| 00:04.31 | *** join/#asterisk sasargen (n=chatzill@68.245.248.89) |
| 00:07.18 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 00:07.47 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 00:12.27 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-2f8d81f2b8350d23) |
| 00:17.41 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 00:22.04 | *** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri) |
| 00:22.43 | *** join/#asterisk drumkilla (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 00:22.43 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 00:25.56 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.4) |
| 00:29.19 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 00:30.01 | *** join/#asterisk watchy (n=watchy@76.196.98.139) |
| 00:30.09 | watchy | do they make ant analog devices that auto answer? |
| 00:32.39 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 00:42.29 | *** part/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
| 00:42.38 | *** join/#asterisk gambolputty (n=BC43599@cpe-76-186-231-222.tx.res.rr.com) |
| 00:46.19 | *** join/#asterisk Segnale007 (n=Pietro@host88-253-dynamic.42-79-r.retail.telecomitalia.it) |
| 00:51.52 | *** join/#asterisk andresmujica (n=andresmu@201.244.108.160) |
| 00:53.33 | *** join/#asterisk Firass-VC22 (n=firass@rza.vikcomm.wwu.edu) |
| 00:59.08 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 01:04.05 | *** join/#asterisk andresmujica (n=andresmu@201.244.108.160) |
| 01:05.10 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:07.23 | *** join/#asterisk chendy (n=chatzill@58.60.97.211) |
| 01:08.01 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-f12cddb10b1414d2) |
| 01:10.28 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 01:17.10 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:26.54 | drmessano^ | I have no spare horses |
| 01:33.58 | *** join/#asterisk [netman] (n=netman@166.Red-88-23-82.staticIP.rima-tde.net) |
| 01:36.11 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:41.43 | *** join/#asterisk etfonhomey (n=chatzill@74-143-196-254.static.insightbb.com) |
| 01:44.08 | *** join/#asterisk tengulre (n=tengulre@124.42.50.9) |
| 01:44.09 | *** join/#asterisk RB2 (n=RB2@pool-71-251-41-182.nwrknj.east.verizon.net) |
| 01:46.54 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 01:48.46 | *** join/#asterisk grx0 (n=dave@64-150-178-3.kansascity.abac.net) |
| 01:49.31 | *** join/#asterisk harry_v (n=pcsuppor@S010600a0c93f6f7e.vs.shawcable.net) |
| 01:50.20 | *** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-219-98.phlapa.east.verizon.net) |
| 01:57.26 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 01:59.42 | *** join/#asterisk jtodd (n=jtodd@249.sub-70-222-193.myvzw.com) |
| 02:02.41 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 02:06.56 | *** join/#asterisk Spirits-Sight (n=christop@c-71-192-91-123.hsd1.ma.comcast.net) |
| 02:08.57 | *** join/#asterisk Coolthreads (n=Coolthre@203.97.238.71) |
| 02:11.47 | Spirits-Sight | Does any one know of a service that allows SMS / TXT msg to be sent through their service? |
| 02:12.03 | Spirits-Sight | besides a cell company |
| 02:14.47 | tengulre | how to play the pcm audio format? |
| 02:15.25 | tengulre | I have a 001.pcm in path /root, how to using PlayBack or BackGround to play it? |
| 02:16.03 | *** join/#asterisk khronos (n=khronos@aquaman.perryinstitute.org) |
| 02:16.06 | khronos | Hi guys. |
| 02:16.24 | khronos | Anybody know how to reset a linksys spa942 to factory defaults remotely? |
| 02:19.07 | Nugget | dunno, but it would be funny if +++ATH0+++ did it. |
| 02:20.28 | *** join/#asterisk illizit (n=hi@c-71-206-66-218.hsd1.fl.comcast.net) |
| 02:21.42 | illizit | hello |
| 02:22.17 | illizit | I'm having an issue with asterisk 1.6, im trying to use SIP over tcp in order to communicate with another PBX |
| 02:22.30 | illizit | it seems asterisk is sending a blank TO field in the SIP packets |
| 02:22.39 | Nugget | clever |
| 02:22.41 | illizit | is this a bug with sip /tcp? |
| 02:22.59 | Nugget | dunno, I set up sip/tcp last weekend to some cisco 7965s and haven't seen that issue |
| 02:23.09 | [TK]D-Fender | illizit: Show us your call attempt at verbose 10, SIP debug enabled |
| 02:23.15 | [TK]D-Fender | ~pb |
| 02:23.16 | jbot | [~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://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 02:23.18 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^ |
| 02:23.42 | illizit | im running 1.6.0.3-rc1 |
| 02:23.46 | illizit | ok |
| 02:26.08 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 02:27.09 | illizit | http://pastebin.com/m9d62490 |
| 02:28.00 | [TK]D-Fender | illizit: I see no SIP DEBUG in there, and no configs either |
| 02:28.07 | carrar | BUSTED!!! |
| 02:28.14 | illizit | lol |
| 02:28.20 | illizit | sry, forgot to enable |
| 02:28.52 | Spirits-Sight | Does any one know of a service that allows SMS / TXT msg to be sent through their service; besides a cell company? |
| 02:30.26 | *** join/#asterisk obnauticus (n=lol@about/windows/regular/obnauticus) |
| 02:33.58 | carrar | Spirits-Sight, what about using the Email to SMS? |
| 02:34.19 | carrar | thats typically included with cell phones |
| 02:35.27 | *** join/#asterisk harry_v (n=pcsuppor@S010600a0c93f6f7e.vs.shawcable.net) |
| 02:36.43 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 02:38.21 | tengulre | I have a 001.pcm in path /root, how to using PlayBack or BackGround to play it? |
| 02:40.43 | illizit | sorry i took so long |
| 02:40.44 | illizit | http://pastebin.com/mc2ca8d7 |
| 02:41.25 | illizit | i seeline 508: UDP is not a valid transport for 'to-OCS' (trunk name). we only use 'TCP'! ending call |
| 02:45.07 | [TK]D-Fender | illizit: and I see no configs |
| 02:46.15 | *** join/#asterisk mike-ekim (n=digiport@24.238.28.155) |
| 02:50.45 | illizit | asterisk configs: |
| 02:50.46 | illizit | http://pastebin.com/mec0cf30 |
| 02:53.11 | tengulre | help |
| 02:57.40 | [TK]D-Fender | tengulre: .pcm is not a file format exteniosn defined by * |
| 02:59.11 | tengulre | [TK]D-Fender, but I want to play that, how to do ? |
| 02:59.36 | JunK-Y | sox it. |
| 03:00.18 | [TK]D-Fender | yup |
| 03:04.19 | illizit | any ideas/ |
| 03:04.21 | illizit | ? |
| 03:04.23 | illizit | :) |
| 03:05.38 | *** join/#asterisk Wayhigh (i=wayhigh@www.kevinlynn.com) |
| 03:05.40 | JunK-Y | [TK]D-Fender: how would you list the full list of active channels? core show channels is not a solution, since it crops the channel name. |
| 03:06.03 | JunK-Y | i think i will create an AMI command like ListChannels |
| 03:06.47 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 03:07.18 | [TK]D-Fender | illizit: not offhand |
| 03:08.05 | [TK]D-Fender | JunK-Y: "core show channels concise" |
| 03:08.32 | russellb | there is already an AMI command to show all channels i think ... |
| 03:08.38 | russellb | aside from the CLI command one |
| 03:08.50 | JunK-Y | its cropping also the channame, no? |
| 03:08.54 | JunK-Y | let me try it. |
| 03:09.15 | JunK-Y | russellb: never found that one (that customer is using ABE) |
| 03:09.32 | russellb | maybe i made it up |
| 03:09.37 | russellb | it's possible |
| 03:09.55 | JunK-Y | what's the name of the ami cmd again? |
| 03:10.25 | russellb | Action: Status |
| 03:11.31 | JunK-Y | that doesnt work |
| 03:11.33 | JunK-Y | *Channel: Name of the channel to query for status |
| 03:11.40 | russellb | optional argument. |
| 03:11.46 | russellb | note the * |
| 03:11.54 | JunK-Y | since u have the know the channelname |
| 03:11.57 | JunK-Y | * is required |
| 03:11.59 | JunK-Y | not optional |
| 03:12.06 | russellb | ok, well i'm looking at the code |
| 03:12.08 | russellb | and it's optional |
| 03:12.08 | russellb | :) |
| 03:12.19 | JunK-Y | <PROTECTED> |
| 03:12.21 | JunK-Y | haaaaaaaaaa |
| 03:12.30 | Spirits-Sight | carrar: thanks for the response I am looking for a way not to have to give out two different numbers, I would like to give only one number out and if that number was to have a SMS sent to it then I would want it to be able to be recieved say through IM program which when responsed to the SMS it would be to a speacial IM screen name so that it would show the person the same phone number that they used, is there any thing like |
| 03:12.30 | JunK-Y | that definetly change the stuff |
| 03:13.10 | JunK-Y | i vote to change it to something more usual, like ListSomething. |
| 03:26.12 | jaytee | phew!!! I swapped out a TE212P card with a new one to test the new one and had a bitch of a time getting my spans to come out of alarm. Took a damn long time for the one connecting to my Nortel system to come up. some funky D-channel crap. Ended up having to reset the card on the Nortel side. |
| 03:27.30 | *** join/#asterisk japerry (n=japerry@drupal.org/user/45640/view) |
| 03:31.20 | illizit | see, right here: [Dec 4 22:29:35] DEBUG[4788] chan_sip.c: Header 4 [ 22]: To: <sip:204.x.x.x> |
| 03:31.34 | illizit | its not filling out the To field correctly :\ |
| 03:32.45 | JunK-Y | [TK]D-Fender: concise works, but any idea why it has a ; instead of a , for local chans? |
| 03:35.08 | [TK]D-Fender | JunK-Y: Nope |
| 03:35.32 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 03:44.29 | *** part/#asterisk DigitalIrony (n=eric@nat/digium/x-ec126581e25c565a) |
| 03:51.07 | *** part/#asterisk Spirits-Sight (n=christop@c-71-192-91-123.hsd1.ma.comcast.net) |
| 03:53.15 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 04:14.16 | illizit | ok |
| 04:14.19 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 04:14.20 | illizit | Fender |
| 04:14.22 | illizit | figured it out |
| 04:14.45 | [TK]D-Fender | illizit: What was it? |
| 04:14.45 | illizit | i downgraded to 1.6.0 ;) |
| 04:14.46 | [TK]D-Fender | ... |
| 04:14.49 | illizit | lol |
| 04:15.02 | illizit | seems like there is a big with sip_chan? |
| 04:22.50 | *** join/#asterisk ryoohki (n=ryoohki@208.96.15.252) |
| 04:22.52 | ryoohki | hello |
| 04:23.37 | *** join/#asterisk jeffik (n=jeffik@69-196-165-181.dsl.teksavvy.com) |
| 04:23.49 | ryoohki | anyone here? |
| 04:25.00 | *** join/#asterisk Strom_C (n=strom@netblock-208-127-61-171.dslextreme.com) |
| 04:25.21 | SkramX | nope |
| 04:25.32 | SkramX | just 241 people |
| 04:25.37 | carrar | Sipping |
| 04:38.26 | drmessano^ | Hmmm |
| 04:38.40 | drmessano^ | Interesting illizit had a problem with SIP TCP |
| 04:38.48 | drmessano^ | Pretty much the issue here |
| 05:02.29 | *** part/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 05:10.30 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 05:14.14 | *** join/#asterisk sah-work (n=Bawbatos@adsl-76-203-172-139.dsl.pltn13.sbcglobal.net) |
| 05:20.54 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 05:27.41 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 05:28.19 | ryoohki | i can't seem to enable vm( getting a fast busy) and an error in the logs "db.c: Unable to find key '131/SIP/149-b7b14a20' in family 'BLKVM'" |
| 05:29.21 | ryoohki | asterisk -V => Asterisk 1.2.18 |
| 05:30.08 | *** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-219-98.phlapa.east.verizon.net) |
| 05:36.09 | ryoohki | i've compared it to a working extension( ext. 238) and there appears to be no difference to ext. 131 |
| 05:37.28 | [TK]D-Fender | ryoohki: pastebin the complete call attempt |
| 05:37.29 | [TK]D-Fender | ~pb |
| 05:37.30 | jbot | [~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://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 05:47.35 | [TK]D-Fender | listens to the crickets crirp |
| 05:51.01 | ryoohki | http://pastebin.com/m79ce409a |
| 05:51.24 | ryoohki | [TK]D-Fender: working on several things at once |
| 05:52.29 | [TK]D-Fender | ryoohki: taht is not normal CLI output of a complete call from beginning to end. no debug mode, verbose 10 |
| 05:56.42 | *** join/#asterisk Bad_Robot- (n=Bad_Robo@cpe-76-173-219-25.socal.res.rr.com) |
| 05:59.21 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 06:01.34 | ryoohki | [TK]D-Fender: no debug mode, verbose 10 ???? |
| 06:02.12 | ryoohki | [TK]D-Fender: thanks for helping. what does that mean? "that is not normal CLI output of a complete call from beginning to end." |
| 06:02.24 | [TK]D-Fender | ryoohki: Tose are only a pile of debug messages. I want complete basic CLI output of the call from beginning to end at verbose 10 |
| 06:03.11 | ryoohki | [TK]D-Fender: and how might i go about that? i diff aroung in /etc/asterisk and found the extension is different from other extensions |
| 06:04.12 | [TK]D-Fender | ryoohki: This has nothing to dow ith you going to * CLI and pastebin the CLI output of a complete call. |
| 06:04.19 | [TK]D-Fender | ryoohki: This is not Raw-Cat Science |
| 06:05.23 | ryoohki | [TK]D-Fender: http://pastebin.com/m32e49407 |
| 06:06.01 | [TK]D-Fender | ryoohki: Again, that is not your CALL OUTPUT |
| 06:06.04 | ryoohki | [TK]D-Fender: you mean, go to the asterisk> prommpt> |
| 06:06.25 | [TK]D-Fender | ryoohki: Yes |
| 06:06.31 | [TK]D-Fender | ASTERISK CLI |
| 06:06.43 | [TK]D-Fender | \what is hard to understand about this? |
| 06:08.16 | ryoohki | [TK]D-Fender: is this what you want? http://pastebin.com/md817ec3 |
| 06:08.36 | ryoohki | [TK]D-Fender: \what is hard to understand about this? => nothing, i've never done it before |
| 06:08.58 | [TK]D-Fender | ryoohki: "set verbose 10 |
| 06:09.07 | ryoohki | [TK]D-Fender: oh ok! |
| 06:12.08 | ryoohki | [TK]D-Fender: http://pastebin.com/m305b5764 |
| 06:13.04 | [TK]D-Fender | ryoohki: "sip show peer 131" |
| 06:15.24 | ryoohki | [TK]D-Fender: i googled "Raw-Cat Science" and see it's only used in #aasterisk, so there must be some asterisk history behind this odd phrase... |
| 06:15.33 | ryoohki | [TK]D-Fender: http://pastebin.com/m4e28a5c7 |
| 06:15.41 | [TK]D-Fender | ryoohki: No, its just ME |
| 06:16.04 | [TK]D-Fender | ryoohki: One of my little inventions. Like the Raw-Cat Lawn Chair |
| 06:16.15 | ryoohki | [TK]D-Fender: is it an acronymn, R.C.S.? |
| 06:16.33 | [TK]D-Fender | ryoohki: Not purposefully |
| 06:16.44 | [TK]D-Fender | ryoohki: Addr->IP : (Unspecified) Port 0 |
| 06:16.53 | ryoohki | [TK]D-Fender: resting your head on a cat in the back yard and then it sprints away? |
| 06:16.56 | [TK]D-Fender | ryoohki: You are calling a sip device that has not registered to * |
| 06:17.04 | [TK]D-Fender | ryoohki: So * cannot call them |
| 06:17.18 | drmessano^ | Nothing like calling no-one |
| 06:17.28 | ryoohki | [TK]D-Fender: how cna i register it? we have a web gui and a mysql backend |
| 06:17.39 | drmessano^ | How do you know if no-one doesn't answer? |
| 06:17.45 | [TK]D-Fender | drmessano^: And don't not use no double negatives neither! |
| 06:18.13 | [TK]D-Fender | ryoohki: Your GUI doesn't mean anything. the phone has not registered to * |
| 06:18.39 | ryoohki | [TK]D-Fender: just use the CLI? wil the CLI modify the files in /etc/asterisk? if the files in /etc/asterisk become out of sync with the mysql database if i edit them directly? |
| 06:18.53 | [TK]D-Fender | ryoohki: Who said this was a config issue? |
| 06:19.06 | [TK]D-Fender | ryoohki: all I know is the device you are calling is not registered. |
| 06:19.16 | [TK]D-Fender | ryoohki: I have no reason to believe it ever even TRIED |
| 06:19.17 | ryoohki | [TK]D-Fender: no one but i assume there must be a file or db entry to be made to fix this |
| 06:19.27 | drmessano^ | ... |
| 06:19.30 | [TK]D-Fender | ryoohki: Says who? |
| 06:19.33 | ryoohki | [TK]D-Fender: there may be no phone at all |
| 06:19.50 | drmessano^ | How can no phone not register to asterisk? |
| 06:19.57 | [TK]D-Fender | ryoohki: what does "no phone at all" mean? |
| 06:20.15 | [TK]D-Fender | ryoohki: You are calling a SIP device you configured. Why would it not exist at all? |
| 06:20.18 | ryoohki | [TK]D-Fender: we may have removed it or given it another extension and then wanted to use the extension to go directly to vm for a catch all vm |
| 06:20.31 | [TK]D-Fender | ryoohki: you don't KNOW? |
| 06:20.37 | drmessano^ | Sounds like someone turned on the improbability drive and ended up with a giant cup of "no tea" |
| 06:21.06 | [TK]D-Fender | ryoohki: as for changing your call handling, FreePBX is NOT supported in here |
| 06:21.08 | ryoohki | [TK]D-Fender: i think a lady here was being harassed and we gave her a new number and another person wanted a dropbox vm so we told him to just use that |
| 06:21.22 | [TK]D-Fender | ryoohki: Less think, more look. |
| 06:21.42 | [TK]D-Fender | ryoohki: and as I said when it comes to configuring your system, it is not supported here |
| 06:21.49 | ryoohki | [TK]D-Fender: if a voip phone is unplugged, should the call go to vm/ |
| 06:21.51 | ryoohki | ? |
| 06:22.06 | [TK]D-Fender | ryoohki: Should it go to VM? that depends on your configuration. |
| 06:22.19 | ryoohki | [TK]D-Fender: that's fine - you've helped me a lot |
| 06:22.27 | [TK]D-Fender | ryoohki: which being FreePBX is not a topic for this place. They have their own channel for support : #freepbx |
| 06:23.16 | ryoohki | [TK]D-Fender: you would think if a phone is unplugged, then the call for the extension would go to the 131 vm box - i would think commmon sense |
| 06:23.26 | [TK]D-Fender | ryoohki: No. |
| 06:23.45 | [TK]D-Fender | ryoohki: your configs do what they are told. Why should I believe that it was configured to act that way? |
| 06:24.11 | [TK]D-Fender | ryoohki: there is no magic. There are no "defaults". Assume = BS |
| 06:24.27 | [TK]D-Fender | there is NOTHING "common" about * |
| 06:24.43 | ryoohki | [TK]D-Fender: the extension was never changed - passwd or anything about it, the voip phone for it was simply moved |
| 06:24.50 | [TK]D-Fender | I've seen people use * as a CRON replacement, and nothing more. |
| 06:25.07 | [TK]D-Fender | ryoohki: And that description proves nothing about how your system was set up. |
| 06:25.13 | ryoohki | [TK]D-Fender: thanks |
| 06:25.23 | *** part/#asterisk ryoohki (n=ryoohki@208.96.15.252) |
| 06:26.46 | mahlon | contemplates * as a cron replacement |
| 06:26.48 | [TK]D-Fender | drmessano^: \o/ |
| 06:26.58 | mahlon | aaaaand no |
| 06:27.14 | [TK]D-Fender | mahlon: I see you've thought long & hard on it |
| 06:27.18 | [TK]D-Fender | :) |
| 06:27.26 | mahlon | :) |
| 06:27.36 | [TK]D-Fender | mahlon: and others who've use * as a jukebox... mine used to make me coffee.... |
| 06:28.18 | mahlon | in the toolbox of available tools... seems like cron is better suited for cron tasks |
| 06:28.28 | mahlon | unless of course... we're talking about timed monkey calls |
| 06:28.39 | drmessano^ | Alison wakes me up every morning |
| 06:28.50 | [TK]D-Fender | mahlon: I did that as a mass-page call-file at my office :) |
| 06:29.06 | [TK]D-Fender | drmessano^: Does she do "naughty" recordings too? ;) |
| 06:29.08 | mahlon | drmessano^: are you sure you're awake when you hear her voice? :) |
| 06:29.49 | drmessano^ | lol |
| 06:29.56 | mahlon | [TK]D-Fender: I had something similar, but it was a single monkey call, once a week. Random developer, random time. |
| 06:30.00 | drmessano^ | Im pretty sure |
| 06:30.06 | drmessano^ | She askes me math questions too |
| 06:30.13 | [TK]D-Fender | Phone Sex causes hearing AIDS :p |
| 06:30.15 | mahlon | Each week, it somehow became an exciting thing: I GOT THE MONKEY CALL!!! |
| 06:30.15 | drmessano^ | asks |
| 06:30.26 | mahlon | Of course, I got it in the middle of the night a few times... |
| 06:31.53 | drmessano^ | Wakeup call is probably the most useful thing my home box does lol |
| 06:32.12 | mahlon | I love how no 'normal' people are excited by phones, until you demo a monkeycall. |
| 06:32.15 | drmessano^ | I just need to cron it now |
| 06:32.50 | mahlon | drmessano^: Isn't there an Allison sample about a "phone in your pants" or somesuch? |
| 06:33.04 | drmessano^ | yep |
| 06:33.26 | mahlon | mmm yeah. good morning, ms. smith. |
| 06:33.50 | drmessano^ | I wanted one of her going "Im feeling frisky".. but that would be too unbelievable to actually hear from a woman |
| 06:34.04 | mahlon | hahah |
| 06:34.35 | [TK]D-Fender | drmessano^: A sad state indeed |
| 06:34.42 | mahlon | It would be a fun social experiment to see how my wife would react to that wake up call. |
| 06:34.50 | [TK]D-Fender | drmessano^: And I don't mean just "bama ;) |
| 06:34.52 | mahlon | "I have no idea who that was, honey." |
| 06:35.07 | drmessano^ | Again.. I doubt she would believe it either |
| 06:37.56 | [TK]D-Fender | alrighty... bed time... |
| 06:37.57 | [TK]D-Fender | later all |
| 06:58.22 | tengulre | how to play pcm audio format file? |
| 07:00.15 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 07:05.55 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 07:11.08 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 07:17.40 | *** join/#asterisk chi6IT41 (n=chigital@services.mivitec.net) |
| 07:17.57 | *** join/#asterisk Daejeo (n=chatzill@118.221.248.181) |
| 07:39.59 | *** join/#asterisk chendy (n=chatzill@58.60.97.211) |
| 07:41.32 | *** join/#asterisk ryoohki (n=ryoohki@208.96.15.252) |
| 07:43.51 | *** join/#asterisk Yourname`_ (i=Yourname@unaffiliated/yourname/x-837320) |
| 07:46.01 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 07:46.01 | *** join/#asterisk jksM (i=jks@193.189.93.254) |
| 07:48.41 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 07:49.16 | *** join/#asterisk xrmx__ (n=user@host5-117-dynamic.60-82-r.retail.telecomitalia.it) |
| 07:55.08 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 07:58.26 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 08:09.23 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 08:17.56 | digime | any asterisk techs for hire? have a few troubleshooting/issues on my box and just need a quick hand if someone could remote in. |
| 08:19.42 | angryuser | digime: describe your problem |
| 08:25.32 | digime | okay |
| 08:26.25 | digime | the issue is that asterisk sees the state of the phones as busy/unavailable even though they are not. i think some feature was depricated when i upgraded from 1.2 to 1.4 and i haven't been able to resolve it. the message suggests reading upgrade.txt for solutions but i couldnt figure it out... |
| 08:27.44 | digime | Does anyone here have a snom m3 wireless sip phone? All my other IP phones can handle multiple calls but asterisk won't send multiple calls to the M3. It thinks the M3 is "busy". |
| 08:40.21 | angryuser | digime: i remeber about some option in sip.conf , try to find example file of * 1.4 it is something related to presense, pff dont remember exactly, just compare files |
| 08:41.02 | angryuser | digime: do you have a call-limit set for your snom m3 ? |
| 08:45.09 | digime | no i dont have a call limit set |
| 08:45.41 | digime | are you talking about within the m3 interface? I don't think so. where would I check? it's damn frustrating, i tell u.. .:) |
| 08:46.34 | digime | all my other sip phones work fine and can accept many calls at once but asterisk thinks the m3 is busy only after one call. i even tried putting the sip configuration into the phone as multiple identities.. but snom support suggested not to do that. they said just use one sip account and it should work |
| 08:47.41 | digime | wow good idea, get a sample file for 1.4's sip.conf and look for changes.. good idea!! |
| 08:49.20 | *** join/#asterisk Mrnick (n=Mrnick@88.197.232.3) |
| 08:49.29 | angryuser | digime: you need to set call-limit to presense work |
| 08:50.00 | digime | ok.. |
| 08:50.26 | digime | "call-limit to presense work" means? |
| 08:50.51 | angryuser | set call-limit = 60 for example , let the phone to manage this call ability |
| 08:51.01 | angryuser | this => his |
| 08:51.06 | digime | where does that setting go? |
| 08:51.22 | digime | and why is the issue only with my m3 and not the other ip phones? |
| 08:51.32 | angryuser | digime: sip.conf for each user or in general for everybody |
| 08:51.53 | digime | ah |
| 08:52.17 | digime | wow i dont have that setting in my config at all! |
| 08:52.24 | digime | you are like some kind of angel! |
| 08:52.38 | digime | okay.. if it works on all the other phones, then i can just go ahead and add it to my sip entry for the snom? |
| 08:52.46 | digime | and what is the exact line I should add? |
| 08:53.46 | angryuser | digime: it am not sure why snom m3 do not manage multiple call's, try to look at it's web interface for any kind of call limit setting |
| 08:54.02 | digime | The call-limit value appears to be the number of calls that can be simultaneously associated with an extension; I've seen a note that setting it down to 1 blocks attended transfers. Advised values I've seen are 4 or 10. Asterisk 1.4 appears to default to zero (= unlimited) which is when the problems start. |
| 08:54.31 | digime | i took screen shots of my m3 and sent it to their support, they said it was all good. but perhaps it's because i need to specify the setting in sip.conf |
| 08:54.51 | angryuser | digime: go and test |
| 08:56.16 | angryuser | digime: and check if you phones are subscribed to a good context |
| 08:56.24 | digime | hmm |
| 08:56.34 | digime | okay, what is the best way to do that/ |
| 08:56.39 | angryuser | *for presense* subscribecontex=something |
| 08:56.52 | angryuser | in CLI sip show subscriptions |
| 08:57.19 | angryuser | subscribecontext* |
| 08:58.22 | digime | ok |
| 08:58.37 | *** join/#asterisk l0gis1c (n=logisic@adsl-75-18-165-92.dsl.pltn13.sbcglobal.net) |
| 08:59.41 | digime | one more: i set up qos on my phones running through a sonicwall, the sonicwall has a default sip timeout of 3600, but my asterisk box was set to 200 or so (in sip.conf), so I changed all my phones to match the sonicwall and time out at 3600. is that correct? |
| 09:00.06 | digime | maxexpirey=3600 |
| 09:00.10 | *** join/#asterisk lucky|aba (n=lucky@ip72-205-200-133.sb.sd.cox.net) |
| 09:00.53 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 09:01.21 | digime | wow! sip show subscriptions only shows one phone listed, the snom m3 that I am having issues with!! why?? |
| 09:01.49 | *** join/#asterisk Aurs (n=Ove_Aurs@apb9hb.ip.ssc.net) |
| 09:02.24 | digime | extension -- (no extension shown!!) |
| 09:02.37 | Aurs | anyone know why the output of show channels sometimes is wrong? |
| 09:02.40 | digime | Last state <none> |
| 09:03.13 | Aurs | (asterisk 1.2, approx 500 sip peers, uptime 5 days) |
| 09:03.29 | digime | 1 active SIP subscription (but I actually have multiple phones registered...) ??? |
| 09:06.33 | nfi|ermes | exten => s,3,Dial(Zap/1-1/03483857355) |
| 09:06.38 | nfi|ermes | what's wrong ? |
| 09:09.41 | nfi|ermes | [Dec 5 11:12:10] WARNING[5690]: app_dial.c:1242 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 09:11.43 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 09:13.15 | nfi|ermes | i tried also Dial(Zap/1/03483857355) |
| 09:13.19 | nfi|ermes | i tried also Dial(Zap/1/3483857355) |
| 09:13.43 | Mrnick | has anybody got an idea why my sip.conf is not appearing in gatsman and other programs connecting remotely but sip show peers works from gastman? |
| 09:15.41 | Aurs | nfi|ermes: looks like chan_zap.so isn't loaded? |
| 09:18.15 | *** join/#asterisk rpr (n=chatzill@33.Red-88-29-0.staticIP.rima-tde.net) |
| 09:18.30 | rpr | Anyone knows if B410P needs an external power feed to actuate in NT mode? |
| 09:20.57 | rpr | hello, anybody here? |
| 09:24.39 | angryuser | rpr: no |
| 09:27.44 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 09:30.40 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 09:31.23 | digime | angryuser: sip show subscriptions only shows one phone listed, the snom m3 that I am having issues with!! |
| 09:31.39 | digime | what are the implications of this? |
| 09:31.53 | *** join/#asterisk awk (n=awk@gw1.security.web.za) |
| 09:32.00 | angryuser | digime: paste here the line |
| 09:33.02 | awk | anyone here good with snom phones, can somebody tell me what ser=phone is supposed to do under function keys on a snom ? if you add that as eg: active -> extesion -> then <sip:ser=phone.... ? |
| 09:34.26 | digime | Peer User Call ID Extension Last state Type Mailbox |
| 09:34.27 | digime | 11.21.333.173 74 zm.9oe926ip -- <none> mwi 74 |
| 09:34.27 | digime | 1 active SIP subscription |
| 09:42.06 | *** join/#asterisk Vale-ICS (n=vale@icsnet.demon.co.uk) |
| 09:44.42 | awk | hrm, nobody know snom phones? |
| 09:47.03 | mort_gib | Sure |
| 09:49.55 | *** join/#asterisk drew (i=drew@whitehat.org) |
| 09:50.54 | drew | has anyone experienced an issue with calls sometimes having one side unable to head the other ? sip device<->asterisk<->iax2 trunk<->asterisk<->sip device |
| 09:51.00 | drew | to hear* |
| 09:51.43 | drew | its intermittant and the other side can sometimes get through (every third attempt or so) |
| 09:56.53 | awk | mort_gib: then do you know the answer to my questions |
| 09:59.10 | mort_gib | ser is a SIP proxy, I have NO clue as to what that means, normally you would use <sip:%extension%@sip.server.net> |
| 09:59.26 | Mrnick | does anybody know why gastman doesn`t use my sip.conf although the "sip show peers" works in gastman? |
| 10:03.28 | awk | mort_gib: well this client tells me he has ser = under there, then his phone does something special where he can then press the flashing key an it picks up that line, heard of this? |
| 10:03.33 | awk | like a pickup but using hints? |
| 10:05.16 | mort_gib | no, you need to look at his extensions.conf |
| 10:07.19 | drew | is there a way to pipe output from asterisk console ? |
| 10:08.03 | awk | mort_gib: I do the dailplan, i just thought the snom had such a feature, which i doubt it did |
| 10:08.21 | awk | mort_gib only way I knew is a pickup group with *8 and maybe adding a feature of *8 to a key on the phone |
| 10:08.26 | mort_gib | No, they don't |
| 10:09.35 | *** part/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-f12cddb10b1414d2) |
| 10:10.02 | mort_gib | :-) no magic around, you have to do that stuff in the dialplan |
| 10:10.23 | awk | :) |
| 10:12.31 | mort_gib | I use *6%exten% for the dedicated pickup... |
| 10:16.27 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 10:19.39 | *** join/#asterisk IguanaNed (n=me@CPE000625db3f84-CM00111ae43f1e.cpe.net.cable.rogers.com) |
| 10:21.22 | *** join/#asterisk subdolus (n=subdolus@subby.afraid.org) |
| 10:22.09 | subdolus | can asterisk detect silence in a channel? eg: someone hangs up but the call doesnt disconnect properly, but after say 10 seconds of silence, it will dial back |
| 10:22.45 | angryuser | subdolus: what do you mean does not disconnect ? |
| 10:26.39 | subdolus | say my providers end disconnects, but my asterisk box thinks its still in the call... |
| 10:29.28 | *** join/#asterisk lou_gr (n=lou_gr@212-70-216-131.ath.static.tee.gr) |
| 10:29.39 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 10:31.23 | subdolus | Ok, never mind that heh |
| 10:31.27 | *** join/#asterisk Paige (n=Paige@2001:470:b:aa:21f:c6ff:fe48:8988) |
| 10:31.33 | Paige | hello |
| 10:31.44 | troubled | hey guys, with sip in asterisk, is it limited to the 8khz that my digium card is as well? |
| 10:32.03 | troubled | just wondering if absolutely everything going in and out of asterisk gets down sampled or not |
| 10:32.13 | Paige | i am getting this error when building asterisk-addons cdr_addon_mysql.c:314: error: ?ast_config_load? undeclared |
| 10:32.25 | Paige | anyeone have any suggestions to try? |
| 10:32.41 | angryuser | subdolus: normally when there is no BYE at the end at the call, so if you missing BYE regulary you have network problems |
| 10:33.36 | subdolus | ok |
| 10:34.20 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 10:38.49 | troubled | anyone? |
| 10:45.57 | *** join/#asterisk qdk (n=qdk@212.27.19.21.bredband.3.dk) |
| 10:53.08 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 10:59.22 | *** part/#asterisk Paige (n=Paige@2001:470:b:aa:21f:c6ff:fe48:8988) |
| 11:01.37 | Daejeo | awk |
| 11:06.18 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:13.57 | *** join/#asterisk Karlitoo (n=Karlitoo@213.137.110.67) |
| 11:14.51 | Karlitoo | does any one know wich old pwlib compiles correctly and doesn't give psasl.o or pldap.o errors |
| 11:14.58 | Karlitoo | I'm trying 1.9.0 |
| 11:15.07 | Karlitoo | but it's giving me those 2 errors |
| 11:16.30 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 11:18.13 | Mrnick | any ideas why sip.conf is not use in gastman or php requests |
| 11:25.17 | *** join/#asterisk cianmaher (n=Cian@84.203.233.126) |
| 11:26.04 | cianmaher | does anyone know is it possible to interrupt a call after 30 mins and play a prompt??? |
| 11:28.47 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 11:34.45 | Karlitoo | can any one help me out with pwlib compiling |
| 11:54.19 | espent | how can i limit call duration with the L-attribute for dial, via originate thorugh AMI? |
| 11:55.13 | *** join/#asterisk jc_yyz2bkk (n=jc@ppp-58-8-60-200.revip2.asianet.co.th) |
| 11:55.50 | *** join/#asterisk Guest25980 (n=root@support.ccxtech.fi) |
| 11:57.38 | jc_yyz2bkk | hi, when making a system recording with freepbx, once you enter *77 is there a message that plays before you are supposed to say your recording? |
| 11:58.52 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 11:59.24 | Guest25980 | Hi, I have problem with TCP SIP. I try to make call from extension with transport=tcp to TCP SIP trunk and I get: ERROR[19343]: chan_sip.c:4003 create_addr_from_peer: 'UDP' is not a valid transport for 'ocs_sip_trunk'. we only use 'TCP'! ending call. |
| 12:03.32 | *** join/#asterisk Segnale007 (n=Pietro@host88-253-dynamic.42-79-r.retail.telecomitalia.it) |
| 12:21.49 | jblack | Guest25980: I suppose you could file a bug. |
| 12:28.38 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 12:29.06 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-c6590b859713b9f8) |
| 12:41.13 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:41.13 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:42.23 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 12:49.07 | *** join/#asterisk uri_ (n=uri@84.201.225.66) |
| 12:49.48 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 12:55.22 | *** part/#asterisk Guest25980 (n=root@support.ccxtech.fi) |
| 12:56.41 | uri_ | Hi, people! Is there any way to setup TON to unknown and NPI to 'isdn-telephony/E.164' in asterisk fo outgoing calls? |
| 13:06.28 | tzafrir_laptop | uri_, in chan_zap/chan_dahdi? pridialplan = unknown |
| 13:11.37 | uri_ | pridialplan = unknown sets both ton and npi to default, but my telecom needs first one in default and second one in isdn-telephony... |
| 13:14.15 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 13:22.22 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 13:22.22 | *** mode/#asterisk [+o russellb] by ChanServ |
| 13:22.36 | *** part/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 13:28.36 | *** join/#asterisk mbranca (n=matteo@93-62-230-243.ip24.fastwebnet.it) |
| 13:38.00 | *** join/#asterisk superc (n=superc@ip-90-186-196-159.web.vodafone.de) |
| 13:38.20 | superc | hi. is there a way to disable packet2packet bridging in asterisk? |
| 13:38.34 | *** join/#asterisk artakamiraghyan (n=artakami@cpc3-york1-0-0-cust404.leed.cable.ntl.com) |
| 13:39.31 | artakamiraghyan | Hi all |
| 13:40.02 | superc | if there is an incoming call and it is bridged theres often the problem that there is no sound at all.. the called person is behind nat. so probably this is because of packet2packet bridging |
| 13:40.36 | artakamiraghyan | I am fesher here, and I am looking advice aboute asterisk hardware |
| 13:40.42 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:41.06 | superc | what hardwrae |
| 13:42.04 | Karlitoo | is there any version of asterisk that can be used with h323 |
| 13:42.12 | superc | I already have disablep2p=yes in rtp.conf ... but no change |
| 13:42.14 | artakamiraghyan | the issue is in the following |
| 13:42.44 | Karlitoo | I cannot compile completely, everything that I need for asterisk and h323 |
| 13:42.44 | superc | Karlitoo I think you have to compile an addon |
| 13:42.54 | Karlitoo | yeah I have |
| 13:42.56 | artakamiraghyan | I need hardware for call centre for up to 100 concurrent cals. |
| 13:44.50 | superc | karlitoo there are also debian packagages available |
| 13:44.59 | superc | asterisk-h323 and asterisk-oh323 |
| 13:47.47 | *** join/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 13:48.21 | *** join/#asterisk madduck (n=madduck@debian/developer/madduck) |
| 13:48.28 | madduck | does asterisk 1.4 support IPv6 in any way? |
| 13:49.51 | *** join/#asterisk ceegee (n=christia@mail.cg-networks.de) |
| 13:49.54 | ceegee | hello |
| 13:50.39 | ceegee | I am looking for a way to get my status leds on my snom 320 on |
| 13:51.09 | ceegee | I programmed an extension that switches a group membership on and off with one button |
| 13:51.32 | superc | what status leds? |
| 13:51.32 | *** join/#asterisk moy (n=moy@69.150.142.130) |
| 13:51.39 | ceegee | now I want to display the status with the led which belongs to the button |
| 13:52.12 | madduck | this is fucked up. Nokia E71 sends headers with IPv6 addresses in IPv4 packets. |
| 13:52.23 | superc | probably you are looking for busy-lamp-field function |
| 13:52.28 | ceegee | if I press the button once, the led should shine, if I press the button a second time the led should go off |
| 13:53.04 | ceegee | I use asterisknow 1.5 running asterisk1.6 packages and I have snom phones with 7.3.x firmware |
| 13:54.14 | ceegee | I have read that func_devstate is a built-in component of asterisk 1.6 so it should work, right? |
| 13:54.33 | dandre | Hello, |
| 13:54.39 | *** part/#asterisk madduck (n=madduck@debian/developer/madduck) |
| 13:55.10 | dandre | Is there any possibility, in asterisk 1.6, to put a call on hold using the manager interface? |
| 13:57.15 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:58.06 | *** join/#asterisk pyite (n=pyite@c-76-21-105-195.hsd1.ca.comcast.net) |
| 14:01.03 | *** join/#asterisk gangans (n=LL@84.36.154.193) |
| 14:01.30 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:02.25 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 14:02.44 | angryuser | dandre: i think you can do all with AMI |
| 14:02.45 | jasonwoot | woot |
| 14:02.54 | gangans | I have a voip gateway connected to PSTN and it supports T.38 standard , what i'd like to implement is a faxing system using this gw with asterisk , when someone call the gw for faxes this should be directed to asterisk and there asterisk just print this fax in a pdf file , any ideas about this issue ? or how to achieve that ? |
| 14:03.03 | *** join/#asterisk vi390 (n=fc@unaffiliated/vi390) |
| 14:03.46 | angryuser | gangans: use hylafax with iaxmodem |
| 14:03.47 | jasonwoot | gangans: hylafax |
| 14:03.49 | ceegee | no ideas? |
| 14:03.52 | jasonwoot | jinx |
| 14:04.06 | dandre | angryuser: so putting a call on hold has been added in 1.6 because in 1.4 this is impossible |
| 14:04.37 | *** join/#asterisk killfill (n=killfill@200.63.96.244) |
| 14:04.40 | killfill | hi |
| 14:05.14 | gangans | angryuser, jasonwoot , thanks for response , any good tutorial on how to setup that with asterisk ? |
| 14:05.39 | angryuser | dandre: hm, try to ask in mailing list to be sure |
| 14:05.51 | killfill | ive got a problem with my asterisk. Just put a celulink (thats the name in english?), a modem i can put a GSM chip on it so i can access de celular network |
| 14:06.04 | killfill | http://pastebin.ca/1276850 |
| 14:06.13 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:06.15 | killfill | thats my config (zapata.conf and zaptel.conf) |
| 14:06.29 | angryuser | gangans: it is pretty simple, hylafax will "see" iaxmodes as /dev/ttyx |
| 14:06.41 | angryuser | iaxmodem* |
| 14:07.03 | killfill | im using "Zap/g1/my_cel_phone" |
| 14:07.23 | killfill | and i see "-- Called g1/my_cel_phone" |
| 14:07.45 | gangans | angryuser, thanks alot I'll try to test that |
| 14:07.52 | killfill | -- Zap/1-1 answered IAX2/6000-459 |
| 14:07.55 | killfill | and there it stais. |
| 14:07.58 | killfill | stays... |
| 14:08.15 | killfill | is it a problem with the celulink hardware? or my configuration on asterisk? |
| 14:09.17 | angryuser | killfill: are you sure that tha number you dialing is ok ? like some provider's require prefix, do you hear any dial tone ? |
| 14:09.18 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:09.31 | angryuser | killfill: do you hear anything ? |
| 14:10.31 | angryuser | are you sure that g1 is has a channel ZAP you need ?, and why do you call it by group ? i suppose your interface is connected to one zap channel |
| 14:11.14 | angryuser | why are you so slow ? |
| 14:13.48 | killfill | angryuser, was in the bathroom.. :P |
| 14:14.04 | killfill | well, the number is ok. its my personal phone.. :P |
| 14:14.34 | killfill | yes, i have it connected to one of the FXO ports |
| 14:14.36 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 14:15.25 | angryuser | killfill: port1 or another port ? |
| 14:15.34 | killfill | port1 |
| 14:15.47 | killfill | others are not connected to anything. |
| 14:16.04 | angryuser | killfill: can you use tha sim card in the phone ? try to verify that it is ok |
| 14:16.27 | killfill | angryuser: yup. the chip works ok. |
| 14:16.41 | killfill | i see nothing when i call the chip neither.. |
| 14:16.49 | killfill | no 'Ring'or anything |
| 14:16.57 | vi390 | hi, I want to work with agis (need support for playing back messages, react on keys pressed etc) the best would be with python. Is there anything known, which works well, and is not to old. Or is the PhpAgi the only usable "agi" at the moment ? |
| 14:17.19 | uri_ | Is there any way to setup TON to unknown and NPI to 'isdn-telephony/E.164' in asterisk fo outgoing calls? |
| 14:17.20 | *** join/#asterisk jtodd (n=jtodd@nat/digium/x-345a703cca1bbe47) |
| 14:17.35 | killfill | angryuser: there is nothing strange in the config is it?.. |
| 14:17.42 | killfill | maybe the modem is just bad. |
| 14:17.47 | angryuser | killfill: can you connect your modem to analog phone directly ? |
| 14:17.49 | killfill | itsnew.ive not use it before.. |
| 14:18.04 | angryuser | killfill: pastebin you zaptel.conf |
| 14:18.10 | killfill | nope. its just for gsm |
| 14:18.19 | killfill | oh its there in the pastebin |
| 14:18.23 | killfill | below |
| 14:18.36 | killfill | fxsks = 1,2,3 |
| 14:18.58 | angryuser | zap show status says what ? |
| 14:19.18 | killfill | Wildcard TDM410P Board 1 OK |
| 14:19.33 | Katty | morning |
| 14:19.38 | killfill | its alwais OK tho.. |
| 14:19.53 | killfill | no difference if i connect the lines or not.. |
| 14:20.24 | angryuser | killfill: pff , your modem is recent ? i mean it is not 5 yrs old ? |
| 14:20.49 | [TK]D-Fender | vi390: PhpAgi is a 3rd party class. This doe not say taht AGI onyl works on PHP. AGI WORKS. |
| 14:21.07 | killfill | well dont know when it was created.. but its a cheap thing i just got in myhands. |
| 14:21.13 | [TK]D-Fender | vi390: AGI Does not care what language you use |
| 14:21.49 | angryuser | killfill: with your operator contract, can you receive and send call or just send ? |
| 14:22.02 | jaytee | Katty, morning |
| 14:22.27 | Katty | hugs jaytee |
| 14:22.29 | killfill | angryuser: send and receive. i test them both before putting the chip on the modem |
| 14:22.33 | killfill | (with my phone) |
| 14:22.38 | Katty | jaytee: i've been very mood swingy last night )= |
| 14:22.49 | Katty | jaytee: got pretty mad at the dog last night. )= |
| 14:22.51 | Katty | feels bad. |
| 14:22.58 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 14:23.03 | angryuser | killfill: ahve you tryed to call your modem and lookk what happens in console ? |
| 14:23.12 | jaytee | hugs Katty |
| 14:23.14 | vi390 | [TK]D-Fender: ok @ language use. But what I saw is, that I have support for Playing back wav files , and react on things, and this in a nice framework, which is easy to handle. otherwise as I understand I could just call a python instance and hand the parameters over somehow... |
| 14:23.20 | killfill | angryuser: yup. i see no reaction on asterisk |
| 14:24.31 | vi390 | [TK]D-Fender: but So far I only found the phpAgi , there are several old python "agi" things, and it seems sort of weird, because its hard to find realiable informations about agis other than Phpagi |
| 14:24.39 | angryuser | killfill: where do you live ? ;) it's a made for us product ? |
| 14:24.45 | angryuser | US* |
| 14:24.58 | [TK]D-Fender | vi390: .... |
| 14:24.59 | [TK]D-Fender | ~book |
| 14:25.00 | jbot | book is probably probably 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://tfot.leifmadsen.com or see ~buybook |
| 14:25.02 | [TK]D-Fender | ^^^^ |
| 14:25.13 | jaytee | Katty, you're just going through a stress period because of all the bullcrap at your work. Just remember, "All I can do IS all I can do" and if other people put unreasonable expectations or demands on you that were never initially part of your accepted and agreed upon position you aren't under any obligation to meet those demands or expectations without a "renegotiation". |
| 14:25.19 | [TK]D-Fender | vi390: 2 whole chapters on it with samples in many languages |
| 14:25.43 | *** join/#asterisk etfonhomey (n=chatzill@www2.askpri.org) |
| 14:26.11 | angryuser | killfill: whatever try to change that port to loopstart (ls) same i zapata.conf |
| 14:26.15 | Katty | jaytee: still shouldn't have yelled at the dog so much )= |
| 14:26.42 | *** join/#asterisk xrmx__ (n=user@host10-182-dynamic.6-79-r.retail.telecomitalia.it) |
| 14:26.44 | vi390 | [TK]D-Fender: you mean the one from voip-info.org (in the wiki) ? |
| 14:26.49 | jaytee | Katty, he'll forgive ya. In fact he's probably forgotten all about within a 15 minute period of it happening. |
| 14:26.58 | [TK]D-Fender | ~book |
| 14:26.59 | jbot | well, book is probably 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://tfot.leifmadsen.com or see ~buybook |
| 14:27.04 | [TK]D-Fender | vi390: the BOOK ^^^^^^^^^^^ |
| 14:27.21 | tzafrir_laptop | coppice, here? |
| 14:27.27 | Katty | jaytee: yeah probably. he asked just as happy as could be this morning |
| 14:27.50 | coppice | no. here |
| 14:28.36 | vi390 | [TK]D-Fender: oh, ok, didnt know about that . thanks gonna read that :-) |
| 14:28.45 | jaytee | Katty, I strongly agree with the bumper sticker that says, "The more I understand humans, the more I prefer the company of my dog." |
| 14:29.32 | Katty | jaytee: (= |
| 14:29.43 | Katty | jaytee: puppy had me up ever 2 hours last night having to go out :/ |
| 14:29.54 | jaytee | Katty, I think your sleekgeek.org site is dead forever from the looks of things. |
| 14:29.57 | Katty | jaytee: then i'd take him out, put him back in the kennel and go to bed. and he'd starting whining AGAIN |
| 14:30.17 | Katty | jaytee: i've noticed the same, wondering if our dydns subscription ran out or something |
| 14:31.07 | jaytee | a kennel? all the dogs we've ever had slept either at the foot of our bed or on the floor next to it. They didn't like sleeping more than 8 feet away. Sense of Duty I imagine. |
| 14:31.32 | coppice | so, you're not a kennel hacker |
| 14:31.39 | Katty | well i can't have the puppy run all over the house yet |
| 14:31.42 | Katty | he's not quite housebroken |
| 14:31.45 | jaytee | ah |
| 14:31.50 | Katty | he's right by the bedroom |
| 14:32.09 | Karlitoo | run asterisk run |
| 14:32.09 | jaytee | coppice? |
| 14:32.12 | Katty | at 6 months, Jager was perfectly house broken, he slept by my side of the bed. |
| 14:32.13 | Karlitoo | lol |
| 14:32.32 | Katty | and occasionally tried to get on the bed and weasel himself between me and ryan in the middle of the night. |
| 14:33.02 | jaytee | I think it was the comedian Steven Wright that wanted to get a dog and name him "Stay" |
| 14:33.05 | Katty | it's hard to fit 3 people in a queen sized bed! |
| 14:33.21 | jasonwoot | Jagermeister is by my side of the bed too |
| 14:33.26 | jasonwoot | wait |
| 14:33.27 | Katty | jasonwoot: ;) |
| 14:33.35 | jaytee | I know a few people that are hard to fit in a queen size bed all by themselves |
| 14:33.50 | *** join/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net) |
| 14:34.10 | Katty | grins. |
| 14:34.14 | Katty | i pray i never have that problem. |
| 14:34.20 | jaytee | me either! |
| 14:34.25 | killfill | angryuser, heh..in Chile. |
| 14:34.40 | killfill | loopstart?.. hm.... |
| 14:35.13 | Katty | i thought about bringing riddick to work with me for the morning, and jsut telling people he had a vet appt. at 11 |
| 14:35.49 | jaytee | kewlstart is just a smarter version of loopstart. if you're in Chile and using switching equipment from the days of Salvador Allende you might try groundstart instead :-) |
| 14:37.30 | jaytee | I'd love to work at one of those "Bring your doggy to work day" places. Much better than "Oh, and don't forget people, friday is Hawaiian shirt day." places. |
| 14:41.28 | *** join/#asterisk gangans (n=LL@84.36.131.35) |
| 14:42.09 | gr0mit | suffered 'bring your kids to work day' |
| 14:43.28 | *** join/#asterisk eric2 (n=nobody@sudbury-cable-69-60-228-64.unitz.ca) |
| 14:44.38 | killfill | jaytee, jaja... |
| 14:44.42 | coppice | a better idea would be "send your kids to work, and stay home and relax day" |
| 14:45.43 | gr0mit | coppice, that is indeed a better plan! |
| 14:46.12 | jaytee | "It appears you've been missing alot of work lately" "I really haven't missed it, Bob" |
| 14:46.27 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 14:48.09 | gr0mit | ROFL - status report from one of my staff :" Code is now all complete, but does not work" |
| 14:48.11 | *** join/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 14:48.47 | jaytee | hahaha! |
| 14:49.07 | jaytee | the operation was a complete success. unfortunately the patient died. |
| 14:49.07 | Katty | gr0mit: i love it. |
| 14:49.19 | Katty | jaytee: don't say that... |
| 14:49.39 | gr0mit | indeed- i had to laugh and laugh! |
| 14:49.58 | Katty | can you imagine giving someone radiation therapy without proving they have cancer? |
| 14:51.04 | *** join/#asterisk snapple42 (n=snapple4@h216-18-80-132.gtconnect.net) |
| 14:51.13 | tzanger | Katty: I've done that to myself by accident |
| 14:51.17 | tzanger | (playing with radioisotopes) |
| 14:51.56 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.4) |
| 14:55.13 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-32732d9c0f3baa83) |
| 14:55.13 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:55.51 | *** join/#asterisk telnettech (i=telnette@gw.percipia.com) |
| 14:56.04 | Katty | tzanger: don't do that. that's very bad. |
| 14:56.41 | tzanger | Katty: indeed |
| 14:58.08 | *** join/#asterisk SiberAIR (n=SibRphre@ip67-93-6-162.z6-93-67.customer.algx.net) |
| 14:58.28 | *** join/#asterisk beniwtv (n=beniwtv@124.Red-83-36-62.staticIP.rima-tde.net) |
| 14:59.31 | beniwtv | Hi all... anyone knows how I can get the IP address from a conneted H.323 soft phone via the dial plan? I'm using SIP_HEADER for SIP channels, but obviously this doesn't work on H.323, but there might be a similar thing |
| 15:01.04 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:03.48 | *** join/#asterisk mog (n=mog@nat/digium/x-cc586e1affc13458) |
| 15:03.48 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:04.21 | *** join/#asterisk yidiyuehan (n=yidiyueh@bb219-74-226-16.singnet.com.sg) |
| 15:04.52 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 15:06.58 | yidiyuehan | hi, anybody knows any special notice for IP phone working in Pakistan? As right now I have a * server in my Singapore office, and initially IP Phone in Pakistan could remotely register to the * and working fine, however, once I reboot the IP phone it failes to register, if I apply the configuration setting of the phone it will register again. Reboot will fail again...... |
| 15:07.25 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 15:08.26 | etfonhomey | yidiyuehan, what kind of phone? |
| 15:08.38 | yidiyuehan | etfonhomey, SIP IP Phone |
| 15:09.03 | yidiyuehan | and the phone is working fine in other places like Newzea Land, |
| 15:09.59 | etfonhomey | yidiyuehan, what brand (make/model) of phone? |
| 15:10.41 | yidiyuehan | etfonhomey, I tested X-Lite, JR-Tech JR820P(China Brand) |
| 15:11.14 | yidiyuehan | etfonhomey, and IAX2 even fails to register although it's working in other places as well.... |
| 15:12.23 | yang | yidiyuehan: ekiga.org |
| 15:13.36 | yidiyuehan | hi, yang, so you wanna me to try ekiga? |
| 15:13.54 | yang | why not |
| 15:14.23 | etfonhomey | yidiyuehan, what do you mean by "if I apply configuration setting of the phone it will register again"? That seems to say that the phone is losing its configuration after a reboot. |
| 15:14.33 | yidiyuehan | yang, right now I doubt the internet configuration of the pakistan side, as normally they need to set a static IP given by their ISP to their PC |
| 15:14.45 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 15:14.47 | yidiyuehan | etfonhomey, the configuration details are still there, |
| 15:15.05 | yidiyuehan | but it seems that it fails to register to the server or failes to send a request signal |
| 15:15.59 | yidiyuehan | and the funny point is, the same model phone works in many other countries with the same server.............. |
| 15:16.19 | *** part/#asterisk superc (n=superc@ip-90-186-196-159.web.vodafone.de) |
| 15:17.13 | lowtek | yidiyuehan: http://en.wikipedia.org/wiki/Internet_censorship_in_Pakistan |
| 15:17.57 | coppice | how can it be censorship if it only stops after a reboot? |
| 15:17.59 | etfonhomey | yidiyuehan, pastebin the sip.conf for that peer and also the sip debug messages for when that peer attempts to register. |
| 15:18.38 | coppice | its more likely the phone is not saving its config settings properly |
| 15:18.40 | lowtek | I'm not saying it has anything to do with it, but if they have "electronic borders" and filter traffic, it could affect any number of things with this SIP/RTP relationship. |
| 15:19.06 | coppice | those things reset every call, not every reboot |
| 15:19.33 | gr0mit | yidiyuehan, better off using IAX2 and running on a non-standard port |
| 15:19.35 | *** join/#asterisk mort_gib (n=mjensen@177.210.244.195.dsl.static.gibconnect.com) |
| 15:19.38 | yidiyuehan | etfonhomer, hang for a second |
| 15:19.51 | gr0mit | if all else fails, use a tunnel |
| 15:20.30 | yidiyuehan | gr0mit, I have tried iax2 with 4569, it's working in one site of pakistan, however, it completes fails to register in another site, |
| 15:20.46 | coppice | its friday night. if all else fails go down the clark quay and get drunk |
| 15:20.48 | gr0mit | so try some other arbitary port |
| 15:21.09 | gr0mit | a port wellknown for harmless activity |
| 15:21.14 | yidiyuehan | like 8090? |
| 15:21.27 | gr0mit | well, pick your ports! |
| 15:21.29 | *** join/#asterisk linuxer_igor (n=linuxer_@mlsrj200152096p202.mls.com.br) |
| 15:21.35 | coppice | clark quay is a port |
| 15:21.38 | yidiyuehan | but it doesn't make sense if 4569 is not blocked in pakistan right? |
| 15:22.23 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 15:24.13 | yidiyuehan | etfonhomer, here is the past bin for SIP client |
| 15:26.36 | artakamiraghyan | Hi |
| 15:26.56 | artakamiraghyan | is anyone familiar with Sangoma A104? |
| 15:28.41 | [TK]D-Fender | artakamiraghyan: Plenty of people are. Just ask your question |
| 15:30.02 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 15:30.25 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 15:32.00 | beniwtv | so anyone knows an answer to my question? :( |
| 15:33.12 | lowtek | Any idea what would cause this error -> WARNING[6293]: translate.c:175 framein: no samples for g729tolin |
| 15:33.21 | [TK]D-Fender | beniwtv: "core show functions" <- go read the list. If you don't see it, it probably doesn't exist |
| 15:33.33 | [TK]D-Fender | beniwtv: And you'll have to make an externals cript to do it. |
| 15:35.06 | bijit | how can I monitor incoming call to a trunk? as blf or any other? |
| 15:35.22 | beniwtv | [TK]D-Fender: It could be an AGI script maybe, but I still would need to know a way to ask Asterisk... hmm... maybe channel variables? Does chan_h323 set any vars that could be of use? |
| 15:35.48 | [TK]D-Fender | bijit: Sure, set up a hint like you would for any other device |
| 15:37.25 | [TK]D-Fender | beniwtv: Set a unique channel variable. then call your AGI. Have your AGI dump each channel "show channel [channel]" until it finds the one where it is set. Then use that to drill to get the H.323 info |
| 15:38.00 | bijit | [TK]D-Fender: ty, I will try that. |
| 15:38.43 | beniwtv | [TK]D-Fender: OK, I'm going to try it :) |
| 15:43.14 | *** join/#asterisk fred-tmft (n=fred-tea@c-69-244-180-112.hsd1.mi.comcast.net) |
| 15:43.32 | *** part/#asterisk fred-tmft (n=fred-tea@c-69-244-180-112.hsd1.mi.comcast.net) |
| 15:43.45 | mort_gib | I need some help with a Polycom IP 6000, it keeps saying check_auth: username mismatch, have <450>, digest has <> |
| 15:44.35 | mort_gib | I'm using the webinterface to configure the device, which is not perfect but I only have that same Polycom on that network |
| 15:48.08 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-c38cc5dfda2eb496) |
| 15:48.08 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:48.44 | angryuser | mort_gib: hello try to activate unsecure=very |
| 15:48.54 | errr | anyone know of a good plugin for outlook that works with vista (both 32 and 64bit) I tried outcall http://code.google.com/p/outcall/ but it crashes and wont run on 2k7 outlook on vista 64b |
| 15:49.06 | mort_gib | in sip.conf?? |
| 15:49.18 | angryuser | mort_gib: yes |
| 15:49.25 | mort_gib | Ok, hang on! |
| 15:49.51 | [TK]D-Fender | angryuser: thats INSECURE, and should be "port,invite" on 1.4+ |
| 15:49.52 | angryuser | mort_gib: somehow your polycom is not sending it's name 450 |
| 15:50.42 | angryuser | [TK]D-Fender: yes allways miss typing that word ;) |
| 15:51.42 | mort_gib | angryuser: Strange though.... |
| 15:52.07 | mort_gib | No, same s**t |
| 15:52.26 | angryuser | do as fender said insecure=port,invite |
| 15:52.31 | mort_gib | tk: used insecure=port,invite |
| 15:52.36 | mort_gib | same problem |
| 15:53.37 | angryuser | do a sip trace see what is sent by that phone, check your credentials |
| 15:53.41 | etfonhomey | mort_gib, let me check the web interface of my polycom. It's been a while since I've used it. |
| 15:54.20 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.130.105) |
| 15:54.22 | mort_gib | etfonhomey: I know, but I only have XP laptops and a OpenBSD firewall in that office... |
| 15:54.30 | mort_gib | -And a load of Snoms :-) |
| 15:54.48 | etfonhomey | mort_gib, pastebin the sip.conf for the peer that you're having problems with. |
| 15:56.04 | mort_gib | http://pastebin.com/d38206c04 |
| 15:56.14 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 15:56.48 | mort_gib | I added username yesterday..... didn't work before or after... |
| 15:56.49 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 15:57.47 | etfonhomey | mort_gib, You made your settings on the Polycom under the Line 1 section of the web interface? |
| 15:58.09 | mort_gib | Hang on, I'll put it up |
| 15:58.18 | etfonhomey | k. |
| 16:02.10 | mort_gib | etfonhomey: Damn, ftp is blocked I gotta go to the clients office hang on a few minutes! |
| 16:02.20 | etfonhomey | mort_gib, I'll be around. |
| 16:02.26 | mort_gib | Thanks! |
| 16:02.31 | etfonhomey | np |
| 16:06.46 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 16:07.04 | bijit | [TK]D-Fender: That means on sip.conf I have to put on all trunks call-limit=99 in order for blf to work right? |
| 16:08.07 | [TK]D-Fender | bijit: "insert large # here" |
| 16:11.17 | *** join/#asterisk telnettech (i=telnette@gw.percipia.com) |
| 16:12.23 | *** join/#asterisk Pagautas (n=bigman@ns.voip.ktu.lt) |
| 16:13.58 | *** join/#asterisk jackson__ (n=jackson@68-115-84-188.dhcp.roch.mn.charter.com) |
| 16:18.03 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-234.gibnet.gi) |
| 16:21.44 | *** join/#asterisk outtolunc (n=me@c-67-188-204-139.hsd1.ca.comcast.net) |
| 16:27.20 | mort_gib | etfonhomey: have a look at www.heimdal-systems.com/Screenshot.png |
| 16:27.27 | etfonhomey | mort_gib, found something for you |
| 16:27.32 | mort_gib | Yeah?? |
| 16:27.33 | etfonhomey | http://www.polycom.com/usa/en/support/voice/soundpoint_ip/VoIP_Technical_Bulletins_pub.html |
| 16:27.42 | etfonhomey | 44011 at the very bottom |
| 16:27.53 | etfonhomey | Your sip.conf looks good. |
| 16:28.05 | etfonhomey | You only need 405 in the address field. |
| 16:28.19 | etfonhomey | (of line 1) |
| 16:28.42 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:29.40 | etfonhomey | mort_gib, sorry, my info was incomplete. You have a secret= in your sip.conf entry for that peer. Throw that in the auth password field. |
| 16:32.54 | *** part/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 16:32.57 | mort_gib | etfonhomey: Tried that already, reading that doc now |
| 16:32.58 | *** join/#asterisk uri_ (n=uri@84.201.225.66) |
| 16:34.30 | *** join/#asterisk Zeeek (n=Zeeek@bdx.resmo.net) |
| 16:34.44 | Zeeek | HAPPY BIRTHDAY ASTERISKKKKKKKKKKK!!!!!! |
| 16:34.48 | digime | hi....... how can i make an alarm clock for asterisk that calls me every day at a certain time? |
| 16:35.13 | Zeeek | digime: croin job and call file |
| 16:35.57 | etfonhomey | mort_gib, under Transport, why did you change it from the default? |
| 16:36.01 | outtolunc | happy bday |
| 16:36.10 | Zeeek | in about ten minutes we open the doors to the party fro asterisk's 9th birthday over at http://voipusersconference.org |
| 16:36.20 | digime | aha ok |
| 16:36.32 | outtolunc | wonders if mark fired up the old hottub we bought him |
| 16:36.43 | mort_gib | etfonhomey: recommendation from a website, but I tried with that setting and the default |
| 16:37.08 | mort_gib | Still no go |
| 16:37.32 | coppice | I think its just about 9 years since the first zapata card ran, too |
| 16:38.09 | etfonhomey | mort_gib, can you give me a sshot of the Servers section under SIP on the web UI? |
| 16:38.10 | coppice | it was some time in December. I only got an E1 version running in January, though :-) |
| 16:38.32 | mort_gib | Sure |
| 16:38.43 | mort_gib | Changed back to defaults |
| 16:39.31 | *** join/#asterisk invalidrecord (n=fares@87.113.52.178) |
| 16:39.41 | invalidrecord | evening |
| 16:40.01 | jasonwoot | not in florida it's not |
| 16:40.20 | invalidrecord | lol |
| 16:40.50 | invalidrecord | but you have ocalla so we ignore you ;l-) |
| 16:42.11 | mort_gib | etfonhomey: www.heimdal-systems.com/Screenshot2.png |
| 16:42.19 | Zeeek | <PROTECTED> |
| 16:45.13 | etfonhomey | mort_gib, are you remotely connecting to this phone? |
| 16:45.36 | mort_gib | I'm in the webinterface, but it's in another office |
| 16:45.40 | mort_gib | no NAT |
| 16:46.06 | *** join/#asterisk mikealeonetti (n=mikel@static-72-68-153-122.nycmny.fios.verizon.net) |
| 16:46.13 | digime | zeeek: do you know of an example online with code? i see alarms for asterisk but they are all gui based |
| 16:47.03 | *** join/#asterisk n3hxs (n=HAMming@63.68.135.4) |
| 16:47.33 | etfonhomey | mort_gib, here's my sip.conf: http://pastebin.com/d40685fa5 |
| 16:49.03 | etfonhomey | mort_gib, putting up screenshots of my working setup. One sec. |
| 16:49.38 | mikealeonetti | when people call in to my company from other SIP phones, my Asterisk doesn't "hear" the DTMF tones generated by the phone. Did I set something up wrong? |
| 16:50.24 | *** join/#asterisk japerry (n=japerry@drupal.org/user/45640/view) |
| 16:50.37 | Zeeek | http://voipusersconference.org |
| 16:51.02 | Zeeek | digime: call and ask your questions on the conference |
| 16:51.09 | digime | cool! |
| 16:51.47 | Zeeek | bye for now |
| 16:52.41 | *** part/#asterisk Zeeek (n=Zeeek@bdx.resmo.net) |
| 16:55.14 | bijit | I have three incoming DID to a same exten. How can I identify by which incmoning trunk is it coming? On the screen of the phone. I've done it through blf thanks to fender. |
| 17:00.58 | *** join/#asterisk CunningPike (n=arodgers@204.239.10.119) |
| 17:02.45 | *** join/#asterisk c4t3l (i=rcallico@equinox.alluvium.com) |
| 17:02.59 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 17:03.02 | c4t3l | hello world |
| 17:06.05 | *** join/#asterisk flohack (n=fhackenb@lancelot.acoveo.com) |
| 17:06.12 | mikealeonetti | hi |
| 17:06.56 | *** join/#asterisk JerJer (n=PhatJ@24-236-197-165.dhcp.aldl.mi.charter.com) |
| 17:07.09 | flohack | Hi! Does someone know how orderlyq works internally? How do they force asterisk to connect a queued call to an agent and how do they play announcements to the caller which is in the asterisk queue? |
| 17:08.32 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 17:08.40 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 17:12.47 | *** join/#asterisk japerry (n=japerry@drupal.org/user/45640/view) |
| 17:14.39 | etfonhomey | mort_gib, any luck? |
| 17:15.38 | telnettech | anybody know of a setting that would affect sending of DTMF over a T-1 from the Asterisk box to an IVR |
| 17:18.08 | angryuser | telnettech: remote IVR ? when you call peer>>T1 >> remote server's IVR ? |
| 17:19.28 | mort_gib | Nope, this really pisses me off... |
| 17:19.47 | mort_gib | I NEVER have any issues with the Snoms |
| 17:20.08 | mort_gib | provisioning or web configuration |
| 17:20.11 | telnettech | angry: site is calling a toll free number to whoever which gets an IVR menu and asked them to choose a number. They are pressing the number to correspond to their choice. It is not responding to the DTMF tones sent from the Asterisk box. It has a T-1 connection from the asterisk box to the Telco provider |
| 17:20.21 | mort_gib | all works really easy and smooth |
| 17:20.34 | telnettech | Im not sure what is on the other end but customer is adamant that it is the asterisk box problem, not far end |
| 17:22.05 | angryuser | telnettech: set the dtmf=auto to the peer calling, and try |
| 17:22.09 | troy- | happy bday asterisk |
| 17:22.13 | angryuser | or rfc2833 |
| 17:22.21 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:22.49 | flohack | Hi! Does someone know how orderlyq works internally? How do they force asterisk to connect a queued call to an agent and how do they play announcements to the caller which is in the asterisk queue? |
| 17:23.02 | telnettech | under the chans_zap.conf? |
| 17:23.29 | *** join/#asterisk diter (n=martin@83.140.111.160.dyn.rp80.se) |
| 17:24.05 | angryuser | telnettech: sip.conf for the peer who is calling |
| 17:24.27 | telnettech | no they are calling out, not into the asterisk box |
| 17:24.40 | telnettech | and it is not just 1 user it is whols site |
| 17:24.57 | *** join/#asterisk Segnale007 (n=Pietro@host247-96-dynamic.7-87-r.retail.telecomitalia.it) |
| 17:24.58 | angryuser | telnettech: please set and try, anyway i am goint out good day ;) |
| 17:25.03 | angryuser | going* |
| 17:25.24 | telnettech | angry i will try |
| 17:34.38 | mikealeonetti | I actually have the same issue |
| 17:34.45 | mikealeonetti | and I will try it as well |
| 17:35.50 | *** join/#asterisk `Sean (i=Un1x@CPE001cc031aa0d-CM0014045acc3c.cpe.net.cable.rogers.com) |
| 17:37.36 | *** join/#asterisk M1s3ry (n=M1s3ry@nat/digium/x-92e23c614d92766a) |
| 17:40.40 | *** join/#asterisk ratdeptrai (n=dfavro@d14-69-67-36.try.wideopenwest.com) |
| 17:45.29 | ratdeptrai | Hi, does anyone have any suggestions for a GUI (or TUI) "manager" app for asterisk that can see current calls, barge in, kick calls, etc. (i.e. the typical AMI functions), that is open-source (other than flash-operator-panel) and runs on Linux? |
| 17:47.43 | ratdeptrai | I tried flash-operator-panel and gastman but neither seemed satisfactory for me. |
| 17:49.04 | *** join/#asterisk Xaviertoor (i=User@189-015-64-095.xd-dynamic.ctbcnetsuper.com.br) |
| 17:51.24 | tzafrir_laptop | ratdeptrai, monast (see #monast) |
| 17:51.37 | tzafrir_laptop | Not as mature as FOP, but quite interesting |
| 17:53.13 | ratdeptrai | tzafrir_laptop: Thanks, I'll take a look at that. |
| 17:53.28 | flohack | Does someone know how orderlyq works internally? How do they force asterisk to connect a queued call to an agent and how do they play announcements to the caller which is in the asterisk queue? |
| 17:57.16 | diter | Hi I have two analog telephones using the same fxs port. If I anwser in one phone I can no longer close the call and answer in the other phone. Is there som smart way to use the cabels in the walls to get to seperet phones ? |
| 17:57.47 | Qwell | diter: no |
| 17:57.52 | Qwell | you need 1 FXS per "line" |
| 17:57.56 | diter | My digital ATA box have two fxo |
| 17:58.24 | diter | sorry meen two fxs |
| 17:59.02 | diter | Qwell so I need to draw new cabels to the box ? |
| 17:59.04 | Qwell | diter: please elaborate. you're saying 2 phones on 1 fxs, but you're also saying you have 2 fxs |
| 18:02.46 | diter | Yes I have 2 fxs and whant to do new cabels. But my parents think it will look uggly when I draw cabels and whant too use the ones in wall. |
| 18:07.03 | *** join/#asterisk Bad_Robot- (n=Bad_Robo@cpe-76-173-219-25.socal.res.rr.com) |
| 18:07.22 | [TK]D-Fender | diter: YES |
| 18:07.30 | *** part/#asterisk mikealeonetti (n=mikel@static-72-68-153-122.nycmny.fios.verizon.net) |
| 18:09.25 | [TK]D-Fender | diter: RJ11 supports 3 pairs. You'll need a spltter adapter to bring it to the SPA but from which point standard wiring allows you to do what you require for separate phones |
| 18:10.04 | Qwell | 3 pairs? yuck... |
| 18:10.15 | Qwell | I highly doubt there's more than 2 in any existing wiring... |
| 18:10.18 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 18:10.25 | Qwell | at least in a house |
| 18:10.56 | Qwell | It's RJ11 6p4c? |
| 18:10.58 | diter | D-Fender: Nice I will try too read more about this RJ11 and 3 pair splitter. Thanks |
| 18:13.00 | Qwell | s/It's/Isn't/ |
| 18:13.04 | invalidrecord | if im building an exchange to handle several companies, should I give each one its own context ? |
| 18:13.35 | invalidrecord | and if so how can i define a prefix so i can dial extensions in each |
| 18:13.53 | *** join/#asterisk LeddyHM (n=none@polar.artica.net) |
| 18:14.29 | Katty | happy |
| 18:25.14 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-4c8f970a9e5c7a94) |
| 18:25.14 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 18:26.26 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-10-13.w86-215.abo.wanadoo.fr) |
| 18:26.40 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-10-13.w86-215.abo.wanadoo.fr) |
| 18:30.41 | *** join/#asterisk qdk (n=qdk@212.27.19.21.bredband.3.dk) |
| 18:35.57 | *** join/#asterisk Psychobilly (n=moi@adsl154-96.kln.forthnet.gr) |
| 18:38.58 | Psychobilly | hello, can somebody suggest a way to access (read/write) the astdb outside asterisk, asterisk manager etc? Im trying with perl and DB_File but i m not very lucky till now |
| 18:39.43 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
| 18:41.24 | [netman] | uses rasterisk -x from bash |
| 18:43.16 | *** join/#asterisk moy (n=moy@189.169.89.20) |
| 18:49.50 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 18:52.49 | *** join/#asterisk chi6IT41 (n=chigital@tmo-100-213.customers.d1-online.com) |
| 18:56.50 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 19:05.19 | *** join/#asterisk intrin (n=intrin@99-196-130-98.cust.wildblue.net) |
| 19:07.57 | *** join/#asterisk M1s3ry (n=M1s3ry@nat/digium/x-6931138422d934f1) |
| 19:08.46 | ratdeptrai | Psychobilly: like netman, I would: "asterisk -r -x 'database put foo bar blaf'" but it's slow as heck. I think it's Berkley format, you could probably directly access the files, but I gave up on astdb some time ago for reasons like these, and now use external DBs. |
| 19:09.07 | Psychobilly | yes i see |
| 19:10.04 | Psychobilly | its Berkeley db v1, DB_File is supposed to be able to access it |
| 19:10.38 | seanbright | are you getting some kind of error while trying to access it? |
| 19:10.41 | Psychobilly | maybe i have to recompile DB_File with db 1.86 |
| 19:11.01 | Psychobilly | seanbright it doesnt read it, and when i cerate a db with it its a different version thanh astdb |
| 19:11.10 | seanbright | i see. |
| 19:11.20 | Psychobilly | $ file astdb |
| 19:11.20 | Psychobilly | astdb: Berkeley DB 1.85/1.86 (Btree, version 3, native byte-order) |
| 19:11.27 | Psychobilly | the original astersisk db |
| 19:11.35 | Psychobilly | $ file astdb.cust |
| 19:11.35 | Psychobilly | astdb.cust: Berkeley DB (Btree, version 9, native byte-order) |
| 19:11.43 | Psychobilly | the one i generated with DB_File |
| 19:11.46 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
| 19:12.23 | seanbright | interesting |
| 19:12.24 | Psychobilly | i guess db_file needs recompile againts db-1.86 |
| 19:12.28 | seanbright | mayhaps |
| 19:12.40 | Psychobilly | dabian doesnt have support for it i guess |
| 19:13.17 | seanbright | can you pastebin the code you are using? i want to try it here |
| 19:13.39 | Psychobilly | sure |
| 19:15.29 | *** join/#asterisk patrick-- (n=patrick@gate.devnull.biz) |
| 19:15.53 | patrick-- | Hey there. is it possible to record an mISDN channel and get just one wav/gsm/mp3 whatever fileformat out of it ? |
| 19:16.16 | Psychobilly | http://pastebin.com/m282f9a49 seanbright |
| 19:16.36 | Psychobilly | this is for reading if u want to write uncomment the proper lines |
| 19:16.44 | Psychobilly | very simple script |
| 19:22.21 | *** join/#asterisk thepacmanfan (n=thepacma@12-218-136-3.client.mchsi.com) |
| 19:23.40 | thepacmanfan | are there any compelling reasons not to make my asterisk server do double-duty as an internal webserver? |
| 19:24.08 | patrick-- | Hey there. is it possible to record an mISDN channel and get just one wav/gsm/mp3 whatever fileformat out of it ? |
| 19:25.20 | thepacmanfan | i don't want voice traffic to suffer, but the machine should be more than capable of performing both functions at once. i don't need the webserver to be public, so it shouldn't put the machine at a higher security risk. |
| 19:25.42 | *** join/#asterisk gangans (n=LL@84.36.131.35) |
| 19:27.35 | thepacmanfan | it should be even less of an issue if i throw another NIC in the server, and connect it directly to the regular network, rather than through my dedicated VoIP switch. |
| 19:28.48 | thepacmanfan | asterisk is handling 10 SIP users, and 5 analog lines through dahdi. |
| 19:30.31 | seanbright | Psychobilly: yeah... not working. weird. |
| 19:31.14 | Psychobilly | seanbright propably needs rebuilding against db v1 |
| 19:34.58 | ratdeptrai | thepacmanfan: depends on the machine, but I would think it would be fine, especially if the web-server is only serving static pages (no heavyweight middleware apps), and you keep asterisk at a higher scheduling priority than the web-server. |
| 19:36.22 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:40.06 | *** join/#asterisk JBrunk (n=jbrunk@rrcs-71-40-184-56.se.biz.rr.com) |
| 19:42.24 | [TK]D-Fender | thepacmanfan: Should be just fine |
| 19:43.01 | JBrunk | i have asterisk 1.6 running on an esxi server, connecting to teliax by iax. i get horrible warble sounds from other calls, but not * itself, all internal messages are fine |
| 19:43.14 | JBrunk | anyone run into this? |
| 19:43.16 | [TK]D-Fender | thepacmanfan: your * needs could be done off a sub GHZ PC. So 1 even vaguely modrn spec system could much more than easily handle this |
| 19:44.46 | grndslm | anybody have any experience with the atcom au-600 ?? |
| 19:45.04 | grndslm | could it be better than an spa ata? |
| 19:47.08 | *** join/#asterisk fabbari (n=mrwho@89-97-140-219.ip17.fastwebnet.it) |
| 19:47.08 | fabbari | Hi all! |
| 19:48.37 | fabbari | I have the following problem: I'm using messagenet as provider, it is defined as a peer and I register for it. If in the peer section I don't put the secret I get the calls coming in, but I can't dial out. If I put the secret I can call out, but I don't get the incoming calls - unauthorized. |
| 19:48.52 | thepacmanfan | ratdeptrai, [TK]D-Fender: the server has a 1.3ghz Celeron CPU, and 256mb of RAM, so hardware-wise it's fine. as an internal webserver, the heaviest thing it will be running is Mediawiki, with no more than 10 simultaneous users. |
| 19:48.55 | fabbari | Should I post the relevant sip.conf? |
| 19:50.12 | thepacmanfan | network congestion is the only performance concern i have; if i'm FTP'ing a large file on or off or the server, voice data could suffer a little... i'm not sure how to do QoS. |
| 19:50.31 | fabbari | I get "Found peer 'messagenet_out' for '<phonenumber>' from 212.97.59.76:5061" |
| 19:50.41 | thepacmanfan | however, i could just add another NIC, and configure Asterisk to use a separate interface than everything else does. |
| 19:50.56 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 19:56.45 | *** join/#asterisk StephenF (n=none@198.144.201.106) |
| 19:57.21 | StephenF | Anyone have any experience with the Polycom SoundPoint 450? |
| 19:57.36 | *** join/#asterisk CGMChris (n=chris@mail.cgmyes.com) |
| 19:57.58 | StephenF | Is it new? |
| 19:58.08 | CGMChris | is there a way to get/use variables from sip.conf (such as "fullname") in my dialplan? ${fullname} doesnt seem to do the trick. Thoughts? |
| 19:59.40 | telnettech | ok guys need some more help (TK and ET?)....I have a customer that is calling a toll free number that the IVR is not responding to DTMF interactions.....i routed over both T-1. Earlier someone on here asked me to put dtmf=auto in sip.conf which i did....If the customer calls the Local Telco number that the Toll free points to the DTMF interaction is fine....any ideas...it is asterisk 1.4.18 with zaptel channels |
| 20:01.37 | *** join/#asterisk LND (n=lee@fazer1-adsl.demon.co.uk) |
| 20:03.01 | *** part/#asterisk LND (n=lee@fazer1-adsl.demon.co.uk) |
| 20:04.48 | *** join/#asterisk dbii (n=dbii@rrcs-67-79-196-168.sw.biz.rr.com) |
| 20:05.04 | flohack | Does someone know how orderlyq works internally? How do they force asterisk to connect a queued call to an agent and how do they play announcements to the caller which is in the asterisk queue? |
| 20:08.04 | [TK]D-Fender | thepacmanfan: I highly doubt you have to care... VoIP traffic is irrelevent |
| 20:09.42 | [TK]D-Fender | CGMChris: Core config values? Or just some custom ones you want to define in each peer for your own purposes? |
| 20:10.25 | [TK]D-Fender | StephenF: Yes, its new. It sits between the ip50X & IP430 and just beloy the 550 in terms or position within the family. |
| 20:10.36 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 20:10.38 | [TK]D-Fender | StephenF: Polycom is introducing too many models now. |
| 20:11.04 | *** join/#asterisk etfonhomey (n=chatzill@www2.askpri.org) |
| 20:11.20 | *** join/#asterisk mohawk (n=mohawk@host217-40-110-154.in-addr.btopenworld.com) |
| 20:12.12 | bijit | which is best for connecting like 6 asterisk box together? iax or sip? |
| 20:13.01 | [TK]D-Fender | bijit: Doesn't matter |
| 20:14.52 | killfill | hey how do i make asterisk show the http requests at port 8088? |
| 20:14.58 | killfill | cannot find my file.. :S |
| 20:15.12 | bijit | even if its miles away? |
| 20:17.54 | bijit | and limited bandwidth 1mb between links. |
| 20:19.19 | x86 | hate how chan_sip adds 'hostname' to the allowed SIP domains, but not 'hostname.f.q.d.n" |
| 20:19.29 | x86 | (even with autodomain = yes) |
| 20:22.51 | fabbari | Using INVITE request as basis request - 29a031252f6df33b2a767e692878a079@sip.messagenet.it |
| 20:22.51 | fabbari | No user '+xxxxxxxxxxxx' in SIP users list |
| 20:22.51 | fabbari | Found peer 'messagenet_out' for '+xxxxxxxxxxxxxxx' from 212.97.59.76:5061 |
| 20:23.49 | fabbari | Then if peer messagenet_out has a secret defined the call is rejected as "Unauthorized". |
| 20:24.19 | fabbari | If I don't define the "secret" the calls can get through, but i won't be able to call out via the same peer. |
| 20:27.51 | fabbari | Asterisk 1.6.0 |
| 20:28.00 | fabbari | ...does anyone have any hint' |
| 20:28.01 | bijit | ~pb |
| 20:28.02 | jbot | [~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://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 20:28.02 | fabbari | ? |
| 20:28.21 | *** join/#asterisk Toerkeium (n=Miranda@201.216.206.221) |
| 20:28.25 | fabbari | sorry.. |
| 20:28.44 | Toerkeium | does anyone know of a softphone for windows CE 6 proffesional? |
| 20:29.12 | Toerkeium | it's a PPC HTC touch |
| 20:29.18 | [TK]D-Fender | fabbari: "insecure=port,invite" |
| 20:29.33 | Toerkeium | I was using SJPhone but it doesn't support windows 6 |
| 20:31.48 | fabbari | tk Thanks! |
| 20:31.58 | StephenF | [TK]D-Fender I feel like the 330 is a little limited for regular office workers, The 450 could be a nice replacement |
| 20:32.15 | telnettech | <PROTECTED> |
| 20:32.16 | telnettech | on is fine....any ideas...it is asterisk 1.4.18 with zaptel channels |
| 20:32.20 | StephenF | I dont know why they dont phase out the 501, 601, etc... |
| 20:32.49 | StephenF | or the 501 and 430 I mean |
| 20:33.01 | telnettech | by the way, it is an elastix gui if that makes a difference |
| 20:33.21 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 20:39.53 | *** join/#asterisk `Sean (i=Un1x@CPE001cc031aa0d-CM0014045acc3c.cpe.net.cable.rogers.com) |
| 20:41.48 | Thorn | hello |
| 20:42.17 | Thorn | is it possible to see which zap fxo channels are currently busy? 'zap show channels' doesn't show status |
| 20:42.54 | CGMChris | [TK]D-Fender: Core config values (i think), assuming "fullname" is a core config value. |
| 20:53.53 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 20:54.45 | [TK]D-Fender | CGMChris: "core show functions like SIP" |
| 20:55.11 | [TK]D-Fender | CGMChris: You could duplicate this as well with extra parallel channel variables |
| 21:01.29 | Thorn | I have set up a queue. when agents receive calls, the caller id is 'asterisk'. is it possible to change it (to queue name for example)? |
| 21:03.51 | [TK]D-Fender | Thorn: "core show function CALLERID" |
| 21:05.30 | Thorn | [TK]D-Fender: cool, didn't know about that, thanks |
| 21:06.07 | Thorn | so syntax would be Set(${CALLERID(name)}=foo} ? |
| 21:08.49 | [TK]D-Fender | Thorn: Yes |
| 21:08.59 | [TK]D-Fender | Thorn: Sorry... CLOSE |
| 21:09.09 | *** join/#asterisk PanGoat (n=PanGoat@node2.sensoryresearch.net) |
| 21:09.10 | *** join/#asterisk sprite-- (n=sprite@12.228.1.97) |
| 21:09.23 | [TK]D-Fender | Thorn: Set(CALLERID(name)=foo} |
| 21:09.40 | Thorn | right, I just found that too |
| 21:10.54 | *** join/#asterisk LND (n=lee@fazer1-adsl.demon.co.uk) |
| 21:11.18 | *** join/#asterisk neurosys (n=neurosys@c-66-229-81-161.hsd1.fl.comcast.net) |
| 21:15.42 | *** join/#asterisk kb3ien (n=kb3ien@isl177-max1.accesshighway.net) |
| 21:17.08 | *** join/#asterisk naif (n=naif@62-101-65-16.ip.fastwebnet.it) |
| 21:17.58 | naif | so |
| 21:18.13 | naif | guys i need one little help for something particular to play with asterisk ISDN DATA CALLS |
| 21:18.53 | naif | i need to call from a GSM phone with CSD V110 |
| 21:18.58 | naif | with two GSM phone |
| 21:19.09 | naif | an ISDN number with at least two lines |
| 21:19.23 | naif | GSM1 -> ISDN | sw pipe| |
| 21:19.30 | naif | GSM1 -> ISDN |sw pipe| |
| 21:19.40 | naif | GSM2 -> ISDN |sw pipe| |
| 21:19.54 | naif | so that GSM1 data call and GSM2 data calls going to the same ISDN number |
| 21:20.05 | naif | will shortcircuit their input/output data buffers |
| 21:20.12 | naif | like if they where GSM1 <--> GSM2 |
| 21:20.24 | naif | but with in the middle GSM1 <--> ISDN <--> GSM2 |
| 21:20.28 | naif | transparently |
| 21:22.09 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:22.26 | naif | i found several way |
| 21:22.55 | naif | app_pppd or ZapRAS replacing system pppd binary, app_pipe with mISDN |
| 21:23.00 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 21:23.14 | naif | but it would need some coding maybe |
| 21:24.01 | [TK]D-Fender | checkout time. Heading home, later all. |
| 21:30.56 | *** join/#asterisk Shotygun (n=thorn@DSL212-235-83-213.bb.netvision.net.il) |
| 21:31.20 | *** join/#asterisk freakazoid0223 (n=mattc@pool-68-238-188-183.phil.east.verizon.net) |
| 21:34.59 | *** part/#asterisk JBrunk (n=jbrunk@rrcs-71-40-184-56.se.biz.rr.com) |
| 21:42.00 | diter | How do I get Dial(SIP/diter,30,tr) to play back a tone beep to let pepole they are ringing me ? |
| 21:46.13 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 21:46.34 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 21:52.20 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:57.20 | *** join/#asterisk qdk (n=qdk@79.138.254.85.bredband.3.dk) |
| 21:59.00 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 22:00.51 | *** join/#asterisk glaz (i=strke@glaciuz.com) |
| 22:01.58 | glaz | took the dCAP exam today! *sigh* |
| 22:02.33 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 22:02.35 | *** join/#asterisk davidef1986 (n=davide@2002:4e86:4444:a:21f:3bff:fe98:a9b1) |
| 22:05.33 | bkruse | glaz: pass? |
| 22:05.47 | bkruse | I need to take that... |
| 22:05.56 | glaz | bkruse: I got my practical result only |
| 22:06.05 | glaz | I did pass, waiting for the theorical part result |
| 22:06.16 | bkruse | written vs lab? |
| 22:06.16 | glaz | I got 106% in the lab |
| 22:06.21 | glaz | yup |
| 22:06.28 | bkruse | Nice Nice |
| 22:06.31 | bkruse | Where did you take it at? |
| 22:06.41 | glaz | at aheeva.com |
| 22:06.57 | bkruse | Oh, interesting. Well congrats! |
| 22:07.12 | glaz | thanks, but you have to pass both at 75% to get it |
| 22:07.18 | glaz | but I'm pretty confident. |
| 22:07.26 | bkruse | become an asterisk expert in 5 days |
| 22:07.30 | bkruse | well, that makes me obsolete :) |
| 22:07.52 | bkruse | glaz: Right, I think you will do well. Usually it's the lab that people get tripped up on, if they do |
| 22:08.00 | glaz | I doubt the bootcamp is enough to get dCAP.. |
| 22:08.59 | bkruse | Bah, it's just an exam. The test is when you are on site and they give you shell access and say go, or rather, when things break :) |
| 22:09.20 | glaz | well the lab is quite hard |
| 22:09.33 | glaz | you have 90 mins to setup a PBX from scratch |
| 22:09.37 | bkruse | Heh, I don't think it would be to bad :) |
| 22:09.39 | glaz | even libpri is not installed yet |
| 22:09.41 | bkruse | For me anyways |
| 22:09.56 | bkruse | I am glad you did well though, good thing to throw on the resume! |
| 22:10.16 | glaz | you have to setup a T1 and a Analog line, a SIP and/or IAX trunk |
| 22:10.24 | glaz | few features, its fun. |
| 22:10.28 | glaz | its a nice test |
| 22:10.40 | bkruse | I have done many many many times :) |
| 22:10.47 | *** part/#asterisk davidef1986 (n=davide@2002:4e86:4444:a:21f:3bff:fe98:a9b1) |
| 22:11.04 | bkruse | glaz: How long have you been using Asterisk? |
| 22:11.06 | glaz | yeah so did I, but still fun after all these years! |
| 22:11.17 | glaz | humm, 2 years I'd say |
| 22:11.21 | glaz | started with 1.2 |
| 22:11.24 | bkruse | Nice |
| 22:11.31 | bkruse | I have a 1.0 copy of BE in my room :D |
| 22:11.47 | glaz | oh, I have the 1.0 BE as well |
| 22:11.49 | bkruse | But I guess 1.2 was around my starting days as well |
| 22:11.55 | bkruse | glaz: Right on! |
| 22:12.20 | *** join/#asterisk bijit (n=benji@201.198.72.142) |
| 22:12.49 | *** join/#asterisk at0mw (n=zoso@ip67-152-66-2.z66-152-67.customer.algx.net) |
| 22:13.01 | glaz | gotta go eat, I'll wait for my written results this week. |
| 22:13.03 | glaz | ttyl |
| 22:13.40 | x86 | anyone familiar with Linksys 962 and 942 phones? |
| 22:13.57 | x86 | I got them both to register, but the 962 wont dial out |
| 22:14.16 | x86 | I know it's registered because it's subscribed to some of my dialplan hints, and it has BLF setup just fine |
| 22:14.21 | glaz | you're missing the sip proxy |
| 22:14.48 | x86 | I've got Proxy and Outbound Proxy both setup on it |
| 22:14.56 | x86 | Register: yes also |
| 22:15.41 | x86 | they both have a flashing red light on the first line key |
| 22:15.52 | x86 | flash flash flash -- wait -- flash flash flash |
| 22:16.04 | x86 | no, just two flashes, wait, two flashes |
| 22:16.24 | x86 | the 942 can call out just fine though |
| 22:16.36 | x86 | and niether is in DND mode either |
| 22:19.13 | x86 | hmm, on the main Info page of the 962, it says Registration State: Failed |
| 22:19.23 | x86 | although Asterisk thinks it registered just fine |
| 22:20.34 | Katty | wonders what an MGI channel is |
| 22:20.37 | x86 | if I hit the flashing red line key, it tells me "Line 1: 7000 (Ready)" |
| 22:20.56 | x86 | any ideas? |
| 22:23.03 | *** join/#asterisk mbranca (n=matteo@93-62-230-243.ip24.fastwebnet.it) |
| 22:23.27 | [TK]D-Fender | x86: PASTEBIN |
| 22:23.46 | Katty | wonders if these el crapo samsung systems have a built in pastebin. |
| 22:24.26 | x86 | [TK]D-Fender: what would you like to see? |
| 22:24.30 | Katty | [TK]D-Fender: the boss told me if i get samsung certified, i get a raise. |
| 22:24.33 | [TK]D-Fender | x86: Everything |
| 22:24.39 | Katty | [TK]D-Fender: and a pretty samsung polo shirt. |
| 22:24.57 | x86 | well AFAIK, I can't pastebin the configs of the phones... and Asterisk seems to be configured correctly |
| 22:25.40 | *** join/#asterisk drgalaxy (n=drgalaxy@adsl-75-55-228-170.dsl.lbcktx.sbcglobal.net) |
| 22:25.50 | x86 | [TK]D-Fender: or is there a way to get the configs out of the 962? |
| 22:26.24 | [TK]D-Fender | x86: IMAGEBIN |
| 22:26.33 | x86 | hmm |
| 22:26.37 | x86 | url? |
| 22:26.37 | [TK]D-Fender | http://imagebin.ca/ |
| 22:26.41 | x86 | sweet |
| 22:27.19 | x86 | now, how to take a screenshot in MacOS X heh |
| 22:28.23 | Katty | hands x86 her camera, and usb cable. |
| 22:29.03 | x86 | hehehe |
| 22:29.30 | Katty | it's easier than putting the lcd panel on the copier machine. |
| 22:30.55 | [TK]D-Fender | \x8You seem to be a running gag of excuses today :p |
| 22:32.32 | *** join/#asterisk DarkRift (n=dark@65.92.170.174) |
| 22:35.04 | jlnt | hey what's the command to meetme |
| 22:35.11 | jlnt | to barge in on a call |
| 22:35.43 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 22:35.51 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 22:37.22 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 22:37.48 | Katty | wonders what this MMC /code/ thing is, and how it works |
| 22:38.15 | x86 | [TK]D-Fender: imagebin.ca is down? |
| 22:38.27 | drgalaxy | is there a way with AMI or AGI to play back a sound file during an established phone call (not in the dial plan)? something like 'barge channel' then 'exec file'? |
| 22:38.51 | x86 | imagebin.org ftw |
| 22:38.53 | [TK]D-Fender | drgalaxy: "core show application chanspy" |
| 22:40.21 | x86 | [TK]D-Fender: http://imagebin.ca/view/7GJ_h9.html |
| 22:40.41 | x86 | [TK]D-Fender: http://imagebin.ca/view/35vgruIt.html |
| 22:40.54 | x86 | [TK]D-Fender: I've got ten screenshots, want urls in /msg? |
| 22:41.05 | x86 | guess I could pastebin the urls |
| 22:41.22 | [TK]D-Fender | x86: And SIP debug of calls, reg attempts, etc |
| 22:43.25 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 22:45.28 | x86 | ok, working on it now |
| 22:46.26 | *** join/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 22:47.36 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-4c8f970a9e5c7a94) |
| 22:50.46 | x86 | [TK]D-Fender: http://pastebin.ca/1277314 |
| 22:50.54 | x86 | there's a LOT of SIP debug there |
| 22:51.45 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 22:53.07 | [TK]D-Fender | x86: I see a hell of a lot of SIP/2.0 401 Unauthorized |
| 22:53.18 | x86 | yeah what's that all about? |
| 22:53.22 | x86 | want to see my sip.conf also? |
| 22:53.39 | x86 | but why is Asterisk saying it's registered properly? |
| 22:53.45 | [TK]D-Fender | x86: of course, and I don't see a good register in there yet |
| 22:54.05 | x86 | and when I go to call extension 7000 from another phone, sip debug shows me that the 962 is rejecting the call saying "Busy here" |
| 22:54.13 | [TK]D-Fender | x86: I see subscribe spam to high hell, but not normal register |
| 22:54.32 | x86 | yeah subscriptions seem to be working fine |
| 22:54.51 | [TK]D-Fender | x86: 191 & 210 |
| 22:54.58 | [TK]D-Fender | x86: your core reg seems bad |
| 22:56.35 | x86 | do I need Use Auth ID? |
| 22:56.47 | x86 | or is just User ID and Password sufficient? |
| 22:58.30 | [TK]D-Fender | x86: W-T-F |
| 22:58.39 | [TK]D-Fender | x86: SRSLY |
| 22:58.45 | x86 | hmm, took it out and it seems to be registered |
| 22:59.02 | x86 | although on my 942 it's setup to use auth ID |
| 22:59.05 | [TK]D-Fender | x86: What the hell are you doing with trying to use the SPA witha SHARELD LINE REG!?! |
| 22:59.21 | x86 | [TK]D-Fender: it works :) |
| 22:59.34 | [TK]D-Fender | x86: * doesn't not support SLA |
| 22:59.43 | x86 | [TK]D-Fender: also have a 932 "sidecar" with 32 line keys |
| 22:59.58 | x86 | [TK]D-Fender: no, it doesn't support SLA, but it does support subscriptions to hints |
| 22:59.59 | [TK]D-Fender | x86: And? thats just presence |
| 23:00.07 | x86 | yeah that's all I want |
| 23:00.13 | [TK]D-Fender | x86: why is your REG set as an SLA reg? |
| 23:00.20 | [TK]D-Fender | http://imagebin.ca/view/7GJ_h9.html |
| 23:00.24 | [TK]D-Fender | x86: SHARED |
| 23:00.32 | x86 | private ftw? |
| 23:00.49 | [TK]D-Fender | x86: Yes |
| 23:01.07 | x86 | cool |
| 23:01.23 | x86 | asterisk doesn't support SLA, but what about SCA? |
| 23:01.46 | [TK]D-Fender | x86: and "user authid=no |
| 23:01.51 | [TK]D-Fender | x86: SCA? |
| 23:02.05 | [TK]D-Fender | x86: Society for Creative Anachronism? |
| 23:02.07 | x86 | aka, can I have a line appearance for a CO line, and be able to check status on it (via a hint or otherwise), and be able to pick it up (aka clamp onto) the line? |
| 23:02.13 | x86 | [TK]D-Fender: shared call appearance |
| 23:02.16 | [TK]D-Fender | x86: NO |
| 23:02.38 | x86 | so the extent of support in asterisk is simply BLF, plain and simple? |
| 23:02.39 | [TK]D-Fender | x86: SCA = SLA |
| 23:02.46 | [TK]D-Fender | x86: correct |
| 23:02.50 | x86 | gotcha |
| 23:02.57 | x86 | that sucks :( |
| 23:03.11 | [TK]D-Fender | x86: As usual you are trying to do everything * does not support and break everything you can in priority sequence :p |
| 23:03.24 | x86 | there is no way to make the line keys do BLF, but if you hit them, make it call a certain extension? |
| 23:03.54 | [TK]D-Fender | x86: Depends on your phone. |
| 23:04.03 | x86 | err, it's a Linksys 962 ;) |
| 23:04.08 | x86 | with 932 sidecar |
| 23:04.10 | [TK]D-Fender | x86: Polycom / Aastra's BLF's are speed-dials as well |
| 23:04.29 | [TK]D-Fender | x86: I don't know Linksys for sure, but I'd suspect they can. |
| 23:04.40 | [TK]D-Fender | (if they don't totally suck) |
| 23:04.48 | x86 | do you see where in the configs that it would support it? |
| 23:05.24 | x86 | hey man, changing the reg from shared to private made the light go solid green instead of flashing red :) |
| 23:07.12 | x86 | [TK]D-Fender: I bet the cryptic BLF subscription format that Linksys uses supports speed-dial also... |
| 23:07.25 | x86 | just having trouble finding documentation on the cryptic format |
| 23:08.57 | [TK]D-Fender | x86: JFGI |
| 23:08.58 | [TK]D-Fender | :) |
| 23:09.07 | x86 | IFTA :) |
| 23:09.19 | x86 | this is what I'm talking about: |
| 23:09.19 | x86 | fnc=blf+sd+cp;sub=7001@10.42.69.252;nme=User B |
| 23:10.06 | x86 | wait, the SD part of that is supposed to be speed dial |
| 23:10.07 | x86 | hmm |
| 23:10.23 | x86 | BLF, Speed Dial, Call Pickup |
| 23:10.47 | [TK]D-Fender | x86: CP = BS |
| 23:11.03 | CunningPike | Hey, hey, hey, now |
| 23:11.05 | [TK]D-Fender | x86: More shit you can't do because * DOES NOT SUPPORT SLA |
| 23:11.10 | *** join/#asterisk xxoxx (n=xxoxx@tor/regular/xxoxx) |
| 23:11.33 | x86 | [TK]D-Fender: yes, yes, I know... i grabbed that from some example and have not yet removed it ;) |
| 23:11.47 | x86 | [TK]D-Fender: was only interested in the SD part, which works now |
| 23:11.55 | [TK]D-Fender | x86: so just fnc=blf+sd;sub=7001@10.42.69.252;nme=User B |
| 23:12.07 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:12.17 | x86 | but, when user 7001 makes a call, the light on the 932 for that subscription stays green :( |
| 23:12.26 | *** join/#asterisk brif8 (n=brif8@rrcs-24-227-64-35.se.biz.rr.com) |
| 23:12.40 | [TK]D-Fender | x86: You said you wanted presence... so sd alone won't cut it |
| 23:12.53 | [TK]D-Fender | x86: blf+sd |
| 23:13.26 | brif8 | Just did a Major Upgrade nor running 1.6.0.2 I'm trying to receive calls via teliax and getting check_auth username mismatch, have <teliax>, digest has <s> |
| 23:13.29 | brif8 | error |
| 23:14.04 | x86 | [TK]D-Fender: yes, that's what I have now |
| 23:14.10 | x86 | I just removed the cp only |
| 23:14.28 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 23:14.38 | [TK]D-Fender | x86: Should have presence + SD |
| 23:14.57 | x86 | isnt BLF == presence? |
| 23:15.09 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
| 23:15.11 | [TK]D-Fender | x86: yes |
| 23:15.29 | x86 | right, so I have: fnc=blf+sd;sub=7001@10.42.69.252;nme=User B |
| 23:15.43 | x86 | but when 7001 makes a call, the light stays green |
| 23:16.43 | *** topic/#asterisk by [TK]D-Fender -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0.2 (2008/12/02), 1.4.22 (2008/10/02), *-Addons 1.6.0.1 (2008/12/02), 1.4.7 (2008/06/04), dahdi-linux-complete 2.0.0+2.0.0 (2008/10/02), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev |
| 23:17.13 | [TK]D-Fender | x86: And your dialplan & phone configs please... |
| 23:17.20 | *** mode/#asterisk [-o [TK]D-Fender] by [TK]D-Fender |
| 23:17.22 | x86 | ah, this is interesting... when I make a call from 7001 and do: sip show subscriptions in the CLI, it says Last State: Idle |
| 23:18.05 | x86 | my parking lot subscriptions + BLF work fine, just not this SIP/7001... |
| 23:18.11 | x86 | [TK]D-Fender: un momento, por favor |
| 23:20.44 | *** join/#asterisk citywok (n=chatzill@65.249.42.130) |
| 23:21.00 | citywok | Is there such a thing as a sip sopftphone with programmable buttons? |
| 23:23.09 | x86 | [TK]D-Fender: http://pastebin.ca/1277341 |
| 23:23.19 | x86 | [TK]D-Fender: my entire extensions.conf and sip.conf |
| 23:24.45 | [TK]D-Fender | x86: UGH |
| 23:24.50 | [TK]D-Fender | x86: 1.4+? |
| 23:24.53 | x86 | 1.6 |
| 23:24.56 | [TK]D-Fender | x86: FFS |
| 23:25.02 | x86 | ? |
| 23:25.09 | [TK]D-Fender | x86: "type=peer" , "call-limit=99" |
| 23:25.22 | [TK]D-Fender | x86: friend = DEAD |
| 23:26.08 | x86 | I thought peer could only receive calls (or place calls, but not both) |
| 23:26.26 | [TK]D-Fender | x86: We have got to get you mainstreamed... |
| 23:26.40 | [TK]D-Fender | x86: This has been common knowledge for years.... |
| 23:27.44 | x86 | hmm |
| 23:27.50 | [TK]D-Fender | x86: And I believe you should remove the junk after the SIP/1224 in your hints. |
| 23:27.55 | x86 | I'm still used to 1.2-land ;) |
| 23:28.43 | x86 | 1224? |
| 23:28.47 | [TK]D-Fender | 1234 |
| 23:29.03 | x86 | hmm, not according to Russell Bryant's blog |
| 23:29.09 | [TK]D-Fender | exten => 7001,hint,SIP/7001,User B <- don't add crap |
| 23:29.20 | x86 | ok, i'll take it out |
| 23:29.30 | x86 | that was pulled right off of russellb's blog |
| 23:29.52 | [TK]D-Fender | x86: You don't just ask for trouble, you mail-order it and then forget that its perishable, goes rancid and becomes a bio-hazard |
| 23:30.21 | [TK]D-Fender | x86: And you have more "include"s that actual extens... |
| 23:30.21 | *** join/#asterisk pc500 (n=fewa@24-117-110-20.cpe.cableone.net) |
| 23:30.27 | [TK]D-Fender | x86: mess! |
| 23:30.29 | pc500 | Is there a win32 T.38 "fax modem" driver so I can use windows fax services or similar? |
| 23:30.36 | x86 | I didn't know going by the author of the code's blog was asking for trouble ;) |
| 23:30.43 | x86 | lesson learned heh |
| 23:31.34 | [TK]D-Fender | x86: x86 Status? |
| 23:32.27 | x86 | not working still |
| 23:32.36 | x86 | works as a speed dial, but BLF no workie |
| 23:33.11 | x86 | I did module reload chan_sip.so, as well as dialplan reload |
| 23:33.33 | x86 | and restarted the 7000 phone (the 962 with the 932 side car) |
| 23:33.57 | [TK]D-Fender | x86: order may be important |
| 23:34.24 | x86 | hmm wait |
| 23:34.26 | *** join/#asterisk roxton (n=roxton@24-177-48-50.dhcp.oxfr.ma.charter.com) |
| 23:35.05 | x86 | I called extension 6000 from my 7001 phone (just to put myself in the parking lot) and both the 6001 (first parking space) and 7001 lamps went red! |
| 23:35.28 | x86 | but if I call, say, 8500 (voicemail exten) from the 7001 phone, the BLF light on 7000 stays green |
| 23:35.53 | x86 | I had been testing it by using 8500 the whole time |
| 23:36.07 | x86 | so if you call voicemail, you're not "busy" ? |
| 23:36.12 | *** join/#asterisk monoexec (n=North@64-42-36-136.atgi.net) |
| 23:36.50 | x86 | anyway, it works fine now :) |
| 23:37.09 | *** part/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net) |
| 23:38.15 | x86 | [TK]D-Fender: now on my 942, how do i use one of the line keys as BLF+SD? |
| 23:38.16 | citywok | [TK]D-Fender: on my actual pbx i have keyset/programmable buttons on my phones, is there anythign similar that can be done in asterisk? |
| 23:38.23 | x86 | it looks like they can only be used for registrations? |
| 23:38.51 | [TK]D-Fender | x86: Might not be possible. I don't really use them so I can't cofirm the status of their firmware. |
| 23:38.59 | x86 | gotcha |
| 23:39.16 | [TK]D-Fender | citywok: Please clarify that. A lot. |
| 23:39.30 | x86 | [TK]D-Fender: is it normal for voicemail to not make presence show "InUse" ? |
| 23:39.34 | [TK]D-Fender | citwith you know... details... and "stuff" |
| 23:39.50 | [TK]D-Fender | x86: it should... |
| 23:39.58 | [TK]D-Fender | x86: when in doubt... |
| 23:40.04 | [TK]D-Fender | ~pb |
| 23:40.05 | jbot | [~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://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 23:40.08 | citywok | Say i want to provide a button for "Log into account code", and it prompts you for your account code, and saves it |
| 23:40.20 | citywok | or "Log into hunt group" |
| 23:40.36 | [TK]D-Fender | citClearly depends on the phone. That sounds like a boring speed-dial to me... |
| 23:40.57 | citywok | [TK]D-Fender: i agree, i have yet to find a softphone with speeddial that actually works |
| 23:41.28 | citywok | phonerlite has speeddial but its a piece of trash -- xlite doesnt appear to have it, and thats my favorite softphone so far |
| 23:41.28 | [TK]D-Fender | citywok: Oh now you want a SOFTPHONE with it? |
| 23:41.48 | [TK]D-Fender | citywok: X-Lite supports it in their contact directory IIRC |
| 23:41.54 | [TK]D-Fender | citywok: eyeBeam for sure |
| 23:42.36 | [TK]D-Fender | citywok: Or since you're talking "soft-phone", you're already on a PC why not make a web-panel for this... |
| 23:42.54 | *** join/#asterisk quentusrex (n=quentusr@c-24-19-34-21.hsd1.wa.comcast.net) |
| 23:42.54 | citywok | web panel & use it in the xlite browser? i thought about that |
| 23:43.28 | quentusrex | Does anyone know of a way to control asterisk for a multi client setup? clients don't need to be able to login, but I need to be able to control the DID's that are forwarded to them, etc. |
| 23:45.29 | [TK]D-Fender | quentusrex: Forwarded to them? Sounds like you need to control what is SENDING calls to them. |
| 23:45.37 | quentusrex | correct. |
| 23:45.46 | [TK]D-Fender | quentusrex: You don't control your phone company, your phone company controls YOU |
| 23:46.18 | [TK]D-Fender | quentusrex: You don't generally get to fire your BOSS. You seem to suffer from a heirarchy disorder... |
| 23:47.33 | quentusrex | I have a few different branch offices that have 'sub accounts' with our voip providor. The problem is that I need to put a server in between the offices and the voip providors now.. |
| 23:47.33 | [TK]D-Fender | quentusrex: Perhaps you should describe the setup you have in mind witha lot better detail. |
| 23:47.34 | quentusrex | I need a way to allow all calls between branches to never leave our systems, also to be able to have a single connection to our voip providors. |
| 23:48.16 | [TK]D-Fender | quentusrex: Well if they all go through a central * server you control, then its all just dialplan. |
| 23:48.50 | quentusrex | [TK]D-Fender: I'd like something graphical or webbased to be able to do the configs for the dialplan.. |
| 23:49.21 | [TK]D-Fender | quentusrex: Then go pick a GUI |
| 23:49.54 | [TK]D-Fender | quentusrex: but good luck with the "fine-grained control" idea |
| 23:51.28 | quentusrex | I was looking at A2Billing, but we don't need calling cards... I've read that this 'could' be used to handle what I need... |
| 23:52.14 | [TK]D-Fender | quentusrex: How often do changes really need to be made? |
| 23:52.24 | quentusrex | once a week if that... |
| 23:52.52 | quentusrex | But I've been tasked with switching some of the managers home phones over to voip too.... |
| 23:52.54 | quentusrex | :( |
| 23:53.16 | quentusrex | "since it's soo much cheaper, then let the executives have it as a perk" |
| 23:53.18 | quentusrex | :( |
| 23:53.46 | monoexec | Hello friends. I have a total of 10 Aastra 9112i's sip phones running off an Asterisk server. Everything works weel, exept for the message waiting indicater (well...the stuttered dial tone). The Aastra tech support people claim that everything is fine on there end and therefore must be the result of a misconfigured asterisk server. Thoughts? |
| 23:54.06 | *** join/#asterisk xuser (n=JJ@unaffiliated/xuser) |
| 23:55.30 | [TK]D-Fender | quentusrex: Hardly seems worth wasting the time on any GUI |
| 23:55.56 | [TK]D-Fender | monoexec: pastebin your sip.conf masking only passwords |
| 23:55.59 | [TK]D-Fender | ~pb |
| 23:55.59 | jbot | [~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://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 23:56.01 | [TK]D-Fender | ^^^^^^^^^^^ |
| 23:56.25 | jaytee | [TK]D-Fender, do you have a macro that does that? :-) |
| 23:56.51 | [TK]D-Fender | jaytee: nope. ~pb + enter, then hold down "^" for a few secs :) |
| 23:57.01 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 23:57.04 | monoexec | ok -- hold on just a sec |
| 23:57.13 | [TK]D-Fender | jaytee: If you were really smart you'd have compared the # of "^" I use and noticed its variable ;) |
| 23:57.23 | phix | hey |
| 23:57.24 | [TK]D-Fender | jaytee: Forensics 101 |
| 23:57.32 | phix | [TK]D-Fender! How's it going? |
| 23:57.48 | [TK]D-Fender | phix: Still breathing... |
| 23:58.06 | [TK]D-Fender | Tomorrow = all-day Lotus Domino up(?)grade |
| 23:58.16 | [TK]D-Fender | Yippy-kai-yay-&^*#%$&#^*%$*&#$ |
| 23:58.44 | [TK]D-Fender | However I'm inches from migrating from NetWare to Samba! |
| 23:58.45 | phix | [TK]D-Fender: That a computer application or a car? |
| 23:58.57 | [TK]D-Fender | phix: Mail Server |
| 23:59.12 | phix | [TK]D-Fender: lol, you're not using samba yet? what's stoping you? |
| 23:59.19 | [TK]D-Fender | phix: all-day Lotus Elise upgrade... I would look forward to :p |
| 23:59.25 | phix | [TK]D-Fender: ah :) hehe |
| 23:59.25 | jaytee | [TK]D-Fender, does your office use Groupwise or Notes? |
| 23:59.36 | [TK]D-Fender | jaytee: Notes. |
| 23:59.37 | phix | Yes Elise's need upgrades :P they suck stock |
| 23:59.46 | jaytee | I beta tested Notes 1.0 |
| 23:59.51 | [TK]D-Fender | jaytee: We're on NetWare 5.0 (1998) |