| 06:07.09 | *** join/#asterisk jbot (i=ibot@rikers.org) |
| 06:07.09 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0.5 (2009/01/23), 1.4.23.1 (2009/01/23), *-Addons 1.6.0.1 (2008/12/02), 1.4.7 (2008/06/04), dahdi-linux 2.1.0.3, dahdi-tools 2.1.0.2 (2008/12/18), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev -=- jbot is back! |
| 06:07.26 | beherit | [TK}-Fender- I have read an article about it. |
| 06:07.40 | SwK | using asterisk? |
| 06:07.47 | beherit | SwK: yes. |
| 06:07.50 | SwK | maybe openser or something else |
| 06:07.56 | SwK | URL? |
| 06:08.04 | beherit | wait let me get it |
| 06:08.10 | beherit | its in my other machine |
| 06:10.37 | beherit | SwK: search in google asterisk cluster and database and result is second link from the top .read the PDF article from astricon.net. |
| 06:16.11 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 06:19.31 | [TK]D-Fender | ok, checkout time... |
| 06:19.34 | [TK]D-Fender | later all |
| 06:20.34 | Khratos | tc |
| 06:22.33 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 06:22.33 | *** mode/#asterisk [+o denon] by ChanServ |
| 06:23.56 | Khratos | is going to sleep |
| 06:24.04 | *** part/#asterisk Khratos (n=Khratos@190.166.129.103) |
| 06:29.42 | kerx_ | weird, it's getting a forbidden, but doesn't register the phone between my trunked SIP peers (two asterisk machines) |
| 06:31.04 | kerx_ | ~book |
| 06:31.05 | jbot | methinks book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
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| 06:44.15 | *** mode/#asterisk [+o denon] by ChanServ |
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| 07:01.21 | *** join/#asterisk jyap (n=jyap@cpe-72-130-222-197.hawaii.res.rr.com) |
| 07:02.20 | jyap | is there a common way/best practice way of mapping a DID to a SIP address? |
| 07:03.08 | bmoraca | jyap: exten=>_XXXXXXX,1,Dial(SIP/XXXXX)...were you expecting something else? |
| 07:04.20 | jyap | i guess i was thinking there was some common function/lookup. that method seems 'inelegant'. |
| 07:05.08 | bmoraca | jyap: it's as elegant as you want it to be. a call comes in to a context, you have to tell it what to do. |
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| 07:09.47 | bmoraca | gotta go...bye bye |
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| 08:00.13 | kerx_ | Someone can help me get my IAX2 channel setu pplease? |
| 08:01.31 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 08:07.36 | beherit | if my * setup is realtime, how can i add sippeer and sipuser in * CLI? or I have to insert all the information inside the DB? |
| 08:08.50 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
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| 08:18.56 | kerx_ | WARNING[5433]: channel.c:3181 ast_channel_make_compatible: No path to translate from SIP/agent103-09f4d480(4) to Local/103@acdqueue-29eb,2(256) |
| 08:21.09 | Dovid | kerx_": Seem like asterisk can't translate between 2 codecs. |
| 08:21.25 | Dovid | has onyone here used OSLEC with DAHDI ? |
| 08:21.43 | kerx_ | Dovid, is it possible that it may be a problem I am dialing w/ IAX trunk and then sending to a acd queue that is supposed to call a SIP phone? |
| 08:21.55 | kerx_ | is it because it's not possible to do IAX2 <--> SIP ? |
| 08:22.32 | Dovid | nah. that u can do |
| 08:22.38 | Dovid | what codecs r u trying to use ? |
| 08:22.46 | kerx_ | g729 |
| 08:22.57 | kerx_ | oh my ip phone may not do g729 |
| 08:23.00 | kerx_ | x-lite |
| 08:25.24 | kerx_ | Dovid, is this possible? |
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| 09:02.33 | hi365 | is there any way to check if the installed (digium) pri card has hs e/c? |
| 09:02.39 | hi365 | hs=wardware |
| 09:02.48 | hi365 | *hardware |
| 09:03.14 | unasi7 | hi. my asterisk pbx got hacked. any ideas how a user could do a outbound call from p.E. "1634642919" (no sip user!)? |
| 09:03.51 | Dovid | unasi7: Most likely because ur default context in sip.conf or iax.conf has permission to make outbound calls |
| 09:04.01 | mort_gib | unasi7: insecure in sip.conf |
| 09:04.12 | unasi7 | okay.. will check |
| 09:04.40 | mort_gib | unasi7: 1. ALWAYS place your asterisk behind a firewall |
| 09:05.01 | mort_gib | 2. Make sure Sip accounts has to authenticate prior to making calls |
| 09:05.32 | mort_gib | I stumbled over this a while back when a SIP device that was unable to receive a call could quite happily place a call |
| 09:05.56 | mort_gib | -Only there are no way for external SIP connections to be made TO my server |
| 09:06.03 | mort_gib | -Still, not good |
| 09:06.04 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-1cc5bae63b20281e) |
| 09:08.32 | unasi7 | mort_gib, so because im in hurry i do ask some questions: i do have the "secret" on each SIP user. But how do i set up a auth? |
| 09:09.10 | mort_gib | that's not it |
| 09:09.19 | Dovid | unasi7: Even if you have sip passwords if in the same context as your general you have in extensions.conf outbound calling people can send calls |
| 09:09.31 | Dovid | via ur box |
| 09:10.19 | unasi7 | Dovid, so if i change from default context to something other, i will be secure? |
| 09:11.04 | Dovid | yes and no |
| 09:11.11 | Dovid | u should learn what the issues are first |
| 09:11.31 | unasi7 | okay. |
| 09:11.46 | kamlh | hi all |
| 09:12.31 | Dovid | there is a file I think it is called SECUTRITY or something of that sort that explains it |
| 09:17.30 | unasi7 | Dovid, so i will read some papers. your right. but for now, if i remove the Dial() (to outbound) in my default context in extensions.conf, outbound calls can not be made for now? |
| 09:17.41 | *** join/#asterisk _Raptor_ (i=raptorbl@andariel.informatik.uni-erlangen.de) |
| 09:17.48 | unasi7 | (just to fix to get time to solve the whole problem) |
| 09:30.02 | unasi7 | if a place my outbound calling and my local sip clients in the context [longdistance], and leave my incoming registration in the [default] conext. right way? |
| 09:35.30 | hi365 | tzafrir_laptop: sup? is there anyway, in software, to see if my card has e/c installed? |
| 09:35.48 | tzafrir_laptop | depends on the card, I guess |
| 09:36.03 | hi365 | er, digium? |
| 09:36.16 | hi365 | Found a Wildcard: Wildcard TE410P (3rd Gen) |
| 09:39.27 | hi365 | tzafrir_laptop: ^^ |
| 09:40.52 | tzafrir_laptop | Sorry. I don't remember |
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| 09:41.00 | *** join/#asterisk keebler (n=keebler@h199.233.20.98.dynamic.ip.windstream.net) |
| 09:41.22 | keebler | Has anyone used a Linksys SPA9000 as JUST an ATA with Asterisk? |
| 09:41.38 | hi365 | np |
| 09:42.02 | keebler | I've tried copying the same config that I have on my PAP2-NA, but I think I'm missing something. |
| 09:43.51 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 09:44.08 | keebler | No one? |
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| 09:59.58 | frogonwheels | keebler: I found using the syslog capabilities of the PAP2T invaluable for diagnosing what was happening. presumably the SPA9000 has it as well..? |
| 10:00.28 | keebler | Don't see the option. And I don't have a PAP2T, just he -NA |
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| 10:15.24 | frogonwheels | keebler: on my pap2t, you click on Admin Login - an then go to the System tab (in Basic view _or_ advanced view) |
| 10:15.46 | frogonwheels | keebler: you can set a 'syslog server' |
| 10:16.03 | frogonwheels | keebler: of course you hav to enable a syslog server to accept logging from outside. |
| 10:16.26 | keebler | Yeah... and can't check just yet, co worker is on it. |
| 10:16.37 | *** join/#asterisk Rabenklaue (n=Rabe@f051097181.adsl.alicedsl.de) |
| 10:16.42 | keebler | Whats does Mapped SIP port mean? |
| 10:17.29 | Rabenklaue | Hi, does anyone knows a small sip application with few dependencies for linux in order to test asterisk without any further hardware uses? |
| 10:18.34 | frogonwheels | Rabenklaue: kiax ? |
| 10:19.19 | *** join/#asterisk coppice (n=chatzill@201.193.17.210.dyn.pacific.net.hk) |
| 10:19.24 | keebler | frogonwheels: The weird thing is we get a dial tone, it registers. But after a couple seconds we get a busy ssignal |
| 10:19.34 | keebler | and never makes a connection |
| 10:20.03 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 10:20.14 | frogonwheels | keebler: you set up a dialplan on the SPA ? |
| 10:20.27 | keebler | Yeah, copied the one from my PAP2 |
| 10:20.55 | Rabenklaue | frogonwheels: Looks nice, also I don't like the qt dependency on a non-KDE desktop |
| 10:20.57 | frogonwheels | keebler: if you dial as soon as you pickup - does anything happen on the asterisk console>? |
| 10:21.14 | keebler | frogonwheels: no |
| 10:21.16 | frogonwheels | Rabenklaue: is linphone one as well? |
| 10:21.34 | frogonwheels | keebler: does it still show up in sip show peers |
| 10:22.30 | keebler | it shows it |
| 10:22.46 | Rabenklaue | frogonwheels: linphone seems to be exactly what I was looking for. Thanks a lot |
| 10:22.47 | frogonwheels | keebler: sip set debug ?? |
| 10:22.54 | frogonwheels | np Rabenklaue |
| 10:23.07 | keebler | hmm enabled |
| 10:23.13 | frogonwheels | now try? |
| 10:23.20 | keebler | busy signal |
| 10:23.45 | BBHoss | Rabenklaue: Ekiga |
| 10:24.00 | frogonwheels | keebler: you _sure_ you can't get syslog on the SPA9000 ? |
| 10:24.10 | BBHoss | or simply netcat, i'm sure you could use it to test sip, RTP is another story though |
| 10:24.44 | frogonwheels | keebler: did you mess round with the Regional/Advanced Tone settings ? |
| 10:26.03 | keebler | nope. and not sure about the syslog |
| 10:26.19 | frogonwheels | keebler: keep looking - googling seems to hint that it is. |
| 10:26.42 | keebler | yeah |
| 10:26.50 | frogonwheels | http://www.astronomywa.net.au/index.php?view=details&id=30%3Apartial-solar-eclipse-in-perth&option=com_eventlist&Itemid=30 |
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| 10:27.40 | keebler | frogonwheels: It has a syslog server |
| 10:27.48 | keebler | or an option |
| 10:27.54 | keebler | do I set it to log on the asterisk server then? |
| 10:28.02 | frogonwheels | keebler: or wherever. |
| 10:28.25 | keebler | just has a blank space where syslog server is |
| 10:28.37 | frogonwheels | keebler: yep - give it an IP address of a linux box. |
| 10:29.24 | frogonwheels | keebler: which distro? |
| 10:29.34 | keebler | no distro |
| 10:29.44 | keebler | FBSD with fresh install |
| 10:29.51 | frogonwheels | keebler: you need to give syslogd '-r' option |
| 10:30.05 | frogonwheels | assuming it's the same. |
| 10:30.42 | frogonwheels | keebler: I'm suspecting something has changed - possibly the dialplan syntax? I have no idea. |
| 10:30.48 | frogonwheels | keebler: hopefully the log willtell you. |
| 10:30.53 | frogonwheels | gl |
| 10:30.55 | keebler | dthank |
| 10:30.56 | keebler | thanks |
| 10:31.22 | keebler | I can, theoretically just copy the PAP2 dialplan to the SPA9000 right? |
| 10:31.37 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 10:31.39 | frogonwheels | good theory. but I really don't know. .That's what i'd have done. |
| 10:36.44 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 10:37.00 | awk | good day, anyone know if 'call forward' one can disable on the snom 360/320 phone themself? eg: prevent or have the option removed under menu to not show it or prevent it? |
| 10:39.08 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 10:45.11 | pif | tzafrir_laptop: hi, do plan a 1.4.23 for debian ? |
| 10:55.05 | tzafrir_laptop | pif, would 1.6 be better? |
| 10:55.33 | tzafrir_laptop | pif, I can probably create one, but it will take some time, and I prefer to spend it on 1.6 |
| 10:55.43 | pif | right now we are happy with the 1.4.x series |
| 10:56.13 | pif | but I'd be interested to try 1.6 too |
| 10:56.31 | awk | tzafrir_laptop don't you think 1.6 IS WAY TO BUGGY for a release? 1.4 still has issues, i'm still submmiting 1.4.23 bugs.. |
| 10:56.53 | awk | i'm stuck on 1.4.22 (HIGHLY PATCHED) to work properly |
| 10:57.26 | pif | awk: do you mean 1.4.23 has new bugs? |
| 10:57.41 | awk | ofcourse |
| 10:57.51 | pif | vs 1.4.22 ?? |
| 10:58.17 | awk | now i'm finding lpc10.c is leaking |
| 10:58.47 | awk | http://bugs.digium.com/view.php?id=14308 here is a bug I was having issues with, luckly resolved |
| 10:59.00 | awk | but after that stil finding the leak |
| 10:59.06 | pif | you are not saying 1.4.23 is regression vs 1.4.22, are you? |
| 10:59.11 | *** join/#asterisk scruz (n=scruz@41.220.73.170) |
| 10:59.37 | scruz | good day |
| 11:00.07 | awk | pif, i'm saying I wont roll out my software with 1.4.23 yet, as i'm not satisfied its stable enough yet, i'm using around 8 patches to make 1.4.22 work properly.. |
| 11:00.54 | *** join/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
| 11:01.05 | pif | I happily run vanilla 1.4.22 , what big issues do you have with it? |
| 11:01.07 | awk | there is no way i would even attempt to let people use 1.6 unless all i want to do all day is submit bug reports and lose 90% of clients |
| 11:01.32 | pif | you bet |
| 11:01.33 | awk | pif, queues, parking BIGGIES |
| 11:01.45 | pif | ah, I don't use these much |
| 11:02.14 | awk | anyway bbl. & |
| 11:02.35 | pif | 1.4.23 changelog says several parking issues solved |
| 11:04.25 | awk | true, but major issues with crashes... latest svn seems ok on test machine except 1 memory leak... but not had enouhg time to actually test it |
| 11:04.28 | awk | properly... |
| 11:05.05 | pif | how many users you have? |
| 11:05.41 | awk | depends on what client, some clients do 22k calls a day outbound |
| 11:05.57 | pif | serious shit :) |
| 11:06.22 | scruz | if a sip channel has a default context defined in sip.conf, does that mean i can't make calls to that channel with a different context? |
| 11:06.33 | pif | any memory leak is bound to become very annoying with 22K/day |
| 11:06.49 | awk | scruz have you included your contex you want to dial? |
| 11:06.56 | scruz | yes i have |
| 11:07.01 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 11:07.13 | awk | pif, that client i was using 1.4.18, highly patched by our coders |
| 11:07.43 | awk | way to scared to move them.. as allot of these issues only happen under extreme load. |
| 11:08.02 | awk | scruz: no then you fine... anyway, work... |
| 11:08.16 | pif | I know, it's scary to upgrade a high-load machine |
| 11:08.42 | scruz | i found the issue...i'd renamed the context in extensions.conf, but the call file still had the old name |
| 11:11.19 | pif | have you guys moved to ael2 yet , or still prefer plain old asterisk extension syntax? |
| 11:12.09 | scruz | who? me? |
| 11:12.26 | pif | anybody :) |
| 11:12.52 | pif | just curious avout AEL2 relevance |
| 11:13.06 | scruz | i guess i use 'plain old asterisk extension syntax', as we use asterisk 1.2 here |
| 11:17.01 | *** join/#asterisk BipBip (n=BipBip@194.65.5.235) |
| 11:20.49 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:21.17 | keebler | Where can I find omfg |
| 11:21.18 | keebler | asjhd |
| 11:21.18 | keebler | ad |
| 11:21.40 | keebler | I just found an IP Phone in my garage.... 4 hours wasted on teh SPA9000 |
| 11:21.42 | keebler | sigh |
| 11:23.38 | tzafrir_laptop | pif, I can spend a bit of time on it to get 1.4.23 buildable with at least a few of the bristuff patches |
| 11:23.59 | tzafrir_laptop | But I can't really spend time on actually testing it |
| 11:24.47 | pif | tzafrir_laptop: could you do a -classic version without bristuff (or only the ISDNGuard patch) ? |
| 11:25.30 | tzafrir_laptop | If I work on it, I need the minimal chan_dahdi adaptations |
| 11:26.02 | pif | 1:1.4.21.2~dfsg-3 is unusable with a Digium TE410 (crashes, 100% CPU, etc.) |
| 11:26.11 | tzafrir_laptop | pif, any bug? |
| 11:26.32 | pif | I reported on the debian bts |
| 11:26.45 | tzafrir_laptop | because a reliable Lenny version is something I'd like to have |
| 11:27.22 | pif | I had to build a vanilla 1.4.22 to have a usable installation |
| 11:27.42 | pif | I recycled your ./debian dir from 1:1.4.21.2~dfsg-3 |
| 11:28.01 | pif | it builds OK on 1.4.23 too |
| 11:28.55 | pif | 1) remove all patches (except ISDNGuard), 2) touch agi/xagi-test.c, 3) debuild |
| 11:28.59 | tzafrir_laptop | pif, which bug? 504741 ? |
| 11:29.07 | pif | hmm wait |
| 11:29.47 | tzafrir_laptop | or 471160? http://bugs.debian.org/471460 http://bugs.debian.org/504471 |
| 11:32.10 | *** join/#asterisk Jandark (n=milad@unaffiliated/slackark) |
| 11:33.30 | Jandark | hi365, I had problem I add exten => _X.,n,NOOP(Bala bala bala bala ) in [from-pstn-custom] but I check out console and I never see this log on console |
| 11:34.06 | hi365 | check to see if Jandark has paypaled him anything... |
| 11:34.19 | hi365 | ...nope nothing there. sorry |
| 11:34.34 | scruz | i'm trying to use asterisk.net to make a call. i've no idea why it fails, but when i try the same thing via telnet, it works...can someone recommend a Ruby/Python/C library for using ami? |
| 11:35.17 | *** join/#asterisk Aurs (n=Ove_Aurs@180.80-202-215.nextgentel.com) |
| 11:37.20 | Aurs | hello. I'm doing some tests with voicemail storage on a nfs mount. My voicemail messages get "speeded up", and cut short when I save them via nfs. (asterisk 1.4.22, centos 5.2, dahdi 2.0.0) |
| 11:37.25 | pif | tzafrir_laptop: I can't find my original bug report, but it looks a lot like http://bugs.debian.org/471460 |
| 11:38.17 | tzafrir_laptop | and still reproduced? |
| 11:39.11 | pif | I haven't retried since, but it realy makes sense to provide a vanilla asterisk anyway, as bristuff is very instrusive |
| 11:39.43 | pif | and make it harder to debug asterisk vs bristuff problems |
| 11:40.03 | tzafrir_laptop | pif, that will be on 1.6 . The Lenny package will not have any drastic changes |
| 11:40.20 | tzafrir_laptop | the separate -bristuff package was a pain to build and maintain |
| 11:40.54 | pif | the vanilla package is not hard to build, just remove the patches and 'debuild' |
| 11:47.51 | Aurs | if I record a file with the Record app and save it to the same nfs location, there is no problem. only on voicemails. does anyone have an idea on how to debug this? |
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| 11:52.27 | *** part/#asterisk rethus (n=rethus@xdsl-84-44-227-105.netcologne.de) |
| 11:54.41 | fiddur | Hi. Is there a way, with realtime queues, get the cmd Queue to set paused on a queue_member when that member doesn't answer? ....or have a macro executed when a member doesn't answer? |
| 11:54.42 | *** join/#asterisk merlin8282 (n=merlin82@88-122-137-192.rev.libertysurf.net) |
| 11:54.51 | merlin8282 | Hi |
| 11:55.11 | merlin8282 | Are there some "Gemeinschaft" user here ? |
| 12:00.00 | merlin8282 | Or does a "gemeinschaft" irc channel exist, or someting similar ? |
| 12:00.32 | merlin8282 | I can english (of course), french and german. |
| 12:00.37 | *** join/#asterisk orn (n=orn@office.sip.is) |
| 12:01.43 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 12:01.45 | orn | I need to insert a semi-colon and some arguments into the SIP header, but if I use \ to escape it, the \ shows up in the SIP header too. How do I correctly escape the ; ? |
| 12:18.32 | florz | orn: it could work if you just assign it with \ to a variable, then copy that variable to itself by setting it again from itself, and then using that variable in the addheader |
| 12:19.41 | orn | florz: Thanks. I'll try that. |
| 12:19.57 | *** join/#asterisk scruz (n=scruz@41.220.73.170) |
| 12:20.05 | scruz | hello again |
| 12:20.20 | scruz | finally solved my originate problems with asterisk.net |
| 12:21.41 | scruz | the problem was the defaults suck. for instance, when originating a call, you *must* specify a timeout for *both* the OriginateAction instance and the SendAction() method call |
| 12:22.14 | orn | florz: Doesn't work. The \ is still sent in the header. I read somewhere that this was a bug in 1.6, but that bug was supposed to have been fixed as far as I could tell. |
| 12:23.21 | scruz | and i think i finally get the way calls work |
| 12:23.51 | *** join/#asterisk path_ (n=path@223-102-21-190.adsl.terra.cl) |
| 12:26.01 | orn | florz: http://bugs.digium.com/view.php?id=14110 |
| 12:26.08 | orn | i guess it's been patched :-) |
| 12:28.35 | beherit | is it possible that after the registration process i will forward the SIP to another * server? |
| 12:33.57 | merlin8282 | In fact, i'm not able to make the "gemeinschaft" software run. What interface would you advise me ? |
| 12:34.09 | merlin8282 | I see that freepbx has a large community |
| 12:37.54 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 12:42.52 | *** join/#asterisk Greek-Boy (n=greek@41.222.89.77) |
| 12:44.44 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 12:45.15 | *** join/#asterisk keebler (n=keebler@h199.233.20.98.dynamic.ip.windstream.net) |
| 12:45.26 | keebler | BOOYA!!! |
| 12:46.07 | keebler | I managed to get 3 blocks while driving and talking on a WRT54G wireless Bridge and Asterisk. |
| 12:47.14 | keebler | Now to study proper dial plans. |
| 12:48.14 | *** part/#asterisk merlin8282 (n=merlin82@88-122-137-192.rev.libertysurf.net) |
| 13:01.10 | *** join/#asterisk angryuser (n=gdobrovo@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr) |
| 13:06.36 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 13:10.51 | *** join/#asterisk marl (n=marl_sco@78.148.238.15) |
| 13:14.04 | *** join/#asterisk elred (i=sauron@fucksheep.org) |
| 13:14.20 | marl | hi folks, could anyone recomend a way to block certain ip's from accessing the manager port? i have an ip that is constantly trying to acces port 5038, and althow it is being refused because it cant be authenticated, it is getting REALLY anoying! i have tried the following in hosts.deny but to no avail :( ALL : 87.117.237.68 |
| 13:15.01 | orkid | asterisk has no ACL capabilities? |
| 13:15.10 | *** join/#asterisk inam (n=IceChat7@116.71.208.231) |
| 13:15.15 | *** join/#asterisk propellerhead (n=yogurt2u@host15.190-30-186.telecom.net.ar) |
| 13:15.30 | inam | helooooooooooooooooooo every body |
| 13:15.55 | inam | i need some help in asterisk about outboud routes |
| 13:16.01 | beherit | marl: try blocking it using your firewall |
| 13:16.12 | marl | was hoping to avoid that :( |
| 13:16.21 | marl | but if thats the only way then i will |
| 13:16.36 | inam | can some one help me |
| 13:16.40 | marl | can u think of a reason why the hosts.deny entry wouldnt work? |
| 13:16.40 | orkid | whats the problem.. u still want this ip accessing other services? |
| 13:16.59 | orkid | because asterisk doesnt use tcp wrappers? |
| 13:17.01 | marl | nope, i want to stop it access anything |
| 13:17.10 | orkid | so stop it at the firewall |
| 13:17.12 | Gido-E | marl use your firewall |
| 13:18.06 | inam | marl u can also stop asterisk access using host.deny file |
| 13:18.10 | inam | it is possible |
| 13:18.46 | marl | shurly i WANT axterisk to use the hosts.deny file? |
| 13:19.02 | inam | yah it's possible |
| 13:19.42 | elred | Hello. I am trying to get the callee number in my dialplan (extensions.conf from a zapata.conf context= line), EXTEN is no good there. Do yo guys know which Asterisk's variable do I have to retrieve to get the number ? I readed the following : http://www.voip-info.org/wiki/view/Asterisk+variables but can't find my need. Thanks |
| 13:19.43 | inam | but tell me about us experties of linux |
| 13:19.44 | beherit | a more complex solution marl is use fail2ban for added security |
| 13:19.46 | marl | ah, it may be posible, but would anyone be willing to point me int he rite direction? lol :) |
| 13:20.06 | *** join/#asterisk ocnarf (n=chatzill@ded-134-126.eglobalreach.net) |
| 13:20.45 | ocnarf | im always seeing this in my CLI translate.c:163 framein: no samples for ulawtolin |
| 13:20.53 | ocnarf | anyone here have an idea what it mean? |
| 13:20.56 | ocnarf | *means |
| 13:21.15 | inam | yah why not |
| 13:21.52 | marl | im using a program that watches the secureity log at he moment and enters ips into hosts.deny that attempt to meny ssh connetions, was trying to expand it to do the same with * |
| 13:22.45 | *** join/#asterisk path_ (n=path@19-117-21-190.adsl.terra.cl) |
| 13:22.57 | inam | which program |
| 13:23.06 | inam | and why u use |
| 13:23.24 | inam | ssh self is too secure |
| 13:24.44 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:25.35 | marl | it is just a script i rote myself, and as some of my servers still use passwords for ssh access, in theroy it is posible to crack the passwords given enough attempts |
| 13:26.56 | inam | look if u wana to deny outside access for ur asterisk simply specify that network's. |
| 13:27.13 | inam | ussing deny file |
| 13:27.20 | inam | f u can |
| 13:28.00 | inam | r give access only to a specified user. in tcp wrapper |
| 13:28.12 | esaym | where is the best place to start trouble shooting when I can accept incoming calls from my provider but when I make an outgoing call I am met with silence? |
| 13:28.28 | inam | through finger command. |
| 13:35.52 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:36.56 | zeeesh | i want to write some perl AGI for calleridnum base or accountcode base .. is there any tutorial or website from where i would able to write some initial code? |
| 13:37.49 | [TK]D-Fender | ~book |
| 13:38.12 | jbot | book is probably probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 13:38.46 | [TK]D-Fender | ^^^^^ |
| 13:40.33 | keebler | [TK]D-Fender: I got everything working! |
| 13:40.38 | keebler | Haven't slept yet. |
| 13:40.42 | [TK]D-Fender | keebler: Yeehaw |
| 13:40.49 | [TK]D-Fender | keebler: what was that last challenge? |
| 13:40.54 | keebler | 950ft with two WRT54Gs acting as bridge. |
| 13:40.58 | keebler | well |
| 13:41.02 | keebler | that wasn't the last challenge |
| 13:41.21 | [TK]D-Fender | keebler: Oh yeah the "lets run * behind an ATA that'll FUBAR outside SIP"... now I remember.... |
| 13:42.03 | keebler | The last challenge was trying to get an SPA9000 to act as a dummy ATA, only to have it not work and I end up finding a SPA901 Phone in a box..... |
| 13:42.03 | [TK]D-Fender | keebler: What are you bridging exactly? |
| 13:42.26 | [TK]D-Fender | SPA-9000? |
| 13:42.29 | keebler | ASterisk is on one end, 8 phones are on the other, in a 1 acre spread. |
| 13:42.34 | [TK]D-Fender | thats the PBX core IIRC |
| 13:43.15 | keebler | [TK]D-Fender: Yeah, well, it has two analog ports, and I was hoping to just utilize them and bypass the FXS/Router crap. |
| 13:43.46 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 13:44.03 | [TK]D-Fender | keebler: Its a bloody PBX... what'd you pick that thing up for? :) |
| 13:44.22 | *** part/#asterisk satish2437 (n=root@122.167.67.58) |
| 13:44.34 | keebler | My ex-boss bought a bunch of crap. |
| 13:44.52 | [TK]D-Fender | keebler: So I can tell from your mention of the SPA-901 ;) |
| 13:45.01 | keebler | Yeah. |
| 13:45.14 | keebler | Oh, would you be interested in a SkyPilot Canopy Network? |
| 13:45.17 | keebler | Only used once. |
| 13:45.18 | keebler | :P |
| 13:46.07 | keebler | Same ex-boss-thegenius. Paid $16K for it with the intention of using it on a rig. yeah.. bad bad idea. |
| 13:46.30 | [TK]D-Fender | "For sale : 1 medium parachute, used once, never opened, slightly stained, best offer" |
| 13:46.42 | keebler | Why use equipment thats supposed to cover an ENTIRE TOWN! to cover one acre? |
| 13:47.01 | keebler | There's a reason he's the EX-boss. Haha |
| 13:47.05 | keebler | I've got his job now. |
| 13:47.12 | [TK]D-Fender | OUCH |
| 13:47.23 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:47.40 | keebler | Yeah, can't talk about the gore-ish details, but it was messy. |
| 13:47.47 | elred | Hello. I am trying to get the callee number in my dialplan (extensions.conf from a zapata.conf context= line), EXTEN is no good there. Do yo guys know which Asterisk's variable do I have to retrieve to get the number ? I readed the following : http://www.voip-info.org/wiki/view/Asterisk+variables but can't find my need. Thanks |
| 13:47.55 | keebler | He really screwed the company over too. |
| 13:48.15 | [TK]D-Fender | elred: "core show function CALLERID" |
| 13:48.24 | Gido-E | CALLERID(num) |
| 13:49.13 | [TK]D-Fender | elred: Go read up on how to use functions on the WIKI or in the docs in your source tarball |
| 13:51.51 | *** join/#asterisk CrazyTux (n=brandon@216.138.104.226) |
| 13:52.13 | marl | LOL, finnaly found why i was getting the attempted logins!!!! had a system monitor setup throug my hosting company, and it was checking with 5038 was open every minite!!!!! |
| 13:52.42 | elred | oops sorry, I wanted to say "the caller", not the callee |
| 13:52.57 | elred | I am already using CALLERID(num) |
| 13:53.01 | [TK]D-Fender | elred: We figured as much |
| 13:53.14 | [TK]D-Fender | elred: Then what do you need? |
| 13:53.32 | *** join/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
| 13:54.07 | elred | the called line number, in case I am using multiple ZAP channel |
| 13:54.37 | elred | well... I guess I have to make some AGI that know which ZAP channel is linked to which physical analog line and find out the called number this way |
| 13:54.37 | elred | ? |
| 13:55.00 | elred | like, for outgoing call you can use any line, but for incoming there is a one line = one number ? |
| 13:55.22 | elred | I am not sure tho, because I am going to use a T2 which handle multiple phone line on a single analog line |
| 13:55.45 | [TK]D-Fender | elred: Analog lines I take it? |
| 13:55.48 | *** join/#asterisk viraptor (n=viraptor@awh178.internetdsl.tpnet.pl) |
| 13:56.15 | elred | yes, that's why I am in a context specified from zapata.conf |
| 13:56.35 | beherit | I'm trying to understand how * work so I installed my * from source without GUI support. I have created two sip user in sip.conf the 1000 and 1001, and two x-lite was able to register using the two sip user, but when 1000 tried calling the extension 1001 kit says extension not found. any idea what did i missed? |
| 13:57.00 | beherit | kit=it |
| 13:57.01 | [TK]D-Fender | elred: then you can either look at the ${CHANNEL} variable and parse it out, or what I would advise is send each zap channel into its own context and set a var yourself |
| 13:57.26 | [TK]D-Fender | beherit: Yes... you didn't configure your DIALPLAN |
| 13:57.49 | elred | [TK]D-Fender : ok thanks ! I will try that. |
| 13:58.01 | [TK]D-Fender | beherit: sip.conf setup is only the tiniest part of * setup. Dialplan is 90%+ of the work to be done |
| 13:58.33 | [TK]D-Fender | elred: You could use a single context and use a common bit of code right at the start to parse it out of course... |
| 13:58.42 | beherit | [TK]-Fender: oh ok any sample dial plan for my scenario? |
| 13:58.47 | [TK]D-Fender | elred: Depending how many channels you have it could be worthwhile |
| 13:58.56 | [TK]D-Fender | ~jerjerguide |
| 13:58.57 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 13:58.58 | [TK]D-Fender | ^^^^ |
| 13:59.11 | [TK]D-Fender | beherit: for "inspirational" value |
| 13:59.18 | [TK]D-Fender | beherit: Go read.... THE BOOK |
| 13:59.21 | [TK]D-Fender | ~book |
| 13:59.22 | jbot | extra, extra, read all about it, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 14:00.14 | viraptor | how do you guys solve the problem of loadbalancing many asterisks + attended transfers? |
| 14:00.36 | viraptor | for example if I load-balance on any field, call can come through PBX-a, but the other leg goes through PBX-b, because they're unrelated before the 'REFER'... |
| 14:01.02 | *** join/#asterisk Pagautas (n=bigman@ns.voip.ktu.lt) |
| 14:06.35 | *** join/#asterisk sasargen (n=chatzill@70-4-14-19.pools.spcsdns.net) |
| 14:07.47 | Aurs | after a few hours of googling: looks like -t will solve my problem with nfs storage for voicemail |
| 14:08.05 | Gido-E | -t? |
| 14:08.06 | *** join/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
| 14:08.55 | Aurs | yes |
| 14:09.02 | Aurs | -t When recording files, write them first into a temporary holding directory, then move them into the final location when done. (from man asterisk) |
| 14:10.30 | Aurs | my voicemails get cut short and speeds up after 5-10 seconds.. so I'll try with -t passed to asterisk and see if that helps.. found this from a svn commit message in 2006.. perhaps it is so obvious that noone else has ever asked this question :) |
| 14:12.50 | *** join/#asterisk sasargen_ (n=chatzill@70-4-14-19.pools.spcsdns.net) |
| 14:16.14 | *** join/#asterisk bn43 (n=dhashen@196.212.81.58) |
| 14:17.14 | bn43 | hello I am fiddling with asterisk and would like to know if there is a 'billing' tool for it - the calls made by an extension and the cost thereof? |
| 14:18.21 | [TK]D-Fender | bn43: Go read up on CDR and lookup "a2billing" on the WIKI |
| 14:18.23 | [TK]D-Fender | ~wikis |
| 14:18.24 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 14:18.25 | [TK]D-Fender | ~book |
| 14:18.25 | jbot | i guess book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 14:21.02 | jsmith | bn43: Aanother popular billing engine for Asterisk (among other things) is Freeside (www.freeside.biz) |
| 14:21.09 | *** join/#asterisk Faustov (i=user@gentoo/user/faustov) |
| 14:21.46 | Faustov | hi, can ranges be defined in exten? like exten => _[5,6]Z would match extensions starting with 5 or 6? |
| 14:23.15 | *** join/#asterisk mcargile (n=mikec@rrcs-24-173-156-170.se.biz.rr.com) |
| 14:25.01 | [TK]D-Fender | Faustov: no "," |
| 14:27.45 | Faustov | [TK]D-Fender: thanks |
| 14:28.46 | *** join/#asterisk DarkRift (n=dark@bas1-sthubert21-1096611958.dsl.bell.ca) |
| 14:30.05 | *** join/#asterisk telnettech (i=telnette@gw.percipia.com) |
| 14:30.49 | *** join/#asterisk xacatecas (n=jkroon@dsl-244-21-00.telkomadsl.co.za) |
| 14:32.15 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 14:33.13 | ruben23 | anyone have idea on this..?http://pastebin.com/m348e6198 |
| 14:34.52 | xacatecas | hi all, i just had some accusations thrown in my direction that asterisk can only handle about 100 concurrent sip registrations/connections. what should be the appropriate response (other than hopefully being able to laugh) |
| 14:34.55 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:34.59 | [TK]D-Fender | ruben23: Yes, either vi(m) crashed and left a swap or someone else is editing it |
| 14:35.42 | [TK]D-Fender | xacatecas: no... pointing & laughing is an appropriate response |
| 14:35.50 | ruben23 | <PROTECTED> |
| 14:36.05 | ruben23 | [TK]D-Fende how am i to correct this..? |
| 14:36.09 | [TK]D-Fender | xacatecas: FFS they sell 8-port PRI cards. thats over 200 friggen ZAP channels all by itself |
| 14:36.29 | [TK]D-Fender | ruben23: make sure noone else is editing it and wipe out the temp file |
| 14:36.52 | *** join/#asterisk sasargen (n=chatzill@70-4-14-19.pools.spcsdns.net) |
| 14:37.12 | ruben23 | <PROTECTED> |
| 14:37.18 | [TK]D-Fender | xacatecas: And SIP load? easily good past 200. Get a decent box and you're fine. what'll kill is massive TRANSCODING or RECORDING |
| 14:37.27 | [TK]D-Fender | ruben23: man rm" <- |
| 14:37.48 | ruben23 | <PROTECTED> |
| 14:37.58 | seanbright | holy hell |
| 14:38.01 | [TK]D-Fender | ruben23: I'd be on the SAME |
| 14:38.03 | [TK]D-Fender | bet* |
| 14:38.08 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:38.41 | tzafrir_laptop | ruben23, as the swap file is newer than the file on the disk, you may prefer to recover |
| 14:38.43 | seanbright | based on some of the questions in here... i should be making $400,000 a year |
| 14:39.05 | Gido-E | seanbright :-) |
| 14:39.17 | [TK]D-Fender | seanbright: OMGZ how do I delete aa file, HAELP!!!! |
| 14:39.28 | Gido-E | seanbright if you want to earn that much of money, STOP working! |
| 14:39.41 | seanbright | i am setting up a phone system for our call center. |
| 14:39.48 | seanbright | 200 agents, 5000 calls a day |
| 14:40.00 | seanbright | how do i make a directory? |
| 14:40.09 | tzafrir_laptop | xacatecas, register 200 devices with sipsak? |
| 14:40.24 | *** join/#asterisk [intra]lanman (n=Raymond@freeswitch/developer/intralanman) |
| 14:40.37 | *** join/#asterisk moy (n=chatzill@bas1-unionville55-1177733953.dsl.bell.ca) |
| 14:41.35 | *** join/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
| 14:42.34 | xacatecas | [TK]D-Fender, exactly what i was hoping to confirm :) |
| 14:42.35 | xacatecas | i was about to say ... unless something seriously changes at around 100 ... i've got 70 SIPs + 30 odd IAX2 connections and I'm not even seeing the load ... |
| 14:44.55 | [TK]D-Fender | xacatecas: Aside from the laughing part, you can sum them up as "FUD" |
| 14:45.01 | xacatecas | tzafrir_laptop, why waste my time? if the guy insists, sure. |
| 14:45.12 | *** join/#asterisk The_Boy_Wonder (n=davidvos@nat/digium/x-2694429e1f61d8e5) |
| 14:45.28 | xacatecas | well, this guy is actually a VoIP fanatic... just insists that you should use some kind of sip registration server in front of asterisk ... not sure exactly how that helps though. |
| 14:45.57 | tzafrir_laptop | xacatecas, or maybe there's a better automated SIP client to emulate that? |
| 14:46.07 | Faustov | damn it, it's almost impossible to match patterns for cellphone calls :< |
| 14:46.34 | [TK]D-Fender | xacatecas: It can help if you get a FLOOD of registrations all at once I suppose |
| 14:49.47 | xacatecas | whatever. i'm sitting at less than 5% CPU on a box with 64 SIP and 15 IAX2 connections ffs, that's close enough for load test. |
| 14:49.58 | *** join/#asterisk andrewn (n=andrew@76-191-212-233.dsl.dynamic.sonic.net) |
| 14:50.14 | [TK]D-Fender | xacatecas: 80% of his claim at 5% load. Thats proof enough for me |
| 14:50.44 | viraptor | xacatecas: registration on a reg. server in front does wonders for voice quality on slower servers - especially if some broken phones try to register every 5 secs or so... and there are hundreds of them |
| 14:50.46 | [TK]D-Fender | xacatecas: don't forget to point & laugh though :) |
| 14:51.09 | [TK]D-Fender | viraptor: And in what Crack-based world would a phone try to register every 5 seconds? |
| 14:52.01 | Faustov | why would 205 match exten => _!0[5-7]XXXXXX if X specifies that the number has to be there? |
| 14:52.05 | viraptor | try to deny registration to some grandstreams - they will fire another one as soon as they get a reject |
| 14:52.39 | [TK]D-Fender | Faustov: never put ! at the beginning. |
| 14:52.41 | tzafrir_laptop | Faustov, hmm.. isn't anything after the '!' practically ignored? |
| 14:52.51 | [TK]D-Fender | tzafrir_laptop: Yup |
| 14:52.59 | Faustov | what! |
| 14:53.01 | [TK]D-Fender | Faustov: You're getting sloppy man... |
| 14:53.10 | viraptor | I'm not even going into stuff like eyebeam and sjphones - they can generate up to 40 simultaneous registrations just because they have a bad day |
| 14:53.27 | [TK]D-Fender | viraptor: GS.... |
| 14:53.29 | [TK]D-Fender | ~gs |
| 14:53.30 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 14:53.35 | [TK]D-Fender | ~grandstream |
| 14:53.35 | jbot | methinks grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 14:53.47 | tzafrir_laptop | [TK]D-Fender, and that is related because? |
| 14:54.20 | xacatecas | i lie, <1%. ok, this guy should go jump in a freezing lake. do the world a favor. |
| 14:54.23 | [TK]D-Fender | tzafrir_laptop: the GS bit? Only because he used them as an example of "stupid phones",in a "you'll get what's commin'" kinda way |
| 14:54.40 | Faustov | [TK]D-Fender: but only ! matches 0 or more characters - i need to find patterns where there is a digit (could be missing), then 0, then [5-7], then 8 digits |
| 14:54.52 | [TK]D-Fender | Faustov: Doesn't work that way |
| 14:55.07 | tzafrir_laptop | Use two different patterns |
| 14:55.08 | [TK]D-Fender | Faustov: there is no "could be a digit here or not" pattern char |
| 14:55.16 | [TK]D-Fender | Faustov: it'll take 2 patterns to do this |
| 14:55.28 | Faustov | hmm ok, actually 4, but fine, i get the idea |
| 14:55.33 | Faustov | need to split it into more patterns |
| 14:55.37 | [TK]D-Fender | Faustov: yup |
| 14:56.27 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 14:56.27 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:56.50 | xacatecas | has been chuckling since he heard it this claim. |
| 14:57.03 | xacatecas | no, those that try to register every 5 seconds gets the iptables treatment. |
| 14:57.50 | *** part/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:58.04 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:58.10 | viraptor | xacatecas: then you get an angry user with his budgetphone and a lot of explaining to do... I guess that's why people are using reg. server as a first line of defence |
| 14:58.19 | [TK]D-Fender | xacatecas: Nah.. burn it.. burn it with FIRE |
| 14:58.57 | xacatecas | viraptor, perhaps, but then I get the opportunity to, how shall we put this, dunk the users head in a bucket of ice? |
| 14:59.01 | viraptor | xacatecas: we've got many thousands of users, but couldn't care less if everyone started registering every second - it doesn't even reach asterisk, so there's nothing to worry about (kinda firewall) |
| 15:00.11 | *** part/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
| 15:00.43 | Faustov | yay, it works :> |
| 15:01.01 | Faustov | now the mofos who call mobile phones after hours in the office hear a not so pleasant words from me :P |
| 15:03.13 | *** join/#asterisk telnettech (i=telnette@gw.percipia.com) |
| 15:05.16 | *** join/#asterisk shido6 (n=shido6@74-132-200-214.dhcp.insightbb.com) |
| 15:07.36 | xacatecas | viraptor, re the gs problem ... yea, we ran into that one. i was wondering why the load got to 10 % ... there is a "delay" option in sip.conf :) |
| 15:07.42 | xacatecas | anyway, i'm off home |
| 15:09.18 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-5cc8ed000f376ea4) |
| 15:09.18 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:16.23 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 15:17.23 | *** join/#asterisk tobias (n=tobias@cpe-069-134-127-101.nc.res.rr.com) |
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| 15:20.17 | Faustov | http://pastebin.com/d12a39ff4 <--- any idea what could be wrong with this? 1.4.22.1 asterisk is supposed to have gotoiftime implemented |
| 15:21.09 | [TK]D-Fender | Faustov: It is implemented. |
| 15:21.25 | [TK]D-Fender | Faustov: Otherwise you'd have gotten a "what app was that again?" warning |
| 15:21.51 | Faustov | [TK]D-Fender: right, and the example is almost copied from the manual on voip-info |
| 15:22.02 | Faustov | so i'm not quite sure what could be wrong here |
| 15:22.11 | [TK]D-Fender | Faustov: Yes... and maybe, just maybe it doesn't MATCH your criteria |
| 15:22.44 | Faustov | [TK]D-Fender: if it doesn't match the criteria, shouldn't it be skipped? |
| 15:23.21 | [TK]D-Fender | Faustov: Do I see a priority # 2 for that exten? No. |
| 15:23.41 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 15:23.42 | [TK]D-Fender | Faustov: AUTOFALLTHROUGH |
| 15:25.05 | angryuser | can someone tell me why when i configure 2 trunk's with the same provider on public ip and i got firbidden for one of them ? ie i nable trunk N1 it works, i enable N2 it works i enable both , only one able to call out, and another get's fobidden |
| 15:25.13 | angryuser | i am sure about credentials |
| 15:25.23 | angryuser | it's not a port issue ? |
| 15:25.49 | angryuser | i have no nat naywhere |
| 15:25.54 | angryuser | anywhere* |
| 15:25.57 | [TK]D-Fender | angryuser: pastebin it all |
| 15:26.09 | Faustov | [TK]D-Fender: i blame lack of sleep, thanks :> |
| 15:26.17 | thansen | anyone have a suggestion for an sms gateway? |
| 15:29.54 | Faustov | thansen: i heard some people use a phone connected via serial to a server to make it send sms |
| 15:30.21 | *** join/#asterisk bluregard (n=matt@66.251.248.13) |
| 15:30.25 | bluregard | hi all |
| 15:30.32 | thansen | I've thought about doing that too...wasn't sure if anyone's had a good experience with that either |
| 15:31.35 | Faustov | i know 1 person who was pretty successful with that and there are manuals |
| 15:32.01 | angryuser | [TK]D-Fender: i aclled them, and they told me that somewhere their system get confused with the same ip and 2 trunks without nat, and to get it working i need to have spare port for each register (as like i need to add NAT) which is a totally stupid |
| 15:33.32 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:34.10 | jameswf | ping jsmith |
| 15:35.57 | bluregard | has anyone in here done faxing with asterisk? |
| 15:36.24 | Faustov | i was lucky enough to talk my bosses out of it :> |
| 15:36.34 | bluregard | haha, yeah, no such luck |
| 15:37.16 | Faustov | even if you collect links to documents regarding how obsolete and deprecated the fax protocol is? |
| 15:40.17 | bluregard | its for a client that refuses to acknowledge the fact that faxing is old and should be replaced |
| 15:41.08 | Gido-E | bluregard, i alway begin to talk about pigeons if fax comes up :-) |
| 15:41.24 | bluregard | pigeons? |
| 15:41.27 | bluregard | ooh |
| 15:41.28 | bluregard | yeah |
| 15:41.36 | bluregard | sorry, its still early |
| 15:41.45 | *** join/#asterisk dlewis (i=c7340d67@about/security/staff/dlewis) |
| 15:41.48 | Gido-E | almost the end of a working day here. |
| 15:41.55 | bluregard | I think pigeons might be a bit more reliable than fax over sip |
| 15:42.02 | jsmith | jameswf: pong |
| 15:42.05 | Gido-E | bluregard yep :-) |
| 15:44.21 | *** join/#asterisk jshriver (n=jshriver@72.240.39.37) |
| 15:44.23 | jshriver | greetings |
| 15:44.29 | jshriver | I'm getting a cdr_sqlite: database is full error |
| 15:44.44 | jshriver | how do you back it up and purge it. |
| 15:49.17 | *** join/#asterisk CrazyTux (n=brandon@216-110-94-230.static.twtelecom.net) |
| 15:51.03 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-6fa536f807133f81) |
| 15:51.03 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:51.25 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
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| 15:55.37 | jameswf | jsmith: know any * techs in MN i can refer out to |
| 15:55.56 | jsmith | jameswf: MN, as in Minnesota? |
| 15:56.02 | jameswf | yeah |
| 15:56.21 | jsmith | Hmmmmn... yeah, I know a guy or two out that direction |
| 15:57.11 | jameswf | if you could get the okay and foreward me contact info I have an end user in ST. Paul who needs some asterisk voodoo done |
| 15:57.12 | *** join/#asterisk manxpower (n=Administ@router.asteriasgi.com) |
| 15:57.58 | [TK]D-Fender | jameswf: IIRC drmessano is out that way.... |
| 15:58.24 | jameswf | drmessano: is in alabama |
| 15:58.36 | [TK]D-Fender | jameswf: I sit corrected. |
| 15:58.49 | *** join/#asterisk ghenry (n=ghenry@92.41.199.171.sub.mbb.three.co.uk) |
| 15:58.51 | [TK]D-Fender | jameswf: there is a prominent membro or two here though that is... |
| 15:59.14 | [TK]D-Fender | jameswf: Somewhat remiss to come up with a nick right now however... |
| 15:59.16 | [TK]D-Fender | member* |
| 15:59.29 | manxpower | looks around |
| 15:59.34 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:00.10 | [TK]D-Fender | jameswf: Time to mass-scan people :p |
| 16:00.45 | jameswf | i looked at the voip info list only name I know is attacom and I dont think they exist anymore |
| 16:02.31 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:04.22 | manxpower | what are you looking for? |
| 16:04.52 | *** join/#asterisk andy-x_l (i=425c01d1@gateway/web/ajax/mibbit.com/x-510848e0a0bf7024) |
| 16:10.55 | *** join/#asterisk mw-home (n=mw-home@99.55.177.158) |
| 16:11.24 | mw-home | What is a good softphone for ubuntu? |
| 16:12.02 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 16:12.20 | *** part/#asterisk unasi7 (n=unasi7@213.144.157.100) |
| 16:13.22 | guax | mw-home: twinkle |
| 16:14.20 | *** join/#asterisk sack (n=sack@224.Red-79-148-188.dynamicIP.rima-tde.net) |
| 16:15.36 | *** join/#asterisk Khratos (n=Khratos@190.80.227.139) |
| 16:15.42 | Khratos | Good afternoon |
| 16:16.37 | mw-home | guax: wow -- didn't know about that one. |
| 16:16.49 | *** join/#asterisk maddog01 (n=minotaur@d221-65-55.commercial.cgocable.net) |
| 16:16.50 | guax | ;) |
| 16:17.55 | Kobaz | mmm |
| 16:17.57 | Kobaz | interesting |
| 16:18.03 | Kobaz | voicepulse is kicking iax out |
| 16:18.05 | Kobaz | All customers using IAX2 must convert to using SIP to continue using VoicePulse services. The IAX2 protocol does not allow for proper utilization of our infrastructure and poses too great of a support cost. |
| 16:18.33 | *** join/#asterisk hfb (n=hfb@pool-96-247-109-136.lsanca.dsl-w.verizon.net) |
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| 16:21.46 | WhiteWolf | when'd that email go out? |
| 16:22.45 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.67) |
| 16:25.43 | jjshoe | Kobaz link? |
| 16:27.17 | *** join/#asterisk dlewis (i=c7340d67@about/security/staff/dlewis) |
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| 16:28.15 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-46-53.w86-215.abo.wanadoo.fr) |
| 16:28.28 | Kobaz | http://campaign.constantcontact.com/render?v=001aqj1QInodGQbs3pAHkhHwksIj-dJmpoWS7IAQZZW0biLtNZrbWdh7U-yWfdvv0_rSw4uQCkLS7sdyz3OPN83TVzUWVxdY-6KmsgI0gWQoPMTzL5WYykPnsNvPvQaB7Bx6F0rrvrGKiU-WgHAQ7U47hikg85uFYgvZywYZoRVjNINNJQknd5DXesy72RsjCs1EGhJsOQ_pVSsRdKpj0jByNcaTEu5VAu2hsc3GBO6yGKcwtFS9D7Je8cP2m9YujJxE2NOIxi0IpU%3D |
| 16:28.42 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-46-53.w86-215.abo.wanadoo.fr) |
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| 16:29.33 | *** join/#asterisk kamh (n=qmpelkam@host-81-190-236-85.wroclaw.mm.pl) |
| 16:29.45 | kamh | ji all |
| 16:29.47 | kamh | hi all |
| 16:35.26 | *** part/#asterisk mcargile (n=mikec@rrcs-24-173-156-170.se.biz.rr.com) |
| 16:38.05 | manxpower | I abandoned IAX several years ago |
| 16:38.32 | *** join/#asterisk voipnet-tech (n=voipnett@216.195.128.62) |
| 16:38.55 | voipnet-tech | good morning #asterisk |
| 16:39.29 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 16:40.50 | Merlin | i'm thinking of backporting whisper from 1.4 to 1.2. is this a totally insane idea? I have a customer that is willing to pay me to do this. |
| 16:41.24 | *** join/#asterisk path_ (n=path@19-117-21-190.adsl.terra.cl) |
| 16:41.34 | dlewis | Kobaz: how has your experience been with voicepulse? |
| 16:44.20 | *** join/#asterisk ickmund (n=ickmund@ada-bcn-fw01.adamoeurope.com) |
| 16:45.48 | Merlin | voicepulse is awful with support |
| 16:47.20 | russellb | Merlin: it's pretty insane |
| 16:47.53 | russellb | you'd basically be 1) writing it all over again, or 2) backport the entire audiohooks API, channel datastores API, and the chanspy application |
| 16:47.56 | russellb | it's not trivial |
| 16:48.12 | manxpower | If it was trivial someone would have done it already |
| 16:48.23 | *** join/#asterisk SparFux (n=raoul@e182017044.adsl.alicedsl.de) |
| 16:48.30 | dlewis | Merlin: how's the service? |
| 16:51.01 | *** join/#asterisk flohack (n=fhackenb@lancelot.acoveo.com) |
| 16:51.23 | *** join/#asterisk ghenry (n=ghenry@92.41.226.101.sub.mbb.three.co.uk) |
| 16:51.43 | flohack | Hi! Is there a way to set the domain of the Contact: header for outgoing sip requests? I tried setting fromdomain and externip with nat=route, but that does not help. |
| 16:52.08 | dlewis | Merlin: you're not to far from me btw... |
| 16:52.14 | dlewis | good to see CT people in here. |
| 16:53.44 | *** join/#asterisk rwaite (n=fieldyca@rrcs-74-218-125-86.central.biz.rr.com) |
| 16:54.55 | *** join/#asterisk rwaite (n=fieldyca@rrcs-74-218-125-86.central.biz.rr.com) |
| 16:57.14 | Merlin | haha |
| 16:57.16 | Merlin | yes, that's true |
| 16:57.49 | Merlin | dlewis: the servies is mediocre... i cannot get DTMF tones to work for example |
| 16:58.05 | Merlin | dlewis: additionally, many carribean countries cannot be called |
| 16:58.10 | dlewis | hmm |
| 16:58.11 | Merlin | this is voicepulse btw |
| 16:58.22 | dlewis | Merlin: what service provider did you move to? |
| 16:58.35 | Merlin | russelb: what parts of the source code would it touch? only modules or the core code? |
| 16:58.48 | russellb | both |
| 16:58.53 | Merlin | dlewis: we tried bandwidth.com, but we can't get dtmf tones working there either |
| 16:58.55 | *** join/#asterisk beniwtv (n=beniwtv@124.Red-83-36-62.staticIP.rima-tde.net) |
| 16:58.59 | Merlin | dlewis: frankly, i don't know who to go to |
| 16:59.03 | dlewis | hmm |
| 16:59.04 | dlewis | Merlin: i'm looking at flowroute as well. |
| 16:59.59 | beniwtv | hi all.. if I call from my sip phone via *, and in the middle of the call I hang up the sip phone, * kills the agi script. This doesn't happen if the other side hangs up. Anyway to prevent that? |
| 17:01.57 | Merlin | russel: if you had to ballpark the number of hours involved in that (assuming a knowledge of C and a reasonable knowledge of the asterisk source code), could you? |
| 17:02.33 | russellb | I can't estimate how much time it would take someone else ... |
| 17:02.35 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:02.54 | Merlin | i understand |
| 17:03.23 | Merlin | it's clearly a waste of time for the asterisk developer team |
| 17:03.28 | Merlin | but if I have a customer willing to pay |
| 17:03.32 | Merlin | maybe it's worth it for someone |
| 17:03.41 | *** join/#asterisk steliosk (n=Stelios@ipa107.2.tellas.gr) |
| 17:03.46 | russellb | money is a good thing, heh |
| 17:03.57 | russellb | I could recommend some people if you didn't want to do it yourself |
| 17:04.04 | beek | Merlin: Wouldn't it be cheaper for the customer if you simply charged them to upgrade from 1.2 to a newer, supported version complete with whisper? |
| 17:04.27 | Merlin | beek: they are not willing to change the asterisk version |
| 17:04.34 | Merlin | beek: they claim they have too much custom code |
| 17:04.40 | Merlin | russe: yes please |
| 17:04.53 | *** join/#asterisk freakazoid0223 (n=matt@pool-72-81-7-242.phlapa.east.verizon.net) |
| 17:05.05 | *** join/#asterisk jcape (n=jcape@209.120.251.66) |
| 17:05.19 | *** join/#asterisk Shaggy64 (n=shaggy63@c-24-1-173-48.hsd1.in.comcast.net) |
| 17:05.26 | beek | Merlin: One thing is for certain... it would be easier to migrate forward their custom code than it would be to backport * code. |
| 17:05.43 | flohack | Is there a way to set the domain of the Contact: header for outgoing sip requests? I tried setting fromdomain and externip with nat=route, but that does not help. |
| 17:05.59 | Merlin | beek: i may have to convince them of them |
| 17:06.07 | Merlin | of that |
| 17:06.37 | beek | Merlin: You can also point out the greater functionality available in the newer releases and the ease of getting support vs the older, now deprecated, version. |
| 17:06.53 | Shaggy64 | Hello everyone, I'm setting up a new box that has version 1.4 on it. I have an existing 1.2 asterisk server i've been using for many years. What is the best way to configure the new asterisk server without having to redo the entire configration. |
| 17:06.54 | Merlin | believe me, I have :) |
| 17:07.12 | beek | Merlin: Good luck. I've had clients like that before... |
| 17:07.22 | Merlin | haha |
| 17:07.24 | Merlin | thank you :) |
| 17:07.52 | beek | Merlin: I kept them just so that I'd eventually have the opportunity to tell them "I told you so." |
| 17:08.17 | Shaggy64 | can I just copy the config? I'm using the asterisk gui on 1.4 |
| 17:08.19 | *** join/#asterisk clintc (n=clintc@n128-227-117-39.xlate.ufl.edu) |
| 17:08.46 | manxpower | Shaggy64: read the upgrade files. Oh, sorry -- I can't help with GUIs |
| 17:09.03 | Khratos | Merlin, are they really decided to pay what it really costs? |
| 17:09.20 | Merlin | hkr: i don't know what it will really cost yet |
| 17:09.24 | Merlin | but they claim they have a real need |
| 17:09.28 | Merlin | and are willing to pay |
| 17:10.00 | beek | Merlin: Did the customer offer up his nubile, young daughter? If not, then perhaps they're not willing to pay enough. |
| 17:10.19 | Khratos | haha |
| 17:11.03 | Shaggy64 | hrmm so what I should really be asking is how to import the configration into the gui? |
| 17:16.09 | Merlin | beek: first born, daughter's hand in marriage, etc. are all on the table |
| 17:16.28 | ickmund | Anyone know of a updated public list of all destinations with prefix codes? |
| 17:19.06 | dlewis | Merlin: got any CT customers that have tried to use asterisk with Optimum Voice? |
| 17:19.31 | carrar | hahaha |
| 17:19.32 | carrar | http://sqlanywhere.blogspot.com/2008/03/unpublished-mysql-faq.html |
| 17:21.26 | *** join/#asterisk n3hxs (n=HAMming@63.68.135.4) |
| 17:25.24 | Merlin | dlewis: sadly, yes |
| 17:26.28 | Merlin | dlewis: as much as I hate optimum, it does work with asterisk. the ATA they use actually have support disconnect supervision |
| 17:26.49 | *** join/#asterisk Avelino (n=Avelino@mail.paterno.com.br) |
| 17:27.04 | flohack | Is there a way to force asterisk into NAT mode? That would apply the externip setting to the Contact header and would solve my problem... |
| 17:27.24 | dlewis | Merlin: which ATA do they use? |
| 17:27.35 | Merlin | dlewis: a Cisco box |
| 17:27.48 | Merlin | cant remember which |
| 17:28.12 | [TK]D-Fender | flohack: set a localnet for a subnet that isn't possible |
| 17:28.33 | dlewis | Merlin: i've been trying to get an ATA that works with optimum voice... I've tried the ht503, but CID doesn't work... My next bet was to try the Linksys Sipura 3102... |
| 17:28.53 | flohack | [TK]D-Fender: Sorry I don't get what you mean...could you please rephrase? |
| 17:29.13 | dlewis | Merlin: could it be the cisco/linksys 3102? |
| 17:29.16 | manxpower | flohack: read the nat docs then you will understand |
| 17:29.28 | manxpower | ~sipnat |
| 17:29.29 | 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 |
| 17:29.44 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:30.30 | flohack | My problem is that incoming SIP calls from the voip gateway arrive at 192.168.1.107 and should be sent using that address as well, but I have to use binaddr=0.0.0.0. The machine has both 192.168.1.1 (which is in the Contact header) and 192.168.1.107 configured on the same interface |
| 17:31.21 | manxpower | flohack: don't expect asterisk to work very well on a machine with two IPs on the *same network* |
| 17:32.24 | flohack | manxpower: I could configure an ip in a different subnet as well, I simply need to make sure that the contact header is set to a specific ip address. |
| 17:33.02 | manxpower | that header is normally set to whatever interface has the best route to the destination |
| 17:34.16 | *** part/#asterisk elred (i=sauron@fucksheep.org) |
| 17:38.29 | *** join/#asterisk kisu (n=kisu@2001:5c0:1100:9900:acbb:dcfd:e13c:5740) |
| 17:38.48 | *** join/#asterisk lucasb (n=lucasb@office.telifon.com) |
| 17:45.54 | flohack | [TK]D-Fender: Ok, I read the docs, I know what you refer to. I set externip=192.168.1.107 nat=yes localnet=10.0.0.0/255.0.0.0 and reloaded the chan_sip.so module. Asterisk still sets the contact header to 192.168.1.1 though... |
| 17:47.04 | *** join/#asterisk interfaith (n=chatzill@ip67-88-184-130.z184-88-67.customer.algx.net) |
| 17:47.05 | [TK]D-Fender | flohack: First WTF is a private IP doing as a your externip, and are you running 2 NICS on the same subnet? If you are you're probably screwed |
| 17:47.25 | [TK]D-Fender | flohack: Make sure * only binds 1 |
| 17:48.42 | interfaith | stun for asterisk: rtp.c shows support for stun ? any hints on where this is provisioned ? some conf file ? |
| 17:48.57 | manxpower | interfaith: you looked in sip.conf.sample? |
| 17:50.09 | interfaith | no sign of stun in the sample conf files here 1.4.2 |
| 17:50.50 | manxpower | interfaith: then I guess there is no STUN support. |
| 17:51.25 | interfaith | well gtalk.c does use it "ast_rtp_stun_request(p->rtp, &sin, username);" |
| 17:51.40 | manxpower | not much good if there is no config option for it |
| 17:51.56 | *** join/#asterisk Rabenklaue (n=Rabe@92.226.207.11) |
| 17:52.18 | interfaith | maybe asterisk-dev will have something to say about it .. will have to dig gtalk.c to see how it is utilized |
| 17:52.42 | manxpower | interfaith: If nothing else report the documentation bug |
| 17:53.17 | interfaith | thx i'll try that ! |
| 17:53.57 | interfaith | as ast/gtalk is ok p2p with google/gtalk but ast/gtalk p2p /ast/gtalk fails ( my tests ) |
| 17:54.28 | [TK]D-Fender | interfaith: 1.6 supports STUN although there is still little need |
| 17:55.31 | interfaith | i'll check that out. though 1.6 may require linux kernel 2.6 as well, dumping the 2.4 users |
| 17:55.37 | flohack | [TK]D-Fender: I use the private IP as external in order to force asterisk to set the contact header to a value I choose. I'm running one nic with two local addresses. The reason is that I use heartbeat, one address is the address of the computer, the other one is the floating ip of the cluster. I cannot bind only one, because the box is connected to two switches. One switch (10.4.4.0 subnet) is the one used by the softphones and the second switch |
| 17:55.38 | flohack | (used by the voip gateway and all cluster boxes) is 192.168.1.0. |
| 17:56.14 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:56.42 | [TK]D-Fender | interfaith: You mean.... you're still plural? |
| 17:57.15 | interfaith | hoping to work via kernel 2.4 |
| 17:57.34 | manxpower | interfaith: you seem bent on making sure your design won't work with Asterisk |
| 17:57.47 | flohack | so there are two floating ips one in the 10.4.4.0 subnet and one in the 192.168.1.0 the 10.4.4 is used by the softphones and the 192.168.1.0 is used by the voip gateway |
| 17:58.03 | interfaith | the upgrade to 2.6 has many hurdles.. to say the least |
| 17:58.24 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 17:58.25 | manxpower | I was not even aware than zaptel 1.4 supported 1.4 |
| 17:58.30 | manxpower | ..er.. 2.4 kernel |
| 17:58.43 | interfaith | thank intel for that.. no need for zaptel in this embedded solution |
| 17:58.59 | manxpower | why should we thank intel? |
| 17:59.32 | interfaith | ha ha they made it impossible to upgrade , only offering 2.4 driver code |
| 17:59.40 | manxpower | it sucks to be you |
| 18:00.16 | interfaith | ill take a look at 1.6 again , thx |
| 18:01.05 | tzafrir_laptop | interfaith, if you're now stuck with a system that does not support kernel 2.6 you're in a mess anyway |
| 18:01.41 | *** join/#asterisk hugenay (n=luther@213-140-11-128.fastres.net) |
| 18:01.52 | hugenay | hello |
| 18:02.06 | jsmith | hello hugenay |
| 18:02.13 | hugenay | hi jsmith |
| 18:02.29 | interfaith | well iax2/sip etc work fine .. just gtalk fails on p2p ast/gtalk p2p ast/gtalk |
| 18:02.55 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 18:03.01 | hugenay | i've a little question...some friend pointed me to asterisk...but before start to studying it i would like to know if it can be useful for my purpose |
| 18:03.03 | interfaith | has anyone found a working ast/nat p2p ast/nat solution ? |
| 18:03.46 | flohack | [TK]D-Fender: Are you still with me? |
| 18:04.53 | hugenay | so the question is: on a linux box i would create this steps: an event occours -> an email is sent and received -> a phone call is started (usb-cable, bluetooth or whatever) to me and when i answer to the call, an audio file is played. |
| 18:05.15 | *** join/#asterisk xacatecas (n=jkroon@dsl-240-158-44.telkomadsl.co.za) |
| 18:06.07 | [TK]D-Fender | hugenay: Sure |
| 18:06.09 | hugenay | does anyone know if asterisk can help me with that, or even if there is some another kind of sw more easy than a complete pbx like asterisk? |
| 18:06.29 | xacatecas | hi all, what's the status of T.38 support? Based on the description of ReceiveFax it looks like it's there, however, I've no idea how to make it work. |
| 18:06.36 | [TK]D-Fender | flohack: You're running a multi-homed PBX... that = trouble... can't help you there. |
| 18:07.19 | xacatecas | [TK]D-Fender, why is multi-homes = trouble? |
| 18:07.25 | hugenay | [TK]D-Fender, so asterisk does that. wonderful. at last i will use it. could you suggest me, if you know some, other solution? |
| 18:07.32 | xacatecas | other than the obvious sip redirect issues ... |
| 18:08.09 | hugenay | [TK]D-Fender, in any case, thank you!!! |
| 18:08.24 | [TK]D-Fender | hugenay: No, I don't know another. What you want from * is easy. The harder part is the script you'll have to write to take that incoming e-mail and process it to prep the dialout. The actual dialout bit isn't a big deal |
| 18:08.42 | *** join/#asterisk masus (i=masus@88.248.14.186) |
| 18:09.40 | *** join/#asterisk chi6IT41 (n=chigital@tmo-100-22.customers.d1-online.com) |
| 18:10.42 | masus | hi all, i have "Asterisk 1.4.22" and have one question if i login as an agent and make a call , i can't quit the current call by pressing * it's something like frozen . can anybody help me please ? |
| 18:10.49 | hugenay | [TK]D-Fender, ok, i suppose that "technically" my req was simply. the hard is find the right sofware |
| 18:10.58 | hugenay | thank for your suggestion |
| 18:11.11 | hugenay | i'll go read docs :) |
| 18:11.16 | *** part/#asterisk beniwtv (n=beniwtv@124.Red-83-36-62.staticIP.rima-tde.net) |
| 18:11.54 | [TK]D-Fender | hugenay: read up on "call files" on the WIKI, this will be your call-out mechanism with a minimal dialplan. |
| 18:11.56 | [TK]D-Fender | ~wikis |
| 18:11.57 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 18:12.45 | [TK]D-Fender | masus: If you are calling out then you need to set that option in your Dial statement. This is not Queue's job |
| 18:13.08 | flohack | [TK]D-Fender: Thanks for your help, I'll rethink my network setup |
| 18:14.15 | masus | [TK]D-Fender: sorry i'm not calling out it's an incoming call |
| 18:14.52 | *** join/#asterisk IvanG (n=IvanG@78.52.233.150) |
| 18:16.19 | [TK]D-Fender | masus: then it depends how your agent is called |
| 18:16.33 | masus | [TK]D-Fender: Action: Originate |
| 18:16.49 | masus | from PHP to manager.conf |
| 18:16.56 | [TK]D-Fender | masus: what does that have to do with a queue? |
| 18:17.10 | [TK]D-Fender | masus: it depends how the QUEUE calls your "agent" |
| 18:17.24 | masus | hmmm |
| 18:17.25 | hugenay | thanks again |
| 18:17.26 | hugenay | bye |
| 18:18.22 | *** join/#asterisk twisted (n=twisted@router.asteriasgi.com) |
| 18:18.22 | *** mode/#asterisk [+o twisted] by ChanServ |
| 18:22.59 | masus | [TK]D-Fender: http://rafb.net/p/zb7bKQ48.html here is teh sample configuration |
| 18:23.07 | masus | can u take a look please ? |
| 18:23.53 | [TK]D-Fender | masus: This does not show me how your agent is called.... |
| 18:24.00 | masus | hmmm |
| 18:24.02 | [TK]D-Fender | masus: please show me an ACTUAL call. |
| 18:24.32 | [TK]D-Fender | masus: and this : exten => s,n,Dial(SIP/${ARG1}${ARG2}); <-- does not contain an option to hangup via "*" |
| 18:24.39 | [TK]D-Fender | masus: that requires a DIAL parameter |
| 18:24.47 | masus | oh now i understand |
| 18:24.52 | masus | ok i'll see one mom |
| 18:30.55 | *** join/#asterisk anakin_ (n=xxxx@a83-132-132-211.cpe.netcabo.pt) |
| 18:36.59 | *** join/#asterisk bn43 (n=dhashen@41.26.239.239) |
| 18:38.29 | SparFux | Sb said with type= the "peer" and the "friend" is basically the same. There was no difference and one should use "peer". But with "peer" calls don't get connected. |
| 18:39.25 | Greek-Boy | has anyone here successfully setup Kannel? |
| 18:39.43 | jsmith | SparFux: Technically, I'm not sure that a peer and friend are *exactly* the same. They're close, but not identical. |
| 18:39.51 | *** join/#asterisk mercutioviz (n=chatzill@freeswitch/developer/msc) |
| 18:39.55 | Gido-E | it is not the same |
| 18:40.23 | SparFux | jsmith: But how come when I change friend to peer in a working setup, it does not even find the called extension anymore? |
| 18:40.30 | Gido-E | cal a peer a peer and a user an user. |
| 18:40.58 | Greek-Boy | and if its a user and peer then its a friend |
| 18:41.00 | [TK]D-Fender | SparFux: pastebin is your friend... |
| 18:41.01 | Gido-E | SparFux are u using callerid in your sipconf? |
| 18:41.25 | [TK]D-Fender | SparFux: There was a large article about how friend & user are being phased out |
| 18:41.34 | bn43 | Hi I have installed asterisk from source on ubuntu hardy and asterisk gui 2 from svn - I'm trying to add a user on the gui but it says number not in preferred range |
| 18:41.39 | [TK]D-Fender | SparFux: So show us what you're doing and we'll advise from there |
| 18:41.45 | bn43 | google does not help |
| 18:42.06 | [TK]D-Fender | bn43: GUI's are not supported in this channel. Please refer to the linked channels in the topic |
| 18:42.24 | bn43 | oh sorry |
| 18:46.26 | Rabenklaue | After the setup of asterisk I'm using a SIP-softphone (ekiga) to call my server. It works as expected. But only locally. If I try to call myself via internet (with a dyndns address) it doesn't work |
| 18:47.02 | Rabenklaue | All ports are open and redirected to my (asterisk)server. Where can I edit this behaviour within the asterisk config files? |
| 18:47.25 | *** join/#asterisk fexy (n=fexy@208.3.217.29) |
| 18:47.54 | SparFux | And now I even got a different error: http://pastebin.com/d422bfbc0 |
| 18:49.08 | [TK]D-Fender | SparFux: please enable SIP DEBUG from CLI, you aren't looking at whats actually happening |
| 18:51.22 | SparFux | sorry, have to go. I'll be right back later on. |
| 18:51.25 | SparFux | Thx so far. |
| 18:51.29 | jsmith | SparFux: The reason is because of the way incoming calls are matched against users/peers/friends. |
| 18:51.40 | jsmith | SparFux: Ping me when you get back and I'll explain in more detail |
| 18:52.45 | [TK]D-Fender | jsmith: Difficult to express how much I trust things when the channel name flags as an IP and not a peer name :) |
| 18:53.06 | [TK]D-Fender | jsmith: But the term "zero" comes to mind... |
| 18:53.07 | *** join/#asterisk ghenry (n=ghenry@92.41.138.186.sub.mbb.three.co.uk) |
| 18:53.21 | jsmith | [TK]D-Fender: Amen, brother... at least we're on the same page, then :-) |
| 18:53.41 | [TK]D-Fender | jsmith: ClueBat at the ready, as always... |
| 18:53.56 | [TK]D-Fender | pets his trust ClueBat (tm) |
| 18:54.02 | [TK]D-Fender | trusty* |
| 18:54.32 | jsmith | grabs his handy clue-by-four and gets to work |
| 18:55.04 | thehar | haha |
| 18:56.00 | jsmith | There's more than one way to educate people :-p |
| 18:56.18 | thehar | I like that idea. |
| 18:56.32 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 18:57.35 | [TK]D-Fender | thehar: strike a point home and make a big impression and the creative flow will have genes pooling in no time! |
| 18:57.55 | [TK]D-Fender | goes to replace the plastic sheeting on the floor |
| 18:58.07 | thehar | haha |
| 18:58.41 | fexy | Is there a way to have asterisk dynamically accept(not reject) SCCP phones? |
| 18:58.54 | thehar | cookie mysql |
| 18:59.00 | fexy | I see mention of using mysql with sip, but none with using skinny. |
| 18:59.01 | thehar | oops.. stupid mouseover window failure! |
| 18:59.10 | *** join/#asterisk edibrac (n=elusive4@206.173.193.34.ptr.us.xo.net) |
| 19:06.16 | *** join/#asterisk chi6IT41 (n=chigital@tmo-104-25.customers.d1-online.com) |
| 19:10.00 | thansen | what's the current state of chan_mobile? |
| 19:10.06 | *** join/#asterisk sekil (n=Ognjen@80.93.247.26) |
| 19:10.19 | thansen | easy to use? work with most cell phones? etc? |
| 19:11.26 | [TK]D-Fender | thehar: http://www.voip-info.org/wiki/view/chan_mobile |
| 19:11.30 | [TK]D-Fender | thansen: rather |
| 19:12.13 | thansen | [TK]D-Fender: I looked at that but didn't look very recent (oct 2007) |
| 19:13.07 | thehar | heh |
| 19:13.52 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 19:16.31 | [TK]D-Fender | thansen: And I'm sure you could look at the tracker # to get newer info, etc. Its called "trying" |
| 19:17.19 | thansen | apologizes for asking a question to a forum |
| 19:17.28 | [TK]D-Fender | thansen: and if you look at the actual page history you'll see it updated all throughout 2008 |
| 19:17.47 | [TK]D-Fender | thansen: So it could very well be "current" |
| 19:18.56 | SparFux | re |
| 19:19.31 | hardwire | bar |
| 19:20.40 | [TK]D-Fender | foo |
| 19:20.45 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:20.45 | SparFux | hehe :-) |
| 19:20.51 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 19:22.01 | SparFux | Thx for your will to help, but I think Ill just keep working friend setup and first have to READ a little more about this stuff. Otherwise I jus twaste your time. |
| 19:25.13 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 19:25.44 | *** join/#asterisk el_- (n=el@mnch-4d0435c9.pool.mediaWays.net) |
| 19:25.48 | el_- | hi all |
| 19:25.56 | edibrac | our telco dropped off a Westell monitoring unit that is between the NIU and CPE -- when it says there are problems on the CPE side, does that necessarily mean the errors are caused on the CPE side? |
| 19:26.34 | edibrac | that a bad NIU or bad crimp on the coax to the NIU could case errors coming from the CPE side? |
| 19:27.30 | el_- | I got the following problem: When I try to call via my asterisk everything is fine... but the one that is called cannot hear me! The same happens when someone tries to call me... I hear the counterpart doesn't ... any ideas? |
| 19:27.41 | el_- | The Asterisk has a public IP and no firewall |
| 19:27.48 | edibrac | The main problem is dropped calls from random HDLC errors, which I've gone through the usual to troubleshoot (interrupts, PCI timing, swapping asterisk cards) |
| 19:28.17 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 19:35.57 | *** join/#asterisk martyn-dev (n=admin@190.24.134.154) |
| 19:36.03 | martyn-dev | Hi everybody |
| 19:36.22 | martyn-dev | I want to know what is the name of the module of control manager.conf (AMI) .. |
| 19:36.58 | fexy | chan_sccp2 works with asterisk realtime |
| 19:37.01 | fexy | sweet |
| 19:37.08 | martyn-dev | really. i've been added a new user on manager.conf.. soo i need reload the configuration.. but i can't reload all.. i need only reload module of manager .. |
| 19:39.53 | SuPrSluG | hi all |
| 19:40.02 | SuPrSluG | what causes Got SIP response 302 "Moved Temporarily" back from ? |
| 19:40.53 | SuPrSluG | we're trying to redirect some number from old system to a new system. |
| 19:40.56 | [TK]D-Fender | martyn-dev: straight "reload" should do it |
| 19:41.11 | [TK]D-Fender | SuPrSluG: Forwarding on the device |
| 19:41.30 | martyn-dev | ok. thanks |
| 19:42.36 | SuPrSluG | from oldpbx -> newpbx (different machine). |
| 19:43.38 | *** join/#asterisk kisu_ (n=kisu@2001:5c0:1100:9900:acbb:dcfd:e13c:5740) |
| 19:44.28 | SuPrSluG | some number work fine, others don't. strange |
| 19:45.24 | *** join/#asterisk telnettech (i=telnette@gw.percipia.com) |
| 19:47.24 | *** join/#asterisk bbryant (n=Brett_Br@adsl-068-016-200-248.sip.chs.bellsouth.net) |
| 19:50.49 | voipnet-tech | anyone got a hack for app_voicemail to work with freepbx voicemail blasting to let me be able to forward an existing message to a distribution group? currently of course just get pbx-invalid played to me since its not a real mailbox... help is appreciated :-p job depends on it working |
| 19:51.13 | interfaith | any call transfer guru at large ? can an agi script setup call transfer via hook flash then dial & hangup ? |
| 19:52.02 | *** join/#asterisk jazzplyer (n=jazzplye@222-154-246-214.adsl.xtra.co.nz) |
| 19:52.08 | [TK]D-Fender | voipnet-tech: Go ask in #freepbx |
| 19:52.32 | interfaith | got it |
| 19:53.11 | [TK]D-Fender | interfaith: No AGI required, and yes, if its on a zaptel analog channel |
| 19:53.25 | interfaith | no zaptel available |
| 19:53.34 | [TK]D-Fender | interfaith: What is your interface? |
| 19:53.45 | interfaith | sip or iax etc |
| 19:53.59 | [TK]D-Fender | interfaith: There is no "hookflash" with those really. |
| 19:54.25 | [TK]D-Fender | intthe devices taht speak it work on channels, and can have multiple actual channels in progress. |
| 19:54.33 | interfaith | true, i was thinking of going to a 'new call' tomake a transfer command |
| 19:54.41 | *** join/#asterisk VoipForces (n=courchea@207.107.190.130) |
| 19:55.04 | interfaith | use the new call to send a command to transfer off the 1st call |
| 19:55.08 | VoipForces | Anyone had errors like "include/linux/types.h:16: error: expected '=', ',', ';', 'asm' or '__attribute__' before '__kernel_dev_t'" trying to compile dahdi under kernel 2.6.28 ? |
| 19:55.38 | voipnet-tech | anyone got a hack for app_voicemail to add distribution groups, and the ability to reply to a group of mailboxes, forward to a group, and to leave messages for multiple groups? |
| 19:56.30 | [TK]D-Fender | voipnet-tech: And we heard you 5 minutes ago.... |
| 19:57.03 | voipnet-tech | i asked a different question |
| 19:57.18 | voipnet-tech | one that you can't tell me to go to #freepbx for |
| 19:58.30 | [TK]D-Fender | voipnet-tech: Good luck canvassing for such major mods.... |
| 19:58.36 | interfaith | just a quiz ? if iax2 was hacked to do nat/hole-punching for easy iax2/nat p2p iax2/nat would that be in demand ? for easy asterisk networking ? |
| 19:59.24 | interfaith | basically a form of iax2 p2p call transfer |
| 19:59.31 | [TK]D-Fender | interfaith: The world at large does not care about IAX2. NAT probems are actually rather rare and the only point for it is INTER ASTERISK as its name implies |
| 19:59.34 | jsmith | interfaith: It's certainly get looked at. If you could do it for SIP as well, that'd be even better :-p |
| 20:00.02 | [TK]D-Fender | interfaith: and PBX's are not "p2p", they are typically client-server |
| 20:00.12 | interfaith | well via developing a ?"virtual" interface that does the hole punch ..no asterisk code change is needed |
| 20:00.59 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 20:00.59 | *** mode/#asterisk [+o russellb] by ChanServ |
| 20:01.03 | interfaith | millions of mini asterisk ippbx coming out of china now |
| 20:01.15 | voipnet-tech | every pbx voicemail since 1985 has had the ability to do distribution groups. why doesn't app_voicemail... seems like something that would have been done by now. just asking if anyone knows anyone who is doing it. voicemail blasting in freepbx is incomplete. |
| 20:02.20 | [TK]D-Fender | voipnet-tech: You'v got the source just like the rest of us, feel free to write it. |
| 20:02.25 | jsmith | voipnet-tech: You can do it in app_voicemail |
| 20:02.37 | [TK]D-Fender | voipnet-tech: And we do have distribution goup capabilities, just not "forward to" |
| 20:02.47 | jsmith | voipnet-tech: Voicemail(123@vmcontext&234@vmcontext&222@vmcontext) |
| 20:02.53 | [TK]D-Fender | jsmith: his angle was the "forward to a group" |
| 20:03.10 | jsmith | [TK]D-Fender: Ah... no forwarding to groups yet (as far as I know), but we accept patches |
| 20:03.10 | [TK]D-Fender | jsmith: as in from a pre-existing VM |
| 20:03.11 | voipnet-tech | customer specifically requests forward to and reply to group capabilities |
| 20:03.29 | [TK]D-Fender | voipnet-tech: Then go code it for them :) |
| 20:03.46 | dlewis | anyone have a good experience with the x100p card? |
| 20:03.58 | voipnet-tech | i'm trying to... just seeking help, or to know if it exists already. i hate reinventing the wheel u know |
| 20:04.02 | [TK]D-Fender | dlewis: Ask a specific question and you might get a speicif answer... |
| 20:04.11 | [TK]D-Fender | voipnet-tech: It doesn't |
| 20:04.48 | [TK]D-Fender | specific* |
| 20:04.49 | voipnet-tech | ok thanks. |
| 20:04.54 | n3hxs | dlewis, yes and no |
| 20:05.13 | jsmith | dlewis: I've used the X100p, and I'll never forget the horrors... 'nuf said? |
| 20:05.23 | dlewis | sure |
| 20:05.36 | *** join/#asterisk edwin_quijada (n=macaruch@200.26.172.98) |
| 20:05.38 | n3hxs | They said it will stop working suddenly, I didn't believe them till it did! |
| 20:05.40 | dlewis | i guess you get what you pay for. |
| 20:05.55 | n3hxs | works as a toy or just to play, |
| 20:05.58 | [TK]D-Fender | ~ygwypf |
| 20:05.59 | jbot | somebody said ygwypf was You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 20:06.02 | *** part/#asterisk martyn-dev (n=admin@190.24.134.154) |
| 20:06.16 | mercutioviz | x100p == torture device to encourage one to get a "real" TDM card |
| 20:06.19 | jsmith | dlewis: (Full disclosure... I now work for Digium, so my views are obviously biased... but my experience with the X100P was long before I started working for them, and I stand by what I said.) |
| 20:06.39 | voipnet-tech | gotta love thumbing thru 9200 lines of code with no real idea what i'm doing :-) |
| 20:07.09 | jsmith | voipnet-tech: There's a possibility you could do it with minivm.... I haven't really played with it |
| 20:07.11 | edwin_quijada | <PROTECTED> |
| 20:07.18 | n3hxs | That is one way to learn. 9199 mistakes 1 completion. |
| 20:07.19 | edwin_quijada | i ahve this error with mysql |
| 20:07.34 | [TK]D-Fender | voipnet-tech: They're in good hands! |
| 20:07.51 | [TK]D-Fender | jsmith: No, trashing the X100P is the "party-line" ;) |
| 20:07.54 | edwin_quijada | this error cdr_addon_mysql.c:159 mysql_log: cdr_mysql: cannot connect to database server localhost |
| 20:08.19 | [TK]D-Fender | jsmith: Oh with the thought of MiniVM... I refer you to.. "FreePBX" ;) |
| 20:08.21 | dlewis | jsmith: thanks for the hnoesty |
| 20:08.28 | jsmith | [TK]D-Fender: I've been trashing it long before I started working for Digium, and I'll still be trashing it afterward. It's just a plain lousy no-good card. |
| 20:08.33 | Khratos | edwin_quijada: did you verify that you correctly set username/password ? |
| 20:08.45 | edwin_quijada | Ibut when I connecct from command line with this user I get conecction |
| 20:08.45 | [TK]D-Fender | dlewis: Its CrapTASTIC! ShitACULAR even! |
| 20:08.55 | edwin_quijada | Khratos: yes |
| 20:09.04 | voipnet-tech | i might as well quit my job now i'm gonna get fired over this. a system was sold and the customer requested this specific functions when we replaced their nortel system... i shouldn't have assumed that * would come with a basic feature. now it's pretty much add it or be fired oh well |
| 20:09.05 | edwin_quijada | I tried from comand line |
| 20:09.11 | Khratos | Well, Edwin, entonces prueba la ruta para el socket de mysql, esta bien seteada? |
| 20:09.32 | Khratos | netstat -l | grep -i mysql |
| 20:09.35 | [TK]D-Fender | voipnet-tech: Go post a bounty or hire a programmer |
| 20:09.41 | edwin_quijada | Khratos: do u speak spanish? |
| 20:09.45 | dlewis | [TK]D-Fender: yea, thanks... looks like I might just get the X100P to test and then try to get a B600D card... |
| 20:09.47 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:09.47 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 20:09.47 | *** join/#asterisk mascool (n=george@adsl-99-178-196-46.dsl.sfldmi.sbcglobal.net) |
| 20:10.01 | Khratos | I think so, but I think this is an english channel |
| 20:10.05 | [TK]D-Fender | lmadsen: Na Na Na Na Na! |
| 20:10.06 | lmadsen | it is |
| 20:10.17 | lmadsen | [TK]D-Fender: I don't want to know your name! |
| 20:10.18 | jsmith | lmadsen: Don't you have work to do? |
| 20:10.20 | edwin_quijada | Khratos: yes! |
| 20:10.21 | jsmith | ducks and hides |
| 20:10.25 | lmadsen | jsmith: nope! I closed all the bugs. |
| 20:10.26 | [TK]D-Fender | lmadsen: I just want... |
| 20:10.31 | lmadsen | [TK]D-Fender: ! ! ! |
| 20:10.47 | mascool | Do you guys know if it's possible to connect a nortel PBX to an asterisk box over a PRI line? (TE110P on the asterisk box) |
| 20:10.55 | voipnet-tech | yup |
| 20:10.59 | voipnet-tech | mascool, yep done that |
| 20:11.00 | *** join/#asterisk exvito (n=exvito@80.172.25.71) |
| 20:11.03 | jsmith | mascool: Yes... |
| 20:11.19 | mascool | voipnet-tech, so basically I would just need to set the correct framing |
| 20:11.32 | mascool | and all the PRI params |
| 20:11.45 | edwin_quijada | Khratos: it seeems the problem is the socket |
| 20:11.56 | Khratos | :) go ahead and fix it |
| 20:12.20 | voipnet-tech | mascool, yes as long as all the params are set correctly on both sides you can use a t1 crossover cable between the two systems, then of course you need the dialplan to account for number dialing on either side |
| 20:12.36 | mascool | voipnet-tech, awesome! thanks! |
| 20:12.39 | dlewis | anyone have any luck hooking up a Panasonix KX-TD1232-7 with asterisk? |
| 20:12.45 | n3hxs | and only set one as the master clock |
| 20:12.48 | edwin_quijada | Khratos: I need to reload asterisk |
| 20:12.49 | edwin_quijada | ? |
| 20:12.59 | Khratos | Yes, or at least modules |
| 20:14.03 | voipnet-tech | mascool,, n3hxs makes a good note, one side should have internal clock, the other should have external... i'm not sure which to suggest would give better clocking. we did it on the * box |
| 20:14.29 | *** part/#asterisk exvito (n=exvito@80.172.25.71) |
| 20:14.35 | mascool | I'll do the same then :) |
| 20:14.50 | Khratos | I'm impatient about when will asteriskdocs -dot- org will come out |
| 20:14.55 | n3hxs | it will sorta run with neither being the master clock, but every so often it will drop all calls. |
| 20:16.12 | voipnet-tech | we actually had a fun time with our system... we actually networked the two systems via fiber. we got a device called an IP tube to do clear channel T1 over fiber, which we did PRI over. the two systems were 3 miles apart. lol memories. haven't any jobs like that in about 5 years |
| 20:18.03 | mascool | voipnet-tech, that sounds like fun .. |
| 20:18.42 | mascool | do you know any good ,affordable 8xFXS gateways ? |
| 20:19.06 | voipnet-tech | i'm biased to adtran and quintum |
| 20:19.07 | mascool | the Digium TDM cards are very expensive .. |
| 20:19.28 | n3hxs | mascool, Depends on who's purse you have |
| 20:19.39 | mascool | voipnet-tech, do you know if it's possible to have 8 separate registration for each port? |
| 20:19.42 | voipnet-tech | grandstream makes one. we've got a couple deployed, but sometimes they're difficult |
| 20:19.48 | voipnet-tech | yes mascool u can |
| 20:19.55 | mascool | sweet |
| 20:20.23 | mascool | voipnet-tech, from now on, I'll only ask you and questions I have, you only have good news |
| 20:20.30 | mascool | and=any* |
| 20:21.20 | voipnet-tech | yay i found a fan |
| 20:21.37 | mascool | sarcastic too |
| 20:21.38 | mascool | :) |
| 20:21.45 | voipnet-tech | too bad i'm probably not keeping my job much longer |
| 20:21.52 | mascool | why is that ? |
| 20:22.12 | voipnet-tech | sold a system with a function i thought * came with default but doesn't... now it's code it or die pretty much |
| 20:22.23 | mascool | oh crap, been there |
| 20:22.36 | edwin_quijada | how can I get the time when an agent login to asterisk |
| 20:22.51 | mascool | voipnet-tech, what do you need to do ? |
| 20:23.11 | voipnet-tech | mascool, voicemail distribution groups with the availablity to forward and reply to one or more groups |
| 20:23.42 | *** join/#asterisk SuPrSluG__ (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 20:23.51 | mascool | good luck, man ... |
| 20:24.03 | [TK]D-Fender | edwin_quijada: go look at your Queue Log |
| 20:24.32 | edwin_quijada | [TK]D-Fender: this information doesnt save into mysql table? |
| 20:24.47 | edwin_quijada | where is Queue log? |
| 20:24.52 | [TK]D-Fender | edwin_quijada: It does if you configure it to |
| 20:25.12 | [TK]D-Fender | edwin_quijada: Look in your var/log/asterisk folder if it saves to CSV as well |
| 20:25.16 | *** join/#asterisk kisu (n=kisu@2001:5c0:1100:9900:acbb:dcfd:e13c:5740) |
| 20:25.46 | *** join/#asterisk Tako-san (n=Tako-san@96.50.64.203) |
| 20:26.21 | Tako-san | Anyone ever got a Sangoma card to be recognized using ESXi? |
| 20:26.34 | mascool | audiocodes any good ? |
| 20:27.00 | [TK]D-Fender | mascool: Sure. What are you looking to do? |
| 20:27.29 | mascool | this is a different job. i need to connect 8 analog lines to a nortel PBX |
| 20:27.52 | edwin_quijada | [TK]D-Fender: this info can be configured to mysql table? |
| 20:28.02 | edwin_quijada | where can I find info about this conF? |
| 20:29.23 | manxpower | voipnet-tech: Asterisk has voicemail groups |
| 20:30.24 | [TK]D-Fender | edwin_quijada: Go read THE BOOK... its all in there.. and in the sample configs |
| 20:30.25 | [TK]D-Fender | ~book |
| 20:30.26 | jbot | well, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 20:32.21 | mascool | wow, the grandstream gateways are 1/4 the price of audiocodes and adtran |
| 20:32.45 | manxpower | mascool: But they work 8 times as bad. |
| 20:32.57 | mascool | yeah, that's what I as thinking .. |
| 20:32.59 | mascool | was* |
| 20:33.18 | manxpower | your nortel supports SIP? |
| 20:33.50 | mascool | I wish ... |
| 20:34.07 | [TK]D-Fender | mascool: What signalling? |
| 20:34.27 | mascool | [TK]D-Fender, it's all analog |
| 20:34.41 | [TK]D-Fender | mascool: what kind of PORT on the nortel? |
| 20:34.41 | mascool | that's all I know, it's got 4 POTS lines from AT&T |
| 20:35.09 | mascool | [TK]D-Fender, what do you mean what kind of port ? |
| 20:35.11 | manxpower | mascool: so what exactly is the problem with plugging 4 analog lines into your nortel? |
| 20:35.15 | [TK]D-Fender | masFXO, or FXS <- |
| 20:35.26 | mascool | it's a nortel Ox32 (MCIS) |
| 20:35.33 | [TK]D-Fender | mascool: And why are you looking for 8 ports? |
| 20:35.38 | manxpower | mascool: my condolences |
| 20:35.51 | Kobaz | is there an equivalent type of command to 'pri' when using loop start, or e&m wink, etc, on a t1... so see the alarms and etc |
| 20:35.54 | mascool | manxpower, :) why ? |
| 20:36.11 | mascool | [TK]D-Fender, we're also expanding from 4 to 8 |
| 20:36.16 | mascool | but that's a secondary issue |
| 20:36.18 | manxpower | Kobaz: no. |
| 20:36.21 | Kobaz | aww |
| 20:36.23 | manxpower | zttool will tell you |
| 20:36.37 | Kobaz | mmm |
| 20:36.39 | *** join/#asterisk rue_mohr (n=rue@24.207.122.10) |
| 20:36.39 | mascool | first we need to figure out how to connect at least the existing 4 ports to an IP gateway |
| 20:37.10 | manxpower | mascool: I've connected Asterisk and a MICS before. |
| 20:37.42 | mascool | manxpower, any tips? |
| 20:37.43 | mascool | :) |
| 20:37.43 | manxpower | In order for it to be reliable we had to use T-1 E&M/Wink between the two systems. |
| 20:38.38 | mascool | manxpower, noted |
| 20:39.02 | rue_mohr | if I get oslec working, is it prettymuch as good as the hardware echo can? |
| 20:39.30 | mascool | manxpower, do you know if the TE110P supports that signaling? |
| 20:39.33 | *** part/#asterisk sekil (n=Ognjen@80.93.247.26) |
| 20:39.49 | *** join/#asterisk rootforce (n=chatzill@office.aircanopy.net) |
| 20:42.32 | [TK]D-Fender | mascool: Sounds like you want to connect to LINE ports, not STATION ports. |
| 20:42.41 | mascool | [TK]D-Fender, yes |
| 20:43.02 | dlewis | Merlin: lol... when I called voipe-pulse twice, for some reason I kept getting disconnected... i guess that'll be my future if I go with them. |
| 20:43.11 | rue_mohr | [TK]D-Fender, was it a nme conflict that made zaptel -> dahdi? |
| 20:44.03 | rootforce | yes |
| 20:44.24 | rue_mohr | so zaptel isn't that old still? |
| 20:44.26 | [TK]D-Fender | masfor that You'd effectively plug them into an ATA. For this you DO have 1 nice cost-effective choice : Linksys SPA-8000 |
| 20:44.42 | [TK]D-Fender | rue_mohr: ...huh? |
| 20:44.53 | rue_mohr | looks like the oslec will only work with zaptel, not working for dahdi |
| 20:44.57 | rootforce | http://blogs.digium.com/2008/05/19/zaptel-project-being-renamed-to-dahdi/ |
| 20:45.05 | [TK]D-Fender | mascool: http://www.voiplink.com/Linksys_SPA_8000_p/linksys-spa-8000.htm |
| 20:45.17 | rootforce | i am not a fan of the 800 |
| 20:45.19 | rootforce | 8000 |
| 20:45.38 | [TK]D-Fender | mascool: Better price : http://www.888voipstore.com/linksys-spa-8000-pr-18784.html |
| 20:45.39 | rootforce | it is essentially an spa2102 with 3 pap2s |
| 20:46.07 | rootforce | in a box |
| 20:46.11 | [TK]D-Fender | rootforce: Or 4 x 2102 as it supports G.729 across all ports, etc |
| 20:46.20 | mascool | awesome, thanks [TK]D-Fender |
| 20:46.25 | [TK]D-Fender | rootforce: I never said it was "awesome, jsut a great value |
| 20:46.43 | rootforce | if you look at the configuration it has ports forwarded to internal devices |
| 20:46.53 | rue_mohr | rootforce, can I still download the last version of 'zaptel' ? |
| 20:47.06 | rootforce | and the first 3 ip addresses in the router portion are taken |
| 20:47.06 | [TK]D-Fender | rootforce: Certainly gets the job done, reliable, good wiring options at a price that can't be beat by anything nearly as decent |
| 20:47.16 | rootforce | you have me there |
| 20:47.26 | rootforce | i have yet to find a good alternative |
| 20:47.39 | rue_mohr | looks like not off digium, all the links are broken |
| 20:47.45 | [TK]D-Fender | rootforce: Yeah, it is 1 port per device and is an internal daisychain lookin' kind of thing, but everyone I know who's used one has been happy |
| 20:47.49 | rootforce | i just had re-registration problems on the 6 lines that are behind nat |
| 20:48.07 | [TK]D-Fender | rootforce: Wierd... |
| 20:48.28 | rootforce | but if you don't use qualify you are ok or if your registration timers are very short |
| 20:48.42 | rue_mohr | I'm cuaght between things, I cant make oslec work with dahdi, and zaptel dosnt' exist anymore |
| 20:48.48 | mascool | rootforce, doesn't qualify help keep the nat holes open ? |
| 20:49.09 | [TK]D-Fender | rue_mohr: I've linked guides for this before... |
| 20:49.09 | rue_mohr | and I have to go back to a worksite, how can I solve this in a reasonable about of time? |
| 20:49.15 | [TK]D-Fender | rue_mohr: Works fine for everyone else |
| 20:49.17 | rue_mohr | I tried it, didn't work |
| 20:49.35 | [TK]D-Fender | rue_mohr: And you can't show us so we can comment on it and help get ti fixed |
| 20:49.37 | rootforce | i should not have said qqualify |
| 20:49.45 | [TK]D-Fender | rue_mohr: Your "hindsight" model is defective |
| 20:49.54 | rue_mohr | everytime I turn on the echo canceling, asterisk dosn't load the dahdi channel driver |
| 20:50.10 | rootforce | what i mean is that if asterisk loses the registration info for some reason such as a sip reload then the other 6 lines will not know to come back |
| 20:50.24 | rootforce | they will still think they are registered |
| 20:50.26 | tzafrir_laptop | rue_mohr, what do you mean by "turn on echo cancelling"? |
| 20:51.01 | rue_mohr | I'll go over it more later, I had no means to know WHY the channel driver disn't come up when echo was turned on (in the file with the channels you turn echo on and off for channels that I cant remmebr the name of right now) |
| 20:51.56 | *** join/#asterisk bmoraca (n=bmoraca@209.60.253.58) |
| 20:52.02 | rue_mohr | echocanceller=oslec,5-8 system.conf when i enable that line the dahdi fails to come up in core show channeltypes |
| 20:52.04 | rue_mohr | bye! |
| 20:52.29 | tzafrir_laptop | this makes dahdi_cfg fail and thus channels are left unconfigured? |
| 20:53.00 | tzafrir_laptop | rue_mohr, look at /proc/dahdi/* in both cases |
| 21:03.01 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 21:10.12 | *** join/#asterisk bn43 (n=dhashen@41.26.216.254) |
| 21:10.59 | bn43 | hi I am trying to register my snom320 phone but they both say NR on display - yet I can go onto web interface and ping them |
| 21:12.10 | bn43 | the log files say Registra number@IP timed out |
| 21:12.13 | *** join/#asterisk ZX81 (n=matt@202.49.106.158) |
| 21:12.31 | ZX81 | hi all, can anyone recommend a multi party video conferencing thingy? |
| 21:12.36 | ZX81 | i.e. media mixer etc |
| 21:12.49 | ZX81 | kinda wanting to stay away from the sip.fontventa.com one |
| 21:13.02 | ZX81 | wouldn't mind using the red9 one if someone knows it works |
| 21:14.03 | SkramX | im confused. what happens if a gotoif() is never matched |
| 21:14.16 | ZX81 | continues in dialplan |
| 21:14.23 | SkramX | how do i overwrite that? |
| 21:14.35 | ZX81 | ~wiki |
| 21:14.40 | ZX81 | hmm |
| 21:14.46 | SkramX | i dont see it on thew wiki |
| 21:14.55 | ZX81 | http://www.voip-info.org/wiki-Asterisk+cmd+GotoIf |
| 21:15.07 | ZX81 | GotoIf(condition?label1[:label2]) |
| 21:15.18 | ZX81 | label1 = true, label2 = false |
| 21:15.29 | ZX81 | i.e. GotoIf($["${CALLERID(num)}" != "304"]?moh:dial2) |
| 21:15.50 | SkramX | exten=>s,2,Read(ACCEPT|mcc-agent-ackcall|1) |
| 21:15.50 | SkramX | exten=>s,3,Gotoif($[${ACCEPT} = 1] ?50) |
| 21:15.53 | SkramX | thats what i have |
| 21:16.03 | SkramX | confusing. |
| 21:16.13 | SkramX | i think i can just do an absolute timeout |
| 21:16.35 | bn43 | Hi can any help me with snom320 phone? they are not registring - show NR on display |
| 21:16.59 | ZX81 | SkramX: so |
| 21:17.08 | ZX81 | will go to 50 |
| 21:17.09 | ZX81 | if true |
| 21:17.12 | ZX81 | and next if not |
| 21:17.13 | SkramX | yes |
| 21:17.17 | ZX81 | if you want to go to 60 if not |
| 21:17.19 | ZX81 | do 50:60 |
| 21:17.28 | SkramX | okay |
| 21:17.29 | SkramX | hmm |
| 21:17.45 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 21:18.23 | SkramX | ZX81: can you take a look at http://pastie.org/371491 ? |
| 21:18.32 | bn43 | anyone? |
| 21:18.43 | SkramX | so i could really clean this up if on line 4/priority3 just do ?50:30 |
| 21:19.43 | *** join/#asterisk mesfet (n=psubiaco@host165-3-static.25-87-b.business.telecomitalia.it) |
| 21:20.55 | ZX81 | or |
| 21:20.55 | ZX81 | change = to != |
| 21:20.55 | ZX81 | and goto 30 |
| 21:20.55 | SkramX | ok |
| 21:21.36 | SkramX | ill try this |
| 21:21.36 | SkramX | thanks |
| 21:23.36 | ZX81 | np |
| 21:23.49 | mesfet | Hello. I've a question. I need to use MySQL application (in the dialplan) to update a field in a database: my problem is that I don't know how to write a newline character. '\n' does not seems to work. |
| 21:24.14 | *** join/#asterisk psilikon (n=joel@cerberus.vicimarketing.com) |
| 21:24.28 | SkramX | ZX81: http://pastie.org/371498 - if i press #, it still connects |
| 21:24.34 | [TK]D-Fender | checkout time... later all |
| 21:25.00 | [TK]D-Fender | SkramX: exten=>s,3,Gotoif($[${ACCEPT} = 1] ?50:30) ;connect!<_ whitespace = BAD |
| 21:25.02 | rootforce | is there a way to reread sip.conf but not lose realtime cached peers? |
| 21:25.17 | [TK]D-Fender | SkramX: And I don't see a 50 for it to land on anway <- |
| 21:25.36 | SkramX | it doesnt need one, does it? |
| 21:25.42 | SkramX | it just connects the caller and words |
| 21:25.58 | [TK]D-Fender | SkramX: you are telling it to jump somewhere that doesn't exist. Not smart |
| 21:26.05 | SkramX | ok |
| 21:26.06 | [TK]D-Fender | SkramX: mand remove the white-space |
| 21:26.06 | SkramX | ill fix that |
| 21:26.12 | [TK]D-Fender | and* |
| 21:26.14 | SkramX | ok |
| 21:26.18 | SkramX | but what about the # issue? |
| 21:26.24 | bn43 | Hi can any help me with snom320 phone? they are not registring - show NR on display |
| 21:27.24 | [TK]D-Fender | SkramX: And you aren't testing for "#", you're testing for "1" and in a broken way where they can choose to enter NOTHING, which breaks your expression |
| 21:27.45 | SkramX | ok |
| 21:27.48 | SkramX | i changed it to != 1 |
| 21:27.50 | [TK]D-Fender | Ok, go run with it... I'm off |
| 21:27.52 | SkramX | and ?30 |
| 21:27.52 | [TK]D-Fender | later |
| 21:27.56 | SkramX | thanks. |
| 21:28.01 | SkramX | ZX81: ? :) |
| 21:30.17 | *** join/#asterisk jaybeals (n=chatzill@216.195.128.62) |
| 21:30.58 | voipnet-tech | sup noob |
| 21:32.54 | jaybeals | boon pus |
| 21:36.17 | *** join/#asterisk asteriskmonkey (n=philip@69.77.169.14) |
| 21:37.21 | asteriskmonkey | on a multihomed systems, can you run asterisk on all ips and selectivly tell which sip clients to bind to which interface/ip? |
| 21:40.07 | manxpower | asteriskmonkey: I have never heard of anyone making that work |
| 21:40.34 | manxpower | Also SIP clients do not bind to ip/ports on the server. |
| 21:40.56 | manxpower | Generally if you leave it the default asterisk should respond on the best interface for the destination |
| 21:41.16 | manxpower | which may or may not be the IP the phone REGISTERED to. |
| 21:42.42 | *** join/#asterisk DarkRift (n=dark@bas1-sthubert21-1279640708.dsl.bell.ca) |
| 21:42.59 | rootforce | you should be able to use linux routing to have an effect on that |
| 21:43.07 | asteriskmonkey | yes my problem is im getting asymetric registration happening :p |
| 21:43.39 | asteriskmonkey | its also causing failures with registrations between to systems... |
| 21:44.13 | asteriskmonkey | ag...seems im going to have to split them up into vservers or jails.. id like to run them native as jailed asterisk tends to go wonky for some reasons. |
| 21:45.29 | manxpower | don't expect asymetric password |
| 21:46.10 | asteriskmonkey | no requests come in on one ip and go out on another cuasing registration failures. |
| 21:46.18 | manxpower | then change your routing |
| 21:46.36 | asteriskmonkey | if i was to do that id have to set up each client staticlly. |
| 21:47.18 | asteriskmonkey | its a single asterisk box with a single gateway... has 2 ips, mean for clients to connected to with 2 different internet connections binding the route to the servers ip through one provider and the other through the other |
| 21:48.14 | bmoraca | asteriskmonkey: why do you need to multihome an asterisk box to begin with? |
| 21:49.58 | asteriskmonkey | i have many clients with dsl and wireless as a backup, i keep 2 sip accounts on the multihomed box and assign the dsl and wireless there own individual routes to the respective ips aswell as the a/b sip accounts so to say and do a simple failover, this also gives me status information on both connections and lets me do a few more things all from one asterisk box. |
| 21:50.58 | *** join/#asterisk D3b|4n (n=D3b_4n@unaffiliated/lynxnica) |
| 21:51.24 | D3b|4n | i have a problem Got SIP response 503 "Service Unavailable" back from *.*.*.* |
| 21:52.42 | bmoraca | asteriskmonkey: sounds like you're going about failover in the wrong way. if it's really an issue, failover should be done at the gateway and any application servers (PBX included) should be completely oblivious to it |
| 21:57.34 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
| 22:00.30 | *** join/#asterisk mags2 (n=mags2@ampulex.whoi.edu) |
| 22:02.01 | mags2 | any recommendations for 24 (or 48) port ATA? ('d really rather use a channel bank or even an extra asterisk box, but there are reasons...) |
| 22:02.15 | Miccster | anyone know where to buy telecom testing tools? I need to get tone testers and punch down tool and stuff for installing customer equipment. Where can I get the best deal on that stuff? |
| 22:02.22 | Miccster | It all seems so expensive. |
| 22:02.35 | asteriskmonkey | Miccster: canada, usa, where? |
| 22:06.07 | n3hxs | Check pimfg.com for some tools. |
| 22:06.26 | bmoraca | mags2: mediatrix or dialogic...www.voiplink.com sells them. |
| 22:06.45 | n3hxs | Miccster, but it has been a couple of years since I bought tools from them. |
| 22:06.47 | mags2 | bmoraca: thanks. what about vega that was recommended to me? |
| 22:07.36 | n3hxs | Miccster, you can get hammers and chisles at Home Depot. |
| 22:08.36 | bmoraca | mags2: no experience with them. |
| 22:09.02 | D3b|4n | i have a problem Got SIP response 503 "Service Unavailable" back from *.*.*.* |
| 22:09.11 | bmoraca | Miccster: the good stuff is expensive. PI sells decent stuff that's fairly cheap. As does Startech. But if you want the good stuff, it's expensive. |
| 22:10.17 | mags2 | bmoraca: k thank you |
| 22:10.17 | mags2 | Miccster: never ever buy a $15 punch tool if you see one. ever. |
| 22:15.34 | n3hxs | Good for 25 pair.;) |
| 22:15.44 | n3hxs | just once. |
| 22:17.14 | *** join/#asterisk ocnarf (n=chatzill@122.2.249.114) |
| 22:18.12 | D3b|4n | i have a problem Got SIP response 503 "Service Unavailable" back from *.*.*.* |
| 22:18.13 | manxpower | Does anyone know the best way to mix AEL and regular extensions.conf stuff? |
| 22:18.14 | *** join/#asterisk JAMMAN2110 (n=James@unaffiliated/jamman2110) |
| 22:18.54 | *** join/#asterisk genoobie (n=genoobie@pool-72-65-17-165.bflony.east.verizon.net) |
| 22:18.59 | genoobie | hey all |
| 22:19.13 | genoobie | still investigating VoIP options / services providers |
| 22:19.36 | JAMMAN2110 | Hello genoobie |
| 22:19.39 | *** join/#asterisk UQlev (n=kvirc@91.184.220.73) |
| 22:19.51 | genoobie | JAMMAN2110, are you in the US? |
| 22:20.00 | JAMMAN2110 | No, New Zealand :) |
| 22:20.06 | genoobie | argh :) |
| 22:20.30 | genoobie | I'm trying to find out if Vonage is the way to go or I should investigate into cheaper alternatives |
| 22:21.00 | *** join/#asterisk BadHAL (n=wut@cpe-72-179-194-139.stx.res.rr.com) |
| 22:21.39 | bmoraca | genoobie: depends on what your requirements are. do you need trunking or just service? |
| 22:21.52 | genoobie | just service |
| 22:21.59 | D3b|4n | help me |
| 22:22.07 | genoobie | just simple residential service |
| 22:22.23 | bmoraca | then they're a dime a dozen. take your pick. if cost is your concern, try Ooma. |
| 22:22.26 | JAMMAN2110 | D3b|4n ? |
| 22:22.34 | D3b|4n | i have a problem |
| 22:22.37 | ocnarf | Guys, I need some advice. I have an asterisk server at location A. Then i have 50 phones to register to server A from location B and another 50 phones to register to server A from location C. |
| 22:22.47 | D3b|4n | <PROTECTED> |
| 22:23.02 | genoobie | bmoraca, are there advantages to using a "bundled" origination / termination pkg? |
| 22:23.06 | ocnarf | will that cause me alot of problem? |
| 22:23.26 | genoobie | Vonage has a pkg that is $25 / mo unlimited |
| 22:23.33 | frogonwheels | D3b|4n: you've said. Have a log at sip set debug and the conversation that happens. possibly more enlightening. |
| 22:23.51 | bmoraca | genoobie: any residential service you get is going to include both origination and termination. |
| 22:23.51 | frogonwheels | D3b|4n: some hint of what you are trying to connect to would be advantageous. |
| 22:24.09 | bmoraca | genoobie: Ooma is $200 one-time-charge with no monthly recurring fee |
| 22:24.27 | bmoraca | genoobie: also, read fine print...Vonage only includes 5000 minutes per month |
| 22:24.41 | codefreeze-lap | manxpower: what would you like to do? |
| 22:25.01 | genoobie | right but 5000/mo is going to be enough |
| 22:25.14 | genoobie | Over a 3 mo period I have 1500 min |
| 22:25.20 | genoobie | (incoming & outgoing) |
| 22:25.45 | *** join/#asterisk bn43 (n=dhashen@41.26.216.254) |
| 22:25.47 | genoobie | hmm...but this Ooma sounds interesting. What about call quality of diff. providers, all pretty similar? |
| 22:26.39 | *** join/#asterisk rpm (n=rpm@S010600055d2cf2e2.cg.shawcable.net) |
| 22:26.46 | bn43 | Hi my snom phones are now registered but I cannot make a call between the 2 phones on the internal lan - the gui shows the extensions are registered |
| 22:27.00 | bmoraca | genoobie: just about. you'll always get sound issues...one-sided calls, static, etc...it's the nature of VoIP. if you can't live with that and potential outages, keep your copper. if you can make the sacrifice to save a little money, then VoIP would be fine for you. |
| 22:27.05 | bn43 | what can I do to troubleshoot? |
| 22:27.22 | genoobie | no I have VoIP and I'm fine with it now |
| 22:27.50 | genoobie | what I was wondering was if there were differences b/w the diff. service providers. |
| 22:27.56 | bmoraca | bn43: for starters, pastebin your sip.conf and extension.conf files. also, what does the asterisk console say when you attempt the call. |
| 22:27.59 | rootforce | always |
| 22:27.59 | bmoraca | ~pb |
| 22:28.14 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 22:28.14 | bmoraca | no jbot? |
| 22:28.17 | bmoraca | there it is |
| 22:28.22 | manxpower | codefreeze-lap: I'd like to use both extensions.conf and extensions.ael |
| 22:28.34 | bn43 | bmoraca: I just have the gui - what is the console? |
| 22:28.37 | codefreeze-lap | manxpower: done. You can already. |
| 22:28.45 | bmoraca | bn43: what GUI? |
| 22:29.03 | bn43 | asterisk-gui |
| 22:30.01 | genoobie | bmoraca, one last question.... |
| 22:30.16 | bn43 | ah - you rang a bell - I went to the command line and typed in "asterisk -r" |
| 22:30.27 | bmoraca | bn43: you realize that is just a front-end for the standard asterisk config files, right? you still have the console.... ok, good |
| 22:30.28 | bn43 | error is that the extensions don't exist |
| 22:30.31 | bmoraca | genoobie: shoot. |
| 22:30.38 | genoobie | suppose I go with this Ooma thing. $250 is the one time fee. What if the company goes under? |
| 22:30.46 | bmoraca | bn43: what's your dialplan look like (extensions.conf)? |
| 22:30.48 | genoobie | or is it secure enough to trust |
| 22:31.06 | bn43 | I am pasting now |
| 22:31.23 | manxpower | codefreeze-lap: My question was "how". i.e. in extensions.ael put #include extensions.conf or in extensions.conf put #include extensions.ael? |
| 22:31.32 | bmoraca | genoobie: i wouldn't trust their business plan as far as I could throw the building they're in. however, if they last a year, you're already ahead if compared to paying the monthly fee of Vonage. |
| 22:31.36 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 22:32.02 | genoobie | that's true. hrm... |
| 22:32.48 | genoobie | decisions, decisions |
| 22:32.56 | bmoraca | genoobie: they're big enough to have been at CES. the device they use is based on asterisk. |
| 22:32.57 | genoobie | Vonage also has an $18 / mo plan |
| 22:33.21 | genoobie | bmoraca what do you do for home service/ |
| 22:33.30 | bmoraca | genoobie: unfortunately, they're not publicly traded, so there's no way to know how well their model is working. |
| 22:33.54 | *** join/#asterisk riddlebox (n=victoria@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:34.12 | bmoraca | genoobie: i have a good, old-fashioned copper landline. i have DSL service and they require it or charge you $20 extra per month...since landline is only $15, it's a no-brainer. |
| 22:34.49 | genoobie | right, I have DSL / dry loop so I pay no monthly landline charges |
| 22:34.53 | bmoraca | genoobie: I also have a VoIP phone that ties in to the PSTN gateway for my hosted PBXes. |
| 22:34.58 | codefreeze-lap | manxpower: no, they both will get read in, you just put the appropriate in each, and both will then get read in when asterisk starts. |
| 22:34.59 | bn43 | http://pastebin.com/macc3b2 |
| 22:35.08 | manxpower | codefreeze-lap: thanks |
| 22:35.18 | *** join/#asterisk MaliutaLap (n=biteme@203.171.192.7) |
| 22:35.21 | genoobie | bmoraca hold a sec, describe something a little more in detail for me |
| 22:35.43 | genoobie | so suppose I get a voip phone. Would I be able to use it on my dry-loop line? |
| 22:35.57 | Qwell | genoobie: umm, rephrase your question |
| 22:36.03 | manxpower | genoobie: does your dry-loop line have IP on it? |
| 22:36.07 | Qwell | dry-loop implies copper with no services |
| 22:36.23 | manxpower | no IP = No Voice over IP |
| 22:36.35 | bmoraca | genoobie: intarwebs is intarwebs. if you can access the net over the data service, then you shouldn't have a problem. |
| 22:36.47 | bn43 | I actually did look at dialplans and don't see anything untoward there |
| 22:37.00 | bn43 | via the gui that is |
| 22:37.03 | bmoraca | bn43: pastebin your sip.conf as well |
| 22:37.09 | genoobie | okay so in principal I could buy a VoIP and then hook up with some service provider for access to PSTN |
| 22:37.10 | bn43 | ok |
| 22:37.29 | genoobie | but I need a service provider for PSTN access correct? |
| 22:37.31 | manxpower | genoobie: yes, as long as you have an ethernet port with connectivity to the internet |
| 22:37.50 | rpm | does anyone here do any type of trunking with asterisk? when i receive a sip invite it always contains the phone number of the pilot user but in the to: header it contains the dialed number.. are there any settings in asterisk 1.4 to tell it to look elsewhere for the called number instead of the invite uri? |
| 22:38.02 | Qwell | manxpower: I'm still waiting for a VoIP phone with PPPoE support |
| 22:38.39 | bmoraca | genoobie: no...your VoIP service IS your PSTN access. to use VoIP you need internet service. the two can coexist on the same wires but are mutually exclusive. |
| 22:38.39 | asteriskmonkey | Qwell: there is a ton of those already |
| 22:39.01 | bmoraca | Qwell: ewwwww. |
| 22:39.08 | Qwell | asteriskmonkey: link? |
| 22:39.10 | genoobie | hmm...I'm not sure if I unerstand. |
| 22:39.13 | genoobie | *understand |
| 22:39.16 | asteriskmonkey | a company call bb makes some |
| 22:39.28 | genoobie | so I have DSL with dry loop |
| 22:39.39 | asteriskmonkey | there out of asia, im sorry i dont have the direct link i tested there gear when i used to work at williams. |
| 22:39.49 | bmoraca | genoobie: do you have an internet connection? |
| 22:39.55 | genoobie | I make about 500 min outgoing each month |
| 22:39.58 | genoobie | bmoraca yes |
| 22:40.02 | Qwell | asteriskmonkey: the arcade game company? |
| 22:40.10 | manxpower | genoobie: we DON'T CARE how you get your internet. |
| 22:40.14 | bmoraca | genoobie: then you can use VoIP. your VoIP service provider is your PSTN gateway. |
| 22:40.15 | asteriskmonkey | lol no the canadian digium supplier |
| 22:40.29 | manxpower | you could get your internet via alian brain waves for all we care. |
| 22:40.36 | asteriskmonkey | williamsglobal.com (canadian supplier digium gear) |
| 22:40.43 | genoobie | right, so when you buy the phone, you are essentially working with a service provider |
| 22:40.44 | bmoraca | manxpower: clearwire has crappy latency :P |
| 22:40.56 | bn43 | http://pastebin.com/d606370c3 |
| 22:41.09 | genoobie | then there are per minute fees, etc. |
| 22:41.47 | bmoraca | genoobie: yes. Most will not allow you to use just any old phone. They'll typically supply you with an ATA to hook up your existing analog phones. and most do not bill based on usage-rates, but rather flat monthly rates. |
| 22:42.42 | genoobie | right. Okay, I like the sound of the Ooma thing, I'd hate to sign up and have crappy call quality thought :) |
| 22:42.45 | genoobie | *though |
| 22:43.20 | bmoraca | bn43: in the asterisk console, do a "sip show peers" and then a "sip show peer XXX" for each of the two extensions. pastebin the output. use putty to capture if you need to. |
| 22:44.02 | bmoraca | genoobie: call quality will be equivalent to any number of other voip service providers. |
| 22:45.04 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) [NETSPLIT VICTIM] |
| 22:45.10 | genoobie | okay. so vonage versus ooma pretty much the same |
| 22:47.07 | bmoraca | genoobie: until the call gets to the provider's network, they have no control over QoS or anything. this is fundamentally different from copper landlines and is what leads to quality issues. because of this, quality shifts from location to location. i do believe, however, that Ooma has a 30-day money-back policy. |
| 22:47.19 | bn43 | http://pastebin.com/d18c66f39 |
| 22:50.22 | bmoraca | bn43: i don't see anywhere in your dialplan where it's set up for extension-to-extension calling. capture the output of an attempt and then pastebin the results. |
| 22:51.02 | bn43 | [Jan 27 00:48:02] NOTICE[18612]: chan_sip.c:14489 handle_request_invite: Call from '619' to extension '+0027011621' rejected because extension not found. |
| 22:51.34 | bn43 | ahh - its using the enum setting on the phone right? |
| 22:52.24 | bmoraca | bn43: just try dialing "621" from that phone |
| 22:52.30 | codefreeze-lap | manxpower: As a side note, you can call macros you define in extensions.con from AEL; in 1.4, you can call macros you define in AEL from extensions.conf; but in trunk/1.6, you have to use GoSub(). |
| 22:52.31 | SparFux | I have a lot of noise on the line. Even when calling from sip to sip via my asterisk all on a local machine. How can I track this down? |
| 22:52.46 | bn43 | from 619? |
| 22:52.48 | *** join/#asterisk rdk2 (n=jeff@75-27-14-205.lightspeed.iplsin.sbcglobal.net) |
| 22:52.51 | *** part/#asterisk jsmith-away (n=njsmith@asterisk/training-and-documentation-guru/jsmith) |
| 22:52.54 | bmoraca | bn43: yes |
| 22:53.06 | manxpower | SparFux: it is impossible to have "noise" in a sip-sip call. You must be experiencing some other issue. Are you using GSM?> |
| 22:53.50 | rdk2 | does anyone else find the gsm files containing the prompts for the voicemail system to be pretty bad sound quality? The rest of my sound quality is good, but those prompts sound pretty bad Anyone have any ideas? |
| 22:53.54 | Miccster | Anyone know where I can get cheap RJ21 connectors for hooking up an SPA8000 to a punch down block? |
| 22:54.00 | bn43 | same : [Jan 27 00:53:11] NOTICE[18612]: chan_sip.c:14489 handle_request_invite: Call from '619' to extension '+0027011621' rejected because extension not found. |
| 22:54.04 | manxpower | ~gsmbug |
| 22:54.05 | jbot | [~gsmbug] there is a bug compiling Asterisk with GCC 4.2 where optimization errors cause GSM transcoding to distort heavily. Compile with GCC 4.1 or lower, or follow this patch : http://bugs.digium.com/view.php?id=11243 Also please read : http://forums.digium.com/viewtopic.php?p=116731&sid=3c65e49ec840380ed20f1c8426382b39 |
| 22:54.12 | SparFux | manxpower: No, pcmu. |
| 22:54.19 | bmoraca | bn43: why is your phone appending all of that other crap to it? |
| 22:54.39 | bn43 | i enabled enum on them when setting them up |
| 22:54.49 | SparFux | manxpower: It must be my sound hardware. But on the other hand, I get clear sound with everything except most voip phones. |
| 22:55.04 | bmoraca | bn43: well turn that off. it's not going to work unless it's part of your dialplan as well |
| 22:55.21 | *** join/#asterisk telecos (n=sergio@107.167.219.87.dynamic.jazztel.es) |
| 22:56.47 | rdk2 | thanks, manxpower, I'll take a look |
| 23:02.35 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 23:03.47 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 23:04.07 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 23:05.52 | manxpower | SparFux: are you using any Cisco/Linksys/SIPura devices? |
| 23:06.39 | SparFux | manxpower: no, I am using Ekiga and Twinkle and one calls the other. |
| 23:06.52 | SparFux | And both are connected to asterisk via sip. |
| 23:06.58 | manxpower | SparFux: You're on your own. |
| 23:07.24 | SparFux | I bet it is the stupid sound hardware, I cannot believe it! |
| 23:07.43 | SparFux | I am searching for the failure for ages and it's the stupid hardware! |
| 23:07.55 | manxpower | I can never believe it when people report they don't have issues with softphones. |
| 23:08.43 | bn43 | bmoraca: still doing it - not sure why as I have reset both phones and did not select the enum option |
| 23:09.06 | bmoraca | bn43: what model phones do you have? |
| 23:09.18 | bn43 | snom320 |
| 23:10.30 | SparFux | manxpower: I have only issues with softphones. Almost nothing works. |
| 23:10.43 | manxpower | SparFux: that has been my experience |
| 23:10.45 | bn43 | ahh - 621 phones to 619 |
| 23:10.51 | SparFux | oh wait, I have to say I use the same input and output device on both softphones. |
| 23:10.55 | bn43 | but not the other way around |
| 23:11.42 | SparFux | manxpower: but I would like to have a softphone with pc speaker ring capability. |
| 23:12.18 | manxpower | SparFux: And I want a billion dollars. |
| 23:12.30 | SparFux | :-P |
| 23:14.05 | SparFux | manx: softphones suck because the sound hardware in PCs sucks. |
| 23:15.22 | bmoraca | bn43: if you're still getting the incorrect extension dial in the console, it's likely a config issue on the phone itself. |
| 23:15.36 | *** join/#asterisk path_ (n=path@pc-15-190-86-200.cm.vtr.net) |
| 23:16.38 | SparFux | manx: and skype works perfectly. There's something at stake here! |
| 23:17.48 | *** join/#asterisk tobias (n=tobias@user-0ce2hu8.cable.mindspring.com) |
| 23:19.53 | manxpower | SparFux: I'm having a lot of trouble caring |
| 23:20.12 | SparFux | Why? |
| 23:20.16 | SparFux | I don't get thiis? |
| 23:20.34 | SparFux | You have trouble caring? So you don't care? Do you like Skype? |
| 23:21.02 | *** part/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 23:21.15 | manxpower | SparFux: I use polycom phones and simply have no problems |
| 23:21.43 | bmoraca | Polycoms are great quality phones |
| 23:22.08 | manxpower | But over and over and over again I hear of people with problems with softphones |
| 23:22.36 | beek | God uses Polycom phones |
| 23:24.14 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 23:24.14 | bmoraca | manxpower: they're cheap. |
| 23:24.35 | thedonvaughn | i only had problems with polycom. I'm a fan of aastra |
| 23:25.12 | SparFux | manx: no problem with skype. It just works. |
| 23:25.12 | manxpower | bmoraca: oddly I tend not to be around cheap people |
| 23:25.23 | manxpower | SparFux: THE USE SKYPE |
| 23:25.31 | SparFux | manx : NO WAY! |
| 23:25.44 | manxpower | SparFux: then stop comparing asterisk to it |
| 23:26.34 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 23:26.41 | SparFux | No, I am not compaaring, I am checking wether some sound issue could be my problem.! |
| 23:28.14 | SparFux | Obviously, contrary to what I was thinking, the sound hardware does not really seem to be my problem. That's what I try to say. How could it possibly be the problem causing noise when there is no noise in skype at all? |
| 23:28.46 | bn43 | bmoraca: thank you - I have reset the phone again and it now works! |
| 23:28.48 | SparFux | But skype sucks, what should I do with a softphone, I cannot place calls to most of the people to? |
| 23:29.26 | *** join/#asterisk [acer]lanman (n=Raymond@216.235.233.182) |
| 23:30.25 | rdk2 | manxpower: thanks for the gsmbug tip. Replaced the files with wavs, and it sounds great now. |
| 23:31.32 | rdk2 | one thing I haven't been able to figure out yet is why asterisk isn't using my custom voicemail greetings -- i recorded them, and they're showing up in the spool directory. The dialplan is sending the caller to the right voicemail box, because the voicemails are showing up in the right place. However, asterisk isn't using the custom gretings I recorded for a box, it's just playing the default ones. Ideas? |
| 23:31.40 | bn43 | now to sleep! g'nite all - thank you again |
| 23:32.08 | *** join/#asterisk nix8n82 (n=nate@63.162.27.243) |
| 23:33.12 | beek | rdk2: Crank up the verbosity and watch the console while you test it. |
| 23:33.40 | rdk2 | beek, I did that, it's playing the default sound files instead of my custom ones |
| 23:35.25 | rdk2 | I'm getting vm-intro.sln instead of my custom greeting |
| 23:35.56 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 23:36.07 | rdk2 | i would suspect that it wasn't going to the right vm boix, but the messages show up in there |
| 23:36.32 | beek | rdk2: Is this for the individual mailboxes? |
| 23:36.53 | rdk2 | beek: yes |
| 23:37.19 | beek | Are you using a 'u' or 'b' in your call to VoiceMail()? |
| 23:38.00 | rdk2 | checking |
| 23:38.31 | *** part/#asterisk SparFux (n=raoul@e182017044.adsl.alicedsl.de) |
| 23:39.08 | rdk2 | beek: neither |
| 23:39.29 | rdk2 | it's just VoiceMail(9001@default) |
| 23:39.29 | beek | rdk2: Which custom message did you record? Busy or Unavailable? |
| 23:39.43 | rdk2 | beek: I put something in both just to check |
| 23:40.02 | beek | rdk2: Try VoiceMail(9001@default,b) |
| 23:40.20 | beek | Dont' forget to reload the dialplan |
| 23:40.47 | rdk2 | ok, i am trying it now |
| 23:41.13 | rdk2 | bingo, that took to my custom busy message |
| 23:41.27 | rdk2 | i wonder why it didn't take me to unavail with nothing there |
| 23:41.28 | *** join/#asterisk dlewis (i=4579ba4a@about/security/staff/dlewis) |
| 23:41.47 | beek | I've always specified which I wanted. |
| 23:41.59 | rdk2 | it works, so I won't argue :) |
| 23:44.15 | rdk2 | thanks for the help, beek |
| 23:44.22 | beek | rdk2: np |
| 23:45.53 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:46.06 | beek | evening jaytee |
| 23:46.13 | jaytee | evening beek |
| 23:50.43 | *** join/#asterisk stevetotaro (n=Steve@pool-71-254-231-87.hrbgpa.east.verizon.net) |
| 23:52.28 | *** join/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 23:52.33 | stencil | ~book |
| 23:52.34 | jbot | i heard book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
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