| 00:01.04 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 00:01.49 | *** join/#asterisk demlak (i=demlak@schwarz-pUnK.de) |
| 00:02.08 | demlak | hi.. anyone able to make a test call to me? SIP |
| 00:02.19 | demlak | will give number |
| 00:03.14 | demlak | im testing askozia PBX |
| 00:05.08 | Peaceful | grandpapadot, I'm provisioning through tftp |
| 00:09.31 | nvrpunk | I have two accounts on the same network with1 DID per each |
| 00:09.38 | *** join/#asterisk ph0ne (n=ph0ne@dsl-207-112-86-159.tor.primus.ca) |
| 00:09.41 | nvrpunk | and im having trouble getting the inbound calls to route |
| 00:09.43 | nvrpunk | to the phone |
| 00:09.53 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 00:09.56 | nvrpunk | anyone have time to look at my configs and give a pointer? |
| 00:10.15 | Qwell | 0x8F73FE42 |
| 00:12.16 | nvrpunk | Qwell, funny |
| 00:12.34 | nvrpunk | the sad part is I actually got the joke |
| 00:12.39 | ph0ne | ~ask |
| 00:12.40 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 00:13.03 | *** part/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 00:13.03 | ph0ne | I thought that was apt |
| 00:13.14 | nvrpunk | no, apt is a package manager |
| 00:13.19 | nvrpunk | you are wrong |
| 00:13.22 | *** part/#asterisk demlak (i=demlak@schwarz-pUnK.de) |
| 00:13.22 | Qwell | ~apt |
| 00:13.23 | jbot | well, apt is the really annoying bot. |
| 00:13.44 | Qwell | jbot: no, apt is not as cool as jbot |
| 00:13.45 | jbot | okay, Qwell |
| 00:13.51 | *** join/#asterisk fskrotzki (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 00:14.45 | ph0ne | why don't you paste bin your config files nvrpunk ? |
| 00:14.50 | grandpapadot | Peaceful: Login to the web console and make sure what your sending matches the config it pulled, i.e., local changes overwrite pulled settings |
| 00:15.07 | nvrpunk | http://www.pastebin.ca/911777 |
| 00:15.15 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 00:15.16 | ph0ne | thanks |
| 00:15.21 | nvrpunk | http://www.pastebin.ca/911782 |
| 00:15.29 | nvrpunk | ones my extensions.conf |
| 00:15.31 | nvrpunk | and my iax |
| 00:16.50 | nvrpunk | the explanation is this, 2 junciont netoworks accounts, both with their own DIDs |
| 00:16.56 | nvrpunk | im trying to get both routing in |
| 00:17.11 | nvrpunk | but im confused as to what context I should put the incoming extension under |
| 00:19.06 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 00:20.18 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 00:21.30 | SteveTotaro | Wayhigh: around? |
| 00:23.44 | Peaceful | grandpapadot, yup config sure matches up |
| 00:24.07 | zobia | anyone knows what can cause RTCP Read too short? |
| 00:24.14 | Peaceful | grandpapadot, I'm gonna have to continue this tommorrow -- thanks for the comments |
| 00:25.29 | *** join/#asterisk DJF6 (n=DJF5@84-105-201-37.cable.quicknet.nl) |
| 00:35.17 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.175.142) |
| 00:52.04 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 00:55.35 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-92ab253e4b73b301) |
| 01:00.48 | *** join/#asterisk isamar (n=isamar@200.254.219.30) |
| 01:02.18 | isamar | need a quick reference of "IF" inside extensions.conf |
| 01:02.44 | isamar | can anybody just copy and paste please? |
| 01:03.08 | isamar | no graphic internet here....... only text terminal :-) |
| 01:04.18 | *** join/#asterisk adjohn (n=adjohn@p4053-ipad401marunouchi.tokyo.ocn.ne.jp) |
| 01:04.52 | drmessano | Sorry, I dont have text |
| 01:04.55 | drmessano | I only have jpg |
| 01:05.09 | JunK-Y | Set(foo=${IF($[ ${x} = 7]?tval:fval)}) |
| 01:06.30 | isamar | hi adjhon... |
| 01:06.37 | isamar | in Japan... |
| 01:06.52 | isamar | I miss Tokyo... believe or not |
| 01:07.54 | Nugget | nihon wa iki desu ne |
| 01:08.13 | drmessano | teo torriate |
| 01:10.10 | grandpapadot | Is there a way to specify a codec prior to placing a call without modifying the peer definition? Somehow through the dial-plan? |
| 01:14.36 | _ShrikE | grandpapadot: See ${SIP_CODEC} in channelvariables.txt |
| 01:14.43 | grandpapadot | Cool! Great, thanks. |
| 01:15.00 | _ShrikE | grandpapadot: lemme know how it works.. i've never used it. |
| 01:16.32 | *** join/#asterisk mjackson (n=happy@69.85.202.188) |
| 01:17.21 | cmantito | quick question, the callerid line in sip.conf/iax.conf, does that override the client specified CID details, or is that used if there are no client specified CID details? |
| 01:19.33 | grandpapadot | _ShrikE: Is seems it still has to be in the allow list ... |
| 01:20.35 | drmessano | What exactly are you trying to do? |
| 01:21.02 | grandpapadot | I want my peer definition for this particular itsp to be g729 but I want to override it in some cases with ulaw. Just a bandwidth conservation technique. |
| 01:21.15 | grandpapadot | .. if it was possible |
| 01:21.31 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:25.52 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:26.38 | mjackson | I'm trying to, through either the manager api or AGI, take an existing call and conference in another party. Is the best way to do this to Redirect the existing channel to a meetme room and then dial the 3rd party and drop them into the meetme room as well? |
| 01:28.43 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) |
| 01:29.25 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 01:33.59 | *** join/#asterisk anthm (n=anthm@70-9-12-51.area4.spcsdns.net) |
| 01:33.59 | *** mode/#asterisk [+o anthm] by ChanServ |
| 01:34.40 | *** join/#asterisk Tigerplug (n=chatzill@89.100.141.160) |
| 01:34.51 | mjackson | When a SIP device is on a call, how can I find out what channel the other party on the call is on? |
| 01:34.52 | Tigerplug | How long are you guys working with Asterisk? |
| 01:39.50 | *** join/#asterisk simonr (n=simonr@125.38.15.204.static.thewire.ca) |
| 01:40.25 | drmessano | 26 hours a day |
| 01:40.38 | *** join/#asterisk nvrpunk (n=root@81.90.21.227) |
| 01:40.59 | nvrpunk | how do I make the voicemail prompt so the user can play back their message? |
| 01:41.07 | nvrpunk | like not HangUp() |
| 01:44.14 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-22-105.lns10.syd7.internode.on.net) |
| 01:44.15 | *** join/#asterisk jameswf-home (n=james@ip72-204-221-181.ph.ph.cox.net) |
| 01:45.50 | *** join/#asterisk PepOSX (n=angeldav@190.72.131.172) |
| 01:48.18 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.ITB.ac.id) |
| 01:59.40 | jameswf-home | tzafrir Debian :http://www.no-clutter.com/gallery/displayimage.php?album=56&pos=40 lmao |
| 02:02.39 | *** join/#asterisk SteveTotaro (n=Elizabet@pool-71-166-102-100.bltmmd.east.verizon.net) |
| 02:03.05 | SteveTotaro | one of my groups http://blog.washingtonpost.com/securityfix/2008/02/research_may_spell_end_of_mobi.html |
| 02:03.16 | drmessano | HAHAH |
| 02:03.45 | SteveTotaro | got my USRP and four TB all ready |
| 02:08.00 | tzanger | yeah I read about that |
| 02:09.15 | *** join/#asterisk delphus (n=delphus@201-43-192-25.dsl.telesp.net.br) |
| 02:09.58 | delphus | sorry if this is quite trivial but, there is anyway to store register=> in realtime database ? |
| 02:10.00 | SteveTotaro | i have been following that for over a year |
| 02:10.31 | SteveTotaro | i can get frames from the BTS no problem just cannot crack the A5 |
| 02:10.52 | SteveTotaro | as soon as they release this, i am going to have my own encrypted BTS |
| 02:11.04 | SteveTotaro | connected to asterisk and a T1 |
| 02:11.11 | jameswf-home | hmm watch american idol or shaq debut.... basket ball sucks |
| 02:16.12 | jameswf-home | like presidential elections http://www.no-clutter.com/gallery/displayimage.php?album=56&pos=98 |
| 02:17.01 | *** join/#asterisk profounded (n=bruiz@nat01-quad3-ext.Rutgers.EDU) |
| 02:17.31 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 02:18.41 | profounded | hey everyone, i just got my asterisk pbx all setup with freepbx (voicepulse is my carrier) and everything is great, except for call forwarding.. it forwards the calls but the line is silent on both sides when answered.. is there any obvious reason for this? |
| 02:21.01 | jameswf-home | ~freepbx |
| 02:21.01 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 02:21.48 | profounded | ill try that, ty |
| 02:22.07 | *** join/#asterisk simonr (n=simonr@125.38.15.204.static.thewire.ca) |
| 02:29.43 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 02:31.18 | *** join/#asterisk Tuari (n=Tuari@cpe-76-183-79-199.tx.res.rr.com) |
| 02:42.18 | mike-ekim | <PROTECTED> |
| 02:42.19 | mike-ekim | exit |
| 02:42.21 | mike-ekim | \quit |
| 02:54.47 | JunK-Y | wow, the moon is orange now. |
| 02:57.08 | jameswf-home | Do you pine for the days when men were men and wrote their own device drivers? |
| 03:00.26 | J4k3 | the moon is on the other side of a lot of clouds here :( |
| 03:04.17 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 03:06.51 | *** join/#asterisk webar7 (n=webart@CPE0080c8f208a5-CM001371173cf8.cpe.net.cable.rogers.com) |
| 03:07.43 | webar7 | my asterisk is trunked to an ITSP using IAX .... inside our firewall all the devices connect to the * box using SIP |
| 03:07.55 | delphus | sorry if this is quite trivial but, is there anyway to store register=> in realtime database ? |
| 03:08.32 | webar7 | if someone calls our DID and enters and extension thnings ring through to the SIP phone on the LAN |
| 03:08.45 | webar7 | but sound is only one way |
| 03:10.20 | styelz | webar7 do you have externip and localnet set ? |
| 03:11.33 | Tigerplug | is there an open source solution to start a VOIP provider business? |
| 03:11.45 | webar7 | styelz, hmm yes but they look wrong |
| 03:12.00 | Tigerplug | something that allows you to manage large numbers of users easily |
| 03:12.09 | webar7 | styelz,thanks I will try that and see if it fixes anything |
| 03:14.00 | jameswf-home | Tigerplug: my guess if your asking here and didnt find it on google you cant manage anything of the sort |
| 03:14.27 | jameswf-home | ~fish |
| 03:14.27 | jbot | i guess fish is FISHFISHFISH! DO THE FISH DANCE! "Give a man a fish and you'll feed him a day. Teach him how to fish and he'll feed himself for the rest of his life." This is so appropriate, instead of asking us to tell you exactly what to do, why not read some docs, then come back and ask specific questions which aren't covered?, or ... |
| 03:15.02 | jameswf-home | ~fire |
| 03:15.03 | jbot | Bender : Light a fire for a man and he's warm for a night. Light a man on fire and he's warm for the rest of his life... |
| 03:15.05 | Tigerplug | jameswf-home - thanks for the input. Very valuable, someday I may know as much as you. Until then I guess I can only hope and look up to you. |
| 03:16.07 | jameswf-home | ~heh |
| 03:16.08 | jbot | heh |
| 03:18.06 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 03:18.06 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 03:18.58 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 03:19.41 | lmadsen | curious... how many lines are your dialplans typically... and what is the purpose of the dialplan at that line length? |
| 03:20.42 | _ShrikE | Replacing MySQL() with func_odbc has made my dialplans much shorter :) |
| 03:23.54 | Tigerplug | guys I have been reading through the Asterisk book and it does explain many things. However, my setup will be completely VOIP and I am just not having any look with it |
| 03:24.01 | _ShrikE | Typically about 300 priorities in a pbx type build for us |
| 03:24.51 | Tigerplug | I have looked at sample config files and read asterixguru.com etc but still havint difficulty. I am a noob on the subject and I have put in the time to read material (maybe not enough), but I'm still having difficulty with it. Anyone suggest some more learning resources? |
| 03:26.39 | plik | Tigerplug: go slow and actually follow the examples in the book, disregarding the stuff for external lines |
| 03:27.56 | Tigerplug | plik -> thats where I have trouble applying it. I had a simple dial plan setup and I was able to make internal calls etc and then when I started following the examples in the book I couldn't get anything working. I think I'll start again, build the system and install from there |
| 03:28.43 | lmadsen | _ShrikE: interesting. My dialplans have gotten a lot more powerful because of func_odbc... but I wasn't using MySQL before. However I know what you mean because I had a client that took about 7 or 8 lines of MySQL in a macro and converted it to 1 line with func_odbc. |
| 03:29.03 | delphus | _ShrikE: how did you put register=> lines into odbc database ? |
| 03:29.14 | _ShrikE | delphus: I didnt |
| 03:29.18 | delphus | damn |
| 03:29.35 | lmadsen | _ShrikE: this dialplan I've been working on for 2 months and is coming to fruition tonight is a redundant system using DUNDi and func_odbc, and it's 880 priorities |
| 03:29.38 | plik | Tigerplug: yeah, try that... change a little at a time, test & play... comment as you go |
| 03:29.49 | lmadsen | oh, and lots of realtime |
| 03:30.05 | _ShrikE | sounds impressive |
| 03:30.06 | lmadsen | well... for sip extensions, queues, and queue_members |
| 03:30.06 | Tigerplug | yup I guess. Any learning resources that you can recommend (from experience)? |
| 03:31.00 | plik | www.voip-info.org was definitely useful, but theres probably as much out-dated or wrong oinfo there as good stuff... |
| 03:31.08 | plik | so good luck with that |
| 03:31.29 | _ShrikE | Tigerplug: have you read the book? |
| 03:32.17 | _ShrikE | err. backreading I see you already have :) |
| 03:32.30 | delphus | lmadsen: is extension realtime supporting t and s priorities now ? |
| 03:33.05 | plik | the book + voip-info + google + common sense + lurking here + willingness to try things = success |
| 03:33.33 | delphus | lmadsen: I mean i and s |
| 03:34.50 | styelz | ~book |
| 03:34.51 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 03:38.36 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 03:38.53 | *** join/#asterisk Kumba_ (n=kumba@62-209.187-72.tampabay.res.rr.com) |
| 03:39.13 | Kumba_ | The volgain in voicemail.conf, does it apply to voicemail in general or just the e-mail? |
| 03:40.07 | jameswf-home | ~wikis |
| 03:40.08 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 03:41.09 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
| 03:41.14 | lmadsen | delphus: I don't put the dialplan in the database -- the dialplan stays as a flatfile and is version controlled with svn, and I use func_odbc to control the dynamic data fromt eh database |
| 03:41.36 | delphus | lmadsen: oh I got it. |
| 03:41.53 | lmadsen | I can't imagine how you would ever program anything by having the dialplan in the database.... :) |
| 03:42.04 | *** join/#asterisk Kumba_ (n=kumba@62-209.187-72.tampabay.res.rr.com) |
| 03:42.18 | lmadsen | chapter 12 -- if you want to get started on how I personally do clustering... read chapter 12 of tfot2 :) |
| 03:42.38 | jameswf-home | I keep my dial plan on a 5.25 floppy in a safe under the bed |
| 03:42.43 | lmadsen | that chapter is the starting block of all the high level dialplan language I utilize and what I learned over the last 2 years |
| 03:42.56 | lmadsen | jameswf-home: I have an 8" on my shelf that I keep mine on |
| 03:43.09 | Wayhigh | sup jameswf |
| 03:43.12 | jameswf-home | thats what she said |
| 03:43.18 | styelz | mines on punch cards burried in a chest on an island |
| 03:44.01 | lmadsen | jameswf-home: http://www.mp3lyrics.org/k/king-missile/detachable-penis/ |
| 03:44.15 | Kumba_ | Sounds like a good christian song |
| 03:44.25 | jameswf-home | jbot: shesaid is <reply>I have an 8" on my shelf that I keep mine on |
| 03:44.26 | jbot | okay, jameswf-home |
| 03:45.26 | *** join/#asterisk wolvenar (n=wolv@71-217-183-21.farg.qwest.net) |
| 03:46.25 | tzanger | heh I've heard that song |
| 03:51.28 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-183-109.hsd1.or.comcast.net) |
| 03:51.51 | *** join/#asterisk Robba (n=rob@203.56.181.15) |
| 03:53.05 | drmessano | heh |
| 03:53.06 | Robba | does anyone know if there are any problems using "xfersound =" |
| 03:53.10 | drmessano | innovations |
| 03:53.18 | drmessano | I have this great idea |
| 03:53.29 | drmessano | A telephone system that runs on linux |
| 03:53.55 | drmessano | I'm going to call it "pound" after my fav button on the keypad |
| 03:54.50 | plik | it'll never work... europeans call tht the hash, ansd some weirdos even say octithorpe |
| 03:55.08 | Robba | lol |
| 03:55.23 | lmadsen | octothorpe forever! |
| 03:56.01 | drmessano | ha |
| 03:56.31 | plik | "when you have finished recording your message, you may hang up or press the octothorpe for more options " |
| 03:57.14 | *** join/#asterisk suvir (n=chatzill@ppp-124-120-140-195.revip2.asianet.co.th) |
| 03:57.29 | Robba | and has anyone had any issues using someone elses IVR on an asterisk box? |
| 03:57.35 | Nivex | Press the octothorpe until it megahertz |
| 03:57.49 | jameswf-home | ~pound |
| 03:57.50 | jbot | ACTION pounds head on desk |
| 03:58.00 | suvir | has anyone ever added a new language to say.c and app_voicemail.c? |
| 03:58.03 | jameswf-home | ~octothorpe |
| 03:58.03 | jbot | i heard octothorpe is ASCII character 35: #; AKA hash, Pound, <shift> 3, gliph, number see http://en.wikipedia.org/wiki/Octothorpe |
| 03:58.13 | jameswf-home | wow jbot is smart |
| 03:58.17 | drmessano | HashPBX |
| 03:58.25 | jameswf-home | octothorpebx |
| 03:58.31 | drmessano | HashPBX "It really smokes" |
| 03:58.35 | plik | "for marijuana, press the hash key" |
| 03:58.47 | J4k3 | ~hashpbx |
| 03:58.58 | drmessano | ~happyclownpbx |
| 03:58.59 | jbot | [HappyClownPBX] is currently in closed beta, approaching 12GB in size, uses Asterisk for it's core, and it pwns |
| 03:59.18 | J4k3 | jbot: hashpbx is <reply> It really smokes! |
| 03:59.19 | jbot | J4k3: okay |
| 03:59.20 | J4k3 | ~hashpbx |
| 03:59.21 | jbot | It really smokes! |
| 03:59.38 | drmessano | ~happyclownphone |
| 03:59.41 | drmessano | Crap |
| 04:00.23 | J4k3 | jbot: blenderpbx is <reply> will it blend? |
| 04:00.25 | jbot | J4k3: okay |
| 04:00.31 | J4k3 | ~blenderpbx |
| 04:00.32 | jbot | will it blend? |
| 04:00.43 | J4k3 | yes, yes it will. |
| 04:02.06 | drmessano | ~happyclownphone |
| 04:02.06 | jbot | [HappyClownPhone] is a happy softphone using the latest GSM codec and uses closed APIs to communicate with HappyClownPBX. It also pwns |
| 04:02.08 | Robba | Where is [TK]? |
| 04:04.49 | J4k3 | jbot: happyclownphone is also gay |
| 04:04.49 | jbot | okay, J4k3 |
| 04:05.15 | J4k3 | haha |
| 04:05.36 | Robba | well drmessano i have decided i will also create a phone system that runs on linux although i will call mine DND |
| 04:05.50 | drmessano | Hmmm |
| 04:05.59 | Robba | for the same reason as you |
| 04:06.14 | jameswf-home | You should call it HTH (hide The Human) |
| 04:06.26 | J4k3 | WCT |
| 04:06.31 | J4k3 | Waste Client's Time |
| 04:06.53 | jameswf-home | maybe WhatNowPBX |
| 04:06.56 | J4k3 | then theres the pressure cooker sales version, IAE |
| 04:06.59 | drmessano | TMITSLMHUOY |
| 04:07.02 | J4k3 | I Annoy Everyone |
| 04:07.05 | Robba | i was thinkin as a bit of a joke i was going to use microsoft narrator to create the files for my IVR lol |
| 04:07.07 | jameswf-home | or AserixkWhatNow |
| 04:07.21 | drmessano | Tell Me If This Sounds Like Me Hanging Up On You |
| 04:07.31 | J4k3 | wtfbox |
| 04:07.38 | jameswf-home | fanboy pbx |
| 04:07.43 | drmessano | Thats Trixbox |
| 04:07.52 | jameswf-home | could be a fork |
| 04:07.52 | J4k3 | ricebox |
| 04:07.58 | J4k3 | the pbx with a spoiler. |
| 04:08.00 | drmessano | Wrought with fanboyism |
| 04:08.10 | Robba | i used to like trixbox |
| 04:08.19 | Robba | but not so much anymore |
| 04:08.26 | J4k3 | fork asterisk, call it pound |
| 04:08.34 | jameswf-home | maybe KornFlakesBox or LuckyCharmsBox |
| 04:08.51 | J4k3 | FRuitLoopOS |
| 04:09.06 | drmessano | CaptainCrunchBox |
| 04:09.24 | drmessano | CookieCrispBox Pro |
| 04:09.26 | jameswf-home | they use to have phones with hex digits.... guess base16 phone numbers didnt make it |
| 04:09.55 | drmessano | Most 2 way radio keypads are 1 - 9, *, #, and A thru D |
| 04:10.00 | drmessano | Err |
| 04:10.02 | drmessano | 0-9 |
| 04:10.25 | jameswf-home | binary phone numbers... please enter your 64bit binary phonenumber |
| 04:10.44 | drmessano | Theres no reason they couldnt implement the extra 4 DTMF tones |
| 04:10.47 | jameswf-home | doh i transposed the 32nd 0 |
| 04:12.30 | drmessano | Dude, call me @ a0987d45fa67b42 |
| 04:12.47 | Qwell | no way, 128 bit hex. ipv6 |
| 04:12.53 | sbingner | jameswf-home, are you serious (hex) |
| 04:13.11 | Qwell | SIP/[2001:470:1f07:77:215:f2ff:fe43:2b13] |
| 04:13.18 | drmessano | Good god |
| 04:13.20 | jameswf-home | no hex for me, I am saving my self for marige |
| 04:13.52 | sbingner | SIP/[2002:ce7e:3ba:1:0:0:0:3] <-- that's me! |
| 04:14.13 | Qwell | sbingner: 1::0:0:3 |
| 04:14.15 | Qwell | ftw |
| 04:14.19 | drmessano | SIP/danny@127.0.0.1 <--- Call me |
| 04:14.32 | sbingner | Qwell, so 1::3? ... I'm missing something |
| 04:14.35 | Qwell | ipv6 short for is awesome |
| 04:14.39 | Qwell | sbingner: umm, yeah |
| 04:14.41 | Qwell | that too |
| 04:14.47 | Qwell | short form* |
| 04:15.08 | drmessano | There's no place like 127.0.0.1 |
| 04:15.09 | sbingner | oh yea I added the 0's to make it longer |
| 04:15.10 | sbingner | lol |
| 04:15.21 | *** join/#asterisk Sniffadog (n=cameron@CPE-121-223-233-73.static.vic.bigpond.net.au) |
| 04:15.26 | Qwell | Address unreachable. :( |
| 04:15.35 | sbingner | you can't hit 2002:ce7e:3ba:1::3 ? |
| 04:15.46 | jameswf-home | id hit that |
| 04:15.52 | Qwell | probably my side |
| 04:16.37 | jameswf-home | my netgear doesnt do ipv6 |
| 04:16.58 | Qwell | my (modified..) Digium AA50 does. |
| 04:16.59 | drmessano | My ISP barely does ipv4 |
| 04:17.07 | *** join/#asterisk webar7 (n=webart@CPE0080c8f208a5-CM001371173cf8.cpe.net.cable.rogers.com) |
| 04:18.03 | *** part/#asterisk Sniffadog (n=cameron@CPE-121-223-233-73.static.vic.bigpond.net.au) |
| 04:18.46 | sbingner | Qwell, I can hit 2001:470:0:64::2 (ipv6.he.net) so yea |
| 04:18.58 | Qwell | yeah,I can't hit anything |
| 04:19.19 | sbingner | Qwell, I stopped using tunnel brokers, and moved to 6to4 -- it's faster and more reliable ;) |
| 04:19.26 | Qwell | eh? |
| 04:19.40 | jameswf-home | Qwell: so cheap he wont buy a $5nic he robs the hw graveyard |
| 04:19.57 | Qwell | hw graveyard? |
| 04:20.03 | sbingner | hardware graveyard |
| 04:20.11 | drmessano | None of my NICs will do ipv6 |
| 04:20.14 | drmessano | :( |
| 04:21.31 | brookshire | i thought ipv6 was just a software thing |
| 04:21.50 | Qwell | eh? |
| 04:21.56 | brookshire | any nic that can do ipv6 can do ipv6 |
| 04:21.58 | Qwell | it's like ipv4, but +2 |
| 04:22.02 | Qwell | yeah, it's protocol level |
| 04:22.06 | brookshire | i mean |
| 04:22.07 | JT | drmessano: :o |
| 04:22.11 | brookshire | ipv4 can do ipv6 |
| 04:22.11 | brookshire | blah |
| 04:22.21 | JT | NICs don't care about IP |
| 04:22.23 | drmessano | Non of my NICs will :( |
| 04:22.26 | JT | they only care about frames |
| 04:22.26 | Qwell | no special hardware needed. just a decent OS |
| 04:22.35 | JT | drmessano: are you serious? |
| 04:22.40 | drmessano | Oh yes |
| 04:22.58 | drmessano | I go to Start > Settings > Control Panel.. NO IPV6 |
| 04:22.58 | brookshire | what are they? 10baseT ? |
| 04:23.03 | drmessano | :( |
| 04:23.04 | brookshire | oh.. windows |
| 04:23.13 | Qwell | drmessano: update to windows 3.11 |
| 04:23.19 | drmessano | Ohhh |
| 04:23.28 | drmessano | Do you think that will help? |
| 04:24.02 | drmessano | I'm running Windows XP.. I guess that's like 2.0 or so? |
| 04:24.15 | drmessano | Ok, I will go buy the update |
| 04:25.35 | brookshire | *speechless* |
| 04:25.57 | drmessano | If I get this IPV6, will my ICQ work faster? |
| 04:26.11 | brookshire | probably not |
| 04:26.13 | Qwell | drmessano: it's more digits, so it'll be slower. it's like adding xml. |
| 04:26.21 | drmessano | Oh crap |
| 04:26.22 | sbingner | drmessano, absolutely! |
| 04:26.58 | brookshire | but everyone in the world can have like 10 public IPs |
| 04:26.58 | sbingner | drmessano, xp = "ipv6 install" and you'll have ipv6 |
| 04:26.58 | Qwell | 10...billion |
| 04:26.58 | drmessano | OHHH |
| 04:27.03 | drmessano | I <3 IPV6 |
| 04:27.18 | Qwell | there are a ridiculous number of addresses with ipv6 |
| 04:27.24 | brookshire | so what happened to IPV5? |
| 04:27.32 | Qwell | we don't talk about ipv5 |
| 04:27.35 | drmessano | So if I have public IP, you can see my hard drive files, yes no yes? |
| 04:27.52 | sbingner | drmessano, yes no yes |
| 04:27.57 | brookshire | Qwell: i think i'm going to wait on IPV-LIVEXP |
| 04:28.00 | drmessano | :S |
| 04:28.05 | Qwell | 3.402823669e+38 |
| 04:28.06 | *** join/#asterisk adjohn (n=adjohn@219.106.248.145) |
| 04:28.08 | Qwell | addresses |
| 04:28.11 | *** join/#asterisk AdamWest (n=Leif@d221-75-88.commercial.cgocable.net) |
| 04:28.11 | *** join/#asterisk bobnormal (n=bob@221.213.47.10) |
| 04:28.14 | brookshire | the microsoft version will be better |
| 04:28.29 | sbingner | lol |
| 04:28.32 | bobnormal | does anyone know how to integrate zoiper with firefox so when users click a sip:// url it initiates a call in zoiper? |
| 04:28.49 | drmessano | My friend told me Microsoft invented linux but they didn't sell it because it was too much like DOS |
| 04:28.59 | drmessano | I guess DOS was better |
| 04:29.01 | drmessano | :( |
| 04:29.16 | sbingner | drmessano, who let the cat out of the bag? |
| 04:29.36 | drmessano | Have you ever noticed how DOS and Linux are the same color?????!!!! |
| 04:29.40 | *** part/#asterisk profounded (n=bruiz@nat01-quad3-ext.Rutgers.EDU) |
| 04:29.50 | sbingner | drmessano, the white in dos seems whiter |
| 04:29.56 | *** join/#asterisk webar7 (n=webart@CPE0080c8f208a5-CM001371173cf8.cpe.net.cable.rogers.com) |
| 04:30.03 | drmessano | Yes, it much faster too |
| 04:30.17 | drmessano | and commands have good names like "format" |
| 04:30.24 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-55-104.lns10.syd7.internode.on.net) |
| 04:30.29 | drmessano | in linux you use like flagglepoop c: to format |
| 04:30.33 | drmessano | Why so complicated? |
| 04:30.52 | sbingner | lol |
| 04:31.00 | Nivex | fraggle poop? |
| 04:31.10 | drmessano | No, that partition |
| 04:31.14 | sbingner | drmessano, no it's "mformat c:" |
| 04:31.39 | drmessano | oic |
| 04:32.08 | drmessano | I install Asterisk to C:\Asterisk |
| 04:32.45 | drmessano | But it no run |
| 04:32.48 | drmessano | Dumb XP |
| 04:33.21 | sweeper | drmessano: u need to edit the registr |
| 04:33.45 | drmessano | Eww no, I delete registry.. too many words |
| 04:35.30 | *** join/#asterisk AdamWest (n=Leif@d221-75-88.commercial.cgocable.net) |
| 04:36.58 | drmessano | So I install Trixbox and my friend on skype can't call my drmessano@asterisk1.local address, WHY???!!!??!?!? |
| 04:38.31 | jameswf-home | tell em to try sippyskype |
| 04:38.56 | sbingner | drmessano, they need to add asterisk1.local to /etc/hosts with your IP |
| 04:39.15 | angryuser | skype is not so bad, first mass good quality soft |
| 04:39.17 | jameswf-home | drmessano: did you reboot |
| 04:39.23 | drmessano | Add? Is that like calculator stuff? |
| 04:39.27 | angryuser | even if it is not open source |
| 04:39.37 | drmessano | Skype is the devil |
| 04:39.37 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:39.41 | jameswf-home | open sores?? ewwww |
| 04:39.48 | sbingner | drmessano, sorry I meant C:\Windows\System32\Drivers\etc\hosts |
| 04:39.51 | angryuser | source* |
| 04:40.45 | sbingner | skype is retarded imo |
| 04:41.00 | angryuser | i mean, add a button on site and being called by skype on you sip device is not so bad idea |
| 04:41.39 | Nugget | the skype client is pretty slick, I have to admit. I wouldn't mind if there was a SIP softphone that was even half as nice as skype. |
| 04:41.50 | angryuser | sbingner so we have 7.1 mil people out there, retarded |
| 04:42.09 | Nugget | and their clever tcp and rfc1918 discovery tricks do a great job piercing NAT where SIP and IAX will just give you fits. |
| 04:42.11 | angryuser | online |
| 04:42.35 | sbingner | angryuser, many more than 7.1 mil people who are retarded |
| 04:42.43 | sbingner | angryuser, I expect about 2/3 of the population is |
| 04:43.08 | angryuser | sbingner , dont see any argument |
| 04:44.20 | *** join/#asterisk pigpen2 (n=pigpen@fw.seamans.cc) |
| 04:44.40 | eric2 | I installed the lic for g729, edited the sip.conf file and set my phone to use g729 too... how do I know its working? |
| 04:44.41 | angryuser | sbingner , the main idea is, adapting to many networks brings more users, and skype can be a big advantage |
| 04:45.08 | eric2 | I even restarted asterisk |
| 04:45.09 | pigpen2 | hi all, I am needing to prove to a customer that their employee's are streaming radio and such, that is causing high latency and borking their IAX trunk. |
| 04:45.37 | pigpen2 | Can anyone suggest any package that can monitor/log/graph protocol bandwidth usage? |
| 04:45.46 | drmessano | Wireshark |
| 04:45.59 | pigpen2 | Ah, the "new" etherreal |
| 04:46.07 | pigpen2 | k, didn't know it could do this... |
| 04:46.12 | pigpen2 | I'll check it out. |
| 04:46.37 | drmessano | Wireshark makes my bed every morning |
| 04:47.17 | pigpen2 | Heh, my wife needs it for house chores....lord knows she won't do it. |
| 04:47.18 | pigpen2 | :) |
| 04:47.29 | drmessano | I order my wife to do hers |
| 04:47.35 | drmessano | Gotta keep her in line |
| 04:47.50 | drmessano | I had her toes cut off so she could stand closer to the sink |
| 04:48.21 | sbingner | lol |
| 04:48.29 | pigpen2 | Yeah, we talk big when they arn't looking. |
| 04:48.47 | drmessano | Hell yes |
| 04:48.51 | jameswf-home | you know why brides where white.... |
| 04:48.54 | jameswf-home | so the dishwasher matches the stove and fridge |
| 04:49.13 | jameswf-home | *wear |
| 04:49.18 | pigpen2 | hahaha |
| 04:49.42 | drmessano | Why do brides smile on the way down the aisle? |
| 04:49.58 | drmessano | Because they know they'll never have to have sex again |
| 04:49.59 | drmessano | O.O |
| 04:50.06 | pigpen2 | hmm..maybe I should be Mormon. Fridge is white, stove is black, sink is beige. |
| 04:50.13 | jameswf-home | you know what the first thing a woman should do when she leaves the battered womens shelter.... |
| 04:50.18 | jameswf-home | the dishes if she knows whats good for her |
| 04:50.24 | drmessano | HAH |
| 04:50.50 | sbingner | lol |
| 04:51.05 | angryuser | <eric2> force codec to g729 |
| 04:51.30 | drmessano | I get into work today and my boss walks up to me.. "I guess everyone now knows you're leaving" |
| 04:51.33 | drmessano | "Oh how?" |
| 04:51.37 | sbingner | eric2, sip show channels will list active codec |
| 04:51.45 | drmessano | "You left a copy of your resignation on the copy machine" |
| 04:51.48 | drmessano | OOPS |
| 04:51.58 | sbingner | LOL nice |
| 04:52.16 | sbingner | is that true? |
| 04:52.20 | drmessano | When I made a copy, it made 12.. I thought I had grabbed them all |
| 04:52.21 | angryuser | you found a new job ? |
| 04:52.27 | drmessano | Hell yes |
| 04:52.34 | Qwell | did your boss already know? heh |
| 04:52.39 | drmessano | Yeah |
| 04:52.58 | drmessano | I came in yesterday and made a copy.. someone left "12" copies on the machine and didnt clear the # out |
| 04:53.05 | Qwell | that might've been awkward |
| 04:53.06 | drmessano | So I hit start and it started running |
| 04:53.20 | drmessano | I grabbed all but one, I guess, and shredded them |
| 04:53.34 | angryuser | whatever it's not like you quit you wife ;) |
| 04:53.39 | drmessano | heh |
| 04:53.43 | angryuser | your* |
| 04:54.05 | drmessano | There's been like 24 hours of high drama |
| 04:54.33 | jameswf-home | drmessano: you need a road trip to SC |
| 04:54.42 | drmessano | yeah I do |
| 04:54.43 | drmessano | brb |
| 04:59.28 | drmessano | back |
| 05:00.21 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 05:01.26 | variable_office | has anyone here dealt with gr303 much? I am confused is it packet based or not? |
| 05:01.28 | drmessano | Looks like the missile hit the spy satellite |
| 05:01.36 | drmessano | ..and france surrendered |
| 05:02.11 | sbingner | lol |
| 05:02.31 | drmessano | Glad someone got it |
| 05:03.40 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 05:04.08 | wolvenar | anyone have experience setting up a bit more advanced stuff with broadvoice as a provider? |
| 05:05.09 | wolvenar | trying to use a single account at bv with multiple numbers and route by incoming # dialed |
| 05:09.36 | jameswf-home | no video of them shooting the ufo i mean satelite |
| 05:10.51 | *** join/#asterisk mmurdock (n=nope@c-24-10-190-87.hsd1.co.comcast.net) |
| 05:13.44 | *** join/#asterisk tengulre11 (n=tengulre@125.71.208.16) |
| 05:13.53 | *** part/#asterisk delphus (n=delphus@201-43-192-25.dsl.telesp.net.br) |
| 05:14.58 | angryuser | <jameswf-home> if you have a camera on 240 km orbit, call me next time we will film it |
| 05:15.03 | angryuser | <PROTECTED> |
| 05:15.29 | angryuser | so gn everybody |
| 05:15.48 | drmessano | heh |
| 05:15.55 | angryuser | &have fun |
| 05:16.31 | jameswf-home | will a webcam on a hobby rocket work |
| 05:17.21 | drmessano | That sounds like a bad cinematography method for porn flicks |
| 05:21.06 | *** join/#asterisk Maxous (n=Maxous@76.97.3.24) |
| 05:24.20 | *** join/#asterisk ahbritto (n=guest@adsl-69-104-3-183.dsl.pltn13.pacbell.net) |
| 05:28.05 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 05:28.05 | *** mode/#asterisk [+o russellb] by ChanServ |
| 05:30.50 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 05:37.25 | *** join/#asterisk UnixDog (n=unixdog@ppp-71-128-4-150.dsl.irvnca.pacbell.net) |
| 05:37.30 | UnixDog | asterisk.boldlygoingnowhere.org |
| 05:37.39 | UnixDog | lol |
| 05:37.43 | UnixDog | had to have it |
| 05:42.25 | *** join/#asterisk terracon (n=greisky@CPE0050da822b70-CM0012254076d6.cpe.net.cable.rogers.com) |
| 05:42.37 | russellb | UnixDog: what? |
| 05:42.59 | UnixDog | dundns has a new domain |
| 05:43.00 | *** join/#asterisk bkw_ (n=brian@70.91.87.57) |
| 05:43.20 | UnixDog | so we are playing with hames |
| 05:43.36 | UnixDog | voip.boldlygoingnowhere.org |
| 05:43.39 | UnixDog | lol |
| 05:55.09 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 05:55.51 | joelsolanki | Hi room |
| 05:56.10 | joelsolanki | want to use ss7 with sangoma 104D. what is the best tested solution available ? |
| 05:56.19 | joelsolanki | chan_ss7 or sangoma SMG ? |
| 06:04.13 | joelsolanki | anybody here ? |
| 06:04.20 | joelsolanki | any experienced guys plz ? |
| 06:06.31 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 06:06.53 | Mavvie | Is there a PHP class which connects to the asterisk manager? |
| 06:07.33 | Mavvie | got one http://lists.digium.com/pipermail/asterisk-users/2004-November/064162.html |
| 06:08.37 | *** join/#asterisk adjohn (n=adjohn@219.106.248.145) |
| 06:10.04 | nvrpunk | is there a way to make g729 sound better for voicemail? |
| 06:18.08 | sweeper | nvrpunk: mm, you might be storing it as something not-g729 |
| 06:18.13 | sweeper | transcoding ftl |
| 06:18.17 | nvrpunk | i got it |
| 06:18.20 | nvrpunk | now :) |
| 06:23.25 | jameswf-home | Mavie you can simply open a socket |
| 06:26.03 | jameswf-home | http://pastebin.ca/912247 < php socket |
| 06:27.38 | jameswf-home | such things are better served as a function then a class... |
| 06:28.06 | Kumba_ | I'd like to submit a feature request: wget compatible links on asterisk.org for downloading :) |
| 06:28.50 | jameswf-home | Kumba they use to be but thats hard to track.. simply type the path it works |
| 06:30.31 | jameswf-home | wget http://downloads.digium.com/project/releases/project-version.tar.gz |
| 06:32.11 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) [NETSPLIT VICTIM] |
| 06:32.14 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) [NETSPLIT VICTIM] |
| 06:32.14 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.ITB.ac.id) [NETSPLIT VICTIM] |
| 06:32.14 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) [NETSPLIT VICTIM] |
| 06:32.14 | *** join/#asterisk zobia (n=laurashr@222.212.75.120) [NETSPLIT VICTIM] |
| 06:32.14 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) [NETSPLIT VICTIM] |
| 06:32.14 | *** join/#asterisk mamacito62 (n=XD@201.208.84.21) [NETSPLIT VICTIM] |
| 06:32.14 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) [NETSPLIT VICTIM] |
| 06:32.14 | *** join/#asterisk BrianR___ (i=brianr@static-72-70-36-11.bstnma.fios.verizon.net) [NETSPLIT VICTIM] |
| 06:32.26 | jameswf-home | kubrick is kinda a crap node |
| 06:33.16 | *** join/#asterisk egecko (n=sam@cpe-76-176-238-225.san.res.rr.com) |
| 06:33.55 | egecko | after editing an extensions.conf file .. is there anything that needs to be reloaded/restarted? I assume that .conf file is read in real-time |
| 06:34.19 | *** join/#asterisk Azam (n=azamzia@58-65-160-140.nayatel.pk) |
| 06:34.56 | Azam | hi mysql_query give me segmentation fault can anyone help me please |
| 06:40.05 | pkunkra | azam, out of curiousity, what made you decide to come into #asterisk |
| 06:40.08 | pkunkra | ? |
| 06:41.06 | Azam | bcuz i am trying to connect to mysql DB in my asterisk application |
| 06:41.17 | pkunkra | oh. it is asterisk related. hah |
| 06:41.36 | pkunkra | no idea. sorry. |
| 06:41.36 | Azam | i thought may be the addon i am using to connect has some bugs |
| 06:41.41 | Corydon76-dig | func_odbc already takes care of the API for you... |
| 06:41.44 | Azam | ok thanks anyways |
| 06:41.54 | drmessano | Why is it so hard to meet hot asterisk loving single women on ICQ? |
| 06:42.13 | pkunkra | drmessano, because there aren't any really. :-) |
| 06:42.16 | Azam | thanks Corydon76-dig |
| 06:42.38 | Corydon76-dig | Go look at configs/func_odbc.conf.sample |
| 06:43.36 | Azam | ok |
| 06:43.37 | pkunkra | drmessano, show me an asterisk girl, and i'll show you a girl with more nerds chasing after her than she knows what to do with. |
| 06:43.49 | drmessano | Amen, bro |
| 06:45.00 | pkunkra | <PROTECTED> |
| 06:45.11 | wolvenar | hmm , |
| 06:45.25 | Corydon76-dig | pkunkra: so you're saying it's prime feeding grounds for gay men? |
| 06:45.34 | pkunkra | hahaha |
| 06:46.06 | pkunkra | now that one i don't know about. i guess the channel with have to answer that. |
| 06:46.40 | Corydon76-dig | Actually, there aren't enough gay men in here... |
| 06:46.48 | Corydon76-dig | Not local (to me) anyway |
| 06:47.00 | pkunkra | too bad. |
| 06:47.28 | pkunkra | but #asterisk probably isn't the best place to go looking, however..... |
| 06:54.22 | drmessano | Damn |
| 06:54.36 | drmessano | Someone should have told me Asterisk won't run on a Kaypro |
| 07:01.35 | *** join/#asterisk Tuari (n=Tuari@cpe-76-183-79-199.tx.res.rr.com) |
| 07:04.13 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:09.42 | nvrpunk | kaypro? |
| 07:10.29 | drmessano | Yeah |
| 07:10.39 | clickonce | If I want to make the caller enter their e.g. social security number and store it in a variable, should I do this using something else than alot of exten => lines? |
| 07:11.37 | Corydon76-dig | "core show application Read" |
| 07:13.51 | clickonce | Thanks! :) |
| 07:15.05 | *** join/#asterisk af_ (n=getsmart@88-149-230-148.dynamic.ngi.it) |
| 07:17.01 | pkunkra | drmessano, kaypro? its a little early for april's fools. |
| 07:17.36 | pkunkra | you can try it on my old 8088 if you like. |
| 07:17.59 | nvrpunk | so question, is it possible to notify certain voip phones that they have voicemail? |
| 07:18.04 | *** join/#asterisk puzzled (n=patrick@53533DDB.cable.casema.nl) |
| 07:18.15 | pkunkra | nvrpuck. yes. i don't know how though. |
| 07:18.20 | pkunkra | i e-mail the messages |
| 07:18.31 | clickonce | Jesus, the Kaypro II computer looks alot like a huge oscillator :P |
| 07:18.35 | pkunkra | 1.6 has some improvments in it though |
| 07:19.03 | pkunkra | clickonce, well. that's because there were probably made around the same time and by the same company at first. |
| 07:19.21 | pkunkra | or |
| 07:19.37 | pkunkra | kaypro based its designs on the oscillator itself. |
| 07:19.54 | pkunkra | since that was all they had back then. |
| 07:19.59 | clickonce | Coule be. |
| 07:21.16 | pkunkra | mac classic was pretty revolutionary back then. |
| 07:21.22 | pkunkra | so it was either pick the oscillator p.c. or the new fangled mac classic. |
| 07:21.54 | drmessano | or a commodore 64 |
| 07:21.58 | pkunkra | speaking of which, i ran into an apple retailer that built its front desk out of about 50-100 mac classics in nyc. |
| 07:22.08 | pkunkra | very tasteful construction work. |
| 07:22.11 | clickonce | Oh, nice =) |
| 07:22.38 | nvrpunk | that makes me wonder id wordstar still exists |
| 07:22.41 | nvrpunk | somewhere |
| 07:22.44 | nvrpunk | if* |
| 07:22.46 | pkunkra | probably. |
| 07:22.54 | pkunkra | try abandonware maybe? |
| 07:23.26 | pkunkra | i used to get good games that were abandonware |
| 07:27.33 | pkunkra | one of my favorites was released as open source eventually. |
| 07:27.43 | pkunkra | folks started developing and fixing it. |
| 07:27.52 | pkunkra | its available on fedora now. |
| 07:28.02 | pkunkra | try looking up "uqm" |
| 07:28.34 | pkunkra | used to play that a lot when i was a teenager. |
| 07:28.57 | drmessano | Thats easy to find, clickonce |
| 07:29.04 | puzzled | clickonce: google around. I've come across it several times |
| 07:29.07 | drmessano | Thats the ringtone on my phone |
| 07:29.21 | clickonce | I found some on YouTube but the quality sucks. :) |
| 07:30.19 | pkunkra | http://www.youtube.com/watch?v=DH7EqDIPfpA |
| 07:30.21 | pkunkra | there you go |
| 07:31.46 | *** join/#asterisk Dayver (n=user@ip65-44-153-126.z153-44-65.customer.algx.net) |
| 07:31.46 | puzzled | clickonce: http://blog.tmcnet.com/blog/tom-keating/mobile-phones/download-24-ringtone.asp |
| 07:32.10 | Dayver | does anyone know how to assign DID to Zap channel? |
| 07:32.22 | Dayver | zapata.conf |
| 07:33.10 | clickonce | puzzled: Thanks! :D |
| 07:33.18 | puzzled | have fun |
| 07:34.30 | *** part/#asterisk UnixDog (n=unixdog@ppp-71-128-4-150.dsl.irvnca.pacbell.net) |
| 07:34.41 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 07:36.46 | Anthony76 | hi, I which transfert call to other asterisk and in my log I have: |
| 07:36.47 | Anthony76 | Feb 21 08:34:10 NOTICE[16522]: chan_sip.c:6932 get_refer_info: Supervised transfer requested, but unable to find callid '3c274d2d668a-41wafpzcsx98@snom360-000413236351'. Both legs must reside on Asterisk box to transfer at this time. |
| 07:37.02 | Anthony76 | How can I fixe ? I use asterisk 1.2.13 |
| 07:37.08 | Anthony76 | thx |
| 07:40.46 | Anthony76 | I use two ipbx and I want transfert an external call to internal number where my phone is connected on the second ipbx |
| 07:43.42 | patrick-- | Hey, my asterisk keeps crashing, when i put someone on hold for too long. what could cause this? (BN4S0 HFC Card) |
| 07:48.56 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 07:49.46 | clickonce | What's it called when all phones ring at the same time when a call is coming in? |
| 07:53.01 | Datax | chaos ? :p |
| 07:53.37 | patrick-- | Is it possible to have the recorded Voicemail messages in mp3 format? |
| 07:53.44 | patrick-- | automatically? |
| 07:56.06 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
| 07:57.15 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 07:58.07 | clickonce | Datax: Nah, I just want more than one phone to ring when a call is coming in. Just like the POTS systems. |
| 08:05.43 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 08:06.03 | clickonce | I've tried Dial(SIP/ext1&SIP/ext2) but it doesn't work. |
| 08:06.13 | clickonce | (Which I took from a sample config) |
| 08:09.40 | *** join/#asterisk Alexandre_fr (n=alex@nat04-bzr.dys1.com) |
| 08:16.21 | *** join/#asterisk oej (n=olle@194.171.177.169) |
| 08:16.25 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 08:17.36 | defswork | clickonce: that looks correct to me |
| 08:18.01 | defswork | clickonce: show application dial - seems to confirm it's correct |
| 08:19.44 | *** join/#asterisk steliosk (n=Stelios@athedsl-290446.home.otenet.gr) |
| 08:19.47 | clickonce | Hmm, this is strange... seems like Asterisk doesn't want to connect to my VoIP provider either. Strange since it's the same sip.conf I used yesterday (without mods) |
| 08:19.55 | clickonce | I don't get any errors anywhere either. |
| 08:20.25 | *** join/#asterisk SomethingISOdd (n=TestMast@S010600a0d1757bfb.cg.shawcable.net) |
| 08:20.28 | *** join/#asterisk egecko (n=sam@cpe-76-176-238-225.san.res.rr.com) |
| 08:20.40 | SomethingISOdd | hello all quick question i hope is there anyway to break into a conversation ? |
| 08:24.49 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-be8bb91660045ef5) |
| 08:26.25 | clickonce | God damnit, my sip.conf is correct but "sip show registry" doesn't show anything. |
| 08:26.53 | sweeper | clickonce: sip show peers? |
| 08:26.59 | styelz | pastebin it someone might look at it |
| 08:27.44 | clickonce | I have a line which resembles my VoIP provider there. (Unmonitored) |
| 08:30.01 | clickonce | http://rafb.net/p/Lbs5da63.html |
| 08:30.04 | clickonce | Worked yesterday |
| 08:30.37 | *** join/#asterisk ^shark_ (n=^shark_@217.194.147.193) |
| 08:30.43 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 08:30.46 | ^shark_ | ^book |
| 08:31.17 | sweeper | ~book |
| 08:31.18 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 08:31.47 | sweeper | clickonce: try putting the registry line in the general section, instead of in the [ext3] section ;) |
| 08:35.10 | *** join/#asterisk oej (n=olle@194.171.177.169) |
| 08:36.06 | clickonce | sweeper: ah, damn |
| 08:36.19 | ^shark_ | sweeper: |
| 08:36.33 | clickonce | sweeper: How friggin' simple. :) |
| 08:37.01 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 08:42.26 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:43.27 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:49.19 | sweeper | yea, asterisk config files are....fun :P |
| 08:49.54 | sweeper | I like freeswitch's notion of xml, now if only fs's configs were actually VALID xml, I might be swayed :P |
| 08:55.48 | *** join/#asterisk g0mb0 (n=root@external.micom.mng.net) |
| 08:59.01 | *** join/#asterisk bronson (n=bronson@adsl-76-199-198-19.dsl.pltn13.sbcglobal.net) |
| 09:00.23 | *** join/#asterisk Sniper_linux1234 (n=michofr@mail.splendor.net) |
| 09:00.57 | Sniper_linux1234 | Hi all, I need to ask in which file we an configure the asterisk to send the local ip instead of localhost(127.0.0.1) ip? |
| 09:06.56 | *** join/#asterisk Maxfactor (n=Maxfacto@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 09:08.33 | clickonce | I need someone to record message for me that I can use in Asterisk :) |
| 09:09.01 | Maxfactor | hello...first time here.... |
| 09:09.02 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:09.19 | Maxfactor | need help real bad |
| 09:09.45 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:10.25 | Maxfactor | been reading alot about asterisk and is interesting.. |
| 09:11.47 | nixguy | Maxfactor: dont ask if you can ask just ask |
| 09:12.09 | Maxfactor | hi nixguy |
| 09:12.21 | nixguy | hi |
| 09:12.26 | Maxfactor | gonna ask you a few questions |
| 09:12.36 | Maxfactor | but don't laugh at me... |
| 09:12.37 | nixguy | im no expert on asterisk |
| 09:12.49 | nixguy | i wont laugh but i might say read up more ... |
| 09:13.05 | Maxfactor | where do i get the SPA 3102? |
| 09:13.15 | nixguy | what is that? |
| 09:13.16 | Maxfactor | and a linux distro |
| 09:13.17 | nixguy | a codec? |
| 09:13.43 | nixguy | you can find a linuxdistro downlodable on the internet |
| 09:13.45 | nixguy | www.debian.org |
| 09:13.54 | nixguy | is the distro im using for my asterisk project |
| 09:14.01 | Maxfactor | is it good? |
| 09:14.13 | Maxfactor | what version? |
| 09:14.19 | nixguy | latest |
| 09:14.24 | nixguy | 4.0 |
| 09:14.28 | Maxfactor | ok |
| 09:14.44 | Maxfactor | so I download it off the internet.. |
| 09:14.46 | nixguy | etch |
| 09:14.51 | nixguy | yes and burn the cd's |
| 09:15.02 | nixguy | how much IT experience have you got? |
| 09:15.03 | Maxfactor | is it zipped? |
| 09:15.14 | nixguy | its an iso image |
| 09:15.18 | Maxfactor | ok |
| 09:15.28 | nixguy | i have a hunch you are going to have a hard time if you've never run linux before |
| 09:15.30 | Maxfactor | just been reading alot... |
| 09:15.40 | Maxfactor | correct |
| 09:15.43 | Maxfactor | mandrake |
| 09:15.47 | *** join/#asterisk XnOSX (i=d491af1a@gateway/web/ajax/mibbit.com/x-2725c1d6737d389e) |
| 09:15.50 | Maxfactor | before.. |
| 09:15.53 | nixguy | ok |
| 09:16.03 | nixguy | debian is a lot less automated |
| 09:16.04 | Maxfactor | and JAMD |
| 09:16.13 | nixguy | less gui and eye candy |
| 09:16.46 | Maxfactor | the distro zipped? |
| 09:17.00 | Maxfactor | then need to unzip.. |
| 09:17.10 | nixguy | i told you |
| 09:17.11 | Maxfactor | and then burn into cd.. |
| 09:17.13 | nixguy | it's an iso image |
| 09:17.22 | nixguy | if you dont know how to burn an iso image |
| 09:17.25 | nixguy | the google for it |
| 09:17.40 | Maxfactor | now... |
| 09:18.00 | Maxfactor | I have a lattop |
| 09:18.06 | Maxfactor | laptop |
| 09:18.15 | Maxfactor | with windows on it |
| 09:18.40 | nixguy | so far your questions have been related to linux not to asterisk |
| 09:18.44 | Maxfactor | don't need windows for now |
| 09:18.45 | nixguy | i recomend you to try #debian |
| 09:18.48 | nixguy | or something like that |
| 09:18.55 | Maxfactor | ok |
| 09:19.10 | Maxfactor | the spa 3000 |
| 09:20.18 | Maxfactor | thanks |
| 09:21.08 | nixguy | np |
| 09:22.13 | Maxfactor | nixguy |
| 09:22.19 | nixguy | yes? |
| 09:22.59 | Maxfactor | besides the debian and asterisk ...do I need some type of card installed on my laptop? |
| 09:23.47 | nixguy | Maxfactor: to do what? |
| 09:24.17 | Maxfactor | to get my system going... |
| 09:24.23 | Maxfactor | pbx |
| 09:24.40 | nixguy | to run a pbx you dont need any card |
| 09:24.51 | nixguy | if you want to be able to call to the pstn |
| 09:24.57 | nixguy | you need some kind of connection to it |
| 09:24.59 | Maxfactor | yes |
| 09:25.24 | nixguy | either a card of some kind or a sip trunk to some operator to bridge you out to the pstn |
| 09:25.31 | nixguy | this is all described in the first pages of the asterisk manual |
| 09:25.34 | nixguy | !asterisk |
| 09:25.38 | nixguy | hmm was it something like that |
| 09:25.40 | nixguy | !book |
| 09:25.43 | nixguy | :) |
| 09:25.44 | nixguy | cant remember |
| 09:25.53 | nixguy | just google for the manual |
| 09:26.21 | Maxfactor | I am reading the future of telephony... |
| 09:26.21 | nixguy | come back when you've read the asterisk book by o'rileys |
| 09:26.32 | Maxfactor | exact |
| 09:26.35 | Maxfactor | doing now |
| 09:27.27 | Maxfactor | how do I get started alltogether? |
| 09:27.39 | nixguy | Maxfactor: in your case not to sound evil or anything |
| 09:27.43 | nixguy | you need to read more |
| 09:27.59 | nixguy | obviouesly you are lacking some fundamentals as when it comes to understanding asterisk and linux |
| 09:28.20 | nixguy | you need a base of knowledge to ask your questions here not to piss people of (at least me) |
| 09:28.44 | Maxfactor | sorry man |
| 09:28.50 | nixguy | im not angry |
| 09:29.18 | nixguy | i'm no asterisk expert but running a asterisk on your laptop if you want it to connect to the pstn desent sound plausible |
| 09:29.34 | nixguy | i belive cards you buy to connect to the pstn are PCI cards |
| 09:29.38 | nixguy | your laptop doesent have pci cards |
| 09:29.50 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:31.02 | Maxfactor | thought I need a different one |
| 09:32.04 | *** join/#asterisk af_ (n=getsmart@88-149-240-203.dynamic.ngi.it) |
| 09:32.33 | nixguy | Maxfactor: read |
| 09:32.34 | nixguy | read |
| 09:32.39 | nixguy | no go my son! |
| 09:33.08 | nixguy | if you arent familiar with the linuxterminal i belive i won't see you untill you've mastered it |
| 09:34.00 | sweeper | ~book |
| 09:34.01 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 09:34.05 | sweeper | failsauce :P |
| 09:34.58 | *** join/#asterisk jarod14 (n=jarod14@ns2.viatelecom.com) |
| 09:35.04 | Maxfactor | nixguy thanks |
| 09:35.10 | nixguy | np |
| 09:36.12 | defswork | nixguy: I ran asterisk on a laptop at home - connected via a spa3102 |
| 09:36.28 | defswork | nixguy: the laptop had a broken screen but otherwise was fine |
| 09:36.30 | defswork | until the HD died :( |
| 09:36.36 | sweeper | : |
| 09:36.37 | sweeper | ( |
| 09:40.35 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-89-86.lns10.syd6.internode.on.net) |
| 09:49.09 | nixguy | defswork: whats spa3102 |
| 09:50.39 | *** join/#asterisk badcfe (i=christia@nothing.beats.deathporn.com) |
| 09:50.55 | badcfe | is it possible to reload the manager.conf without restarting * ? |
| 09:51.11 | nixguy | badcfe: thereis reload manager |
| 09:51.12 | nixguy | command |
| 09:51.16 | nixguy | maybe thats what you are after |
| 09:52.13 | badcfe | nixguy: shouldve become manager reload no? oh i see its considered core, so the command doesnt start off by the functionality name |
| 09:52.19 | badcfe | nixguy: thanks! |
| 09:52.36 | nixguy | np |
| 09:53.32 | badcfe | i got 1.4.17 and theres no "manager reload" in CLI, i can do "reload manager" only. huh, it tells me Please use 'module reload' instead. |
| 09:58.16 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 09:58.32 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-3d6dc5b9de3347b3) |
| 09:59.16 | *** join/#asterisk xezz (n=phob@athedsl-238842.home.otenet.gr) |
| 10:00.10 | xezz | hello, i have an isdn PRA (30 channels), i am planning to user asterisk+sangoma a101d, is the card suitable for the PRA line ? |
| 10:00.38 | *** join/#asterisk tsubasafr (n=sylvain@ABordeaux-156-1-32-205.w86-213.abo.wanadoo.fr) |
| 10:00.41 | tsubasafr | hi ! |
| 10:01.19 | Chris-NB | hi |
| 10:01.26 | Chris-NB | anyone using a cisco 7970 phone? |
| 10:01.51 | Chris-NB | or anyone knows if it is possible to get the BlindXfer Softkey on 7970? 7960 had this softkey |
| 10:02.57 | *** join/#asterisk oej (n=olle@194.171.177.169) |
| 10:04.17 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 10:04.21 | joelsolanki | Hi room |
| 10:04.41 | joelsolanki | anybody using asterisk + sangoma ss7 ? |
| 10:04.41 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 10:04.45 | tsubasafr | sorry for my question, but you provide the asterisk daemon for Asterisk Recording Interface |
| 10:05.05 | tsubasafr | because it seem the daemon not running unable to connect to localhost:5038 (Connection refused) |
| 10:05.14 | joelsolanki | I m in process to choose either sangoma ss7 or chan_ss7 |
| 10:05.23 | joelsolanki | need some feedback / talk on this. |
| 10:05.25 | joelsolanki | anyone plz ? |
| 10:06.57 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 10:07.25 | tsubasafr | Who provides ARI ? ( asterisk recording interface ) |
| 10:08.05 | SomethingISOdd | hello all quick question i hope is there anyway to break into a conversation ? |
| 10:08.13 | Sniper_linux1234 | Hey all |
| 10:08.19 | sweeper | SomethingISOdd: there is |
| 10:08.28 | sweeper | it's called "barging" |
| 10:08.49 | SomethingISOdd | ok let me look that up thanks sweeper |
| 10:08.53 | Sniper_linux1234 | I have freepbx installed on my machine..the issue is in the INVITE message the freepbx is sending the localhost IP(127.0.0.1) instead of its local IP |
| 10:09.07 | Sniper_linux1234 | do you have any Idea about houw to fix it? |
| 10:12.11 | SomethingISOdd | sweeper what version of asterisk did that come out in? |
| 10:12.23 | sweeper | SomethingISOdd: pretty old |
| 10:12.27 | SomethingISOdd | i tried core show function baring and barging and both say its invaild command |
| 10:12.38 | SomethingISOdd | i`m using Asterisk 1.4.6 |
| 10:14.23 | sweeper | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ChanSpy |
| 10:14.38 | sweeper | http://www.voip-info.org/wiki/view/Asterisk+cmd+ZapBarge |
| 10:14.40 | SomethingISOdd | sweeper chanspy only listens doesnt it |
| 10:16.21 | SomethingISOdd | sweeper perfect thank you |
| 10:16.47 | sweeper | SomethingISOdd: chanspy can also talk |
| 10:17.00 | sweeper | the w option allows that |
| 10:17.26 | SomethingISOdd | sweeper what i am wondering if i can play a recording instead of talking.. |
| 10:17.45 | sweeper | fo sho |
| 10:17.54 | SomethingISOdd | ? |
| 10:18.18 | sweeper | ARG SWARMING ANTS ARE BITING ME |
| 10:18.26 | *** join/#asterisk ToTo (n=ToTo@209.8.41.152) |
| 10:18.31 | sweeper | chanspy hooks one channel up to another |
| 10:18.44 | patrick-- | does anyone have a sample dialplan with callerID on it for me to have a look at? i think ive built up my dialplan a bit wrong |
| 10:18.59 | sweeper | actually, there might be an easier way |
| 10:19.06 | SomethingISOdd | sweeper ok? |
| 10:20.24 | SomethingISOdd | sweeper i wonder if i could setup a dsp, to play the recording on the execution of the chanspy command |
| 10:20.41 | *** join/#asterisk af_ (n=getsmart@88-149-240-203.dynamic.ngi.it) |
| 10:21.47 | sweeper | SomethingISOdd: you could, but that's nasty |
| 10:21.53 | SomethingISOdd | ya |
| 10:21.57 | SomethingISOdd | <PROTECTED> |
| 10:22.59 | sweeper | looks like your best bet is to use chanspy to hook up a dummy channel that uses Playback() |
| 10:23.19 | SomethingISOdd | ok let me go a head up how to create a dummy channel :-) |
| 10:24.14 | SomethingISOdd | thanks sweeper |
| 10:24.20 | sweeper | n/p |
| 10:28.12 | badcfe | on * 1.4.2 when i do ${STRFTIME(|GMT+1|%Y%m%d-%H%M%S)} i get the right local time. on 1.4.17 i dont. on the 1.4.17 i have to do just ${STRFTIME(||%Y%m%d-%H%M%S)}, wich gives me correct local time. |
| 10:28.23 | *** join/#asterisk vale-ICS (n=vale@icsnet.demon.co.uk) |
| 10:28.41 | badcfe | is this a difference between * versions or rather my time setop on the gnu/linux. |
| 10:29.07 | sweeper | I blame asterisk! |
| 10:29.22 | badcfe | ofcourse doing date command on the two shows the same thing, that ive verified.. |
| 10:29.28 | badcfe | sweeper: u do? |
| 10:29.36 | badcfe | sweeper: or is it ironic? |
| 10:29.54 | sweeper | well, assuming you're doing these tests on the same box, yea |
| 10:30.16 | sweeper | to keep things working across several versions, you could always run a shell command that returns the properly formatted time |
| 10:30.17 | badcfe | sweeper: its on different boxes. but theyr both debian with same ntp config and showing same date. |
| 10:30.19 | clickonce | Guys, when someone call me or I call them, I hear them perferctly while they barely hear me. I haven't added any encoding/quality stuff to extensions.conf nor sip.conf. What do you recommend that I add to get the best possible quality? (Not exceeding 256kbit/s) |
| 10:30.29 | sweeper | badcfe: then yea, asterisk! |
| 10:31.09 | sweeper | clickonce: are they hearing choppy audio or staticy audio, or echo? |
| 10:31.15 | badcfe | sweeper: thing is that all the gmt localtime unixtime timezones conversion locatlisation and so on on the lurky nix boxes there may be some differences in the setup, and thats why asterisk behaves differently on the two. theyr both debian etch tho |
| 10:31.30 | clickonce | sweeper: I kinda disappear from time to time. (Mostly 100% of the time) |
| 10:31.33 | sweeper | choppy = bandwidth, static = codecs, echo = nasty analog things |
| 10:31.59 | sweeper | badcfe: so use UMT! |
| 10:32.58 | sweeper | clickonce: sounds like chop to me. trying killing all internet traffic except for your voip |
| 10:33.16 | sweeper | I'm assuming you have decent bw available |
| 10:33.20 | clickonce | It's as dead as it can be and it's a 100Mbit/s connection. |
| 10:33.31 | clickonce | And according to the ISP it should be 100/100 |
| 10:33.37 | sweeper | hmm |
| 10:33.53 | sweeper | that's where your asterisk box is, or where your sip client is, or both? |
| 10:34.12 | clickonce | I suspected it could be the belkin wlan AP, but, when I call my softphone client on my WinBox (not wlan) it works fine. |
| 10:34.17 | clickonce | both |
| 10:34.53 | clickonce | (I know belkin isn't state-of-the-art but I borrowed it from my GF, but, as said, it works find internally) |
| 10:35.07 | badcfe | how many simultaneous calls can a all-intel quad1333 with enough mem take doin only forwarding of alaw-alaw ? |
| 10:35.42 | badcfe | 100? then youll get jitter introduced from the box? |
| 10:35.44 | sweeper | badcfe: test and find out. usually single-box limit is around 300 calls, but there's lots and lots of variables |
| 10:35.47 | clickonce | sweeper: It sounds like 24kbit/s music :) |
| 10:36.25 | sweeper | oh, so they hear SOMETHING the whole time, just not always something intelligible? |
| 10:36.45 | sweeper | I am going to be eaten alive by ant proto-queens |
| 10:36.47 | clickonce | Yes, always something, but it's like low-bit crap and the volume goes up and down. |
| 10:37.20 | sweeper | well, try disallow=all; allow=ulaw |
| 10:37.36 | sweeper | in sip.conf's general section |
| 10:37.52 | sweeper | and make sure you don't have any contradicting directives elsewhere |
| 10:38.39 | clickonce | Was that for me or badcfe? |
| 10:39.23 | sweeper | for you |
| 10:39.44 | *** join/#asterisk exvito (n=exvito@195.245.132.93) |
| 10:40.38 | clickonce | That sounds better internally, let me make an external call. |
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| 10:43.26 | clickonce | Can I increase the quality even more? By changing allow=ulaw... |
| 10:43.53 | sweeper | well, ulaw is equivalent to pstn calls |
| 10:43.59 | clickonce | ah |
| 10:43.59 | sweeper | and is pretty standard |
| 10:44.12 | clickonce | Okay, then I'll let it be. :) |
| 10:44.26 | sweeper | so unles you're calling, for example, someone with HD audio codecs on their client, you're not gonna get much better |
| 10:44.31 | clickonce | Perhaps it gets better when I get my new Asterisk box along with a new 10/100 switch and a SPA-962 |
| 10:44.49 | clickonce | Instead of a going through a Belkin WLAN AP :) |
| 10:44.51 | sweeper | perhaps |
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| 10:52.31 | nebojsajsimic | hi all |
| 10:52.53 | sweeper | hihi |
| 10:58.29 | nebojsajsimic | can someoane help with phpagi i make call with dial and it works fine but i can't set MOH |
| 10:58.35 | nebojsajsimic | is there any idea |
| 10:58.54 | nebojsajsimic | $zovi = $agi->exec_dial(SIP,$broj,25,m); |
| 10:59.12 | nebojsajsimic | works ok but when i try |
| 10:59.25 | nebojsajsimic | <PROTECTED> |
| 11:00.04 | nebojsajsimic | it doesnot and from ext 1,1,Dial (sip......,m(xxx)); works too |
| 11:01.01 | nebojsajsimic | i think that is broblem to send "(xxx)" as param to exec_dial any idea |
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| 11:08.24 | nebojsajsimic | <PROTECTED> |
| 11:08.25 | nebojsajsimic | <PROTECTED> |
| 11:08.25 | nebojsajsimic | <PROTECTED> |
| 11:08.25 | nebojsajsimic | <PROTECTED> |
| 11:08.25 | nebojsajsimic | <PROTECTED> |
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| 11:08.57 | nebojsajsimic | and then play defoult moh to answer |
| 11:08.59 | nebojsajsimic | ... |
| 11:09.37 | nebojsajsimic | ani idea why my moh class start then dial then stop and change to defoult |
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| 11:14.47 | nebojsajsimic | and it work from extensions |
| 11:14.48 | clickonce | To enable more than one incoming call and more than one outgoing call on the same POTS number I have from my SIP provider, do I need some support at the provider site as well? |
| 11:15.55 | clickonce | Currently I have Asterisk programmed so that I can call in on my POTS number, press 5, enter a number, press # and asteriskd dials that number out, but, I get a busy tone. I suspect my SIP provider doesn't have support for more than one call at the same time over the same SIP trunk or is it my asterisk that is misconfigured? |
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| 11:19.09 | nixguy | from what version of asterisk is iax2 supported? |
| 11:21.53 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
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| 11:30.32 | sweeper | clickonce: probably the provider |
| 11:32.00 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 11:32.33 | clickonce | sweeper: Better call them then. |
| 11:34.19 | sweeper | yep |
| 11:34.31 | sweeper | some providers charge more for each concurrent call |
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| 11:42.46 | marl | hi can anyone help me? i have a * box setup with a dedicated disa number, basicly i dial into the disa number, * uses my CID to authenticate my call, and then allows me to dial outwards, problem i have is if i setup a basic phone (sony ericson) with a number like: 01418761234p01415641232 * dials the second number after the pause without any problems, but when i dial the same number from my xda2 OR if i only dial the disa number and try to manually enter the |
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| 12:06.46 | srd2 | how do I setup an extension to say, if callerid matches 123 or 456 it'll go to something else? |
| 12:08.25 | mosty | Use GotoIf and the CALLERID function |
| 12:08.47 | marl | exten => extensionnumber,n,GotoIf($["${CALLERID(num)}" = "CIDOFMOBILE"]?disa-matt,s,1) |
| 12:09.02 | marl | works great for my disa auth via CID |
| 12:09.16 | srd2 | disa? |
| 12:09.59 | marl | allows me to dial in from my mobile, and then place an outbound call as if i was sittig in the office |
| 12:11.14 | srd2 | ah, same here, except I use mine to let me call long distance/int |
| 12:11.38 | marl | yup, can use it for that as well |
| 12:12.14 | marl | anyone know if autofallthrough can be set within an extension context without affecting the rest of the dial plan, or is it a global only option? |
| 12:13.32 | srd2 | I used to have it so, I could spoof my callerid through one of my outgoing iax and have fun with people at work, making it look like they were calling themselves lol |
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| 12:13.58 | marl | lol |
| 12:14.30 | mosty | srd2, just use the n priority |
| 12:14.57 | mosty | you should not need to rely on auto fall through, if you do then it's a sign of a bad dialplan design |
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| 12:16.25 | marl | mosty, were u talking to me there about autofallthrough? |
| 12:17.27 | mosty | yes |
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| 12:18.16 | marl | if i ensure that all my extension definitions end with exten number => n,HangUp() that should be ok shouldnt it? |
| 12:18.25 | marl | (all my extensions end with this at the mo) |
| 12:19.17 | mosty | yes, assuming you don't have any crazy Goto's |
| 12:20.38 | marl | i dont always set i and s prioritys |
| 12:20.39 | mosty | you probably also want something to happen for i, t and T extensions |
| 12:21.25 | marl | or t and T prioritys |
| 12:22.03 | srd2 | got an interesting one, even tho I have language=en, it doesn't always play files from the en sub-directory in sounds, even when the file in that dir exists |
| 12:22.05 | mosty | depends on the context, if you need those obviously |
| 12:22.25 | marl | ok thanks :) ill see if this solves the problem :) |
| 12:22.41 | srd2 | so, say I go into voicemail, I'll get a mixture of two different voices like (a)you have (b) # (a) messages |
| 12:25.47 | srd2 | played around with it to, like changing it from en to uk, both in language= and the sub-directory name, still does it tho |
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| 12:28.10 | mosty | only in voicemail? |
| 12:30.00 | srd2 | em |
| 12:30.17 | srd2 | nope not just voicemail |
| 12:32.31 | marl | mosty, ive set autofallthrough to no, but i still dont get a chance to dial a number when i dial in to my disa number :( i have digit and response being set to 10 but it falls through as soon as i hit the first digit :( any ideas? |
| 12:32.56 | mosty | i know nothing about disa, sorry |
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| 12:34.46 | marl | its not specific to disa, its something to do with the timeouts etc :( |
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| 12:48.37 | defswork | srd2: got some files missing ? |
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| 12:49.05 | srd2 | nope |
| 12:51.28 | zeeesh | using rhel4, when service iptables is stop, asterisk is working fine .. when its stoped ... call connecting but unable to hear voice .. how hear .. IVR,,, voices... ??????? |
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| 12:59.11 | clickonce | Hmm, which one do you recommend, ext..conf or ext..ael? |
| 12:59.43 | SAL123 | I get an "481 Call Leg/Transaction Does Not Exist" from my provider, when trying to call-transfer as reply for REFER packegt. He uses Asterisk. Can somebody help me, what to suggest him to fix this problem? |
| 13:10.18 | lirakis | morning all |
| 13:12.08 | codefreeze | clickonce: I'm definitely biased toward extensions.ael |
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| 13:15.52 | mosty | clickonce, one major benefit of extensions.conf is that you will find a lot more help/docs/etc. even if it is quite an ugly language |
| 13:16.16 | clickonce | Okay :( |
| 13:16.17 | clickonce | :) |
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| 13:28.10 | zeeesh | when "service iptables is stop" asterisk server is working fine... when "service iptables start" unable to hear voice,IVR, although call is connected untill i disconnected it ????????? |
| 13:28.49 | *** join/#asterisk Migrane (n=DirtyD@ool-18bddaa0.dyn.optonline.net) |
| 13:28.54 | lmadsen | zeeesh: you haven't opened the appropriate ports, obviously |
| 13:29.02 | jameswf-home | ~ports |
| 13:29.03 | jbot | i heard ports is http://www.debian.org/ports/, or http://www.isi.edu/in-notes/iana/assignments/port-numbers, or the FreeBSD ports system etc etc, or http://www.portforward.com/routers.htm |
| 13:29.10 | jameswf-home | doh |
| 13:29.11 | Migrane | Hi. I'm trying to config a t1 pri line for the first time in my life. |
| 13:29.49 | jameswf-home | congrats |
| 13:30.21 | lmadsen | zeeesh: see chapter 4, around page 97 of TFoT2 |
| 13:30.28 | Migrane | Right now, the back of my digium card shows a yellow alarm, from my asterisk CLI.. pri intense debug shows a continues "Unnumbered frame: SAPI: 00....." msg over and over. |
| 13:31.05 | Migrane | Is that frame from my side or the remote side? |
| 13:31.08 | jameswf-home | Migrane: pastebin your zaptel.conf |
| 13:31.14 | Migrane | k |
| 13:31.17 | jameswf-home | ~pb |
| 13:31.18 | 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 |
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| 13:34.08 | zeeesh | <lmadsen>:port 5060 is opened . coz .. my sip client is registering by using 5060 ,,, its making calls .. but i m unable to hear any voice and IVRs .. when i just enter the command at my asterisk server "service iptables stop" .. asterisk server working fine .. ???? |
| 13:34.32 | lmadsen | zeeesh: RTP carries the audio, not SIP |
| 13:34.38 | jameswf-home | zeeesh: 10000-20000 |
| 13:34.44 | lmadsen | jameswf-home: don't make it too easy :) |
| 13:34.47 | clickonce | In ext..ael I have context welcomemenu { ..., in ext..conf I want to goto that context, but exten => 4561,1,Goto(welcomemenu) doesn't work, I assume it's not the proper way to goto it, so, how should it be done? |
| 13:34.57 | jameswf-home | oh sorry retract that |
| 13:34.58 | lmadsen | and by 10000-20000, jameswf-home means whatever you have configured in rtp.conf |
| 13:34.59 | *** join/#asterisk oej (n=olle@194.171.177.169) |
| 13:35.22 | jameswf-home | ~buybook |
| 13:35.23 | jbot | You can buy "Asterisk the Future of Telephony" at http://www.oreilly.com/catalog/9780596510480/ so go buy it SERIOUSLY |
| 13:35.42 | lmadsen | clickonce: that would only goto a priority label called welcomemenu -- you need Goto(welcomemenu,EXTENSION,PRIORITY) |
| 13:35.54 | lmadsen | change EXTENSION and PRIORITY to something useful |
| 13:36.04 | Migrane | here it my zaptel.conf jameswf-home http://pastebin.com/m2ea63a68 |
| 13:36.18 | mosty | goto with only one argument assumed that the argument is a priority (from memory) |
| 13:36.20 | clickonce | lmadsen: ah, okay |
| 13:36.42 | jameswf-home | span=1,1,3,esf,b8zs is your problem LBO is almost never 3 |
| 13:36.53 | jameswf-home | make LBO 0 |
| 13:37.40 | Migrane | ok.. |
| 13:37.45 | Migrane | let me try |
| 13:37.51 | jameswf-home | 0 works 99.9999999999% of the time |
| 13:38.27 | jameswf-home | just like d4,ami wtf who uses that |
| 13:38.39 | jameswf-home | kentuckey :) |
| 13:38.50 | patrick-- | I keep on loosing my parked callers. i put them on park and then after a few seconds waiting i get a busy tone and they are somewhere lost in my asterisk :D how can i get them back? |
| 13:39.03 | patrick-- | and in most cases the pbx crashes if the caller waits too long |
| 13:39.06 | jameswf-home | patrick--: what asterisk version |
| 13:39.07 | srd2 | What about that 0.00000000001% of the time? |
| 13:39.10 | *** join/#asterisk pylinuxian (n=pylinuxi@adsl196-19-53-217-196.adsl196-10.iam.net.ma) |
| 13:39.17 | pylinuxian | hi every1 |
| 13:39.23 | patrick-- | jameswf-home: 1.4.18 |
| 13:39.31 | jameswf-home | oh snap |
| 13:39.33 | jameswf-home | :) |
| 13:39.33 | pylinuxian | I have got a question regarding E1 |
| 13:39.36 | patrick-- | ? |
| 13:39.39 | jameswf-home | ~ask |
| 13:39.40 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 13:40.13 | Migrane | james: I still have a yellow alarm when I do zap show status |
| 13:40.21 | patrick-- | Im using asterisk 1.4.18 with mISDN and beroNet BN4S0, where Port 1 is TE and 2-4 are NT for ISDN Phones. |
| 13:40.36 | pylinuxian | ok ! i ask : I need to inteface with a voice gateway model QUINTUM DX2030 |
| 13:40.37 | jameswf-home | Migrane: did you ztcfg |
| 13:40.47 | Migrane | james: I did. |
| 13:41.01 | pylinuxian | from my DIGIUM TE212P Card |
| 13:41.12 | jameswf-home | ok next change timing to 0 if that doesnt work get a loop back |
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| 13:41.38 | pylinuxian | has anybody tries to interface with a Quintum E1 port ? |
| 13:41.41 | jameswf-home | james must hit the commute see you all in 1.25 hours |
| 13:44.42 | *** join/#asterisk pylinuxian (n=pylinuxi@adsl196-19-53-217-196.adsl196-10.iam.net.ma) |
| 13:44.44 | patrick-- | can anyone tell me how i can get parked callers back on the line? |
| 13:45.01 | pylinuxian | got disc |
| 13:45.09 | ManxPower | patrick--: dial the number that was read back to the person that parked the call. |
| 13:45.10 | pylinuxian | <pylinuxian> has anybody tries to interface with a Quintum E1 port ? |
| 13:45.22 | patrick-- | that was read? |
| 13:45.34 | pylinuxian | well ? |
| 13:45.41 | pylinuxian | as i got disc i don't know |
| 13:45.42 | ManxPower | pylinuxian: How is the Quintum different from all the other T-1/E-1 devices out there? |
| 13:46.02 | patrick-- | ManxPower: I park calls by Pressing the R button on my Phone |
| 13:46.13 | pylinuxian | well its a voice gateway ... & its my first time ... |
| 13:46.19 | ManxPower | patrick--: when you park a call you should hear the parking lot number. |
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| 13:46.29 | ManxPower | patrick--: R = Flash |
| 13:46.39 | patrick-- | so what does that mean? |
| 13:46.39 | robl^ | patrick--: when you park a call, a voice tells you a number like "seven zero one". Just pick up a phone and dial those digitis |
| 13:46.41 | ManxPower | You need to do an ATTENDED transfer to the parking extenson |
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| 13:47.09 | pylinuxian | so ? |
| 13:47.10 | *** part/#asterisk SteveTotaro (n=Elizabet@pool-71-166-102-100.bltmmd.east.verizon.net) |
| 13:47.25 | patrick-- | when i redirect calls, i press R, dial the target number and then just put the phone down |
| 13:47.41 | ManxPower | patrick--: then you are doing it wrong. |
| 13:47.43 | [TK]D-Fender | patrick--: You have to do an attended transfer to 700. |
| 13:48.03 | ManxPower | How about you dial the number, wait for the number to be read back to you, then complete the transfer. |
| 13:48.16 | [TK]D-Fender | patrick--: Having "include => parkedcalls" in your phone's context. |
| 13:48.25 | patrick-- | sigh |
| 13:48.34 | [TK]D-Fender | patrick--: While in the transfer * will read back which lot # you have to DIAL to get them back |
| 13:48.41 | patrick-- | i need to use the R button |
| 13:48.55 | ManxPower | patrick--: The R button is just a way of doing a transfer. |
| 13:48.58 | [TK]D-Fender | patrick--: No, you DON'T. That is not how this works. |
| 13:49.06 | [TK]D-Fender | patrick--: Time to do it properly. |
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| 13:49.52 | tzanger | [TK]D-Fender: http://www.mixdown.ca/~andrew/dump/stfu.jpg <-- I want that on a poster |
| 13:51.01 | kyron | tzanger, nice poster ;) |
| 13:53.14 | [TK]D-Fender | tzanger: Get me a better res, I have a banner-capable inkjet here :) |
| 13:53.21 | robl^ | [TK]D-Fender: don't just love when someone asks you how to do something and then after you tell them, they tell you they don't want to do it that way and can't understand why it doesn't work? |
| 13:53.33 | tzanger | :-) I know, I am emailing a friend of mine right now to see if she can create a high-res vesion of this. |
| 13:53.48 | tzanger | She can do the font, and I'm sure she can find a pic of an institution of higher learning similar to the background |
| 13:53.57 | tzanger | the private though might be a problem |
| 13:54.10 | tzanger | she can put it together and I'll get something done up... I so want that for a poster though |
| 13:55.25 | ManxPower | robl^: pretty standard around here. |
| 13:56.28 | kyron | robl^, kids are always like that |
| 13:58.10 | ManxPower | good things robl^ is too smart to have kids. |
| 13:58.14 | robl^ | ManxPower: same here in my office. "Robert, why can't I dial international long distance?" "You forgot to dial 011 before the country code." "But I don't to dial those extra digits." |
| 13:58.52 | tzanger | what would hte name of the private be in that pic? It's a famous picture with the double-cup from WW2-era... |
| 13:59.01 | ManxPower | "Sorry, we limit Interntional calls to people with the special code. Don't tell anyone, that special code is "011" |
| 13:59.19 | robl^ | I have an 18 yr puppy -- he's close enough to being a "kid" |
| 13:59.28 | michael-i | Hi everyone. I think I'm missing something regarding nat=yes|no|never in sip.conf. What situations would having nat=yes cause problems? Is it normally safe just to set it? |
| 13:59.53 | ManxPower | michael-i: a few less common phones have had issues with nat=yes in the past. |
| 14:00.10 | ManxPower | Yes, it is normally safe to set it, even on non-NAT connections. |
| 14:00.33 | michael-i | ManxPower: thanks, that's the conclusion my reading had led me to. Just looking for any gotchas. |
| 14:00.42 | clickonce | Yay! Got queueing working. |
| 14:00.58 | robl^ | I usually leave nat=ye, only time it tends to be an issue is if the phone it self has NAT features turned on. |
| 14:01.55 | *** join/#asterisk steliosk (n=Stelios@athedsl-105216.home.otenet.gr) |
| 14:01.56 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:02.02 | ManxPower | I find the best way to get people to use something is to forbid them from using it. |
| 14:03.09 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 14:03.31 | robl^ | ManxPower: true.. but then we have people eager to use things.. they just don't want to take 30 secs to learn how to use it correctly -- so they yell at me for 2 hours |
| 14:03.53 | *** join/#asterisk ruied (n=ruied@pal-213-228-184-21.netvisao.pt) |
| 14:04.34 | [TK]D-Fender | ManxPower: Perverse psychology :p |
| 14:04.36 | ManxPower | robl^: that's the genius, they can't complain if they are not supposed to be using whatever it is. |
| 14:05.17 | robl^ | hahah |
| 14:05.35 | ManxPower | for a while we had people NOT dialing the 1 before the 504 area code. |
| 14:05.49 | JenniferAkemi- | has anyone here used the ipod touch to do sip? |
| 14:05.52 | stansmith | hello? |
| 14:06.02 | ManxPower | So, I put in _9504NXXXXXX,1,Playback(must-dial-1-and-areacode) |
| 14:06.21 | ManxPower | stansmith: say something interesting and someone might respond. |
| 14:07.45 | stansmith | has anyone here used the app_swift module? I have installed it, I would just like to talk about a couple issues |
| 14:07.59 | [TK]D-Fender | ManxPower: You could always just ADD the "1" and not nag your users. My dialplans adapt to 7-10-11 digit all seamlessly |
| 14:08.15 | x86 | [TK]D-Fender: amen... mine too |
| 14:08.23 | ManxPower | [TK]D-Fender: we could do that, but we don't. |
| 14:08.36 | x86 | ManxPower: that's user abuse ;) |
| 14:08.46 | [TK]D-Fender | ManxPower: Surprising since getting idiots to change isn't your forte... |
| 14:08.50 | ManxPower | x86: no, that is the standard dialplan instructions. |
| 14:09.22 | ManxPower | [TK]D-Fender: I don't CARE if they can't call someone. They can follow the directions or not use the phone. |
| 14:09.56 | robl^ | [TK]D-Fender: we do 10/11 easily.. but no 7 digit dialing (except for internal site dialing (3 digit office code) + (4 digit extension). we have 3 area codes in our city that are all LOCAL, so it is mandatory to always dial areacode |
| 14:10.03 | ManxPower | It's pretty simple. Calls within same area code 9+7-digit number. All other calls 9+1+area code+ 7-digit number. |
| 14:10.07 | [TK]D-Fender | ManxPower: Wow, forcing them to evolve beyond Homo Verbratis Gellatinous, I'm impressed! |
| 14:10.36 | [TK]D-Fender | robl^: Here we can still assume 1 for the most-part |
| 14:11.13 | *** join/#asterisk oej (n=olle@194.171.177.169) |
| 14:11.15 | ManxPower | Unfortunatly, we are now starting to use carriers that have local and toll calls in the same area code. |
| 14:11.24 | ManxPower | That complicates things a litt.e. |
| 14:11.51 | robl^ | eww! then you have to look at the excahnges |
| 14:12.20 | ManxPower | robl^: nope. we let the carrier play a message telling them what to do. |
| 14:12.57 | tzanger | woohoo! I found a nice high res version of that guy |
| 14:13.05 | tzanger | I can buy the original poster for $25 too |
| 14:13.37 | stansmith | ou |
| 14:13.39 | [TK]D-Fender | tzanger: Worth it for the cost / quality of printing it yourself. |
| 14:13.54 | tzanger | yep, but that's not just what I want |
| 14:13.57 | tzanger | I need the whole thing put together |
| 14:13.59 | *** part/#asterisk SAL123 (i=ondrejj@work.salstar.sk) |
| 14:14.34 | robl^ | ManxPower: ahhh. we have to be a little more selective here. if something is considered non-local, the call has to be flagged and require a billing code before the call completes. |
| 14:15.00 | ManxPower | robl^: *nod* We should do that and may do that in the future. |
| 14:15.10 | *** join/#asterisk oej (n=olle@194.171.177.169) |
| 14:15.12 | JenniferAkemi- | robl^: do you do that in a database? |
| 14:15.25 | ManxPower | For most of the offices all calls to Louisiana and Mississippi are free. |
| 14:15.55 | ManxPower | Toll calls come out of a pool of something like 20,000 mins/month that is included in the phone service. |
| 14:16.30 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 14:17.03 | robl^ | JenniferAkemi-: yes. it integrates with our internal systems.. EVERYTHING is tagged with a Client / Matter number and we use a central database that integrates with the phone, finances, HR, email, document management systems, etc. all hightly integrated |
| 14:17.30 | ManxPower | robl^: Your users bill the client for every phone call, so you need to do that |
| 14:17.32 | JenniferAkemi- | robl^: that's great |
| 14:18.08 | JenniferAkemi- | robl^: my task now that I have the basic asterisk working with phones and a pri is to make it be database driven |
| 14:18.13 | robl^ | ManxPower: every phone call, postage stamp, photocopy, printed document, cup of coffe, text message.. |
| 14:18.47 | robl^ | if they could figure out a way to bill toilet paper usage back to a client, theywould |
| 14:19.04 | JenniferAkemi- | do you work at a law firm or something? |
| 14:19.34 | ManxPower | If lawyers spent less time finding ways to squeeze money out of clients and more time lawyering...well...the would be better lawyers. |
| 14:19.36 | robl^ | JenniferAkemi-: yes. I work at a very large law firm. |
| 14:19.52 | ManxPower | robl^: you really need to get me a consulting gig there. 8-) |
| 14:20.13 | *** join/#asterisk adjohn (n=adjohn@p4053-ipad401marunouchi.tokyo.ocn.ne.jp) |
| 14:21.07 | robl^ | ManxPower: we already have nearly 120 I. T. / Telecom ppl in the firm. I am sure we could squeeze in another |
| 14:21.16 | stansmith | does X11 have that much of an impact on resources if the server has a dual core xeon and 2 gigs of ram? |
| 14:21.25 | stansmith | while running asterisk |
| 14:21.30 | [TK]D-Fender | stansmith: No, thats fine |
| 14:21.31 | ManxPower | robl^: Yeah, but I'm better at networking and telecom and all of them combined |
| 14:22.12 | ManxPower | stansmith: it's not X11, it's the fact that almost all graphics chips lock interrupts for long enough to corrupt audio and signalling information |
| 14:22.16 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:22.52 | ManxPower | If you are only doing VoIP with no telecom cards in the server, you might be able to get away with it. |
| 14:23.15 | stansmith | i read not using X11 is reccomended, and im having audio quality issues using a digium card |
| 14:23.30 | robl^ | if you have a digium card, turn off X |
| 14:23.34 | stansmith | yea ok |
| 14:23.38 | ManxPower | stansmith: Good thing you know how to fix that. |
| 14:23.46 | stansmith | ManxPower: ? |
| 14:23.55 | ManxPower | stansmith: you fix it by not running X |
| 14:24.00 | [TK]D-Fender | stansmith: Stop X and see if that helps. And while you're at it you should be describing everything involved in these calls. You could waste a long time guessing what it could be if you don't tell us what you're actually doing. |
| 14:24.51 | stansmith | setting up an IVR on new hardware |
| 14:25.00 | stansmith | i set it up already on old hardware and it works great |
| 14:25.09 | *** join/#asterisk BadHorsie (n=sebas@201.198.239.167) |
| 14:25.29 | stansmith | the new hardware is a HP ML350..and as a part of HP, you need to install a lot of their proliant stuff to get the drivers for the nic card |
| 14:25.35 | stansmith | and to install that, you need X11 |
| 14:25.49 | ManxPower | then turn off X after it's installed |
| 14:26.40 | BadHorsie | hi, i'm still running asterisk 1.2.14, and for some reason, when sporadically i see that show channels (or action: status) doesn't show some channels that are currently active, but, after a while it shows them back with their same ID and i know the call hasn't dropped coz i'm chanspy'ing on it, anybody has an idea of why could this be happening? |
| 14:26.51 | ManxPower | No matter how much you scream, complain, and otherwise refuse to accept reality it does not change the fact that running graphics on the server (X11, frame buffer, etc) will frequently cause problems |
| 14:27.22 | ManxPower | BadHorsie: first upgrade to the latest 1.2.x |
| 14:27.41 | stansmith | ManxPower: I would rather not run X11, I would like to use Arch linux, but my boss recommends using CentOS or RHES |
| 14:28.07 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 14:28.09 | ManxPower | neither CentOS nor RHES require X11 to run. |
| 14:28.20 | stansmith | correct, but i need to install those HP drivers |
| 14:28.23 | stansmith | and that requires X11 |
| 14:28.24 | BadHorsie | 1.2 is not deprecated yet right? |
| 14:28.36 | ManxPower | so install the drivers then turn off X |
| 14:28.50 | [TK]D-Fender | BadHorsie: Yes, there are no more bug fixes coming. It is DEAD |
| 14:28.51 | BadHorsie | HP drivers requiring X11, now THAT is loathsome |
| 14:28.52 | stansmith | yea we have already come to this conclusion haha |
| 14:28.59 | ManxPower | BadHorsie: 1.2 is no longer maintained except for maybe security fixes |
| 14:29.10 | robl^ | stansmith: once the systems is all installed and running, you can turn off X. it's just a program that runs on the box like anything else |
| 14:29.10 | [TK]D-Fender | stansmith: Just bloody-well TEST it already |
| 14:29.21 | BadHorsie | time to search for a migration guide from 1.2 to 1.4 then. |
| 14:29.23 | ManxPower | stansmith: "chkconfig dm off" |
| 14:29.39 | stansmith | dm = desktop manager? |
| 14:29.53 | ManxPower | BadHorsie: it's called "upgrade.txt" or similar in 1.2 and 1.4, you should read the files in both versions |
| 14:30.10 | ManxPower | stansmith: Display Manager, it's what starts X on most distros |
| 14:30.28 | stansmith | hm...thats good to know, i thought i would change the runlevel |
| 14:30.37 | ManxPower | you can do that too. |
| 14:30.59 | ruied | I have the following error: " Unable to open file '/var/lib/asterisk/moh/fpm-world-mix': No such file or directory". but the file exists. where is the 'fpm-world-mix-file' defined? |
| 14:31.35 | ManxPower | ruied: Where is the file defined? |
| 14:31.50 | ManxPower | ruied: MoH is configured in musiconhold.conf |
| 14:32.47 | ManxPower | ruied: what is the file extension in the results of this command: ls -l /var/lib/asterisk/moh/fpm-world-mix.* |
| 14:33.50 | ruied | ManxPower, .wav |
| 14:34.03 | ManxPower | ruied: and when do you get that error? |
| 14:34.20 | ruied | when I put a call on hold... |
| 14:34.46 | ManxPower | I guess you need to put the contents of /etc/asterisk/musiconhold.conf on pastebin.ca then |
| 14:35.16 | ManxPower | ruied: and what is the file size reported by the "ls" |
| 14:35.35 | ruied | ManxPower, it's the default configuration... |
| 14:35.48 | stansmith | ruied: are you loading the wav format modules before loading res_musiconhold ? |
| 14:35.58 | ManxPower | ruied: "default configuration" changes depending on the release. |
| 14:36.10 | *** join/#asterisk rpm (n=russell@S01060014f6e07140.cg.shawcable.net) |
| 14:36.22 | [TK]D-Fender | ManxPower: You don't stop X from loading by removing it with chkconfig... you just change your bloody run-level :p |
| 14:36.32 | stansmith | lol |
| 14:36.55 | hmm-home | not in all distro's |
| 14:37.04 | stansmith | RH-based? |
| 14:37.10 | [TK]D-Fender | Disclaimer : I suck at Linux and get by on 30% Instinct, and 70% Google. |
| 14:37.15 | hmm-home | some call gdm in run level 3 |
| 14:37.18 | ruied | stansmith, maybe not, going to check, it's the default configuration... |
| 14:37.32 | stansmith | ruied: does that mean you typed "make samples" during compilation? |
| 14:37.44 | ruied | yes |
| 14:38.02 | ManxPower | ruied: the sample config files are NOT designed to work. They are designed to show examples of many things. |
| 14:39.05 | ruied | stansmith, ManxPower Wht I normally do is a make samples and then cut what I don't need... maybe it is a bad idea... |
| 14:39.15 | stansmith | i do the same |
| 14:39.42 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:39.44 | ManxPower | Too bad I never got to see that musiconhold.conf. Now I have to go to work. |
| 14:39.47 | stansmith | im in the process of whittling the system down to just whats needed though |
| 14:41.01 | ruied | ManxPower, sorry, have a good work! I'll get there.... |
| 14:41.06 | *** join/#asterisk tobias (n=tobias@cpe-066-057-253-087.nc.res.rr.com) |
| 14:41.47 | *** join/#asterisk Sniper_linu (n=michofr@mail.splendor.net) |
| 14:42.00 | stansmith | anyone here use Cepstral TTS? |
| 14:42.31 | [TK]D-Fender | ruied: pastebin your msuiconhold.conf , "ls -la /var/lib/asterisk/moh", and the CLI output of your failed attempt at verbose 10 |
| 14:42.32 | [TK]D-Fender | ~pb |
| 14:42.33 | 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 |
| 14:42.42 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 14:42.45 | *** join/#asterisk jbigbee (n=jbigbee@216.207.245.1) |
| 14:43.36 | Sniper_linu | Hi All,I have an asterisk server installed on a Centos machine... I have a problem when making a call that the asterisk server is not sending the codecs supported by the asterisk it self |
| 14:43.52 | Sniper_linu | my question is where I can define the codecs supported by the asterisk server? |
| 14:44.14 | [TK]D-Fender | Sniper_linu: in the appropriate channel-driver's config file. |
| 14:44.43 | [TK]D-Fender | Sniper_linu: typcailly sip.conf / iax.con / users.conf (for those hapless chumps running the UNSUPPORTED GUI) |
| 14:44.53 | *** join/#asterisk JoseBravo (n=jbravo@190.156.225.15) |
| 14:45.24 | BadHorsie | i just saw that with action: status it shows up to 100 lines of output, so maybe under high load i will only get 100 lines no matter how many channels are active, right? |
| 14:45.31 | Sniper_linu | [TK]D-Fender, DO you mean that the codec lists is defined in all these files? |
| 14:45.51 | stansmith | Sniper_linu: if you are only using SIP, you only need to worry about sip.conf |
| 14:45.55 | stansmith | for instance.. |
| 14:46.11 | Sniper_linu | stansmith, let me che ck this fileand let you know |
| 14:46.32 | JoseBravo | Im receiving call from a SIP over Internet to my PBX, with out problems, and I have a FAX connected over a Linksys Phone Adapter, but I can't receive faxes over the SIP. I have connected to same PBX a FXO and the fax works fine. Any idea? |
| 14:46.56 | *** join/#asterisk hmmhesays (n=hmmhesay@24-119-176-74.cpe.cableone.net) |
| 14:47.26 | Sniper_linu | stansmith, I have g711 and g729 codecs defined there, as follow: |
| 14:47.27 | Sniper_linu | bindport=5060; UDP Port to bind to (SIP standard port is 5060) |
| 14:47.27 | BadHorsie | which makes me think action: status is not the right way to gather the channel information, what would be the proper way to gather ALL the active channels (no matter how many they are) from the AMI? |
| 14:47.27 | Sniper_linu | bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) |
| 14:47.27 | Sniper_linu | disallow=all |
| 14:47.27 | Sniper_linu | allow=ulaw |
| 14:47.27 | Sniper_linu | allow=g711 |
| 14:47.36 | stansmith | pastebin |
| 14:47.41 | stansmith | ~pastebin |
| 14:47.41 | jbot | [~pastebin] 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 |
| 14:48.04 | ruied | [TK]D-Fender, http://paste.uni.cc/18367 |
| 14:48.26 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 14:50.06 | JoseBravo | In other words, My PBX have two incoming trunks (SIP over Internet and FXO), I also have a FAX connected to my PBX, if anyone send me a FAX over the FXO works fine, but if he send over SIP the fax dosn't works. I can receive voice calls from SIP and FXO. Any idea? |
| 14:50.47 | jameswf-home | lmadsen: ping. |
| 14:51.45 | ruied | [TK]D-Fender, It seems a g729 translation problem... doesn't Digium B410P came with g729 licences |
| 14:52.43 | lmadsen | jameswf: yo |
| 14:53.34 | JT | ruied: why would it come with any G.729 licenses? |
| 14:53.42 | [TK]D-Fender | ruied: NO |
| 14:53.51 | lmadsen | jameswf: you have exactly 10 seconds, then I gotta leave |
| 14:53.56 | *** join/#asterisk JayTee52 (n=jforde@207-67-84-188.static.twtelecom.net) |
| 14:54.04 | jbigbee | ruied, The TC400B is the card that comes with g.729 |
| 14:54.43 | [TK]D-Fender | Sniper_linu>allow=g711 <- nope. G.711u = ulaw, G.711a = alaw |
| 14:54.47 | jbigbee | ruied, with the latest drivers it supports about 120 g.729 liceses |
| 14:54.53 | [TK]D-Fender | Sniper_linu: Get your formatting right |
| 14:55.29 | *** join/#asterisk Docfxit (n=none@cpe-72-129-118-135.socal.res.rr.com) |
| 14:55.53 | *** join/#asterisk ZaVoid (n=zavoid@66-95-182-90.client.dsl.net) |
| 14:55.54 | Sniper_linu | [TK]D-Fender, you mean instead of allow=g711 I should put G.711=ulaw? |
| 14:55.56 | [TK]D-Fender | JoseBravo: fax over SIP is likely to fail. compression, jitter, delay, etc will kill faxes |
| 14:56.05 | [TK]D-Fender | Sniper_linu: "allow=ulaw" |
| 14:56.05 | ZaVoid | everyone see that lunar eclipse last night? |
| 14:56.13 | ruied | hmm, ok, so it seems that is the problem... |
| 14:56.20 | hmmhesays | blood moon |
| 14:56.21 | ruied | thanks... |
| 14:56.22 | hmmhesays | it was sweet |
| 14:56.46 | Sniper_linu | [TK]D-Fender, I have it already |
| 14:57.13 | [TK]D-Fender | Sniper_linu: pastebin the complete failed call attempt at verbsoe 10, SIP debug enabled. |
| 14:57.13 | ZaVoid | http://www.flickr.com/photos/zavoid/2280330377/in/photostream/ |
| 14:57.15 | [TK]D-Fender | ~pb |
| 14:57.15 | 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 |
| 14:57.17 | [TK]D-Fender | ^^^^^^^^ |
| 14:57.20 | ZaVoid | picutre of it THE ELCIPSE |
| 14:57.38 | stansmith | elcipse, lol |
| 14:57.45 | Sniper_linu | [TK]D-Fender, ok I'll do that |
| 14:57.50 | hmmhesays | nice shots |
| 14:58.28 | *** join/#asterisk webar7 (n=webart@CPE0080c8f208a5-CM001371173cf8.cpe.net.cable.rogers.com) |
| 14:58.49 | JenniferAkemi- | oops. i guess you book guys already know about the typo on page 159, but just in case you don't it has MCARO_EXTEN instead of MACRO_EXTEN |
| 14:59.04 | JenniferAkemi- | ps. thanks for the book. love it. |
| 14:59.11 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 14:59.32 | webar7 | I am almost there :) .... now when I am in the console and type sip show peers I get : |
| 14:59.35 | webar7 | 222/222 (Unspecified) D 0 UNKNOWN |
| 14:59.35 | *** join/#asterisk _foxfire_ (n=_foxfire@cica-adm.fe.up.pt) |
| 14:59.37 | hmm-home | ahh the book |
| 14:59.58 | hmmhesays | Is it good I i've never read a page out of it |
| 15:00.03 | webar7 | but if I drop my firewall then I can see the peer |
| 15:00.05 | Nivex | ZaVoid: nice! Mine didn't turn out as good |
| 15:00.13 | JenniferAkemi- | it's good for if you don't know anything when you start hmmhesays |
| 15:00.17 | JenniferAkemi- | which was me |
| 15:00.26 | ZaVoid | thanks Nivex |
| 15:00.34 | webar7 | I have port 5060 open for my LAN so shouldn't I be able to see the phone? |
| 15:00.38 | JenniferAkemi- | and i bet there are some things in it that you don't even know about that could be totally useful |
| 15:00.45 | JenniferAkemi- | like the (!) template thing |
| 15:01.01 | webar7 | do I need another set of ports open for the SIP registration to work? |
| 15:01.07 | Sniper_linu | [TK]D-Fender, http://pastebin.com/m2f70b352 |
| 15:01.36 | Sniper_linu | This is the pastein address, the SIP packet has been captured on the PSTN switch Server |
| 15:01.47 | [TK]D-Fender | Sniper_linu: I said COMPLETE. |
| 15:01.58 | Sniper_linu | OK |
| 15:01.59 | stansmith | lol |
| 15:02.22 | [TK]D-Fender | Sniper_linu: and include your [general] section of sip.conf, and any peers matched in your attempt |
| 15:03.03 | Sniper_linu | [TK]D-Fender, I'll do that, thx a lot |
| 15:03.04 | *** join/#asterisk wmaulik (n=wmaulik@158.59.192.218) |
| 15:03.52 | JoseBravo | Anyone can helpme? |
| 15:05.02 | webar7 | isn't port 5060 sufficient? |
| 15:05.17 | [TK]D-Fender | webar7: NO |
| 15:05.18 | stansmith | webar7: im taking a wild shot, but you opened tcp and udp? |
| 15:05.19 | Sniper_linu | [TK]D-Fender, the SIP request is on the following link |
| 15:05.22 | Sniper_linu | [TK]D-Fender, http://pastebin.com/m591775da |
| 15:05.26 | [TK]D-Fender | webar7: 10000-20000 as well |
| 15:05.37 | Sniper_linu | [TK]D-Fender, I'll paste now the sip.conf file |
| 15:05.39 | webar7 | stansmith, yeah |
| 15:06.02 | webar7 | [TK]D-Fender, yeah I did that part fron the rtp.conf |
| 15:06.07 | [TK]D-Fender | Sniper_linu: Where is this debug coming from? |
| 15:06.25 | webar7 | I'm sort of trunking different channels |
| 15:06.39 | Sniper_linu | [TK]D-Fender, This SIP packets has been captured on the PSTN gateway |
| 15:06.42 | [TK]D-Fender | webar7: Are you trying to have a remote phone register to an * beox that is behind NAT? |
| 15:06.49 | [TK]D-Fender | Sniper_linu: obtained how? |
| 15:06.55 | webar7 | calls come in and out via IAX but then go from the asterisk box to the phones using SIP |
| 15:06.56 | Sniper_linu | [TK]D-Fender, by a snoop |
| 15:07.12 | [TK]D-Fender | Sniper_linu: NO. I want' * CLI SIP debug at verbose 10 like I asked the first time. |
| 15:07.50 | Sniper_linu | [TK]D-Fender, Can you check please the SDP on the INVITE packet sent by the asterisk server? |
| 15:08.08 | webar7 | [TK]D-Fender, basically [internet cloud+POTS]<----->DID@ITSP<------IAX2----->[my LAN with *box]<---SIP--->phones |
| 15:08.40 | _foxfire_ | hello guys, i am having a serious problem here at our university, we are running asterisk 1.2 and we want to upgrade to version 1.4. I 've changed the configuration files to be compatible with 1.4. Our system has an Digium, Inc. Wildcard TE210P, when we start getting some calls , the hole thing crashed , after an cold reboot it took about 5-10 minutes to crash again. |
| 15:08.43 | [TK]D-Fender | Sniper_linu: Please provide the output I have have now repeatedly requested. |
| 15:08.44 | Sniper_linu | [TK]D-Fender, if you check it then you'll find that the asterisk server is not trying to negotiate any codec there |
| 15:08.45 | webar7 | [TK]D-Fender, since I am "translating" a system that worked under 1.2 to 1.4 I am fixing things but missing others |
| 15:08.54 | Sniper_linu | [TK]D-Fender, |
| 15:09.00 | Sniper_linu | Let me do that plz |
| 15:09.12 | _foxfire_ | unfortunatly the system is in production so i could not play araound and had to revert to 1.2 , the kernel dump pointed to an zaptel problem |
| 15:09.24 | Sniper_linu | [TK]D-Fender, In the meantime can you check please the sip.conf file? |
| 15:09.40 | [TK]D-Fender | _foxfire_: Here's hoping that you upgraded Zaptel to an appropriate version as well, all from scratch |
| 15:10.06 | _foxfire_ | iup upgraded all the modules to the last one |
| 15:10.17 | [TK]D-Fender | Sniper_linu: sip.conf is where? |
| 15:10.32 | Sniper_linu | [TK]D-Fender, http://pastebin.com/m5b6d0a55 |
| 15:10.47 | *** join/#asterisk Brat3 (n=brast@unaffiliated/brat3) |
| 15:11.08 | [TK]D-Fender | Sniper_linu: Where is your peer section header in there? |
| 15:11.25 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 15:11.43 | Sniper_linu | [TK]D-Fender, this is all the file |
| 15:11.49 | webar7 | [TK]D-Fender, basically the SIP phones aren't visible because of a firewall issue I guess ... |
| 15:12.15 | webar7 | [TK]D-Fender, but I have opened every port between the phones and the * box :) |
| 15:12.16 | Sniper_linu | [TK]D-Fender, Should I add anything? |
| 15:12.23 | [TK]D-Fender | Sniper_linu: fine, continue to get the infor I first asked for. |
| 15:12.35 | [TK]D-Fender | webar7: NOT ENOUGH : go read : |
| 15:12.36 | [TK]D-Fender | ~sipnat |
| 15:12.37 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 15:12.38 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 15:12.39 | [TK]D-Fender | ^^^^^^^^^^ |
| 15:12.46 | Sniper_linu | [TK]D-Fender, ok let me do that please |
| 15:13.08 | *** join/#asterisk Brat3 (n=brast@unaffiliated/brat3) |
| 15:13.18 | webar7 | [TK]D-Fender, ok but tell me that what I am trying to do is OK :) |
| 15:13.19 | _foxfire_ | running asterisk 1.4.18 and zaptel 1.4.8 libpri 1.4.3 |
| 15:13.32 | webar7 | [TK]D-Fender, [internet cloud+POTS]<----->DID@ITSP<------IAX2----->[my LAN with *box]<---SIP--->phones |
| 15:13.44 | webar7 | SIP on the LAN only |
| 15:14.05 | webar7 | with the trunk to my ITSP using IAX2 |
| 15:14.37 | webar7 | [TK]D-Fender, I'm afraid I forgot to turn on some simple channel2channel thingie somewhere |
| 15:15.39 | [TK]D-Fender | _foxfire_: update to Zaptel 1.4.9 and confirm that it is loading successfully then try to start * manually and pastebin the complete output of all of this. |
| 15:15.41 | [TK]D-Fender | ~pb |
| 15:15.42 | 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 |
| 15:15.43 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 15:16.08 | [TK]D-Fender | webar7: pastebin is your friend. Grab some meaningful output for us to examine. |
| 15:16.16 | webar7 | my calls are only sip calls between the phone and the * box ... after that they go via the trunk ... |
| 15:16.18 | webar7 | ok |
| 15:17.51 | _foxfire_ | ok D-Fender will do that , can't do it now , will do it as soon as possible |
| 15:18.24 | _foxfire_ | thanx, did notice that 1.4.9 was out |
| 15:18.48 | _foxfire_ | i meant i did not notice that 1.4.9 was out ;-) |
| 15:21.34 | JoseBravo | [TK]D-Fender how can I receive dax over Internet without SIP? |
| 15:22.00 | [TK]D-Fender | JoseBravo: Go read up on T.38 |
| 15:23.08 | *** join/#asterisk Maxfactor (n=Maxfacto@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 15:23.52 | *** join/#asterisk msetim (n=msetim@200.195.161.164) |
| 15:24.06 | msetim | hi guys |
| 15:24.40 | webar7 | ok |
| 15:25.04 | msetim | How can put a call of an agent in Hold? |
| 15:26.01 | robl^ | msetim: press the HOLD button on a phone? |
| 15:26.34 | coppice | [TK]D-Fender: that's not a nice thing to tell someone to do :-) |
| 15:26.42 | msetim | msetim, without it... I would like to create an application... like #XX |
| 15:26.53 | Maxfactor | g'day all |
| 15:26.59 | msetim | robl^, without it... I would like to create an application... like #XX |
| 15:27.34 | robl^ | msetim: then park the call |
| 15:27.44 | wmaulik | can anyone link any tutorials that explains how to configure the pots line for an asterisk server |
| 15:28.07 | Maxfactor | new to the site...starting from scratch lol |
| 15:28.25 | msetim | robl^, right :) |
| 15:28.59 | msetim | robl^, however it is not "easy to use" to an agent working dialy in a Callcenter. |
| 15:29.29 | robl^ | msetim: are you using analog phones or IP phones? |
| 15:30.50 | msetim | robl^, we are using headset connected with ATA Linksys |
| 15:31.34 | Maxfactor | where will I get a good image of debian? |
| 15:31.48 | Maxfactor | wrong channel |
| 15:31.49 | stansmith | www.goodbye-microsoft.com |
| 15:31.52 | stansmith | oops |
| 15:31.55 | JayTee52 | wmaulik, try the O'Reilly book Asterisk, The Future of Telephony. It's available as a free PDF download on O'Reilly's site. |
| 15:32.23 | *** part/#asterisk exvito (n=exvito@195.245.132.93) |
| 15:32.34 | wmaulik | thanks, do you have the link for the O'Reilly site |
| 15:33.04 | robl^ | msetim: so.. analog. do an attendend transfer to 700 and park the call. you can use a dtmf button sequence for that. |
| 15:33.39 | webar7 | ok does this look bad ? http://rafb.net/p/XtDnML72.html |
| 15:34.12 | robl^ | msetim: if you want easier to use, buy your call center people phones that have "hold" buttons |
| 15:34.30 | JayTee52 | wmaulik, try www.oreilly.com |
| 15:34.34 | msetim | robl^, right :) |
| 15:34.38 | msetim | robl^, thanks |
| 15:34.38 | wmaulik | thanks |
| 15:36.17 | jameswf | ~buybook |
| 15:36.18 | jbot | You can buy "Asterisk the Future of Telephony" at http://www.oreilly.com/catalog/9780596510480/ so go buy it SERIOUSLY |
| 15:37.07 | *** join/#asterisk oppassum (n=op@host-64-179-56-117.gra.choiceone.net) |
| 15:37.21 | oppassum | hey all - newb question -- how do i enable "presence" and "buddywatch"? |
| 15:37.41 | wmaulik | jaytee52 it seems that this book costs money, do you know of any free tutorials |
| 15:37.58 | robl^ | ~book |
| 15:37.59 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 15:38.16 | robl^ | wmaulik: download a PDF of the book for free |
| 15:38.32 | wmaulik | ok |
| 15:38.37 | webar7 | OMFG it works |
| 15:39.15 | webar7 | phone [222] is at IP phone address 192.168.28 |
| 15:39.35 | wmaulik | it worked tyvm |
| 15:39.50 | *** join/#asterisk Anthony_Senna (n=Anthony@LAubervilliers-153-52-32-150.w217-128.abo.wanadoo.fr) |
| 15:40.13 | webar7 | and the asterisk box is at 192.168.2.4 .... in the host= line of the entry for [222] in my sip.conf file I had 192.168.2.8 .... |
| 15:40.25 | webar7 | (the address of the phone) |
| 15:40.49 | Anthony_Senna | hello everybody. I use asterisk 1.4.17 And I have this error when I try to hangup a call: Failed to authenticate on BYE to ..... How can I fixe? |
| 15:40.49 | webar7 | but phone/user 222 registers on the asterisk box ... 192.168.2.4 |
| 15:40.56 | webar7 | so ... |
| 15:42.49 | webar7 | http://rafb.net/p/XtDnML72.html doesn't work ... but http://rafb.net/p/mlxBb499.html does |
| 15:43.29 | webar7 | [TK]D-Fender, if you tell me that was my problem I will have to kill myself ... |
| 15:43.53 | webar7 | I will attempt to kill myself and hopefully be revived |
| 15:44.00 | webar7 | :) |
| 15:44.03 | JayTee52 | robl^, thanks for doing the book thing, I'd forgotten about it and couldn't find the link for the free download. |
| 15:44.41 | webar7 | JayTee52, it is updated for 1.4 which is essential reading! |
| 15:44.45 | Anthony_Senna | or somtimes I have this: Failed to authenticate on INVITE |
| 15:45.04 | [TK]D-Fender | webar7: Well you never even bothered to set your codecs. That is very bad |
| 15:45.07 | JayTee52 | yeah, I've got the 2nd edition PDF. I just couldn't find the link to give to wmaulik |
| 15:45.13 | Poincare | anyone have a clue about why 'hint' works on sip channels defined in the config file but not on sip channels defined via res_mysql? |
| 15:45.31 | robl^ | webar7: your domain= is wrong in both |
| 15:45.48 | webar7 | ? |
| 15:45.52 | wmaulik | thats fine, but ya having an actual text book is much better than jumping for web tutorial to web tutorial |
| 15:46.17 | webar7 | robl^, what should it be .. I got that tip from a user group |
| 15:46.48 | JayTee52 | wmaulik, you can also reference www.voip-info.org and www.asteriskguru.com. They both have loads of usefull info. |
| 15:47.03 | oppassum | hey all - newb question -- how do i enable "presence" and "buddywatch"? |
| 15:48.04 | webar7 | robl^, I do get messages like ---> chan_sip.c:15051 handle_request_register: Registration from '222 <sip:222@192.168.2.4>' failed for '192.168.2.8' - Peer is not supposed to register |
| 15:48.06 | robl^ | domain=192.168.2.8,internal is wrong.. should either be a domain name or an IP.. drop the ",internal" at least |
| 15:48.19 | webar7 | oh |
| 15:48.30 | webar7 | I thought the context could go in there |
| 15:48.32 | webar7 | of |
| 15:48.37 | webar7 | ok |
| 15:49.23 | Sniper_linu | [TK]D-Fender, still here? |
| 15:49.30 | robl^ | context doesn't go there. context is asterisk specific.. domain is a SIP protocol requirement. you will get weird results |
| 15:49.59 | webar7 | ok |
| 15:50.12 | [TK]D-Fender | Sniper_linu: Yes, finally have it? |
| 15:50.14 | JoseBravo | How can I know what codec Im using in my SIP peer? |
| 15:50.46 | Sniper_linu | [TK]D-Fender, Yes I'll paste it soon |
| 15:50.49 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:51.32 | *** join/#asterisk [T]ank (n=ckwall@c-71-195-199-107.hsd1.ut.comcast.net) |
| 15:51.45 | [T]ank | Anyone else here using binfone? |
| 15:52.56 | *** join/#asterisk eelcob (n=niemand@wc-198.r-195-35-139.atwork.nl) |
| 15:54.37 | JoseBravo | How can I know what codec Im using in my SIP peer? |
| 15:54.59 | [T]ank | JoseBravo: sip show channels |
| 15:55.30 | [T]ank | that will show you the codec you are currently using on connected calls. |
| 15:55.56 | [T]ank | if you are just interested in seeing what you have a peer set to either look at it in the sip.conf of do a sip show channele <sip peer> |
| 15:56.01 | [T]ank | sip show channel |
| 15:56.03 | [T]ank | soory |
| 15:56.05 | [T]ank | sorry |
| 15:56.13 | [T]ank | spelling is not my thing this morning apparently |
| 15:57.43 | [T]ank | can anyone recommend a good residential voip provider? I am currently with binfone and they have been down for some reason for the last 24 hours. i have a pay per minute plan at about two cents per minute. recommendations would be appreciated. |
| 15:57.45 | _foxfire_ | D-Fender , i am compiling zaptel as you recomended , there are several warning that pop up , i was ignoring it earlier because modules are build and loaded anyway without any problems |
| 15:57.59 | _foxfire_ | WARNING: "_spin_unlock_irqrestore" [/usr/src/zaptel-1.4.9/kernel/wct4xxp/wct4xxp.ko] undefined! |
| 15:58.20 | Sniper_linu | [TK]D-Fender, Please check the following link |
| 15:58.24 | Sniper_linu | [TK]D-Fender, http://pastebin.com/m4e6e53d1 |
| 15:58.32 | *** part/#asterisk Alexandre_fr (n=alex@nat04-bzr.dys1.com) |
| 15:59.34 | [TK]D-Fender | Sniper_linu: I asked for sip debug, not core debu, and you are using FreePBX which is not supported here. |
| 15:59.37 | [TK]D-Fender | ~freepbx |
| 15:59.37 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 16:01.54 | robl^ | [TK]D-Fender: Why do I get this on my screen? http://pastebin.ca/912707 |
| 16:02.23 | file | robl^: you have a virus! quick, run! |
| 16:03.01 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 16:04.00 | drmessano | Virus = Very Yes |
| 16:04.04 | drmessano | Computer Over! |
| 16:05.05 | Poincare | I'm using Asterisk 1.4.17 with the mysql addons for sip configuration. When I do a 'core show hints', sip channels configured in mysql are always 'Idle', sip channels configured in sip.conf work properly. Any 'hints' for me? |
| 16:05.09 | _foxfire_ | D-Fender by the way i am running aD-fender i am running kernel version 2.6.17.13-smp , is it worth upgrading to the last one ? |
| 16:05.38 | *** join/#asterisk egecko (n=sam@cpe-76-176-238-225.san.res.rr.com) |
| 16:05.45 | file | Poincare: they don't work with realtime. |
| 16:05.48 | *** join/#asterisk mib_3vvtcz9a (i=0ca5bc82@gateway/web/ajax/mibbit.com/x-1ad027264634f938) |
| 16:06.04 | Poincare | file: it worked in 1.2 but not anymore in 1.4 ??? |
| 16:06.30 | file | it was never designed to work with realtime... they require keeping stuff in memory, which doesn't happen with realtime unless you use caching |
| 16:07.11 | Poincare | file: ah ok, and caching can be enabled or just isn't implemented? |
| 16:07.26 | file | it's a sip.conf configuration option |
| 16:07.36 | x86 | rtcachefriends=yes, generally |
| 16:07.42 | file | I *think* it works with caching... haven't tested it |
| 16:07.45 | Poincare | file: ah ok, I'll try that, thanks |
| 16:07.47 | x86 | there's another cache option too |
| 16:07.53 | x86 | forgot it though |
| 16:07.55 | mib_3vvtcz9a | I am running meetme on asterisk 1.14.13 and after a while for some users the audio is changed to oneway.(listen only) Did anybody experienced this? |
| 16:08.22 | *** join/#asterisk CVirus (n=GoD@196.205.192.157) |
| 16:08.36 | mib_3vvtcz9a | Sorry, I am running meetme on asterisk 1.4.13 |
| 16:08.53 | x86 | file: how do you do hints for a range of extensions? say I've got extensions 200-299 and devices SIP/200-SIP/299 |
| 16:09.01 | x86 | file: can i do those with one simple line? |
| 16:09.24 | file | you can not, there is a patch on the bug tracker that tries to implement it but I do not remember the number |
| 16:09.37 | x86 | oh weak |
| 16:09.48 | x86 | ok |
| 16:09.59 | anonymouz666 | corydon wrote this patch |
| 16:11.24 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 16:11.51 | mib_3vvtcz9a | Anybody? |
| 16:13.20 | RoyK | http://karlsbakk.net/fun/ventilation-pipe-art.jpg |
| 16:13.21 | hmmhesays | x86, with a fancy bash script that writes them out for you |
| 16:13.23 | x86 | file: yeah seems like even with caching it only shows Idle, or unavailable if that device is not registered |
| 16:13.35 | x86 | hmmhesays: perl one-liner you mean? :) |
| 16:13.42 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 16:14.13 | hmmhesays | x86 that works too |
| 16:14.20 | *** join/#asterisk adorah (n=Michael@87.69.130.248) |
| 16:14.51 | x86 | seems hint support in asterisk is still not very mature yet |
| 16:14.55 | x86 | will give it some more time :) |
| 16:15.04 | Poincare | file: that does the trick, thanks for your help |
| 16:15.20 | file | so we've got one where it doesn't work, and one where it does |
| 16:15.30 | *** join/#asterisk seanbright (i=seanbrig@65.207.74.18) |
| 16:16.07 | drmessano | If Asterisk only does half as much as it needs to, it still does 3x as much as the other guys |
| 16:16.12 | x86 | file: hmm interesting |
| 16:16.39 | x86 | file: rtcachefriends=yes, rtupdate=yes, rtautoclear=yes # in my sip.conf |
| 16:16.50 | hmmhesays | drmessano, heh |
| 16:17.01 | drmessano | When I was interviewing for this job, they read the stuff about Asterisk on my resume, and asked me if I thought I could work with Cisco VoIP stuff... |
| 16:17.02 | x86 | file: using 1.4.12.1 |
| 16:17.17 | drmessano | I told them "Yeah, I may have to unlearn a few things.. but sure" |
| 16:17.44 | x86 | drmessano: did you tell them you know how to flash cisco phones with the SIP firmware and use them with asterisk too? :) |
| 16:18.20 | drmessano | No, but they have no idea what's in store for them :) |
| 16:18.30 | hmmhesays | cisco's + asterisk = PITA |
| 16:18.35 | hmmhesays | go with a poly |
| 16:18.38 | drmessano | Yep |
| 16:19.16 | hmm-home | I need to find an old version of skype for linux, coming up short |
| 16:19.20 | drmessano | "Did you guys know you can sell a customer Asterisk and Polycom, undercut the other guy, and have a much greater profit margin?" |
| 16:19.25 | drmessano | "....oh?" |
| 16:19.38 | hmm-home | I try and stay away from the end user market |
| 16:19.43 | mib_3vvtcz9a | Can anybody help me out why is dropping audio in meetme conference? |
| 16:19.44 | drmessano | "Oh yeah, and it does more" |
| 16:19.46 | JenniferAkemi- | if i have two offices, and two phones, both for the same person, is it generally considered annoying if i only give them one mailbox? |
| 16:19.57 | hmm-home | JenniferAkemi- I do it all the time |
| 16:20.03 | JenniferAkemi- | on the one hand it seems liek it would be easier to check etc |
| 16:20.11 | stansmith | asl? |
| 16:20.26 | hmm-home | just make sure when they go to check it they are only able to access one box, a well placed func IF takes care of it |
| 16:20.28 | *** join/#asterisk timeshell (n=timeshel@gw.lusi.on.ca) |
| 16:20.28 | JenniferAkemi- | but on the other hand, if you're sitting at DeskA and someone leaves a message at the phone at DeskB, then your MWI light turns on at DeskA that might bea nnoying |
| 16:20.29 | drmessano | Thats how I would do it.. Unless the person is one of those that likes things seperated |
| 16:20.34 | timeshell | Greetings |
| 16:20.57 | timeshell | Is anyone here familiar with the Panasonic Globarange phones that are preprogrammed to use JoiP? |
| 16:21.01 | drmessano | <stansmith> asl? <-- Who was that for? |
| 16:21.03 | JenniferAkemi- | ok thanks guys. |
| 16:21.18 | drmessano | I'm a guy, dude.. stop hitting on me |
| 16:21.20 | timeshell | Are they locked to JoIP or can they be connected to a Asterisk? |
| 16:21.47 | stansmith | cat ran across the keyboard, sorry |
| 16:21.53 | JenniferAkemi- | haha |
| 16:22.07 | drmessano | and hit A, S, and L, followed by ? .... ? |
| 16:22.20 | JenniferAkemi- | do you have a function key bound to "asl?" |
| 16:22.24 | stansmith | yea, amazing how it doesnt have a thumb but could hit shift + ? |
| 16:22.28 | Anthony_Senna | How can I fixe this message: Re-invite to non-existing call leg on other UA. When I hangup a call, my phone ringing during 1 second and hangup after |
| 16:22.28 | hmm-home | now thats how you hit on someone |
| 16:22.38 | hmm-home | canreinvite=no |
| 16:23.23 | pylinuxian | hi - everyone ! anybody has any experience with Quantum Tenor DX2010 |
| 16:23.24 | pylinuxian | ? |
| 16:23.46 | JenniferAkemi- | so calls on my g729 codec sound great. But the DTMFs aren't very good. The asterisk box doesnt' recognize them at all. |
| 16:23.52 | drmessano | JenniferAkemi-, do you even allow PM's in your IRC client? |
| 16:24.06 | drmessano | If you do, you are brave |
| 16:24.09 | hmm-home | only from ugly trolls like me |
| 16:24.13 | JenniferAkemi- | heh |
| 16:24.19 | drmessano | ha |
| 16:24.22 | pylinuxian | I need to connect my Digium Card to a Quintum DX2030 |
| 16:24.23 | stansmith | teehee |
| 16:24.29 | pylinuxian | anybody can help ? |
| 16:24.29 | JenniferAkemi- | i'm just good at ignoring people |
| 16:24.35 | JenniferAkemi- | (if necessary) |
| 16:24.37 | hmm-home | pylinuxian: fun |
| 16:24.42 | mib_3vvtcz9a | Can I ask for help? |
| 16:24.56 | stansmith | mib_3vvtcz9a: no |
| 16:24.59 | JenniferAkemi- | mostly i'm up for being amused though. |
| 16:25.01 | hmmhesays | don't ask to ask |
| 16:25.02 | drmessano | Being a female on IRC is bad enough, being one on the sausagefest known as freenode has to be worth hazardous duty pay |
| 16:25.05 | [TK]D-Fender | JenniferAkemi-: you should be using rfc2833 |
| 16:25.07 | pylinuxian | anybody has any previouse exp in doing this ?. |
| 16:25.12 | hmmhesays | pylinuxian, yes |
| 16:25.14 | x86 | drmessano: hah |
| 16:25.18 | JenniferAkemi- | thanks [TK]D-Fender |
| 16:25.29 | pylinuxian | hmmhesays, good |
| 16:25.36 | drmessano | "ZOMGGG A WOMAN!!!1112!!ONES!!! ASL??? ASL??? ZOMG" |
| 16:25.39 | *** join/#asterisk eelcob (n=niemand@wc-198.r-195-35-139.atwork.nl) |
| 16:25.41 | mib_3vvtcz9a | stansmith; thanks |
| 16:25.51 | stansmith | my personal fave is "OMG BEWBZ LOL" |
| 16:25.57 | drmessano | ha |
| 16:26.01 | JenniferAkemi- | in my younger days i used to hang out on #sex on efnet :P |
| 16:26.02 | JenniferAkemi- | haha |
| 16:26.05 | pylinuxian | hmmhesays, can i see your zaptel.conf & zapata.conf iles ? |
| 16:26.12 | pylinuxian | files |
| 16:26.23 | drmessano | JenniferAkemi-: Did you go through a lot of keyboards? |
| 16:26.24 | hmmhesays | pylinuxian, um no. Each configuration is different |
| 16:26.27 | mib_3vvtcz9a | I have issues with meetme conference |
| 16:26.28 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:26.28 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:26.32 | x86 | drmessano: i always say legal/yes please/old enough.... seems to scare most people off ;) |
| 16:26.35 | pylinuxian | well, I can get started |
| 16:26.38 | x86 | err |
| 16:26.47 | x86 | drmessano: legal/yes please/anywhere you want it |
| 16:26.49 | hmmhesays | on a dx2030 i'm assuming your using E1 |
| 16:26.52 | pylinuxian | are you interfacing E1 or T1 ? |
| 16:26.55 | drmessano | On IRC, all you need is "Are you pinay?" |
| 16:26.58 | JenniferAkemi- | heh. gotta go run. my year old is coming home early and she hates when i am on the treadmill. |
| 16:27.00 | pylinuxian | yep E1 |
| 16:27.02 | drmessano | A simple yes/no will suffice |
| 16:27.06 | JenniferAkemi- | two year old i mean |
| 16:27.07 | pylinuxian | from equal at france |
| 16:27.11 | x86 | drmessano: pinay? |
| 16:27.14 | pylinuxian | e-qual is the provider |
| 16:27.20 | hmmhesays | pri, or cas? |
| 16:27.24 | pylinuxian | pri |
| 16:27.45 | drmessano | I dont know the etemology, but pinay = from the philippines |
| 16:27.57 | pylinuxian | my card is Digium TE212P |
| 16:28.07 | pylinuxian | 2 ports |
| 16:28.10 | hmmhesays | ok, thats pretty straight forward then |
| 16:28.17 | x86 | drmessano: http://en.wikipedia.org/wiki/Pinay |
| 16:28.19 | hmmhesays | you want asterisk to be the master or the slave? |
| 16:28.26 | drmessano | I've gotten thousands of PMs from group of islands |
| 16:28.26 | x86 | drmessano: it's a female from the phillipines |
| 16:28.30 | drmessano | Ah ok |
| 16:28.32 | x86 | drmessano: aka Filipina |
| 16:28.37 | pylinuxian | slave you mean clock side ? |
| 16:28.39 | x86 | drmessano: wikipedia++ :) |
| 16:28.39 | *** join/#asterisk jbigbee (n=jbigbee@216.207.245.1) |
| 16:28.58 | drmessano | There I go.. Never once bothered to look it up, but I used it lol |
| 16:29.07 | x86 | hehehe |
| 16:29.56 | *** join/#asterisk SteveTotaro (n=Elizabet@c-69-243-124-5.hsd1.md.comcast.net) |
| 16:30.14 | hmm-home | slave/master user/network whatever you want to call it |
| 16:30.32 | pylinuxian | ok then : Master |
| 16:31.19 | *** join/#asterisk arguile (i=user224@KTNRON06-1242488957.sdsl.bell.ca) |
| 16:31.46 | hmm-home | should be pretty straight forward, where are you having your troubles? |
| 16:31.50 | *** join/#asterisk afed (n=rooot@2001:470:1f07:360:211:11ff:fec4:45f1) |
| 16:32.03 | drmessano | ~asteriskcat |
| 16:32.04 | jbot | asteriskcat is probably not amused |
| 16:32.05 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:32.17 | pylinuxian | well, its my first interfacing with DX2030 hardware |
| 16:32.17 | [TK]D-Fender | pylinuxian: what are you connecting to each end of this unit, and why? |
| 16:32.28 | hmm-home | [TK]D-Fender: its a quintum |
| 16:32.33 | [TK]D-Fender | hmmI see that. |
| 16:32.41 | SteveTotaro | quintum rulez |
| 16:32.42 | hmmhesays | I'm having multiple personalities today |
| 16:32.43 | *** part/#asterisk arguile (i=user224@KTNRON06-1242488957.sdsl.bell.ca) |
| 16:32.49 | x86 | quintum? |
| 16:32.51 | [TK]D-Fender | hmm-home: Doesn't say HOW he's trying to use it |
| 16:32.52 | mib_3vvtcz9a | Hi everyone. Does anybody experience oneway audio in conference? |
| 16:32.55 | hmmhesays | I use quintum pretty heavily for ds3 and under installs |
| 16:33.11 | SteveTotaro | i have setup four quintum tenor ax 24 FXS boxes |
| 16:33.15 | pylinuxian | Asterisk <--> Digium TE212P <---> DX2030 |
| 16:33.30 | hmmhesays | SteveTotaro, I've probably done 3-400 quintum installs in my life |
| 16:33.33 | x86 | I'm happy with CAC for channel banks |
| 16:34.09 | [TK]D-Fender | pylinuxian: Since the DX2030 speaks SIP, why go through it in the first place since * can ALREADY talk SIP? |
| 16:34.09 | hmmhesays | I'm going to guess, radius |
| 16:34.09 | grandpapadot | I have a number of Queues (asterisk 1.2.26), they have members. Some members are in more than one queue. Is there a way for the queue to check if a member is on the phone before sending a call? I know it won't send the call if the member pulled a call from the queue, but what about other calls? |
| 16:34.15 | *** part/#asterisk afed (n=rooot@2001:470:1f07:360:211:11ff:fec4:45f1) |
| 16:34.18 | stansmith | does anyone here run Arch linux for use with asterisk? |
| 16:34.20 | pylinuxian | well, I need more advice then i thought then |
| 16:34.23 | [TK]D-Fender | pylinuxian: Thats like speaking english to a translator, having that translator speak french to another translator who'll translate it back to enlish. |
| 16:34.31 | *** join/#asterisk zeeesh (i=zeeesh@203.215.179.43) |
| 16:34.47 | hmmhesays | [TK]D-Fender, radius would be a huge reason, or h.323 translation |
| 16:34.47 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 16:34.47 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:34.52 | hmmhesays | there are many reasons why you would do that |
| 16:35.00 | [TK]D-Fender | hmmhesays: None of which I've heard |
| 16:35.04 | pylinuxian | [TK]D-Fender : I need to do other stuff than just place sip calls |
| 16:35.18 | [TK]D-Fender | pylinuxian: So indeed, what exactly are you trying to accomplish with it? |
| 16:35.37 | hmmhesays | [TK]D-Fender, radius accounting and authentication would be the big one |
| 16:35.45 | lirakis | grandpapadot: if it is not a call from the queue.. it will try and send calls to the agent :( (from my experience) .. id love to figure out how to make this not happen |
| 16:35.51 | hmmhesays | asterisk + radius = suck. |
| 16:36.00 | lirakis | if .. any one has a way to do it.. id be very happy :) |
| 16:36.05 | grandpapadot | lirakis: yea, that's what we're having. hmm.... |
| 16:36.07 | [TK]D-Fender | pylinuxian: So * already can talk SIP, and has a PRI interface of its own. I still don't see what this DX2030 is suppsed to be doing for you. |
| 16:36.13 | pylinuxian | asterisk + other applications <-----> Digium TE212P <----> Quintum DX2030 from provider for long distance calls |
| 16:36.28 | x86 | quintum is a full VoIP switch eh? |
| 16:36.32 | x86 | no need for asterisk? |
| 16:36.35 | hmmhesays | ok its ignore hmmhesays day |
| 16:36.38 | [TK]D-Fender | pylinuxian: How is the DX2030 connected to this provider? |
| 16:36.46 | pylinuxian | via satellite link |
| 16:36.54 | [TK]D-Fender | x86: its a simple SIP/PRI gateway |
| 16:37.06 | hmmhesays | hardly simple |
| 16:37.09 | pylinuxian | PRI in here |
| 16:37.19 | stansmith | hardly simple or simply hard? |
| 16:37.20 | [TK]D-Fender | hmmhesays: You can tell me 100 MORE possible uses that have NOTHING to do with his needs if you want to, but really, whats the point? |
| 16:37.36 | [TK]D-Fender | pylinuxian: So you have a PRI from the telco? |
| 16:37.40 | grandpapadot | Hmm.. maybe tackle this from the phone's side, silence the call waiting ring ... |
| 16:37.43 | mib_3vvtcz9a | TK]D-Fender: can you help me out with some audio issues? |
| 16:37.44 | pylinuxian | yep PRI |
| 16:37.45 | hmmhesays | maybe he's using radius, maybe he's using h.323, maybe he's transcoding g.723 over a satellite link |
| 16:37.59 | [TK]D-Fender | pylinuxian: Then what are you doing as far as SIP is concerned? |
| 16:38.07 | hmmhesays | all of which are good reasons to have that gateway sitting there |
| 16:38.34 | ruied | I need an opinion, I'm choosing my sip phone codec, I'm planning PCMU for the Intranet and g729 transcoding for sip ans isdn trunks... is it a good solution? |
| 16:38.52 | ruied | (several phones) |
| 16:38.57 | pylinuxian | [TK]D-Fender : I don't understand you, you mean i can place call on the DX2030 without using my digium card ? |
| 16:39.23 | [TK]D-Fender | pylinuxian: you can do this DIRECTLY through your Digium card and have no NEED for the DX2030 at all. |
| 16:39.51 | [TK]D-Fender | pylinuxian: Plug your TE212P into your telco-provided PRI and you're DONE. You have no need of the DX2030 in this case |
| 16:41.01 | pylinuxian | well, actually the DX2030 is the actuall PRI interface that i have here .... there is no other PRI interface as far as i can see |
| 16:41.08 | hmmhesays | unless he needs g.723 and or radius |
| 16:42.27 | pylinuxian | Interface: |
| 16:42.28 | pylinuxian | T1/E1 and Fractional T1/E1 with a built in CSU. |
| 16:42.28 | pylinuxian | T1/E1 Signaling: |
| 16:42.28 | pylinuxian | Channel Associated Signaling (CAS) |
| 16:42.28 | pylinuxian | Common Channel Signaling (CCS) |
| 16:44.05 | pylinuxian | so ? |
| 16:44.31 | [TK]D-Fender | pylinuxian: Take the PRI jack out of the DX2030, plug it into the TE212P and you're DONE. |
| 16:45.27 | pylinuxian | [TK]D-Fender : satellite dish <-----> reciever <----> quintum <-----> digium card <---> asterisk |
| 16:45.53 | pylinuxian | i must go thru dx2030 to get to link of service provider |
| 16:45.58 | [TK]D-Fender | pylinuxian: what does this receiver pass to your Quintum? |
| 16:46.32 | pylinuxian | voice |
| 16:46.42 | *** join/#asterisk Corydon76-vcch (i=red@pdpc/supporter/bronze/Corydon76-home) |
| 16:46.42 | *** mode/#asterisk [+o Corydon76-vcch] by ChanServ |
| 16:46.49 | pylinuxian | its a VoIP provider |
| 16:47.07 | pylinuxian | over satellite |
| 16:47.12 | stansmith | thats sick |
| 16:47.18 | SteveTotaro | that is lag city |
| 16:47.25 | stansmith | is it? |
| 16:47.28 | [TK]D-Fender | pylinuxian: So this "receiver" is talking SIP to your quintum? |
| 16:47.42 | pylinuxian | how would i know ? |
| 16:47.48 | *** join/#asterisk Dovid (n=Dovid@bzq-79-177-125-106.red.bezeqint.net) |
| 16:47.55 | pylinuxian | I know only that the quintum is providing E1 |
| 16:48.05 | [TK]D-Fender | pylinuxian: Holy crap, you have NO CLUE what you're even doing NOW. |
| 16:48.27 | [TK]D-Fender | pylinuxian: Go call a tech to figure out what you've got now and why its set up the way it is |
| 16:48.37 | SteveTotaro | send me your quintum gear, i hear it is no good ;) |
| 16:48.40 | drmessano | [11:36] <[TK]D-Fender> pylinuxian: So you have a PRI from the telco? [11:36] <pylinuxian> yep PRI <-- FAIL.. the dx2030 is his "PRI" |
| 16:48.56 | [TK]D-Fender | drmessano: I WANT MY WEEKEND BACK |
| 16:49.04 | drmessano | Sorry dude :( |
| 16:49.23 | Dovid | ever since upgrading to 1.4X I am unable to use MP3's for MOH. What am i missing ? |
| 16:49.42 | [TK]D-Fender | drmessano: we oughtta form some sort of club.... |
| 16:49.42 | [TK]D-Fender | Dovid: asterisk-addons? |
| 16:49.49 | drmessano | [TK]D-Fender: Sometimes "No" means "No", and "Yes" means "I have no idea what I am talking about, applesauce, yes I have a PRI" |
| 16:49.58 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 16:50.08 | [TK]D-Fender | sdalkfkl;asdsf;klasdjfkl;asdklfui9pewrasowekhewkh;mhvwovioasfdfunvasofnb |
| 16:50.22 | stansmith | Dovid: you need mpg123 if you want to play back mp3 moh |
| 16:50.29 | pylinuxian | [TK]D-Fender : what is wrong with my setup ? |
| 16:50.34 | [TK]D-Fender | stansmith: NO. |
| 16:50.40 | Dovid | stansmith: Argh !!!! |
| 16:50.43 | *** join/#asterisk methods[laptop] (n=daquino@69.60.204.9) |
| 16:50.45 | stansmith | wah! |
| 16:50.55 | [TK]D-Fender | pylinuxian: We can't tell, you can't even adequately describe it. |
| 16:51.03 | stansmith | [TK]D-Fender: dont kill the messenger..... http://www.asterisktopics.com/?p=14 |
| 16:51.04 | drmessano | [TK]D-Fender: Don't get frustrated, let the bunny handle it: http://ratonland.org/img/articles/bunny-pancake.gif |
| 16:51.27 | stansmith | someone else send [TK]D-Fender a link |
| 16:51.28 | methods[laptop] | when you run asterisk does it keep recreating the ctl file constantly ? |
| 16:51.29 | [TK]D-Fender | drmessano: Comedy gold! |
| 16:51.51 | drmessano | Hmm |
| 16:52.10 | [TK]D-Fender | stansmith: That guide is BS. |
| 16:52.17 | stansmith | i didnt write it |
| 16:52.21 | pylinuxian | [TK]D-Fender : I ll get back to you in a while |
| 16:52.29 | stansmith | thats what happens when you google |
| 16:52.38 | [TK]D-Fender | stansmith: You only spewed its misinformation to others :) |
| 16:52.51 | methods[laptop] | why would the startup script remove the ctl file ? |
| 16:53.13 | x86 | so these Quintum devices are used basically to terminate T1/E1 lines and do SIP/H323 trunking to the phone system? |
| 16:53.21 | drmessano | ~wut |
| 16:53.22 | jbot | methinks wut is a lamer way of saying what |
| 16:53.26 | drmessano | crap |
| 16:53.29 | stansmith | [TK]D-Fender: im not seeing how that guide is BS, even if that mp3 playback is wrong, the rest of it looks legit |
| 16:53.42 | drmessano | ~uhwut |
| 16:53.43 | pylinuxian | x86 : yes |
| 16:53.54 | *** join/#asterisk joelsolanki (i=joelsola@220.224.109.92) |
| 16:54.04 | x86 | seems kind of cool, actually |
| 16:54.10 | drmessano | We need a #jobot-commits channel |
| 16:54.12 | pylinuxian | x86 : but for me they only terminate E1 lines |
| 16:54.14 | drmessano | We need a #jbot-commits channel |
| 16:54.22 | drmessano | ~uhwut |
| 16:54.44 | pylinuxian | the rest of what they do i don't care |
| 16:54.54 | joelsolanki | Hi |
| 16:55.00 | stansmith | ~hi joelsolanki |
| 16:55.01 | jbot | Many greetings, joelsolanki, most strange traveller, to this IRCdom of plenty. |
| 16:55.08 | drmessano | ~uhwut |
| 16:55.11 | drmessano | Odd |
| 16:55.47 | joelsolanki | is there any proven ss7 implementation available other than sangoma SMG ? i want to use with sangoma A104D card |
| 16:55.59 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 16:56.05 | signius | why dint the thumbs work on that guide tho |
| 16:56.19 | *** join/#asterisk simbol76ss (n=simbol@ip-212-26.sn1.eutelia.it) |
| 16:56.24 | javar | smsq support sip? |
| 16:56.27 | drmessano | Ok, so I have a POTS line from my telco |
| 16:56.39 | drmessano | Ok well, it's kind of a POTS line |
| 16:56.41 | x86 | I want to see an SS7 over IP implementation with asterisk... that'd be hawt |
| 16:56.50 | drmessano | Has anyone ever heard of Vontage? |
| 16:57.19 | joelsolanki | the opensource available implementation is chan_ss7 |
| 16:57.24 | stansmith | voip provider, no? |
| 16:57.36 | [TK]D-Fender | stansmith: plenty of errors in there and hard links to ancient versions |
| 16:57.37 | drmessano | Google is my ISP and I guess Vontage is my AT&T |
| 16:57.42 | simbol76ss | in che senso |
| 16:57.43 | joelsolanki | want to check if any body using chan_ss7 on production server ? |
| 16:57.47 | simbol76ss | voip provider?? |
| 16:57.50 | stansmith | zzzz.. |
| 16:58.26 | drmessano | Wut I want to no is, can I make Google calls on Vontage like Myspace without Firefox??!?!?!??! |
| 16:58.36 | drmessano | Any1? |
| 16:59.04 | defswork | so, how does jabber integration work ? |
| 16:59.27 | drmessano | You can send messages to Jabber clients from within the dialplan |
| 16:59.34 | Dovid | stansmith: I have it installed. |
| 16:59.38 | defswork | that it ? |
| 16:59.50 | defswork | more annoying than useful :) |
| 16:59.59 | drmessano | You can do Gtalk voice as well, but I could care less about that |
| 17:00.12 | *** join/#asterisk kahless_ (n=kahless@i577AC452.versanet.de) |
| 17:00.24 | defswork | I install jabber servers at all my customer sites - was wondering if it might be of use |
| 17:00.27 | kahless_ | hi all |
| 17:00.30 | drmessano | I use it |
| 17:00.34 | stansmith | ~hi kahless_ |
| 17:00.35 | jbot | Many greetings, kahless_, most strange traveller, to this IRCdom of plenty. |
| 17:00.59 | defswork | drmessano: for what kind of messages ? |
| 17:01.03 | joelsolanki | any guys using ss7 in asterisk /? |
| 17:01.04 | davevg-btwtech | not annoying if you want to use the jabber messaging for screen pops |
| 17:01.11 | drmessano | I sent out call notifications and can have it send me server stats when I dial certain extensions |
| 17:01.28 | mib_3vvtcz9a | meetme audio drop audio on asterisk 1.4? What I am doing wrong? |
| 17:01.48 | drmessano | It's kinda neat as an additional notification if doing any voicemail blasting |
| 17:01.57 | defswork | davevg-btwtech: you can just do screen pops ? is that a client specific thing or built into xmpp ? |
| 17:02.13 | *** join/#asterisk DarKnesS_WolF (n=sherif@unaffiliated/sherif) |
| 17:02.26 | drmessano | Openfire does screen pops |
| 17:02.31 | drmessano | Openfire/Spark |
| 17:02.50 | davevg-btwtech | defswork, we use jabber for screen pops, each client has a java app using the smack library |
| 17:02.53 | drmessano | Use it with the Asterisk-IM module |
| 17:02.57 | kahless_ | is there any asterisk (or out of the box thigy) which supports isdn configuartion via gui? |
| 17:03.08 | defswork | Are they just status notifications ? |
| 17:03.16 | drmessano | Call notifications |
| 17:03.23 | drmessano | Status notifications |
| 17:03.26 | drmessano | Integrated dialing |
| 17:03.43 | drmessano | Right click and call another extension |
| 17:03.51 | drmessano | Popup to dial any number |
| 17:04.14 | defswork | I might have a play then - would be useful for a popup showing each incoming call at least |
| 17:04.26 | *** join/#asterisk simbol76ss (n=simbol@ip-212-26.sn1.eutelia.it) |
| 17:04.34 | drmessano | Are you using ejabberd? |
| 17:04.43 | Dovid | stansmith: any other ideas ? |
| 17:04.51 | defswork | no - I've got jabberd and jabberd2 |
| 17:04.56 | drmessano | oh |
| 17:05.05 | drmessano | Openfire is the way to go with Asterisk |
| 17:05.27 | defswork | I put a jabber server on all my office installs so customers can msg each other and also msg me |
| 17:05.30 | stansmith | Dovid: how do your modules.conf and musiconhold.conf files look? |
| 17:05.44 | defswork | got some scripts I wrote to prepopulate and repopulate rosters |
| 17:05.56 | defswork | much easier with jabberd2-mysql |
| 17:07.05 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
| 17:07.12 | defswork | I don't fancy openfire |
| 17:07.20 | defswork | java :o |
| 17:07.28 | stansmith | java = tits! |
| 17:08.49 | webar7 | sigh |
| 17:08.54 | webar7 | ejabberd is nice |
| 17:09.28 | webar7 | whey do I keep getting this message? |
| 17:09.31 | webar7 | chan_sip.c:15051 handle_request_register: Registration from '222 <sip:222@192.168.2.4>' failed for '192.168.2.8' - Not a local domain |
| 17:09.47 | webar7 | "Not a local domain" is driving me nuts |
| 17:09.56 | drmessano | defswork, you mean like adding new users to everyones rosters? |
| 17:10.01 | webar7 | is it a DNS SRV thing? |
| 17:10.03 | defswork | drmessano: yes |
| 17:10.06 | drmessano | Oh god |
| 17:10.14 | drmessano | Openfire does all this from the GUI.. two clicks |
| 17:10.21 | drmessano | Web GUI |
| 17:10.31 | defswork | so everyone sees everyone in the same company, departmentalised etc.. |
| 17:10.34 | webar7 | ejabberd has a web gui too |
| 17:10.51 | defswork | you kids and your web guis |
| 17:10.51 | drmessano | Yes, but it doesnt have Asterisk support |
| 17:10.53 | drmessano | Move along |
| 17:11.29 | drmessano | My point, defswork, was that it has really tight asterisk integration, and the support for groups is outstanding |
| 17:11.45 | defswork | yeah - but it's java :) |
| 17:11.55 | drmessano | ...which means it's better? |
| 17:12.14 | drmessano | Could be worse.. WTF is an ERLANG? |
| 17:12.25 | defswork | heh true :) |
| 17:12.37 | webar7 | java is open source :-) sun will make it available real soon now |
| 17:12.40 | drmessano | I thought an ERLANG was 2 kilometers across a pasture in dublin |
| 17:13.22 | drmessano | I've been using Openfire for over a year now.. it's damn solid |
| 17:13.31 | drmessano | Even Spark is 10x better than it was even 8 months ago |
| 17:13.41 | drmessano | The asterisk stuff pwns |
| 17:13.43 | defswork | I've been using jabberd for at least 5 |
| 17:13.52 | drmessano | lol |
| 17:14.15 | drmessano | I was playing with XMPP back in the late 90s :P |
| 17:14.39 | stansmith | sun jdk crushes gcj in terms of performance |
| 17:15.02 | robl^ | I've been using Ms. PacMan since the 80s! |
| 17:15.12 | defswork | can't remember when I first started using it - I was working in Edinburgh at the time |
| 17:15.21 | drmessano | I got PacMan back in 79 |
| 17:15.24 | defswork | robl^: I heard she was using you |
| 17:15.41 | stansmith | ooo! |
| 17:15.44 | stansmith | burn |
| 17:15.46 | defswork | she's still seeing Mr Pacman on the side |
| 17:15.48 | drmessano | Pacman was the worst first person shooter ever |
| 17:16.06 | robl^ | defswork: shhh. let me keep up with my fantasy of being the user. its less painful and traumatic that way |
| 17:16.21 | defswork | robl^: don't aspire to be a user |
| 17:16.27 | defswork | Admins rule! |
| 17:16.51 | *** join/#asterisk digime (n=digime@99.145.104.206) |
| 17:17.20 | digime | Looking for a toll free DID provider in USA, any suggestions? |
| 17:18.16 | defswork | anything interesting happening on the adhearsion front ? |
| 17:18.53 | defswork | I'm doing some rails work at the moment - wouldn't mind playing with some adhearsion asterisk stuff |
| 17:19.13 | *** part/#asterisk joelsolanki (i=joelsola@220.224.109.92) |
| 17:20.25 | kahless_ | is there any asterisk (or out of the box thigy) version which supports isdn configuartion via gui? |
| 17:20.37 | defswork | kahless_: get away |
| 17:20.47 | defswork | kahless_: whats the point of that - you only do it once |
| 17:21.34 | kahless_ | yeah, but i dont spam, i only ask twice |
| 17:21.48 | defswork | ? |
| 17:21.56 | defswork | you only setup ISDN once - then it's done |
| 17:22.07 | defswork | what would the point of a web gui for it be |
| 17:22.19 | stansmith | kahless_: respond |
| 17:22.23 | kahless_ | .. oops (sorry im too tired, totally misunderstood you) |
| 17:22.40 | defswork | the sangoma install autocreates the 3 files needed |
| 17:23.08 | defswork | then you just need to spend 3 hours hunting through the mostly wrong information about configuring it on the internet |
| 17:23.13 | defswork | :) |
| 17:23.31 | *** part/#asterisk [T]ank (n=ckwall@c-71-195-199-107.hsd1.ut.comcast.net) |
| 17:23.44 | kahless_ | if isdn is working, then i can configure the other stuff with the gui? |
| 17:23.52 | defswork | other stuff ? |
| 17:24.12 | defswork | if you mean dial plans and stuff - freepbx |
| 17:24.25 | kahless_ | defswork: like voicebox |
| 17:25.11 | kahless_ | defswork: not only voicebox, voip too |
| 17:25.16 | defswork | freepbx |
| 17:25.30 | robl^ | most asterisk configuration GUIs are limit much of the flexibility in asterisk -- restricting you to doing things a certain way.. plus.. they make a mess out of the configuration files |
| 17:25.54 | defswork | robl^: freepbx is pretty good imho |
| 17:26.06 | Migrane | does asterisk support caller id with name on a PRI line? |
| 17:26.17 | Migrane | I'm getting number now, no name. |
| 17:26.26 | defswork | Migrane: your telco provides the name ? |
| 17:26.29 | Qwell | Migrane: nothing does. you need to call your telco to set it. |
| 17:26.51 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 17:27.00 | Migrane | so if they set it on their side, I won't have to do anything on mine? |
| 17:27.12 | Qwell | correct. if they allow setting it. |
| 17:27.18 | Qwell | and I assume you mean sending? |
| 17:27.26 | defswork | heh - I assumed receiving |
| 17:27.32 | Migrane | nono, Im not getting name on my incoming calls. |
| 17:27.41 | Qwell | oh, well tell your telco you want name |
| 17:27.53 | defswork | Migrane: name of who ? |
| 17:27.54 | robl^ | defswork: its not BAD, but it still restrictive and tends to create overly complex config files that are difficult to debug.. but if somoene is willing to work within those limits, then go ahead with freepbx --at least that is my impression of it |
| 17:28.00 | stansmith | CID info is sent through the copper, correct? |
| 17:28.00 | Migrane | the name of the incoming caller. |
| 17:28.10 | defswork | Migrane: how would your telco know that ? |
| 17:28.23 | Qwell | they would look it up in the big callerid database |
| 17:28.26 | Migrane | caller id with name? |
| 17:28.28 | javar | <PROTECTED> |
| 17:28.34 | twisted | poof |
| 17:28.55 | robl^ | hey twisted one |
| 17:29.01 | defswork | Migrane: I lookup the CLI against a phonebook to get the name |
| 17:29.33 | twisted | Migrane: you have to ask your PRI vendor for name |
| 17:29.42 | Qwell | personally, I just memorize the phone book, so when I get an incoming call, I know who it's from. |
| 17:29.44 | defswork | Qwell: there's such a thing ? |
| 17:29.52 | Qwell | defswork: how do you think phonebooks are made? |
| 17:30.09 | twisted | Qwell: magic! |
| 17:30.22 | defswork | Qwell: I'm ex-directory |
| 17:30.38 | Migrane | defs: Doesnt it work just the same as caller-id with name on pots lines? |
| 17:30.48 | robl^ | I know this one! I saw it on "How Things Are Made".. Starts with a tree that gets turned into paper |
| 17:30.59 | defswork | Migrane: we don't have such a thing in the uk |
| 17:31.10 | javar | somebody knows if smsq works over SIP? |
| 17:31.28 | Migrane | defs: That explains it.. You guys are still driving on the left side of the road as well... |
| 17:31.42 | defswork | Migrane: we're waiting for you to join us |
| 17:32.10 | Migrane | Migrane: I actually do drive on the left, friday when I'm loaded. |
| 17:32.19 | robl^ | Migrane: shh.. they're still upset about the whole tea into the ocean business. |
| 17:32.34 | twisted | i invented the phone. |
| 17:32.36 | defswork | robl^: yeah that gutted us that did |
| 17:32.39 | stansmith | my uncles friend did |
| 17:32.44 | SteveTotaro | it was a harbor not an ocean |
| 17:32.48 | defswork | so much dunking opportunity wasted |
| 17:32.50 | Migrane | I invented the internet. |
| 17:32.58 | twisted | Migrane: you're Al Gore? |
| 17:33.01 | twisted | sweet! |
| 17:33.08 | stansmith | i used to kick it with denzel washington |
| 17:33.14 | robl^ | SteveTotaro: true.. But I like to exagerate a bit ;-) |
| 17:33.16 | defswork | can you kick it ? |
| 17:33.20 | twisted | s/denzel/george |
| 17:33.24 | stansmith | lol |
| 17:33.40 | *** join/#asterisk lonebobwhite (n=rleblanc@74.231.171.198) |
| 17:33.45 | lonebobwhite | sterisk |
| 17:33.56 | stansmith | lonebobwhite: o yea? |
| 17:33.56 | twisted | lonebobwhite : a- |
| 17:34.52 | *** join/#asterisk angryuser (i=nononon@df01t2-213-44-81-225.d4.club-internet.fr) |
| 17:36.16 | *** join/#asterisk Peaceful (n=peaceful@70.102.57.178) |
| 17:36.28 | stansmith | cha-ching! |
| 17:36.48 | Peaceful | My polycom 550 won't dial 3-digit numbers--weird! |
| 17:36.54 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 17:37.10 | Peaceful | I just tried upgrading to the 2.2.2 polycom firmware, and it STILL won't dial 3-digit numbers! |
| 17:37.17 | Peaceful | Won't even send the request to asterisk |
| 17:37.25 | Peaceful | Has anyone ever encountered that before? |
| 17:37.26 | Qwell | Peaceful: fix your phones dialplan |
| 17:37.46 | Peaceful | Qwell, I just reset it to the default digitmap for 2.2.2, if that's what you mean |
| 17:37.56 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 17:38.03 | Qwell | does the default digitmap know about 3 digit numbers? |
| 17:38.09 | Qwell | (the answer may surprise you) |
| 17:38.15 | Peaceful | dialplan.digitmap="[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT" |
| 17:38.25 | Peaceful | The first one ought to match 911, at least |
| 17:38.34 | Peaceful | and 911 gives me the same error |
| 17:38.51 | Peaceful | I tried adding 3xx, for my 300-level extensions, and that didn't change anything |
| 17:39.07 | Peaceful | (rebooted the phone and verified the config on the phone, of course) |
| 17:39.33 | drmessano | Whatever happened to letting Asterisk manage it.. |
| 17:39.49 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:39.51 | Peaceful | manage...what? |
| 17:40.22 | drmessano | More problems are caused by stupid phone dialplans than anything else |
| 17:40.46 | drmessano | You make it generic as hell, and asterisk will tell you if you did something wrong |
| 17:40.54 | Peaceful | drmessano, regardless of the dialplan, the phone won't let you send a 3-digit number at all. Not even pressing the "send" soft-button. |
| 17:41.15 | drmessano | [12:37] <Qwell> does the default digitmap know about 3 digit numbers? <--- O.o |
| 17:41.23 | [TK]D-Fender | Peaceful: show us your dialplan |
| 17:41.55 | drmessano | Everyones diaplan is perfect until .. it's not |
| 17:41.56 | Peaceful | The dialplan itself actually works; if I add "3xx" as a pattern, pressing "301" will make the "send" soft button disappear, but the phone says "Enter more digits" and doesn't send anything to asterisk. |
| 17:42.04 | Peaceful | [TK]D-Fender, I DID ^^ |
| 17:42.21 | [TK]D-Fender | Peaceful: that does not account for 3XX |
| 17:42.21 | Peaceful | here it is again: dialplan.digitmap="[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT" |
| 17:42.33 | Peaceful | you're right, like I said, I set it back to default |
| 17:42.45 | [TK]D-Fender | Peaceful: And you're wondering why it doesn't work? |
| 17:43.02 | [TK]D-Fender | Peaceful: > insert "duh" here < |
| 17:43.07 | Peaceful | the problem's not the auto-dialing, it's that the phone refuses to send 3-digit extensions at all, whether auto-dialed due to the digitmap or manually entered with the "send" soft button |
| 17:43.07 | drmessano | Yeah |
| 17:43.22 | Peaceful | The phone displays "Enter more digits" |
| 17:43.33 | Peaceful | I'm using 911 to test right now |
| 17:43.40 | Peaceful | or 211, or 311, or 411, etc. |
| 17:43.50 | angryuser | souns like a stupid problem |
| 17:43.55 | angryuser | sounds* |
| 17:43.56 | *** join/#asterisk LiNeTuX (n=LiNeTuX@98.205.205.68.cfl.res.rr.com) |
| 17:43.56 | drmessano | It is |
| 17:44.05 | [TK]D-Fender | Peaceful: sounds like you should be staring at sip debug while you're at it. |
| 17:44.32 | drmessano | [*x.|x.] |
| 17:44.40 | *** join/#asterisk zobia (n=laurashr@222.212.72.170) |
| 17:44.41 | Peaceful | [TK]D-Fender, that's worth a try. Maybe it is actually sending "something" to asterisk... |
| 17:44.52 | zobia | hello i am using 1.4.18 |
| 17:44.55 | Peaceful | though the console shows nothing with debug and verbose to 99 |
| 17:45.05 | stansmith | zobia: im using 1.4.17 |
| 17:45.07 | zobia | but i keep get RTCP Read too short when i using sip trunk. anyone knows why this happened? |
| 17:45.14 | drmessano | ZOMG |
| 17:45.18 | drmessano | I am on 1.4.18 too! |
| 17:45.24 | drmessano | LETS BE FRUNDS! |
| 17:45.27 | stansmith | LOL! |
| 17:45.29 | LiNeTuX | heh |
| 17:45.34 | stansmith | drmessano: add him to your top 8 |
| 17:45.42 | Qwell | top 8 is so 2006 |
| 17:45.50 | drmessano | Top24, bitches |
| 17:45.53 | stansmith | lol |
| 17:45.57 | zobia | stansmith:i was using 1.4.17 2 days ago. just upgrade to this one |
| 17:46.03 | Peaceful | [TK]D-Fender, so I turned on sip debug for the IP of the polycom phone, entered "911", got NO output, and the phone displays "Enter more digits" |
| 17:46.06 | robl^ | not MyFaves? |
| 17:46.06 | drmessano | zobia: You rock |
| 17:46.16 | stansmith | zobia: ive been developing on top of 1.4.17, to risky to upgrade despite newer versions |
| 17:46.19 | drmessano | I TOTALLY JUST upgraded TOO |
| 17:46.41 | drmessano | I was like, 1.4.17 is so teh kewl |
| 17:46.43 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 17:46.43 | *** mode/#asterisk [+o anthm] by ChanServ |
| 17:46.50 | LiNeTuX | drmessano: you forgot to say "LIKE" |
| 17:46.51 | drmessano | and liek, russellb was like, "zomg, 1.4.18" |
| 17:46.57 | drmessano | and I was all like "ZOMG, NO WAI" |
| 17:47.01 | drmessano | and he was liek "Yah" |
| 17:47.05 | zobia | ok. so any one knows why i get RTCP Read too short using sip trunk for 1.4 version |
| 17:47.07 | zobia | ? |
| 17:47.07 | drmessano | and I was like "ZUPGRADED!" |
| 17:47.35 | stansmith | zobia: are you trying to execute Read() in the dialplan? |
| 17:47.41 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:47.46 | drmessano | Oh, he actually has a problem |
| 17:47.50 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:47.50 | clickonce | The default voice files, invalid, etc where are they located? I can't find them anywhere on my disk. |
| 17:47.54 | stansmith | lol |
| 17:47.57 | zobia | stansmith: no i don't use read() |
| 17:47.59 | drmessano | ^5 |
| 17:48.26 | zobia | stansmith: i think it happened when it using voicemail() |
| 17:48.36 | russellb | drmessano: ORLY?! |
| 17:48.44 | stansmith | ~pb zobia |
| 17:48.51 | angryuser | i wonder whne 1.6 will be released, or at least RC1 |
| 17:48.57 | stansmith | ~pb | zobia |
| 17:49.03 | Peaceful | Soooo...polycom's just don't handle 3-digit numbers? Must make it dangerous for emergencies... |
| 17:49.08 | stansmith | zobia: pastebin the error |
| 17:49.13 | zobia | ok |
| 17:49.52 | Peaceful | [TK]D-Fender, I messed up. I mistyped the sip debug command. |
| 17:49.57 | [TK]D-Fender | Peaceful: You've messed something up... |
| 17:50.00 | Peaceful | [TK]D-Fender, now there IS some output |
| 17:50.01 | zobia | [Feb 21 17:41:21] WARNING[26795]: rtp.c:891 ast_rtcp_read: RTCP Read too short |
| 17:50.02 | x86 | Peaceful: polycom will support anything you want |
| 17:50.11 | zobia | stansmith: [Feb 21 17:41:21] WARNING[26795]: rtp.c:891 ast_rtcp_read: RTCP Read too short |
| 17:50.16 | *** part/#asterisk simbol76ss (n=simbol@ip-212-26.sn1.eutelia.it) |
| 17:50.17 | stansmith | ~pb |
| 17:50.17 | 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 |
| 17:50.19 | Peaceful | ok, so let's see what this sip debug output means... |
| 17:50.30 | x86 | Peaceful: play with the dialplan in the phone |
| 17:51.19 | Peaceful | x86, [TK]D-Fender: Here's my sip debug that I'm looking through: http://pastebin.com/m5c1d24fb |
| 17:51.45 | *** topic/#asterisk by russellb -> Asterisk: The Open Source Telephony Application Platform (asterisk.org) -=- Asterisk, -Addons, Libpri 1.6.0-beta4 (2008/02/21), Asterisk 1.4.18 (2008/02/07), *-Addons 1.4.5 (2007/12/1), Zaptel 1.4.9 (2008/02/18), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox |
| 17:51.51 | zobia | stansmith:http://pastebin.ca/912841 |
| 17:51.54 | drmessano | [*x.|x.] |
| 17:51.56 | drmessano | YAY |
| 17:52.14 | *** topic/#asterisk by russellb -> Asterisk: The Open Source Telephony Application Platform (asterisk.org) -=- Asterisk 1.6.0-beta4 (2008/02/21), Asterisk 1.4.18 (2008/02/07), *-Addons 1.4.6 (2008/02/21), Zaptel 1.4.9 (2008/02/18), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox |
| 17:52.32 | *** topic/#asterisk by russellb -> Asterisk: The Open Source Telephony Application Platform (asterisk.org) -=- Asterisk 1.6.0-beta4 (2008/02/21), Asterisk 1.4.18 (2008/02/07), *-Addons 1.4.6, 1.6.0-beta2 (2008/02/21), Zaptel 1.4.9 (2008/02/18), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox |
| 17:52.51 | LiNeTuX | Question: is there a reason to include a dialrule on a trunk if the only route using that trunk already has a dial pattern associated with it? |
| 17:53.06 | zobia | stansmith: i am not sure it's becoz of voicemail(). it just show on CLI so frequently. |
| 17:53.37 | stansmith | zobia: you got to post more then that |
| 17:54.02 | zobia | stansmith: each line are the same like that except timestamp |
| 17:54.28 | drmessano | beta3? |
| 17:54.40 | russellb | what about beta3 |
| 17:54.47 | drmessano | -Addons 1.4.6, 1.6.0-beta2 (2008/02/21), |
| 17:54.49 | russellb | beta3 had some silly codec/format bugs |
| 17:54.56 | drmessano | ok |
| 17:55.07 | russellb | nah, -addons is only at beta2 |
| 17:55.12 | drmessano | OH |
| 17:55.16 | drmessano | Im dumb |
| 17:55.20 | russellb | yes, you are. |
| 17:55.23 | russellb | j/k! <3 |
| 17:55.29 | drmessano | LOL |
| 17:55.34 | drmessano | I was misreading the commas |
| 17:55.45 | drmessano | Addons 1.4.6 and 1.6.0beta2 |
| 17:55.49 | drmessano | :( |
| 17:55.50 | russellb | mhm |
| 17:55.51 | drmessano | SAD FACE |
| 17:56.04 | drmessano | I was thinking 1.6.0 went back a beta |
| 17:56.12 | russellb | that would be amusing :) |
| 17:56.20 | drmessano | "Screw it, going back to Beta 2..... .1" |
| 17:56.39 | russellb | :-D |
| 17:56.40 | drmessano | A beta x.1 release would be hilarious |
| 17:56.44 | drmessano | "OH, COME ON!" |
| 17:57.01 | russellb | lol |
| 17:57.07 | russellb | beta5.3 |
| 17:57.14 | russellb | beta5.3.1 |
| 17:57.22 | russellb | beta5.3.1-patch3 |
| 17:57.24 | clickonce | What format would you recommend for "record"? I.e. for recording voicemail messages. |
| 17:57.26 | zobia | stansmith: any idea? |
| 17:57.27 | drmessano | No worse than Wine 0.97.516.12.11alpha6 |
| 17:57.30 | stansmith | http://forums.digium.com/viewtopic.php?t=13114&highlight=&sid=949335ae5d5eddc10771bea0d7443599 |
| 17:57.38 | stansmith | not sure, but that might help you pin point it a little more |
| 17:57.44 | drmessano | Wine "screw 1.0, we're having fun with the Betas" |
| 17:58.03 | zobia | stansmith: thank you . let me read |
| 17:58.12 | Qwell | drmessano: I think they broke 0.100 |
| 17:58.18 | drmessano | HA |
| 17:58.24 | JayTee52 | we could be more like Microsoft and call all our alpha code versions beta and our beta code versions RTM..........on second thought let's not. |
| 17:58.25 | drmessano | No way.. now thats funny |
| 17:58.28 | Qwell | wait, no, they cheated |
| 17:58.30 | Qwell | 0.9.55 |
| 17:58.35 | russellb | that's not even numerically valid ... |
| 17:58.36 | russellb | oh well |
| 17:58.38 | drmessano | I always love 0.9 releases that go to 0.10 |
| 17:58.48 | russellb | i guess we do that too |
| 17:58.49 | russellb | oh well |
| 17:58.54 | russellb | 1.4.9, 1.4.10 |
| 17:58.56 | russellb | :) |
| 17:58.59 | stansmith | 0.10 > 0.9, duh |
| 17:59.02 | drmessano | Sure.. but 0.x is the kicker |
| 17:59.08 | Wayhigh | I vote for you being like Oracle and only fixing known bugs 6 months later.. |
| 17:59.12 | Peaceful | So the closest thing that I can find to an explanation to the 3-digit not dialing is: "SIP/2.0 484 Address Incomplete" on line 264 of http://pastebin.com/m5c1d24fb |
| 17:59.14 | drmessano | 0.99 >>>> 0.100 |
| 17:59.17 | clickonce | stansmith: 0.10 < 0.9 :) |
| 17:59.29 | stansmith | ok, i was joking |
| 17:59.31 | drmessano | That's the easy way to never hit 1.0 |
| 17:59.31 | clickonce | stansmith: if it had been 0.09 it would have been correct |
| 17:59.39 | stansmith | clickonce: i know lol |
| 17:59.47 | russellb | better use 0.00009, and leave some room |
| 18:00.28 | drmessano | I can deal with the Asterisk release system.. At least 1.4.18 is more honest than 3 full version releases a year because the numbers ran out |
| 18:00.33 | drmessano | Asterisk 11.0 coming in may! |
| 18:01.01 | russellb | :) |
| 18:01.11 | outtolunc | Asterisk 3000 |
| 18:01.15 | russellb | asterisk 1.6 <3 |
| 18:01.16 | outtolunc | GT |
| 18:01.19 | russellb | 1.6 is the hotness |
| 18:01.32 | russellb | that is all |
| 18:01.38 | drmessano | I'm gonna load 1.6 later on |
| 18:01.49 | drmessano | I'm not scared anymore |
| 18:02.04 | russellb | :) |
| 18:02.09 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 18:02.14 | russellb | drmessano: let me know how it goes |
| 18:02.25 | russellb | don't forget to read UPGRADE.txt, though |
| 18:02.26 | drmessano | You know I will |
| 18:02.33 | russellb | and fix your dialplan ... s/|/,/g |
| 18:02.36 | drmessano | You'll hear ALLLLL about it |
| 18:03.06 | *** join/#asterisk mmmToop (n=michaelt@dsl-243-239-90.telkomadsl.co.za) |
| 18:03.06 | drmessano | I just got a ream of paper to print upgrade.txt |
| 18:03.23 | russellb | lol |
| 18:04.37 | drmessano | I was reading through chan_sip.c the other day and noticed how unfunny the comments are |
| 18:04.42 | russellb | haha |
| 18:04.45 | drmessano | You need more comment drama |
| 18:04.50 | russellb | yeah, we don't have enough humor in our code |
| 18:05.15 | drmessano | Fixed this crap that russell threw in, guess he was drunk at the time |
| 18:06.15 | generalhan | hey all, its been a long time since i messed with my TE210, can some one take a look at my zaptel/zapata config sections to make sure that i have this properly setup, please ! http://pastebin.com/d7a97f452 |
| 18:06.43 | generalhan | rather the TDM40B is what i need to be sure is setup correctly. the TE is working just fine |
| 18:06.46 | clickonce | Is it possible to send a beep to the client? I.e. record message after the beep. |
| 18:06.57 | drmessano | So anyway |
| 18:07.10 | generalhan | clickonce: there is a beep file in the sounds folder |
| 18:07.28 | drmessano | I guess I need to generate some patches for more funny |
| 18:07.47 | [TK]D-Fender | Peaceful: pastebin your dialplan |
| 18:07.51 | russellb | drmessano: go for it |
| 18:08.00 | [TK]D-Fender | Peaceful: from ASTERISK |
| 18:08.04 | russellb | we need some silly CLI commands ... |
| 18:08.17 | Daviey | !halt |
| 18:08.46 | stansmith | is there some kind of repository of sound clips to use with asterisk? if so, i would like to contribute one |
| 18:08.56 | hmmhesays | finally I got my skype trunk stable on this machien |
| 18:08.58 | hmmhesays | *machine |
| 18:09.08 | Daviey | russellb: !sl would be quite silly (if installed) |
| 18:09.09 | drmessano | heh |
| 18:09.20 | drmessano | core show asl |
| 18:09.40 | Daviey | hmmhesays: using which channel? |
| 18:09.44 | hmmhesays | chanskype |
| 18:10.04 | drmessano | Using that wacky skype thing that james posted? |
| 18:10.05 | generalhan | the best one EVER is the "... has been brutally murdered and mutalated by the teletubbies" ! lol |
| 18:10.49 | drmessano | I tried a few variations of skype interfacing just so I could say I have it set up,and all of them sucked |
| 18:10.57 | Daviey | hmmhesays: i don't think reproducing chanskype would be too hard - i refuse to pay :) |
| 18:11.05 | drmessano | One sucked so bad, I lost my spare car keys in the vortex |
| 18:11.19 | Daviey | It just uses the API and xvnc |
| 18:11.27 | hmmhesays | Daviey: probably not, they're just piping audio through different fifo's |
| 18:11.39 | mort_gib | hmmhesays: have a look at voipstunt |
| 18:11.47 | *** join/#asterisk activo (n=haryv@S010600146cf497f9.vs.shawcable.net) |
| 18:11.51 | zamba | suggestions for sip softphones under linux? |
| 18:12.03 | activo | xlite |
| 18:12.15 | Daviey | ekiga |
| 18:12.15 | hmmhesays | why? |
| 18:12.25 | zamba | ekiga does strange things |
| 18:12.34 | Daviey | gizmo |
| 18:12.37 | zamba | like disconnecting a call after exactly 30 seconds every time |
| 18:12.41 | mort_gib | They are SIP native, and are free for loads of countries |
| 18:12.45 | drmessano | Xlite |
| 18:12.50 | drmessano | Xlite is really solid |
| 18:12.52 | mort_gib | So no need for chan_skype ;-) |
| 18:12.56 | activo | very solid |
| 18:13.06 | zamba | gizmo can register with a sip proxy? |
| 18:13.07 | Daviey | mort_gib: and it peers with skype? |
| 18:13.09 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 18:13.16 | *** join/#asterisk af_ (n=getsmart@88-149-230-204.dynamic.ngi.it) |
| 18:13.23 | drmessano | It supports Speex now, so you can use something other than ULAW/ALAW that's NOT GSM |
| 18:13.26 | mort_gib | Not as far as I know |
| 18:13.28 | drmessano | Which is VERY cool |
| 18:13.30 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 18:13.35 | drmessano | Screw skype |
| 18:13.38 | drmessano | This isnt #skype |
| 18:13.43 | mort_gib | Yes, skrew Skype |
| 18:13.47 | drmessano | ~skype |
| 18:13.48 | jbot | [~skype] Skype is a free VoIP software and service using a closed client and propritary protocol. Only commercial channel drivers exist, all of which are poor implementations of convoluted hacks. In general, forget about using Skype with Asterisk... |
| 18:13.52 | drmessano | Skype uses ONE client |
| 18:13.54 | drmessano | Skype |
| 18:14.00 | drmessano | Its proprietary |
| 18:14.05 | hmmhesays | I just have to figure out how to change my sound buffer size |
| 18:14.11 | drmessano | and every effort to make it work with Asterisk as an UGLY HACK |
| 18:14.15 | drmessano | is* |
| 18:14.16 | Daviey | and interfacing it with * is OT |
| 18:14.27 | zamba | Daviey: do you have to sign up with gizmo to use it? |
| 18:14.28 | Daviey | is not Off topic |
| 18:14.35 | Daviey | zamba: not afaik |
| 18:14.36 | activo | come to taling about xlite wonder which laptop has the best built in mic/speakers for using it or somw how could support a blue tooth wireless headset. |
| 18:14.39 | Peaceful | [TK]D-Fender, You're right. 911 simply wasn't on the dialplan. Wacky! And the parked calls (my 300's) are simply not functioning. Probably has to do with my trying to upgrade to 1.4 and then reverting to 1.2 and not reverting some config setting. |
| 18:14.44 | drmessano | No, but asking if Xlite works with Skype is OT |
| 18:15.01 | Peaceful | drmessano, [TK]D-Fender, x86: Thanks for all the help! |
| 18:15.08 | Daviey | zamba: actually mentions * on the sign on screen :) |
| 18:15.14 | [TK]D-Fender | Peaceful: Go pummel yourself with a halibut |
| 18:15.22 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 18:15.27 | yangvnc | I experience the following issue with ISDN, the incoming call comes in properly - by the outgoing call when I call number 041710598 I see on my VOIP phone 503710598 - http://openpaste.org/en/5221/ |
| 18:15.36 | activo | TK give me the Halibut :) |
| 18:15.52 | drmessano | You can not only set up a SIP extension to Asterisk in the Gizmo client, but you can use Gizmo PEERS in Asterisk |
| 18:16.14 | drmessano | "Not the dialplan" |
| 18:16.14 | drmessano | It never is |
| 18:16.28 | drmessano | [x] |
| 18:16.48 | drmessano | [x.] rather |
| 18:16.57 | clickonce | generalhan: thanks! |
| 18:17.23 | zobia | stansmith: i read that link. if it's gsm file problem. the only gsm file i have and using is the voicemailmain fucntion's gsm file. do you know how to make this kind of file 33 multiple size? |
| 18:17.26 | activo | Its been a while since I have been here. Doing remote trouble calls and deployments can be tiring :) put in 1,000 kms yesterday. Never been to the Kootnies of BC before |
| 18:17.43 | *** join/#asterisk draygon-w (n=draygon@216.52.176.254) |
| 18:17.45 | *** join/#asterisk [T]ank (n=ckwall@c-71-195-199-107.hsd1.ut.comcast.net) |
| 18:17.55 | stansmith | zobia: no i do not, sorry |
| 18:18.00 | [T]ank | anyone using dell poweredge 1550 or similar for their servers? |
| 18:18.20 | stansmith | [T]ank: hp ml350..but are you having some kind of issue with that server? |
| 18:18.31 | activo | tank, trying to recall is that a 2u server? |
| 18:18.44 | zobia | stansmith: no roblem. let me figure it out |
| 18:18.54 | zobia | stansmith:thank you |
| 18:18.59 | stansmith | np |
| 18:19.16 | [T]ank | stansmith: no... looking for assistance on configuring wake on lan. online resources not getting me far, and I know many of you all are using dell poweredge. |
| 18:19.19 | [T]ank | its a 1u |
| 18:19.22 | *** join/#asterisk atik7 (n=chatzill@122.53.193.241) |
| 18:19.54 | activo | tank, ask Hc he has one at one our clients locations. |
| 18:20.03 | stansmith | [T]ank: ahh, im trying to deploy this system on a HP server, and the proliant (what HP calls its drivers n such) seem to be interfering with things |
| 18:20.07 | activo | Looks like he left |
| 18:20.16 | *** join/#asterisk Leiste (n=m@dialbs-213-023-181-002.static.arcor-ip.net) |
| 18:20.22 | stansmith | just wondering if you were having some of the same issues as me |
| 18:20.46 | activo | with the voice quality? |
| 18:20.49 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 18:20.49 | *** mode/#asterisk [+o anthm] by ChanServ |
| 18:20.58 | stansmith | activo: me? |
| 18:21.10 | atik7 | Hi, is this right place for asking asterisk user question? |
| 18:21.18 | [T]ank | stansmith: what issues are you having. I use the dl3XX series servers and have had some headaches that I have figured out |
| 18:22.10 | Leiste | ...Hey guys, I'm trying to get the BLF running with ring state (Polycom 650, Asterisk 1.4.18 or Astrisk 1.2) has anybody some expieriences with that? |
| 18:22.47 | [TK]D-Fender | Leiste: Polycom reports "ringing" as "in use". Thats a notification error on their part |
| 18:23.11 | [TK]D-Fender | Leiste: Aastra's presence supports reports it properly |
| 18:23.18 | stansmith | [T]ank: im using the app_swift module, and it seems to be bogging the system down |
| 18:23.30 | stansmith | but i have the exact same stuff running on old hardware (non HP) and it works perfect |
| 18:23.39 | *** join/#asterisk JenniferAkemi (n=akemi@206-248-164-21.dsl.teksavvy.com) |
| 18:23.43 | drmessano | asl? |
| 18:23.44 | stansmith | i reinstalled without the HP drivers and it seems to be working better |
| 18:23.58 | stansmith | drmessano: 20/m/ohio, why do u ask? |
| 18:23.59 | [T]ank | yeah... hp has a way of doing that. sorry, i dont have an answer for you there. |
| 18:24.02 | drmessano | lol |
| 18:24.09 | JenniferAkemi | i need a new name. |
| 18:24.17 | Leiste | [TK]D-Fender: So you mean we are not able to make that work? |
| 18:24.32 | atik7 | i am having some problem with asterisk playback, it doesnt move to next dialplan command, its keep stop on playback, any one knows what is the couse for this? |
| 18:24.33 | drmessano | Yes you do, but first, ASL? |
| 18:24.37 | stansmith | [T]ank: nah its cool man, im gettin to the bottom of it, what were your issues with that server? |
| 18:24.46 | *** join/#asterisk BCS-Satori (n=somewher@75.148.21.113) |
| 18:24.49 | [TK]D-Fender | Leiste: If you are asking what I jsut specified, then no. You can get "in use", but "ringing" will get lumped into that category. |
| 18:24.49 | Maq | damn nickserv |
| 18:25.06 | activo | [TK]D-Fender ever hear of askerisk used ask for DTMF prompts to fetch PDF work orders from a web server and fax them to a work site? We have a case where this would help streamline our operations. |
| 18:25.15 | [TK]D-Fender | atik7: pastebin your dialplan, and the CLI output at verbose 10 of your attempt. |
| 18:25.17 | [TK]D-Fender | ~pb |
| 18:25.18 | 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 |
| 18:25.19 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 18:25.29 | stansmith | ^^^ in case you didnt see it ^^^ |
| 18:25.30 | [T]ank | stansmith: irq sharing I had to turn off half of the servers hardware to get my t1 card to be on its own irq |
| 18:25.42 | [TK]D-Fender | activo: Doable |
| 18:26.34 | stansmith | [T]ank: yea, i had a similiar problem, the pci-x bus wasnt working with the test tdm400p card i had, moving it to a pci-x bus with higher speed seemed to solve it...that irq conflict, good to know though |
| 18:26.36 | activo | TK it probebly is. Now the next question which languages would be the most likely use for this? I know almost nothing about perl but know it could be used. |
| 18:26.39 | stansmith | ill keep that in the back of my mind |
| 18:27.01 | atik7 | *CLI> core set verbose 10 |
| 18:27.02 | atik7 | Verbosity was 3 and is now 10 |
| 18:27.04 | [T]ank | it sucks... because i had to turn off my redundant nic |
| 18:27.05 | atik7 | *CLI> -- Executing [500@default:1] Playback("SIP/6404285496-08202b78", "demo-abouttotry") in new stack |
| 18:27.09 | atik7 | <PROTECTED> |
| 18:27.14 | [T]ank | because it shared the same irq as the pcix slot |
| 18:27.55 | stansmith | calling digium, they said the cards require 2.2 compliance, i called hp and they said all my pci-x bus' are 1.08 compliant, so im scratching my head as to how it works |
| 18:28.04 | atik7 | then it keepwait here, and no audio |
| 18:28.05 | [T]ank | lol |
| 18:28.37 | [TK]D-Fender | atik7: PASTEBIN what I requested please. |
| 18:29.17 | [TK]D-Fender | activo: Whatever you feel like using |
| 18:29.24 | *** join/#asterisk egecko (n=sam@cpe-76-176-238-225.san.res.rr.com) |
| 18:31.01 | *** join/#asterisk Rei-chan (n=lazy@c-75-64-155-65.hsd1.tn.comcast.net) |
| 18:31.12 | atik7 | <PROTECTED> |
| 18:31.13 | atik7 | <PROTECTED> |
| 18:31.15 | atik7 | <PROTECTED> |
| 18:31.20 | stansmith | LOL |
| 18:31.22 | robl^ | PCIX != PCI |
| 18:31.28 | activo | probebly true. Often we are in the field and the cleints have a firewall locked out network so we cannot just log into our site and download and print the work order. This would be a simple solution to just call the asterisk server, log in, request a work order number then it prompts me for the fax number and sends it to me. |
| 18:31.30 | stansmith | the man loves not using pastebin |
| 18:32.22 | Wayhigh | I've got some strange pci slots in my server board.. 64bit pci slots that can run at 66Mhz? |
| 18:32.28 | [TK]D-Fender | atik7: LAST TIME. Pastebin your dialplan direct from extensions.conf, and the complete CLI output of your attempt. Do not spam in here |
| 18:32.40 | activo | agi |
| 18:32.42 | activo | :) |
| 18:32.56 | Wayhigh | activo: too high.. AMT for me.. |
| 18:33.04 | Wayhigh | (ok ok.. bad pun..) |
| 18:33.16 | activo | ? |
| 18:33.20 | atik7 | ok , i got it now, this is my first time using irc, |
| 18:33.31 | Rei-chan | Is there an FAQ on getting Asterisk to compile for Ubuntu 7.10? I hit the no termcap support problem, and found only dead forum topics in the support forum on both sides. |
| 18:33.43 | *** join/#asterisk thedonvaughn (i=jayson@unaffiliated/printk) |
| 18:33.56 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 18:34.08 | clickonce | generalhan: Where are the sounds folder located? I can't find it. |
| 18:34.09 | stansmith | Rei-chan: if you follow the asterisk book, which lists the prereqs..you can usually just replace [package name]-devel with [package-name]-dev |
| 18:34.15 | Wayhigh | AMT = alternative minimum tax.. it's the stuff you get hit from by the .gov in order to smack you down for being successful and make you more in line with all the other poor people. |
| 18:34.30 | stansmith | Rei-chan: what part is the compilation complainin about? |
| 18:34.41 | Rei-chan | configure. No termcap support found. |
| 18:34.45 | clickonce | clickonce: And when I built asterisk I saw something about ringtones, where are those? I've updatedb'd and locate'ed the files without any results. |
| 18:34.59 | clickonce | generalhan: ^^ |
| 18:35.14 | Rei-chan | and aptitude does not have a termcap lib package. I'm told that termcap is ancient and no longer supported and the application should be checking for ncurses. |
| 18:35.56 | [TK]D-Fender | Rei-chan: NCURSES , NCURSES-DEVEL , ETC |
| 18:36.37 | stansmith | Rei-chan: yea, did you get ncurses-dev? |
| 18:37.09 | Leiste | soory |
| 18:37.52 | Rei-chan | There we go, yay. Where's this book? |
| 18:37.57 | stansmith | ~book |
| 18:37.58 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 18:38.05 | *** join/#asterisk guillote_GNU (n=guillote@host208.190-30-106.telecom.net.ar) |
| 18:38.08 | Leiste | <[TK]D-Fender> but with snom it work fines -doesn't it? |
| 18:38.26 | Rei-chan | Ok, that's what I've been using excerpts from in various ubuntu tutorials, thanks. |
| 18:38.36 | [TK]D-Fender | Leiste: Maybe, I don't use Snom |
| 18:38.41 | stansmith | Rei-chan: page 39..it lists RH package names but its not too hard to translate it to .debs |
| 18:40.42 | stansmith | runlevel 3 doesnt have X11 running? [confirm/deny] |
| 18:40.59 | clickonce | confirm |
| 18:41.05 | clickonce | X11 usually goes in 5 |
| 18:41.20 | clickonce | At least on the dists I've been using. |
| 18:41.24 | Nivex | unless you are under Debian/Ubuntu, in which case everything runs at runlevel 2 |
| 18:41.27 | lirakis | is there any indicator that a callfile has been succesfully processed? |
| 18:41.38 | stansmith | yea, 5 = 3 with X11...or so /etc/inittab says |
| 18:41.42 | clickonce | Well Ubuntu is a windows-crap-wannabe so... |
| 18:41.46 | lirakis | I guess... the cdr .. maybe |
| 18:41.49 | lirakis | hmm... |
| 18:41.49 | clickonce | I couldn't care less about it. |
| 18:42.46 | drmessano | Ubuntu needs a better name and KDE by default |
| 18:43.04 | stansmith | kubuntu |
| 18:43.14 | drmessano | Yeah, thats not "Ubunut" |
| 18:43.14 | clickonce | KDE is so very bloated... It used to be good. |
| 18:43.17 | drmessano | Yeah, thats not "Ubuntu" |
| 18:43.26 | drmessano | KDE 4 looks awesome |
| 18:43.27 | stansmith | it is the essence of ubuntu |
| 18:44.46 | drmessano | Kubuntu as an option is still Ubuntu 9.10 Goatse Giver not having KDE when Mr n00bsmith installs it |
| 18:44.58 | clickonce | haha |
| 18:45.15 | stansmith | 9.10 really gonna be called that? lol |
| 18:45.21 | stansmith | wait |
| 18:45.23 | drmessano | It should be |
| 18:45.25 | stansmith | ive been tricked! |
| 18:45.27 | Leiste | [TK]F - so you get the busy signn, but not the blinking while it is ringing - write? Are you able to pick the call up at his moment? |
| 18:45.43 | drmessano | and Ubuntu server is a joke.. |
| 18:45.48 | drmessano | It needs work |
| 18:46.10 | [TK]D-Fender | Leiste: yOU SEE HIM AS "BUSY". wHAT YOU do IS UP TO YOUR DIALPLAN AND WHAT YOU DIAL |
| 18:46.24 | stansmith | the man loves caps lock and then using shift still |
| 18:46.40 | clickonce | lol |
| 18:47.12 | [TK]D-Fender | stansmith: I work in all-caps a lot and miss it occasionally. |
| 18:47.16 | drmessano | Usually that means we're at NEWBCOM 4 |
| 18:47.30 | drmessano | NEWBCOM 5 is "~GTFO" |
| 18:47.32 | [TK]D-Fender | stansmith: at least you can tell what if anything I actually intended on emphasizing. |
| 18:47.32 | *** join/#asterisk af_ (n=getsmart@88-149-240-36.dynamic.ngi.it) |
| 18:47.40 | stansmith | lol yea |
| 18:47.44 | drmessano | ~gtfo |
| 18:47.44 | jbot | Sorry sir, I won't bother you anymore. |
| 18:47.49 | [TK]D-Fender | drmessano: that'd be "NEWBCON". |
| 18:47.59 | drmessano | Uh yeah |
| 18:48.03 | drmessano | Duh |
| 18:48.20 | drmessano | For some reason I didn't think it sounded right lol |
| 18:48.28 | drmessano | DEFCON = NEWBCON |
| 18:48.32 | stansmith | newbcom sounds more authentic |
| 18:48.34 | Leiste | [TK] F : Yes, thats right :) - I'd like to see the blinking on my phone if A calls B - and when B is not avaible - I'd like to pick the call up via pressing the button auf the Polycom phone |
| 18:48.45 | drmessano | "NEWBCON 4, scramble the bombers" |
| 18:48.45 | [TK]D-Fender | DEFCON = DEFense CONdition. |
| 18:48.49 | Leiste | you know what i mean? |
| 18:49.07 | drmessano | Dude, I totally saw Wargames 1000 times.. |
| 18:49.34 | [TK]D-Fender | Leiste: You can still pick it up by pressing the speed-dial you linked to his presence. it won't BLINK like you'd like it to, but you can still see ti at least. the actual act of grabbing his call is up to you and your dialplan. |
| 18:49.55 | [TK]D-Fender | drmessano: I partially saw it once, does that count? |
| 18:50.04 | drmessano | "Take us to DEFCON 1, and get me the president on the line" |
| 18:50.16 | drmessano | heh |
| 18:50.23 | denon | west wing re-runs? |
| 18:50.26 | drmessano | Wargames is obligatory, you have to see it, own it, etc |
| 18:50.29 | denon | oh, wargames |
| 18:50.50 | Leiste | [TK] F - What means ti? |
| 18:50.52 | drmessano | Asterisk == The WOPR |
| 18:50.58 | *** join/#asterisk simbol76ss (n=simbol@host65-209-dynamic.3-87-r.retail.telecomitalia.it) |
| 18:51.03 | denon | heheh |
| 18:51.38 | x86 | WOPR? |
| 18:51.40 | [TK]D-Fender | Leiste: s/ti/it/ |
| 18:51.43 | drmessano | I've heard asterisk has a hidden CLI function to scramble the bombers |
| 18:51.55 | drmessano | War Operations Planned Response |
| 18:52.03 | drmessano | The WOPR |
| 18:52.09 | drmessano | Hey denon |
| 18:52.20 | JunK-Y | sure, its !/usr/sbin/start_bomb |
| 18:52.22 | drmessano | Google for CPE1704TKS |
| 18:52.45 | Leiste | [TK]F - Thanks for your help! Last question - what SIP Version do you run with your 650? sorry for the ti question!!! you see that I'am not a native speaker...greetings from Hamburg,Germany |
| 18:52.48 | drmessano | Sadly, I remember that from the Movie |
| 18:53.10 | drmessano | That was the launch code |
| 18:53.13 | [TK]D-Fender | Leiste: I don't have a 650, and if I did, it'd probably be SIP 3.0.0 |
| 18:53.38 | Leiste | OK - I'running it! Fine |
| 18:53.45 | Leiste | Bye @all |
| 18:58.45 | *** part/#asterisk [T]ank (n=ckwall@c-71-195-199-107.hsd1.ut.comcast.net) |
| 19:02.48 | *** join/#asterisk angom (n=angom@200.79.141.128.dsl.dyn.telnor.net) |
| 19:06.24 | *** join/#asterisk neoalex (n=chatzill@user-0ccens8.cable.mindspring.com) |
| 19:06.52 | neoalex | hi, what options do I have for sending SMS from asterisk |
| 19:07.25 | Daviey | neoalex: Using what sms sending hardware? |
| 19:07.33 | Daviey | ie, end user hardware? |
| 19:07.54 | *** join/#asterisk AndyGraybeal (n=andy@node84.32.251.72.1dial.com) |
| 19:08.17 | neoalex | no hardware, just a a 3rd party gateway like bayham for example |
| 19:08.42 | neoalex | or SMS through my land-line though as far as I know that's only possible in europe |
| 19:08.46 | Daviey | If you aren't using some funky sip phone, i'd not use asterisk at all for that |
| 19:08.53 | Daviey | probably perl/bash scripts |
| 19:09.20 | neoalex | well yes, but asterisk has to run one of those scripts, basically I want to send an SMS when I have a new voice-mail |
| 19:09.26 | Daviey | thats if you use a net sms gateway |
| 19:09.36 | Daviey | erm |
| 19:10.13 | neoalex | do you happen to know other gateways? |
| 19:10.19 | Daviey | Three ways come to mind: |
| 19:10.25 | Daviey | 1) monitor voicemail folder |
| 19:10.36 | Daviey | 2) write in an agi script into the dialplan |
| 19:10.56 | Daviey | 3) email notification of vm to sms@localhost |
| 19:11.04 | *** join/#asterisk cjk (i=ldidelot@d212-66-83-208.cust.tele2.lu) |
| 19:11.05 | Daviey | where sms@localhost = procmail |
| 19:13.40 | drmessano | why 3/ |
| 19:13.40 | drmessano | why 3? |
| 19:14.02 | drmessano | <PROTECTED> |
| 19:14.25 | Daviey | drmessano: thats not sms :) |
| 19:14.42 | drmessano | No, same difference |
| 19:14.43 | Daviey | not every network/country supports that |
| 19:15.06 | drmessano | Well, the US supports it, and we're behind the rest of the world |
| 19:15.15 | drmessano | So it can't be that uncommon |
| 19:15.44 | stansmith | america r0x! |
| 19:15.48 | drmessano | I've been sending emails to cell phones for 10 years now |
| 19:15.54 | Daviey | good for you |
| 19:16.12 | drmessano | Wow, you're an ass |
| 19:16.20 | stansmith | me? |
| 19:16.23 | x86 | but that's not SMS drmessano |
| 19:16.23 | drmessano | No |
| 19:16.39 | drmessano | Whats the difference how it gets there? |
| 19:17.05 | Daviey | 19:14:43 < Daviey> not every network/country supports that |
| 19:17.07 | x86 | for example, I have to pay for inbound email support on my phone (so that it shows up like a text message would) |
| 19:17.08 | neoalex | yeah, it's not sms, we're doing that now, and for some reason my boss wants me to find a way to send true sms, though for the life of me I can't see why |
| 19:17.09 | drmessano | Pagers and cell phones have supported email > phone for years.. thats how I have always done it |
| 19:17.24 | cjk | hi, i have a problem. when i take a call there is a delay of approximately 1 second till the call is established. the first 2 works are cut off. any idea? later in the call there is no delay |
| 19:17.29 | x86 | drmessano: some carriers you have to pay for that, where as SMS may be included with the plan, etc |
| 19:17.39 | drmessano | SMS is only going to cost more in the long run if you do it that way |
| 19:17.41 | Daviey | cjk: * behind a NAT? |
| 19:17.55 | cjk | Daviey, no |
| 19:18.07 | JoseBravo | How can I change the codec that use one SIP trunk? |
| 19:18.08 | stansmith | cjk: Answer(2); ? |
| 19:18.19 | drmessano | x86: Most carriers support it without having "EMAIL" |
| 19:18.22 | neoalex | drmessano: I know, I'm not paying for it though |
| 19:18.29 | Daviey | JoseBravo: hint: disallow all, allow $codec |
| 19:18.41 | JoseBravo | Daviey: Thanks |
| 19:18.44 | neoalex | JoseBravo: in the trunk say disallow=all then allow= some codec |
| 19:18.48 | cjk | stansmith, why should i answer a call in the dialplan. this will cost money. i answer if the call with my phone |
| 19:19.02 | stansmith | misunderstood, chill |
| 19:20.10 | JoseBravo | Daviey, done... how can I check what codec is the SIP using? |
| 19:20.58 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 19:21.26 | Daviey | JoseBravo: make a call, then sip show channels |
| 19:22.02 | Daviey | (under title FORMAT) |
| 19:23.09 | JoseBravo | Daviey see my output: http://www.pastebin.ca/912976 |
| 19:23.24 | yangvnc | I experience the following issue with zaptel ISDN, the incoming call comes in properly - by the outgoing call when I call number 041710598 I see on my VOIP phone 503710598 - http://openpaste.org/en/5221/ |
| 19:23.32 | Daviey | JoseBravo: using gsm? |
| 19:23.45 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:23.45 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 19:25.02 | defsdoor | yangvnc: outgoing cli is usually set by the telco |
| 19:25.57 | yangvnc | defsdoor: do you know how to fix it, it cannot dial 503710598 (the number doesnt exist) it doesnt connect me |
| 19:27.05 | defsdoor | what's the cli when you dial a normal phone ? |
| 19:27.13 | JoseBravo | Daviey, I haved g711 and changed for ulaw and now shows ulaw.. |
| 19:28.54 | yangvnc | defsdoor: what do you mean with normal phone? |
| 19:28.54 | *** join/#asterisk dswillia (n=me2@vpn.choicepay.com) |
| 19:29.25 | defsdoor | you are dialling out over your isdn line to a voip service |
| 19:29.32 | yangvnc | no |
| 19:29.41 | yangvnc | i am dialing over isdn to a pstn number |
| 19:29.46 | SteveTotaro | i support SMS |
| 19:29.49 | yangvnc | and that always comes up |
| 19:29.52 | SteveTotaro | with Kannel |
| 19:30.14 | defsdoor | yangvnc: aaah - so the "I see on my VOIP phone" was just to mislead ? |
| 19:30.25 | defsdoor | yangvnc: speak to telco |
| 19:30.30 | dswillia | hey all, quick newb questions i am needing to find a list of announcements that are played during our conference bridge usage, what config file are those found in? |
| 19:30.52 | yangvnc | defsdoor: well, do you think my zapata.conf might be wrong? |
| 19:30.59 | activo | What is the lowest power server for a office of 10 users that can be deployed? Perfer embeded with the nessesary pci-x or pci slot capability to take fxo/fxs cards. |
| 19:31.02 | yangvnc | defsdoor: or signaling? |
| 19:31.09 | defsdoor | yangvnc: no |
| 19:31.15 | activo | Ie, a server that consumes the least amount of power |
| 19:31.20 | yangvnc | defsdoor: i can show you my zapata.conf if you think there is a fix for it |
| 19:31.25 | defsdoor | yangvnc: unless your telco lets you set the cli to anything you please |
| 19:31.29 | neoalex | SteveTotaro: do you use any hardware with Kannel? |
| 19:31.41 | neoalex | or just a phone |
| 19:31.42 | SteveTotaro | ~stevetotaro |
| 19:31.43 | jbot | somebody said stevetotaro was an IRC nub |
| 19:31.55 | stansmith | ~stansmith |
| 19:32.00 | yangvnc | defsdoor: but how can i dial out to "any" number if (503) always comes first |
| 19:32.02 | stansmith | :( |
| 19:32.21 | SteveTotaro | i use a whole bunch of bluetooth phones |
| 19:32.26 | SteveTotaro | to send sms |
| 19:32.27 | defsdoor | yangvnc: you aren't making sense |
| 19:32.28 | [TK]D-Fender | yangvnc: pastebin your configs |
| 19:32.42 | yangvnc | [TK]D-Fender: sure |
| 19:32.46 | neoalex | it works via bluetooth too, that's cool |
| 19:33.07 | neoalex | can you use chanmobile at the same time? |
| 19:33.09 | [TK]D-Fender | dswillia: its only in the source |
| 19:33.27 | SteveTotaro | well once paired via bluez it is just a modem as far as linux is concerned |
| 19:33.49 | SteveTotaro | chan mobile seems to only support one phone per dongle |
| 19:34.07 | SteveTotaro | with Kannel, i can connect a BUNCH of phones on one dongle |
| 19:34.11 | drmessano | heh, dongle |
| 19:34.23 | Daviey | SteveTotaro: you use SMS() for that? |
| 19:34.29 | activo | SteveTotaro which ones |
| 19:34.33 | SteveTotaro | no, i use Kannel |
| 19:35.19 | SteveTotaro | with kannel, you can send sms by hitting a properly formed url with username password destination message all part of the url |
| 19:35.36 | *** join/#asterisk DJF6 (n=DJF5@84-105-201-37.cable.quicknet.nl) |
| 19:35.57 | SteveTotaro | so system(lynx${url}) or whatever |
| 19:36.02 | Daviey | SteveTotaro: how does the outbound message get to Kannel? |
| 19:36.11 | Daviey | ahh, i see |
| 19:36.26 | SteveTotaro | it works very well |
| 19:36.38 | stansmith | hey, computer science is no more about computers than astronomy is about telescopes |
| 19:36.39 | SteveTotaro | each phone gives me 1 sms per second |
| 19:36.53 | Daviey | geez, how many do you send? |
| 19:37.13 | yangvnc | defsdoor: [TK]D-Fender http://openpaste.org/en/5222/ |
| 19:37.14 | SteveTotaro | not many now but my new startup could be sending thousands |
| 19:37.18 | SteveTotaro | in bursts |
| 19:37.31 | Daviey | SteveTotaro: wouldn't a bulk sms net gateway be cheaper? |
| 19:37.41 | SteveTotaro | not in the US |
| 19:37.43 | Daviey | oh |
| 19:37.48 | SteveTotaro | i can receive too |
| 19:38.03 | SteveTotaro | a short code in the us is $500/mo |
| 19:38.10 | Daviey | US ftl :( |
| 19:38.11 | SteveTotaro | or something like that |
| 19:38.30 | *** join/#asterisk Docfxit (n=none@ip-64-32-143-214.lax.megapath.net) |
| 19:38.40 | [TK]D-Fender | yangvnc: -- Executing [041710598@buster:1] Dial("SIP/30-c000a180", "ZAP/g1/041710598") in new stack |
| 19:38.41 | SteveTotaro | right now, per five phones I pay $125 for unlimited SMS in and out |
| 19:38.46 | [TK]D-Fender | yangvnc: yOU DON'T have A "GROUP=1" |
| 19:39.07 | Daviey | SteveTotaro: *unlimited*? |
| 19:39.10 | [TK]D-Fender | stupid caps |
| 19:39.29 | yangvnc | [TK]D-Fender: Yeah, I am not familiar with groups...whoops that i need to change in zapata.conf |
| 19:39.31 | SteveTotaro | yes, unlimited plan is $20 extra on a family plan |
| 19:39.47 | SteveTotaro | family plan can have up to five phones |
| 19:40.01 | yangvnc | ZAP/g1/041710598 is related g1=group1 ? |
| 19:40.08 | Daviey | I wonder if they moan if you send thousands per week |
| 19:40.24 | [TK]D-Fender | yangvnc: *yes* |
| 19:40.25 | SteveTotaro | i have done it several times for testing |
| 19:40.52 | SteveTotaro | 10,000 was the most i sent just to get an idea of scaling and throughput |
| 19:41.06 | neoalex | in a week or month? |
| 19:41.15 | SteveTotaro | Kannel didn't even cough |
| 19:41.23 | SteveTotaro | all at once queued up in Kannel |
| 19:41.38 | neoalex | wooow... nice |
| 19:41.41 | SteveTotaro | so ten a second total |
| 19:41.51 | Daviey | shame you can't spoof sender id tho (for genuine purpose) |
| 19:42.03 | SteveTotaro | what is 10,000 divided by 60? |
| 19:42.22 | DJF6 | [Win]+[R] > calc > [ENTER] |
| 19:42.47 | SteveTotaro | no workie in linux |
| 19:42.47 | *** join/#asterisk nvrpunk (n=root@81.90.21.227) |
| 19:42.59 | DJF6 | should be a calc in your menu ;) |
| 19:43.10 | SteveTotaro | i am on the linux cli |
| 19:43.17 | Daviey | alt + f2 => gcalctool |
| 19:43.24 | neoalex | 166.666666666666 |
| 19:43.25 | DJF6 | #!/bin/php |
| 19:43.28 | DJF6 | <?php |
| 19:43.30 | Daviey | php!! |
| 19:43.31 | neoalex | 6666666666666 |
| 19:43.38 | DJF6 | var_dump(10000/60); ?> |
| 19:43.41 | defsdoor | never heard of bc ? |
| 19:43.55 | nvrpunk | say, I have an extension 1004, but right now in my extensions.conf it doesnt just dial at 4 digits, you have to hit dial on the phone, how would I make it realize 4 digits is an acceptable amount to autodial? |
| 19:44.02 | SteveTotaro | i just wanted someone else to do my math for me |
| 19:44.07 | SteveTotaro | ;) |
| 19:44.18 | defsdoor | nvrpunk: dial plans on the phone config |
| 19:44.30 | neoalex | nvrpunk: what phone? |
| 19:44.31 | nvrpunk | defsdoor, ah |
| 19:44.36 | nvrpunk | links 922 |
| 19:44.39 | nvrpunk | sys& |
| 19:44.40 | drmessano | defsdoor: Dialplans on phones are NEVER a problem |
| 19:44.44 | *** join/#asterisk stack_ (n=sgerstac@mail.edpaymentsystems.com) |
| 19:44.45 | SteveTotaro | bc is boston college |
| 19:44.46 | drmessano | Ever |
| 19:44.52 | Daviey | $ echo $(( 10000 / 60 )) |
| 19:44.56 | neoalex | nvrpunk: there's usually an auto dial delay option in the phone |
| 19:45.04 | defsdoor | drmessano: pardon ? |
| 19:45.20 | nvrpunk | couldnt i do some sort of pattern matching? |
| 19:45.23 | defsdoor | drmessano: you saying users must always tell phone to dial instead ? |
| 19:45.24 | yangvnc | Another question which I have is, that we got this weird situation now, I have always used asterisk on a public IP, and now we have just one public IP and router behind it 192.168.1.1 and asterisk NAted IP is 192.168.1.5 , How can I make this asterisk to be able to accept calls from outside...Do I need to forward some ports from the router? |
| 19:45.26 | nvrpunk | i was thinking it was asterisk |
| 19:45.28 | nvrpunk | not the phone |
| 19:45.31 | drmessano | <sarcasm /> |
| 19:45.36 | drmessano | It's never the phone |
| 19:45.41 | defsdoor | nvrpunk: most phones dial plans are patterns |
| 19:45.44 | drmessano | Diaplans are never a problem |
| 19:45.50 | drmessano | Dialplans |
| 19:45.55 | drmessano | ~dialplan |
| 19:45.56 | jbot | from memory, dialplan is the thing configured in extensions.conf |
| 19:46.00 | drmessano | ~dialplans |
| 19:46.18 | stack_ | Our asterisk server connects to a SIP peer named 'trunk1', so if we want to reach an extension at that site, I dial SIP/trunk1/101. If I try to use that extension in a call queue, the queue marks these entries as invalid. Any ideas on what I'm doing wrong? |
| 19:46.31 | clickonce | I found this wonderful SPA-932 addon as well :) |
| 19:46.31 | SteveTotaro | dialplan can be configured in database, jbot is old skewl |
| 19:46.35 | [TK]D-Fender | yangvnc: Read up : |
| 19:46.37 | [TK]D-Fender | ~sipnat |
| 19:46.38 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 19:46.40 | [TK]D-Fender | ^^^^^^^^^^ |
| 19:46.54 | neoalex | try SIP/101@trunk1 |
| 19:47.12 | neoalex | stack_: try SIP/101@trunk1 |
| 19:47.18 | drmessano | ~phone dialplans |
| 19:47.18 | jbot | <sarcasm> phone dialplans are never a problem </sarcasm> |
| 19:47.30 | drmessano | bah |
| 19:47.35 | stack_ | neoalex: okay, thanks |
| 19:47.46 | drmessano | ~phone dialplans |
| 19:47.46 | jbot | <sarcasm> phone dialplans are never the problem </sarcasm> |
| 19:47.46 | neoalex | you missed the sarcasm part drmessano ? |
| 19:47.49 | drmessano | better |
| 19:48.01 | drmessano | _the_ |
| 19:48.02 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:48.13 | SteveTotaro | my siptrunk is acting funny |
| 19:48.16 | drmessano | No, just the statement of |
| 19:49.06 | drmessano | "I am having a problem dialing out.. When I dial 83 digits, there's 20 seconds of silence and then 911 is dialed. It's not the phone dialplan, any ideas?" |
| 19:50.20 | defsdoor | that sounds like your dial plan timeout on the phone to me - have you checked it ? |
| 19:50.34 | drmessano | It's not the dialplan... Let me paste it |
| 19:50.55 | defsdoor | in irc of course |
| 19:50.58 | stansmith | LOL |
| 19:51.02 | stack_ | neoalex, that seems to have done it, thanks |
| 19:51.13 | drmessano | [45840584985||||911|||,L84748||#*|[1-56]3.14{^^}] |
| 19:51.25 | drmessano | Anything wrong with that? SEE, NO.. THOUGHT SO |
| 19:51.28 | drmessano | NOW HALP ME |
| 19:51.48 | stansmith | LOL |
| 19:52.04 | drmessano | Maybe it's my IAX.conf |
| 19:52.11 | defsdoor | drmessano: I can help you but first you must lick the poles of a 9volt battery |
| 19:52.34 | drmessano | heh |
| 19:52.36 | yangvnc | [TK]D-Fender: thanks |
| 19:53.03 | drmessano | Ok, I think I narrowed it down |
| 19:53.06 | [TK]D-Fender | drmessano: Not enough PI! |
| 19:53.22 | drmessano | Its not asterisk, and it's not my phone.. Can I paste my autoexec.bat and config.sys in hear? |
| 19:53.42 | clickonce | lol |
| 19:53.52 | clickonce | hear? jesus :P |
| 19:53.58 | defsdoor | I have a bunch of english sound files - do I just replace the existing with them or can I put them in a different location ? |
| 19:54.24 | drmessano | I don't mean to be a jerk (yes I do), but if I hear, one more time, "It's the dialplan in the phone" |
| 19:54.26 | clickonce | defsdoor: Whatever you want :) |
| 19:54.47 | drmessano | After hours of "No, it cant be" |
| 19:54.57 | DJF6 | what is the syntax a dailplan excists of? google couldn't awnser that question :s |
| 19:54.59 | drmessano | 911 works, and Pi works |
| 19:55.15 | drmessano | So I changed my phones to extension 911 and 3.14 |
| 19:55.20 | yangvnc | [TK]D-Fender: very great thanks, which time are you online tomorow? |
| 19:55.22 | drmessano | So I think i'm set |
| 19:55.50 | drmessano | ~asteriskcat |
| 19:55.51 | jbot | i guess asteriskcat is not amused |
| 19:56.22 | drmessano | Damn |
| 19:56.23 | [TK]D-Fender | yangvnc: depends |
| 19:56.42 | drmessano | I have been asked to create the job posting for my job |
| 19:56.44 | drmessano | Hmm |
| 19:57.08 | [TK]D-Fender | drmessano: I had a guy here stuck with the same. Serious shafting. |
| 19:57.22 | [TK]D-Fender | drmessano: Time to make your life sound as miserable as possible!@ |
| 19:57.28 | drmessano | "Must enjoy crappy hours, 2am phone calls for MySpace problems, low pay, and painful urination" |
| 19:57.42 | *** join/#asterisk joelsolanki (i=joelsola@220.224.114.170) |
| 19:58.02 | yangvnc | [TK]D-Fender: I will be around experiencing troubles probably 9am UTC+1 , at that time nearly nobody speaks here |
| 19:58.05 | drmessano | [TK]D-Fender: You should at least know me a LITTLE by now.... |
| 19:58.10 | drmessano | Im gonna make this thing SHINE |
| 19:58.20 | drmessano | I am gonna make it sound like the best job EVAR |
| 19:58.27 | drmessano | </evil> |
| 19:58.32 | [TK]D-Fender | drmessano: time to beak out the Turd Polish (TM) |
| 19:58.47 | [TK]D-Fender | yangvnc: What troubles? |
| 19:58.56 | yangvnc | [TK]D-Fender: NAT issues perhaps |
| 19:58.56 | drmessano | "Do you want an EXCITING career in a FAST PACED industy with EXCITING benefits and painful urination" |
| 19:59.03 | drmessano | Ok, not the last part |
| 19:59.23 | [TK]D-Fender | yangvnc: Just go read the guide and follow it |
| 19:59.33 | defsdoor | drmessano: have my job |
| 19:59.47 | yangvnc | [TK]D-Fender: i surelly will do that ! |
| 19:59.50 | drmessano | lol |
| 20:00.03 | drmessano | Take mine |
| 20:00.46 | drmessano | Fun, but boring an low pay |
| 20:00.49 | drmessano | and* |
| 20:01.08 | drmessano | In 3 or 4 years, they may be ready for VoIP |
| 20:01.14 | defsdoor | my problem is I am paid too much |
| 20:01.29 | defsdoor | so I can't leave and do my own thing |
| 20:01.38 | drmessano | Ah |
| 20:01.45 | drmessano | So you're not challenged, but the pay is good |
| 20:01.52 | stansmith | drmessano: what is your job? |
| 20:02.17 | drmessano | For the next two weeks I am the IT Director and Engineer for 7 radio stations |
| 20:02.19 | defsdoor | I do my own thing pretty much - but we got taken over last april |
| 20:02.46 | clickonce | Ey! I think the dialplan in the UniPhone (P990i WLAN SIP software) is screwed... whatever number I dial... I get an error... |
| 20:03.01 | drmessano | Radio pays lousy and continues to be technologically behind |
| 20:03.17 | activo | My job can be stressfull as a bell/ibm contractor but at least I get a 1 year old van to use all the time :) |
| 20:03.45 | jameswf | Radio is like BSD :)) |
| 20:03.52 | stansmith | burn! |
| 20:04.03 | drmessano | Thats another one of my problems.. Its VERY stressful, but not for the right reasons |
| 20:04.27 | jameswf | ~unixdog |
| 20:04.28 | jbot | <unixdog> Everyone use BSD gah linux sux progress is overrated use my project gah |
| 20:04.43 | stansmith | lol? |
| 20:04.46 | activo | drmessano ever work on the traciever equipment? Is the FCC licence still required to work on the gear? I still have mine 15 years running. |
| 20:05.09 | drmessano | Yep, done a share of RF work |
| 20:05.27 | drmessano | Monday: We decided we're going to give every listener in the area a computer to listen to our streaming stations on.. You're building them.. need them done by Friday |
| 20:05.28 | activo | so fcc lience still required? |
| 20:05.35 | drmessano | Not anymore |
| 20:05.40 | activo | Interesting |
| 20:05.46 | jameswf | mmmmm tower work, never a cold day whrn your in front of a dish,,,,, oh look cancer |
| 20:05.51 | drmessano | Tuesday <> Thursday: FAST, WE NEED THOSE BUILT |
| 20:06.18 | activo | Its most likely still used for Aviation and ship based work |
| 20:06.19 | drmessano | Friday: You know what, we decided it was a bad idea.. Take those 11,000 machines you already imaged and send them back |
| 20:06.47 | jameswf | I didnt like being up on 25ft poles I wouldnt wanna be on a 150ft tower |
| 20:06.57 | drmessano | I'll go up about 40 ft |
| 20:06.59 | drmessano | Thats IT |
| 20:07.01 | activo | I thought of tower work |
| 20:07.17 | clickonce | I'd sure as hell climb a tower server. |
| 20:07.21 | activo | Nice being in the mountains in the fresh air :) |
| 20:07.24 | drmessano | Thats enough to do grounding and fix minor issues with line management |
| 20:07.34 | drmessano | heh |
| 20:07.34 | [TK]D-Fender | PULL!!!!!! |
| 20:07.44 | [TK]D-Fender | <SHOOMP> |
| 20:07.45 | joelsolanki | Hi all |
| 20:07.51 | stack_ | I have a queue with two members. if a member is "In Use", the queue still tries to dial that person, even though they are on the phone. Is there a way to stop this? |
| 20:07.57 | drmessano | I was dared to go up 60 feet.. |
| 20:08.01 | jameswf | the best part when I worked for "the company" was parking my truck in the road and shutting off a lane of traffic when there was a 30 ft easement... |
| 20:08.04 | drmessano | I got to 40 "Ok, this kinda scary" |
| 20:08.12 | joelsolanki | implementing chan_ss7 with sangoma A104D card on production server is stable ? |
| 20:08.14 | drmessano | 41 "Im gonna die" |
| 20:08.15 | joelsolanki | any body using it ? |
| 20:08.52 | jameswf | the resident sangoma guy stepped out |
| 20:09.09 | drmessano | ROFL |
| 20:09.18 | joelsolanki | who is that ? |
| 20:09.21 | activo | drmessano our company just finished up a network install with fiber and cable in a 140,000 sq foot warehouse with 50 foot Ceilings ;) Warehouse was of course massive. |
| 20:09.25 | jameswf | Steve |
| 20:09.41 | joelsolanki | hmm. is he is from sangoma ? |
| 20:09.45 | drmessano | Stevegoma |
| 20:09.54 | jameswf | nah just a fanboy |
| 20:10.03 | drmessano | Heh |
| 20:10.07 | joelsolanki | oh ok. |
| 20:10.09 | jameswf | Do you use Stvegoma cards |
| 20:10.12 | drmessano | I MAY survive a 50 foot lift |
| 20:10.22 | activo | dont see a stevegoma here. |
| 20:10.26 | joelsolanki | i dont see him online right now |
| 20:10.26 | drmessano | Hands white from clutching the basket |
| 20:10.33 | drmessano | Stevegoma is elusive |
| 20:10.42 | activo | Dr, the lift was pretty big. |
| 20:10.44 | jameswf | 25ft on a phone pole in a windstorm your expected to use both hands to work.... |
| 20:10.49 | Daviey | Stevegoma is an irc noob - so maybe he got lost :) |
| 20:11.01 | joelsolanki | i use sangoma |
| 20:11.02 | sweeper | jameswf: clench those thighs boy! |
| 20:11.02 | joelsolanki | oh ok |
| 20:11.15 | jameswf | ~stevegoma is oh nevermind |
| 20:11.15 | jbot | okay, jameswf |
| 20:11.20 | activo | Ever install a sangoma on a micro itx? |
| 20:11.30 | Daviey | sounds fun... |
| 20:11.37 | drmessano | Stevegoma has 2 kids.. he named them FXO and FXS.. guess which one is a boy and which one is a girl.. |
| 20:11.37 | activo | via micro itx? |
| 20:11.47 | joelsolanki | :) |
| 20:11.52 | jameswf | you mean a foft Ramora on a 6'' board |
| 20:12.10 | jameswf | FXS=boy |
| 20:12.21 | jameswf | put your FXS in your FXO |
| 20:12.31 | Daviey | :O |
| 20:12.45 | stack_ | I have a queue with two members. if a member is "In Use", the queue still tries to dial that person, even though they are on the phone. Is there a way to stop this? |
| 20:12.48 | jameswf | wanna se my FX-OH face |
| 20:13.16 | activo | One of our clients lost there dsl or T-1 one circuit to Bell and all there phones and work stations were dead in the water :) |
| 20:14.20 | activo | Guess thay never heard of the Single point of Failure network model ;) |
| 20:14.55 | jameswf | redundancy is overrated |
| 20:14.56 | stansmith | jameswf: LOL |
| 20:14.58 | AndyGraybeal | activo: what would they have done to get around Bell's failure? |
| 20:15.17 | jameswf | chan_mobile |
| 20:15.48 | drmessano | HAHA |
| 20:16.25 | drmessano | I hear a that a satellite backup with a virtual PRI works wonders |
| 20:16.30 | JoseBravo | Anyone can recomend me a good SIP client for windows and free? |
| 20:16.41 | *** join/#asterisk blaylock (n=blaylock@snap.helixsystems.com) |
| 20:16.42 | drmessano | X-Lite |
| 20:16.44 | jameswf | we are getting ready to add a satelite link |
| 20:16.52 | jameswf | *microwave |
| 20:16.57 | blaylock | what is the diff between cdr-csv and cdr-custom? |
| 20:16.57 | drmessano | Ah |
| 20:17.23 | jameswf | 8 megs trunked to WA for backup T1 |
| 20:17.34 | drmessano | niice |
| 20:17.45 | drmessano | Using Sangoma cards? |
| 20:17.52 | jameswf | nah Openvox |
| 20:17.55 | jameswf | :) |
| 20:18.09 | JoseBravo | drmessano do you know one that support transfer calls? |
| 20:18.16 | drmessano | What are those $20 chinese knockoffs |
| 20:18.19 | drmessano | X-Lite |
| 20:18.43 | codefreeze | blaylock: cdr-csv has a fixed format, and cdr-custom allows you to modify the format |
| 20:18.59 | jameswf | oddly Ramora cards dont have an fcc logo |
| 20:19.00 | blaylock | ahh gotcha |
| 20:19.11 | drmessano | What about a KGB logo? |
| 20:19.15 | blaylock | codefreeze, thanks man |
| 20:19.20 | JoseBravo | drmessano my x-lite says to upgrade to eyebeam to transfer feature... |
| 20:19.37 | blaylock | codefreeze, but otherwise all calls are logged to both correct? |
| 20:19.52 | drmessano | Ah |
| 20:19.55 | drmessano | Voiper maybe |
| 20:20.13 | codefreeze | yep, and if you configure both, you'll end up with fairly duplicate sets. |
| 20:20.37 | codefreeze | uh, blaylock ... see above ^ |
| 20:21.13 | drmessano | Ok, bbiab |
| 20:21.31 | blaylock | codefreeze, thanks again |
| 20:22.09 | *** part/#asterisk mmmToop (n=michaelt@dsl-243-239-90.telkomadsl.co.za) |
| 20:22.15 | BCS-Satori | ~voip provider |
| 20:22.20 | BCS-Satori | doh! |
| 20:22.22 | J4k3 | ~itsp |
| 20:22.23 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 20:22.39 | J4k3 | ~ronpaul |
| 20:22.47 | BCS-Satori | ~itsplist-us |
| 20:22.47 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, or http://www.jnctn.com, or http://www.bandwidth.com, or http://vitelity.net |
| 20:22.51 | J4k3 | wtfbbq |
| 20:22.55 | J4k3 | ~wtfbbq |
| 20:22.55 | jbot | wtf |
| 20:23.01 | J4k3 | ~bbq |
| 20:23.02 | jbot | wtf |
| 20:23.28 | stansmith | jbot: LOL |
| 20:23.28 | jbot | [lol] stands for Laughing Out Loud. It is grammatically incorrect to use LOL in the first person; use 'heh' or 'haha' instead. If you want to use LOL, do '/me lol' instead. |
| 20:23.48 | J4k3 | jbot: lol is also for aolers |
| 20:23.48 | jbot | J4k3: cannot alter locked factoids |
| 20:24.16 | hmodes | hrmm... just got a 1.6-beta4 release email, but it's not on downloads.digium.com |
| 20:24.18 | stansmith | jbot been acting different lately |
| 20:24.18 | hmodes | 'whoops' |
| 20:24.29 | jameswf | ~idnms |
| 20:24.29 | jbot | Why would a Wookiee, an eight-foot tall Wookiee, want to live on Endor, with a bunch of two-foot tall Ewoks? That does not make sense! But more important, you have to ask yourself: What does this have to do with this channel? Nothing |
| 20:24.53 | *** join/#asterisk webtech_m33 (i=webtech@webtech.m33access.com) |
| 20:25.14 | J4k3 | ~kenya |
| 20:25.15 | jbot | i heard kenya is Where can you find Lions? Only http://mastaile.mine.nu/kenya1.mov ! |
| 20:25.32 | J4k3 | omg I Got beat to the kenya lions |
| 20:25.54 | webtech_m33 | does anyone know where i can find some perl examples for Asterisk::Manager |
| 20:26.42 | jameswf | ~perl |
| 20:26.43 | jbot | hmm... perl is at http://www.handhelds.org/z/wiki/Perl or at http://www.perl.com, or a knitting stitch, or the Pathologically Eclectic rubbish Lister, or that other "P" language |
| 20:27.05 | *** join/#asterisk goodmove (n=yves@69.57.246.162) |
| 20:27.28 | J4k3 | jbot: kenya is also http://www.weebls-stuff.com/toons/kenya/ |
| 20:27.28 | jbot | J4k3: okay |
| 20:27.55 | signius | Has anyone here messed around with the pbx-in-a-flash and if so whats your opinion of it ? |
| 20:28.08 | webtech_m33 | yeah the perl.com one |
| 20:28.17 | goodmove | Hello all |
| 20:28.50 | goodmove | I have an unusual problem that I have been battling |
| 20:29.08 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 20:29.09 | *** mode/#asterisk [+o anthm] by ChanServ |
| 20:29.37 | goodmove | I am attempting to set up Asterisk to use Mysql using Redhat Linux |
| 20:30.20 | webtech_m33 | for mysql support you have to load asterisk-addons |
| 20:30.32 | goodmove | Mysql client and server are working ok but when I compile the asterisk-addons it does not recognise that mysql is installed |
| 20:30.51 | webtech_m33 | you need the mysql dev libs |
| 20:30.58 | webtech_m33 | so it can find mysql.h |
| 20:31.14 | webtech_m33 | not sure what rpm in redhat that package is |
| 20:31.31 | webtech_m33 | debian it's libmysqlclient15-dev |
| 20:31.39 | goodmove | I used rpm -qa | grep mysql and I see |
| 20:32.03 | Shaun2222 | rpm -qa|grep -i mysql |
| 20:32.09 | *** join/#asterisk ruied (n=ruied@bl7-221-245.dsl.telepac.pt) |
| 20:33.31 | goodmove | I see the following packages |
| 20:34.16 | blaylock | jbot: shut up |
| 20:34.16 | jbot | yes, master blaylock |
| 20:34.25 | Shaun2222 | mysql.h comes from the mysql-devel pacakge on Redhat based distros normally. |
| 20:34.25 | *** join/#asterisk horsesgofaster (n=chatzill@ool-44c4e9ea.dyn.optonline.net) |
| 20:34.43 | goodmove | libdbi-dbd-mysql-0.8.1a-1.2.2, mysql-5.0.22-2.1, mysql-server-5.0.22-1.1 |
| 20:34.45 | *** join/#asterisk GreyFoxx (i=greg@out.of.phaze.org) |
| 20:34.57 | Shaun2222 | goodmove: what os? |
| 20:35.27 | *** part/#asterisk horsesgofaster (n=chatzill@ool-44c4e9ea.dyn.optonline.net) |
| 20:35.30 | goodmove | the os is Redhat linux Enterprise 5.0 |
| 20:36.01 | Shaun2222 | i think RHEL 5 uses yum now... i wont pay for RH so i use CentOS 5 which is the same.... |
| 20:36.05 | Shaun2222 | yum install mysql-devel |
| 20:36.30 | goodmove | Shaun2222, I am assuming that the libdbi-.... package is the development library.. |
| 20:36.41 | Shaun2222 | no |
| 20:36.47 | Shaun2222 | thats for perl |
| 20:37.00 | Shaun2222 | mysql-devel is what you want |
| 20:37.04 | goodmove | Ok |
| 20:38.12 | goodmove | do you have any idea of how I could get mysql-devel using yum? |
| 20:38.26 | goodmove | I mean commands.. |
| 20:38.37 | Shaun2222 | yum install mysql-devel |
| 20:38.57 | goodmove | thank you shaun2222 |
| 20:39.09 | webtech_m33 | after that is install then try install asterisk-addons |
| 20:39.13 | webtech_m33 | and it should install |
| 20:40.33 | goodmove | Webtech, Thanks for the tip will do.. |
| 20:40.55 | Shaun2222 | goodmove: it will put mysql.h in /usr/include/mysql/mysql.h |
| 20:41.12 | Shaun2222 | cant remember if you have to specify that when building the addons... |
| 20:41.19 | Shaun2222 | you'll find out soon enough when you try to build. |
| 20:42.24 | Shaun2222 | one of the asterisk packages had some issues finding somthign where i had to specify the path but i think that was only a 64bit issue and with /usr/lib |
| 20:42.39 | goodmove | Thanks a million guys, I am off to try your suggestions. my yum commnad failed because I did not register my RedHat. I will take care of this then proceed to tryout what you have suggested |
| 20:43.19 | Shaun2222 | ya, you'll have to register... |
| 20:45.41 | clickonce | Ey, can someone record voice message for me? I hate recording them on my own. |
| 20:46.06 | webtech_m33 | i think you can use sox and record your own |
| 20:46.18 | webtech_m33 | or use some tts to make files |
| 20:46.30 | clickonce | I know how to... |
| 20:46.40 | clickonce | I just don't like my voice :P |
| 20:46.42 | stansmith | i use the cepstral TTS allison smith voice, its pretty good |
| 20:46.47 | stansmith | i have her say " |
| 20:47.01 | clickonce | Windows XP Text-To-Speech :) |
| 20:47.05 | webtech_m33 | yeah tts |
| 20:47.40 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:49.42 | webtech_m33 | anyone good at scripting with Asterisk Manager Interface |
| 20:49.51 | clickonce | Darn, I only have Sam. |
| 20:51.30 | webtech_m33 | i have this perl mod ... http://search.cpan.org/~jamesgol/asterisk-perl-0.10/lib/Asterisk/Manager.pm but i am not sure how to get Action => 'SIPPeers' to work |
| 20:52.03 | *** part/#asterisk GreyFoxx (i=greg@out.of.phaze.org) |
| 20:53.46 | stansmith | webtech_m33: did you look at manager-test.pl that came with the package? |
| 20:54.12 | webtech_m33 | yeah .. it works for events and calls, but i want to get all of the sip peers |
| 20:54.28 | webtech_m33 | and put them into an array then dump it to a web page |
| 20:57.02 | webtech_m33 | the manager-test.pl makes a connection at stays connected in a eventloop |
| 20:57.23 | webtech_m33 | i want to get in, grap all of the SIPPeers and get out |
| 20:58.38 | stansmith | connect, grab the sippeers and disconnect, whats the problem? |
| 20:58.42 | webtech_m33 | i shoot an email off the to writer dude to see if he can help me |
| 20:59.00 | webtech_m33 | the commands i do to get the sippeers doesn't work |
| 20:59.10 | webtech_m33 | rint STDERR $astman->sendcommand( Action => 'SIPPeers'); |
| 20:59.20 | webtech_m33 | with p in the front |
| 20:59.39 | webtech_m33 | i am not sure how to make it haappen |
| 20:59.48 | stansmith | have you ever used the perl AGI before? |
| 20:59.55 | webtech_m33 | twice |
| 20:59.59 | stansmith | hm.. |
| 21:00.48 | *** join/#asterisk arguile (n=arguile@KTNRON06-1242488957.sdsl.bell.ca) |
| 21:01.40 | stansmith | webtech_m33: maybe just $astman->sendcommand( Action => 'SIPPeers' ); |
| 21:01.41 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 21:01.46 | stansmith | i havent used the perl AGI for AMI |
| 21:01.47 | defsdoor | I've installed some new sounds (a lovely sounding chap) into sounds/en_GB and he sounds great apart from when extension numbers are read out in voice mail where it switches back to Allison - I have .pcm files in digits though |
| 21:02.15 | *** join/#asterisk DrkShadow (n=chatzill@host-72-175-240-62.static.bresnan.net) |
| 21:04.41 | *** join/#asterisk Nasra (n=maxshipp@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 21:06.00 | *** join/#asterisk maszlo (n=reckenro@65.223.240.146) |
| 21:06.03 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:06.03 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 21:07.36 | maszlo | i am looking to get my cdr problem figured out. it is not inserting records into the database anymore, it was working at one time and then after a update it no longer works. |
| 21:07.47 | *** join/#asterisk lunaphyte__ (n=lunaphyt@70.90.148.3) |
| 21:08.08 | webtech_m33 | what did you update? |
| 21:08.12 | codefreeze | maszlo: what version of asterisk? |
| 21:08.13 | maszlo | it does not show that the insert failed in /var/log/asterisk/full |
| 21:08.51 | maszlo | we are running trixbox |
| 21:09.00 | russellb | ~trixbox |
| 21:09.00 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 21:09.37 | maszlo | i hate when bots yell at me |
| 21:10.14 | maszlo | is there a file that enables the cdr? |
| 21:10.36 | defsdoor | http://pastebin.ca/913123 < is this a big ? It's playing the sounds out of sounds/en_GB for all apart from digits |
| 21:10.44 | defsdoor | s/big/bug |
| 21:10.48 | codefreeze | maszlo: yes, the config files, with the appropriate i/f detected during the "configure" run... |
| 21:12.11 | codefreeze | maszlo: the config files are in /etc/asterisk, usually. But if trixbox does a lot of twiddling, that may be only part of the story. For instance, there's no /var/log/full stuff with asterisk plain. |
| 21:12.36 | maszlo | it seems at though it is not trying. we had a incorrect password before and it erroring in the logs not it does not do that. |
| 21:12.54 | maszlo | seem as if it is not enabled |
| 21:13.19 | webtech_m33 | my guess would be that the cdr_addon_mysql.so is not loading.. i would guess it's missing |
| 21:13.46 | webtech_m33 | but i couldn't tell you where trixs hides it |
| 21:14.13 | maszlo | that would be loaded when asterisk starts? |
| 21:14.20 | webtech_m33 | yeah |
| 21:14.31 | webtech_m33 | it's in the modules.conf |
| 21:14.35 | webtech_m33 | sometimes |
| 21:14.52 | maszlo | i will give it look |
| 21:14.59 | webtech_m33 | mine auto loads .. but i didn't compl the addons |
| 21:15.06 | webtech_m33 | so it didn't load |
| 21:17.12 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 21:17.55 | maszlo | i found the cdr_addon_mysql.so it is not in the modules.conf |
| 21:18.19 | webtech_m33 | you may have to add it |
| 21:18.31 | webtech_m33 | but if i am wrong asterisk will CRASH on boot |
| 21:18.35 | webtech_m33 | or a restart |
| 21:19.09 | maszlo | has autoload=yes would that bring in them from a directory or somthing |
| 21:20.05 | maszlo | there are actually zero loads in the files, they are al noload |
| 21:20.14 | webtech_m33 | yeah the autoloads all of the .so |
| 21:20.25 | webtech_m33 | so you are missing the file |
| 21:20.37 | webtech_m33 | i would check with the trixs chat and see if they know |
| 21:21.18 | Nasra | aren't they supposed to give you support? |
| 21:22.07 | webtech_m33 | i know you can load trixs with out support |
| 21:22.09 | [TK]D-Fender | LOL |
| 21:22.48 | codefreeze | Nasra: I'd imagine its just like here; you toss in your problem, and pray someone takes pity on you... |
| 21:23.05 | cmantito | yay for pity |
| 21:23.06 | [TK]D-Fender | Support? Trixbox? This isn't a WondreBra you know, more like Kleenex stuffing :p |
| 21:23.09 | maszlo | haha |
| 21:23.32 | webtech_m33 | maszlo> i would check with trixs, if you did an update, and now it doesn't work.. sounds like a bug to me |
| 21:23.40 | maszlo | we dont pay for support, it was loaded on a rhino system we bought |
| 21:23.43 | [TK]D-Fender | maszlo: Yup, you're up a creek. You've got distro blow-up issues and nobody here wants to hear about them. |
| 21:24.31 | maszlo | thats a blunt way of putting it out there |
| 21:24.42 | [TK]D-Fender | maszlo: I thought this said it all : |
| 21:24.44 | [TK]D-Fender | ~freepbx |
| 21:24.44 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 21:24.46 | [TK]D-Fender | ~trixbox |
| 21:24.47 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 21:25.14 | [TK]D-Fender | maszlo: Most would have gotten the "Why am I here?" a little quicker. |
| 21:25.33 | [TK]D-Fender | maszlo: When in doubt start over with your install and say a prayer |
| 21:25.52 | [TK]D-Fender | maszlo: Because you have already sold your soul to the lowest bidder |
| 21:26.02 | maszlo | its not like calls are not going through, just the reports are not working |
| 21:26.21 | [TK]D-Fender | maszlo: Well you know where to go. Best of luck with that. |
| 21:26.34 | [TK]D-Fender | alrighty, checkout time, BBL |
| 21:27.17 | maszlo | i have had no prior experience with any sort of pbx, i dont feel it was the lowest bidder.. i do appreciate the help |
| 21:27.42 | maszlo | thanks again |
| 21:27.48 | webtech_m33 | sorry i can't help more.. when i first load the cdr for mysql |
| 21:28.02 | webtech_m33 | it took me a few times to get it to load |
| 21:28.13 | cmantito | I had no problem with cdr_addon_mysql |
| 21:28.32 | webtech_m33 | yeah i forgot to do a ./configure |
| 21:28.40 | webtech_m33 | so it didn't find mysql.h |
| 21:28.42 | stansmith | silly webtech_m33 |
| 21:28.48 | webtech_m33 | my bad |
| 21:28.50 | cmantito | biggest thing I can suggest is make sure the module is loaded |
| 21:29.06 | webtech_m33 | or try updating asterisk-addons |
| 21:37.09 | eric2 | I'm with voip carrier A and am trying to call someone with voip carrier B, with a simple Dial(SIP/4165551234) I get a fast busy - congestion, what needs to change? |
| 21:37.37 | Mavvie | eric2: you didn't specifiy where to find that SIP host. |
| 21:38.33 | webtech_m33 | 4165551234@carrierb.com |
| 21:38.36 | eric2 | something like: Dial(SIP/4165551234@somePlace.com |
| 21:38.38 | eric2 | ah |
| 21:38.53 | *** join/#asterisk husimon (n=nhuisman@aeko.IfA.Hawaii.Edu) |
| 21:38.58 | eric2 | how will I know what carrier that other person is with if I only have a number? |
| 21:39.00 | husimon | does anyone know much bout xml services on 79xx? |
| 21:39.04 | husimon | phones |
| 21:39.20 | husimon | I was trying out the directories.xml file and it's totally useless, only lets you have 32 entries |
| 21:39.28 | husimon | wanted to duplicate that via an xml service instead. |
| 21:39.36 | Shaun2222 | anybody know how i can acheive waht i'm trying to do? http://lists.digium.com/pipermail/asterisk-users/2008-February/206103.html |
| 21:41.08 | eric2 | what if I'm routing all my calls out through my virtual pri, then my pri provider doesn't have the various carriers? |
| 21:42.34 | husimon | anyone use xml for a directory services on cisco 7940s? |
| 21:43.25 | Nugget | I did a long time ago, but gave up because in practice it wasn't very useful. |
| 21:43.59 | Nugget | why would I want to use my phone to look things up when I have my computer, right there, a foot to the left of the phone? |
| 21:44.16 | Nugget | navigating the cisco menus is more of a pain than dialing a number. |
| 21:44.40 | Nugget | and if you really want to get fancy you can get asterisk to dial a call from a web page and transfer it to your phone, which is a lot more useful |
| 21:45.44 | husimon | Nugget, yeah I think you're probably right |
| 21:45.47 | husimon | Nugget, oh well |
| 21:46.16 | husimon | Nugget, unfortunately the menu is so clunky with just a number pad |
| 21:46.22 | Nugget | yeah |
| 21:46.24 | husimon | hard to type in someones name |
| 21:46.37 | husimon | now what would be neat |
| 21:46.42 | Nugget | and, at least on the sip firmware, the cisco implementation is REALLY flaky |
| 21:46.45 | husimon | is a predictive text auto completion thing |
| 21:47.23 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:48.00 | stansmith | deeper '07 what can i do? |
| 21:51.00 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 21:51.12 | stansmith | Strom_C: sup buddy!!! |
| 21:51.29 | stansmith | its me, your best friend! |
| 21:52.19 | husimon | laugh |
| 21:55.34 | husimon | mmm this black bean chili is so good :) |
| 21:55.53 | stansmith | husimon: o yea? |
| 21:55.56 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 21:56.31 | husimon | yeah I made it last night |
| 21:57.04 | stansmith | can i have some? |
| 21:57.12 | *** join/#asterisk ThatKidKel (i=user@cm-64-221-169-156.dhcp.southerncoastalcable.net) |
| 21:57.24 | husimon | sure let me open an IAX trunk and pour some in. |
| 21:57.26 | *** join/#asterisk bsdwarrior (n=mfahey@fahey.enter.net) |
| 21:57.29 | *** part/#asterisk ThatKidKel (i=user@cm-64-221-169-156.dhcp.southerncoastalcable.net) |
| 21:57.40 | stansmith | lol |
| 21:57.56 | bsdwarrior | I can't get a phone to register with asterisk from a remote location. any know if I need to set the proxy address ,etc? |
| 21:58.39 | husimon | bsdwarrior, I'd guess it's an issue of firewall ports not being open |
| 21:58.51 | *** join/#asterisk dexpdx (n=dexpdx@66-162-134-242.static.twtelecom.net) |
| 21:59.17 | bsdwarrior | husimon, I dont remember the setting to tell the phone ip to connect to. |
| 21:59.36 | dexpdx | <PROTECTED> |
| 21:59.43 | dexpdx | anyone ever seen that one before? |
| 22:00.05 | stansmith | bsdwarrior: are you using SIP? |
| 22:00.16 | husimon | man my office mates are not going to like me later |
| 22:00.20 | husimon | ....chili |
| 22:00.22 | stansmith | husimon: lol |
| 22:00.44 | husimon | i remember eating chili for like 3 days straight |
| 22:00.49 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 22:00.54 | *** join/#asterisk mikemking (n=mking@pool-72-78-140-9.phlapa.east.verizon.net) |
| 22:00.57 | husimon | i realized I was eating too much of it when my pee started to smell like cumin |
| 22:01.54 | *** part/#asterisk JayTee52 (n=jforde@207-67-84-188.static.twtelecom.net) |
| 22:02.35 | mikemking | I'm using realtime queues, and dynamic members (or realtime static members) and I am trying to figure out how I can not send a call to a member who is already on a queue call. Can anyone help? |
| 22:02.40 | *** join/#asterisk drfreeze (n=Jim@207.191.114.82) |
| 22:02.47 | drfreeze | Anyone know a reason I am getting Congestion problems when I have open lines? |
| 22:03.28 | mikemking | drfreeze: any log errors? |
| 22:03.45 | drfreeze | mikemking: lemme check |
| 22:04.52 | tzanger | haha |
| 22:05.07 | tzanger | <PROTECTED> |
| 22:05.21 | mikemking | I know that I can disable call waiting on the device, but that is not an option in this case |
| 22:06.26 | drfreeze | there are a few data format errors that got emails |
| 22:07.20 | mikemking | drfreeze: what are the errors? Hearing fast-busy is often a sign of an error, not just of congestion |
| 22:09.16 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:09.22 | drfreeze | Feb 21 12:03:25 talky sendmail[22091]: m1LI3Jij022089: to=<user@domain.com>, ctladdr=<root@localhost.localdomain> (0/0), delay=00:00:0 |
| 22:09.25 | drfreeze | 6, xdelay=00:00:05, mailer=esmtp, pri=120427, relay=mail.domain.com. [1.2.3.4], dsn=5.6.0, stat=Data format error |
| 22:09.26 | webtech_m33 | well l8r all |
| 22:10.09 | *** part/#asterisk webtech_m33 (i=webtech@webtech.m33access.com) |
| 22:10.41 | dexpdx | drfreeze: those are mail errors |
| 22:10.47 | dexpdx | not asterisk errors |
| 22:10.57 | drfreeze | yeap. no asterisk errors other than congestion |
| 22:11.13 | drfreeze | the event_log is empty |
| 22:11.15 | mikemking | freeze: have you watched the asterisk CLI while making a call? |
| 22:11.21 | dexpdx | asterisk -vdgr |
| 22:11.23 | drfreeze | yes |
| 22:11.28 | dexpdx | Congestion() |
| 22:11.32 | dexpdx | check your dialplan |
| 22:11.52 | mikemking | and what messages go by when you get congestion? |
| 22:12.03 | drfreeze | the uptime is 20 weeks |
| 22:12.59 | mikemking | if you hear congestion, you should see a warning or error in the CLI. Otherwise, Congestion() was called by the dialplan |
| 22:13.54 | drfreeze | have had a few instances of congestion in the past - 600 logged cases over 1.5 years - 212 occurances today |
| 22:14.22 | drfreeze | the staff said they could not make outgoing calls - only one person at a time |
| 22:14.23 | mikemking | are these inbound or outbound calls? |
| 22:14.26 | husimon | drfreeze, i'd check your zaptel device |
| 22:14.28 | drfreeze | otehrs got a fast busy |
| 22:14.28 | [hC] | What is it about my sip peers that makes them reset to username = s after a call comes in from them? sigh. |
| 22:14.46 | drfreeze | husimon: zaptel looked good. no errors in status |
| 22:14.55 | mikemking | freeze: are you sending calls out zap or a sip trunk? |
| 22:14.56 | drfreeze | mikemking: outbound |
| 22:15.09 | husimon | drfreeze, i'd definitely load up the cli and turn on debug and make two calls and see what happens |
| 22:15.28 | drfreeze | husimon: k. but it seems to be working now |
| 22:15.45 | husimon | drfreeze, do you log your cli to a file? |
| 22:15.57 | drfreeze | I'll watch it for a bit. They don't seem to call when the problem is occuring |
| 22:16.02 | drfreeze | husimon: yes |
| 22:16.06 | drmessano | I got a $12 bluetooth USB thingo from CompUSA |
| 22:16.09 | mikemking | that's always the way :-) |
| 22:16.14 | husimon | drmessano, what's it do? |
| 22:16.26 | husimon | drfreeze, odd that they don't call when the phones don't work ;) |
| 22:16.26 | drmessano | USB Bluetooth Adapter |
| 22:16.41 | drmessano | Where is Stevegoma when you need him |
| 22:18.43 | *** join/#asterisk GlobeTrotter (n=eric@196.40.26.98) |
| 22:19.44 | drfreeze | on sip show channels, does Tx: ACK mean the line is being used? |
| 22:24.49 | drfreeze | mikemking: http://pastie.textmate.org/private/egclwqns1huzyqpvjkn4q |
| 22:24.56 | drfreeze | husimon: see pastie above |
| 22:25.56 | Mavvie | putnopvut: thanks! |
| 22:26.24 | drfreeze | 50.97 is ext 510 |
| 22:26.55 | putnopvut | Mavvie: are you the reporter on issue 11917? |
| 22:27.00 | Mavvie | putnopvut: yes I am. |
| 22:27.00 | drfreeze | So, does Got SIP response 500 "Internal Server Error" back from 192.168.50.97 mean that my phone is going bad? |
| 22:27.05 | putnopvut | Mavvie: Ah, okay :) |
| 22:27.15 | *** join/#asterisk Beave (n=beave@DCC.SEND.startkeylogger.000.telephreak.org) |
| 22:27.29 | mikemking | drfreeze: might be your PSTN connection |
| 22:27.50 | drfreeze | ah, so maybe the adtran board is going bad |
| 22:29.02 | mikemking | or your carrier is congested ;-) |
| 22:29.47 | drfreeze | sure |
| 22:30.23 | drfreeze | mikemking: what do you think the internalserver error from the phon emeans? |
| 22:32.53 | mikemking | drfreeze: not sure on that one |
| 22:34.18 | signius | is it a 500 server error and what phones you using ? |
| 22:34.43 | drfreeze | signius: polycom 501 |
| 22:35.31 | signius | http://bugs.digium.com/view.php?id=3798 |
| 22:35.43 | signius | does that maybe point you in the right direction ? |
| 22:35.45 | *** join/#asterisk lonebobwhite (n=rleblanc@74.231.171.198) |
| 22:38.33 | signius | Have you got the buddy watch setting activated ? |
| 22:38.57 | drmessano | Anyone know what chipset is needed to make chan_mobile work? |
| 22:38.59 | drmessano | Well |
| 22:39.08 | drmessano | What kind of bluetooth chipset is supported |
| 22:39.10 | Qwell | drmessano: any? |
| 22:39.17 | drmessano | ? Could be |
| 22:39.19 | Qwell | if Linux supports it, so will chan_mobile |
| 22:39.28 | drmessano | I guess I asked that stupidly |
| 22:39.33 | signius | There seems to be an article for trixbox that seems to be related to this and polycom 501s |
| 22:39.34 | Qwell | there are only like 3 major ones, and iirc, Linux supports them all |
| 22:39.40 | drmessano | Was trying to be simple about it |
| 22:39.41 | drmessano | Ok |
| 22:39.52 | signius | http://www.trixbox.org/forums/trixbox-forums/help/constantly-getting-incoming-call-got-sip-response-500-errors |
| 22:40.04 | Qwell | signius: are you using trixbox? |
| 22:40.07 | drmessano | I got this $12 one from CompUSA and I was gonna go back and get another |
| 22:40.18 | Qwell | or, drfreeze rather |
| 22:40.24 | Qwell | drmessano: another Doctor! |
| 22:40.34 | drmessano | ZOMG |
| 22:40.55 | drmessano | drfreeze, you are trying my patients |
| 22:41.10 | drmessano | :) |
| 22:41.20 | drfreeze | signius: hmm |
| 22:41.49 | drmessano | Dinner time, brb |
| 22:42.30 | drfreeze | drmessano: not using trixbox |
| 22:44.00 | signius | i know you didnt say you were using trixbox but trixbox does use asterisk and it might be related to the same element is what i was thinking |
| 22:45.24 | drfreeze | looks like the server error is harmless and can be fixed with rebooting the phones |
| 22:45.36 | *** join/#asterisk lonebobwhite (n=rleblanc@74.231.171.198) |
| 22:45.50 | drfreeze | thanks everyone for your help |
| 22:47.29 | Mavvie | putnopvut: The original DTMF debug was invalid because of the path it took. I tried to strike it out (HTML wise) but it kind of failed. |
| 22:47.48 | putnopvut | Mavvie: ah, okay. I'm not sure why it works for me but not for you. |
| 22:48.04 | putnopvut | There's nothing odd in your DTMF debug messages either. |
| 22:50.02 | Mavvie | putnopvut: that is with asterisk 1.4.18 you did it with? |
| 22:50.13 | putnopvut | Correct. |
| 22:50.26 | putnopvut | I tried both the tagged 1.4.18 and the latest SVN revision of 1.4. |
| 22:50.27 | *** join/#asterisk antonyo14 (n=tony@209.101.229.196) |
| 22:50.28 | Mavvie | And with an FXO or FXS card or with a PRI ? |
| 22:50.34 | *** join/#asterisk nvrpunk (n=root@81.90.21.227) |
| 22:50.36 | putnopvut | FXS port. |
| 22:50.59 | nvrpunk | question, some phone numbers register our dtmf and some dont |
| 22:51.09 | nvrpunk | would that be a setting at the pstn gateway? |
| 22:51.15 | nvrpunk | or possibly something on our side |
| 22:51.15 | antonyo14 | I am wondering if anyone has any experience with static when using speakerphone |
| 22:51.17 | nvrpunk | causing it |
| 22:51.56 | Mavvie | putnopvut: It was an PRI card for me. I just did a test with a SIP phone and it indeed worked fine. |
| 22:52.21 | putnopvut | Hmmm...interesting. |
| 22:53.04 | putnopvut | I wonder if anyone here has a PRI card set up that they could test on...? |
| 22:54.09 | Mavvie | putnopvut: I just tried it on the 1.2 instance we have still running and it worked fine there. Also with a PRI. |
| 22:54.39 | putnopvut | Mavvie: are you the one who sent an e-mail to the asterisk-dev list about this issue or is that someone else? |
| 22:54.56 | Mavvie | putnopvut: that was me. |
| 22:55.11 | putnopvut | Okay, because if it was someone else, I was going to ask if he was using a PRI. |
| 22:55.20 | putnopvut | But obviously he is :) |
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| 23:12.03 | drmessano | odd |
| 23:12.10 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 23:12.11 | sun_moon | hello dr |
| 23:12.14 | drmessano | howdy |
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| 23:12.27 | sun_moon | long time no hear howz atlanta |
| 23:12.40 | drmessano | Dunno.. Im in Augusta |
| 23:13.10 | sun_moon | sorry I meant to type augusta and instead typed atlanta |
| 23:13.23 | drmessano | heh |
| 23:13.36 | sun_moon | so whats going on with the latest release ofasterisk ? |
| 23:14.03 | drmessano | I dunno.. I think it still makes phone calls |
| 23:14.19 | sun_moon | :) |
| 23:14.25 | sun_moon | that was a good one |
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| 23:20.07 | drmessano | $6.60 for a USB Bluetooth |
| 23:20.15 | drmessano | Shipped |
| 23:20.16 | sun_moon | oh cool |
| 23:20.30 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) [NETSPLIT VICTIM] |
| 23:20.33 | drmessano | $1.00 + $5.60 S&H |
| 23:21.16 | drmessano | Issue #2 |
| 23:21.21 | Wayhigh | any idea what chipset? some of them you can do the handsfree with |
| 23:21.32 | drmessano | Well |
| 23:21.35 | drmessano | I dunno |
| 23:22.10 | drmessano | How does one USB do the handsfree and one not? |
| 23:22.28 | Wayhigh | some don't support the profile in their drivers for some reason |
| 23:22.37 | Qwell | drivers? linux. |
| 23:23.16 | Wayhigh | the only reason it matters is because if you have the handsfree profile setup you can bridge your cell phone calls |
| 23:23.27 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 23:24.11 | Wayhigh | I'm trying to find a way of having a verizon prepaid that wont cost $30/mo to use it daily.. |
| 23:24.55 | *** join/#asterisk terracon (n=greisky@CPE0050da822b70-CM0012254076d6.cpe.net.cable.rogers.com) |
| 23:25.00 | Wayhigh | then I could bridge the prepaid to my in-network verizon phones and if they get angry you're only out a prepaid :) |
| 23:25.23 | drmessano | I was thinking it would be cool to use Chan_mobile with an extra phone.. blah blah blah...... and then I realized the utility of having it work with MY phone when it's sitting here |
| 23:29.28 | *** join/#asterisk Dovid (n=Dovid@bzq-79-178-182-3.red.bezeqint.net) |
| 23:29.35 | Dovid | hi. Can anyone help me with this ? |
| 23:29.35 | Dovid | http://pastebin.ca/913323 |
| 23:30.34 | Mavvie | Dovid: if you explain what it is, what happens and what you expect to happen. |
| 23:31.00 | Dovid | lol |
| 23:31.20 | Dovid | I am have an E1 and I am unable to make calls. I know that is a broad statement. |
| 23:31.28 | Dovid | i get back congestion from my carrier |
| 23:31.44 | Mavvie | Dovid: do they require a valid CallerID? |
| 23:31.57 | Mavvie | alling Number (len= 4) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) |
| 23:31.58 | Mavvie | Presentation: Number not available (67) '' ] |
| 23:32.08 | Dovid | ahhhhhhhhhhhhh |
| 23:32.16 | Dovid | i will check the CID |
| 23:32.17 | Dovid | one sec |
| 23:34.08 | Dovid | nope. CID is being set |
| 23:34.11 | [hC] | ~book |
| 23:34.12 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 23:34.26 | drmessano | If only there was a book |
| 23:34.34 | Mavvie | so, give a new trace. |
| 23:34.57 | mvanbaak | Qwell: we need drivers on bsd |
| 23:35.07 | Dovid | ok |
| 23:36.25 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 23:36.48 | Corydon76-vcch | mvanbaak: you find me someone local to write the drivers, and I'll be happy to loan him the cards |
| 23:37.17 | drmessano | Wow |
| 23:37.27 | drmessano | http://www.virtualhosting.com/blog/2008/wide-open-voip-top-50-open-source-voip-apps/ |
| 23:37.43 | drmessano | Asterisk was the top PBX, and somehow down the list a few, Trixbox showed up.. |
| 23:37.50 | Mavvie | kernel work. my favourite! seven reboots in twelve minutes, and you still have no idea why even the simplest example program doesn't work. |
| 23:37.57 | drmessano | So either someone is a dumbass, or Asterisk made the list twice |
| 23:38.03 | Dovid | http://pastebin.ca/913334 |
| 23:38.39 | Dovid | i wonder what this is: [Feb 22 01:35:38] WARNING[7640] pbx.c: Ignoring entry 'CALLERID(num)XXXXXXXXXXwith no = (and not last 'options' entry) |
| 23:38.47 | Dovid | oh |
| 23:38.48 | Dovid | geex |
| 23:39.15 | drmessano | Evolution PBX as well |
| 23:39.26 | Dovid | hmm. now i get a new error |
| 23:39.29 | drmessano | Unless they change the core |
| 23:40.08 | Mavvie | Ext: 1 Cause: Unallocated (unassigned) number (1) |
| 23:40.10 | Corydon76-vcch | drmessano: also inaccurate, but the that's the case in any of the "top *" articles out there |
| 23:40.18 | Dovid | > Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) |
| 23:40.18 | Dovid | > Presentation: Number not available (67) 'XXXXXXXXXX' ] |
| 23:40.20 | drmessano | Agreed.. |
| 23:40.25 | drmessano | They're all pretty bogus |
| 23:40.26 | Corydon76-vcch | drmessano: how exactly is Sangoma "open hardware"? |
| 23:40.28 | Dovid | sorry for the 2 lines |
| 23:40.40 | Mavvie | Dovid: but still: Presentation: Number not available (67) '' |
| 23:41.08 | Dovid | Mavvie: What does that mean ? |
| 23:41.09 | Corydon76-vcch | I know of zero commercial vendors who have "open hardware" |
| 23:41.19 | drmessano | True.. The fact they write drivers for open source systems doesnt exactly fit |
| 23:41.26 | Mavvie | Dovid: that you didn't set tthe callerid. |
| 23:41.49 | *** join/#asterisk fbnts (n=thomas@host86-128-11-159.range86-128.btcentralplus.com) |
| 23:41.51 | Dovid | but i DID :( :( :( :( |
| 23:41.57 | Dovid | i guess they dont like it |
| 23:42.01 | drmessano | and EIKGA |
| 23:42.08 | drmessano | EKIGA anyone? |
| 23:42.10 | Mavvie | Dovid: read line 5. |
| 23:42.15 | Mavvie | and read line 6. |
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| 23:42.36 | fbnts | Hi, has anyone use the ExtensionState Manager AGI function? I am trying to check the status of our SIP Phones but it always returns 0 regardless of if the phone is in use or ringing |
| 23:42.57 | Dovid | Mavvie: How do I block my CID on a PRI ? |
| 23:43.04 | Dovid | does that depend on the carrier ? |
| 23:43.12 | Corydon76-vcch | And describing Digium as "one of the leading providers" is dishonest, as well. Digium is the progenitor of Asterisk. Without Digium, there would be no Asterisk. |
| 23:43.24 | Mavvie | Dovid: SetCallerPres(prohib) |
| 23:43.42 | Dovid | thnx |
| 23:43.58 | Dovid | i am using Set(CALLERID(num)=XXXXXXXXX) is that correct for a PRI ? |
| 23:44.35 | Mavvie | depends on if you replac ethe X's with a valid number or not. |
| 23:44.36 | Dovid | seems they dont like blcked CID |
| 23:44.37 | Dovid | Presentation: Presentation prohibited of network provided number (35) '' ] |
| 23:44.49 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) |
| 23:45.01 | Dovid | Mavie: it is valid. just wonderd if that was correct (not that it shouldnt be but maybe for a PRI.........) |
| 23:45.06 | drmessano | HA |
| 23:45.11 | drmessano | "Digium is one of the leading providers of Asterisk’s open source PBX software" |
| 23:45.15 | drmessano | Yeah |
| 23:45.18 | drmessano | or "The one" |
| 23:45.24 | Mavvie | Dovid: first issue is to make the basics work. Later on, you can do funky stuff. |
| 23:46.00 | drmessano | This list was probably make up by one of those Skype using losers over on TMCnet |
| 23:46.10 | drmessano | "Asterisk not supports SIP" |
| 23:46.14 | drmessano | "Asterisk now supports SIP" |
| 23:46.26 | drmessano | and other highly riveting articles |
| 23:46.57 | drmessano | "Facebook's next big thing: VoIP" Oh, pardon, there's been no VoIP apps for facebook yet? |
| 23:47.54 | Dovid | Mavvie: goto give them a ring tomorrow |
| 23:48.47 | drmessano | and lets not forget: |
| 23:48.52 | drmessano | "MobiCents is billed as “the most popular Open Source Service Logic Execution Environment (JSLEE) and SIP Application Server for the Java platform." |
| 23:48.53 | Mavvie | Dovid: they will tell you: you're not sending the caller id. |
| 23:49.57 | drmessano | Thats a way to have a niche |
| 23:51.06 | drmessano | "HappyClownPBX is the fastest open source asterisk based PBX branded specifically for use in circuses, childrens hospitals, and as a mobile application platform for pedo's in red vans with 'Free candy' painted on the side" |
| 23:51.09 | drmessano | Nice niche :) |
| 23:51.34 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 23:52.08 | drmessano | I'm easily the most qualified doctor in here that has an italian last name |
| 23:52.10 | drmessano | O.O |
| 23:52.57 | Dovid | Mavvie: And I am |
| 23:53.09 | Dovid | exten => _[0-9*#]!, 1, Set(CALLERID(num)=0788187123) |
| 23:53.11 | Mavvie | Dovid: not according to the last paste you did. |
| 23:53.14 | *** part/#asterisk beek (n=klinebl@65.211.106.243) |
| 23:53.27 | Dovid | oh last paste was blocked. I changed it back |
| 23:53.45 | Dovid | now I get the error that I got b4 (67) |
| 23:53.57 | drmessano | Qwell: I have 3 of those $6 Bluetooth USB DONGLES coming, so I guess I am gonna see how this crap works |
| 23:58.13 | Dovid | Mavvie: Thanks for the help |
| 23:59.26 | *** join/#asterisk SteveTotaro (n=root@pool-71-166-102-100.bltmmd.east.verizon.net) |
| 23:59.56 | SteveTotaro | I am such a nice guy |
