You mean fixing things that aren't broke and setting up future problems when 2 people sharing a CC try to use it and they cut each other off all the time |
| 20:07.38 | [TK]D-Fender | SMRT |
| 20:07.41 | AeroCloud | so how would I find a channel? |
| 20:07.44 | asif | hey anyone here using MySQL CDR with Asterisk 1.6? |
| 20:07.49 | AeroCloud | not fixing anything not broken |
| 20:07.53 | [TK]D-Fender | AeroCloud: Parse out the channel list |
| 20:08.02 | [TK]D-Fender | AeroCloud: core show channels concise |
| 20:08.08 | [TK]D-Fender | AeroCloud: any of the Ami functions, etc |
| 20:10.31 | AeroCloud | if there are multiple servers? |
| 20:10.46 | AeroCloud | lets say we have 20 asterisk servers handling this |
| 20:11.10 | asif | how do i customize mysql cdr? include only the fields i want, is it possible? |
| 20:11.19 | KyleK | AeroCloud: then you need to check all servers somehow |
| 20:11.22 | AeroCloud | asif... write your own AGI CDR |
| 20:11.28 | AeroCloud | thats what I had to do |
| 20:11.37 | KyleK | erm |
| 20:13.24 | [TK]D-Fender | AeroCloud: I guess you had really better think of the PROGRAMMING this will require. |
| 20:13.57 | AeroCloud | yeah its for a large company that supports 200,000 concurrent users |
| 20:14.06 | AeroCloud | there are 100's of boxes |
| 20:14.13 | asif | but can't i use Asterisk's cdr? |
| 20:14.35 | KyleK | asif: the cdr_mysql doesn't look like you can skip fields |
| 20:14.58 | AeroCloud | you can write a cronjob that will pull only the fields you want and insert them into a different table |
| 20:15.03 | KyleK | you could just ignore them when you use the cdr records, or try the odbc cdr |
| 20:15.10 | [TK]D-Fender | asif: Or a trigger on add. |
| 20:15.24 | KyleK | ooo I like that one |
| 20:15.29 | asif | Hmm... |
| 20:15.58 | asif | ok but can i include custom fields? read somewhere it's possible in 1.6 |
| 20:16.13 | AeroCloud | you can specify the accountcode and userfield |
| 20:16.28 | AeroCloud | use those 2 fields, and pipe delimit the multiple items |
| 20:16.38 | AeroCloud | parse it out and put it into multiple columns |
| 20:17.11 | *** join/#asterisk markgreene (n=markgree@71-12-177-173.dhcp.leds.al.charter.com) |
| 20:17.15 | AeroCloud | mysql cdr, is pretty fixed as it is.. you can specify the other forms of cdr though |
| 20:18.06 | AeroCloud | if you need to, change the column type/size of the accountcode/userfield if its too small |
| 20:23.15 | asif | that's an idea. |
| 20:27.28 | *** join/#asterisk [KS] (n=Ask@dynamic-155-129-241-92-cmtsbl.broadband.blic.net) |
| 20:27.41 | [KS] | good evening |
| 20:29.04 | [KS] | no good evening? :] |
| 20:29.19 | Naikrovek | ge |
| 20:29.19 | asif | hey good evening! :) |
| 20:29.41 | [KS] | hey :) |
| 20:29.53 | [KS] | okay to ask questions here about asterisk here? :) |
| 20:29.59 | asif | thanks for the tip AeroCloud |
| 20:30.03 | iflux | re all |
| 20:30.05 | iflux | sup asif |
| 20:30.23 | asif | hey iflux |
| 20:30.43 | asif | i'm already using accountcode and userfield |
| 20:31.48 | [KS] | could I get some help about DAHDI here? |
| 20:32.22 | timeshell | That's your first wish of 3. The answer is yes. |
| 20:32.33 | [KS] | good |
| 20:32.39 | [KS] | just one question tho, so I am good :] |
| 20:32.42 | [TK]D-Fender | ~ask |
| 20:32.43 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 20:32.44 | [TK]D-Fender | ^^^^^^ |
| 20:32.50 | timeshell | ~questions |
| 20:32.51 | infobot | remember, there are no stupid questions, just stupid people. <http://www.catb.org/~esr/faqs/smart-questions.html> |
| 20:33.07 | [KS] | I am trying to install DAHDI on RPS |
| 20:33.17 | timeshell | On what? |
| 20:33.20 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 20:33.36 | [KS] | rps |
| 20:33.39 | [KS] | real private server |
| 20:33.41 | [KS] | they netboot |
| 20:33.54 | [KS] | iSCSI |
| 20:33.59 | timeshell | ok |
| 20:34.02 | [KS] | anyway, linux-headers-2.6.28.4-xxxx-std-ipv4-32 is my kernel |
| 20:34.04 | Naikrovek | neat |
| 20:34.18 | [KS] | source and headers are required for dahdi |
| 20:34.28 | [KS] | so I downloaded 2.6.28.4 source |
| 20:34.35 | [KS] | with config for xxx-std-ipv4-32 |
| 20:34.44 | [KS] | which I got from my RPS provider |
| 20:35.06 | [KS] | so, the question is, how do I tell DAHDI when using make all, where to look for source |
| 20:37.03 | [KS] | no clue? :] |
| 20:37.12 | timeshell | checking |
| 20:37.14 | timeshell | lol |
| 20:37.14 | wcselby | what happens when you try to make all now? |
| 20:37.34 | [KS] | You do not appear to have the sources for the 2.6.28.4-xxxx-std-ipv4-32 kernel installed. |
| 20:37.41 | timeshell | Where'd you get the kernel source from? Is it the source for your std-ipv4-32 kernel binary? |
| 20:37.50 | timeshell | yep |
| 20:37.52 | timeshell | lol |
| 20:37.54 | timeshell | That was my question |
| 20:37.56 | wcselby | and did you install the source? |
| 20:38.01 | wcselby | bleh, stupid question |
| 20:38.02 | wcselby | it's source |
| 20:38.09 | timeshell | http://forum.ovh.com/archive/index.php/t-36208.html |
| 20:38.10 | [KS] | lol :] |
| 20:38.23 | [KS] | I installed asterisk |
| 20:38.51 | timeshell | [KS] That link might help you. It's a french forum, but has some english guidance in it. |
| 20:39.01 | timeshell | Use google to translate rest of it. :p |
| 20:39.05 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 20:40.41 | asif | So is it like Asterisk 1.6 supports custom cdr fields, but not with MySQL? |
| 20:40.57 | wcselby | [KS] - look for the Hoggins! post near the end of the page timeshell linked you |
| 20:41.19 | [KS] | OVH is dumb |
| 20:41.28 | [KS] | my server is with OVH btw :) |
| 20:41.39 | timeshell | [KS] You need the source for that kernel. |
| 20:41.46 | [KS] | I know |
| 20:41.46 | wcselby | haven't ever used that, so I don't know how to help |
| 20:41.52 | [KS] | figured |
| 20:41.57 | [KS] | I need to recompile kernel with their config |
| 20:42.00 | [KS] | and boot with that |
| 20:42.16 | timeshell | You might be able to fool it somehow |
| 20:42.24 | timeshell | But can't be sure it will work. |
| 20:43.08 | [KS] | I download 2.6.8.24 and their .conf |
| 20:43.14 | [KS] | I'll compile it and add it to LILO |
| 20:43.24 | [KS] | hopefully it will not terminate my server :] |
| 20:43.50 | timeshell | Rename the 2.6.8.24 folder on their server to the folder name it's expecting? |
| 20:44.11 | [KS] | there is no their 2.6.8.24 folder |
| 20:44.21 | [KS] | I downloaded kernel source for that version |
| 20:44.32 | [KS] | so I will compile it now against their .conf file |
| 20:44.36 | [KS] | and hopefully it will work |
| 20:44.39 | [KS] | :] |
| 20:45.13 | [KS] | well, make has commenced :P |
| 20:46.22 | dustybin | anybody here using queue features? |
| 20:46.23 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 20:51.27 | wcselby | dustybin - some |
| 20:51.30 | wcselby | what's your question? |
| 20:51.46 | [KS] | lol timeshell |
| 20:52.02 | [KS] | ln -s /usr/src/linux-2.6.8.24 /usr/src/linux |
| 20:52.10 | [KS] | seems to be doing the job :] |
| 20:52.27 | timeshell | Glad I could help |
| 20:52.29 | timeshell | :) |
| 20:52.56 | [KS] | I am not going to interrupt the kernel compile tho |
| 20:53.05 | [KS] | Maybe this "make all" will fail after all |
| 20:53.29 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 20:58.05 | *** join/#asterisk simcop2387 (n=simcop23@p3m/member/simcop2387) |
| 20:59.51 | *** part/#asterisk simcop2387 (n=simcop23@p3m/member/simcop2387) |
| 21:05.52 | dustybin | is this a good guide to follow for queues? |
| 21:05.53 | dustybin | http://www.voipplanet.com/backgrounders/article.php/3633901 |
| 21:06.54 | [TK]D-Fender | dustybin: probably somewhat... it is 3 years old |
| 21:07.09 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.117.25) |
| 21:07.13 | DelphiWorld | hi all! |
| 21:07.19 | dustybin | the book isnt clear with regards to queues |
| 21:07.20 | [TK]D-Fender | dustybin: And using 1.0 stuff that is shattered in 1.6 |
| 21:07.24 | DelphiWorld | please IaxTel is allready existe? or discontinued? |
| 21:07.26 | DelphiWorld | i get: |
| 21:07.27 | DelphiWorld | 503 - Service Not Available |
| 21:07.28 | [TK]D-Fender | dustybin: read the SAMPLE configs |
| 21:07.35 | dustybin | i am! |
| 21:07.36 | [TK]D-Fender | dustybin: and the app instructions |
| 21:07.47 | eppigy | HALIBURTON STOCK HOLDER PICNIC DAY |
| 21:08.25 | dustybin | i dont understand how a queue gets activated |
| 21:08.39 | hesco | I'm encountering an error in the CLI*> console reading "Huh? Child handler, but nobody there?", apparently emanating from: child_handler() in asterisk-1.6.0.3-rc1/main/asterisk.c Can anyone here offer some illumination on what this error may be telling me? |
| 21:08.50 | dustybin | do you need to manually put in: if line is busy then > goto queue ? |
| 21:09.16 | dustybin | confusing stuff |
| 21:09.55 | p3nguin | dustybin: You'll dial an extension to get into a queue. |
| 21:10.27 | p3nguin | The line will never be busy when you have queues. |
| 21:10.27 | dustybin | p3nguin: but you dont want to be put in a queue if somebody answers |
| 21:10.28 | p3nguin | actually you do. |
| 21:10.34 | p3nguin | That's the point of a queue. |
| 21:10.35 | wcselby | dustybin - instead of doing a "Dial()" command, you do a "Queue()" command |
| 21:10.36 | dustybin | is double confusued |
| 21:10.38 | wcselby | in extensions.conf |
| 21:10.50 | dustybin | Queue can also Dial? |
| 21:11.18 | wcselby | it's a whole separate concept |
| 21:11.28 | dustybin | is triple confused |
| 21:11.34 | p3nguin | Call comes in. Your greeting plays and then your caller will enter into a queue. At this time, the queue agents' phones will begin ringing as defined. |
| 21:11.37 | *** part/#asterisk DelphiWorld (n=Miranda@41.201.117.25) |
| 21:12.18 | dustybin | i see.. |
| 21:12.39 | dustybin | but you dont want a greet to play if the queue is empty |
| 21:12.43 | hesco | runninf 1.6, core show applications like Queue gives me nothing. That looks useful, how would I add that command |
| 21:12.44 | dustybin | you just want it to ring normally |
| 21:13.02 | p3nguin | That's done with the queue. |
| 21:13.08 | dustybin | right.. |
| 21:13.20 | dustybin | so all i need to edit is |
| 21:13.26 | dustybin | extensions.conf, change dial to queue |
| 21:13.32 | dustybin | and also edit queues.conf |
| 21:13.34 | p3nguin | If agents 2010, 2011, and 2012 are all logged in and waiting for calls in the queue, then the caller will enter into the queue and 2010 might be able to answer the call right away. |
| 21:13.54 | wcselby | dustybin, http://www.voip-info.org/wiki/view/Asterisk+call+queues read this and it should help you start to udnerstand |
| 21:13.56 | dustybin | what is a agent? |
| 21:14.03 | dustybin | a caller? |
| 21:14.21 | dustybin | oe 007 :P |
| 21:14.24 | dustybin | *or |
| 21:14.29 | p3nguin | The agent is the person accepting calls at your call center. You can also make the queue ringall, which will call all agents' at once. |
| 21:14.39 | wcselby | agent is someone who is supposed to answer a call that comes into a queue. |
| 21:14.50 | dustybin | so im the agent! |
| 21:14.52 | *** join/#asterisk tamiel (n=tamiel@ip-7.net-81-220-254.rev.numericable.fr) |
| 21:15.04 | p3nguin | If you are waiting for calls in the queue, you are the agent. |
| 21:15.13 | dustybin | ok |
| 21:15.19 | dustybin | agent 007 |
| 21:16.11 | TJNII | You don't have to use agents, either. You can just define specific phones to ring in a queue. Like member => SIP/1214 |
| 21:16.27 | p3nguin | The agent number will be your number you use to log in and out of the queues. |
| 21:16.27 | dustybin | there are 3 configs!!!!!!! queues.conf agents.conf and extensions.conf |
| 21:16.33 | dustybin | 3 configs for a queue! |
| 21:16.50 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 21:16.51 | TJNII | Not really.... |
| 21:16.53 | wcselby | dustybin - once you get your head around them, queue's are some of the simplest things to configure in asterisk |
| 21:17.03 | dustybin | o_0 ok.. |
| 21:17.14 | dustybin | i will take your word for that |
| 21:17.52 | p3nguin | It's not that bad once you learn what the compoents do. |
| 21:18.14 | dustybin | i will try again with the book, which i own a hard copy of |
| 21:18.25 | *** part/#asterisk korihor (n=korihor@190.77.83.180) |
| 21:19.27 | dustybin | ok |
| 21:19.48 | dustybin | if i do this: exten = > 007,1,Answer() |
| 21:20.12 | dustybin | exten = > 007,2,Queue(me) |
| 21:20.23 | dustybin | am i right in thinking that i will need a |
| 21:20.26 | dustybin | [me] |
| 21:20.32 | dustybin | blah blah inside queues.conf ? |
| 21:20.45 | p3nguin | If you have a queue named 'me' in there, then you are on the right track. |
| 21:20.52 | dustybin | right i see |
| 21:21.04 | dustybin | do i also need to edit agents.conf ? if so what for? |
| 21:21.26 | TJNII | You only need to edit agents.conf id you are using agents. |
| 21:21.28 | p3nguin | You can statically assign members to the queue. |
| 21:21.47 | TJNII | Like member => SIP?1234 |
| 21:21.47 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 21:21.58 | TJNII | s/SIP?1234/SIP//1234/ |
| 21:22.00 | p3nguin | It's usually not recommended because there are some drawbacks, but it sounds to me like you won't need to log in and out. |
| 21:22.00 | dustybin | am i using agents or not? |
| 21:22.14 | p3nguin | Do you want to be able to log in and out of hte queue? |
| 21:22.16 | TJNII | To start, I would recommend not :) |
| 21:22.21 | dustybin | right ok! |
| 21:22.28 | dustybin | no i dont need anything complicated |
| 21:22.36 | dustybin | maybe a message like |
| 21:22.41 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 21:22.45 | p3nguin | If you'll always be at the same phone, don't worry about agents right now. |
| 21:22.47 | dustybin | "you are number 3 in the queue" |
| 21:22.54 | p3nguin | It's built in. |
| 21:23.04 | dustybin | "sorry to keep you waiting, dustybin will be with you soon" |
| 21:23.09 | dustybin | ace |
| 21:23.18 | p3nguin | If I call into my own queue right now, it says "your call is now first in line..." |
| 21:23.27 | dustybin | ace |
| 21:23.44 | dustybin | its time to edit my queue context |
| 21:28.13 | *** part/#asterisk asif (n=chatzill@61.17.225.68) |
| 21:32.03 | *** join/#asterisk davidandgoliath (n=David@out.clearnet.com) |
| 21:33.04 | *** join/#asterisk asif (n=chatzill@61.17.225.68) |
| 21:33.21 | asif | how do i join asterisk forums? |
| 21:33.35 | asif | i can't find a register link anywhere! :-/ |
| 21:36.19 | [KS] | Registration from '<...>' failed for 'IP' - No matching peer found |
| 21:36.22 | [KS] | means what |
| 21:36.24 | [KS] | ? |
| 21:36.37 | wcselby | asif - http://forums.digium.com/profile.php?mode=register |
| 21:36.58 | dustybin | wow there are a _LOT_ of options in queues.conf |
| 21:37.06 | dustybin | at the end i have put: member => Agent/1010 |
| 21:37.29 | dustybin | 1010 is the extension for my phone |
| 21:37.47 | p3nguin | dustybin: Comment that out for now, then add "member => SIP/1010" after it. |
| 21:38.04 | dustybin | IAX2/10101 |
| 21:38.10 | dustybin | *1010 |
| 21:38.14 | asif | wcselby, tried that already. came across this message: |
| 21:38.16 | Naikrovek | [KS]: you have host set for the extension. if it uses dhcp set it to dynamic : host=dynamic |
| 21:38.18 | asif | "Please visit the front page of either www.asterisk.org or www.asterisknow.org to create a user account. Doing so will provide you with one set of login credentials for you to log into all three websites (Asterisk.org, AsteriskNOW.org and the Asterisk Forums at forums.digium.com). " |
| 21:38.19 | p3nguin | You don't want to use agents right now; just add your self as a static queue member for now. |
| 21:38.42 | asif | can't find any link to register at either of these sites... :| |
| 21:38.54 | [KS] | Naikrovek, I don't understand |
| 21:38.59 | wcselby | asif - http://www.asterisk.org/user/register |
| 21:39.01 | p3nguin | dustybin: It's very easy that way. |
| 21:39.20 | wcselby | it's on the bottom right under the "User Login" box...It says "Create new account" |
| 21:39.22 | asif | woof! thanks wcselby :) |
| 21:39.51 | wcselby | woof? lol np |
| 21:39.54 | Naikrovek | [KS]: in the config for that extension/device, set host=dynamic. you have it set to an IP that it's not using, I think... |
| 21:39.55 | asif | think my eyes need testing :) |
| 21:40.27 | dustybin | i created all the queue options under the [general] context |
| 21:40.42 | dustybin | inside extensions.conf, i have replaced dial with queue |
| 21:40.44 | dustybin | exten => s,2,Queue(general) |
| 21:40.49 | wcselby | dustybin - now create a new context for your queue |
| 21:40.59 | wcselby | you can't use general, I'm pretty sure |
| 21:41.21 | dustybin | wcselby: shall i rename the [general] context in queues.conf ? |
| 21:41.22 | TJNII | Yea, general is for general options, not for a specific queue |
| 21:41.26 | TJNII | No. |
| 21:41.35 | wcselby | i.e make a new context named [queue1] and then put queue specific options, along with your "member => IAX2/1010" line |
| 21:41.40 | dustybin | do i need to put a context inside queues? |
| 21:41.52 | p3nguin | dustybin: Just create a queue my some other name at the bottom of the queues.conf. |
| 21:42.03 | wcselby | then your extensions.conf line would be "exten => s,2,Queue(queue1) |
| 21:42.05 | wcselby | " |
| 21:42.06 | p3nguin | by some other name, rather |
| 21:42.10 | dustybin | how annoying, that means i need to copy and paste the whole thing |
| 21:42.16 | p3nguin | maybe |
| 21:42.25 | p3nguin | should have done it right from the get-go, though. :) |
| 21:42.31 | wcselby | dustybin - not really |
| 21:42.42 | dustybin | i just uncommented a load of lines on queues.conf |
| 21:42.57 | p3nguin | You should probably undo that. |
| 21:43.02 | dustybin | grrrr |
| 21:43.05 | p3nguin | The standard commented file is fine. |
| 21:43.14 | p3nguin | Just add your own queue context in the bottom. |
| 21:43.16 | p3nguin | That is all. |
| 21:43.29 | dustybin | that involves serious copy and pasting |
| 21:43.34 | p3nguin | Shouldn't. |
| 21:43.36 | TJNII | Now, if you are setting up queues make sure to put appropriate thought into the hold music to use. Personally, I'm a bug fan of this for queue music: http://tinyurl.com/nf24fs. Would not sound out of place in any corporate queue or elevator. |
| 21:43.44 | dustybin | p3nguin: how else will i know what to put? |
| 21:43.53 | TJNII | grins |
| 21:43.58 | dustybin | re-comments queues.conf |
| 21:44.03 | p3nguin | My entire queue context has a total of 11 lines. |
| 21:45.10 | p3nguin | I use piano and dulcimer music for MoH/queues. |
| 21:45.34 | TJNII | That could be really good or really bad. |
| 21:45.53 | p3nguin | Ripped them off a CD as wave files, then encoded with sox to the right bitrate. |
| 21:46.17 | TJNII | I also have a couple of lounge versions of "Stairway to Heaven" and Rick Astly's "Never gonna give you up" or whatever the real title is. |
| 21:46.26 | [TK]D-Fender | p3nguin: I like to start soft... something like Slayer or Pantera, maybe a little Voivod |
| 21:46.43 | TJNII | (Obviously, this is a non-professional setup.) |
| 21:46.48 | dustybin | ok queues re-commented |
| 21:46.52 | p3nguin | Then follow up with some Disturbed, Godsmack, Seather? |
| 21:47.23 | dustybin | the sample config is not user friendly |
| 21:47.27 | bmoraca | TJNII: why the hell do all of the album covers in that recommended albums section have pictures of women's hips? |
| 21:47.34 | p3nguin | dustybin: You can ignore most of it. |
| 21:47.45 | TJNII | bmoraca: I honestly don't know. Because they are Brazillian? |
| 21:47.51 | dustybin | the asterisk oreilly big shows me nothing at all |
| 21:47.55 | dustybin | *book |
| 21:47.59 | bmoraca | i wasn't complaining...i just thought it was weird |
| 21:48.03 | dustybin | google is full of old stuff |
| 21:48.06 | dustybin | im d00med. |
| 21:48.08 | p3nguin | dustybin: On that voipplanet article you found... |
| 21:48.34 | p3nguin | dustybin: Grab the queue1 context off it and stick it in the bottom of your queues.conf. |
| 21:48.53 | dustybin | i lost the link |
| 21:49.00 | dustybin | what about the queues.conf from this: http://voip-info.capatres.com/wiki/view/Asterisk+cmd+Queue.html |
| 21:49.01 | p3nguin | http://www.voipplanet.com/backgrounders/article.php/3633901 |
| 21:49.16 | dustybin | ok! |
| 21:49.18 | p3nguin | Just use the voipplanet context. |
| 21:49.39 | p3nguin | Then delete the line: announce = queue-queue1 |
| 21:49.43 | p3nguin | You probably don't want that. |
| 21:49.44 | [KS] | Call from '3003' to extension '1' rejected because extension not found. |
| 21:49.48 | [KS] | sorry, and this? |
| 21:50.20 | p3nguin | dustybin: Then comment out the three member lines that are there (save them for later when you decide to use agents). |
| 21:50.35 | dustybin | aye done that |
| 21:50.39 | dustybin | i put member => IAX2/1000 at the end |
| 21:50.42 | p3nguin | dustybin: Then add your member => IAX2/1010 line. |
| 21:50.46 | dustybin | yep |
| 21:50.51 | p3nguin | Save, exit. |
| 21:50.57 | p3nguin | reload asterisk. |
| 21:51.36 | dustybin | p3nguin: before i do that, i need to make sure my extensions.conf is correct |
| 21:51.46 | p3nguin | Ah. I thought you did that earlier. |
| 21:52.03 | p3nguin | You'll want any incoming call to be stuck into the queue almost right away. |
| 21:52.26 | p3nguin | Maybe play your standard greeting or whatever, then into the queue they go. |
| 21:53.15 | dustybin | p3nguin: i need to change this line: |
| 21:53.16 | wcselby | [KS] - check the context that 3003 starts in |
| 21:53.18 | dustybin | exten => s,2,Dial(IAX2/1000,10) |
| 21:53.34 | dustybin | exten => s,2,Queue(IAX2/1000) ? |
| 21:53.36 | dustybin | or |
| 21:53.37 | wcselby | dustybin - exten => s,2,Queue(queue1,10] |
| 21:53.45 | dustybin | thanks :D |
| 21:53.51 | p3nguin | or Queue(queue1,twh) |
| 21:54.04 | p3nguin | whatever options you want. |
| 21:54.06 | dustybin | right, now it makes sense |
| 21:54.13 | dustybin | thanks :D |
| 21:54.44 | p3nguin | So once the caller hits that line for the extension, he is sent into the queue, where your phone will ring. |
| 21:54.58 | p3nguin | You can either answer it or wait a while. It should keep ringing until you pick up. |
| 21:55.06 | dustybin | its time to test |
| 21:55.43 | wcselby | dustybin - make sure you use the right syntax, not sure I gave it to your properly |
| 21:55.55 | dustybin | dialplan reload |
| 21:56.08 | dustybin | queues reload.. |
| 21:56.13 | p3nguin | You'll need to reload the entire thing since you changed some other files. |
| 21:56.21 | p3nguin | Just "reload" is enough to do it all. |
| 21:56.27 | dustybin | ace |
| 21:56.45 | p3nguin | Depending on your verbose level, you might see it read all the files. |
| 21:57.25 | dustybin | [Sep 11 22:56:33] NOTICE[22132]: app_queue.c:5449 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action. == Parsing '/home/asterisk/asterisk/bin/asterisk/queues.conf': |
| 21:57.58 | p3nguin | shrug |
| 21:58.14 | p3nguin | I don't have any queuerules.conf, so that's not an issue for me. |
| 21:59.56 | dustybin | now i need to ring myself twice |
| 22:00.10 | p3nguin | simultaneously? |
| 22:00.17 | dustybin | to test the queue |
| 22:00.34 | p3nguin | Do you have the capability to do that? |
| 22:00.40 | dustybin | no lol |
| 22:00.47 | dustybin | i can ring in with my cellphone |
| 22:00.53 | dustybin | and i have a soft phone on my laptop |
| 22:01.01 | dustybin | i could do that |
| 22:01.10 | p3nguin | I could call you if you wanted. |
| 22:01.15 | dustybin | o_0 |
| 22:01.27 | dustybin | i will try now |
| 22:01.44 | p3nguin | Let me know if you need an extra caller. |
| 22:01.50 | dustybin | ok :D |
| 22:01.57 | TJNII | Yea, paste your phone number |
| 22:02.00 | p3nguin | heh |
| 22:02.03 | p3nguin | not here! |
| 22:02.03 | dustybin | O-O |
| 22:02.04 | TJNII | grins in a predatory manner |
| 22:02.37 | wcselby | 900-734-0976 |
| 22:02.45 | wcselby | if you want to call and chat or whatever |
| 22:02.55 | wcselby | grins in an *evil* way |
| 22:03.02 | TJNII | Heheh. For only $4.95/min, right? |
| 22:03.10 | dustybin | haha i forgot i am closed at the moment |
| 22:03.18 | dustybin | i better change the time |
| 22:06.36 | *** join/#asterisk blkry (n=chatzill@64.147.222.130) |
| 22:08.10 | dustybin | it worked!! your call is now next in line, please wait for the next representative! |
| 22:08.13 | dustybin | thanks people :D |
| 22:08.28 | p3nguin | :) |
| 22:11.09 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 22:13.29 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 22:27.19 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
| 22:34.18 | asif | Hey, has cdr_yada met it's end? |
| 22:34.33 | *** join/#asterisk blkry (n=chatzill@64.147.222.130) |
| 22:35.28 | *** join/#asterisk ross` (n=ross@67.52.47.70) |
| 22:35.33 | ross` | hello all |
| 22:35.39 | asif | hi ross! |
| 22:41.44 | *** join/#asterisk davidandgoliath (n=David@out.clearnet.com) |
| 22:46.53 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 22:50.35 | ethicx | hi everyone how do I turn debugging off in asterisk 1.4? |
| 22:51.52 | *** join/#asterisk ian6 (i=blank@slu.ms) |
| 22:51.54 | p3nguin | Look in asterisk.conf. |
| 22:52.24 | p3nguin | comment out the debug = line. |
| 22:52.37 | ian6 | is it possible to have call queues use a followme line for an agent? |
| 22:53.04 | [TK]D-Fender | ian6: if it hits the dialpla it will do whatever your dialplan tells it to |
| 22:53.06 | box2 | ethicx: sip set debug off |
| 22:53.14 | ethicx | never mind I just found it |
| 22:53.21 | ethicx | box2: Yup, thanks man. |
| 22:54.45 | dustybin | p3nguin: one last question, how can i put a timeout on the queue, so after a certain amount of time, the queue will exit and continue my dialplan, or jump to another context? |
| 22:55.15 | dustybin | ie. i want my caller to go in a queue, then after a certain amount of time, i want the caller to jump to my voicemail menu |
| 22:55.29 | [TK]D-Fender | duccontinue |
| 22:55.39 | dustybin | googles |
| 22:56.37 | dustybin | google shows nothing.. |
| 22:56.40 | dustybin | checks book |
| 22:57.29 | dustybin | nothing listed in the index for 'duccontinue' |
| 22:57.46 | dustybin | [FAIL] |
| 22:58.29 | p3nguin | exten => s,2,Queue(queue1|twh|||300) |
| 22:58.35 | p3nguin | maybe that |
| 22:58.45 | dustybin | 300 = the timeout? |
| 22:58.50 | dustybin | seconds |
| 22:59.43 | p3nguin | Yes, 5 minutes. |
| 22:59.57 | dustybin | thanks :D |
| 22:59.58 | dustybin | perfect |
| 23:00.10 | p3nguin | exten => s,3,Voicemail(u1010) |
| 23:00.15 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:01.28 | p3nguin | You might want to set that timeout to more like 70 seconds for testing. |
| 23:01.34 | [TK]D-Fender | dustybin: warning p3nguin's syntax is pre 1.6 |
| 23:01.44 | p3nguin | I use 1.4. Sorry. |
| 23:02.40 | p3nguin | exten => s,2,Queue(queue1,twh,,,300) |
| 23:02.43 | p3nguin | better? |
| 23:03.06 | ross` | hi, can i get someones opinion on how good the debian build of asterisk is |
| 23:03.19 | ross` | i.e. should i compile from source, or it installing through apt-get alright |
| 23:03.40 | p3nguin | You should probably use your package manager for everything. |
| 23:04.01 | ian6 | [TK]D-Fender: hmm. Yeah I guess that works, never done this before and the documentation wasn't super clear on it. Thanks. |
| 23:04.13 | L2Logic | 1 |
| 23:05.02 | dustybin | is this pre 1.6 ==> Queue(testq2|t|||45) ? |
| 23:05.08 | p3nguin | yes |
| 23:05.14 | p3nguin | Do you have 1.6? |
| 23:05.18 | dustybin | blimin heck, the whole of net is pre 1.6 |
| 23:05.21 | dustybin | yes |
| 23:05.45 | p3nguin | Change | to , |
| 23:05.53 | dustybin | aye ok |
| 23:05.54 | [TK]D-Fender | dustybin: and check the parm orders |
| 23:06.24 | p3nguin | 1.6 - Queue(queuename[,options[,URL[,announceoverride[,timeout[,AGI[,macro[,gosub[,rule]]]]]]]]) |
| 23:06.27 | ross` | p3nguin: do you have a good background to make that statement |
| 23:06.48 | ross` | im talking specifically about installing from the debian repos or installing from source |
| 23:08.52 | p3nguin | ross`: I know absolutely nothing about how they built it, but I bet they did it correctly. It's always best to use the package manager to install things, even if you build it yourself and then make it into a package. checkinstall can help you make a package if you insite on compiling from source. |
| 23:09.10 | p3nguin | insist, rather |
| 23:09.24 | davidandgoliath | curious, when a user is 'in' a queue, is there a button that allows them to leave a voicemail at any point in time |
| 23:09.27 | davidandgoliath | or can one be assigned? |
| 23:09.29 | ross` | p3nguin: you are totaly wrong in my opinion |
| 23:09.42 | p3nguin | ross`: Suit yourself. |
| 23:09.59 | dustybin | p3nguin: when will you upgrade to 1.6 ? |
| 23:10.01 | p3nguin | If you think they did it wrong, don't use it. |
| 23:10.22 | p3nguin | dustybin: If Arch ever gives me a build for 1.6, I might upgrade. |
| 23:10.46 | ross` | p3nguin: |
| 23:10.48 | ross` | are you serious |
| 23:10.52 | ross` | lol |
| 23:10.56 | p3nguin | about what? |
| 23:10.56 | dustybin | p3nguin: compile yourself! |
| 23:11.00 | davidandgoliath | whistles quietly. |
| 23:11.15 | ross` | so you let the AUR and ABS boss you around |
| 23:11.16 | ross` | dude |
| 23:11.24 | ross` | package repos are SLOW |
| 23:11.28 | ross` | they rarely have the latest/best version |
| 23:11.31 | ross` | lol |
| 23:11.35 | p3nguin | I guess you don't know how Arch does things. I would be compiling myself, regardless. |
| 23:11.36 | ross` | well arch is pretty good about that |
| 23:11.36 | ross` | actually |
| 23:11.53 | ross` | p3nguin: define your definition of how arch does things |
| 23:11.56 | davidandgoliath | Does anyone give a shit about how slow package 'repos' are? :p |
| 23:11.57 | p3nguin | I could even make my own pkgbuild for 1.6, but why should I? |
| 23:12.00 | davidandgoliath | speed it up, build your own. ;) |
| 23:12.04 | ross` | do you mean a yaourt script |
| 23:12.04 | ross` | ? |
| 23:12.06 | drmessano | Why even ask if you're going to argue, you dumb shit? |
| 23:12.34 | drmessano | I mean, "I can see your point, ross`, though you seem to have none" |
| 23:12.34 | p3nguin | drmessano: That's just how people are. |
| 23:12.57 | ross` | drmessano: i asked p3nguin a quetion |
| 23:13.09 | ross` | and he answered without any knowledge of the question |
| 23:13.11 | ross` | =/ |
| 23:13.23 | p3nguin | dustybin: Anyway, I don't see anything missing from 1.4, so I probably won't upgrade soon. |
| 23:13.40 | dustybin | p3nguin: the whole syntax has changed!! |
| 23:14.06 | ross` | dustybin: tahts a reason to stay with 1.4 in my opinion :p |
| 23:14.21 | p3nguin | ross`: My answer was based purely on the fact that you should always use your package manager. I even told you that if you INSIST on compiling it yourself, use checkinstall to make your compilation into a package. If you don't like that response, move on. |
| 23:15.02 | drmessano | ross`: The fac that you're silly enough to ask it in the first place is pretty interesting. If the last 10 builds in the repo were great, does this mean this one isn't a bomb? what about the last 10 being bombs, and this one being great? What sort of data can you expect to grep from this line of questioning? |
| 23:15.06 | ross` | p3nguin: im trying to find out from somoene who knows enough about the debian package to accurately compare it from compiling from source =/ |
| 23:15.11 | p3nguin | ross`: My opinion was that the build in the repo was probably done correctly. |
| 23:15.24 | ross` | arghhh |
| 23:15.27 | ross` | ahh |
| 23:15.30 | ross` | !! |
| 23:15.30 | dustybin | ross`: http://www.thinkdebian.org/archives/828 |
| 23:16.06 | dustybin | part 2 will be setting up asterisk configs :D |
| 23:16.06 | drmessano | My opinion is that no one can tell me this build in the repo is any reflection on the last 10 or the next 10, so why the hell does it matter? |
| 23:16.13 | drmessano | Try it.. if it doesnt work, dont use it |
| 23:18.58 | wcselby | hey ross` - google is your friend |
| 23:19.40 | p3nguin | dustybin: The fact that the syntax has changed doesn't really promote my upgrading. |
| 23:20.34 | p3nguin | Actually, I haven't really ran into a lot of things that 1.6 syntax doesn't work on in my current version. |
| 23:20.59 | p3nguin | I also do not have a very elaborate setup. |
| 23:21.28 | dustybin | aye ok |
| 23:23.21 | dustybin | the only thing i use the CLI for is reloading configs, surely it must be capable of doing more exciting stuff? |
| 23:23.42 | drmessano | Its a server, dude |
| 23:23.53 | p3nguin | I do all configuration on the cli, as well. |
| 23:23.58 | dustybin | aye ok |
| 23:24.05 | drmessano | It abides |
| 23:26.59 | _ShrikE | Shut the fuck up donny |
| 23:28.39 | dustybin | <PROTECTED> |
| 23:29.49 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:30.04 | drmessano | lol |
| 23:30.17 | drmessano | Fucking nihilists |
| 23:30.29 | _ShrikE | pissed on my carpet |
| 23:31.09 | jaytee | that rug really tied the room together |
| 23:31.51 | drmessano | heh |
| 23:32.11 | jaytee | nice marmot! |
| 23:32.15 | drmessano | I don't roll on Shabbos |
| 23:32.49 | drmessano | MARK IT ZERO |
| 23:40.27 | *** join/#asterisk ESCulapio__ (n=ESCulapi@148.Red-217-127-193.staticIP.rima-tde.net) |
| 23:40.38 | jaytee | Hell, I can get you a toe by 3 o'clock this afternoon... with nail polish. These fucking amateurs... |
| 23:40.57 | ESCulapio__ | quien habla espanol ? |
| 23:42.12 | *** join/#asterisk neurosys (n=vinix@c-71-196-19-254.hsd1.fl.comcast.net) |
| 23:43.21 | bmoraca | buzz killington |
| 23:47.17 | dlynes | ESCulapio__: /join #asterisk-es |
| 23:47.32 | ESCulapio__ | no hay nadie |
| 23:47.36 | ESCulapio__ | nadie responde |
| 23:47.40 | dlynes | ah |
| 23:48.00 | dlynes | nues habla anglais |
| 23:48.01 | p3nguin | Can you try to ask your question in ingles? |
| 23:49.00 | dlynes | ESCulapio__: comprende ingles? |
| 23:49.06 | ESCulapio__ | yes |
| 23:49.15 | ESCulapio__ | pero no lo escribo |
| 23:50.43 | dlynes | ESCulapio__: so what's the problem then? |
| 23:51.27 | ESCulapio__ | quiero poner un telefono en asterisk que cuando levante pueda hablar sin marcar numero |
| 23:51.39 | dlynes | ESCulapio__: in English? |
| 23:52.00 | wcselby | and that's all folks |
| 23:52.22 | p3nguin | He said he can't scribe in english. |
| 23:52.28 | dlynes | p3nguin: oh |
| 23:52.34 | p3nguin | understand it, but not write it. |
| 23:52.54 | dlynes | p3nguin: heh...if he could write french, maybe I could help him :) |
| 23:53.04 | *** join/#asterisk manxpower (n=EWieling@50.sub-70-222-21.myvzw.com) |
| 23:53.05 | dlynes | p3nguin: but I don't know spanish |
| 23:53.12 | ESCulapio__ | lift a telephone and automatically dial |
| 23:53.12 | manxpower | ~answers |
| 23:53.13 | infobot | somebody said answers was Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 23:53.14 | p3nguin | Translation tools indicate that he's wanting to be able to pick up a phone and it auto-dials. |
| 23:53.23 | manxpower | ESCulapio__: analog or SIP? |
| 23:53.28 | ESCulapio__ | sip |
| 23:53.41 | dlynes | ESCulapio__: which particular sip phone? |
| 23:53.42 | manxpower | ESCulapio__: that is the function of the SIP Phone, not Asterisk. |
| 23:53.58 | manxpower | Asterisk controls that function on analog, but not SIP. |
| 23:54.19 | manxpower | ESCulapio__: the common term are "PLAR", "ARD", and "Bat Phone" |
| 23:54.19 | dlynes | manxpower: well...depends on whether it's a sip device providing analog or a zaptel channel |
| 23:54.34 | manxpower | dlynes: no SIP devices provide zap channels |
| 23:54.52 | dlynes | manxpower: but they do provide analog signals to analog phones |
| 23:55.03 | dlynes | manxpower: asterisk just doesn't consider it analog |
| 23:55.30 | ESCulapio__ | as are made with Analog |
| 23:55.36 | neurosys | I have a couple of sangoma a400s. There's some subtle steady line noise on some of the channels but only a couple. how would i troubleshoot this? Its not connected to the PSTN yet. Its just the breakout box. and it starts when the img. loads @ wanpipe loadup. |
| 23:56.13 | manxpower | neurosys: contact Sangoma. Be sure to mention there are NOT PSTN lines connected at this point |
| 23:57.00 | ESCulapio__ | dlynes, the phone, polycom and Grandstream |
| 23:57.26 | neurosys | manxpower: ok. emailed their techdesk |
| 23:59.15 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |