| 00:15.18 | *** join/#asterisk gambolputty (n=BC43599@cpe-76-186-225-57.tx.res.rr.com) |
| 00:15.27 | *** part/#asterisk gambolputty (n=BC43599@cpe-76-186-225-57.tx.res.rr.com) |
| 00:15.36 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:31.41 | *** join/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:38.46 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:38.46 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 00:43.41 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 00:44.07 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 00:50.56 | *** join/#asterisk angryuser (n=angryuse@90-156-167-83.reverse.alphalink.fr) |
| 00:59.52 | KavanS | how can I tell if I have support for cdr_mysql? |
| 01:12.50 | lmadsen | it builds? |
| 01:32.10 | *** join/#asterisk suma (n=chinnapa@cpe-76-168-177-23.socal.res.rr.com) |
| 01:36.26 | mmlj4 | it loads? |
| 01:40.47 | apeiron | Using it does what you expect? (like silently truncating oversized CHAR columns) |
| 01:45.08 | lmadsen | it makes you popcorn? |
| 02:13.49 | *** join/#asterisk jcoffi1 (n=jcoffi@75.147.155.89) |
| 02:16.41 | *** join/#asterisk elitecoder (n=liq@apollo.bullethost.com) |
| 02:19.00 | elitecoder | I made an extensions.conf that is virtually empty, did a dialplan reload, then dialplan show. I see a lot of stuff I didn't put in extensions.conf - where is it coming from? |
| 02:24.25 | lmadsen | elitecoder: there will be some stuff for parking and such that gets loaded |
| 02:26.23 | elitecoder | /join/win 1 |
| 02:26.40 | KavanS | anyone have a suggestion for a good cdr stats viewer? |
| 02:26.53 | KavanS | I'm playing with asterisk-web-stats, but getting buggy results |
| 02:27.07 | elitecoder | -= 26 extensions (63 priorities) in 24 contexts. =- |
| 02:28.57 | lmadsen | elitecoder: did you reload your dialplan? |
| 02:29.03 | elitecoder | Does that look reasonable for an almost empty extensions.conf? |
| 02:29.05 | lmadsen | they have to be coming from the extensions.conf file |
| 02:29.09 | lmadsen | no |
| 02:30.10 | elitecoder | I swear I have like 6 context. all empty. |
| 02:30.14 | elitecoder | contexts* |
| 02:36.18 | *** join/#asterisk jcoffi (n=jcoffi@75.147.155.89) |
| 02:48.39 | *** part/#asterisk elitecoder (n=liq@apollo.bullethost.com) |
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| 02:51.30 | *** mode/#asterisk [+o Mog] by ChanServ |
| 02:51.47 | carrar | swear |
| 03:01.04 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 03:02.24 | KavanS | christ |
| 03:07.03 | *** join/#asterisk jcoffi1 (n=jcoffi@75.147.155.89) |
| 03:19.07 | *** join/#asterisk path_ (n=path_@pc-15-190-86-200.cm.vtr.net) |
| 03:39.02 | *** join/#asterisk CunningPike (n=arodgers@S01060014bf81366b.vc.shawcable.net) |
| 03:45.33 | raden_work | i bought aastra 9133i phones and it said they are compatiable with g.711 and g.729 and in the user manual it says g.729 annex b will they still work with asteris k ? |
| 03:56.01 | mmlj4 | have you tried it? |
| 04:02.14 | raden_work | there not here yet unfortunatley |
| 04:23.06 | areay | when i dial my external number from a regular phone, i can hear my menus and such, but no sound can be transmitted from an outside line into asterisk... i'm behind NAT and i'm pretty sure that's set up right |
| 04:23.49 | *** join/#asterisk jcoffi (n=jcoffi@75.147.155.89) |
| 04:24.39 | areay | what would cause sound to go one way, but not the other? |
| 04:27.00 | *** join/#asterisk killown (n=geek@unaffiliated/killown) |
| 04:29.14 | areay | my ports are forwarded correctly, here's my sip.conf: http://paste.ubuntu.com/131401/ (i'm getting one-way audio) |
| 04:33.36 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 04:33.48 | Kobaz | areay: your ports arent forwarded correctly |
| 04:34.29 | [TK]D-Fender | areay: [voiptalk] should be "nat=no" |
| 04:34.35 | [TK]D-Fender | areay: and confirm your ports please |
| 04:34.54 | [TK]D-Fender | raden_work: Yes they work fine |
| 04:36.08 | areay | Kobaz, [TK]D-Fender: i'll change nat to "no"; my ports are 5060 and 10001-20000 and they're forwarded correctly |
| 04:36.28 | [TK]D-Fender | areay: show us |
| 04:36.43 | RypPn | voiptalk are g729-only |
| 04:36.50 | Kobaz | areay: obviously something is wrong, if you only get one side of the audio |
| 04:37.09 | areay | RypPn, what does that mean? |
| 04:37.11 | Kobaz | areay: 99% of the time... it's the nat/ports on your router |
| 04:37.19 | [TK]D-Fender | Kobaz: "nat=yes" is a very likely problem |
| 04:37.21 | areay | Kobaz, I've triple-checked em |
| 04:37.29 | areay | [TK]D-Fender, retrying now |
| 04:37.29 | [TK]D-Fender | areay: SHOW US |
| 04:37.42 | areay | [TK]D-Fender, u wanna see the sip debug? |
| 04:38.27 | [TK]D-Fender | areay: Sure, and I want to see your forwarding |
| 04:38.29 | Kobaz | a snippit of rtp debug, firewall rules... and packet header dump |
| 04:38.42 | raden_work | [TK]D-Fender, is there anything special i will have todo digium says they dont support g.729b |
| 04:38.58 | [TK]D-Fender | raden_work: It should work fine, just wait till it arrives. |
| 04:39.12 | raden_work | lol |
| 04:41.31 | areay | [TK]D-Fender, sip debug: http://paste.ubuntu.com/131402/ ... the way my router deals with forwarding is through something it calls "game and application sharing"... i have an application called "Asterisk" set up, here's the forwarding rules: http://paste.ubuntu.com/131403/ |
| 04:42.11 | [TK]D-Fender | areay: should be 10000-20000 normally as well |
| 04:42.59 | areay | [TK]D-Fender, ah ok i'll change that |
| 04:43.42 | *** join/#asterisk crakrjak (n=merc@rrcs-70-62-156-154.central.biz.rr.com) |
| 04:45.16 | crakrjak | hi there, i was wondering if i could get a recommendation on where to get a free SIP number for incoming calls |
| 04:45.19 | areay | [TK]D-Fender, is everything set up as it should be? do i need to use a proxy? |
| 04:45.46 | areay | crakrjak, ekiga.net offer free sip numbers |
| 04:46.12 | crakrjak | cool, i found a bunch of free ones just wondering what people are happy with, ill check that one out, thanks |
| 04:47.49 | *** join/#asterisk snowboarder04 (n=un@serv.bemail.co.uk) |
| 04:47.50 | crakrjak | so far i found sipnumber,voxalot and ipkall (i think) heh |
| 04:48.00 | *** join/#asterisk uluatu (n=denis@201.47.21.191.dynamic.adsl.gvt.net.br) |
| 04:48.37 | uluatu | Who are taking the CDR issues at the bugtracker? |
| 04:48.44 | [TK]D-Fender | areay: So far I'm not seeing it.... |
| 04:49.36 | areay | [TK]D-Fender, "it" being the problem, yeah? or do you need more info |
| 04:49.46 | [TK]D-Fender | areay: the problem... |
| 04:51.28 | *** part/#asterisk jcoffi (n=jcoffi@75.147.155.89) |
| 04:51.47 | areay | [TK]D-Fender, cool... i've read the "book" pretty much in its entirety, and i've configured this installation from scratch, using the config files... it's given me a much better understanding of how everything works |
| 04:53.12 | areay | [TK]D-Fender, i can get another static ip from my isp... if i dmz that to asterisk, will it solve my problems? |
| 04:53.45 | [TK]D-Fender | areay: for now, why not try it |
| 04:54.14 | [TK]D-Fender | areay: and double check your actual WAN IP. and then triple check it. The hard way |
| 04:55.08 | areay | [TK]D-Fender, yeah i wanna check everything else... i dont wanna pay extra if i dont have to, and plus they take 48 hours to get it up and running... and it might not even fix my problem |
| 04:55.51 | [TK]D-Fender | areay: Shouldn't need a fixed IP at all. |
| 04:56.03 | [TK]D-Fender | areay: Does make things easier however |
| 05:00.53 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 05:07.20 | *** join/#asterisk RichardLynch (n=RichardL@c-98-193-37-55.hsd1.il.comcast.net) |
| 05:09.44 | RichardLynch | Where would I look for somebody who has done some EAGI work, preferably with PHP? I tried a few emails from here: http://www.voip-info.org/tiki-index.php?page=Asterisk+consultants+USA and got very little response, none of it useful yet. |
| 05:11.32 | [TK]D-Fender | RichardLynch: What are you trying to do with it? |
| 05:11.49 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-209-183.lns10.mel6.internode.on.net) |
| 05:12.07 | RichardLynch | Take the audio in, bounce it through sox to add an effect, and use MOH to stream it back real-time. |
| 05:12.27 | areay | [TK]D-Fender, something wrong with my router i think... i checked on http://seemyport.info/ to see if port 5060 was open and it says no.... |
| 05:12.59 | [TK]D-Fender | areay: that lookup is TCP <----- |
| 05:13.25 | RichardLynch | Having the devil of a time just getting pipe 3 open and pulling in data reliably (stdin 0, stdout 1, stderr 2, ??? 3). |
| 05:13.46 | RichardLynch | Probably just doing something stupid, but the sample code is in Perl, and doesn't look that tricky. |
| 05:18.36 | RichardLynch | Hmmm. As soon as I explain what I'm trying to do, nobody wants to help... [shrug] |
| 05:18.38 | areay | [TK]D-Fender, i forwarded the tcp ports too to make sure everything was working as it should... its saying they're open now |
| 05:24.11 | *** join/#asterisk jcoffi (n=jcoffi@75.147.155.89) |
| 05:33.01 | *** join/#asterisk trymi1 (n=please@203.81.71.107) |
| 05:46.45 | *** join/#asterisk SunnyDP (n=scan@bas7-montrealak-1128744958.dsl.bell.ca) |
| 05:49.26 | trymi1 | Hi all |
| 05:49.39 | trymi1 | can you help me ? in Asterisk |
| 05:49.52 | trymi1 | I am beginner for asterisk |
| 05:50.03 | trymi1 | everybody there? |
| 05:50.04 | SunnyDP | trymi1: hello friend, i am wondering if anyone has heard about sip foundry |
| 05:50.22 | trymi1 | hi Sunny |
| 05:50.32 | trymi1 | can you help me in asterisk? |
| 05:51.24 | trymi1 | my asterisk can`t call to PSTN phone |
| 05:51.31 | trymi1 | can you help me? |
| 05:54.12 | SunnyDP | trymi1: ican help you with voip but not asterisk |
| 05:54.39 | trymi1 | just sip foundry? |
| 05:55.32 | SunnyDP | trymi1: no, i meant VOiP in general, protocols, etc... i am myself discovering sip foundry next week |
| 05:56.09 | trymi1 | sunny I am interesting you |
| 05:56.10 | trymi1 | :) |
| 05:56.19 | trymi1 | yes pls help me for VoIP |
| 05:58.22 | SunnyDP | trymi1: :D |
| 05:59.21 | *** part/#asterisk RichardLynch (n=RichardL@c-98-193-37-55.hsd1.il.comcast.net) |
| 05:59.21 | trymi1 | now I build 1 astersik server |
| 05:59.50 | trymi1 | can call to other extension but can`t outgoing to PSTN |
| 06:01.25 | trymi1 | hi sunny |
| 06:01.48 | SunnyDP | trymi1: yes |
| 06:02.00 | SunnyDP | you have softphones ? |
| 06:02.17 | trymi1 | I have xlite |
| 06:05.24 | SunnyDP | xlite ? |
| 06:05.37 | trymi1 | yes |
| 06:06.30 | trymi1 | I can call from my ext 100 to 101 |
| 06:06.56 | trymi1 | I have sip truk account |
| 06:07.00 | trymi1 | trunk |
| 06:07.23 | trymi1 | I can call to anywhere with this account with xlite |
| 06:07.43 | trymi1 | I setup trunk in my asterisk |
| 06:07.50 | trymi1 | but not work |
| 06:07.51 | trymi1 | :) |
| 06:08.02 | trymi1 | I don`t know what wrong |
| 06:09.33 | SunnyDP | ahhh ok |
| 06:09.44 | k-man | if you have a custom moh handler, does asterisk suspend the process while its not required? |
| 06:09.47 | SunnyDP | just wait here for support it will come |
| 06:10.03 | trymi1 | yes sunny |
| 06:10.04 | trymi1 | :) |
| 06:13.21 | SunnyDP | trymi1: where are you from brother ? |
| 06:13.38 | trymi1 | from myanmar |
| 06:14.13 | trymi1 | why? :) |
| 06:15.52 | trymi1 | and you? |
| 06:17.26 | SunnyDP | myanmar ??? i am in canada |
| 06:18.13 | SunnyDP | trymi1: ahhh burma :D beautiful jewelry |
| 06:18.20 | trymi1 | :) |
| 06:18.24 | trymi1 | yes burma |
| 06:20.22 | trymi1 | will you be do for SIP foundry? |
| 06:21.52 | SunnyDP | yes |
| 06:22.30 | SunnyDP | next week i will start do you know about sip foundry? |
| 06:23.27 | trymi1 | I hear but I did not test |
| 06:24.57 | trymi1 | can you explain me different of softswitch and PBX ? |
| 06:25.39 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
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| 06:39.59 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-23-21.w86-215.abo.wanadoo.fr) |
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| 07:32.06 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 07:37.28 | KavanS | does anyone use h264 video on ekiga w/asterisk 1.4? |
| 07:41.11 | *** join/#asterisk j_kroon (n=jkroon@dsl-240-188-106.telkomadsl.co.za) |
| 07:41.50 | j_kroon | hi all, why would setting jbenable=yes, jbimpl=adaptive in sip.conf cause one-way voice when transferring/putting on hold dahdi channels? |
| 07:42.12 | j_kroon | I receive a warning like this in the CLI: abstract_jb.c:472 create_jb: Failed to put first frame in the jitterbuffer on channel 'DAHDI/1-1' |
| 07:42.46 | j_kroon | at the same time the voice going out from asterisk to the dahdi channel dies and it sounds to the caller like you've got a dead channel. |
| 07:46.00 | *** join/#asterisk WeazelON (n=deazel@mail2.tikalnetworks.com) |
| 07:47.58 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 07:57.07 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
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| 08:02.00 | *** part/#asterisk trymi1 (n=please@203.81.71.107) |
| 08:12.31 | *** join/#asterisk sergey (n=sergey@sergey-home.iks.ru) |
| 08:21.17 | *** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net) |
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| 08:48.02 | *** join/#asterisk Qapf (n=qapf@173.8.140.60) |
| 09:07.05 | *** join/#asterisk trajce (n=blazko@77.28.221.114) |
| 09:07.27 | trajce | hi all |
| 09:07.46 | trajce | I am trying to troubleshoot my first asterisk box |
| 09:08.27 | trajce | The last thing that I can't get it to work is call pickup *8 |
| 09:09.05 | trajce | local pickup works just fine |
| 09:11.01 | trajce | but when i call the outside line, and try to pickup the call from an extension, the operators phone stops ringing and from the phone that I try to pickup I can only hear noise |
| 09:11.17 | trajce | also the outside phone keeps ringing |
| 09:11.38 | trajce | can someone help me troubeshoot this issue? |
| 09:26.57 | j_kroon | try dialing inbound :) |
| 09:31.40 | trajce | <PROTECTED> |
| 09:32.45 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 09:35.54 | tzafrir_laptop | trajce, try looking at a traec |
| 09:36.03 | tzafrir_laptop | with a verbocity level of 3 |
| 09:36.14 | tzafrir_laptop | to see the flow of the dialplan |
| 09:37.45 | *** join/#asterisk ZeNN (n=abc@ip224-160-173-82.adsl2.static.versatel.nl) |
| 09:43.25 | j_kroon | trajce, oh, misread your call the outside line as "dialing out". |
| 09:49.11 | *** join/#asterisk TrentCreek (i=kvirc@adsl-66-136-99-93.dsl.hrlntx.swbell.net) |
| 09:53.57 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
| 10:11.44 | *** join/#asterisk hi365 (n=hi365@85.130.230.240) |
| 10:12.12 | freddyk | tzafrir: hello, can you help me about a sip issue on trunk ? |
| 10:13.32 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 10:22.49 | ZeNN | asterisk 1.6.0.6 / gui 2.0.4 / misdn 1.1.9 | i've got misdn working with asterisk in the sense that asterisk recognizes a fritzcard on the pci bus and lists it in the asterisk cli as available. But the GUI (under misdn config) it gives me the error: No mISDN Card detected (found 0 spans) |
| 10:23.32 | ZeNN | is this a bug in the gui ? and is there a way to debug de gui ? |
| 10:25.09 | tzafrir_laptop | freddyk, well, if I can. I'm not sure I'm the best SIP expert around |
| 10:25.21 | *** join/#asterisk hi365 (n=hi365@85.130.230.240) |
| 10:28.44 | freddyk | tzafrir: i have some problems after upgrading from 1.6.0.6 to 1.6.1 trunk |
| 10:29.56 | freddyk | tzafrir: i got some errors about registrations and non resolving domains |
| 10:30.41 | freddyk | tzafrir: same configuration worked on 1.6.0.6 |
| 10:33.08 | *** join/#asterisk Failrar (n=Failrar@fsm.xs4all.nl) |
| 10:34.55 | tzafrir_laptop | Could you be more specific? what error? Can you pastebin relevant parts of the configuration? |
| 10:39.46 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 10:45.04 | *** join/#asterisk zaafouri (n=cz@196.203.51.238) |
| 11:03.09 | *** join/#asterisk lbt (n=david@78.32.229.233) |
| 11:26.29 | *** join/#asterisk smooth_penguin (n=smoove_@59.95.12.112) |
| 11:26.33 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 11:26.53 | smooth_penguin | hey can I run ekiga on the same machine where asterisk is setup? |
| 11:27.09 | smooth_penguin | is an absolute newb to asterisk |
| 11:27.53 | smooth_penguin | asterisk, for sip listens on 5060 |
| 11:28.10 | smooth_penguin | can ekiga run on anothe port? |
| 11:28.23 | *** join/#asterisk hohum (n=dcorbe@206.71.169.115) |
| 11:30.43 | tzafrir_laptop | smooth_penguin, yes |
| 11:30.57 | tzafrir_laptop | IIRC it will default to doing so if port 5060 is not free |
| 11:58.03 | *** join/#asterisk fsero (n=mort@84.122.53.51.dyn.user.ono.com) |
| 12:02.06 | k-man | does asterisk cache moh internaly? |
| 12:02.27 | k-man | so that if you hangup and dial moh again, it can start really quickly? |
| 12:03.09 | k-man | hmm... no, i think its mplayer caching |
| 12:03.50 | smooth_penguin | tzafrir_laptop, yeah I changed the port ekiga runs on |
| 12:03.53 | smooth_penguin | now its fine |
| 12:04.06 | smooth_penguin | got my first tt-monkeys :> |
| 12:04.14 | *** join/#asterisk edjo_ (n=abla@p578b7856.dip0.t-ipconnect.de) |
| 12:06.14 | *** join/#asterisk nima0102 (n=nima@91.98.221.195) |
| 12:10.15 | k-man | is there any way to get asterisk to kill the custom music player when not required in moh, instead of suspending and resuming it? |
| 12:14.06 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 12:21.57 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
| 12:22.33 | EmleyMoor | Is there any way to get an old-fashioned "purr" dialtone on a Zap phone? |
| 12:23.28 | EmleyMoor | (used in the Uk universally 1923-1976 and in some areas into the 1990s) |
| 12:24.48 | k-man | EmleyMoor: that sounds nice |
| 12:25.13 | k-man | EmleyMoor: you know, remember the first uk electronic phones? (or maybe early, not first) that had this cool ringer? |
| 12:25.26 | k-man | i wish i could find that ringer again - id love to use it |
| 12:25.27 | EmleyMoor | Trimphones? |
| 12:25.33 | k-man | yeah! trimphones |
| 12:25.41 | k-man | your memory is better than mine |
| 12:25.45 | k-man | i forgot they were called that |
| 12:25.48 | EmleyMoor | "preep preep" |
| 12:26.05 | k-man | yeah something like that |
| 12:27.13 | k-man | i think that ringer would have real street cred if you have it in your mobile phone |
| 12:27.39 | EmleyMoor | k-man: I had it on one of mine once |
| 12:28.07 | EmleyMoor | (in fact I think it's on my N70) |
| 12:28.25 | coppice | EmleyMoor: are you talking about the ringback where you simply actually heard the purr of the ring going to the called phone? |
| 12:28.46 | k-man | EmleyMoor: can you email it to me by any chance? |
| 12:28.55 | EmleyMoor | coppice: No - the dialtone... |
| 12:29.25 | EmleyMoor | k-man: If I can find it, of course... |
| 12:29.56 | EmleyMoor | coppice: Was replaced by an eletronic "hmmm" in stages from 1976 |
| 12:30.27 | k-man | is this it? http://www.polyphonicringtonez.com/free-mp3-ringtone/36516-70s_telephone-trimphone |
| 12:30.37 | *** join/#asterisk propellerhead (n=yogurt2u@host215.190-138-92.telecom.net.ar) |
| 12:30.49 | EmleyMoor | Don't think mine is an mp3 |
| 12:30.59 | coppice | oh, sorry. in the old strowger exchanges that was something similar. just a 33.3Hz purr from some electromechnical gadget if I remember rightly. the change came with the first electronic exchanges (TXE4) |
| 12:31.04 | *** join/#asterisk af_ (n=getsmart@88-149-240-204.dynamic.ngi.it) |
| 12:31.58 | EmleyMoor | coppice: Yes - exact frequency varied from one village to the next |
| 12:32.20 | EmleyMoor | k-man: I will find where I got it from on the web |
| 12:32.59 | coppice | those electromechanical supervisory generator were highly variable to say the least. It made tone detection on the PSTN kinda fun :-) |
| 12:34.26 | coppice | you'd probably need a recording of the purr to really try to mimic its character. it probably had an interesting harmonic mixture :-) |
| 12:34.46 | EmleyMoor | k-man: http://www.telephonesuk.co.uk/sounds/trimphone.wav |
| 12:36.39 | k-man | EmleyMoor: thanks |
| 12:36.39 | EmleyMoor | I think I mp3'd it |
| 12:37.06 | k-man | bah! stupid windows - its refusing to play sounds at the moment - ill have to reboot to hear it |
| 12:37.26 | EmleyMoor | I have trimphone.mp3 on my desktop |
| 12:39.37 | k-man | made from that wav? |
| 12:39.48 | EmleyMoor | Yes |
| 12:41.55 | k-man | EmleyMoor: ill try them out tomorrow |
| 12:42.07 | k-man | its too late to be fiddling with that here- it might wake my wife up |
| 12:42.45 | k-man | i left the uk some time ago, so havent heard taht sound for a long time |
| 12:43.13 | EmleyMoor | I visited someone a couple of years ago who still has a trimphone |
| 12:43.22 | k-man | wow, thats cool |
| 12:43.31 | k-man | did them bring them out with DTMF? |
| 12:43.52 | EmleyMoor | I've even got a couple of earth recall switches for them somewhere |
| 12:44.02 | EmleyMoor | There were DTMF ones... |
| 12:44.04 | k-man | oh? whats thta? |
| 12:44.48 | EmleyMoor | A litle push switch added lower front for use on cortain switchboards |
| 12:44.52 | EmleyMoor | certain |
| 12:45.02 | k-man | what did it do? |
| 12:45.53 | EmleyMoor | Earthed one side of the line briefly causing dial tone to return - for enquiry calls/transfer mainly |
| 12:46.09 | EmleyMoor | Hookflash/TBR came later |
| 12:46.42 | k-man | oh, i see |
| 12:46.51 | k-man | sort of an out of band signal |
| 12:47.10 | EmleyMoor | Trimphone 786 was a DTMF version |
| 12:48.46 | k-man | your a bit of an export |
| 12:48.51 | k-man | oops, expert i mean |
| 12:50.01 | EmleyMoor | I just happen to like the old phones... got a 746 on my bedside table and an 8785 that just needs two new washers and a battery for the amplifier (and the bracket putting up) |
| 12:50.23 | EmleyMoor | (the 746 is on my Asterisk) |
| 12:50.34 | k-man | EmleyMoor: thats very cool |
| 12:50.54 | k-man | the meld of old tech to new tech - i like it |
| 12:51.11 | k-man | is that why youre trying to get the old dial tone going? |
| 12:51.36 | EmleyMoor | Partly - if it can be done I can see it having drama use |
| 12:58.55 | EmleyMoor | I've got a CD full of N diagrams and stuff |
| 12:59.21 | k-man | n diagrams? |
| 13:01.14 | EmleyMoor | Circuit diagrams for the old phones, mainly... but other interesting stuff too |
| 13:02.48 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 13:03.14 | k-man | oh right |
| 13:03.16 | k-man | cool |
| 13:03.50 | k-man | well - i just finished my little script and setup to play the radio, streamed from the internet for moh |
| 13:04.03 | k-man | so now i can use my cordless phone as a sort of radio |
| 13:04.19 | EmleyMoor | I've been considering that as an option... |
| 13:04.24 | k-man | i just dial an extension to listen |
| 13:04.26 | k-man | its cool |
| 13:04.28 | k-man | works well |
| 13:04.43 | EmleyMoor | Not worked out how to use a stream as moh yet |
| 13:05.00 | k-man | and i made the extensions the same as the frequency |
| 13:05.11 | k-man | so, to listend to AM 702, i just dial 702 |
| 13:05.31 | bougyman | i have a 48k moh stream backed by mpd on one extension that I listen to from time to time. |
| 13:05.38 | k-man | EmleyMoor: its easy once you know how - i can send you my setup if you like |
| 13:05.51 | EmleyMoor | It would be interesting to see |
| 13:05.51 | bougyman | just for kicks, the quality ended up being as good as streaming through a media player, though, heh. |
| 13:06.08 | k-man | hehe |
| 13:06.18 | bougyman | and I'm off-hook (bonus, no calls) |
| 13:07.30 | coppice | I've seen a DTMF phone with a dial. people build the silliest things |
| 13:07.59 | bougyman | i was wondering if a rotary phone would work on an astribank. |
| 13:08.07 | bougyman | been meaning to find one at a garage sale and test it out. |
| 13:08.16 | bougyman | it'd be a fun prank to replace the CEOs phone with a rotary. |
| 13:08.34 | tzafrir_laptop | bougyman, works on an astribank just as it should work on any zaptel/dahdi fxs adapter |
| 13:08.56 | bougyman | tzafrir_laptop: good to know. |
| 13:08.59 | EmleyMoor | I saw a film once where one of the characters was baffled by a rotary phone |
| 13:09.18 | bougyman | tzafrir_laptop: the astribank gave me hell with the zaptel drivers. |
| 13:09.27 | tzafrir_laptop | though you can test that by "dialing" with the hook key :-) |
| 13:09.34 | bougyman | once I got dahdi installed it was happy, though. |
| 13:10.06 | bougyman | tech support wasn't much of a help, unfortunately. |
| 13:10.28 | *** join/#asterisk littlerock (n=littlero@58.31.43.131) |
| 13:10.35 | bougyman | they just kept telling me to install CentOS and * |
| 13:10.46 | k-man | god, can you remember how painfuly slow it was to dial out on a rotary phone? |
| 13:11.06 | k-man | and when exchanges first started properly recognising dtmf and how fast they connected? |
| 13:11.10 | k-man | it was like, instant |
| 13:11.19 | k-man | as soon as you pressed the last digit, it would start ringing |
| 13:11.21 | coppice | not so long ago I watched a 1930s comedy film where someone pulled out a rotary dial phone in a car, and started using it. The adults laughed, but the kids were so used to cellphones in everyone's pocket that couldn't see the joke |
| 13:13.17 | EmleyMoor | I do sometimes dial on my 746 |
| 13:14.30 | k-man | the problem with streaming the radio though, is that when moh is not required, asterisk suspends the process. so mplayer stops receiving date (which is fine) |
| 13:15.14 | k-man | but when asterisk resumes the process, mplayer keeps playing its buffer, which lasts maybe 15 seconds, then it stops playing for a second while it starts the stream again |
| 13:15.18 | k-man | its a bit annoying |
| 13:16.15 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 13:16.41 | k-man | anyway |
| 13:16.45 | k-man | bedtime |
| 13:16.49 | k-man | night all |
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| 13:48.42 | ZeNN | does /usr/sbin/misdn-init be chowned under a regular user instead of root in order for the asterisk gui (2.0.4) to use that command ? |
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| 15:38.24 | littlerock | I installed asterisk 1.6 and dahdi, however meetme doesn't work, the error message is "can not open pseudo device". |
| 15:39.06 | littlerock | I can not insmod dahdi.ko and dahdi_dummy.ko, any idea? thx |
| 15:41.54 | *** join/#asterisk cm_5_1_2 (n=cm_5_1_2@58.252.229.215) |
| 15:42.39 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
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| 15:56.47 | *** join/#asterisk DragoraN (n=DragoraN@chello089173068056.chello.sk) |
| 15:56.51 | DragoraN | hi all |
| 15:57.08 | DragoraN | is it possible via AGI to initiate call to someone and tell him some message/audiofile ? |
| 15:58.36 | henk | it is possible. i don't know what agi has to do with it... |
| 15:58.56 | *** join/#asterisk _gm (n=gmustafa@115.186.106.37) |
| 15:59.00 | [TK]D-Fender | DragoraN: AGI says that you're already IN a call. How/why is that so? |
| 15:59.26 | [TK]D-Fender | DragoraN: Sounds like you should be issuing a Call-File or AMI Originate |
| 16:00.06 | henk | are there any reliable iax phones at all? which are still being developed? and opensource? |
| 16:00.45 | DragoraN | [TK]D-Fender: so, what |
| 16:03.06 | littlerock | anyone can use meetme() application ? |
| 16:03.08 | *** join/#asterisk smooth_penguin (n=smoove_@59.95.13.183) |
| 16:03.38 | [TK]D-Fender | littlerock: Anyone with a timing source installed. |
| 16:04.05 | littlerock | can I use dahdi_dummy.ko instead? |
| 16:04.21 | [TK]D-Fender | littlerock: That is a timing source... |
| 16:05.37 | littlerock | :) |
| 16:06.00 | littlerock | I can not insmod dahdi.ko |
| 16:06.03 | littlerock | any idea? |
| 16:06.43 | [TK]D-Fender | littlerock: modprobe dahdi && modprobe dahdi_dummy |
| 16:08.32 | littlerock | can not find dahdi |
| 16:11.03 | *** join/#asterisk raden_work (n=jon@adsl-99-139-235-165.dsl.applwi.sbcglobal.net) |
| 16:11.51 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
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| 16:14.20 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 16:14.48 | [TK]D-Fender | littlerock: reinstall it. Pastebin the entire process |
| 16:14.58 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 16:19.41 | *** join/#asterisk propellerhead (n=yogurt2u@host215.190-138-92.telecom.net.ar) |
| 16:29.12 | *** join/#asterisk AndyCrawford (n=andy@dynamic-65-161-142-80.tvscable.com) |
| 16:30.21 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 16:34.00 | axisys | how can I tell which port is fsx and which port is fxo on my pci card from outside? |
| 16:34.16 | axisys | it is already mounted in my system .. but there is no label outside |
| 16:34.44 | jaytee | is it a tdm410? |
| 16:35.09 | axisys | when i plug my 10 dollar phone and i press the buttons on the phones.. i here the tone when connecting to some port and not for the other two |
| 16:35.29 | axisys | it is from openvox |
| 16:35.42 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 16:35.47 | jaytee | dunno, check the docs for the card |
| 16:36.15 | axisys | i am guessing the one where I hear the phone buttons to beam while pressing are the FXS ports |
| 16:36.25 | axisys | s/beam/beep/ |
| 16:37.47 | axisys | jaytee: here is the doc http://www.openvox.cn/download/other_docs/A400_FAQ(English).pdf |
| 16:37.50 | apeiron | eyes jbot |
| 16:38.07 | apeiron | That is quite possibly the most annoying feature I have ever seen in a bot. |
| 16:38.30 | apeiron | s/possibly/die/e |
| 16:38.42 | apeiron | ... heh |
| 16:39.50 | jaytee | axisys, bad link |
| 16:39.57 | Chainsaw | apeiron: It'll only trigger on a normally terminated sed. It won't do e, g, etc. |
| 16:40.18 | apeiron | Chainsaw, Does it do anything actually useful? |
| 16:40.25 | Chainsaw | Of course. |
| 16:40.27 | apeiron | wants to /ignore it but doesn't want to miss useful stuff |
| 16:41.38 | jaytee | axisys, it appears that openvox has "cloned" the color scheme of Digium cards. The red modules are FXO and the green ones are fxs. If you open the case and look at the card you should be able to tell which is which. |
| 16:41.56 | axisys | jaytee: but how do I know how they map to? |
| 16:42.16 | axisys | which two ports are mapped to red and which two mapped to green ? |
| 16:43.16 | axisys | why would the phone buttons make no beep when plug into the top two ports? |
| 16:43.24 | jaytee | the end plate where the RJ-11 jacks connect should have the channel numbers stamped on them 1 through 4 and the card module connectors are labeled as well so a VISUAL inspection of the card should tell you that. This isn't rocket science, or as [TK]D-Fender would say "RAW-CAT" science. |
| 16:44.03 | jaytee | most likely you've got it right and the top two are fxo which is a "dry" line meaning no power to the phone. |
| 16:44.04 | axisys | stamped on them.. hmmm ... let me check again |
| 16:46.56 | axisys | nothing stamped outside..let me open the box |
| 16:47.40 | axisys | i was hoping some linux tool or asterisk command would show somethign when plug my phone into the ports |
| 16:47.49 | *** join/#asterisk path_ (n=path_@pc-15-190-86-200.cm.vtr.net) |
| 16:48.07 | jaytee | did you run dahdi_tool? |
| 16:48.53 | *** join/#asterisk astassistant (n=skip@65.126.63.1) |
| 16:49.08 | *** part/#asterisk astassistant (n=skip@65.126.63.1) |
| 16:52.26 | axisys | it says j1, j2, j3 and j4.. i am guessing that is channel number.. i am using 1.4.x w/ zaptel |
| 16:54.25 | axisys | ztcfg -v says this http://pastebin.com/f75669bca |
| 16:55.15 | axisys | my guess was right.. phone button beeps when pressed if connected to fxs ports |
| 16:55.55 | jaytee | axisys, sorry I thought you were running dahdi not zaptel. that would be zttool instead of dahdi_tool but it may not have been compiled unless you installed the necessary dependencies and set the option by running make menuselect when you compiled zaptel. |
| 16:56.18 | axisys | jaytee: zttool works |
| 16:56.53 | axisys | it gives me a characted based window and says OK for the driver |
| 16:57.12 | jaytee | cool, then all you need to do is make sure /etc/zaptel.conf is setup correctly and /etc/asterisk/zapata.conf is setup to provide the correct signalling for each channel. |
| 16:57.25 | axisys | when I select it .. i get another window |
| 16:57.27 | jaytee | and a context. use the examples in the OpenVox docs and the book |
| 16:58.12 | jaytee | and remember that the signalling is BACKWARDS from what you think it should be. FXO ports use fxs signalling and vice versa for the FXS ports |
| 16:58.19 | axisys | jaytee: would have been nice if I connect the phone and get a message in the system saying something connected to fxs port 1 .. something like that |
| 16:58.30 | axisys | jaytee: yep |
| 16:58.33 | *** join/#asterisk Badrobot- (n=Badrobot@cpe-76-173-233-75.socal.res.rr.com) |
| 16:58.46 | jaytee | axisys, people in hell want ice water...go figure. |
| 17:00.00 | henk | come on guys... someone please tell me there is a good, free, iax-softphone for linux! |
| 17:00.18 | Chainsaw | I only know of a decent SIP one. It's called Ekiga. |
| 17:00.19 | jaytee | axisys, and remember to restart the zaptel service after you've finished setting up the zaptel.conf and zapata.conf files. |
| 17:00.30 | jaytee | henk, try Zoiper |
| 17:00.48 | tzafrir_laptop | axisys, set verbosity level to 3. you won't get a message when connecting the phone |
| 17:01.01 | tzafrir_laptop | But you'll get a channel generated when picking up the phone |
| 17:01.10 | tzafrir_laptop | Not to mention a dialtone |
| 17:01.58 | tzafrir_laptop | rather: run ztcfg after editing zaptel.conf . run 'zap restart' (or just restart asterisk) after editing zapata.conf |
| 17:02.00 | henk | Chainsaw: thanks, ekiga is alright. it seems to be getting critical if two ekigas on the same nat'ed network try to connect to one server. had several problems with that setup... |
| 17:02.37 | tzafrir_laptop | henk, canreinvite=no; :-( |
| 17:02.39 | Chainsaw | henk: NAT is always ugly. |
| 17:02.51 | Chainsaw | henk: I don't see how SIP -> IAX2 will magically make that go away. |
| 17:02.55 | axisys | i dont see any zapata.conf file.. only zaptel.conf |
| 17:03.11 | henk | jaytee: i did try that, but imho it is really bad designed... some strange setup routine trying to get root access without telling what for exactly, and so on... |
| 17:03.22 | henk | interface just disappears but zoiper still runs... |
| 17:03.28 | henk | tzafrir_laptop: hm? |
| 17:03.47 | tzafrir_laptop | axisys, /etc/zaptel.conf , /etc/asterisk/zapata.conf . .What version of Asterisk is it? |
| 17:03.56 | henk | Chainsaw: afaik has iax not as many problems with nat as sip has. isn't that so? |
| 17:04.11 | axisys | tzafrir_laptop: 1.4.23.2 |
| 17:04.16 | tzafrir_laptop | henk: to get the audio path through the remote server :-( |
| 17:04.35 | axisys | tzafrir_laptop: dont see zapata.conf |
| 17:04.49 | henk | tzafrir_laptop: i'm not even at the point where i care about audio... |
| 17:05.13 | axisys | tzafrir_laptop: ls /etc/asterisk/za* |
| 17:05.14 | axisys | ls: cannot access /etc/asterisk/za*: No such file or directory |
| 17:05.17 | tzafrir_laptop | axisys, there's only a sample chan_dahdi.conf . But if you built asterisk vs. zaptel (rather than dahdi) you should call it zapata.conf |
| 17:05.29 | Chainsaw | henk: Everything has problems with NAT. That's because it's a horrible hack. |
| 17:05.56 | Chainsaw | henk: Note the suggestion from tzafrir_laptop. |
| 17:05.58 | axisys | tzafrir_laptop: i see the chan_dahdi.conf .. yes i am using zaptel.. |
| 17:06.12 | henk | Chainsaw: hm ok... thanks |
| 17:06.26 | axisys | tzafrir_laptop: cp chan_dahdi.conf zapata.conf (done) |
| 17:07.43 | axisys | tzafrir_laptop: http://pastebin.com/f22a19082 |
| 17:10.40 | henk | tzafrir_laptop: canreinvite is already configured... |
| 17:15.45 | *** join/#asterisk eric256 (n=Administ@c-75-71-188-88.hsd1.co.comcast.net) |
| 17:16.50 | tzafrir_laptop | axisys, what is the output of lszaptel ? |
| 17:20.59 | kb3ien | anyone tried to overide the location of the pid file: i'm getting this : Unable to open pid file '/var/run/asterisk.pid': Permission denied |
| 17:21.06 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:21.39 | kb3ien | <PROTECTED> |
| 17:22.42 | eric256 | could somebody look at this sip debug dump and tell me if they know why its getting declined? http://rafb.net/p/nShhj829.html |
| 17:23.06 | eric256 | i'm on extension 6000, outbound calls work fine, but calls to other extensions always come back "rejected" |
| 17:23.24 | axisys | tzafrir_laptop: http://pastebin.com/f390b313 |
| 17:23.52 | axisys | tzafrir_laptop: i have nothing connected to them.. why saying in use? |
| 17:24.24 | tzafrir_laptop | axisys, so I see 4 channels. In Use: probably by Asterisk |
| 17:24.33 | axisys | tzafrir_laptop: oh ok |
| 17:24.40 | [TK]D-Fender | eric256: because * hung up before anything answered |
| 17:24.47 | tzafrir_laptop | So it seems to be OK. What's the output of: asterisk -rx 'dahdi show channels' |
| 17:25.04 | [TK]D-Fender | eric256: - Executing [vmret@from-internal:2] Hangup("SIP/6000-087d8f88", "") in new stack |
| 17:25.18 | eric256 | [TK]D-Fender: hmm and now it says number not valid for several of the extensions... |
| 17:25.22 | axisys | tzafrir_laptop: http://pastebin.com/f96a3246 |
| 17:25.38 | [TK]D-Fender | eric256: Funny this doesn't say that... the exten is valid, go look at what you're DOING |
| 17:26.21 | eric256 | no that call just said call rejected |
| 17:26.25 | areay | I have my server set up, my ports are forwarded, and asterisk is configured to work behind NAT... But I'm still getting one-way audio... Any suggestions? |
| 17:26.40 | eric256 | the other calls said invalid number... /me goes to review the call plan |
| 17:27.38 | [TK]D-Fender | are check the firewall on the server |
| 17:27.47 | [TK]D-Fender | areay: check the firewall on the server |
| 17:28.06 | areay | [TK]D-Fender, i thought about that... just apt-get remove --purge firestarter yeah? |
| 17:28.13 | [TK]D-Fender | areay: NO |
| 17:28.20 | [TK]D-Fender | areay: I said go LOOK AT IT |
| 17:28.33 | [TK]D-Fender | areay: Stop trying to burn the evidence |
| 17:28.38 | areay | [TK]D-Fender, lol |
| 17:29.23 | kb3ien | ah found it, the problem is [directories](!) the (1) needs to come out. weird... |
| 17:30.09 | areay | [TK]D-Fender, i know it's not secure, but right now i have ports 5060 and 10000-20000 open to all |
| 17:30.21 | [TK]D-Fender | areay: "iptables --flush" |
| 17:33.44 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 17:36.57 | eric256 | okay i discovered unsurprisingly that i'm a moron |
| 17:36.58 | eric256 | lol |
| 17:37.29 | eric256 | i included a file and one of its extensions+priority was overwriting an existing one...now i just have to figure out how to get around that. |
| 17:44.46 | *** join/#asterisk areay (n=areay@93-97-161-123.zone5.bethere.co.uk) |
| 17:45.24 | areay | [TK]D-Fender, now it wont register to my sip trunk |
| 17:46.47 | areay | [TK]D-Fender, ok i got it now...i messed with my bindaddr before and forgot... |
| 17:47.29 | areay | [TK]D-Fender, still one-way audio tho... |
| 17:49.13 | RypPn | areay is this the voiptalk trunk from yesterday still ? |
| 17:50.46 | areay | RypPn, yeah its the same trunk lol... i've reinstalled asterisk about 6 times now, been workin on it for a week... I'm gonna be an expert by the time this is set up lol |
| 17:53.01 | areay | RypPn, you said something about voiptalk yesterday... i forgot what tho |
| 17:53.11 | RypPn | areay I PM'ed you |
| 17:59.26 | *** join/#asterisk raden_work (n=jon@adsl-99-139-235-165.dsl.applwi.sbcglobal.net) |
| 18:00.00 | raden_work | is there a particular OS that seams to work better with asterisk ? SUSE / Debian / Ubuntu ? |
| 18:01.32 | _BBV_ | centos :) |
| 18:02.01 | Kumba_ | :( |
| 18:06.15 | *** join/#asterisk inv_arp (n=junya@b07s03mr.corenetworks.net) |
| 18:09.10 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 18:09.41 | *** join/#asterisk freddyk (n=freddy@host124-190-dynamic.51-82-r.retail.telecomitalia.it) |
| 18:09.53 | freddyk | anyone could help with asterisk 1.6.1 trunk ? |
| 18:10.01 | freddyk | i have problems with sip register |
| 18:10.30 | axisys | why is the call to my pstn number from my cell phone failing like this http://pastebin.com/f3b00ba57 ? I am using this as my extension file http://pastebin.com/f5736782e (sanitized) |
| 18:11.10 | *** join/#asterisk nima0102 (n=nima@91.98.221.195) |
| 18:11.23 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 18:13.02 | kaldemar | axisys: because you don't have anything matching s in your default context. fix context parameters in zapata.conf and then fix your dialplan to match s. |
| 18:13.35 | axisys | kaldemar: i dialed my home number.. why is it trying to match `s' ? |
| 18:14.42 | kaldemar | the channel driver works that way. it makes no difference what number you dial. assuming you have an analog line. |
| 18:14.53 | axisys | kaldemar: yes |
| 18:15.07 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 18:16.07 | axisys | kaldemar: if I uncomment the default section here http://pastebin.com/f5736782e .. it works |
| 18:16.20 | axisys | kaldemar: hmm .. so it must need a `s' pattern |
| 18:17.19 | kaldemar | a pattern has another meaning in dialplans, s is just an extension. |
| 18:17.33 | axisys | kaldemar: k |
| 18:19.44 | *** join/#asterisk freddyk (n=freddy@host124-190-dynamic.51-82-r.retail.telecomitalia.it) |
| 18:26.02 | *** join/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com) |
| 18:30.45 | *** join/#asterisk km2 (n=x@cpe-74-64-12-212.nyc.res.rr.com) |
| 18:32.59 | axisys | how do I tell see the callerid here http://pastebin.com/fb5e7185 ? |
| 18:33.38 | axisys | and the number that is being called .. i set the verbose to 255 and still not seeing more than above paste |
| 18:41.08 | kaldemar | axisys: set a noop in the dialplan that prints it. |
| 18:41.29 | kaldemar | NoOp(${CALLERID(num)}) |
| 18:44.10 | Qwell | raden_work: if it means anything, most of the developers at Digium use Debian (or a deriv, like Ubuntu) to do their development |
| 18:44.36 | raden_work | i was just thinking ubuntu server edition so its not soo bloated |
| 18:44.40 | Qwell | but, Linux is Linux |
| 18:45.05 | Qwell | it really shouldn't matter what you use - just use whatever you're familiar with |
| 18:45.34 | Qwell | (assuming you're familiar with any of them..) |
| 18:51.04 | *** join/#asterisk UQlev (n=kvirc@91.184.221.31) |
| 18:52.55 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 18:53.57 | *** join/#asterisk CapriCoRN^80 (n=int@207.176.6.153) |
| 18:54.00 | CapriCoRN^80 | hi |
| 18:54.39 | CapriCoRN^80 | how we can improve the voice quality over * ? |
| 18:55.05 | *** join/#asterisk whitehat (n=whitehat@unaffiliated/whitehat) |
| 18:55.14 | Qwell | CapriCoRN^80: use higher quality codecs/endpoints |
| 18:56.34 | CapriCoRN^80 | Qwell: any link that contains such info ? |
| 18:58.15 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 18:58.17 | UQlev | CapriCoRN^80: voice quality depends on 1) your audio settings 2) codec selected |
| 18:58.38 | CapriCoRN^80 | UQlev: ok |
| 18:58.49 | CapriCoRN^80 | UQlev: can we do some audio settings in * server ? |
| 18:59.14 | UQlev | CapriCoRN^80: at 1st your have to check audiorecording on your client PC |
| 18:59.32 | UQlev | CapriCoRN^80: makes no sense to do any audio settings on * |
| 18:59.52 | UQlev | CapriCoRN^80: * only transmits what you have recorded |
| 19:00.11 | *** join/#asterisk ManxPower (n=manxpowe@user-24-236-95-236.knology.net) |
| 19:00.16 | CapriCoRN^80 | UQlev: agree . but may be the companies that are providing some commercial softwares put such options |
| 19:00.47 | UQlev | what options? |
| 19:00.59 | UQlev | to improve voice quality? |
| 19:01.35 | CapriCoRN^80 | actually one of my friend was telling me that he downloaded some commercial software that is build using * and in that i did some voice relating settings |
| 19:01.53 | CapriCoRN^80 | audio related settings |
| 19:01.55 | CapriCoRN^80 | anyways |
| 19:02.12 | CapriCoRN^80 | UQlev: can you refere me some sites that contains such info |
| 19:02.23 | UQlev | give me some examples of audiosettings on * pls |
| 19:03.50 | UQlev | CapriCoRN^80: I know only that * can restrict using particular codec(s) or allow all |
| 19:03.51 | CapriCoRN^80 | UQlev: i am not sure about that i told you that my friend got some commercial software and as i said may be the company has put some other options which he thought is related to * |
| 19:04.59 | CapriCoRN^80 | UQlev: http://pastebin.com/m65a6c844 |
| 19:05.21 | CapriCoRN^80 | what about bandwidth=high ? you what is the purpose of it ? |
| 19:06.20 | UQlev | bandwidth hi means to select codecs like ulaw/alaw |
| 19:06.42 | CapriCoRN^80 | ok |
| 19:06.47 | UQlev | ulaw/alaw give better quality but need wider band |
| 19:06.55 | CapriCoRN^80 | did you check my pastbin ? |
| 19:07.22 | UQlev | yes, I see only selection of codecs |
| 19:07.41 | CapriCoRN^80 | UQlev: is that ok ? |
| 19:07.54 | *** join/#asterisk pmhaddad (n=pmhaddad@141.219.87.43) |
| 19:07.54 | UQlev | yes, nothing particular |
| 19:08.09 | UQlev | codecs selection mostly depends on your client |
| 19:08.25 | UQlev | I have no clients supporting 729 yet |
| 19:09.20 | axisys | i am getting loop detected http://pastebin.com/f6b183bfd .. trying to ring the the phone (while calling from cell) and then switch to voicemail.. here is the extension file http://pastebin.com/f23296d40 |
| 19:11.11 | CapriCoRN^80 | UQlev: should i put bandwith=high in each user setting ? |
| 19:11.47 | UQlev | CapriCoRN^80: you may safely put allow=all |
| 19:12.12 | CapriCoRN^80 | ok |
| 19:12.56 | UQlev | CapriCoRN^80: what is your internet connection and how many clients simultaneous? |
| 19:13.28 | CapriCoRN^80 | 10 Mbps speed at server side and 30 clients so far |
| 19:14.24 | UQlev | CapriCoRN^80: 30x60=1.8mbps |
| 19:14.54 | UQlev | even for hiband |
| 19:15.57 | CapriCoRN^80 | UQlev: ok |
| 19:16.03 | UQlev | you may safely increase twice your customers before to force them to use gsm |
| 19:16.56 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 19:17.13 | CapriCoRN^80 | UQlev: but the voice quality is not good as it should be even with 3 customer is using it |
| 19:17.51 | UQlev | the reason of bad voice quality in something else |
| 19:18.23 | UQlev | CapriCoRN^80: * usually doesn't transcode, only switch packets |
| 19:19.52 | UQlev | sip clients are also very sensitive to other applications running, e.g. some clients may suffer if you open pdf file during phone call |
| 19:20.14 | UQlev | and whole talk may be broken |
| 19:24.37 | *** join/#asterisk CrazyTux (n=brandon@99-53-97-235.lightspeed.cyprtx.sbcglobal.net) |
| 19:32.53 | *** part/#asterisk mphill_ (n=mphill@CPE-70-92-241-36.wi.res.rr.com) |
| 19:33.57 | apeiron | Opening a PDF file? Really? |
| 19:34.00 | apeiron | That seems a bit... extreme. |
| 19:34.22 | apeiron | I find it unlikely that a dual core 2.5GHz box would have problems running a SIP client and a PDF viewer. o_O |
| 19:34.47 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 19:35.58 | UQlev | apeiron: not everyone running dualcore CPU, I am still on a celeron |
| 19:36.14 | apeiron | UQlev, you poor poor bastard |
| 19:36.45 | henk | smartphones have even less power... |
| 19:36.50 | UQlev | considers apeiron as rich bastard |
| 19:37.31 | apeiron | .. |
| 19:37.34 | apeiron | UQlev, http://www.newegg.com/Product/Product.aspx?Item=N82E16819116063 |
| 19:38.11 | apeiron | You can go $10 less than that if you're willing to have a dual-core Celery. |
| 19:38.53 | *** join/#asterisk nima0102 (n=nima@91.98.221.195) |
| 19:39.18 | henk | --> matching mainboard --> matching ram --> matching graphics card --> 200$ or more |
| 19:39.39 | apeiron | -50 or so for graphics, don't need unless you're gaming |
| 19:40.54 | UQlev | thinks that apeiron uses all-sockets-compatible mobos |
| 19:41.41 | apeiron | All I'm saying is that there's very little reason to still be on a Celery unless you don't have your own source of income. |
| 19:41.43 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 19:42.02 | *** part/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 19:42.04 | CapriCoRN^80 | UQlev: ok |
| 19:42.20 | henk | or more important things to use your money for or a very low income ... |
| 19:42.54 | *** join/#asterisk mesfet (n=iw3grx@host165-3-static.25-87-b.business.telecomitalia.it) |
| 19:43.05 | EmleyMoor | I am hoping my last Celeron box lasts until at least the squeeze release |
| 19:43.06 | UQlev | apeiron: I am not uset to "shoot parrots from a canon" |
| 19:43.22 | apeiron | UQlev, Pardon? |
| 19:43.38 | EmleyMoor | It may be possible to decommission it if I am lucky |
| 19:44.56 | UQlev | apeiron: I just want to thank you very much for your advise but I have been dealing with computers since 1985 and know what is reasonable |
| 19:45.16 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:45.17 | apeiron | I beg to differ. Celery processors are not reasonable. :) |
| 19:47.25 | UQlev | apeiron: my office PC is 2 GHz Celeron, and it is enough, for my daughter I buy Core 2quad 3.0 GHz for Acad and it is also enough |
| 19:48.30 | UQlev | apeiron: I am aware that there are some other CPU beside Celeron |
| 19:48.40 | apeiron | I, um, never said you weren't. |
| 19:49.01 | *** join/#asterisk Dovid (n=annon@tony09-121-90.inter.net.il) |
| 19:53.55 | *** join/#asterisk keith4_ (n=keith@216-164-144-176.c3-0.eas-ubr9.atw-eas.pa.cable.rcn.com) |
| 19:54.32 | keith4_ | I upgraded my kernel to 2.6.26, rebuilt the zaptel driver, but can't insert ztdummy: ztdummy: Unknown symbol rtc_register |
| 19:54.46 | keith4_ | same for "rtc_unregister" and "rtc_control" |
| 19:55.17 | Chainsaw | keith4: Please make sure that you have the real RTC driver in your kernel, and not the emulation. |
| 19:56.44 | keith4_ | how would I do that? |
| 19:58.24 | tzafrir_laptop | freddyk, I'm really not a SIP expert, but please provide some data |
| 19:58.36 | tzafrir_laptop | what messages did you get? |
| 19:59.02 | tzafrir_laptop | the standard questions: what did you do? What have you expected to happen? What actually has happened? |
| 20:02.40 | freddyk | tzafrir_laptop: just a message from acl.c regarding resolve [peer name] failed. |
| 20:02.57 | freddyk | tzafrir_laptop: cut & paste it. wait |
| 20:03.22 | tzafrir_laptop | ~pb |
| 20:03.23 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 20:05.55 | *** join/#asterisk Magicblaze007 (n=sony@fl-67-233-204-182.dhcp.embarqhsd.net) |
| 20:06.38 | freddyk | tzafir_laptop: it's just one raw, here it is [Mar 15 21:05:50] WARNING[5697]: acl.c:385 ast_get_ip_or_srv: Unable to lookup 'trunk' |
| 20:07.14 | freddyk | tzafrir_laptop: and, [Mar 15 21:06:49] WARNING[5743]: chan_sip.c:4917 create_addr: No such host: trunk |
| 20:07.49 | freddyk | tzafrir_laptop: this configuration was perfectly working on 1.6.0.6 |
| 20:09.12 | tzafrir_laptop | freddyk, is 'trunk' an explicit peer host? |
| 20:09.51 | freddyk | tzafrir_laptop: it's defined as [trunk] in sip.conf |
| 20:10.27 | tzafrir_laptop | is there any keyword in that entry that should have overriden 'trunk' as the peer hostname? |
| 20:11.07 | freddyk | tzafrir_laptop: host=10.10.10.1 |
| 20:12.39 | tzafrir_laptop | freddyk, I suppose it might help if you pastebin the relevant parts of the configuration |
| 20:13.02 | freddyk | ok |
| 20:27.57 | freddyk | tzafrir_laptop: http://pastebin.com/m5499aafa |
| 20:28.06 | freddyk | tzafrir_laptop: take a look to configuration |
| 20:33.25 | axisys | i still cant fig out why I am getting Loop detected http://pastebin.com/f3701ad56 |
| 20:34.04 | tzafrir_laptop | freddyk, when you run 'sip show peers', do you see the correct host names? |
| 20:34.12 | tzafrir_laptop | or 'trunk*'? |
| 20:36.19 | freddyk | trunk was a dummy i used before |
| 20:36.49 | freddyk | when i do sip show peers i see correct peer informations |
| 20:38.03 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 20:40.08 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 20:46.29 | freddyk | tzafrir: BRB going out for dinner |
| 20:49.59 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 20:50.45 | *** join/#asterisk CrazyTux (n=brandon@99-53-97-235.lightspeed.cyprtx.sbcglobal.net) |
| 20:54.19 | axisys | sip: http://pastebin.com/f66ca2f5e and extension: http://pastebin.com/f43803e3c |
| 20:54.43 | axisys | i am sure something simple here that is causing the loop |
| 20:54.56 | axisys | but i can't fig it out |
| 21:04.36 | keith4_ | will dahdi 2.1.x work with asterisk 1.4? |
| 21:08.19 | *** join/#asterisk saxa (n=sasa@host242-95-static.223-217-b.business.telecomitalia.it) |
| 21:08.24 | saxa | hi ppl |
| 21:08.34 | saxa | I have some doubts |
| 21:08.55 | saxa | I have 2 offices |
| 21:09.09 | saxa | one in Brazil and another in Italy |
| 21:09.34 | saxa | so I want to do that when the phone rings in Brazil I get it transferred via internet to Italy |
| 21:09.46 | saxa | I think this is possible to do with Asterisk |
| 21:10.00 | Chainsaw | Should be, yes. |
| 21:10.11 | saxa | so I was looking at digium.com and I tought I should buy one IAXy |
| 21:10.23 | saxa | or do I need some other kind of equipment |
| 21:10.39 | saxa | also because I will have to use a phone over there in Brazil |
| 21:11.05 | *** join/#asterisk FunkyGMT (n=Hayes@modemcable107.105-58-74.mc.videotron.ca) |
| 21:11.36 | FunkyGMT | Question: I'm having problem to set the channel language on a local channel. Anybody knows how I can set that ? |
| 21:12.14 | saxa | Chainsaw: do you have any advices how should I set up the thing ? |
| 21:12.22 | saxa | I'm new to voip thing |
| 21:12.41 | Chainsaw | saxa: Well, it really depends on how you *want* to set it up. |
| 21:12.57 | Chainsaw | saxa: Do you want to keep your existing phones in Brazil & Italy, or are you okay with changing them. |
| 21:13.17 | saxa | i have no phones yet in Brazil |
| 21:13.20 | saxa | just in Italy |
| 21:13.20 | Chainsaw | saxa: What kind of connection do you have to the phone network in Brazil, is there an existing phone network connection in Italy that has to work with the system, etc. |
| 21:13.47 | saxa | I have a adsl here in Italy and I will have that in Brazil too |
| 21:13.51 | Chainsaw | Right. And what kind of connection to the phone network do you have in Italy? ISDN (BRI/PRI)? |
| 21:14.02 | saxa | no just analog |
| 21:14.06 | Chainsaw | Okay, so just a single copper phoneline that you get for "free" with ADSL, k. |
| 21:14.11 | saxa | yup |
| 21:14.19 | saxa | no |
| 21:14.33 | saxa | actually I have 3 copper phone lines here in Italy |
| 21:14.50 | saxa | one is for internet , one is for phone another is for a fax |
| 21:15.04 | *** join/#asterisk k-man_ (n=jason@unaffiliated/k-man) |
| 21:15.07 | Chainsaw | Okay. And you'd want to "transfer" two of those to Brazil if you're there? |
| 21:15.38 | saxa | possibly, but the primary thing is to transfer from Brazil to Italy |
| 21:15.53 | Chainsaw | But you just told me you have no phone network in Brazil yet. |
| 21:16.01 | Chainsaw | Or at least, no connection to it in your office. |
| 21:16.06 | saxa | no, but I will buy one |
| 21:16.11 | Chainsaw | What kind? |
| 21:16.20 | saxa | analog simple one ? |
| 21:16.23 | Chainsaw | Okay. |
| 21:16.27 | Chainsaw | And what kind of phone will you be using? |
| 21:16.30 | saxa | depends, I need to decide still |
| 21:16.54 | saxa | I can buy a voip phone for there |
| 21:16.58 | Chainsaw | I can't advise you what building blocks to put in your network until you know what the surrounding areas look like. |
| 21:17.21 | Chainsaw | Okay. In that case you would need a FXO interface on your Asterisk server (or appliance, or other device) to talk to the Brazil phone network. |
| 21:17.34 | saxa | yup |
| 21:17.43 | saxa | I tought about IAXy for that |
| 21:17.48 | Chainsaw | The SIP phone in Brazil can talk directly to that. If it's not a problem to have another SIP phone in Italy, you hook that up to your ADSL router there and you're mostly done. |
| 21:18.41 | saxa | i was thinking to set up a server in brazil to connect to the phone line, normal one |
| 21:18.50 | saxa | normal copper phone line |
| 21:18.54 | Chainsaw | I normally build things based around servers, so I'm not too familiar with Digium's commercial offerings. |
| 21:19.12 | saxa | then asterisk would transfer the incoming calls from brazil to Italy |
| 21:19.53 | saxa | the first thing is to know if its better to buy a PCI card and run a server 24/7 |
| 21:20.06 | saxa | in Brazil or to do it with IAXy |
| 21:20.16 | Chainsaw | Can you link me to this IAXy product please? |
| 21:20.37 | saxa | http://www.digium.com/en/products/analog/s101i.php |
| 21:21.13 | *** join/#asterisk UQlev (n=kvirc@91.184.221.31) |
| 21:21.13 | Chainsaw | saxa: But that's an FXS port. |
| 21:21.32 | Chainsaw | saxa: i.e. it's a connection point for analog phones, connecting them to an Asterisk server instead of to a regular phone network. |
| 21:21.48 | Chainsaw | saxa: You want an FXO to talk to the regular phone network, and feed that into an Asterisk server instead of a phone. Right? |
| 21:22.11 | Chainsaw | You could use that IAXy device at one end, if you want to use an analog phone instead of a SIP one. |
| 21:22.48 | saxa | Chainsaw: no, I want to connect the Brazilian phone, which will be a copper most probably and transfer all the incoming calls from that line to my office in Italy |
| 21:23.06 | Chainsaw | saxa: Then that device is not suitable, sorry. You want an FXO, not an FXS. |
| 21:23.29 | saxa | I tought to connect that IAXy in brazil to a server also in the same office |
| 21:23.30 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 21:23.41 | Chainsaw | Well, you wouldn't have to. |
| 21:23.48 | Chainsaw | You can get PCI cards with FXO & FXS modules on the same card. |
| 21:23.51 | saxa | which will receive incoming calls and transfer them trough internet to Italy |
| 21:24.05 | Chainsaw | So you could get say... 2x FXO to connect to the phone network (and have one spare for later) and 2x FXS to connect analog phones. |
| 21:24.21 | Chainsaw | That would be in Brazil. |
| 21:24.34 | Chainsaw | And then in Italy, you can just connect a SIP phone to your ADSL router to take any Brazilian calls. |
| 21:24.45 | saxa | thats it |
| 21:25.13 | saxa | Chainsaw: so I shoud buy an digium card for my server I will set up in Brazil correct ? |
| 21:25.17 | Chainsaw | Also gives you options in Brazil, in that you can use analog phones or SIP phones, but you won't be locked in to either. |
| 21:25.31 | saxa | ok |
| 21:25.37 | Chainsaw | saxa: I would say yes, a WildCard 410 with 2x FXO and 2x FXS. |
| 21:25.49 | Chainsaw | That gives you most flexibility on the Brazil end of the network. |
| 21:25.55 | saxa | I tought that IAXy connects a phone line to the asterisk server |
| 21:26.10 | Chainsaw | It does, yes. But you don't always need one. |
| 21:26.24 | Chainsaw | The IAXy is more of a CPE (Customer Premises Equipment). |
| 21:26.32 | saxa | the IAXy basically is used what for ? |
| 21:26.37 | Chainsaw | i.e. some random VoiP provider wants to offer something you just plug your old phone into. |
| 21:26.55 | saxa | oh ok, inverse of what I tought |
| 21:27.02 | Chainsaw | So you disconnect from the phone network entirely, plug your existing handsets into the IAXy and go through their VoiP network for your calls. |
| 21:27.18 | saxa | ok |
| 21:27.27 | Chainsaw | You could use one at the Italy end of your network. |
| 21:27.35 | saxa | of course |
| 21:27.37 | Chainsaw | To connect an analog phone for your Brazil calls, instead of a SIP phone. |
| 21:27.45 | saxa | in case i would like to ring my normal fone here |
| 21:27.46 | Chainsaw | But in Brazil, no, I don't see a need for it. |
| 21:28.16 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 21:28.20 | saxa | Chainsaw: great, many thanks for clarifying me the thing |
| 21:29.07 | *** join/#asterisk AndyML (n=quassel@pool-98-111-140-229.phlapa.fios.verizon.net) |
| 21:29.10 | Chainsaw | saxa: Any time. I find it's best to draw up a diagram of how everything's going to work before you actually go out and buy hardware. |
| 21:29.31 | saxa | of course |
| 21:29.49 | saxa | I will try to ,also I will try to see whats best thing |
| 21:30.16 | saxa | Chainsaw: can you tell me , the model is WILDCARD 410 ? |
| 21:30.34 | saxa | is under digital or analog products ? |
| 21:31.09 | Chainsaw | It's an analog product. |
| 21:31.11 | AndyML | i'm having problems with dahdi - it 'starts', dahdi_cfg -vvv shows the appropriate stuff, but asterisk won't even run commands starting with 'dahdi' (dahdi show status, etc) i've tried module load chan_dahdi.so etc - no luck. - http://pastebin.com/m2b028afc |
| 21:32.15 | saxa | Chainsaw: so it would be a AEX410 ? |
| 21:32.24 | mvanbaak | AndyML: did you install asterisk first, and later added dahdi ? if so you have to recompile asterisk to pickup dahdi |
| 21:32.30 | Chainsaw | saxa: Can you link me to it? |
| 21:32.35 | saxa | Chainsaw: or a TDM410 |
| 21:32.41 | Chainsaw | TDM410 sounds more like it. |
| 21:32.45 | AndyML | i did recompile asterisk... i suppose i could do it again? |
| 21:32.46 | saxa | http://www.digium.com/en/products/analog/aex410.php |
| 21:32.52 | AndyML | is anythign else worth checking in the meantime? |
| 21:33.11 | Chainsaw | saxa: Ah, it seems the AEX410 is just the PCI-Express edition of the TDM410 (which is a regular PCI card). |
| 21:33.22 | mvanbaak | AndyML: make clean; ./configure; make; make install |
| 21:33.24 | Chainsaw | saxa: So it depends on what slots your server has to offer. |
| 21:33.34 | AndyML | thanks mvanbaak |
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| 21:34.32 | AndyML | stevetotaro: hey there |
| 21:34.45 | AndyML | I'm andy from Anteil - Jim's friend |
| 21:35.43 | saxa | Chainsaw: so those would be the right cards correct ? |
| 21:36.12 | saxa | but how do I connect the phone line into it ? it seems that it has a RJ45 and not a RJ11 |
| 21:36.21 | Chainsaw | saxa: Yes, those look right. And as I said, 2x FXS 2x FXO. |
| 21:36.39 | Chainsaw | saxa: Those are RJ11 ports. |
| 21:36.47 | saxa | all 4 ? |
| 21:37.02 | Chainsaw | Yes. |
| 21:37.03 | saxa | RJ11 is the small one like in analog phones correct ? |
| 21:37.07 | Chainsaw | The very same. |
| 21:37.11 | stevetotaro | hi Andy |
| 21:37.35 | Chainsaw | And the little lights are there to tell you what that port is (FXO or FXS). |
| 21:37.52 | Chainsaw | Because it depends on the modules you install (or have Digium install for you as buy it) on the card. |
| 21:38.10 | saxa | Chainsaw: on, so when I plug in my phone line in the card, the card will automatically know that is a phone line ? |
| 21:38.41 | stevetotaro | oh, hey Andy, how do feel? |
| 21:38.42 | Chainsaw | saxa: Well, you determine whether the port is for connection to a phone network (FXO) or to a phone handset (FXS). |
| 21:39.05 | Chainsaw | saxa: So the card won't "know" it. You'll have made that choice a lot earlier. |
| 21:39.19 | saxa | then in another socket I will use for a voip phone ? |
| 21:39.22 | Chainsaw | saxa: And you tell the driver how you have configured your 4 ports (by choosing a red or green module for each of the four ports). |
| 21:39.26 | AndyML | stevetotaro: feelin' good, thanks. yourself? |
| 21:39.33 | Chainsaw | saxa: Your VoiP phone will connect over ethernet, not RJ11. |
| 21:39.48 | saxa | oh ok |
| 21:40.00 | Chainsaw | saxa: Those two FXS ports would be for analog phones. |
| 21:40.03 | *** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net) |
| 21:40.06 | Chainsaw | saxa: Or an analog phone and a fax. |
| 21:40.19 | stevetotaro | not bad, just made the drive down from L-Town to south of Baltimore |
| 21:40.26 | saxa | ok, so the card will connect astersk server to the phone line, and I will use a voip phone directly in my local router correct ? |
| 21:40.42 | Chainsaw | saxa: Correct. |
| 21:41.11 | saxa | good, so basically I will have for example 2 lines in brasil in the same card connected |
| 21:41.25 | saxa | if I will want to have 2 lines |
| 21:41.27 | Chainsaw | Indeed. |
| 21:41.35 | Chainsaw | Then you connect your two FXO ports to the Brasil phone network. |
| 21:41.41 | saxa | at the beginning i will have one only. |
| 21:41.42 | Chainsaw | And any analog phones to the two FXS ports on the cards. |
| 21:41.53 | Chainsaw | *nod* Then you just leave one FXO open for later use. |
| 21:42.36 | saxa | ok, so this is configured when I buy the card, 2 FXO and 2FXS |
| 21:43.08 | Chainsaw | *nod* You can select it then. |
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| 21:44.07 | saxa | ok ok, thx Chainsaw |
| 21:44.36 | saxa | sorry but I just want to understand before I do any kind of buying |
| 21:44.43 | Chainsaw | *nod* That's fine. |
| 21:44.56 | Chainsaw | If you do decide to go for the PCI Express card, I would advise to purchase the echo cancellation. |
| 21:45.02 | Chainsaw | That way your PC/server won't have to do that job. |
| 21:45.17 | saxa | ok |
| 21:45.28 | Chainsaw | (As you know there's always some echo on analog phones, and you can leave it to that purple module to handle, or to the driver, where it will cost CPU time) |
| 21:45.31 | saxa | actually I need to buy a PC for server use also |
| 21:46.01 | saxa | Chainsaw: is it ok to buy a 64bit cpu ? |
| 21:46.23 | Chainsaw | saxa: Yes, no problem with that. |
| 21:46.37 | Chainsaw | saxa: Even Asterisk 1.2 works perfectly alright on a 64-bit system. |
| 21:48.13 | saxa | i compilled asterisk here on my amd64 without problems, but was version 1.6 |
| 21:48.29 | saxa | so i think it will be everyhting ok |
| 21:48.40 | Chainsaw | Yes, I would indeed advise to go with 1.6 for new installs. |
| 21:48.49 | saxa | good |
| 21:49.21 | saxa | then its just a matter of configuration i think to do the transfer from brazilian asterisk to the italian one |
| 21:49.36 | saxa | i need to run asterisk also on my italian side correct ? |
| 21:49.45 | Chainsaw | Not necessarily, no. |
| 21:49.55 | Chainsaw | You could connect your SIP phone in Italy with your Asterisk server in Brazil. |
| 21:50.24 | saxa | oh I can plug the voip phone into my adsl and configure it to use the server in brazil |
| 21:50.28 | Chainsaw | Indeed. |
| 21:50.30 | saxa | ok |
| 21:50.40 | saxa | now there is another problem i will probably face |
| 21:50.47 | Chainsaw | I'm listening. |
| 21:51.00 | saxa | i dont know if in brazil i will have a static IP |
| 21:51.18 | Chainsaw | You could mitigate that with a service like dyndns. |
| 21:51.25 | Chainsaw | And have your router manage the hostname. |
| 21:51.37 | saxa | oh yes, of course |
| 21:51.40 | Chainsaw | So then your asterisk server would be on something like saxa-brazil.dyndns.org |
| 21:51.59 | Chainsaw | It's not ideal, but not a deal breaker :) |
| 21:52.15 | saxa | is it possible if I run also asteris in italy |
| 21:52.24 | saxa | to connect those 2 servers together ? |
| 21:52.27 | Chainsaw | Certainly. |
| 21:52.32 | Chainsaw | There's where you would use IAX. |
| 21:52.43 | saxa | i mean, that if the server in brazil gets out of game for some reason |
| 21:52.53 | saxa | then when it comes back up |
| 21:52.55 | Chainsaw | That's a protocol to have two Asterisk servers communicate. You could then also transfer calls from Brazil to Italy and back. |
| 21:53.01 | saxa | it looks for asterisk in italy ? |
| 21:53.21 | Chainsaw | Well, seeing how your phone line is still physically in Brazil, it wouldn't help you with availability. |
| 21:53.35 | Chainsaw | (Unless you're planning to also be availble over SIP) |
| 21:53.42 | *** join/#asterisk RonDutt (n=thedot@c-24-17-159-108.hsd1.wa.comcast.net) |
| 21:53.53 | saxa | Chainsaw: i was saying because i'm not sure what kind of internet connection will I have there |
| 21:54.18 | RonDutt | Having a little trouble with inbound calls...I get a ss-noservice then congestion. Any ideas on how to go about diagnosing this? |
| 21:54.19 | saxa | also another thing is the bandwith |
| 21:54.46 | saxa | probably that adsl in brasil will be used by many people |
| 21:55.04 | saxa | so i think it will be a 1MB link |
| 21:55.12 | Chainsaw | saxa: You should avoid a congested link. |
| 21:55.20 | saxa | not much, but it should suffice for phone calls correct ? |
| 21:55.23 | Chainsaw | saxa: VoiP, like all phone services, depends on a low latency. |
| 21:55.40 | Chainsaw | saxa: That does mean that there is barely any error handling, or correction for missed packets (due to congestion). |
| 21:55.50 | saxa | ok got it |
| 21:55.56 | Chainsaw | saxa: So if you share that 1mbit line with say... 3 others, and 2 of them are running BitTorrent. |
| 21:55.59 | saxa | then is better i order my own adsl |
| 21:56.14 | saxa | Chainsaw: hopefully not :)) |
| 21:56.20 | Chainsaw | Yo....ur pho....ne...calls w...ill so...und li....ke thi....s |
| 21:56.24 | saxa | but yes, who knows |
| 21:56.40 | saxa | Chainsaw: heheh |
| 21:56.58 | saxa | ok, got it |
| 21:57.09 | Chainsaw | 1mbit is fine, if you have it to yourself. Avoid heavy usage or consider using a router with QoS (quality-of-service) routing that lets VoiP traffic go first if it's busy. |
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| 21:59.49 | RonDutt | Bah! |
| 22:01.50 | RonDutt | Woohoo! |
| 22:03.04 | saxa | Chainsaw: many many thanks for your help !!! |
| 22:03.15 | Chainsaw | saxa: You're welcome :) |
| 22:03.22 | saxa | I go to sleep now, as its alredy late here |
| 22:03.33 | saxa | i will be back in here soon :) |
| 22:03.34 | RonDutt | Having a context line = very helpful ;) |
| 22:03.50 | saxa | night ppl |
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| 22:08.55 | Chainsaw | saxa: Night. |
| 22:11.53 | wonderworld | sigh |
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| 22:37.10 | sprite-- | Any updates on this issue? http://forums.digium.com/viewtopic.php?p=125675&sid=499c6f96ca3dca2da855d1ad45887efd |
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| 23:48.28 | snowboarder04 | anyone had any experience with FreeSpeech (a UK SIP provider)? Asterisk is saying the SIP registration to them is active, however no calls are being routed to the inbound route catch-all :/ |
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| 23:52.09 | [TK]D-Fender | snowboarder04: "inbound route"? Pardon? Go to CLI and show us SIP DEBUG for the failed attempt |
| 23:52.12 | [TK]D-Fender | ~pb |
| 23:52.12 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 23:52.13 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 23:58.11 | snowboarder04 | [TK]D-Fender: http://paste2.org/p/165164 <--- the call comes in at line 20, from @redsquared.com |
| 23:59.58 | [TK]D-Fender | snowboarder04: They don't seem to respond to the 407 Proxy Auth request. Could be they expect you to receive the call without challenge. add "insecure=port,invite" to your peer |