| 00:03.04 | *** join/#asterisk jks (jks@193.189.93.254) |
| 00:07.15 | *** part/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 00:07.16 | *** join/#asterisk bsdmail (~dig@67.228.177.47) |
| 00:09.11 | bsdmail | what is the best tts tool and free, with portuguese support? |
| 00:11.28 | hardwire | hmmm.. I can't seem to visualize how I can be a t.38 endpoint and then relay. |
| 00:11.29 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 00:13.48 | hardwire | specifically from sip.conf (1.6.2) |
| 00:14.11 | hardwire | anybody have experience accepting a fax over TDM then originating a t.38 call over SIP? |
| 00:28.01 | *** join/#asterisk nentis (~nentis@173-11-4-145-oregon.hfc.comcastbusiness.net) |
| 00:28.46 | nentis | can anyone recommend a iax provider? Business use, need to port two numbers and have 4 in/out lines. Using trixbox. |
| 00:29.02 | nentis | hm. Voicepulse has a trixbox module it appears. |
| 00:30.00 | *** join/#asterisk ruied (~ruied@bl10-126-116.dsl.telepac.pt) |
| 00:32.03 | *** join/#asterisk nix8n82 (~AndChat@63.162.27.14) |
| 00:32.23 | *** join/#asterisk crochat (~crochat@158-89.60-188.cust.bluewin.ch) |
| 00:39.43 | *** join/#asterisk ttwhy (~tekkno@p4FECFC5A.dip.t-dialin.net) |
| 00:40.26 | Katty | ryan just called me a mental patient |
| 00:40.29 | Katty | ;O |
| 00:40.31 | Katty | ;P |
| 00:41.10 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 00:42.12 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 00:42.51 | eppigy | Katty: PUNCH HIM IN THE FACE |
| 00:43.08 | *** part/#asterisk nny (~scott@64.203.239.83) |
| 00:44.48 | Katty | eppigy: it was a joke. |
| 00:45.13 | eppigy | oh |
| 00:45.21 | eppigy | well then restrain yourself |
| 00:45.24 | eppigy | are you crazy |
| 00:45.46 | Katty | yesh |
| 00:45.53 | *** join/#asterisk nix8n82 (~AndChat@63.162.27.14) |
| 00:46.39 | eppigy | me2 |
| 00:47.36 | *** join/#asterisk ex-parrot (~ex-parrot@2401:f000:3:0:21a:4dff:fe0d:7e59) |
| 00:47.54 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 00:48.05 | ex-parrot | hi all.... I've followed the instructions in the novavox international settings PDF but I am not getting the right output in dmesg.... I don't get the "wcfxo: DAA mode is 'AUST' " line |
| 00:48.13 | ex-parrot | can anyone tell me how important this is? |
| 00:50.49 | raden_work | OMFG |
| 00:51.18 | raden_work | You do not appear to have the sources for the 2.6.31.8-0.1-default kernel installed. |
| 00:51.43 | raden_work | Status: out-of-date (version 2.6.31.8-0.1.1 installed) |
| 00:51.52 | raden_work | TTSSUSA-1000:/home/tss/SOURCE/dahdi-linux-complete-2.2.1+2.2.1 # uname -r |
| 00:51.53 | raden_work | 2.6.31.8-0.1-default |
| 00:52.09 | raden_work | what am i doing wrong ? DADHI HATE ME ? |
| 00:52.20 | raden_work | [TK]D-Fender, ? |
| 00:52.23 | raden_work | eppigy, ? |
| 00:52.26 | raden_work | bmoraca_work, ? |
| 00:52.43 | raden_work | Katty, ? |
| 00:53.19 | *** join/#asterisk rossand (~aross@dhcp-233-179.tb-classrooms.carleton.ca) |
| 00:54.38 | Katty | ex-parrot: ohaider. |
| 00:55.26 | ex-parrot | Katty: not sure I follow you there? |
| 00:55.44 | Katty | ex-parrot: o hai der |
| 00:55.48 | ex-parrot | aha! :) |
| 01:00.41 | thehar | apparently asteris doesn't like to have strace attached to it's process |
| 01:01.03 | thehar | asterisk* |
| 01:01.11 | Katty | hi thehar |
| 01:01.19 | *** join/#asterisk cweagans (~cweagans@71-33-110-201.bois.qwest.net) |
| 01:01.38 | cweagans | does anybody feel like giving me a hand with an unruly Cisco 7940? |
| 01:01.44 | Katty | sure |
| 01:01.48 | Katty | gimmie a sledgehammer |
| 01:01.56 | cweagans | my thoughts exactly =P |
| 01:02.09 | Katty | (=< |
| 01:02.18 | cweagans | I think I've got the config correct, but it says UNPROVISIONED on the phone, and I'm really really new at this, so I'm not sure how to debug further |
| 01:02.47 | cweagans | I can share my screen or whatever if that'd make it easier. |
| 01:03.07 | cweagans | and also, if it helps, there's a six pack on me up for grabs |
| 01:03.39 | raden_work | anyone have any idea why dahdi will not installl ??????????????? |
| 01:03.43 | *** part/#asterisk ex-parrot (~ex-parrot@2401:f000:3:0:21a:4dff:fe0d:7e59) |
| 01:04.08 | cweagans | raden_work: my guess? You have an extra question mark somewhere.. |
| 01:04.14 | cweagans | =P |
| 01:04.16 | raden_work | lmao |
| 01:04.18 | *** join/#asterisk Kumbang (~kumbang@125.163.83.153) |
| 01:07.34 | cweagans | free beer (or drinks of your choice) for somebody that can help me with my Asterisk PBX + a Cisco 7940G. I can't figure out how to configure the phone via TFTP and I'm really new to Asterisk and am not sure how to further debug. <--accidentally posted that in #drupal |
| 01:07.36 | cweagans | >.< |
| 01:08.41 | raden_work | cweagans, i wish i had the time to help u but im in my own dilema right now :( |
| 01:08.57 | cweagans | raden_work: what distro? |
| 01:09.11 | raden_work | opensuse 11.2 |
| 01:09.17 | raden_work | everything from console |
| 01:09.25 | cweagans | what's the error? |
| 01:09.34 | raden_work | You do not appear to have the sources for the 2.6.31.8-0.1-default kernel installed. |
| 01:09.36 | raden_work | ^^^^ dahdi |
| 01:09.47 | raden_work | Status: out-of-date (version 2.6.31.8-0.1.1 installed) |
| 01:09.59 | raden_work | <raden_work> TTSSUSA-1000:/home/tss/SOURCE/dahdi-linux-complete-2.2.1+2.2.1 # uname -r |
| 01:09.59 | raden_work | <raden_work> 2.6.31.8-0.1-default |
| 01:10.28 | cweagans | er. well install the kernel sources (you might need the kernel headers too...not sure. i needed it on ubuntu |
| 01:10.40 | cweagans | ) |
| 01:11.47 | raden_work | kernel headers ? |
| 01:11.59 | raden_work | kernel source is installed |
| 01:12.05 | raden_work | <PROTECTED> |
| 01:12.05 | cweagans | yeah, in ubuntu there's a package called kernel-headers |
| 01:12.10 | raden_work | lemee look |
| 01:12.28 | raden_work | im in wisconsin servers in colorado |
| 01:13.15 | raden_work | hope |
| 01:13.17 | raden_work | nope |
| 01:14.07 | raden_work | linux-kernel-headers |
| 01:14.13 | raden_work | there not a version that old to choose :( |
| 01:14.31 | raden_work | 2.6.31-3.4 |
| 01:16.16 | *** join/#asterisk RypPn (~TuMbL@rosscom.co.uk) |
| 01:18.07 | *** join/#asterisk creativx (~creadurex@197.82-134-19.bkkb.no) |
| 01:20.21 | raden_work | well..... |
| 01:20.30 | cweagans | hmm...no ideea |
| 01:20.33 | cweagans | idea* |
| 01:21.22 | raden_work | why doesnt all this shit just come in one package and just disable what a person does not need ? |
| 01:21.42 | raden_work | only thing i might need dahdi for is meet me |
| 01:21.49 | cweagans | heh, because that would make it easy ;) |
| 01:24.06 | *** join/#asterisk pentanol (~Unknown@91.195.60.231) |
| 01:26.05 | raden_work | cweagans, updating kernel to newer version |
| 01:26.12 | raden_work | what can i do for you while that compiling |
| 01:27.28 | cweagans | So, I've got this Cisco 7940 and I can't get it to connect to Asterisk. The SIP firmware is already loaded, but the phone isn't grabbing the settings from the config files on my TFTP server |
| 01:30.23 | cweagans | in addition, I"m really new to Asterisk (and phone systems in general), so I'm not sure how to debug further |
| 01:37.49 | raden_work | cweagans, why will it not connect to asterisk ? |
| 01:38.25 | cweagans | raden_work: I don't know. It just says UNPROVISIONED on the phone and it doesn't show up when I run 'sip show peers' in the asterisk cli |
| 01:39.03 | raden_work | let me see you sip.conf |
| 01:39.44 | cweagans | raden_work: http://pastebin.com/m7baa41b |
| 01:40.07 | raden_work | the phone on internal network ? |
| 01:40.24 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 01:40.41 | cweagans | yeah |
| 01:41.00 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 01:41.09 | raden_work | try something more simple to start to get things working |
| 01:41.50 | cweagans | I'm only trying with the first phone |
| 01:41.57 | cweagans | extension 200 |
| 01:42.03 | cweagans | that's the only phone plugged in |
| 01:42.35 | cvnet | in your dialplan can you assing values to variables? |
| 01:42.50 | cvnet | lets say OriginalCid = $extend |
| 01:43.17 | raden_work | http://pastebin.com/m6250bfb6 |
| 01:43.45 | raden_work | OMFG my server did not reboot :( |
| 01:45.51 | raden_work | wheeeew came online |
| 01:45.55 | raden_work | that was a long 6 min boot |
| 01:46.56 | *** part/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 01:48.05 | cvnet | $extend holds the value of the did, what holds the callerID of the caller? |
| 01:48.48 | raden_work | cweagans, ????? |
| 01:49.02 | raden_work | OMFG this is ridiculous i just want to install dahdi |
| 01:49.14 | cweagans | lol |
| 01:49.15 | *** join/#asterisk [netman] (~netman@40.Red-88-17-244.dynamicIP.rima-tde.net) |
| 01:49.27 | cweagans | ubuntu... i just did sudo apt-get install asterisk and it worked ;) |
| 01:49.29 | raden_work | ive never had anything from source be this difficult |
| 01:49.36 | cweagans | something to be said about ubuntu for stuff like this =P |
| 01:49.58 | raden_work | cweagans, ubuntu will be going on next 2 servers when the go out |
| 01:50.02 | raden_work | in about 9 months |
| 01:50.15 | raden_work | i know opensuse very well was not going to switch on a time crunch |
| 01:50.22 | *** join/#asterisk girlny (~girlny@CPE00195b4be142-CM001a668ec076.cpe.net.cable.rogers.com) |
| 01:51.23 | girlny | can someone tell me if these warnings are normal ?? http://pastebin.ca/1794155 thanks in advancee |
| 01:52.39 | brettnem | girlny: those are normal for freepbx/trix |
| 01:53.15 | raden_work | You do not appear to have the sources for the 2.6.31.12-0.1-default kernel installed. |
| 01:53.23 | girlny | ok thanks allot brettnem |
| 01:53.28 | raden_work | Version: 2.6.31.12-0.1.1 |
| 01:53.28 | raden_work | Arch: noarch |
| 01:53.28 | raden_work | Vendor: openSUSE |
| 01:53.28 | raden_work | Installed: Yes |
| 01:54.33 | cvnet | which variable holds callers CallerID ? |
| 01:56.00 | cvnet | ${CALLERID(num)} THANKS FOR NOTHING |
| 01:56.37 | ManxPower-work | cvnet: all that info is in asterisk.pdf and channelvariables.tex in your Asterisk doc/ directory |
| 01:57.00 | girlny | brettnem what about this http://pastebin.com/m91c7e9d |
| 01:57.47 | cvnet | im reading but its soo confusing |
| 01:58.55 | ManxPower-work | cvnet: read the pdf version |
| 01:59.02 | *** join/#asterisk Faithful (~Faithful@ns.linuxterminal.com) |
| 01:59.19 | raden_work | anyone ? |
| 01:59.27 | ManxPower-work | Bitch at Digium if you can't read it, they are the ones that stopped including the .txt version. |
| 02:00.04 | carrar | Digium gets paid by Adobe |
| 02:00.34 | kuku | I have a queue using ael, but when the queue never times out ( meaning if none of the agents pickup the call ) - it just stays in queue. |
| 02:04.15 | *** join/#asterisk ketema (~ketema@2001:470:5:138:217:f2ff:fe05:1e70) |
| 02:05.24 | *** join/#asterisk CoderForLife (~Miranda@cpe-174-101-155-51.cinci.res.rr.com) |
| 02:08.42 | cweagans | anybody? free beer to anybody that can help me get this phone running |
| 02:08.46 | cweagans | Cisco 7940G |
| 02:08.54 | cweagans | with asterisk, that is |
| 02:08.59 | cweagans | SIP firmware is already loaded. |
| 02:09.08 | carrar | HOW MUCH FREE BEER |
| 02:09.17 | girlny | lol |
| 02:09.18 | cweagans | I dunno? 6 pack? |
| 02:09.27 | cweagans | 12 pack? |
| 02:09.35 | cweagans | the better question is this: |
| 02:09.41 | cweagans | what will it cost me to get this running? |
| 02:10.06 | carrar | What have you done so far? |
| 02:10.42 | cweagans | TFTP server is working, SIP firmware is loaded, the config files are in place (I think...not sure on this one) |
| 02:11.02 | cweagans | the phone says UNPROVISIONED and 'sip show peers' doesn't show any active connections |
| 02:11.24 | cweagans | can share screen if needed |
| 02:11.26 | carrar | binpast your sip.conf and cisco config cnf |
| 02:11.31 | carrar | binpaste |
| 02:11.39 | cweagans | carrar: which cisco configs? |
| 02:11.53 | carrar | SIP{MAC}.cnf |
| 02:11.59 | cweagans | carrar: kk |
| 02:12.01 | girlny | can someone check this this for me is this normal http://pastebin.com/md84e0ce |
| 02:12.11 | carrar | and possible SIPDefault.cnf |
| 02:12.18 | carrar | also |
| 02:12.23 | carrar | Did you reset your phone? |
| 02:12.26 | carrar | might try that first |
| 02:12.33 | cweagans | 'reset'? |
| 02:12.42 | cweagans | power off and back on work? |
| 02:12.44 | carrar | do the tftp log files show it asking for the files? |
| 02:12.49 | cweagans | yes |
| 02:13.00 | cweagans | I updated the phone to the sip firmware from this tftp server |
| 02:13.02 | carrar | hold # while powering on |
| 02:13.12 | carrar | 123456789*0# |
| 02:13.13 | carrar | 2 |
| 02:13.35 | carrar | ok then it "should be" ok |
| 02:13.51 | cweagans | carrar: ok, I'll try the reset. In the meantime: http://pastebin.com/m11927a9f |
| 02:14.02 | *** join/#asterisk titter (~titter@c-76-101-240-142.hsd1.fl.comcast.net) |
| 02:14.41 | carrar | make your display name short name and name all the same |
| 02:14.43 | carrar | "200" |
| 02:15.04 | carrar | Quote your proxy IP |
| 02:15.07 | carrar | "1.1.1.1" |
| 02:15.09 | cweagans | in the SIP[mac]? |
| 02:15.13 | carrar | yeah |
| 02:15.37 | *** join/#asterisk Kumbang (~kumbang@125.163.83.153) |
| 02:15.54 | carrar | put name as" "200" |
| 02:15.57 | spenguin[w0rk] | girlny: is mysql running? |
| 02:15.58 | carrar | put name as: "200" |
| 02:16.23 | carrar | remove quotes from the password |
| 02:16.25 | spenguin[w0rk] | or what about that tpost |
| 02:17.30 | carrar | http://pastebin.com/m3b217019 |
| 02:17.37 | carrar | try that |
| 02:17.44 | carrar | err |
| 02:17.45 | *** join/#asterisk TJNII (~TJNII@207.189.199.62) |
| 02:17.48 | carrar | hang on |
| 02:18.07 | carrar | http://pastebin.com/m10d742ce |
| 02:18.43 | carrar | didn't see your sip.conf |
| 02:20.05 | cweagans | carrar: sip.conf: http://pastebin.com/m52be8169 |
| 02:20.55 | *** join/#asterisk Arsenick (~y@modemcable022.82-21-96.mc.videotron.ca) |
| 02:21.24 | carrar | try the cnf changes |
| 02:22.20 | carrar | I would change: callerid=Darren Willey <200> |
| 02:22.34 | *** join/#asterisk girlny (~girlny@CPE00195b4be142-CM001a668ec076.cpe.net.cable.rogers.com) |
| 02:22.49 | girlny | is this normal or important [Feb 11 20:35:18] ERROR[2058] res_config_ldap.c: No directory URL or host found. |
| 02:23.10 | carrar | cweagans, did you reboot with http://pastebin.com/m3b217019 |
| 02:23.57 | carrar | err I mean http://pastebin.com/m10d742ce |
| 02:23.59 | carrar | sorry |
| 02:25.08 | raden_work | OK now that im back on my track to success |
| 02:25.13 | raden_work | that was ridiculous |
| 02:26.39 | carrar | cweagans, might add: proxy_register: 1 |
| 02:26.50 | carrar | nm you haev that |
| 02:32.17 | raden_work | does duhndi need to be installed before asterisk ? |
| 02:33.18 | cweagans | carrar: the config changes worked. I'll have to keep playing around with it |
| 02:33.22 | cweagans | but I need to go grab some food |
| 02:33.25 | cweagans | bbl :0 |
| 02:33.27 | cweagans | :) * |
| 02:33.28 | carrar | beer please |
| 02:33.32 | cweagans | yea |
| 02:33.34 | cweagans | you gonna be around? |
| 02:33.40 | carrar | possibly |
| 02:33.42 | cweagans | like 30-40 minutes? |
| 02:33.48 | cweagans | if not, email me |
| 02:33.52 | cweagans | my nick @gmail.com |
| 02:33.57 | carrar | jsut pay it forward :) |
| 02:34.37 | cweagans | oh, c'mon. I owe ya some beer |
| 02:34.37 | carrar | no you don't |
| 02:34.40 | carrar | unless you live in Seattle |
| 02:35.02 | cweagans | well, I don't. But I have friends that do. And also, I could just like..paypal you some money =P |
| 02:35.20 | carrar | hehe, just pay it forward is good |
| 02:35.26 | cweagans | alright man, if you're sure :) |
| 02:35.29 | cweagans | thanks for your help! |
| 02:35.35 | carrar | np |
| 02:35.41 | cweagans | :) |
| 02:37.12 | girlny | is thiss error normal [Feb 11 20:35:18] ERROR[2058] res_config_ldap.c: No directory URL or host found. |
| 02:37.14 | prometheanfire | is CRC-CCITT still needed in 1.6? |
| 02:38.55 | carrar | Are you trying to use LDAP? |
| 02:41.00 | girlny | whats that |
| 02:41.02 | girlny | ? |
| 02:41.13 | girlny | i only use sip |
| 02:41.26 | *** join/#asterisk rue_house (~rue@24.207.119.38) |
| 02:41.28 | prometheanfire | then you don't use it |
| 02:41.35 | prometheanfire | :F |
| 02:41.37 | prometheanfire | :D |
| 02:42.13 | rue_house | how can I test the audio levels from voip phones, the 1mw comes out loud enough... |
| 02:42.55 | carrar | You could add: noload => res_config_ldap.so to your modules.conf |
| 02:43.30 | girlny | perfect , so sure i dont need it ?? |
| 02:43.43 | carrar | if you are using ldap you need it |
| 02:43.47 | girlny | i only use sip protocol no iax .. |
| 02:43.55 | carrar | sip != ldap |
| 02:44.07 | raden_work | anyone use dundi ? |
| 02:45.07 | girlny | lol u confususe me now lol im a rokie on this i only use sip i have no idea whats ldap do i need it ??? |
| 02:47.28 | girlny | does ldap complement sip ?? |
| 02:48.18 | Katty | cloudy with a chance of meatballs was CUTE :> |
| 02:48.55 | carrar | girlny, your sip peers might use ldap |
| 02:49.09 | carrar | to authenitcate |
| 02:49.14 | carrar | but you would know this |
| 02:49.18 | carrar | if you set it up |
| 02:49.28 | girlny | is it very commun ? |
| 02:49.41 | carrar | in a ldap enviroment it probably is |
| 02:49.55 | Katty | hugs on carrar |
| 02:50.00 | carrar | but it's not the default |
| 02:50.19 | carrar | feeds Katty more Makers Mark |
| 02:50.32 | girlny | i only use the typical cisco .. and dlink adapter and soft phone in my cellphone |
| 02:50.34 | carrar | FREE HUGS!! |
| 02:50.39 | carrar | hugs katty back |
| 02:50.43 | Katty | FREE HUGS!!!! |
| 02:50.54 | carrar | AUTHENTICATION ACCEPT!! |
| 02:50.57 | Katty | eppigy: FREE HUGS |
| 02:51.02 | Katty | eppigy: GET THEM WHILE THEY"RE FRESH |
| 02:51.05 | Katty | hugs eppigy |
| 02:51.24 | carrar | girlny, probably safe to say you don't use LDAP |
| 02:51.40 | Katty | hugs jblack |
| 02:51.42 | Katty | hugs jaytee |
| 02:52.38 | girlny | ok , if u want to conect a sip softphone in my cell using tcp instead of udp will i need that module |
| 02:52.55 | Katty | ha |
| 02:52.59 | Katty | sip clients on cellphones |
| 02:52.59 | girlny | becase i know i cant conect using tcp for some reason |
| 02:53.10 | girlny | ya i only pay dat a |
| 02:53.25 | Katty | good luck with that. it competes with the cellphone companies |
| 02:53.35 | girlny | is the future ,lol |
| 02:53.42 | girlny | it work fine |
| 02:53.57 | girlny | some carries block port 5060 |
| 02:54.43 | carrar | can always change the port |
| 02:54.52 | carrar | if the client supports it |
| 02:55.15 | *** join/#asterisk OrNix (~ornix@l151-249-47.static.cn.ru) |
| 02:55.40 | girlny | ya it does but some how i cant conect using tcp to my box i can to other boxes but not mine port is open on the firewall |
| 03:02.37 | girlny | lol wikipidia told me i definaly dont need it thanks carrar thanks wikipidia |
| 03:03.18 | girlny | i will bring u a beer when i return to Seattle |
| 03:03.43 | Katty | hmm. |
| 03:03.50 | Katty | stomach says hungry. |
| 03:04.04 | Katty | or does it ^_- |
| 03:04.07 | *** join/#asterisk mtipping (~chatzilla@cpe-24-31-134-176.maine.res.rr.com) |
| 03:04.14 | Katty | carrar: you have a hard time between MUNCHIES and hungry? |
| 03:05.39 | girlny | u should try the master cleanse will teach u the differences real good |
| 03:09.09 | girlny | is there any setting in asterisk that avoids tcp setup ? |
| 03:09.43 | Katty | try the what ^_- |
| 03:12.18 | raden_work | yea !!!! asterisk installed with dahdi and dundi and everything works |
| 03:12.27 | raden_work | now i can do it in like 5 min next time :) |
| 03:12.29 | Katty | wooooooooooooooooooooo |
| 03:12.53 | raden_work | dahdi is not a user friendly install need to write howto |
| 03:12.58 | girlny | congrantsss |
| 03:13.08 | *** join/#asterisk Xetrov` (~xetrov@unaffiliated/xetrov/x-827361) |
| 03:13.17 | Katty | ladies and gentlemen of congrants. |
| 03:13.20 | Katty | mister president |
| 03:13.21 | raden_work | for some reason that was the most stressful asterisk install ever |
| 03:13.22 | Katty | madam speaker. |
| 03:13.26 | raden_work | lol |
| 03:13.29 | Katty | I HAVE A DREAM |
| 03:13.34 | Katty | that ONE DAY |
| 03:13.37 | raden_work | now to get realtime working with mysql |
| 03:13.51 | Katty | dundi will be accepted |
| 03:13.58 | Katty | and not just accepted |
| 03:14.00 | Katty | DOCUMENTED |
| 03:14.12 | raden_work | ill be working on that as well |
| 03:14.12 | Katty | <PROTECTED> |
| 03:14.26 | raden_work | we have 232 people on our signup list like freaking wildfire :( |
| 03:14.44 | Katty | why is that bad |
| 03:14.52 | raden_work | I dont even have the servers in place |
| 03:14.57 | raden_work | i need time to test everything |
| 03:15.06 | Katty | and you won't get servers and time? |
| 03:15.12 | Katty | or are you just feelin the stress |
| 03:15.12 | raden_work | billboard wraps are printed and waiting to go up |
| 03:15.46 | Katty | i would like to make a recommendation then |
| 03:15.46 | raden_work | Katty, Im in electronics this is a side thing the company started they are starting to put much more focus here but i have many other responsibilities so time is always a factor |
| 03:16.19 | Katty | SMILE!!!! not only does it relieve stress, it reduces your blood pressure too :> |
| 03:16.20 | raden_work | plus were down to 2 employees and 4 contractors from 6 and 14 last year |
| 03:16.29 | raden_work | sooo yea race to make money race to get everything done |
| 03:16.41 | raden_work | have a seminar here next week with over 200 business owners coming |
| 03:16.47 | Katty | stop typing |
| 03:16.48 | Katty | SMILE |
| 03:16.50 | Katty | ^- |
| 03:16.53 | Katty | are you smiling? |
| 03:16.54 | Katty | DO EET |
| 03:17.01 | raden_work | NOOO :P |
| 03:17.07 | raden_work | u smile maybe i will :) |
| 03:17.15 | Katty | i am smiling! |
| 03:17.22 | Katty | and eating popcorn |
| 03:17.25 | Katty | which i'm sure makes quite a face |
| 03:17.33 | raden_work | (o= |
| 03:17.41 | raden_work | LMAO |
| 03:18.41 | Katty | did you know smiles are contagious? |
| 03:18.49 | Katty | and few things are contagious. yawning is another one. |
| 03:18.50 | raden_work | some :) |
| 03:18.54 | raden_work | :D |
| 03:19.10 | Katty | real smiles are contagious |
| 03:19.14 | raden_work | Katty, you ever mess with asterisk realtime database ? |
| 03:19.18 | Katty | nope |
| 03:19.28 | Katty | i tinker with realtime smiling tho |
| 03:20.13 | raden_work | =)))))) |
| 03:20.21 | raden_work | 8-) |
| 03:21.09 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 03:21.26 | raden_work | I need to get home have a good night katty |
| 03:21.36 | raden_work | hugs Katty =) |
| 03:21.43 | Katty | byebye |
| 03:22.16 | raden_work | night |
| 03:23.06 | *** join/#asterisk Kumbang (~kumbang@125.163.83.153) |
| 03:23.28 | *** join/#asterisk Caplain (shayne@caplain.loves.boys.fbi.gov.silverelitez.org) |
| 03:24.24 | KavanS | does anyone know the last version of asterisk that had zaptel support? |
| 03:28.54 | raden_work | 1.4 something |
| 03:30.18 | girlny | con someone help set up asterik sip to work over tcp |
| 03:30.26 | *** join/#asterisk Kumbang (~kumbang@rusnas.paume.itb.ac.id) |
| 03:30.30 | KavanS | Raden, hehe yeah...I'm using 1.4.18.1 now |
| 03:30.33 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 03:30.36 | KavanS | I tried 1.4.29 earlier today and no chan_zap support |
| 03:30.42 | KavanS | I tried googling, but I must be using wrong terms |
| 03:31.14 | KavanS | was hoping to find the latest and greatest that still has zap support...I use iaxmodem/hylafax and don't want to risk as much |
| 03:39.22 | *** join/#asterisk toortog (nesm6@38.111.17.107) |
| 03:39.32 | toortog | hello |
| 03:42.57 | ManxPower-work | KavanS: the chan_dahdi.so acts like, uses the same config files, and has the same cli commands as chan_zap.so did. |
| 03:43.29 | ManxPower-work | Only the actual name of the channel driver changed. |
| 03:46.51 | raden_work | KavanS, whats wrong with dahdi ? |
| 03:48.09 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 03:48.45 | cweagans|away | anybody have an image I can look at as an example for replacing the phone image on Cisco 7940s? |
| 03:49.21 | p3nguin | Uh, what? |
| 03:49.41 | raden_work | p3nguin, heya |
| 03:49.45 | p3nguin | hi |
| 03:49.50 | raden_work | how it going ? |
| 03:50.09 | cweagans|away | for the Cisco 7940s, you have to create an image to a certain spec. dimensions, color depth, etc. I'm trying to make one, but it'd help to see an example of one that was already done. |
| 03:50.30 | raden_work | cweagans|away, rtfm ? |
| 03:51.13 | cweagans | that's the other problem. I can't seem to find that particular manual to read >.< |
| 03:51.18 | raden_work | cweagans, i will presume you have the cisco SDK ? |
| 03:51.23 | cweagans | heh. no. |
| 03:51.45 | raden_work | then your SOL |
| 03:51.45 | cweagans | do I need that? |
| 03:51.49 | raden_work | yeah |
| 03:51.53 | p3nguin | cweagans|away: That's simple. http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79xx |
| 03:52.17 | carrar | cweagans , http://www.osburn.com/asterisk.bmp |
| 03:52.35 | p3nguin | cweagans: Windows Bitmap form (*.BMP) with 256 colors and 90 x 56 pixels |
| 03:52.54 | p3nguin | cweagans: Only two colors are displayed, black or white. The image must be saved in greyscale format. In GIMP it is Image->Mode->Greyscale. |
| 03:52.58 | cweagans | carrar: you sure I don't owe you some beer? =P |
| 03:53.03 | cweagans | thanks p3nguin :) |
| 03:53.04 | carrar | heh |
| 03:53.09 | carrar | no, not really sure |
| 03:53.18 | raden_work | lol |
| 03:53.35 | p3nguin | cweagans: The image can only have black, white, and two shades of gray. |
| 03:54.00 | raden_work | they really went all out LOL |
| 03:54.08 | p3nguin | yeah |
| 03:54.17 | carrar | even the 7970 lack a lot as far as images as well |
| 03:54.33 | carrar | I use a 7941 on my desk |
| 03:54.49 | carrar | it has much better resolution then the 7940 |
| 03:55.14 | raden_work | later all im out |
| 03:56.42 | p3nguin | carrar: http://www.wsu.edu/~brians/errors/than.html |
| 03:57.24 | carrar | yeah I know the diff |
| 03:57.31 | carrar | sometimes I just don't care |
| 03:57.52 | p3nguin | Someone that knows wouldn't make such a mistake. |
| 03:58.16 | carrar | spelling is not formost on my mind |
| 03:58.48 | p3nguin | Another typical response of someone that doesn't know which word to use. |
| 03:59.01 | carrar | I also really don't care |
| 03:59.11 | carrar | but carry on if it makes you happy |
| 04:00.44 | ChannelZ | he doesn't get it makes him look like a big cock |
| 04:01.01 | p3nguin | Pfft, you think I care about my appearance here? |
| 04:01.22 | ChannelZ | No, apparently you care deeply for everone else's though |
| 04:01.28 | p3nguin | I would rather people learn the words used in communications. |
| 04:01.32 | *** join/#asterisk titter (~titter@c-76-101-240-142.hsd1.fl.comcast.net) |
| 04:01.47 | ChannelZ | And this is the venue that is most appropriate? Go teach school. |
| 04:02.08 | carrar | You got a lot of correcting to do if you are gonna hang out in this channel |
| 04:02.12 | carrar | haha |
| 04:02.14 | ChannelZ | Or keep looking like a big cock, whichever |
| 04:02.18 | p3nguin | If I have to "look like a big cock" to get someone else to learn English, that's fine by me. |
| 04:03.14 | carrar | Visit the Tagata Jinja Hounen Matsuri in Japan |
| 04:03.29 | ChannelZ | wanders off to read icanhazcheezburger |
| 04:03.35 | p3nguin | Next time I'm in the area, I'll try to remember it. |
| 04:03.43 | carrar | http://www.yamasa.org/japan/english/destinations/aichi/tagata_1_600.html |
| 04:03.44 | ChannelZ | This site must make your balls explode |
| 04:03.46 | carrar | NWS |
| 04:03.51 | toortog | what u mean rite english |
| 04:03.54 | toortog | lol |
| 04:04.13 | p3nguin | penile shrine? |
| 04:05.11 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 04:05.29 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 04:05.40 | mtipping | Hey folks. I'm new to asterisk, trying to install asterisknow, and running into a problem right off the bat. |
| 04:06.00 | mtipping | I stat the install, boot from disc, get the first screen, hit enter, it goes to work and quickly stops with a "kernel NULL pointer dereference" error. |
| 04:06.17 | carrar | maybe a bad copy |
| 04:06.23 | carrar | what are you installing it on? |
| 04:06.24 | p3nguin | Did you check the media? |
| 04:06.33 | mtipping | i tried reburning the image a couple times, no dice |
| 04:06.51 | p3nguin | They don't seem to provide the md5sum for the image, but the media does have a self checker built right in. |
| 04:06.52 | carrar | try downloading it again? |
| 04:07.03 | mtipping | that too |
| 04:07.10 | *** join/#asterisk titter` (~titter@c-76-101-240-142.hsd1.fl.comcast.net) |
| 04:07.49 | p3nguin | Are you using the 32 or 64-bit image? |
| 04:07.51 | mtipping | the machine is running XP right now - I don't have the specs right here, but it's well above the minimums recommended for asterisk |
| 04:07.55 | mtipping | 32 |
| 04:07.57 | p3nguin | ed671b5b76caf28ea08a2028d8930ae6 AsteriskNOW-1.5.0-i386-1of1.iso |
| 04:08.00 | p3nguin | Take an md5sum of your iso image and compare it to mine. |
| 04:09.38 | p3nguin | When using AsteriskNOW, it isn't the specs required by Asterisk that matters, but those of CentOS 5. |
| 04:09.39 | mtipping | I'll check... |
| 04:10.02 | p3nguin | But if you're running XP on it, I doubt hardware specs will be the shortcoming. |
| 04:10.18 | p3nguin | It's possibly a bad image or bad optical drive. |
| 04:10.37 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 04:11.34 | cweagans | where should I start looking to figure out why I get the error message: NOTICE[2379]: chan_sip.c:12035 handle_request_invite: Call from '201' to extension '200' rejected because extension not found |
| 04:11.52 | cweagans | I've got 200 and 201 on my desk, and they both say their extension on the phone |
| 04:12.08 | p3nguin | cweagans: sip set debug |
| 04:12.14 | mtipping | I've tried several different linux images - all fail at the same step, and the drive seems to work for other CDs |
| 04:12.19 | p3nguin | cweagans: Check to see where it is looking for 200. |
| 04:12.46 | p3nguin | mtipping: Are you burning the ISO images correctly? |
| 04:12.54 | p3nguin | mtipping: What procedure are you using to burn? |
| 04:14.01 | mtipping | ed671b5b76caf28ea08a2028d8930ae6 *AsteriskNOW-1.5.0-i386-1of1.iso |
| 04:14.14 | p3nguin | Looks like you got a good download. |
| 04:14.23 | mtipping | I've used both nero and NTI media maker |
| 04:14.45 | p3nguin | You're using the "burn image" selection in Nero Burning ROM? |
| 04:15.01 | *** join/#asterisk girlny (~girlny@CPE00195b4be142-CM001a668ec076.cpe.net.cable.rogers.com) |
| 04:15.06 | girlny | how can i fix this |
| 04:15.09 | girlny | /msg NickServ identify <password>. |
| 04:15.33 | girlny | i mean this |
| 04:15.36 | mtipping | burn cd image, yes |
| 04:15.36 | girlny | [Feb 11 23:12:45] ERROR[2158] chan_sip.c: 'TCP' is not a valid transport for '2255'. we only use 'UDP'! ending call |
| 04:16.04 | *** join/#asterisk xpot-mobile (~xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 04:16.36 | p3nguin | girlny: Stop trying to use TCP for SIP. |
| 04:17.54 | carrar | cweagans: make sure both of those extensions exist in your context of internal |
| 04:18.27 | carrar | (in extensions.conf) |
| 04:18.43 | mtipping | Any ideas for what else I should check or try? |
| 04:18.59 | carrar | download again |
| 04:19.15 | carrar | oh you already checked the checksum? |
| 04:19.38 | mtipping | yeah, it matches penguin's |
| 04:20.04 | carrar | try it on another pc |
| 04:20.22 | p3nguin | They really need to put that on the web site so people don't have to potentially waste a CD to fail the media checker. |
| 04:22.02 | cweagans | carrar: that did it. my context was named wrong |
| 04:22.12 | p3nguin | It can't be that much work to add a hyperlink. <a href="AsteriskNOW-1.5.0-i386-1of1.iso.md5">md5sum</a> |
| 04:22.15 | mtipping | I've downloaded and burned it a few times, on different media using different programs (also tried trixbox) I don't think it's the CDs - are there any hardware problems you know of that could be keeping it from installing? |
| 04:22.36 | cweagans | mtipping: the user ;) |
| 04:22.52 | toortog | lol |
| 04:22.54 | p3nguin | mtipping: You can also md5sum the CD if you think it's a bad burn. |
| 04:23.04 | p3nguin | md5sum /dev/cdrom |
| 04:23.43 | mtipping | I'll check them... |
| 04:25.08 | carrar | haha |
| 04:25.14 | carrar | I love looking at binpastes |
| 04:25.15 | carrar | http://pastebin.com/m29829584 |
| 04:25.27 | carrar | "Automated Spyware Installation" |
| 04:26.09 | rue_house | I spent another 2 hours on tech support with panasonic trying to get a call from a co line, to ring a group of phones and then go to a specific voicemail box |
| 04:28.44 | girlny | what does this mean Unable to set SIP RTP TOS to 184, may be you have no root privileges |
| 04:30.35 | carrar | run asterisk as root |
| 04:30.43 | girlny | [Feb 11 23:23:47] WARNING[2173] netsock.c: Unable to set SIP RTP TOS to 184, may be you have no root privileges |
| 04:31.09 | carrar | or set your TOS using iptables |
| 04:32.33 | girlny | carrar how do i go about doing that |
| 04:32.38 | carrar | things girlny is running freepbx |
| 04:32.42 | carrar | thinks |
| 04:32.47 | girlny | and how will it affect me if i dont fix it |
| 04:32.59 | carrar | your log file will eventually fill up |
| 04:33.02 | carrar | thats it |
| 04:33.05 | *** join/#asterisk came0 (~came0@167.83.189.72.cfl.res.rr.com) |
| 04:33.45 | girlny | like whats the warning telling does this affect call quality ? |
| 04:33.58 | carrar | if you use TOS in your network |
| 04:34.01 | carrar | then yes |
| 04:34.39 | carrar | I suspect you are not using tos packet marking |
| 04:34.47 | carrar | but run asterisk as root to resolve that |
| 04:34.57 | carrar | boot that freepbx to the curb |
| 04:35.17 | girlny | any easy way to fix that |
| 04:35.23 | carrar | RUN ASTERISK AS ROOT |
| 04:36.06 | girlny | isnt that less secure |
| 04:36.07 | girlny | ? |
| 04:36.49 | carrar | is having two doors on your car less secure? |
| 04:36.58 | carrar | vs 4 |
| 04:38.23 | carrar | If someone really wants into your server, I suspect there are other ways besides trying to exploit asterisk |
| 04:38.52 | girlny | i have freepbx so i guess i cant use asterisk as root |
| 04:39.00 | carrar | dump freebpx |
| 04:39.08 | carrar | and install Asterisk from source |
| 04:39.27 | carrar | Since this is what THIS channel is all about |
| 04:39.37 | *** part/#asterisk toortog (nesm6@38.111.17.107) |
| 04:39.39 | girlny | carrar i wish i had the knolege to run my pbx straight from asterik unfurtunally i dont |
| 04:39.47 | carrar | You can |
| 04:39.48 | cweagans | girlny: or, if you're not up to installing from source, install ubuntu and then 'sudo apt-get install asterisk' |
| 04:39.57 | cweagans | girlny: i just started using asterisk three days ago |
| 04:40.02 | carrar | that works too |
| 04:40.06 | girlny | i know how to installed |
| 04:40.18 | carrar | then don't install freebps |
| 04:40.19 | cweagans | girlny: and am doing it straight from the cli. No freepbx needed |
| 04:40.28 | girlny | like i installed freepbx form start |
| 04:41.11 | mtipping | know of an easy way I can check the md5sum of a cd in vista? |
| 04:41.40 | girlny | isnt very hard to manage a pbx straight from asterik |
| 04:41.52 | carrar | http://tinyurl.com/yj3meo6 |
| 04:42.27 | cweagans | girlny: first, you manage -asterisk- from the linux command line. Secondly, no. It's not. As I said, I just started three days ago. There's plenty of reference material to get you most of the way there. |
| 04:43.18 | mtipping | haha, I use md5summer, but it doesn't seem to check CDs, only images |
| 04:48.25 | girlny | can you use ARI (asterisk recording interface) , straight from asterisk ? |
| 04:49.08 | girlny | like no freepbx |
| 04:55.30 | KavanS | Raden, was afk for a bit |
| 04:55.37 | KavanS | Raden, well dahdi I'm not sure will work with iaxmodem? |
| 04:55.44 | KavanS | I want to be sure I can just plug hylafax right back in |
| 04:56.01 | carrar | iaxmodem connects via IAX |
| 04:56.09 | carrar | whats that to do with dadhi |
| 04:56.45 | carrar | and Hylafax connects to iaxmodem |
| 04:56.56 | p3nguin | If you receive a fax call over a dahdi channel and send it to iaxmodem, why wouldn't it work? |
| 04:57.02 | Raden | KavanS, why not ? |
| 04:57.08 | KavanS | ahh ok...maybe I am misunderstanding how this work |
| 04:57.10 | KavanS | *works |
| 04:57.10 | carrar | send it to iaxmodem via IAX |
| 04:57.45 | Raden | i never even used zaptel and that seemed clear |
| 04:58.05 | carrar | asterisk -> IAX -> IAX Modem -> faxgetty -> hfaxd |
| 04:58.11 | p3nguin | Dial(IAX2/iaxmodem) |
| 04:58.31 | KavanS | right I understand that part, I guess I'm not understanding how the t400p is configured to take faxes...looking into this now |
| 04:58.55 | carrar | you would pass g711 |
| 04:58.59 | carrar | not t.38 |
| 04:59.26 | Raden | *yawn* |
| 04:59.39 | KavanS | heh yeah, I will read more...no worries |
| 04:59.45 | KavanS | just want to get pointed in the right direction |
| 05:00.24 | carrar | You can use t.38 modem via h.323 |
| 05:00.32 | carrar | http://www.hylafax.org/content/HylaFAX_Connectors |
| 05:07.56 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 05:08.11 | KavanS | roger that, ty for link |
| 05:08.41 | carrar | with a T1 connect to the save box as hylafax, iaxmodem works just fine |
| 05:08.45 | carrar | same |
| 05:08.50 | carrar | with g711 |
| 05:09.06 | carrar | no need for any h.323 crap |
| 05:09.08 | Raden | is faxing over VOIP ever going to work ? |
| 05:09.31 | carrar | without QoS and ovre g711 |
| 05:09.34 | carrar | no |
| 05:09.51 | Raden | still cant get it to work |
| 05:09.56 | Raden | I can fax on vonage all day long |
| 05:10.02 | Raden | never got it to work on any other network |
| 05:10.27 | carrar | bbl |
| 05:12.55 | cweagans | is there somewhere that I can download tracks for music on hold that are already encoded properly? |
| 05:16.56 | p3nguin | raden: I can fax over both SIP and IAX2 without problems. |
| 05:17.08 | Raden | p3nguin, whats your trick whats your network ? |
| 05:17.15 | Raden | and what adapter u using |
| 05:17.19 | Raden | protocals ? |
| 05:17.28 | Raden | what type of connection |
| 05:17.50 | p3nguin | raden: No tricks, no adapters. I can receive faxes through sipgate and I can send faxes through voip.ms. |
| 05:17.55 | p3nguin | raden: cable internet |
| 05:18.12 | p3nguin | raden: Asterisk 1.4.29 with FFA. |
| 05:18.13 | Raden | so you dont actually have a fax machine ? |
| 05:18.17 | p3nguin | right |
| 05:18.25 | Raden | yeah i use a service like that |
| 05:18.27 | p3nguin | But if I did, I'm sure it would still work. |
| 05:18.35 | p3nguin | A service? |
| 05:18.36 | Raden | but omfg no one can figure how to scan to a file folder :( |
| 05:19.53 | KavanS | what about fax detection in dahdi? |
| 05:20.09 | Raden | p3nguin, i cant get fax to work over G711 failure rate over 80% |
| 05:20.28 | p3nguin | raden: Maybe you have REALLY shitty internet connectivity. |
| 05:20.36 | p3nguin | raden: Go run a pingtest. |
| 05:20.44 | p3nguin | pingtest.net, that is. |
| 05:21.06 | Raden | 32 ms to provider |
| 05:21.14 | Raden | 28 mg to google DNS |
| 05:22.56 | Raden | pingtest 22ms ping 0 packet loss 2ms jitter |
| 05:23.36 | Raden | that good LOL |
| 05:28.44 | *** join/#asterisk _Raptor_ (raptorblue@andariel.informatik.uni-erlangen.de) |
| 05:30.34 | *** join/#asterisk titter` (~titter@c-76-101-240-142.hsd1.fl.comcast.net) |
| 05:41.45 | p3nguin | Yeah, I would expect that to be good enough. |
| 05:42.18 | girlny | i get 22 to ec2 |
| 05:42.23 | girlny | 22ms |
| 05:43.09 | p3nguin | http://www.pingtest.net/result/10229174.png |
| 05:47.48 | girlny | http://www.pingtest.net/result/10229316.png |
| 05:48.29 | girlny | i get 18 ms ping and 1 ms |
| 05:48.32 | girlny | jitter |
| 05:57.05 | *** part/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 05:59.21 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 06:01.06 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 06:06.26 | *** join/#asterisk titter` (~titter@c-76-101-240-142.hsd1.fl.comcast.net) |
| 06:08.00 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 06:26.22 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-fvjuonipefpavkyx) |
| 06:42.01 | *** join/#asterisk Heretic (~fallen@dsl-246-85-127.telkomadsl.co.za) |
| 06:42.57 | Heretic | lo all |
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| 07:46.32 | centrex | if I'm connecting a previous pbx to an asterisk for voicemail only, and using the serial for smdi, is there a way to communicate to the asterisk system over that serial or do I have to have another type of connection, fxo/fxs or sip etc... to record/play voicemails? I can't use the serial for that can I? |
| 07:47.23 | fenrus | using a serial interface as interconnect between two servers ? |
| 07:48.19 | fenrus | my guess is that this will only be trouble. Legacy stuff tends to be fubar.. Anything wrong with ethernet? |
| 07:49.26 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 07:49.36 | centrex | yeah, the other systemm doesn't have one apparently |
| 07:50.11 | centrex | thanks |
| 07:50.14 | fenrus | add one? ;) |
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| 08:08.49 | juancferrer | I just setup a new 1.6 installation on a virtual ubuntu 9.10 running on amazon EC2. In sip.conf, I defined one trunk (type=peer) and one device (type=friend) with extension 100 |
| 08:09.22 | juancferrer | If I place a .call file, I can have asterisk call my cellphone and everything works great, I can do the demo and the echo test is excellent |
| 08:10.14 | juancferrer | Using the sip phone extension 100, I can call extension 's' and in the console i see it running the demo test, but I hear nothing |
| 08:10.50 | juancferrer | using X-lite |
| 08:11.14 | juancferrer | what am I doing wrong? |
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| 08:14.13 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:15.04 | juancferrer | so basically, using a cellphone it works great, but using a sip phone client, i get nothing in either direction |
| 08:26.26 | *** join/#asterisk icyValk77 (~icyValk77@cl-670.lon-02.gb.sixxs.net) |
| 08:28.24 | *** join/#asterisk c0rnoTa (~c0rnoTa@178.176.146.168) |
| 08:29.38 | c0rnoTa | morning, all |
| 08:29.39 | c0rnoTa | :) |
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| 08:52.51 | rizwank | I'm having trouble getting AGI commands working --- How does one pass multiple arguments back? the | seems to be no longer supported, but the commas aren't being interpreted either |
| 08:55.55 | juancferrer | like setting variables? |
| 08:57.32 | juancferrer | any time I call an agi script the only way I pass data back to asterisk is by setting variables |
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| 09:00.20 | *** join/#asterisk joelsolanki (joelsolank@202.160.161.94) |
| 09:00.30 | rizwank | I'm trying to use EXEC to run commands |
| 09:00.32 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
| 09:00.36 | shamelessn00b | hi \ |
| 09:00.39 | rizwank | READ isnt't working but playback is for instance |
| 09:00.51 | shamelessn00b | anyone used asterisk-java? |
| 09:00.52 | joelsolanki | is shell function needed to be compiled or comes by default in asterisk ? |
| 09:00.57 | shamelessn00b | for agis and stuff?? |
| 09:01.13 | shamelessn00b | .join #tomcat |
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| 09:14.56 | *** join/#asterisk kamh (~kamh@xdsl-1814.wroclaw.dialog.net.pl) |
| 09:15.06 | kamh | good morning |
| 09:15.34 | c0rnoTa | morning |
| 09:21.27 | *** join/#asterisk nicknick (~administr@host213-123-201-13.in-addr.btopenworld.com) |
| 09:22.27 | *** join/#asterisk romb (romb@unaffiliated/romb-work/x-7222485) |
| 09:22.31 | kaldemar | joelsolanki: it's a separate module, but it should be enabled by default. what version are you using? |
| 09:23.57 | joelsolanki | oh sorry back |
| 09:24.01 | Pimmetje | juancferrer: I never got X-lite to work with my asterisk Zoiper works for mee (there is a limited free version) |
| 09:24.24 | romb | hello all |
| 09:24.33 | romb | i have problem with attended transfer |
| 09:24.42 | juancferrer | ha, i really hope it's not x-lite, i've wasted about 2 hours trying to fix it...i guess i'll try another client |
| 09:24.45 | joelsolanki | i m using Asterisk 1.4.22 |
| 09:24.46 | romb | 101 is calling to 102 |
| 09:25.04 | romb | 102 attended tarnsfer to 108 |
| 09:25.09 | romb | 108 answer |
| 09:25.13 | joelsolanki | kaldemar: core show functions doesnt show shell function :( |
| 09:25.28 | romb | 102 hears beets like still dilaing |
| 09:25.30 | romb | http://pastebin.ca/1794424 |
| 09:25.31 | romb | log |
| 09:26.00 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 09:26.02 | joelsolanki | kaldemar: any idea? |
| 09:26.30 | juancferrer | fuck, it was x-lite |
| 09:26.50 | juancferrer | oh well, blink on OSX works |
| 09:26.54 | *** join/#asterisk titter` (~titter@c-76-101-240-142.hsd1.fl.comcast.net) |
| 09:26.57 | kaldemar | joelsolanki: care to answer by question about your version? func_shell was introduced in 1.6.0. |
| 09:27.15 | kaldemar | joelsolanki: oh sorry, missed your answer. so, you don't have the function. |
| 09:27.20 | joelsolanki | :) |
| 09:27.24 | joelsolanki | got it. |
| 09:27.32 | joelsolanki | so i need 1.6.0 :( |
| 09:27.51 | joelsolanki | actually i m working on getting something work done so i neeed shell function |
| 09:27.54 | joelsolanki | here is what i want |
| 09:28.00 | juancferrer | x-lite is in the trash |
| 09:28.03 | kaldemar | or backport the module or do what you're aiming at by some other means. |
| 09:28.19 | *** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net) |
| 09:28.26 | joelsolanki | hmm. |
| 09:28.37 | joelsolanki | here is what i wantt o do |
| 09:29.06 | joelsolanki | when people dial 8080 i will ask them the number to dial and that number i want to save in mysql database |
| 09:29.15 | *** join/#asterisk tiav (~tiav@mx.fr.smartjog.net) |
| 09:29.27 | joelsolanki | do u think any way to get the customer's dialed number to be stored in database ? |
| 09:29.55 | juancferrer | joelsolanki: AGI |
| 09:30.19 | joelsolanki | hmm. AGI. |
| 09:30.26 | joelsolanki | i have to study it it then. |
| 09:30.31 | juancferrer | it's easy |
| 09:30.39 | joelsolanki | :) |
| 09:30.59 | kaldemar | joelsolanki: you can do that with app System aswell. |
| 09:31.37 | juancferrer | oh, yeah, System, that would be easier |
| 09:31.40 | kaldemar | if you prefer to use a shell script to do the database insert. |
| 09:31.51 | juancferrer | but AGI is easy too |
| 09:32.08 | joelsolanki | i see |
| 09:32.15 | joelsolanki | app system |
| 09:32.19 | joelsolanki | let me see what it is |
| 09:32.23 | JT | couldn't you use CURL too? |
| 09:32.36 | juancferrer | with AGI your script gets passed in a bunch of extra info that'll come in handy |
| 09:34.00 | joelsolanki | got it. |
| 09:34.12 | joelsolanki | System app and Agi both can fix my stuff |
| 09:34.25 | joelsolanki | i will check that out and i will check curl too |
| 09:34.40 | joelsolanki | but looks like AGI can do much more. |
| 09:35.03 | JT | just keep the performance overhead in mind |
| 09:36.01 | juancferrer | fastagi is the fastest i believe |
| 09:36.16 | JT | well yes |
| 09:36.23 | JT | but not faster than dialplan |
| 09:39.09 | joelsolanki | yes correct |
| 09:41.40 | Pimmetje | Does anyone know if there is a plugin or somting to store quick dall numbers with the realnumber and the name in a database and that show the nam of the users when he/she calls and use the quickdall to call? |
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| 09:52.40 | mike8 | What are the options when you want to connect asterisk with skypeout to call land lines? |
| 09:52.49 | *** part/#asterisk Porks (~BARAD-DUR@unaffiliated/porks) |
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| 09:55.28 | mike8 | is the skype channel module from digium the only option for that? |
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| 10:27.57 | micw | hi |
| 10:28.21 | micw | i have an asterisk installation with a 4 port isdn card, using the zap channel driver |
| 10:28.37 | micw | on the card 2 ports are connected |
| 10:29.05 | micw | while i can receive more than 2 calls at a time, i cannot dial out anymore if 2 lines are busy |
| 10:29.23 | micw | in zapata.conf is: |
| 10:29.26 | micw | bchan=1,2 |
| 10:29.27 | micw | bchan=4,5 |
| 10:29.37 | micw | and a group with channel => 1,2,4,5 |
| 10:30.00 | kaldemar | pastebin the real configuration and the dial line. |
| 10:30.50 | *** join/#asterisk jandem (~jandem@unaffiliated/jandem) |
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| 10:37.43 | micw | http://pastebin.com/m1cec3f0c |
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| 10:38.45 | micw | all incoming (independent of chan 1,2,3,4) are in context isdn-in |
| 10:38.51 | micw | so this part works fine |
| 10:39.38 | micw | dialing is done via: |
| 10:39.41 | micw | exten => s,n(call),Dial(ZAP/g1/${ARG1},${ARG4},${ARG5}) |
| 10:39.56 | micw | ${ARG1} - EXTEN ${ARG4} - timeout ${ARG5} - options |
| 10:40.17 | kaldemar | micw: bchan is not a zapata.conf parameter, you can drop those. |
| 10:40.33 | micw | ok |
| 10:41.07 | micw | timeout is 100, options is "" |
| 10:41.09 | *** join/#asterisk asteriskATmarmuD (~mundt@193.158.65.23) |
| 10:42.45 | micw | the result is: DIALSTATUS: CONGESTION, HANGUPCAUSE: 34 |
| 10:42.55 | micw | when 2 lines are in use |
| 10:43.31 | kaldemar | can you show a real CLI output for a failed call? |
| 10:44.09 | *** join/#asterisk smooth_penguin (~smoove@59.95.46.254) |
| 10:44.55 | *** part/#asterisk troubled (~troubled@unaffiliated/troubled) |
| 10:45.14 | micw | http://pastebin.com/d1f734b9a |
| 10:45.26 | micw | i replaces the numbery by X/Y |
| 10:47.36 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.178.71.dsl.dyn.forthnet.gr) |
| 10:49.16 | Ad-Hoc | hi |
| 10:52.25 | *** join/#asterisk viq (~viq@unaffiliated/viq) |
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| 10:59.31 | *** part/#asterisk darkskiez_ (~dz@62-50-207-156.client.stsn.net) |
| 11:02.50 | micw | kaldemar, any idea what outbound is only working with 2 lines? |
| 11:08.36 | kaldemar | based on what you've showed so far, no, if you really can receive more than 2 calls at a time. |
| 11:12.11 | *** join/#asterisk kruemeltee (~Maddin@port-92-198-62-82.static.qsc.de) |
| 11:12.20 | kruemeltee | hello again :-) |
| 11:13.04 | Ickmund | I've got a SPA942 connected to an * 1.6.1.11. I can call other extensions with it, but not receive. DEVICE_STATE says INUSE. What could cause this? |
| 11:14.35 | micw | definitely. |
| 11:14.49 | micw | i restarted asterisk and will see if it was maybe a bug |
| 11:14.51 | kruemeltee | does anybody knows about a working GUI for configuring * without SQL? I mean a WebGUI or something like that, that is able to use the config * files directly? |
| 11:19.54 | mike8 | kruemeltee: freepbx? |
| 11:20.46 | kruemeltee | I don't know it ... but thanks ... I'll give it a try ;-) |
| 11:33.39 | *** join/#asterisk viraptor (~viraptor@212.11.65.66) |
| 11:34.14 | *** join/#asterisk viq (~viq@unaffiliated/viq) |
| 11:35.35 | viraptor | hi, does anyone know in which version did asterisk gain support for sending the audio back to the stream source (instead of the advertised port) |
| 11:39.26 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 11:41.37 | *** join/#asterisk [psy] (~psy0rz@lounge.datux.nl) |
| 11:42.56 | [psy] | while running a sipp stresstest on asterisk 1.4.x, with a dahdi_dummy driver on 2.6.27.45, we can hang a kernel. anyone any idea how to fix this or where to report it? |
| 11:48.32 | kaldemar | [psy]: https://issues.asterisk.org , http://www.asterisk.org/developers/bug-guidelines |
| 11:48.43 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 11:48.48 | *** join/#asterisk rhp (~c3f0bccd@gateway/web/freenode/x-sepdhekgtcgofqga) |
| 11:49.17 | kaldemar | [psy]: search the issue site first for an already open issue. |
| 11:50.35 | rhp | Hi all. Yesterday I configured asterisk for the first time. I am shooting for a simple setup with a couple of SIP users that want to talk to each other using head-sets. Using the tutorials from voip-info.org I got to a situation where we can connect, but we do not hear any sound at the other side. Any thoughts? |
| 11:50.50 | kaldemar | viraptor: what exactly are you talking about? SIP and NAT? |
| 11:51.26 | kaldemar | rhp: is there a NAT involved in the network setup? |
| 11:52.19 | rhp | I'm using NAT to connect to the internet, over which I have a VPN, over which I connect to the asterisk server. |
| 11:52.47 | viraptor | kaldemar: yes sip&nat scenario - I think I saw an option that allowed you to send replies to the host that starts sending the audio to asterisk, even if it's not the address that was reported in sdp |
| 11:53.32 | rhp | kaldemar: when I try the tutorial that sends me a mp3, I do hear the sound. |
| 11:54.09 | fenrus | can your non-vpn -connected phones talnk to eachother? |
| 11:54.29 | [psy] | thx kaldemar |
| 11:54.49 | rhp | fenrus: I didn't try that yet. |
| 11:55.14 | [psy] | 710 issues :D |
| 11:55.24 | *** join/#asterisk E-bola (~bola@smtp.techbiz.dk) |
| 11:55.46 | E-bola | In 1.6.2 why isnt this a valid cli command? sip set debug 192.168.2.231 |
| 11:55.58 | [psy] | uuh 598 |
| 11:56.02 | E-bola | help says: sip set debug {on|off|ip|peer} Enable/Disable SIP debugging |
| 11:56.41 | *** join/#asterisk MatBoy (~MatBoy@wiljewelwetenhe.xs4all.nl) |
| 11:57.25 | E-bola | hmm figured it out now |
| 11:57.48 | kaldemar | ~sipnat |
| 11:57.50 | 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 |
| 11:57.53 | E-bola | very counter intuitive help.... |
| 11:58.01 | kaldemar | viraptor: ^^ that will help you |
| 11:58.03 | fenrus | E-bola, what was the syntax? |
| 11:58.17 | fenrus | E-bola, was it sip set debug ip <ip>? |
| 11:58.48 | E-bola | yep |
| 12:04.25 | rhp | For debugging, I tried connecting to the server from another machine (both on the 'other side' of the VPN). However the system hangs while registering. How can I debug why it is doing that? |
| 12:04.45 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 12:05.19 | fenrus | rhp, does the server/phones in the subnet haver routing to the vpn-network ? |
| 12:05.27 | fenrus | can you ping between the nets etc |
| 12:06.09 | rhp | Yes |
| 12:06.24 | rhp | I seem to have 'full access' to the systems. |
| 12:07.20 | E-bola | has a spa 941 that doesnt wanna attempt to register with asterisk although it sends tons of debug saying its trying to.... |
| 12:08.33 | *** join/#asterisk pawz (~pawz@ppp118-208-108-108.lns20.bne4.internode.on.net) |
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| 12:12.26 | slashtom | does anyone have any experience with the Druid Linux distro (comes bundled with Asterisk and othe VoIP stuff, based on CentOS) |
| 12:12.43 | slashtom | i'm trying to upgrade from Asterisk 1.4.23.1 to 1.4.29 |
| 12:13.25 | slashtom | having given up with yum, i'm compiling from source. but any replacement is causing /usr/sbin/safe_asterisk to seg fault |
| 12:14.01 | slashtom | it's seg faulting at the start of a while loop, and googlin around that seems to be a bit of a red herring |
| 12:14.32 | slashtom | am i missing something obvious, but is there a more sensible way to upgrade Asterisk on a CentOS system? |
| 12:14.40 | *** part/#asterisk [psy] (~psy0rz@lounge.datux.nl) |
| 12:14.42 | slashtom | CentOS based* |
| 12:14.47 | *** join/#asterisk nightrid3r (kvirc@41.214.144.44) |
| 12:17.40 | viraptor | kaldemar: doesn't really help me - I know the configuration details, just wanted to know if asterisk can start sending the audio to a discovered port and if yes, since which version |
| 12:19.25 | kaldemar | viraptor: it can, and that tutorial tells you what to do to achieve that. the nat options have been around since pre 1.0 versions. |
| 12:20.25 | *** join/#asterisk titter` (~titter@c-76-101-240-142.hsd1.fl.comcast.net) |
| 12:21.32 | rhp | fenrus: connecting two persons that are not on the vpn also does not work. |
| 12:22.03 | rhp | In Zoiper the microphone bar gives a signal, but the speaker bar not. |
| 12:29.41 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:29.41 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:30.21 | rhp | fenrus: with nat=yes, no difference. |
| 12:30.39 | fenrus | no firewalls in the phones that prevent rtp ? |
| 12:30.44 | fenrus | *softphones |
| 12:30.47 | kaldemar | rhp: where did you put nat=yes? |
| 12:30.50 | rhp | We are using zoiper |
| 12:31.09 | rhp | kaldemar: in sip.conf in the user-sections |
| 12:31.27 | kaldemar | rhp: you should take a look at the sipnat tutorial aswell: |
| 12:31.30 | kaldemar | ~sipnat |
| 12:31.53 | 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 |
| 12:31.53 | rhp | I did |
| 12:31.53 | fenrus | well, calling internally should not be a problem |
| 12:31.54 | fenrus | it's regurlar routing. |
| 12:31.56 | kaldemar | rhp: and you have externip set also? |
| 12:32.01 | rhp | Problem is that I do not know about the exact topology of the network |
| 12:32.04 | rhp | externip? |
| 12:32.49 | fenrus | rhp, possible to pastebin a ptraceroute from each direction towards the other ? |
| 12:33.30 | kaldemar | rhp: if you don't know the topology, you'll have a hard time getting this working. however, best way to debug this is by enabling sip debug in CLI. you'll see the sip message trace when calls are made. |
| 12:33.41 | rhp | As far as the VPN is concerned, we are all on the same subnet with mask 255.255.240.0 |
| 12:33.53 | rhp | kaldemar: I do see these messages. |
| 12:34.46 | Pimmetje | I cant get call forwarding to work my asterisk version : Asterisk 1.4.26.2 config: http://pastebin.ca/1794526 Can someone have a look |
| 12:35.14 | Pimmetje | It's not the full config |
| 12:35.45 | rhp | fenrus: is ptraceroute available for windows? |
| 12:35.58 | rhp | Our desktop systems are windows. |
| 12:37.29 | fenrus | rhp, traceroute :D |
| 12:37.29 | fenrus | was a typeo, then my session timed out |
| 12:37.29 | fenrus | darn umts connections ;) |
| 12:37.51 | fenrus | disabling reinvites might be worth a try aswell |
| 12:38.15 | rhp | So, you want a traceroute from clientA to clientB, or from clientA to asterisk? |
| 12:38.29 | fenrus | A -> B and then B -> A |
| 12:38.34 | rhp | ok |
| 12:38.57 | fenrus | probably really hard to detect any nat-machines anyways |
| 12:39.40 | *** join/#asterisk Victor_Yure (~victor@unaffiliated/victoryure/x-837844) |
| 12:40.08 | kaldemar | rhp: put them on a pastebin and someone will surely take a look at the trace. |
| 12:41.31 | *** join/#asterisk michael-i (~michael-i@141.41.40.185) |
| 12:44.10 | fenrus | now i need to run, will be back in an hour or so :) |
| 12:44.13 | *** join/#asterisk titter (~titter@c-76-101-240-142.hsd1.fl.comcast.net) |
| 12:45.11 | rhp | kaldemar: see http://pastebin.org/90612 |
| 12:45.25 | rhp | The other way around is similar (only one line of trace) |
| 12:45.33 | *** join/#asterisk Dovid (~annon@213.8.121.90) |
| 12:46.14 | viraptor | kaldemar: wherever I look, those links say nothing about the port discovery based on incoming audio stream - do you know the exact name of the option I was asking about? |
| 12:46.25 | rhp | See http://pastebin.org/90615 for the other way |
| 12:46.28 | kaldemar | rhp: a sip debug, not a traceroute output. |
| 12:46.49 | rhp | ok, can do that too |
| 12:47.29 | Pimmetje | that trace route is tracing yourself ? I have two time the same ip :D |
| 12:47.42 | rhp | Yes, I noticed. |
| 12:48.12 | michael-i | I'm having a bit of trouble writing some failover dialplan code. I have one SIP trunk purposefully setup so it cannot register to test the failover. It fails as expected but then fails to dial the second trunk. Here is the log output and extensions.conf context: http://pastebin.ca/1794543 |
| 12:48.31 | kaldemar | viraptor: nat=yes for a device. |
| 12:48.34 | rhp | http://pastebin.org/90616 holds the sip debug |
| 12:48.59 | michael-i | Dialing that trunk directly works everytime. |
| 12:50.49 | viraptor | kaldemar: doesn't work - asterisk starts sending to the "wrong / advertised" port, but when the device starts sending rtp, it doesn't switch to the port the rtp comes from |
| 12:52.54 | kaldemar | rhp: the whole call. and enable also verbosity with core set verbose 10. |
| 12:54.59 | *** join/#asterisk krynnotaur (~krynnotau@unaffiliated/krynnotaur) |
| 12:56.04 | *** join/#asterisk krynnotaur (~krynnotau@unaffiliated/krynnotaur) |
| 12:57.02 | rhp | kaldemar: sorry, http://pastebin.org/90618. |
| 13:00.37 | plundra | I can't have a queuerule be applied ever n-th second, can I? +5 every 30 second for example? I'd have to put the caller back in over and over again for that type of behaviour? |
| 13:03.28 | plundra | Ah! Silly me, another line for my rule of course. Don't know why but I thought there was one penaltychange per rule. |
| 13:05.03 | kaldemar | rhp: maybe your zoipers don't play nice with re-invites. what version of asterisk are you using? |
| 13:05.20 | rhp | Asterisk PBX 1.6.2.2-1 |
| 13:06.59 | rhp | We are using zoiper because that was suggested by voip-info.org. |
| 13:08.10 | *** join/#asterisk titter` (~titter@c-76-101-240-142.hsd1.fl.comcast.net) |
| 13:08.17 | michael-i | Do I need to reset the stack or channel variables between dialing attempts? It seems once the first Dial fails with CONGESTION, the second isn't even attempted (circuit-busy) |
| 13:08.38 | plundra | Ok, so this is weird. When I check my incoming caller with "core show channel <foo>", I see that QUEUE_MAX_PENALTY=16, which is what I expected, but no new members were called. |
| 13:08.59 | plundra | I have one member with penalty 1, which got called from the begining, but also a few with 15, but they were never tried. |
| 13:17.06 | kaldemar | rhp: put directmedia=no under [general] in sip.conf and try again |
| 13:17.20 | rhp | ok |
| 13:18.22 | kaldemar | michael-i: your dialplan determines the behavior, not variable contents by themselves. |
| 13:20.27 | *** join/#asterisk crazybyte (~crzp@unaffiliated/crazypenguin/x-000001) |
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| 13:21.15 | michael-i | kaldemar: I'm just curious why this second dial is never executed... http://pastebin.ca/1794543 |
| 13:21.32 | *** join/#asterisk ChrisWi (~admin@mx2.wwserver.net) |
| 13:21.48 | michael-i | I've now tried with a dead IAX account, dead DAHDI analog port, etc... every time the second dial is skipped with circuit-busy |
| 13:23.52 | rhp | kaldemar: no difference. |
| 13:23.59 | rhp | with or without nat. |
| 13:24.13 | rhp | with or without directmedia |
| 13:24.31 | *** join/#asterisk thazza (~thazza@124-254-81-140-static-dsl.ispone.net.au) |
| 13:24.47 | kaldemar | michael-i: your CLI output doesn't match the configuration. output says "Goto (SIP-PHONE-1,failover,1)" and configuration says "Goto(failover,1)". |
| 13:25.48 | michael-i | kaldemar: isn't that just how it's displayed in logs? |
| 13:25.53 | kaldemar | michael-i: no. |
| 13:28.00 | *** join/#asterisk mnick86 (~mnick86@188.195.88.246) |
| 13:28.28 | mnick86 | I have MALLOC_DEBUG on, but I am missing the "core show locks" command . any suggestions ? |
| 13:29.49 | michael-i | kaldemar: I'm pretty sure it's just displayed that way. I just put in a NoOp() as the first priority of the GoTo target and it's being executed |
| 13:30.19 | *** join/#asterisk e4 (~e4@rrcs-76-79-59-194.west.biz.rr.com) |
| 13:30.30 | kaldemar | michael-i: gah, it is. i thought i was looking at different verbosity output. :P |
| 13:33.29 | mnick86 | what do I have to do to enable "core show locks" ?! |
| 13:34.41 | ManxPower-work | mnick86: What version of Asterisk? |
| 13:35.06 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-0-16-133.dhcp.embarqhsd.net) |
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| 13:35.39 | ManxPower-work | Dallas has 9" of snow? |
| 13:35.54 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 13:36.04 | mnick86 | ManxPower-work: 1.6.2 |
| 13:36.49 | Katty | morning |
| 13:36.58 | IsUp | morning |
| 13:37.38 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 13:37.55 | ManxPower-work | mnick86: I don't see anything obvious in the release notes about locks. |
| 13:38.48 | russellb | mnick86: compile with "DEBUG_THREADS" enabled in menuselect |
| 13:39.54 | *** join/#asterisk asteriskATmarmuD (~mundt@193.158.65.23) |
| 13:40.33 | *** join/#asterisk nix8n82 (~AndChat@63.162.27.14) |
| 13:44.19 | angryuser | ~book |
| 13:44.20 | 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 |
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| 14:11.48 | benngard | prbably a very stupid question but i ask anyway, can u from a cell phone, send a sms to an * extension letting the sms application answer and save it as a text file? did some tests, but just endde up with:-- SMS TX 7F 00 [Feb 12 14:47:09] NOTICE[30629]:app_sms.c:1800 sms_process: bad stop bit |
| 14:12.52 | Naikrovek | i don't think so, but i'm not sure. as far as I know * doesn't do SMS |
| 14:13.00 | Naikrovek | but let me tell you |
| 14:13.04 | Naikrovek | i'm a fuck up and wrong all the time |
| 14:13.11 | benngard | :) |
| 14:13.25 | Naikrovek | i'm so depressed today i can feel it in my chest |
| 14:13.28 | Naikrovek | and legs |
| 14:13.39 | Naikrovek | like a disease |
| 14:13.45 | Naikrovek | trying to end me |
| 14:13.47 | Naikrovek | it sucks |
| 14:14.10 | ManxPower-work | benngard: no! |
| 14:14.24 | benngard | i was afraid for that answer :( |
| 14:14.52 | ManxPower-work | benngard: read /doc/sms.txt |
| 14:15.00 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
| 14:15.45 | ManxPower-work | Asterisk supports SMS over PSTN (landline) using FSK. Carriers in the USA/Canada do not support that method of SMS |
| 14:21.09 | *** join/#asterisk jaytee (~jforde@unaffiliated/jaytee) |
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| 14:40.58 | *** join/#asterisk Yahto (~Yahto@port143.ds1-gr.adsl.cybercity.dk) |
| 14:41.02 | Yahto | Hey |
| 14:41.21 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.9.210) |
| 14:41.25 | Yahto | I was told that this is a good place to get some help with asterisk so here i am. |
| 14:42.05 | Naikrovek | ask your question |
| 14:42.25 | Yahto | i am setting up a asterisk server atm with 2 numbers(sip trunk) and i want the sever to be abel to use number 1 first and if that is bussy then use second when ppl call out |
| 14:43.08 | Yahto | i thought i had it right but well not workign as i want it 2 |
| 14:43.15 | ManxPower-work | Yahto: see macro-std-exten in extensions.conf.sample to see how to read the HANGUPCAUSE. |
| 14:43.16 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
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| 14:43.40 | ManxPower-work | I don't understand why you would want to call a busy number twice, however. |
| 14:44.01 | Naikrovek | i think he's talking about outgoing caller id |
| 14:44.37 | Yahto | i dont want to call a bussy number ill try to explain abit more |
| 14:44.43 | Yahto | i have 2 out going numbers |
| 14:44.48 | Naikrovek | he has two "lines" (an imaginary concept but let's run with it) and when one is in use, he wants the caller ID when he calls out the "second line" to be number of the "second line" |
| 14:45.06 | ManxPower-work | *nod* Imaginary lines are always a bitch to work with. |
| 14:45.09 | Naikrovek | indeed |
| 14:45.19 | Naikrovek | Yahto: did i describe that correctly? |
| 14:45.26 | Yahto | yep |
| 14:45.28 | Naikrovek | okay |
| 14:45.30 | Naikrovek | first thing |
| 14:45.31 | brettnem | is it just me or is pastebin.ca not responding? |
| 14:45.35 | Naikrovek | there are no "lines" in asterisk |
| 14:45.37 | ManxPower-work | Hopefully he's reading that document I pointed him to. |
| 14:45.40 | Naikrovek | yeah |
| 14:45.42 | Naikrovek | just do that |
| 14:46.01 | Naikrovek | you can have one phone number with 100,000 "lines" if you want |
| 14:46.16 | Naikrovek | or you can have the first "line" call the second when a call comes in |
| 14:46.26 | Naikrovek | so that way you don't need to worry about outgoing callerid |
| 14:47.58 | *** join/#asterisk RobH_ (~robh@2620:0:860:2:21e:c2ff:fe03:2465) |
| 14:48.09 | Yahto | ill take a look at it i proberly be back but thanks for the pointer for now |
| 14:48.16 | Naikrovek | no prob |
| 14:48.22 | Naikrovek | ask questions and we'll give answers |
| 14:48.31 | Naikrovek | hopefully without attitude but no promises |
| 14:48.34 | Naikrovek | i can promise i wont |
| 14:48.39 | Naikrovek | but i can't promise that for anyone else |
| 14:48.47 | Yahto | hehe :) |
| 14:49.02 | *** join/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 14:50.47 | brettnem | Hey all, I'm having trouble with attended transfers with a 1.4 SVN build of asterisk. Using polycom's running build 3.1.1. Call drops on second transfer. From sip trace I see 202, then a NOTIFY with a 481 in it. That seems to cause the middle phone to drop the call. If I roll back asterisk to latest stable, transfers with the same phones, configs, firmwares, etc, work perfectly.. Here's my log: http://st.pastebin.com/d45d06855 |
| 14:51.09 | rocksfrow | does anybody have any tips on how to debug my digium card? |
| 14:51.24 | rocksfrow | everything we working fine..then when i came back over the weekend..all circuts are busy..and the digium card is blinking red |
| 14:51.25 | brettnem | of particular interest, it seems that there may be some URI encoding issues from this line: |
| 14:51.27 | brettnem | [Feb 11 22:44:37] DEBUG[13163] chan_sip.c: Looking for callid f8bee9f6-48f5c55f-1073902c%40192.168.192.25%3Bto-tag%3Das153c4837%3Bfrom-tag%3D991A9EED-2B6F3472 (fromtag totag ) |
| 14:51.47 | rocksfrow | currently anybody who calls, get's a busy signal |
| 14:52.17 | brettnem | rocksfrow: what kind of interface is it? |
| 14:52.21 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 14:52.40 | rocksfrow | 02:01.0 Communication controller: Digium, Inc. Wildcard TE205P (rev 02) |
| 14:52.52 | plundra | Does the queue-app not reevaluate what QUEUE_MAX_PENATLY is set to, while the caller is in the queue? Because it _is_ increasing, with help of queuerules, but still higher penalty memebers are never called. |
| 14:53.06 | rocksfrow | the lines are analog, but connects via the wildcard with a rj45 interface |
| 14:53.28 | rocksfrow | there is one line that is a separate pair |
| 14:53.30 | rocksfrow | used for hte fax machine |
| 14:53.48 | rocksfrow | so i tried dialing out directly from that line, and it works |
| 14:53.49 | ManxPower-work | The TE205P does not support analog. |
| 14:53.56 | brettnem | correct me if I'm wrong, but that's not an analog card? |
| 14:54.01 | rocksfrow | .... |
| 14:54.26 | rocksfrow | this is the confusion i have/had with the telecom guy |
| 14:54.37 | rocksfrow | but..he says they're analog lines |
| 14:54.48 | rocksfrow | but to this t1 like interface |
| 14:54.57 | brettnem | rocksfrow: THAT is a dual port T1/E1 card.. you cannot plug analog lines into it.. |
| 14:54.59 | ManxPower-work | rocksfrow: if you plug an analog phone into it, do you get dialtone? |
| 14:55.10 | ManxPower-work | (the telco line, not Asterisk) |
| 14:55.11 | rocksfrow | an anog phone line is NOT plugged into it |
| 14:55.23 | ManxPower-work | rocksfrow: plug one into it and find out for SURE if it's analog. |
| 14:55.32 | rocksfrow | oh you mean into the port where the rj is coming rom'? |
| 14:55.34 | rocksfrow | from** |
| 14:55.43 | rocksfrow | sorry for being so daft.. |
| 14:55.45 | ManxPower-work | I mean plug a phone into the line coming from the telco |
| 14:55.59 | rocksfrow | the line coming from the telco is rj45, not analog |
| 14:56.00 | ManxPower-work | If you get dialtone then you know you have an analog line. |
| 14:56.06 | rocksfrow | the rj45 comes from a boxx |
| 14:56.11 | rocksfrow | thats owned by the telcom i assume |
| 14:56.16 | ManxPower-work | what in the world makes you think RJ45 means "not analog" |
| 14:56.18 | brettnem | rj45 is an physical definition of a port, not a protocol |
| 14:56.23 | rocksfrow | ok yes |
| 14:56.26 | brettnem | :) |
| 14:56.27 | rocksfrow | i thought thats what u guys were saying |
| 14:56.28 | rocksfrow | one minute |
| 14:56.30 | rocksfrow | let me go try that |
| 14:56.31 | ManxPower-work | In any case, you have my advice. Follow it or not, I don't care. |
| 14:56.33 | rocksfrow | srry..and thanks |
| 14:56.36 | rocksfrow | yes i am i am..lol |
| 14:56.41 | rocksfrow | let me try that |
| 14:56.42 | rocksfrow | one min |
| 14:56.58 | ManxPower-work | rocksfrow: 5 mins spent on this could save you hours and hours. |
| 14:57.40 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 14:58.33 | brettnem | any ideas on my transfer issue? I think it might be a bug since rolling back to stable fixes it |
| 14:59.02 | rocksfrow | okay so... |
| 14:59.05 | *** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 14:59.18 | rocksfrow | why in the world would the telecom tell me i have analog |
| 14:59.25 | *** join/#asterisk puzzled (~patrick@94.157.69.223) |
| 14:59.36 | rocksfrow | the box says 'AdTran' on it |
| 14:59.43 | brettnem | box |
| 14:59.52 | rocksfrow | the line from the telco goes into that box, whcih has the rj45 interface |
| 14:59.54 | rocksfrow | sorry, i know.. |
| 14:59.59 | ManxPower-work | Yes. We have hundreds of both analog and t-1 boxes |
| 14:59.59 | rocksfrow | i dont know what else to call it |
| 15:00.13 | rocksfrow | well i plugged the phone into the rj45 port |
| 15:00.15 | ManxPower-work | A device that turns T-1 into analog ports is called a "Channel bank" |
| 15:00.18 | rocksfrow | and i hear noise, not a dial tone |
| 15:00.25 | ManxPower-work | What MODEL of Adtran is that? |
| 15:00.34 | rocksfrow | hrm trying to findo ut |
| 15:00.36 | rocksfrow | one min |
| 15:00.47 | brettnem | this used to work?! |
| 15:01.01 | rocksfrow | yesss |
| 15:01.05 | rocksfrow | for yrs |
| 15:01.37 | brettnem | heh, was that T1 always plugged into the adtran? or did it used to be connected direct to the TE205P? |
| 15:01.47 | rocksfrow | nothing has changed |
| 15:01.54 | rocksfrow | this setup was done by the old sysadmin |
| 15:02.01 | *** part/#asterisk benngard (~benngard@213.88.138.230) |
| 15:02.05 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 15:02.10 | brettnem | cable ninjas did it |
| 15:02.11 | rocksfrow | 02:01.0 Communication controller: Digium, Inc. Wildcard TE205P (rev 02) |
| 15:02.14 | rocksfrow | thats from lspci |
| 15:02.22 | rocksfrow | the adtran is connencted to that card |
| 15:02.25 | rocksfrow | via one rj45 |
| 15:02.29 | ManxPower-work | I'm still waiting for the adtran model number |
| 15:02.31 | rocksfrow | my fax is hooked up to a handyton |
| 15:02.41 | rocksfrow | directly from a single pair that comes out of the adtran box |
| 15:02.47 | rocksfrow | yes im trying |
| 15:02.48 | ManxPower-work | Chances are it's a Total Access 750 or 850, but we would have to confirm that. |
| 15:02.49 | brettnem | yeah, get that adtran model.. |
| 15:03.22 | rocksfrow | this thing is so plain |
| 15:03.32 | rocksfrow | i cant find any model # lol |
| 15:04.00 | brettnem | special edition |
| 15:04.13 | rocksfrow | can you elaborate one what you think it is manx? |
| 15:04.27 | ManxPower-work | rocksfrow: Adtran makes over 100 products. |
| 15:04.57 | rocksfrow | well shit |
| 15:05.14 | rocksfrow | there isnt a way i can determine if this is my digium card |
| 15:05.18 | rocksfrow | or the adtran box/telecom? |
| 15:05.18 | ManxPower-work | rocksfrow: at this point you'll just have to wander around aimlessly until you figure out what changes. |
| 15:05.25 | rocksfrow | ..what changes? |
| 15:05.27 | ManxPower-work | changed, that is. |
| 15:05.32 | rocksfrow | nothing has been changed |
| 15:05.34 | rocksfrow | something is broken |
| 15:05.37 | rocksfrow | it was working friday |
| 15:05.39 | rocksfrow | monday morning is not |
| 15:05.40 | ManxPower-work | It was working, not it's not. Something changed. |
| 15:05.47 | rocksfrow | ....changed? |
| 15:05.49 | rocksfrow | lol wtf |
| 15:05.57 | rocksfrow | somebody snuck in here over the weekend? |
| 15:05.58 | ManxPower-work | you CANNOT know if the telco changed your line or not. |
| 15:06.09 | ManxPower-work | is this an GUIfied Asterisk? |
| 15:06.13 | rocksfrow | freepbx |
| 15:06.19 | *** join/#asterisk p4p4 (~P4p4@248.121.113.82.net.de.o2.com) |
| 15:06.19 | rocksfrow | please dont forward me to #freepbx |
| 15:06.21 | ManxPower-work | *nod* I wonder if it updated itself. |
| 15:06.30 | rocksfrow | manx |
| 15:06.34 | rocksfrow | the voip card binking red |
| 15:06.36 | rocksfrow | blinking red |
| 15:06.42 | rocksfrow | that doesnt tell me shits broke? |
| 15:07.03 | rocksfrow | evertyhing internal works fine |
| 15:07.07 | rocksfrow | dialing internal extensions.etc |
| 15:07.13 | ManxPower-work | No, that could mean the card is not configured, it could mean the line is bad, it could mean your PC broke, it could mean the drivers are corrupted. |
| 15:07.30 | rocksfrow | damn ok |
| 15:07.37 | brettnem | are you SURE that the telco line is connected to the adtran and not the server? |
| 15:07.41 | rocksfrow | well, the cards configured |
| 15:07.48 | ManxPower-work | are you running DAHDI or Zaptel? |
| 15:07.49 | rocksfrow | positive..lol |
| 15:07.52 | rocksfrow | zap |
| 15:07.57 | rocksfrow | ran ztcfg |
| 15:08.00 | rocksfrow | it gives no errors |
| 15:08.03 | ManxPower-work | what does "zttool" give you? |
| 15:08.24 | rocksfrow | not installed, lol |
| 15:08.30 | rocksfrow | ztcfg is..weird |
| 15:08.32 | ManxPower-work | *eye roll* |
| 15:08.32 | rocksfrow | let me instal it |
| 15:08.39 | rocksfrow | dude...please |
| 15:08.46 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 15:08.47 | ManxPower-work | it would have been installed automatically. |
| 15:08.52 | rocksfrow | yeah you'd think |
| 15:09.03 | ManxPower-work | put the output of "cat /proc/zaptel/1" on pastebin.ca |
| 15:09.06 | ariel_ | hello everyone |
| 15:09.25 | rocksfrow | looks good |
| 15:09.39 | rocksfrow | one sec |
| 15:10.13 | angryuser | Good day, what was the syntax to cut the end of the var ? ${VAR:1} the fist, i need the last ${VAR:-1} ?? |
| 15:10.27 | rocksfrow | ManxPower-work, http://pastebin.com/m25867b41 |
| 15:10.41 | ManxPower-work | ous |
| 15:10.42 | ManxPower-work | 30 secs ago keyboardjs |
| 15:10.42 | ManxPower-work | 37 secs ago EmpJoe |
| 15:10.42 | ManxPower-work | 45 secs ago Buggzie |
| 15:10.42 | ManxPower-work | 45 secs ago dd354 |
| 15:10.42 | ManxPower-work | 45 secs ago adi |
| 15:10.43 | ManxPower-work | 1 min ago Shawn Gadwa |
| 15:10.43 | ManxPower-work | 1 min ago Make new post Search Pastebin News Want to buy pastebin.com? Own a little bit of Internet history and develop it further! For news and feedback see my blog. Free subdomainsWant your own xyz.pastebin.com sub-domain for your community? Just type the address into your browser address bar. See help for detailsAboutPastebin is a tool for collaborative debugging or editing, See help for details. Cre |
| 15:10.43 | ManxPower-work | Posted by Anonymous on Fri 12 Feb 15:10 |
| 15:10.44 | ManxPower-work | report abuse | download | new postPlease indicate why this post is abusive, and provide any other useful information.Spam / advertising / junk |
| 15:10.45 | ManxPower-work | Personal details |
| 15:10.45 | ManxPower-work | Proprietary code |
| 15:10.45 | ManxPower-work | Other |
| 15:10.46 | ManxPower-work | comments (optional) |
| 15:10.54 | *** join/#asterisk titter (~titter@c-76-101-240-142.hsd1.fl.comcast.net) |
| 15:11.03 | rocksfrow | ? |
| 15:11.15 | ManxPower-work | Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" B8ZS/ESF RED |
| 15:11.17 | *** join/#asterisk Yoe (~wouter@samba.grep.be) |
| 15:11.21 | ManxPower-work | I flooded the channel. |
| 15:11.28 | ManxPower-work | RED = NO LINE DETECTED |
| 15:11.29 | Yoe | so -- how do I check from within asterisk whether a variable has a value? |
| 15:11.31 | rocksfrow | ManxPower-work, http://pastebin.com/m2620ee26 |
| 15:11.40 | rocksfrow | i pasted the ztcfg output as well |
| 15:11.59 | Yoe | with the old 'DBget()' application, that would jump to N+101 if the variable wasn't set, but apparently the new DB() function doesn't do that? |
| 15:12.12 | ManxPower-work | Yoe: you are reading out of date documentation |
| 15:12.41 | Yoe | ManxPower-work: correct, I'm asking what the new ways are :-) |
| 15:12.57 | ManxPower-work | You check if a variable has a value by something like $["${VAR}" = ""] |
| 15:13.10 | Yoe | I have the first edition of the asterisk book, but not the newer version |
| 15:13.19 | ManxPower-work | Yoe: You were asking about variables, but needing information about a function. |
| 15:13.32 | ManxPower-work | Yoe: well download the new one then. |
| 15:13.44 | Yoe | right, so I do a Set(VAR=${DB(foo/var)}), and then a GotoIf with that condition? |
| 15:14.00 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 15:14.08 | ManxPower-work | assuming everything is right, yes. |
| 15:14.13 | Yoe | 'kay, thanks |
| 15:14.15 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:14.15 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:14.24 | ManxPower-work | Also remember if you put quotes on one side of the = you need them on the otherside as well. |
| 15:14.34 | rocksfrow | ManxPower-work, ...? |
| 15:14.35 | Yoe | sensible |
| 15:14.43 | rocksfrow | did you say red li\ght = no detected to me? |
| 15:14.49 | ManxPower-work | rocksfrow: yes, I did. |
| 15:14.56 | rocksfrow | no line detected as in.. |
| 15:14.58 | rocksfrow | telecom fucked up? |
| 15:15.04 | ManxPower-work | as in no line detected. |
| 15:15.17 | rocksfrow | .....sorry for being daft i just dont know how to take that |
| 15:15.34 | rocksfrow | whats interesting is..on the adtran box..i unplug it 2 lights go red |
| 15:15.37 | rocksfrow | i plug it in |
| 15:15.45 | rocksfrow | red turns green and the other goes out |
| 15:15.46 | rocksfrow | after a min |
| 15:15.49 | rocksfrow | red l ight comes on |
| 15:15.54 | rocksfrow | this is on the adtran box |
| 15:15.59 | Yoe | works, thanks |
| 15:16.11 | rocksfrow | im pretty sure that red light ont he adtran box is labeled alm |
| 15:16.17 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:16.22 | rocksfrow | is that good enough evidence to get the telco to come out here? |
| 15:16.29 | rocksfrow | should i try the other port ont he card? |
| 15:18.27 | *** part/#asterisk Yoe (~wouter@samba.grep.be) |
| 15:18.55 | ManxPower-work | you should contact your telco and say "I have a red alarm" |
| 15:19.21 | ManxPower-work | rocksfrow: where are you located? |
| 15:19.30 | rocksfrow | what did you say this box is porbably called again? |
| 15:19.45 | ManxPower-work | Channel bank |
| 15:20.27 | ManxPower-work | Covad is currently having a "major outage" in the NJ area. |
| 15:20.45 | ManxPower-work | Just saw it is also Level 3 |
| 15:21.30 | ManxPower-work | Likely fiber cut in the area. |
| 15:23.49 | *** join/#asterisk Keeper82 (~Keeper@pat-mi.eni.it) |
| 15:24.25 | *** join/#asterisk garymc (~chatzilla@host81-139-136-16.in-addr.btopenworld.com) |
| 15:24.36 | Keeper82 | Hi everyone, I'm trying to send a fax but asterisk keeps telling that it can't read the TIFF file (permissions are right) |
| 15:24.47 | ManxPower-work | Keeper82: pastebin the error |
| 15:24.51 | ManxPower-work | ~pb |
| 15:24.52 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 15:24.58 | Keeper82 | ok |
| 15:25.19 | ManxPower-work | Keeper82: If you are using a GUI don't waste everyone's time. Ask on #FreePBX (or whatever GUI) |
| 15:25.32 | Keeper82 | no, i installed asterisk from source |
| 15:26.01 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:26.43 | ManxPower-work | Keeper82: good. Just let us know when you have the pastebin |
| 15:26.48 | Keeper82 | http://pastebin.com/m547246e6 |
| 15:27.10 | Keeper82 | I ony changed the real number to xxxxxxx :P |
| 15:27.18 | *** join/#asterisk moy (~moy@74.12.129.100) |
| 15:28.36 | ManxPower-work | The only 2 reasons for that error that I can think of is an invalid TIFF file or permissions. How many bytes does ls -l /tmp/prova.tif show? |
| 15:28.59 | Keeper82 | 39864 |
| 15:29.09 | Keeper82 | and I can open it with an image editor |
| 15:29.20 | ManxPower-work | what is the owner/group of that file? |
| 15:29.25 | Keeper82 | permission are 777 |
| 15:29.35 | ManxPower-work | That's not what I asked. |
| 15:29.53 | Keeper82 | superadmin:superadmin |
| 15:30.01 | ManxPower-work | and what user is Asterisk running as? |
| 15:32.24 | ManxPower-work | many. many applications will refuse to open files in /tmp with 777 permissions |
| 15:32.46 | Keeper82 | i didn't changed it so it must be root |
| 15:32.57 | ManxPower-work | ps -auxwww | grep asterisk |
| 15:33.03 | Pimmetje | asterisk wont run as root |
| 15:33.10 | Pimmetje | in default settings |
| 15:33.15 | ManxPower-work | Pimmetje: Asterisk runs as root just fine. |
| 15:33.30 | Pimmetje | @my machine it compained |
| 15:33.40 | [TK]D-Fender | wonders where a "default" got invented.... |
| 15:33.43 | Pimmetje | that is does not want to run as rioot :D |
| 15:33.44 | ManxPower-work | Pimmetje: then your machine is messed up. |
| 15:33.49 | Pimmetje | hehe |
| 15:33.52 | Pimmetje | could be |
| 15:33.53 | Keeper82 | I confirm root |
| 15:34.06 | ManxPower-work | Pimmetje: maybe you did something stupid and installed an Asterisk GUI. |
| 15:34.17 | ManxPower-work | Keeper82: then I have no more suggestions. |
| 15:34.32 | Pimmetje | No i build asterisk from source |
| 15:34.53 | Pimmetje | and then tried to put a webapp on it for voicemail |
| 15:35.01 | Pimmetje | never got it working :D |
| 15:35.13 | ManxPower-work | voicemail web app != Asterisk. |
| 15:35.18 | *** join/#asterisk chilicuil (~sistemas@unaffiliated/chilicuil) |
| 15:35.18 | Pimmetje | But it my hobby box so i dnt care :D |
| 15:35.45 | Pimmetje | ManxPower-work: I know |
| 15:36.02 | Keeper82 | I even created a simple C program which opens that file and it works (obviously) |
| 15:36.15 | ManxPower-work | Keeper82: you ran that inside of Asterisk? |
| 15:36.22 | Keeper82 | nope |
| 15:36.25 | ManxPower-work | (via System or AGI)? |
| 15:36.33 | ManxPower-work | Well then it's a pretty pointless test isn't it? |
| 15:36.39 | Keeper82 | I tried just to see if my libtiff was working or not |
| 15:36.49 | ManxPower-work | try it inside of Asterisk |
| 15:36.55 | Keeper82 | how? |
| 15:36.59 | Pimmetje | The only thing i never got working that i really would like is call transfer |
| 15:37.11 | ManxPower-work | exten => 666,1,System(/path/to/app) |
| 15:37.20 | Keeper82 | ah ok |
| 15:37.35 | ManxPower-work | Pimmetje: I transfer by pushing the TRANSFER button on my phone. Odd it doesn't work for you. |
| 15:38.05 | *** join/#asterisk came0 (~came0@rrcs-71-42-53-182.se.biz.rr.com) |
| 15:38.06 | Pimmetje | I cant get call forwarding to work my asterisk version : Asterisk 1.4.26.2 config: http://pastebin.ca/1794526 Can someone have a look |
| 15:38.11 | Pimmetje | copy past |
| 15:38.18 | Pimmetje | do not have a tranfer button |
| 15:38.20 | Naikrovek | transfer is a phone thing, is it not |
| 15:38.31 | Naikrovek | what kind of phones do you have |
| 15:38.32 | Pimmetje | like to di it with #1 or so |
| 15:39.53 | Pimmetje | as far as i understand the stuff i read i should be possible to make #1 or something do the same thing as the transfer button |
| 15:40.10 | Pimmetje | or do is miss something here? |
| 15:40.31 | slashtom | i'm looking for an alternative Linux distro to use on an embedded i686 board for running an Asterisk system, to replace Druid http://voiceroute.org/ |
| 15:40.39 | slashtom | any recommendations? |
| 15:41.10 | leifmadsen | Slashman: astlinux? |
| 15:41.59 | slashtom | thanks |
| 15:42.14 | leifmadsen | Pimmetje: you've enabled the 't' and/or 'T' flags in your Dial() and enabled the appropriate things in the features.conf file? |
| 15:42.47 | ManxPower-work | leifmadsen: *gasp* Just like in the Asterisk book! |
| 15:42.54 | leifmadsen | OMG! OMG! OMG! |
| 15:42.57 | Pimmetje | I hope so ;) http://pastebin.ca/1794526 part pof my config |
| 15:43.02 | leifmadsen | I didn't look at the pastebin because it didn't load |
| 15:43.07 | leifmadsen | that, and I'm working on writing documetnatin |
| 15:43.39 | rocksfrow | is there anyway to debug traffic going through my digium card? |
| 15:43.45 | Pimmetje | i indead does not load atm :( |
| 15:43.52 | Pimmetje | crap :P |
| 15:43.58 | rocksfrow | its just ringing...and i dont know how to debug whether its even making it to the server |
| 15:44.22 | *** part/#asterisk thazza (~thazza@124-254-81-140-static-dsl.ispone.net.au) |
| 15:46.20 | *** part/#asterisk garymc (~chatzilla@host81-139-136-16.in-addr.btopenworld.com) |
| 15:46.45 | *** join/#asterisk imcdona (imcdona@173.160.189.69) |
| 15:49.12 | *** join/#asterisk darkskiez__ (~dz@62-50-207-42.client.stsn.net) |
| 15:50.12 | tzafrir | slashtom, what do you actually need? |
| 15:50.22 | tzafrir | Droid is a bit heavy-duty |
| 15:50.40 | tzafrir | How do you intend to manage it? What do you mean by "i686 board"? |
| 15:50.41 | ManxPower-work | rocksfrow: the asterisk console will show the incoming call |
| 15:50.44 | *** join/#asterisk ttl- (~patrick@d5153A420.access.telenet.be) |
| 15:51.06 | tzafrir | What do you actually need to do with it? (hardware? no. of concurrent calls? codecs?) |
| 15:51.34 | tzafrir | slashtom, also: is disk space an issue? Is memory an issue? |
| 15:51.41 | ManxPower-work | BTW Covad and Level 3 are having a significant outage in the NY/NJ area because of a fiber cut. |
| 15:55.06 | kruemeltee | I have to go now ... bye bye |
| 15:55.24 | Naikrovek | fiber cuts still happen? |
| 15:55.34 | Naikrovek | i thought all those backhoe operators were fired |
| 15:56.09 | Naikrovek | or maybe it's some dude diggin up his front yard or something |
| 15:56.49 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 15:57.02 | mnick86 | any ideas how to track that bug down ?! https://issues.asterisk.org/view.php?id=16784 |
| 15:58.42 | brettnem | hey anyone have any ideas about my polycom transfer bug I posted above? :D |
| 15:59.38 | Naikrovek | brettnem: i looked but didn't notice anything |
| 15:59.48 | Naikrovek | but i'm a dumbass so don't rely on my input |
| 16:01.03 | brettnem | hah |
| 16:01.28 | brettnem | thanks for looking tho |
| 16:01.31 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 16:01.50 | brettnem | it seems to be a bug... if I roll from svn back to stable, it works perfectly |
| 16:01.59 | Naikrovek | brettnem: could be |
| 16:02.47 | brettnem | blind transfers work.. just not attended |
| 16:04.12 | Keeper82 | ManxPower-work, I tried to execute my program inside asterisk and it works |
| 16:07.49 | *** join/#asterisk cherva (~cherva@93.152.158.160) |
| 16:08.08 | ManxPower-work | brettnem: do you have the allowTransferonProceeding set in the Polycom? (I don't recall the *exact* option, but can look it up) |
| 16:08.25 | ManxPower-work | brettnem: I suspect a bug too, but worth trying that option. |
| 16:08.41 | cherva | How Can I make a whitelist of numbers for a specific extension ? |
| 16:09.19 | rocksfrow | Feb 12 11:07:37 NOTICE[4761] app_dial.c: Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion) |
| 16:10.17 | rocksfrow | anybody know how to interpret that? |
| 16:10.23 | rocksfrow | congestion means...? |
| 16:10.33 | rocksfrow | there are no calls currently on the system |
| 16:11.58 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 16:12.05 | *** join/#asterisk cherva (~cherva@93.152.158.160) |
| 16:12.27 | brettnem | ManxPower-work: I'll give it a shot.. |
| 16:12.56 | ManxPower-work | rocksfrow: do you still have a red alarm? |
| 16:13.18 | brettnem | ManxPower-work: However that sounds like something that happens before a call is established.. however all of my calls are being set up. .I'm not trying to transfer a ringing call or anything.. |
| 16:13.50 | rocksfrow | ManxPower-work, dude, while i was on hold..the red light went away |
| 16:14.00 | rocksfrow | i unplugged/plugged it in a couple of times..and it came on |
| 16:14.11 | rocksfrow | the digium card shows a green light |
| 16:14.22 | rocksfrow | no busy signal when you call the # externally, but rather a never-ending ring |
| 16:14.36 | ManxPower-work | do you see anything on the console when the call comes in? |
| 16:14.54 | Kobaz | Naikrovek: good news |
| 16:15.12 | *** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 16:15.28 | Naikrovek | Kobaz: yes? |
| 16:15.39 | rocksfrow | ManxPower-work, the console? |
| 16:15.42 | Kobaz | Naikrovek: so i had to buy some more tools, but i took out the piston |
| 16:15.50 | rocksfrow | this is a freepbx system..can i still use it? |
| 16:15.52 | Naikrovek | Kobaz: how does it look |
| 16:15.57 | Kobaz | Naikrovek: the entire cylinder looks perfect.. the piston however... has seen better days |
| 16:16.06 | ManxPower-work | rocksfrow: Ah, I forgot you are one of those people. I cannot help you firther. |
| 16:16.08 | Kobaz | which is great news... the lack of compression is surely coming by way of the piston |
| 16:16.09 | ManxPower-work | And further too. |
| 16:16.11 | *** join/#asterisk hfb (~hfb@pool-96-247-114-78.lsanca.dsl-w.verizon.net) |
| 16:16.33 | rocksfrow | ManxPower-work, one of those people? what somebody who is desperately looking for help from experts on the topic? |
| 16:16.35 | Kobaz | Naikrovek: there's a big hole in the side of the piston, and there's a nice grove going right up to the upper plate |
| 16:16.38 | rocksfrow | my customer service system is down.. |
| 16:16.43 | rocksfrow | i appreciate anybodys help. |
| 16:16.48 | cherva | How Can I limit an extension to be able to call only some numbers I want to specify |
| 16:16.50 | ManxPower-work | rocksfrow: no, someone that is into the GUI they don't even know what the console is. |
| 16:17.18 | rocksfrow | ManxPower-work, without help from somebody like yourself how will I ever? |
| 16:17.34 | Naikrovek | Kobaz: that's how it got in the crank case then |
| 16:17.35 | rocksfrow | unfortunately i dont have days to research..hence why i'm reaching out.. |
| 16:17.38 | rocksfrow | i've got a live system down. |
| 16:17.38 | ManxPower-work | ~freepbx |
| 16:17.39 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 16:17.41 | Kobaz | Naikrovek: hah, yeah |
| 16:17.42 | Naikrovek | Kobaz: that piston would have shattered soon |
| 16:17.50 | ManxPower-work | now you know why we don't support FreePBX here. |
| 16:17.52 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 16:18.08 | Kobaz | Naikrovek: it looks like someone tried to cut the piston in half with a dull knife... it's quite misshapen too |
| 16:18.09 | Naikrovek | Kobaz: aluminum piston? |
| 16:18.12 | Kobaz | i think so |
| 16:18.41 | Naikrovek | Kobaz: wow. yeah aluminum pistons are aluminum so they fail and not the cylinder wall |
| 16:18.48 | Kobaz | ah, perfect |
| 16:21.09 | *** join/#asterisk jasonwert-work (~chatzilla@adsl-99-27-170-70.dsl.klmzmi.sbcglobal.net) |
| 16:23.20 | Naikrovek | so new piston on order/installed? |
| 16:23.32 | Kobaz | grubbing for parts as we speek |
| 16:23.34 | Kobaz | speak |
| 16:25.38 | bmoraca_work | rocksfrow: congestion on a T1 could indicate that the line is in alarm. have you tried flapping the interface (either by restarting it or unplugging and plugging it back in)? |
| 16:27.23 | rocksfrow | bmoraca_work, the interface as in the telco box thing? lol |
| 16:27.45 | bmoraca_work | rocksfrow: the interface as in the actual port on your PBX |
| 16:28.02 | bmoraca_work | rocksfrow: you don't touch the "telco box thing" |
| 16:28.55 | ManxPower-work | rocksfrow: Where are you located? |
| 16:29.09 | bmoraca_work | rocksfrow: also, some things which might help are pastebins of "pri show span #" where # is your span. also, an debug of the PRI might indicate that you're either not sending or not receiving data over it. this will usually indicate that the D channel is down, in which case flapping the interface will sometimes fix it. |
| 16:29.13 | p3nguin | How can I get AgentCallbackLogin() to logout the agent without having to enter # for the new extension? Normally, it asks for a new extension, and if you press #, it says "Agent logged off." I want to hard-code something so that if you call an extension (for example *18) it will log off the agent without having to provide the # keypress. |
| 16:29.22 | ManxPower-work | bmoraca_work: he's CAS |
| 16:29.30 | bmoraca_work | ManxPower-work: ewwww. why? |
| 16:29.41 | p3nguin | AgentCallbackLogin(${AGENT},,#) doesn't do it. |
| 16:29.42 | ManxPower-work | bmoraca_work: I would not even want to speculate on that. |
| 16:30.19 | ManxPower-work | bmoraca_work: he's also a FreePBX user. |
| 16:30.39 | bmoraca_work | rocksfrow: if rebooting your system doesn't work, call the telco and make sure there isn't a line problem. i have precisely 1 hour of CAS experience with Asterisk (enough to know that it's not worth the extra channel) |
| 16:31.26 | ManxPower-work | bmoraca_work: his line was in RED alarm when he started asking an hour ago. It "magically" turned green. That's why I wanted his location, so I would know if the major outage I've been seeing applies to him. |
| 16:31.32 | ManxPower-work | He never told me his location. |
| 16:31.44 | bmoraca_work | ManxPower-work: well, Zaptel/Dahdi need to be configured manually even when using freepbx...so he's technically in the right place...although without more information, it'll be very hard to help |
| 16:31.46 | bmoraca_work | ahhh |
| 16:32.19 | ManxPower-work | bmoraca_work: "significant" Level 3 and Covad outage in NYC/NJ area |
| 16:32.32 | bmoraca_work | wow, that must suck |
| 16:32.59 | bmoraca_work | good thing i'm on the left coast |
| 16:34.13 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 16:34.22 | ManxPower-work | bmoraca_work: only 4 or 5 of our customers have reported outages this morning, so it must not be happening to everyone. |
| 16:34.23 | Naikrovek | Kobaz: you really need to update that blog, brah |
| 16:34.40 | Kobaz | Naikrovek: haha... i should write about my new small engine extravaganza |
| 16:34.49 | Naikrovek | "extravaganza" lol |
| 16:35.02 | Naikrovek | not the word i would chosen, your word is better |
| 16:35.39 | Kobaz | well it's not a new small engine... but it's a new extravaganza |
| 16:36.03 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:37.00 | Kobaz | anyways, back to coding |
| 16:37.13 | Naikrovek | have a good weekend |
| 16:37.19 | Kobaz | actually. i should write myself a check first, before i forget... i'm getting poor |
| 16:37.24 | Kobaz | you too |
| 16:39.10 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:41.55 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:42.23 | *** join/#asterisk Geminizer (~johndoe@cpe-76-180-27-4.buffalo.res.rr.com) |
| 16:45.39 | *** join/#asterisk rubberneck (~chatzilla@ext-52.sagetelecom.net) |
| 16:48.41 | *** join/#asterisk quintana (~sylvain@aghnar.doowan.net) |
| 16:49.03 | *** join/#asterisk Orbixx (Orbixx@office.exoware.net) |
| 16:49.09 | Orbixx | Are commas valid in callerids? |
| 16:49.39 | Kobaz | try it out and see |
| 16:50.19 | Orbixx | I have, but I'm unsure if there are multiple problems involved or not. |
| 16:50.26 | Kobaz | it's certainly not allowed in the numeric part... but i'm not sure what's the limit for the name part |
| 16:50.27 | Orbixx | So I'm trying to eliminate a comma being in a callerid as a problem. |
| 16:52.11 | ManxPower-work | Orbixx: yes, commas are allowed in Caller*ID Name. |
| 16:52.25 | ManxPower-work | However, Asterisk's SET statment can have issues with commas. |
| 16:53.32 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:53.33 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 16:56.24 | *** join/#asterisk etfonhomey (~etfonhome@74-143-192-74.static.insightbb.com) |
| 17:00.31 | Orbixx | Does AGI's set callerid allow for the name and number to be set? |
| 17:00.37 | Orbixx | And how does it work syntactically, if so? |
| 17:02.50 | Katty | peeks in |
| 17:03.17 | Katty | GOOD MORNING SUNSHINES |
| 17:03.50 | ManxPower-work | Orbixx: yes, see the documentation for your AGI library. |
| 17:04.03 | ManxPower-work | Robert Dobbs <666> |
| 17:04.08 | ManxPower-work | that is a valid callerid setting |
| 17:04.14 | ManxPower-work | remeber to not use quotes. |
| 17:05.02 | Katty | hugs ManxPower-work |
| 17:06.27 | *** join/#asterisk darkskiez_ (~dz@62-50-207-156.client.stsn.net) |
| 17:07.40 | *** join/#asterisk Tim_Toady (~moi@77.49.167.4.dsl.dyn.forthnet.gr) |
| 17:10.10 | *** join/#asterisk sip83 (sip83@69.196.159.201) |
| 17:13.36 | *** join/#asterisk voipmonk (~shido6@dhcp64-134-174-228.safa.lax.wayport.net) |
| 17:13.50 | sip83 | Hi.. I have a problem with a Dial command. When I use option "F" it does not continue execution of the dial plan when the source hangs up. I'm running Asterisk 1.6.0.21, is there a known bug with this option? |
| 17:14.23 | p3nguin | Maybe F isn't the right option. |
| 17:14.37 | sip83 | If, however, I use option "g" and the destination hangs up, then it continues with execution. |
| 17:14.57 | sip83 | What might be the right option? I'm trying to run a System command upon the end of a call. |
| 17:15.11 | ManxPower-work | What is the F option? |
| 17:15.24 | p3nguin | Does "core show application Dial" show the 'F' option? |
| 17:15.35 | p3nguin | I use 1.4 and it's not listed. |
| 17:15.38 | Katty | hi p3nguin |
| 17:15.40 | ManxPower-work | sip83: "g" only applies when the DESITNATION hangs up. |
| 17:15.53 | ManxPower-work | When the SOURCE hangs up, the diaplan will jump to exten => h |
| 17:15.54 | sip83 | From the documentation: "Proceed with dialplan execution at the next priority in the current extension if the source channel hangs up." |
| 17:15.55 | p3nguin | Hello, katty. |
| 17:16.14 | sip83 | Yes, but that isn't working either.. it doesn't seem to jump to h either |
| 17:16.22 | ManxPower-work | sip83: handy option. Did you get that from the CLI or fro the wiki? |
| 17:16.25 | Katty | p3nguin: i will be up in st. louis in a few weekends. |
| 17:16.43 | p3nguin | katty: On business or pleasure? |
| 17:16.49 | Katty | p3nguin: shopping. |
| 17:16.51 | sip83 | I got it from asterisk.org/docs |
| 17:17.07 | raden_work | morning katty |
| 17:17.15 | Katty | hugs raden_work |
| 17:17.28 | raden_work | how are you today ? |
| 17:17.31 | ManxPower-work | sip83: the official documentation is "core show application dial". Don't use external documentation. |
| 17:17.45 | Katty | raden_work: pretty good so far |
| 17:18.13 | *** join/#asterisk RobH (~robh@2620:0:860:2:21e:c2ff:fe03:2465) |
| 17:19.02 | ManxPower-work | Anyone know what special control 21 is? " -- Zap/22-1 requested special control 21, passing it to SIP/4407-088be1b0" |
| 17:20.07 | Kobaz | fbi tapping in? |
| 17:20.29 | p3nguin | katty: You going to hit up the Mills or the Galleria? |
| 17:21.03 | hardwire | anybody working for broadvoice in here can shut their eyes real quick |
| 17:21.10 | hardwire | BROADVOICE CUSTOMER SUPPORT SUCKS AN EGG! |
| 17:21.13 | Katty | p3nguin: definately the galleria |
| 17:21.16 | hardwire | ok.. I'm done.. you can open them again. |
| 17:21.19 | Katty | p3nguin: can't miss sephora |
| 17:22.13 | KavanS | lol |
| 17:22.37 | *** join/#asterisk chilicuil (~sistemas@unaffiliated/chilicuil) |
| 17:22.43 | ManxPower-work | Sephora sounds like an adult toy shop. |
| 17:24.23 | *** join/#asterisk mnt_real (~sinan@bas1-montreal43-1177754737.dsl.bell.ca) |
| 17:24.32 | Katty | ManxPower-work: beauty, skincare, makeup, bath products, etc |
| 17:25.22 | hardwire | ManxPower-work: sapphic erotica? |
| 17:25.26 | sip83 | ManxPower-work: Thank you... I found that the "F" option is not supported in my version. |
| 17:25.36 | *** join/#asterisk Carlos_PHX (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 17:25.48 | sip83 | However, that still leaves me with no way to continue execution of the dial plan when the caller hangs up.. |
| 17:25.58 | hardwire | g? |
| 17:26.20 | sip83 | That only works the the callee hangs up.. and in this case, I'm calling the console/dsp for a page |
| 17:26.32 | sip83 | *works when the |
| 17:26.46 | hardwire | calling it from a what? |
| 17:26.52 | *** join/#asterisk chilicuil (~sistemas@unaffiliated/chilicuil) |
| 17:26.52 | hardwire | a phone? |
| 17:26.55 | sip83 | a SIP extension |
| 17:26.56 | sip83 | yes |
| 17:27.13 | Katty | i think people who are recieving money from the goverment should be subjected to random drug testing |
| 17:27.21 | Katty | just like people working for the goverment, and recieving a paycheck that way |
| 17:27.23 | hardwire | so your sip phone calls in, dials console/dsp, and the g flag doesn't continue on? |
| 17:27.27 | hardwire | g works both ways I thought. |
| 17:27.37 | hardwire | at least it does with my calling card platform |
| 17:28.03 | sip83 | No, it only works when the destination hangs up.. It's weird, I know.. but I've tested this for the last several hours.. |
| 17:28.25 | hardwire | tried it with phone to phone? |
| 17:28.28 | sip83 | Yes |
| 17:28.54 | hardwire | ok. |
| 17:30.19 | sip83 | right now, I seem to have it working with HS(10)g |
| 17:30.45 | hardwire | is it just for paging? |
| 17:30.57 | sip83 | but this means that the caller has to either press * or if they forget, there will be a 10 second delay before the next priority is execute |
| 17:31.00 | sip83 | *executed |
| 17:31.02 | sip83 | Yes |
| 17:31.09 | hardwire | can I recommend delayed paging? |
| 17:31.20 | sip83 | What do you mean by that? |
| 17:31.45 | hardwire | paging caller calls in and is asked to record a message, press # if they accept it |
| 17:32.14 | hardwire | then a call file is made in the asterisk spool directory to do the playback. |
| 17:32.29 | hardwire | that way you don't get feedback |
| 17:32.34 | sip83 | Ahh.. yes, I've seen this.. |
| 17:32.39 | hardwire | and pages are more intelligable because people think about it first. |
| 17:32.45 | hardwire | did I spell intelligable right? |
| 17:32.47 | hardwire | hehe |
| 17:33.29 | sip83 | Hehe, yes.. I see your point.. but there would be no sync as the current setup has paging over the handsets in realtime and then the overhead would be delayed |
| 17:33.52 | hardwire | you can have it do the same for all handsets too |
| 17:34.33 | hardwire | create a meetme.. call all handsets and have them join.. do the same with the console channel.. then playback into the meetme |
| 17:34.36 | sip83 | I seem to have it working as I want.. it is just the command to unmute a channel on the mixer that is not being executed after the hangup |
| 17:34.53 | hardwire | otherwise you get a buzz? |
| 17:35.09 | sip83 | No, no buzz.. it is the music that is being played that I am muting.. |
| 17:35.26 | hardwire | because you're using your sound card for hold music? |
| 17:35.29 | sip83 | I got this solution from: http://www.ossramblings.com/overhead_paging_with_asterisk |
| 17:35.32 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 17:35.37 | sip83 | not hold music.. for overhead music |
| 17:35.43 | hardwire | ah ok |
| 17:36.20 | Katty | http://i.imgur.com/ygUwr.jpg <- oops, someone forgot their drink on their vehicle |
| 17:36.30 | hardwire | sip83: can I suggest not using console/dsp at all? |
| 17:36.38 | hardwire | and instead using pulseaudio? |
| 17:36.41 | sip83 | It works quite well.. but Asterisk neiter executes the next line nor the h extensions |
| 17:36.45 | sip83 | Hmmm.. what's that? |
| 17:37.14 | sip83 | Ahh.. is this like Jack? |
| 17:37.19 | hardwire | sip83: well do you need the overhead to be in sync with all the phones for pages? |
| 17:37.45 | sip83 | Yes, it would be nice.. |
| 17:37.55 | hardwire | then that might not work all that well |
| 17:38.31 | sip83 | Hmm.. ok, well I'll use this solution for now and just let all users know that they should press * to have the music come back on.. and if they forget, then it will come back after 10 seconds. |
| 17:38.33 | hardwire | hmm.. I have one more suggestion |
| 17:38.40 | sip83 | Oh, sure :) |
| 17:38.40 | hardwire | and it may be one you can use. |
| 17:38.53 | hardwire | are you familiar with perl/python? |
| 17:38.56 | sip83 | Yes |
| 17:38.58 | Katty | http://img705.imageshack.us/img705/3786/6a00c2252b03c78e1d00fa9.jpg <- Cheez-it flavored Lipbalm |
| 17:39.03 | Katty | ^- eww. |
| 17:39.04 | hardwire | ok both have manager API hooks |
| 17:39.29 | hardwire | if you can spawn a process that immediately mutes line-in then connects to asterisk, it can look for console/dsp being hung up |
| 17:39.43 | hardwire | on an event basis |
| 17:39.52 | hardwire | that way you can fade out line in and fade it back in cleanly |
| 17:40.10 | hardwire | and somewhat immediately |
| 17:40.21 | sip83 | Hmmm.. that sounds like a good solution :) |
| 17:40.27 | hardwire | it's friday |
| 17:40.33 | Katty | http://notepad-plus.sourceforge.net/commun/images/linux-evil.png <- Why linux is evil. |
| 17:40.34 | Qwell | hardwire: prove it |
| 17:40.45 | sip83 | I'll have to research this some more |
| 17:40.46 | hardwire | FRIDAYS ARE FOR ALL THOSE GREAT IDEAS THAT YOUR WEEKEND WILL MAKE YOU FORGET. |
| 17:40.52 | hardwire | Qwell: no |
| 17:40.59 | Katty | hardwire: orly |
| 17:41.11 | hardwire | Katty: YARLY |
| 17:41.14 | Katty | :> |
| 17:41.23 | Katty | <3 |
| 17:42.02 | Katty | i think i'm gonna close my bank of america account |
| 17:42.10 | *** join/#asterisk DerkKo (~afernande@75-149-178-131-Miami.hfc.comcastbusiness.net) |
| 17:42.15 | Katty | i'm tired of supporting their antics |
| 17:42.41 | DerkKo | Small question.... Im trying to use mixmonitor command from CLI to record a sip channel. I want the recording to be a .wav instead of .raw |
| 17:42.51 | DerkKo | what arguments to i need to send to make this happen ? |
| 17:42.56 | ManxPower-work | I don't think you can use mixmonitor from the CLI |
| 17:43.15 | ManxPower-work | Ah, nifty. That must new to 1.4 |
| 17:43.30 | Katty | DerkKo: sox? |
| 17:43.37 | p3nguin | katty: My daughter would LOVE that lip balm. She has a thing for both Cheez-Its and for lip balm, so I'm sure she would enjoy the two in combination. |
| 17:43.41 | DerkKo | you can |
| 17:43.42 | DerkKo | idr-2850-06*CLI> mixmonitor start SIP/10.20.2.198-00000029 wav,record |
| 17:43.43 | DerkKo | <PROTECTED> |
| 17:43.48 | ManxPower-work | DerkKo: "help mixmonitor" wasn't helpful to you? |
| 17:43.49 | hardwire | Katty: don't close it.. let me have it |
| 17:43.54 | Katty | p3nguin: your daughter is crazy. |
| 17:43.57 | Katty | p3nguin: JUST LIKE YOU |
| 17:44.02 | p3nguin | /: |
| 17:44.03 | hardwire | Katty: I need my to get an AlaskaAir Visa :) |
| 17:44.22 | Katty | i need an ing direct visa debit card |
| 17:44.28 | DerkKo | no help mixmonitor only shows a syntax which is even wrong |
| 17:44.29 | DerkKo | mixmonitor {start|stop} Execute a MixMonitor command |
| 17:44.34 | Katty | DerkKo: sox. |
| 17:44.50 | p3nguin | derkko: core show application MixMonitor |
| 17:44.53 | ManxPower-work | The optional arguments are passed to the |
| 17:44.53 | ManxPower-work | MixMonitor application when the 'start' command is used. |
| 17:44.56 | hardwire | Katty: pantz |
| 17:44.58 | ariel_ | Katty: so what has Bank of America done? |
| 17:45.07 | ManxPower-work | p3nguin: that will only show him the dialplan mixmonitor, not the CLI mixmonitor |
| 17:45.08 | Katty | ariel_: you mean what haven't they done? |
| 17:45.20 | Katty | ariel_: well on reddit this morning, they forclosed a house which a couple paid cash on 5 years ago |
| 17:45.23 | ariel_ | ok |
| 17:45.23 | ManxPower-work | What has B of A done that most other banks have not done? |
| 17:45.30 | Katty | ariel_: they said they had 'accidentally' forclosed the 'wrong house' |
| 17:45.30 | DerkKo | i dont want to have to execute an external application to change the .raw file to .wav |
| 17:45.38 | Katty | ariel_: but the couple had to take it to court for anything to be done about it |
| 17:45.56 | Katty | ariel_: but that's not it really |
| 17:46.18 | p3nguin | How can a bank take a house that they have no interest in? |
| 17:46.37 | ManxPower-work | p3nguin: who will stop them? |
| 17:46.51 | p3nguin | manxpower-work: Who will facilitate them? |
| 17:47.03 | ManxPower-work | p3nguin: law enforcement, if they have the paperwork |
| 17:47.05 | p3nguin | manxpower-work: That's like if I said I'm taking your car. |
| 17:47.11 | ariel_ | OH I see but that is really a general all round mix up with the attoney's, As there going to have to pay allot to this couple |
| 17:47.22 | ManxPower-work | p3nguin: You mean it's like if a finance company says they are taking my car. |
| 17:47.51 | p3nguin | manxpower-work: If the owners have paperwork too, why would law enforcement overrule the owners? |
| 17:48.00 | hardwire | sip83: yay? |
| 17:48.04 | ManxPower-work | if the finance company has paperwork that says they are going to take the car then they will take the car. |
| 17:48.16 | ManxPower-work | p3nguin: because law enforcement is not paid to think. |
| 17:48.43 | p3nguin | At least here they are smart enough to know the difference between criminal and civil matters. |
| 17:48.48 | ManxPower-work | You seem to think the law is fair or something silly like that. |
| 17:49.38 | p3nguin | In civil matters, they tell you to get an attorney and take it to court. They don't help one party just because they make a claim against another party. |
| 17:49.39 | ManxPower-work | p3nguin: Mortgage companies use the local law enforcement ALL THE TIME to evict people. |
| 17:49.43 | KavanS | p3nguin, just move the car to someone else's house |
| 17:49.50 | KavanS | p3nguin, and pay up on it... |
| 17:50.03 | KavanS | path of least resistance they will take </yoda> |
| 17:50.43 | ManxPower-work | Obviously once the law suits settled things are different. |
| 17:50.54 | p3nguin | If someone tried to take my car(s) which I clearly can show proof of outright ownership on, someone is going to be arrested for theft. |
| 17:50.56 | KavanS | it's different with cars...it's not a house |
| 17:51.05 | KavanS | I'm pretty sure law enforcement does not get involved for car retrieval |
| 17:51.08 | KavanS | I could be wrong about this... |
| 17:51.20 | Katty | http://consumerist.com/2010/01/bank-of-america-seizes-wrong-house-causes-big-stink-no-really.html |
| 17:51.21 | ManxPower-work | KavanS: Generally not, as I understand it. |
| 17:51.23 | KavanS | but I worked for a finance company that dealt with car loans...and they would repo from time to time |
| 17:51.29 | KavanS | the lady told me all sorts of stories... |
| 17:51.49 | ManxPower-work | Repo doesn't usually involve kicking people out of their car, does it? |
| 17:51.57 | KavanS | at the end of the day the lesson was: if you haven't paid up, make sure the car is nowhere near your families/routine places of frequenting/own house |
| 17:52.06 | ManxPower-work | Just "stealing it back" when the owner isn't looking. |
| 17:52.11 | KavanS | exactly... |
| 17:52.14 | KavanS | just stealing it back |
| 17:52.19 | p3nguin | In a repo, the cops usually only get involved if there is a crinimal act such as disorderly conduct or assult. |
| 17:52.26 | KavanS | but with the law on your side I suppose ;) |
| 17:52.40 | p3nguin | or worse, of course. |
| 17:52.53 | KavanS | yeah, no car no repo |
| 17:53.47 | ManxPower-work | We had a local Sherif that, at least for a while, refused to remove people from a house that was foreclosed on, if the person's living there were renters and current on rent. |
| 17:54.06 | ManxPower-work | (i.e. the renters had nothing to do with the owner of the property not making payments) |
| 17:54.11 | spenguin[w0rk] | hi :) |
| 17:54.18 | Kobaz | A Bank of America spokesman told the paper that the bank feels the lawsuit "has no merit." |
| 17:54.21 | Kobaz | hah |
| 17:54.23 | ariel_ | house eviction requires a court order |
| 17:54.26 | Kobaz | that's messed up |
| 17:56.36 | p3nguin | ariel_: And if the court order is not obeyed, then it becomes a criminal situation and law enforcement has a right to get involved. |
| 17:59.10 | raden_work | heya p3nguin |
| 18:00.21 | *** join/#asterisk guax (~guax@unaffiliated/guaxinim) |
| 18:00.23 | guax | !book |
| 18:00.25 | guax | oops |
| 18:00.27 | guax | ~book |
| 18:00.28 | 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 |
| 18:00.46 | *** join/#asterisk albertoandrade (~albertoan@201.22.11.186.dynamic.adsl.gvt.net.br) |
| 18:09.19 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 18:09.40 | Katty | bored. |
| 18:09.53 | Katty | so bored maybe i'll chatroulette |
| 18:12.32 | guax | uia, someone is going wild here |
| 18:13.13 | carrar | Sounds risky |
| 18:14.50 | *** join/#asterisk titter (~titter@c-98-208-158-125.hsd1.fl.comcast.net) |
| 18:15.01 | Geminizer | Hello all. Does running the following do anything: asterisk -rx 'restart now' ? |
| 18:15.03 | *** join/#asterisk drako (~luisjose@nelug/coreteam/luisjose) |
| 18:15.28 | p3nguin | geminizer: I would expect it to restart asterisk right now. |
| 18:15.37 | rubberneck | Geminizer: yes it restarts the asterisk service right now |
| 18:16.02 | Geminizer | hmm... then I would expect all dialplans tied into asterisk would reload as well ? |
| 18:16.14 | rubberneck | Geminizer: Yes it stops the service |
| 18:16.22 | p3nguin | geminizer: Right, but you don't need to restart it to reload the dialplan. |
| 18:16.27 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:16.37 | Geminizer | right... I did 'dialplan reload' |
| 18:16.52 | rubberneck | Geminizer: Youdont really want to run restart now while calls are in progress. |
| 18:17.04 | Orbixx | Does anybody know the syntax for the AGI command 'set callerid' to set the name AND number? |
| 18:17.13 | p3nguin | I always prefer restart gracefully. |
| 18:17.36 | Geminizer | rubberneck: I completely agree... working on a development machine :) |
| 18:18.17 | p3nguin | orbixx: In dialplan it's Set(CALLERID(all)=name <12345>). Does that help any in AGI? |
| 18:18.38 | Orbixx | p3nguin: AGI seems to just want SET CALLERID <number> |
| 18:18.56 | Orbixx | I can find no more documentation than that. |
| 18:19.17 | carrar | http://search.cpan.org/~jamesgol/asterisk-perl-1.01/lib/Asterisk/AGI.pm |
| 18:19.27 | Geminizer | Or even better: http://www.voip-info.org/wiki/view/Asterisk+AGI |
| 18:19.40 | Geminizer | there is agi_callerid and agi_calleridname |
| 18:19.57 | Orbixx | ARGH |
| 18:20.00 | Orbixx | How did I miss that! |
| 18:20.05 | Orbixx | Thanks Geminizer. |
| 18:20.15 | Geminizer | It's no problem :) |
| 18:20.19 | Orbixx | Wait. |
| 18:20.33 | Orbixx | Geminizer: That's not quite right. |
| 18:20.42 | Orbixx | They're variables passed to AGI scripts, not the other way round. |
| 18:20.48 | Orbixx | Scroll a little further down to 'AGI commands'. |
| 18:21.03 | guax | the best way to set is to set the variable SET VARIABLE CALLERID(all)="", this will set everything. |
| 18:21.18 | *** join/#asterisk Alagar (~Administr@122.164.33.82) |
| 18:21.20 | guax | it work for me in a rather complex agi |
| 18:21.34 | *** part/#asterisk ManxPower-work (~EWieling@216.186.151.147) |
| 18:22.28 | Orbixx | guax: It needs to be in AGI, but you have nevertheless given me an idea. |
| 18:22.35 | guax | the agi_callerid and agi_calleridname are params passed to the agi, just a boot up on environment its running |
| 18:22.44 | guax | Orbixx, it is in agi |
| 18:23.06 | Orbixx | guax: Oh sorry, I misinterpreted you. |
| 18:23.28 | guax | just use the set variable in agi as you do in dialplan |
| 18:23.33 | guax | it support funcions like callerid |
| 18:23.37 | guax | functions* |
| 18:23.39 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 18:29.11 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
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| 18:36.05 | *** part/#asterisk rue_house (~rue@24.207.119.38) |
| 19:06.28 | *** join/#asterisk Amorsen (~Amorsen@94.127.50.7) |
| 19:16.37 | *** join/#asterisk autojack (~owen@216.93.177.252) |
| 19:17.54 | autojack | can anyone recommend a good "bring your own device," "pay as you go" VOIP termination provider in the US? I currently use callwithus.com, and overall they're great, but I have persistent issues with one-way-audio that we've been unable to resolve and I wanted to see if switching carriers eliminated them. |
| 19:18.49 | autojack | I have a full Asterisk server with a DID that routes to it, and then I pass calls to callwithus via SIP. |
| 19:22.56 | Kobaz | ~itsp-us |
| 19:23.08 | Kobaz | ~itsp-usa |
| 19:23.12 | Kobaz | ~itsp |
| 19:23.13 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 19:23.20 | Kobaz | ~itsplist-us |
| 19:23.21 | infobot | itsplist-us is, like, Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net, or http://flowroute.com , http://voip.ms |
| 19:23.59 | autojack | word. |
| 19:24.47 | *** join/#asterisk Badrobot- (~badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 19:24.55 | rubberneck | ~itsplist-us |
| 19:24.56 | infobot | itsplist-us is probably Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net, or http://flowroute.com , http://voip.ms |
| 19:24.57 | autojack | the nice thing about callwithus is that there's no signup or cancellation fee and no monthly fee. I just pay for usage. |
| 19:25.15 | p3nguin | autojack: VoIP.ms or Flowroute |
| 19:26.14 | *** join/#asterisk frek818 (~herman@cpe-98-154-78-151.socal.res.rr.com) |
| 19:34.37 | *** join/#asterisk socain (~socain00@74.255.249.66) |
| 19:36.09 | autojack | cool. thanks :) |
| 19:36.33 | socain | is it possible to port a DID from a long distance location to your PRI? If so do they typically just bill you for the long distance? |
| 19:38.43 | *** part/#asterisk moldy (~rene@unaffiliated/moldy) |
| 19:39.15 | autojack | hmm. on this rates page, what is meant by "first interval" and "sub interval" http://www.flowroute.com/services/rates/ |
| 19:47.11 | benngard | SWEDEN - FREEPHONE 4620 30 sec 6 sec 0.1587 Current <- what the hell is SWEDEN - FREEPHONE, i am from sweden and i never heard about ot |
| 19:47.31 | *** join/#asterisk Tech_Travis (~Travis@mail.techglia.com) |
| 19:47.54 | autojack | I think it means free-dial numbers. |
| 19:48.00 | autojack | in the US they are 1-800 numbers |
| 19:48.10 | autojack | like the kind of thing businesses have so you can call them for no charge. |
| 19:49.53 | benngard | do they realy "route" free-dial" number outside your country? |
| 19:50.28 | autojack | they say they do. |
| 19:50.28 | *** join/#asterisk Z_God (~julius@schwartzenberg.xs4all.nl) |
| 19:50.29 | p3nguin | autojack: It means that if you make a 10 second call, you will be billed for 30 seconds. If you make a 34 second call, you will be billed for 36 seconds. |
| 19:50.46 | autojack | p3nguin: aha, OK that makes sense. thanks :) |
| 19:50.59 | autojack | I DEMAND PER-SECOND BILLING |
| 19:51.01 | autojack | :) |
| 19:51.05 | Z_God | I'm trying out the jingle channel, but it seems asterisk isn't even noticing the incoming phonecalls |
| 19:51.06 | benngard | we have 020- numbers in sweden and i am pretty sure u cant dial them from for example us |
| 19:51.06 | p3nguin | Most numbers in the US are 6 second intervals. |
| 19:51.09 | autojack | flowroute looks like they're worth a try. |
| 19:51.16 | p3nguin | So if you make a 10 second call, you would pay for 12 seconds. |
| 19:51.23 | Z_God | how can I verify whether asterisk is seeing a phonecall? |
| 19:51.27 | autojack | benngard: normally you would not be able to, I think. |
| 19:51.32 | Z_God | I do see the incoming xml |
| 19:51.44 | p3nguin | benngard: I can call you. |
| 19:51.48 | autojack | benngard: but if your VOIP provider has a termination gateway in sweden, it should cost them nothing to route to that number. |
| 19:51.57 | autojack | at least, nothing on the PSTN side. |
| 19:52.13 | benngard | autojack: got u! |
| 19:52.31 | autojack | that's just a guess, I'm not really a tel-networking guy :) |
| 19:53.10 | benngard | guess u are right when i am thinking about it |
| 19:54.03 | *** join/#asterisk bawls12342 (~mike@brndmb0239w-ds01-54-57.dynamic.mts.net) |
| 19:54.16 | benngard | but how do u dial that 020 number? +46-20-... yea, must be the way to do it |
| 19:54.30 | autojack | most likely. |
| 19:54.50 | p3nguin | z_god: Is there a debug setting for the channel driver? |
| 19:55.05 | Z_God | p3nguin: yes |
| 19:55.08 | p3nguin | benngard: 0114620XXXXXXXX |
| 19:55.25 | Z_God | p3nguin: when I disable it the xml is gone |
| 19:55.36 | p3nguin | benngard: That's how I would dial it. |
| 19:55.50 | sier | hi |
| 19:55.57 | Z_God | don't channels report 'calls' to asterisk's core? |
| 19:56.00 | p3nguin | benngard: My ITSP allows both 011 and 00 prefixes for international calling. |
| 19:56.28 | p3nguin | z_god: For SIP, you would "sip set debug" and watch what happens. |
| 19:56.50 | Z_God | ah yeah I looked at jabber now, I guess I should look at jingle |
| 19:57.41 | spenguin[w0rk] | how sucessful are dundi networks? |
| 19:57.48 | Z_God | p3nguin: jingle set debug doesn't seem to exist |
| 19:58.37 | *** part/#asterisk autojack (~owen@216.93.177.252) |
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| 20:00.16 | benngard | p3nguin: we must try that, not tonight,i am to tired, but if i let u "route" a swedish 020 number through my *, i terminate the call... will not cost either u or me a cent :)' |
| 20:01.25 | p3nguin | Hmm. I don't have anyone in Sweden that I need to call. I was just saying how I would call your number if I needed to call you. |
| 20:02.13 | benngard | nobody likes us swedes :( |
| 20:03.09 | *** join/#asterisk mrprozac (~mrprozac@62.59.46.85) |
| 20:06.21 | bawls12342 | I have just setup an asterisknow server using the live cd. I can't get any sip phones to register, I do not even see a sip.conf file in /etc/asterisk. Any help would be great |
| 20:07.29 | benngard | ~asterisknow |
| 20:07.30 | infobot | i guess asterisknow is based on Asterisk, but is difficult to support in #asterisk for a number of reasons. Please seek support in #asterisknow instead. |
| 20:07.39 | carrar | benngard, not true!! ABBA is from Sweden! |
| 20:07.52 | *** join/#asterisk niekie (quasselcor@CAcert/Assurer/niekie) |
| 20:07.53 | Qwell | carrar: nobody likes abba |
| 20:08.05 | *** join/#asterisk drako (~luisjose@nelug/coreteam/luisjose) |
| 20:08.19 | benngard | and björn borg not forget about peter foppa forsberg! |
| 20:08.20 | carrar | Though the band 'Europe' is from Sweden so that counter acts ABBA |
| 20:08.27 | carrar | heh |
| 20:09.32 | *** join/#asterisk moldy (~rene@unaffiliated/moldy) |
| 20:10.09 | benngard | http://www.youtube.com/watch?v=7_IKcMl_a9A |
| 20:11.17 | *** join/#asterisk voipmonk (~shido6@216.217.58.154) |
| 20:11.19 | *** join/#asterisk bawls12342 (~mike@brndmb0239w-ds01-54-57.dynamic.mts.net) |
| 20:11.47 | moldy | hi |
| 20:13.16 | bawls12342 | can someone please tell me how to tell is asterisk is accepting sip connections? |
| 20:13.34 | voipmonk | you can look at the sip debug info bawls12342 |
| 20:13.37 | voipmonk | :) |
| 20:13.47 | voipmonk | then work the problem from what asterik is telling you :) |
| 20:14.26 | moldy | hmm, it seems that for the siemens gigaset IP phones, you cannot buy separate additional phones? can anyone confirm this? |
| 20:15.04 | benngard | in sweden u can for sure |
| 20:15.29 | benngard | i do it every month |
| 20:15.33 | moldy | benngard: for which model? |
| 20:15.44 | benngard | doesnt matter |
| 20:16.08 | moldy | benngard: do the additional handsets have their own model number (this seems to be the case for the analog phones)? |
| 20:16.25 | moldy | e.g. "c47h" for "c470" |
| 20:16.35 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 20:16.49 | benngard | nop, it they will be the sam |
| 20:16.54 | benngard | same |
| 20:17.25 | benngard | but ofc u have to go into "service mode" and do some tricks |
| 20:17.32 | moldy | ofc? |
| 20:17.48 | p3nguin | ufc is better! |
| 20:18.07 | moldy | i don't know what ofc or ufc mean |
| 20:18.27 | moldy | google turns up stuff for model numbers like "a58h ip"... those seem to be the additional handsets |
| 20:19.08 | benngard | well known secret about siemens gigaset is: trun phone off! press 1 4 7 and poer it up |
| 20:19.14 | benngard | power* |
| 20:19.17 | moldy | though, interestingly, i don't find any german stores selling it |
| 20:20.12 | benngard | dial let me think 7200 or 76200 then u hit the service menu and can do alot of fun stuff |
| 20:20.36 | benngard | i fire up the program sec |
| 20:21.46 | benngard | it was 76200 |
| 20:21.54 | Geminizer | has anyone ever had this problem: I have an AGI script being called in a dialplan, followed by a busy signal ... when I call into that dialplan, I immediately get the busy signal (no ringing). However, when I modify the AGI script to include more agi functions, calling into the corresponding dialplan results in a couple of rings, followed by a busy signal. It turns out the second version of the agi script is non-functional. And |
| 20:21.55 | Geminizer | the calls are right, because it's literally a copy and paste of the first set of AGI calls (e.g. $AGI->stream_file(...)) |
| 20:22.09 | moldy | it seems that they don't sell additional handsets in germany :( |
| 20:22.17 | benngard | :( |
| 20:23.40 | spenguin[w0rk] | does dundi basically mean free calls? |
| 20:24.29 | moldy | ah, the a58h is compatible with a580 ip... |
| 20:24.44 | *** join/#asterisk krefik (krefik@arm.generacja.pl) |
| 20:24.50 | krefik | hi |
| 20:29.22 | Geminizer | is there a way an agi script can return values to be used by the dialplan that called said script? |
| 20:33.36 | mrprozac | has got bad timing |
| 20:34.05 | mrprozac | so i just got my2 Linksys SPA942 phones by mail order. Perfect time for the server to crash, and not being on the same location as the server totally makes my day. ;_; |
| 20:34.57 | *** join/#asterisk MichaelGG (~mgg@197.164.148.190.dsl.intelnet.net.gt) |
| 20:36.36 | Geminizer | ~book |
| 20:36.37 | 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 |
| 20:38.23 | spenguin[w0rk] | mrprozac: how much for those phones? |
| 20:40.39 | MichaelGG | So, how can I force override the codecs offered on an outbound leg? Like user calls in with codec g729, we connect to a peer with ulaw and g729, peer selects ulaw, we're stuck transcoding |
| 20:40.56 | MichaelGG | I saw some reference to SIP_CODEC or PREFERRED_CODEC but are those the right things? |
| 20:42.37 | mrprozac | 94 a piece |
| 20:43.10 | mrprozac | was on the phhone with technical staff, it's back online again. yay i can't test again :) |
| 20:43.28 | spenguin[w0rk] | mrprozac: thats pretty expensive |
| 20:44.14 | mrprozac | nah, the cheapest price i could find wsas 93 Tax/vat not included |
| 20:44.41 | mrprozac | i bought it together with a Cisco 8 ports Switch with PoE |
| 20:50.33 | *** join/#asterisk fibres (~no@cpc2-nfds1-0-0-cust1021.lei3.cable.ntl.com) |
| 20:55.00 | MichaelGG | or, how can I know the codecs offered on an incoming invite? |
| 20:59.09 | *** join/#asterisk Failrar (~Failrar@5ED66E6D.cable.ziggo.nl) |
| 21:02.38 | *** join/#asterisk gnufan (~hardev@76.91.81.190) |
| 21:03.33 | sbrath | If I want a station, who's a member of a queue, and who's on the phone, and everyone else in the queue is busy, can I ring the station even when the line is busy. |
| 21:04.25 | sbrath | Basicly asking the station to see that their is another call, and give them a chance to grab it, but if they don't grab it, it goes back into the queue and waits? |
| 21:04.37 | gnufan | i'm trying to implement g729 codec into my asterisk box... do i need to have IP phones that compatible w/ g729 codec? |
| 21:04.41 | Katty | let's go to quiznos!!!! |
| 21:04.48 | Katty | WHO"S WITH ME |
| 21:05.07 | sbrath | Or should I just set up two accounts on each phone, and make the first one a login/out member of the queue, and the second line a perm member of a backup-queue? |
| 21:05.09 | Qwell | Katty: quiznos? eww |
| 21:05.16 | Katty | Qwell: :P |
| 21:05.33 | sbrath | Fat Sandwitch !! |
| 21:05.54 | Katty | yeah cause i'm so totally fat. |
| 21:05.55 | ariel_ | it's Sushi night for me.... |
| 21:06.15 | gnufan | i'm trying to implement g729 codec into my asterisk box... do i need to have IP phones that compatible w/ g729 codec? |
| 21:06.29 | ariel_ | no |
| 21:06.57 | gnufan | so any IP phone will work g729 codec? |
| 21:08.03 | gnufan | I know Speex needs compatible phones, wasn't sure about g729. |
| 21:09.01 | gnufan | ? |
| 21:12.02 | staffmember | anyone know why i have no sip options from the CLI under asterisknow |
| 21:12.05 | staffmember | its a fresh install |
| 21:12.09 | staffmember | but the CLI has nothing for sip |
| 21:13.43 | socain | asterisk restart now will probably bring sip back with astnow |
| 21:14.00 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 21:14.14 | socain | that happened to me on the first astnow launch |
| 21:14.16 | *** join/#asterisk Gugge (~gugge@vlan2.dlxhosting.dk) |
| 21:14.46 | staffmember | asterisk -r | reload? |
| 21:14.55 | socain | no reload. restart now |
| 21:14.55 | staffmember | or should i reboot the entire system |
| 21:15.01 | staffmember | okay |
| 21:15.01 | staffmember | hold |
| 21:15.35 | socain | asterisk -vvvvvvvcr will get you back into the cli |
| 21:15.59 | *** join/#asterisk titter (~titter@c-98-208-158-125.hsd1.fl.comcast.net) |
| 21:17.05 | staffmember | yeah, sip still now loaded tho |
| 21:17.13 | staffmember | i dont get why this would happen off of a fresh install |
| 21:18.03 | socain | it must start asterisk before somethign has initialized on chan_sip or something |
| 21:18.33 | socain | maybe it does take a reboot. don't really remember, but i had the same thing.... |
| 21:18.34 | *** part/#asterisk gnufan (~hardev@76.91.81.190) |
| 21:21.53 | *** join/#asterisk voipmonk (~shido6@216.217.58.154) |
| 21:24.06 | Katty | omnomnomnoms lunch |
| 21:27.16 | socain | any luck staffmember? i'm closing shop... |
| 21:28.49 | Z_God | p3nguin: it seems the namespaces in the jingle.h header were outdated, I've got at least some negotiation going on now :) |
| 21:29.25 | Katty | is anyone else near St. Louis, MO? |
| 21:29.34 | Katty | just curious. |
| 21:32.44 | raden_work | bmoraca_work, ? |
| 21:32.47 | *** join/#asterisk halstead|wccls (~halstead_@208.71.200.68) |
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| 21:36.43 | *** part/#asterisk Tech_Travis (~Travis@mail.techglia.com) |
| 21:38.32 | *** join/#asterisk ManxPower-work (~EWieling@216.186.151.147) |
| 21:41.28 | leifmadsen | Katty: all the people who live in St. Louis likely are |
| 21:42.48 | *** join/#asterisk MichaelGG_ (~mgg@128.1.148.190.dsl.intelnet.net.gt) |
| 21:47.58 | *** join/#asterisk usn (~usn@p5B09687F.dip.t-dialin.net) |
| 21:49.10 | *** join/#asterisk ChrisWi (~admin@mx2.wwserver.net) |
| 21:49.35 | usn | Hi, I'm about to migrate my asterisk from 1.2 to 1.4. Have found and detonated several mines, but now I'm stalled. Starting the server with "asterisk -U asterisk -G dialout -vvvf" and I get: |
| 21:49.43 | *** join/#asterisk Geminizer (~johndoe@cpe-76-180-27-4.buffalo.res.rr.com) |
| 21:49.46 | *** join/#asterisk smooth_penguin (~smoove@59.162.86.164) |
| 21:49.51 | usn | [Feb 12 22:41:16] WARNING[18366]: pbx.c:2981 ast_register_application: Already have an application 'Directory' |
| 21:49.55 | Geminizer | hey guys... how do I drop a call using the CLI ? |
| 21:50.28 | usn | Can somebody point me to the right direction, please? |
| 21:52.00 | paulc | usn: did you build from source? |
| 21:52.17 | usn | No, used the 1.4.21 from debian lenny |
| 21:53.05 | *** join/#asterisk cnu (cnu@161.80-203-43.nextgentel.com) |
| 21:54.42 | usn | The last message before the error mentioned above is "app_zapateller.so => (Block Telemarketers with Special Information Tone)" - I'm using CAPI, not ZAP |
| 21:56.31 | russellb | ls /usr/lib/modules/*directory* |
| 21:56.49 | russellb | they include multiple versions of the module, you need to edit /etc/asterisk/modules.conf to only load the one you want |
| 21:57.16 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 21:57.52 | usn | the CAPI module is loaded without problems |
| 21:57.59 | leifmadsen | russellb means: /usr/lib/asterisk/modules/*directory* |
| 21:58.13 | russellb | leifmadsen: yes. |
| 21:58.36 | russellb | that warning is completely unrelated to CAPI, zap, etc. |
| 21:58.42 | usn | okay |
| 21:59.00 | Geminizer | is there a "hard hangup" (as opposed to 'soft hangup') ? |
| 21:59.38 | usn | /usr/lib/asterisk/modules contains a lot of libs, but what should I expect to see there? |
| 22:00.00 | leifmadsen | usn: hence the: ls /usr/lib/asterisk/modules/*directory* |
| 22:00.08 | leifmadsen | usn: which would only show the app_directory.so file |
| 22:00.28 | usn | # ll /usr/lib/asterisk/modules/*directory* |
| 22:00.28 | usn | -rw-r--r-- 1 root root 21112 14. Dez 20:43 /usr/lib/asterisk/modules/app_directory_odbc.so |
| 22:00.28 | usn | -rw-r--r-- 1 root root 16720 14. Dez 20:43 /usr/lib/asterisk/modules/app_directory.so |
| 22:00.38 | leifmadsen | honestly, it sounds like you should have deleted the /usr/lib/asterisk/modules/* contents prior to re-installing asterisk |
| 22:00.43 | leifmadsen | usn: that's the exactly problem |
| 22:01.11 | leifmadsen | usn: you have two modules that are conflicting. Use modules.conf to disable one of them (likely the ODBC one since you're not going to be using the database connectivity I presume) |
| 22:01.13 | usn | thanks for your help, but I still don't get it in the whole I'm afraid |
| 22:01.27 | *** join/#asterisk SaiSoma (~SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 22:01.30 | usn | ok |
| 22:02.18 | leifmadsen | app_directory_odbc.so is for database connectivity of the app_directory application. You also have the non-ODBC compiled version, and you should only load one of them. That's why you're getting the conflict, because you've loaded one of them, and when the other tries to load, Asterisk says, "oops, app_directory has already been loaded into memory" |
| 22:02.26 | usn | no string like directory is somewhere in /etc/asterisk |
| 22:02.38 | leifmadsen | shakes his head |
| 22:02.41 | leifmadsen | *modules.conf* |
| 22:02.57 | leifmadsen | look at the formatting -- you need to disable a module from loading |
| 22:03.09 | leifmadsen | if you look in modules.conf, you'll see an example of how to not load a module. |
| 22:03.17 | usn | omg ;) |
| 22:03.24 | usn | It was a long day ;) |
| 22:03.28 | *** part/#asterisk voipmonk (~shido6@216.217.58.154) |
| 22:04.18 | *** join/#asterisk xpot-mobile (~xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 22:05.06 | usn | leifmadsen, russellb - thanks a lot, not at least for your patience. |
| 22:06.17 | leifmadsen | that's why I almost always just rm -f /usr/lib/asterisk/modules/* before I do 'make install' |
| 22:06.27 | leifmadsen | less problems with module compatibility that way |
| 22:07.14 | usn | my trouble is, the last version was self built, but I am using a prebuilt version now |
| 22:14.49 | *** join/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
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| 22:22.23 | raden_work | <PROTECTED> |
| 22:22.35 | raden_work | Depends on: mysqlclient(E) |
| 22:22.58 | leifmadsen | you're missing some libraries |
| 22:23.04 | raden_work | Name: mysql-client |
| 22:23.04 | raden_work | Version: 5.1.36-6.7.2 |
| 22:23.05 | ManxPower-work | I would have said depends on mysql-devel |
| 22:23.19 | ManxPower-work | or mysql-dev, I don't remember which. |
| 22:23.24 | leifmadsen | ManxPower-work: or mysqlclient devel |
| 22:23.46 | leifmadsen | ManxPower-work: depends on which flavour of Linux (RedHat tends to be -devel, Ubuntu tends to be -dev) |
| 22:23.46 | raden_work | leifmadsen, thats installed as well, was first thing i did |
| 22:24.01 | ManxPower-work | or even mysql-devel-5.0.77-4.el5_4.1 |
| 22:24.13 | leifmadsen | ManxPower-work: for a specific version, yes :) |
| 22:24.31 | leifmadsen | raden_work: I'd check the configure.log file (I think that's what it's called) to see what its expecting |
| 22:24.39 | raden_work | where ? |
| 22:24.49 | *** join/#asterisk bmg505 (~leon@196-209-79-222-rndf-esr-5.dynamic.isadsl.co.za) |
| 22:24.56 | leifmadsen | in your asterisk source |
| 22:25.30 | *** join/#asterisk fifer (~fifer@67.208.108.228) |
| 22:25.35 | raden_work | asterisk addon source ? |
| 22:25.43 | fifer | Any 480i users here? |
| 22:25.55 | leifmadsen | raden_work: whichever directory you're running ./configure from |
| 22:26.35 | *** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net) |
| 22:26.55 | raden_work | leifmadsen, was libmysqld-devel |
| 22:27.09 | leifmadsen | always depends on which flavour of linux it is :) |
| 22:27.35 | fifer | Trying to track down some old Aastra 480i firmware files, I need to upgrade a phone from 1.0.0.78 to the newest but it looks like I need some older files to bridge the gap |
| 22:27.36 | raden_work | what does ,/configure do anyway i still dont get the whole process ? |
| 22:27.41 | fifer | Already talked to Aastra, no help |
| 22:32.01 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 22:36.05 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 22:37.20 | fifer | I'm looking for a very cheap or even free source for a sip or iax trunk to do some testing with on a new system. Any advice? |
| 22:38.31 | leifmadsen | ipkall |
| 22:38.40 | fifer | thanks! |
| 22:38.46 | leifmadsen | raden_work: it basically searches for libraries on your computer so asterisk now what modules it can compile |
| 22:38.58 | raden_work | gotcha |
| 22:39.10 | *** join/#asterisk drako (~luisjose@nelug/coreteam/luisjose) |
| 22:39.40 | raden_work | Feels like 90 degrees in here :( |
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| 23:00.36 | *** join/#asterisk thansen (~thansen@c-76-27-110-194.hsd1.ut.comcast.net) |
| 23:00.58 | thansen | I have an agi script that keeps spewing this channel.c:2480 ast_waitfordigit_full: write() failed: Resource temporarily unavailable |
| 23:01.42 | thansen | it does it during get option and some other weird things...anyone have some pointers? |
| 23:03.08 | *** join/#asterisk TSM2 (~the_softw@87-194-32-212.bethere.co.uk) |
| 23:03.26 | thansen | this one too... file.c:1292 waitstream_core: write() failed: Resource temporarily unavailable |
| 23:03.42 | ManxPower-work | I've only ever seen that if the caller hangs up. |
| 23:04.04 | leifmadsen | ya, I see that lots when a file is playing and someone hangs up |
| 23:04.54 | ManxPower-work | thansen: Is your AGI catching the signals Asterisk sends when the channel hangs up (I don't recall if it's SIGHUP or SIGTERM, look it up if you need to) |
| 23:07.43 | thansen | ManxPower-work: sorry, had a quick call... |
| 23:07.57 | thansen | ok, well I'm just doing IVR stuff at this point |
| 23:08.14 | thansen | so I'm definately still on the line cause I can hear playback etc |
| 23:10.28 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 23:11.46 | thansen | http://pastebin.ca/1794860 |
| 23:12.07 | thansen | about halft way through the readback it start spitting that crap out |
| 23:23.36 | *** join/#asterisk lesouvage (~lesouvage@82.73.69.76) |
| 23:25.07 | *** join/#asterisk xpot-mobile (~xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
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| 23:27.02 | *** part/#asterisk fibres (~no@cpc2-nfds1-0-0-cust1021.lei3.cable.ntl.com) |
| 23:28.01 | lesouvage | I have just build dahdi from dahdi-linux-complete-2.2.1+2.2.1.tar.gz but if I run make menuselect to configure the build of Asterisk picking app_meetme from the list is not possible. dahdi and dahdi_dymmy are loaded. Any suggestion? |
| 23:32.16 | Z_God | how can I allow the speex codec for jingle? |
| 23:32.26 | Z_God | I added allow=speex, but it's not being advertised |
| 23:34.30 | Z_God | I should use a different subversion branch :S |
| 23:34.57 | leifmadsen | lesouvage: re-run ./configure in asterisk after installing dahdi |
| 23:35.12 | leifmadsen | Z_God: the codec_speex is built right? |
| 23:36.23 | Z_God | leifmadsen: yep |
| 23:36.54 | lesouvage | leifmadsen: thanks, I knew I forgot something obvious. |
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| 23:41.23 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
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| 23:52.03 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 23:53.27 | Z_God | getting a segfault from the jingle branch! seems it breaks with ipv6 addresses .... |
| 23:56.37 | *** join/#asterisk DocAwesome (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:56.38 | *** mode/#asterisk [+o DocAwesome] by ChanServ |
| 23:57.35 | jaytee | everytime I see that nick I think of Doc Octopus from Spiderman for some strange reason. |
