| 00:00.27 | hmmhesays | or various other places |
| 00:00.44 | *** join/#asterisk RypPn (i=TuMbL@80.177.214.249) |
| 00:03.56 | *** part/#asterisk frenzy (i=user@unaffiliated/frenzy) |
| 00:12.43 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:13.18 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-d557b6bf0cda5c84) |
| 00:16.46 | *** join/#asterisk MontStuck (n=jtknapp@70.54.226.58) |
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| 00:31.23 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
| 00:33.05 | franck | Hi, I have an extension which is busy, I do not know why, nor how can I check the status of the extension. Looking at packets, asterisk does not even try to send packets to the IP of this extension. What can I do? |
| 00:35.56 | *** join/#asterisk ccaron (n=cedric@85-218-16-115.dclient.lsne.ch) |
| 00:42.17 | *** join/#asterisk rnovotny22 (n=ro085181@c-69-180-187-19.hsd1.mn.comcast.net) |
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| 01:05.40 | *** join/#asterisk LakeSolon (n=blake@12-202-201-70.client.mchsi.com) |
| 01:09.36 | *** join/#asterisk Hoondie (n=andre@RN-2PARK-1800-01.pipenetworks.com) |
| 01:11.52 | Hoondie | hey people.. i have a bit of a problem.. i have an asterisk server that is behind NAT, ports 5060 - 5070 and 10000-20000 are port forwarded to it, at another site I have a GrandStream BT100 that is configured to use the asterisk server, it seems to register fine, i get one way audio and a if i do a sip debug i get a whole heap of retransmitting messages.. |
| 01:12.10 | Hoondie | any ideas? |
| 01:12.21 | JT | why forward 5060-5070? |
| 01:12.39 | Hoondie | i noticed somewhere something used 5070 |
| 01:12.49 | JT | imaginary ;) |
| 01:12.59 | Hoondie | lol.. maybe |
| 01:13.10 | Hoondie | it shouldn't hurt to add a few more ports though? |
| 01:13.34 | JT | it doesn't make much sense though |
| 01:13.41 | JT | and is it tcp or udp? |
| 01:14.06 | Hoondie | 10000-20000 is UDP, 5060-5070 is both |
| 01:18.18 | Hoondie | any ideas? |
| 01:23.45 | JT | 5060 is udp |
| 01:23.47 | JT | and only 5060 |
| 01:23.54 | JT | ~sipnat |
| 01:23.54 | 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 |
| 01:24.04 | JT | also, grandstream sucks |
| 01:24.08 | Hoondie | yea, i know |
| 01:24.51 | Hoondie | the thing is, it seems to be sending SIP messages back and forth, but it seems to not work for one, maybe the grandstream is not sending it back? |
| 01:26.32 | JT | it's obviously an RTP problem, not sip |
| 01:29.21 | Hoondie | hmm, weird.. i changed the 5060 - 5060 UDP/TCP to just 5060 UDP, works fine now |
| 01:30.40 | Hoondie | thanks for your help |
| 01:33.26 | JT | that is strange, but okay |
| 01:49.09 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 02:01.59 | marl | anyone tell me why the following dial command wont timeout? exten => 01415351234,n,dial(ZAP/2/1470w0123456,20,gr) |
| 02:04.57 | *** join/#asterisk tru_`z24 (n=truz_`24@74-129-166-232.dhcp.insightbb.com) |
| 02:12.57 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 02:17.55 | riddlebox | can somone tell me why when I call extension to extension I hear no ringing on either end? here is my extensions.conf http://pastebin.ca/753261 |
| 02:18.20 | *** join/#asterisk peanut- (n=tokarev@cpe-70-113-100-193.austin.res.rr.com) |
| 02:18.27 | peanut- | ~sipnat |
| 02:18.28 | 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 |
| 02:22.18 | flenders | riddlebox: what extension are you trying to reach and which context? |
| 02:22.36 | riddlebox | internal, and 525 |
| 02:23.19 | flenders | is it dialing SIP/522? |
| 02:23.24 | flenders | is SIP/522 registered? |
| 02:23.38 | flenders | what error message do you get on the CLI? |
| 02:23.41 | riddlebox | SIP/522 |
| 02:24.05 | peanut- | anyone sucessfully connect a SIP phone behind NAT to an asterisk box behind another NAT? |
| 02:24.21 | riddlebox | no errors, it just goes to voicemail |
| 02:24.45 | flenders | so SIP.522 registered? |
| 02:24.48 | flenders | SIP/522 |
| 02:24.55 | riddlebox | yeah its registered |
| 02:25.06 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 02:25.23 | riddlebox | I am calling from SIP/522, I meant I am calling SIP/525 |
| 02:25.40 | flenders | your dial command is dialing SIP.522 |
| 02:25.44 | flenders | SIP/522 |
| 02:26.42 | riddlebox | ohh crap |
| 02:26.49 | flenders | exten => 525,1,Dial(SIP/522,20) |
| 02:26.52 | riddlebox | I am a moron |
| 02:26.56 | flenders | :D |
| 02:28.00 | riddlebox | other than that, does the dialplan look ok, I am redoing it to try to update it to the newer functions |
| 02:28.17 | flenders | well, you don't need the Answer() |
| 02:28.35 | flenders | on the extensions, I mean |
| 02:29.34 | *** join/#asterisk SirThomas_Home (n=SirThoma@209.169.199.174) |
| 02:29.51 | Hoondie | hmm.. why would something work before i left for lunch.. and after lunch it's now broken? |
| 02:29.55 | Hoondie | this sucks |
| 02:29.57 | riddlebox | really I need to read the new version of the book |
| 02:30.27 | robin_sz | Hoondie: wild guess here .. but, well, something has chnaged |
| 02:30.27 | Hoondie | Is anyone else using DD-WRT? |
| 02:30.49 | Hoondie | i think it might be the router, the port forwards occationally screw up |
| 02:31.21 | flenders | riddlebox: what handsets are you using? |
| 02:31.39 | *** join/#asterisk jsaunders (n=nevermin@70.70.0.33) |
| 02:31.45 | riddlebox | Grandstream GXP2000's and a sipura 2100 |
| 02:32.09 | *** join/#asterisk L2SHO (n=adam@static-host-24-149-138-156.patmedia.net) |
| 02:32.32 | jsaunders | So I finally get my copy of Asterisk Business Edition and during installation I'm getting flippin' Python errors that are causing the install to abort. Lovely. |
| 02:32.40 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 02:32.48 | fujin_ | pwned |
| 02:32.50 | fujin_ | that's what you get |
| 02:32.55 | fujin_ | for buying a sillysoft |
| 02:33.29 | Hoondie | anyone know why i'm getting unreachable on a peer when i do sip show peers? |
| 02:33.45 | jsaunders | Sorry fujin_.... didn't realize you were such a broke ass that it's way out of your budget to help support a growing software company. Your just one of those people who wants everything for free. |
| 02:33.52 | riddlebox | flenders, another thing I am trying to fix is that when I call someone and they answer, a couple seconds after they answer I hear an extra ring |
| 02:33.58 | fujin_ | lol |
| 02:34.00 | fujin_ | no need to rage |
| 02:34.05 | jsaunders | No need to be a loser. |
| 02:34.12 | fujin_ | ring Digium support |
| 02:34.13 | fujin_ | seriously |
| 02:34.30 | jsaunders | Oh I will. Just felt like venting in a public forum. And now that that's done... I'm outtie. |
| 02:34.46 | fujin_ | What a buttsecks |
| 02:34.48 | riddlebox | what the hell |
| 02:35.10 | riddlebox | you can support digium by buying their hardware..... |
| 02:35.58 | fujin_ | meh |
| 02:36.03 | fujin_ | I'll support the devs that hang out in here |
| 02:36.05 | fujin_ | ^5 devs! |
| 02:36.30 | riddlebox | ~book |
| 02:36.31 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 02:36.56 | *** join/#asterisk linxroute (n=VietPhon@222.252.108.5) |
| 02:37.08 | *** join/#asterisk blq (n=Bl@dslb-088-064-147-021.pools.arcor-ip.net) |
| 02:37.37 | linxroute | hi there |
| 02:37.38 | *** join/#asterisk axscode (n=axscode@58.56.49.60.klj02-home.tm.net.my) |
| 02:40.08 | Hoondie | anyone know why i'm getting unreachable on a peer when i do sip show peers?? |
| 02:40.18 | *** join/#asterisk axscode (n=axscode@58.56.49.60.klj02-home.tm.net.my) |
| 02:40.43 | axscode | hi, what gcc version will asterisk needs? |
| 02:44.38 | *** join/#asterisk axscode (n=axscode@14.57.49.60.klj02-home.tm.net.my) |
| 02:48.11 | *** join/#asterisk axscode (n=axscode@14.57.49.60.klj02-home.tm.net.my) |
| 02:49.23 | axscode | hi, what gcc version will asterisk needs? sorry i got disconnected |
| 02:52.39 | JT | axscode: you got disconnected about 4 times |
| 03:00.48 | [TK]D-Fender | Hoondie, because * tried to contact the other side and failed, or because the other side never registered in the first place and * has no clue where to send calls, or if you're using "qualify=yes" and it times out. |
| 03:01.38 | Hoondie | it registers fine, but then times out.. |
| 03:01.56 | Hoondie | and it's only like 60ms round trip |
| 03:06.28 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 03:07.12 | linxroute | :) |
| 03:07.16 | linxroute | hi there |
| 03:07.28 | linxroute | i want to change incomming caller id |
| 03:07.36 | linxroute | can anyone please help |
| 03:07.48 | *** join/#asterisk axscode (n=axscode@14.57.49.60.klj02-home.tm.net.my) |
| 03:07.51 | [TK]D-Fender | Hoondie, Any NAT involved in thie path between the server & phone? |
| 03:08.03 | linxroute | oh TK |
| 03:08.06 | [TK]D-Fender | linxroute, "show function CALLERID" |
| 03:08.16 | linxroute | i used your script |
| 03:08.19 | [TK]D-Fender | linxroute, I already COMPLETELY ANSWERED THIS last time... |
| 03:08.46 | linxroute | exten => s,1,GotoIf($["${CALLERID(num):2}"!="04"]?3) |
| 03:08.57 | linxroute | sorry to bother you again |
| 03:09.11 | Hoondie | [TK]D-Fender: yea, NAT at both ends.. it worked before i went out to eat lunch.. i plugged the phone in again to test if it was still working and what do you know, it stopped working.. i was at lunch so it's not like i changed anything |
| 03:09.21 | linxroute | could you please explain for me the "?" question mark site |
| 03:09.27 | linxroute | sign |
| 03:09.30 | linxroute | is correct ? |
| 03:09.32 | [TK]D-Fender | Hoondie, Read this, NOW : |
| 03:09.35 | [TK]D-Fender | Sipnat |
| 03:09.37 | [TK]D-Fender | ~sipnat |
| 03:09.38 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 03:09.47 | Hoondie | [TK]D-Fender: i have, like 3 times |
| 03:09.58 | [TK]D-Fender | linxroute, the "?" is just a seperator ofr the GotoIf. |
| 03:10.11 | [TK]D-Fender | Hoondie, then pastebin your sip.conf masing only passwords |
| 03:10.12 | [TK]D-Fender | ~pb |
| 03:10.13 | jbot | A Pastebin is a place to paste your stuff 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 |
| 03:10.14 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^ |
| 03:10.15 | Hoondie | i think it might actually be the router |
| 03:10.20 | [TK]D-Fender | Hoondie, And we'll see what you missed |
| 03:10.32 | [TK]D-Fender | Hoondie, That is possible. What brands are they? |
| 03:10.41 | Hoondie | [TK]D-Fender: i'm going to try installing sipath.. |
| 03:10.51 | Hoondie | [TK]D-Fender: it's a WRT54G with DD-WRT on it.. |
| 03:11.01 | [TK]D-Fender | Hoondie, And the other side? |
| 03:11.14 | Hoondie | The other end is a cisco 1800 |
| 03:11.35 | [TK]D-Fender | Hoondie, make sure it isn't doing ANY sip transform. |
| 03:12.07 | Hoondie | it worked about an hour ago.. made calls in both directions and it worked.. |
| 03:12.27 | [TK]D-Fender | Hoondie, Well pastebin your sip.conf as I asked and we'll see if you missed something. |
| 03:12.29 | Hoondie | I'll give SipAtH a go on the DD-WRT, it's a sip/rtp proxy, that *should* fix it |
| 03:13.11 | [TK]D-Fender | Hoondie, Avoid... |
| 03:13.18 | [TK]D-Fender | Hoondie, Try to fix this within * first. |
| 03:13.27 | [TK]D-Fender | Hoondie, PB up your configs and we'll see there first |
| 03:13.37 | *** join/#asterisk mitcheloc (n=mitchel@ppp-67-126-240-11.dsl.irvnca.pacbell.net) |
| 03:14.33 | *** join/#asterisk axscode (n=axscode@14.57.49.60.klj02-home.tm.net.my) |
| 03:14.35 | Hoondie | [TK]D-Fender: http://pastebin.ca/753303 |
| 03:16.24 | linxroute | exten => s,1,GotoIf($["${CALLERID(num):2}"!="04"]?3) |
| 03:16.24 | linxroute | exten => s,2,Set(CALLERID(num)=${CALLERID(num):2}) |
| 03:16.24 | linxroute | exten => s,3,Set(TIMEOUT(respose)=4) |
| 03:16.24 | linxroute | exten => s,4,Background(introen) |
| 03:16.24 | linxroute | exten => i,1,Playback(pbx-invalid) |
| 03:16.24 | linxroute | exten => i,2,Goto(incomming,s,1) |
| 03:16.26 | linxroute | exten => t,1,Dial(SCCP/4005&SIP/4000,30,Ww) |
| 03:16.28 | linxroute | exten => t,2,Hangup |
| 03:16.31 | [TK]D-Fender | Hoondie, Ok, that looks 100% fine if your DNS resolves properly. Chech that.... then I'd first suspect the CISCO as being at fault before the Linksys. |
| 03:16.33 | linxroute | here's script |
| 03:16.38 | linxroute | my incomming |
| 03:16.42 | [TK]D-Fender | linxroute, do NOT spam like that in here again. |
| 03:16.50 | Hoondie | DNS works fine, i use it to log into the box via ssh all the time |
| 03:16.56 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:17.15 | linxroute | sorry |
| 03:17.17 | Hoondie | [TK]D-Fender: getting the same error on another external phone that's behind a WRT54G |
| 03:17.24 | Hoondie | with DD-WRT on it again |
| 03:17.24 | [TK]D-Fender | Hoondie, Ok, check the Cisco side.. they can be NASTY where NAT is concerned. I unfortunately can't give you any of the specifics though. |
| 03:17.32 | linxroute | hic |
| 03:17.35 | [TK]D-Fender | Hoondie, hrm |
| 03:18.05 | [TK]D-Fender | linxroute, So, what about it? |
| 03:18.12 | Hoondie | [TK]D-Fender: the cisco end did work before, i went to lunch and not working again :( |
| 03:18.31 | linxroute | sorry for bother you but still when there's imcomming call |
| 03:18.45 | linxroute | the box does not remove |
| 03:18.50 | linxroute | the are code |
| 03:19.38 | [TK]D-Fender | linxroute, PASTEBIN the CLI output of the failed call and do "NoOp(CallerID is -${CALLERID(num)}-) as "s,1" and bump up the jump in GotoIf. |
| 03:19.53 | [TK]D-Fender | linxroute, And then again rigth before you do yoru menu |
| 03:20.07 | Hoondie | [TK]D-Fender: check out this page, search the body for dd-wrt : http://www.voip-info.org/wiki/view/NAT+and+VOIP |
| 03:20.23 | Hoondie | [TK]D-Fender: do you think that would have anything to do with it? |
| 03:20.44 | Hoondie | i have sp2, said it's not tested.. |
| 03:20.47 | [TK]D-Fender | Hoondie, thats a huge page :) |
| 03:21.05 | Hoondie | the last bullet point in the workarounds section |
| 03:21.08 | *** join/#asterisk axscode (n=axscode@14.57.49.60.klj02-home.tm.net.my) |
| 03:21.25 | [TK]D-Fender | 'different approaches for making sip devices work behind a dd-wrt router' |
| 03:21.26 | [TK]D-Fender | <PROTECTED> |
| 03:21.27 | [TK]D-Fender | Ah |
| 03:21.34 | [TK]D-Fender | well... they recommend it :) |
| 03:21.45 | [TK]D-Fender | Hoondie, Guess its SOMETHING to try, right? :) |
| 03:21.57 | [TK]D-Fender | Hoondie, Trust those with experience... |
| 03:22.14 | Hoondie | yea, i have sp2, dun know if i want to downgrade.. |
| 03:22.32 | Hoondie | it's weird... sometimes i'll reboot the router and something will change.. |
| 03:23.00 | [TK]D-Fender | Hoondie, You already have a full linux server... let go of letting a silly Linksys get delusions of grandeur :p |
| 03:23.40 | *** join/#asterisk antlers (n=antlers@ip70-173-89-173.lv.lv.cox.net) |
| 03:24.59 | *** join/#asterisk axscode (n=axscode@14.57.49.60.klj02-home.tm.net.my) |
| 03:26.02 | Hoondie | [TK]D-Fender: yea, i should just use it as an access point.. i have a cisco 800 series just acting as a DSL brigde that would probably the the job a bit better than the WRT |
| 03:26.04 | linxroute | TK Fender, thanks for you help |
| 03:26.12 | linxroute | http://pastebin.com/m1b725d15 here's my paste bin |
| 03:26.24 | linxroute | it does not show up any error |
| 03:26.55 | [TK]D-Fender | linxroute, You didn't add the NoOp's I told you to add. |
| 03:27.35 | *** join/#asterisk axscode (n=axscode@14.57.49.60.klj02-home.tm.net.my) |
| 03:28.29 | linxroute | yeapppppppppppp |
| 03:28.46 | linxroute | thanks to you again |
| 03:28.56 | linxroute | really sorry to bother you |
| 03:29.04 | linxroute | it's work now |
| 03:29.22 | [TK]D-Fender | ... |
| 03:29.27 | linxroute | you are really asterisk expert |
| 03:30.17 | linxroute | how come you know so much |
| 03:30.20 | *** join/#asterisk axscode (n=axscode@14.57.49.60.klj02-home.tm.net.my) |
| 03:30.23 | linxroute | gosh |
| 03:30.56 | linxroute | thanks alot |
| 03:31.01 | [TK]D-Fender | linxroute, np |
| 03:31.08 | [TK]D-Fender | linxroute, I've jsut done this for a while |
| 03:32.15 | flenders | Hoondie: sorry to jump in... I never used dd-wrt voip firmware, and always had an asterisk server running behind my router |
| 03:32.38 | flenders | and also have phones behind other routers that work just fine |
| 03:32.54 | Hoondie | you using dd-wrt? |
| 03:32.58 | flenders | yeah |
| 03:33.01 | Hoondie | hmm |
| 03:33.15 | [TK]D-Fender | Ok, you two run with that a bit :) |
| 03:33.26 | Hoondie | what version you running now? |
| 03:33.30 | linxroute | one site is linksys and another site with cisco ? |
| 03:33.38 | linxroute | 1800 hoondie ? |
| 03:33.41 | Hoondie | yea |
| 03:33.48 | linxroute | adsl module ? |
| 03:33.51 | flenders | all I ever did on my router was forward ports TCP/5060, UDP/5060 and UDP/10000-20000 to my asterisk box |
| 03:33.53 | Hoondie | nope |
| 03:33.58 | linxroute | bri ? |
| 03:34.10 | Hoondie | Do i need TCP 5060? |
| 03:34.26 | linxroute | asterisk is udp |
| 03:34.27 | flenders | Hoondie: DD-WRT v23 SP2 vpn |
| 03:34.39 | Hoondie | linxroute: ethernet at the moment |
| 03:35.24 | linxroute | ?? |
| 03:35.28 | linxroute | remote site with ethernet |
| 03:36.12 | Hoondie | the router is in one of our datacenters |
| 03:36.20 | Hoondie | it's connecting to a switch |
| 03:36.43 | Hoondie | that goes to a 7100, that is connected to one of our upstream providers |
| 03:37.44 | tzanger | in sip show peer <foo>, what's defaddr->ip? what's that map to in sip.conf? |
| 03:38.04 | linxroute | and the sip server is behind the linksys router ? |
| 03:38.08 | *** join/#asterisk axscode (n=axscode@14.57.49.60.klj02-home.tm.net.my) |
| 03:38.23 | linxroute | the phone does not register ? or register but with one way voice ? |
| 03:38.29 | Hoondie | well, i'm at work now.. testing from work.. the * server is at home |
| 03:38.48 | Hoondie | it registers, it's marked as unreachable if i do sip show peers |
| 03:38.56 | Hoondie | i don't know if i have two way audio at the moment |
| 03:38.58 | linxroute | sure |
| 03:39.17 | linxroute | so you are under two NAT |
| 03:39.24 | Hoondie | yea |
| 03:40.34 | linxroute | did you set in the sip.conf |
| 03:40.39 | linxroute | nat=yes |
| 03:40.42 | JT | you do not need TCP 5060 |
| 03:40.43 | linxroute | for the phone ? |
| 03:40.52 | [TK]D-Fender | tzanger, defaultip = Dotted.Quad.IP.Addr : Default IP address of client host= is specified as DYNAMIC. Used if client has not been registered at any other IP address. Valid only for type=peer. |
| 03:41.00 | Hoondie | JT: it worked then stopped again :( |
| 03:41.05 | Hoondie | yea, nat=yes |
| 03:41.14 | tzanger | [TK]D-Fender: hmm |
| 03:41.24 | linxroute | one moment |
| 03:42.05 | [TK]D-Fender | linxroute, He set everything right for *, I already checked. |
| 03:43.15 | linxroute | i had the same problem before with my cisco 7940 connect to my home * box |
| 03:43.57 | linxroute | but when i added the "externhost=x.x.x.x/mask |
| 03:44.02 | linxroute | sorry |
| 03:44.09 | Hoondie | got that in there already |
| 03:44.12 | linxroute | externip |
| 03:44.29 | linxroute | it's work okay |
| 03:45.37 | linxroute | hoondie |
| 03:45.42 | linxroute | what's your phone ? |
| 03:46.04 | Hoondie | Grandstream BT100 |
| 03:46.07 | Hoondie | crappy phone |
| 03:46.26 | tzanger | this makes no fucking sense |
| 03:46.31 | linxroute | hard phone might get problems |
| 03:46.35 | linxroute | have you try like |
| 03:46.38 | linxroute | XLite ? |
| 03:46.44 | tzanger | [TK]D-Fender: * box a and b. neither changed in months. I upgraded a but did NOT change any configs |
| 03:46.45 | Hoondie | XLite crashes on this system |
| 03:46.59 | tzanger | now B rejects calls from A (it isn't matching its peer on B now) |
| 03:47.06 | tzanger | IPs are static, no nat... |
| 03:48.55 | [TK]D-Fender | tzanger, IMO, defaultip = cop-out. let things register, or set a fixed host. Anything else = half-assed |
| 03:49.09 | tzanger | it is fixed host |
| 03:49.09 | linxroute | with a softphone |
| 03:49.11 | tzanger | acl=yes |
| 03:49.21 | linxroute | we can get more debug informations |
| 03:49.23 | tzanger | I just tried defaultip as a last chance attempt to figure out wtf changed |
| 03:50.21 | tzanger | it's not matching the peer entry it matched before |
| 03:52.42 | *** join/#asterisk bmg505 (n=leon@196.209.183.44) |
| 03:53.05 | *** join/#asterisk polerin (n=erin@c-71-228-222-87.hsd1.tn.comcast.net) |
| 03:53.07 | tzanger | deny=0.0.0.0/0, permit=a.b.c.d/32 |
| 03:57.13 | polerin | feh. anyone have a good tutorial for setting up incoming/outgoing fax stuff? Searched but 1/2 of the stuff is just forum posts or brief mentions of asterfax |
| 03:57.34 | denon | polerin: sure, that's easy. Buy an AS5400. |
| 03:57.35 | linxroute | have anyone here used PIKA cards before ? |
| 03:57.42 | denon | 1-step tutorial |
| 03:57.59 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 03:58.04 | linxroute | PIKA cards have fax port |
| 03:58.10 | linxroute | with on board DSP |
| 03:58.31 | linxroute | with very good result, well that's what they said |
| 03:59.02 | denon | shrugs, Ive seen millions of faxes go through as5300s and as5400s with practically zero issues |
| 03:59.08 | denon | hard for me to argue with that |
| 04:00.08 | polerin | let me rephrase. Does anyone have a link to a tutorial on setting up asterisk with SpanDSP/what have you. I've seen one or two, but most of them are a bit dated. I don't have the resources to go buy cards, so even a slightly shakey solution would be good right now. |
| 04:00.29 | denon | polerin: you won't be happy with it |
| 04:00.38 | denon | it's not slightly shakey, it's slightly functional |
| 04:00.42 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-69-66.dsl.irvnca.pacbell.net) |
| 04:01.01 | denon | I'm not just being a technical biggot, I'm trying to save you days of frustration |
| 04:01.30 | polerin | I understand and got the point. doesn't help i'm trying to do it over sip through broadvoice either. |
| 04:01.34 | linxroute | sorry denon, if i may ask you AS5300 use with asterisk ? |
| 04:01.51 | denon | nope |
| 04:01.56 | denon | but you could pass off a PRI to it from * |
| 04:02.25 | polerin | more of a "I have no choice" kinda thing, if there is any kind of possiblity of it even working in the slightet |
| 04:02.35 | polerin | slightest.. feh |
| 04:03.16 | franck | How do you reset dynamically Hints? |
| 04:03.23 | polerin | denon: no tutorial == sad polerin, but I'll live ;p |
| 04:04.35 | *** join/#asterisk ahattar (n=ahaha@pool-71-172-246-220.nwrknj.fios.verizon.net) |
| 04:05.15 | denon | someday you'll thank me |
| 04:06.19 | polerin | denon: try "if I don't get faxing working my partner may loose a client and I may loose my house." frustration is affordable at this point. |
| 04:06.44 | linxroute | polerin |
| 04:06.50 | denon | polerin: so having a fax solution that works on maybe 1 out of 10 faxes is better .. how? |
| 04:06.55 | linxroute | i tried all possible way |
| 04:07.03 | linxroute | but like denon said |
| 04:07.05 | denon | if you can't provide a good solution in-house, sub it out, or use other equipment |
| 04:07.09 | linxroute | it's very crappy result |
| 04:07.14 | polerin | if it meens I have to go with some crappy web fax service, fine, but I don't like the idea of HIPAA sensitive material hitting someone elses servers |
| 04:07.50 | linxroute | no real dsp service |
| 04:07.54 | linxroute | very bad result |
| 04:07.57 | polerin | (yeah I got that. crappy result, 1 out of 10 faxes. check. Knew that before I joined :P) |
| 04:07.57 | denon | clients get extremely frustrated when faxes just disappear |
| 04:08.06 | denon | or when they get an error, and have to re-send over and over |
| 04:08.28 | polerin | I'm aware. I'm human too :P |
| 04:08.28 | denon | we've all been down this road |
| 04:08.28 | linxroute | PIKA card is affordable |
| 04:08.30 | denon | we've all thought it'd be cool if * could natively do it |
| 04:08.35 | linxroute | with good result for faxing |
| 04:08.36 | denon | but at this point, it's just not there yet |
| 04:08.43 | linxroute | well that's what they said |
| 04:08.51 | linxroute | and asterisk compatible too |
| 04:08.55 | polerin | linux: pika card is $50 bucks? and do you ahve $50? |
| 04:08.59 | denon | you can get as5300s pretty cheap on fleabay |
| 04:09.01 | polerin | because I don't :P |
| 04:09.42 | tzanger | yes as5300s are pretty decent |
| 04:09.47 | tzanger | at least as far as I am concerned |
| 04:09.57 | tzanger | I helped build a dialup empire with as5248s and now maxtnts |
| 04:10.13 | polerin | define cheap, because what i'm seeing is like 1,699 |
| 04:10.26 | denon | Cisco AS5300 8 E1 240 Modems 128MB DRAM 2x AC PS 5300 |
| 04:10.30 | denon | 2 grand |
| 04:10.37 | tzanger | polerin: compared to what I think is the part you're missing |
| 04:10.38 | denon | that's a lotta simultaneous faxes for 2k |
| 04:10.41 | polerin | denon: if I had two grand I'd pay my house payment. |
| 04:10.59 | denon | polerin: perhaps you should consider another line of work, if you can't invest in your business properly |
| 04:11.17 | denon | or simply sub it out to someone with the proper equipment |
| 04:11.40 | denon | you could do so seamlessly, then get it back once you have a proper solution in place |
| 04:11.42 | polerin | denon: that would be nice, but for a small business that's running on $3k of equipment total, 2k for a fax machine is way WAY out of spec |
| 04:11.57 | denon | right, then providing fax service is beyond the scope of your business |
| 04:12.51 | denon | anyway, if you must do soft faxes, there are docs out there .. Ive seen em, but I dont know where they are .. a few googles should get you there |
| 04:12.51 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-20-238.lns10.syd7.internode.on.net) |
| 04:13.15 | polerin | i'm looking for a slightly glorified fax machine denon. |
| 04:13.16 | tzanger | I've done spandsp with asterisk. it isn't stable |
| 04:13.29 | tzanger | your best bet is iaxmodem + hylafax |
| 04:13.32 | denon | tzanger: that's what we've all been trying to explain |
| 04:13.49 | denon | polerin: I understand .. so run hylafax with an analog modem, and pipe it to email |
| 04:13.49 | tzanger | sometimes people need to do it the hard way. I'm generally one of those people |
| 04:13.57 | polerin | no analog line. |
| 04:14.05 | denon | have asterisk give you an analog line |
| 04:14.11 | tzanger | polerin: use an ATA then |
| 04:14.13 | denon | oh, you're getting it via sip? |
| 04:14.17 | tzanger | you'll get T38 for under $100 |
| 04:14.18 | polerin | ding. |
| 04:14.22 | denon | your upstream won't do t.38 properly anyway |
| 04:14.26 | denon | if they say they will, they're lying |
| 04:14.35 | polerin | and no hard output as of this second. |
| 04:14.56 | denon | tzanger: we're told the budget for this project is less than 1 dollar |
| 04:14.59 | polerin | feh. I needed to give it a try before going to a hardline with a per min charge or something. |
| 04:15.02 | polerin | lol |
| 04:15.21 | polerin | that is the aim, reality.. well.. you get what you pay for. |
| 04:15.26 | tzanger | polerin: you will *not* have success with spandsp (or ANY soft fax without t38) and VOIP. Period. Full-stop. |
| 04:15.29 | polerin | trust me, I understand that concept very well. |
| 04:15.46 | Corydon76-dig | polerin: go with efax or another online provider |
| 04:16.02 | Corydon76-dig | unless you're going to get a hardline for faxes |
| 04:16.15 | tzanger | that's just the reality of things |
| 04:16.22 | polerin | I prolly will end up having to :/ |
| 04:16.27 | tzanger | I've heard that callweaver can do proper t38 faxing, but I've not tested it myself |
| 04:16.58 | polerin | Corydon76 knows I'm a bit hard headed and need to bang it out myself :P |
| 04:16.58 | alrs | tzanger: I was going to test that, but discovered instead that Callweaver is an effective OpenVZ DOS tool |
| 04:17.21 | Corydon76-dig | polerin: how's business, btw? |
| 04:17.27 | tzanger | alrs: you're trying ot run a realtime application on virtualized hardware? |
| 04:17.40 | tzanger | has everyone been taking crazy pills? |
| 04:17.45 | denon | uh huh. |
| 04:17.58 | denon | it's sunday night, apparently the crazy pills get delivered slightly before monday |
| 04:17.58 | alrs | tzanger: I run my personal Asterisk server on an OpenVZ VPS. Works. |
| 04:18.09 | polerin | each client is paying more than anticipated, but therapists are hard to get to actually listen. all three of her clients right now are absolutly blown away with the actuall service ;) |
| 04:18.12 | tzanger | alrs: ah okay |
| 04:18.16 | denon | I guess it's so they're in full effect monday morning |
| 04:18.39 | tzanger | home server for experimentation and testing the limits of the wife acceptance ratio. Been there, doing that. :-) |
| 04:18.40 | Corydon76-dig | Good... that's what an investor likes to hear |
| 04:18.50 | polerin | denon: I don't take my crazy pills, the giant purple elephant told me they were poison. AND HE WAS RIGHT |
| 04:21.04 | polerin | wow apparently THAT killed the channel again :P |
| 04:21.11 | Corydon76-dig | As all of the men in this channel know, the crazy pills are estrogen... |
| 04:21.19 | polerin | phahah |
| 04:21.29 | polerin | don't get me started on 'crazy' corydon. |
| 04:21.53 | polerin | tell you some stories from this past couple months :P |
| 04:22.11 | Corydon76-dig | Yes, I'm the only one crazy enough to do the accounting for a small charity... |
| 04:22.29 | *** join/#asterisk axscode (n=axscode@14.57.49.60.klj02-home.tm.net.my) |
| 04:22.47 | tzanger | hah |
| 04:23.01 | Corydon76-dig | 'sokay, I'm about to start doing the accounting for a small chapter of a trade union |
| 04:23.18 | JT | charity, haha |
| 04:23.26 | JT | as if trade unions are charities |
| 04:23.36 | Corydon76-dig | JT: two different sets of books |
| 04:23.47 | polerin | see, estrogen has Nothing to do with the crazy. |
| 04:23.48 | Corydon76-dig | The charity is a real 501(c)(3) |
| 04:24.00 | tzanger | polerin: who said it did? |
| 04:24.14 | Corydon76-dig | trade union is completely separate |
| 04:24.28 | polerin | tzanger: corydon did :P |
| 04:24.48 | Corydon76-dig | polerin: I never said anything about lacking estrogen |
| 04:25.01 | tzanger | haha |
| 04:25.40 | Corydon76-dig | polerin: granted, I have less estrogen than you... |
| 04:25.50 | denon | wow.. |
| 04:25.56 | denon | I leave the channel window for 10 seconds. . |
| 04:25.59 | denon | and this is what happens |
| 04:26.08 | Corydon76-dig | What happens? |
| 04:26.13 | denon | that giant purple elephant is looking normal now |
| 04:26.19 | denon | estrogen wars |
| 04:27.03 | *** join/#asterisk axscode (n=axscode@14.57.49.60.klj02-home.tm.net.my) |
| 04:27.39 | polerin | denon: no, he has way less estrogen than I. trust me. |
| 04:27.42 | polerin | anyway |
| 04:28.37 | *** join/#asterisk axscode (n=axscode@14.57.49.60.klj02-home.tm.net.my) |
| 04:28.54 | Corydon76-dig | Yeah, I have no tits... |
| 04:29.33 | Corydon76-dig | You have to admit, though... you have a nice rack... |
| 04:30.58 | polerin | :P |
| 04:31.00 | denon | yeah, the HP racks are pretty nice |
| 04:31.05 | denon | the APC ones feel a little cheap |
| 04:31.20 | denon | and the telco frames aren't worth using at all |
| 04:31.57 | Corydon76-dig | denon: I dunno, I kinda like the shelves of a telco rack |
| 04:32.40 | *** join/#asterisk axscode (n=axscode@14.57.49.60.klj02-home.tm.net.my) |
| 04:32.58 | JT | how do apc racks feel cheap? |
| 04:33.57 | MaliutaWrk | can't say I've seen an apc rack |
| 04:34.41 | denon | JT: they cost less than HP ones. so they feel cheaper. ;) |
| 04:35.10 | Corydon76-dig | My rack was free for the taking |
| 04:35.19 | denon | MaliutaWrk: you have now: http://www.apc.com/products/family/index.cfm?id=301 |
| 04:35.20 | Corydon76-dig | Just had to go pick it up |
| 04:36.03 | JT | apc 1070mm racks are pretty nice |
| 04:36.14 | polerin | meph |
| 04:36.16 | denon | Corydon76-dig: yeah, friend of mine got a *really* nice rack that way. someone needed to clean the garage |
| 04:36.25 | polerin | i'm taking my ass to bed. |
| 04:36.38 | denon | like a $5k rack |
| 04:36.48 | polerin | work in the morning and I have to get up early to figure out what to do with this craptastic haircut I got today |
| 04:36.59 | denon | return it |
| 04:37.04 | denon | hope you kept the receipt |
| 04:37.08 | polerin | yeah that's a great idea. |
| 04:37.20 | denon | if they don't take it back, just charge it back on your CC |
| 04:37.26 | denon | that's what everyone does these days, stupid consumers |
| 04:37.31 | polerin | :P cute... anyway |
| 04:37.33 | polerin | sleeps |
| 04:37.46 | Corydon76-dig | What's wrong with the haircut? |
| 04:37.56 | Corydon76-dig | Oh, and BTW, we all missed you at PN |
| 04:38.11 | tzanger | looks like the sip nat handling in svn trunk has changed enough to break things between today and 8000 revs ago |
| 04:38.13 | tzanger | :-) |
| 04:38.34 | denon | talk to file |
| 04:38.36 | tzanger | nat=no isn't good enough anymore for an asterisk box sitting directly on the 'net |
| 04:38.38 | denon | he's always to blame |
| 04:38.47 | tzanger | nat=never isn't fucking up the rport now |
| 04:38.55 | tzanger | but the far end isn't responding right just yet |
| 04:38.59 | tzanger | but at least it's not failing |
| 04:41.12 | polerin | Corydon76-dig: yeah... I decided I didn't need the stress at that point, plus it was cheeper to go help a comic friend |
| 04:41.45 | polerin | and the hair cut? it's like half way between a bob and a mullet. which is really bizzare. short bangs don't do well on my face I've discovered |
| 04:41.55 | polerin | s/cheeper/cheaper |
| 04:42.34 | Corydon76-dig | polerin: why stress? |
| 04:43.20 | Corydon76-dig | PN this year you would have just been an attendee |
| 04:44.05 | polerin | "hey this customer's mousepad is frozen.. push the modem for me!" "hey this screen has a lot of emphasis push my modem for me" "hey I can't find my ass with both hands, even if someone else IS holding the flashlight for me.. push my modem?" |
| 04:44.33 | polerin | Corydon76-dig: since when have I ever been able to just attend anything? (push my modem for me?) |
| 04:45.19 | Corydon76-dig | Heh |
| 04:45.39 | Corydon76-dig | Some things always go wrong every year |
| 04:46.03 | Corydon76-dig | even this year |
| 04:46.23 | polerin | no comment :) |
| 04:46.26 | Corydon76-dig | but the video recordings were highly redundant |
| 04:46.49 | polerin | good. bout fucking time. You realize why they wern't last year right? |
| 04:47.12 | Corydon76-dig | Weren't redundant? |
| 04:47.16 | polerin | yeah. |
| 04:47.28 | Corydon76-dig | Because we hadn't had a catastrophic failure at that point |
| 04:47.37 | polerin | phsst. we had one the year before. |
| 04:47.55 | Corydon76-dig | not one where we lost all the videos from one day |
| 04:48.02 | polerin | I was told that equipment would be brought. equipment was NOT brought. by 2 different people :P |
| 04:48.10 | polerin | and yeah, it was :) |
| 04:48.18 | polerin | or most of one day |
| 04:48.19 | polerin | anyway |
| 04:48.20 | polerin | bed |
| 04:48.27 | Corydon76-dig | Well, AV was responsible for 2 different recording media this year |
| 04:48.31 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:48.44 | Corydon76-dig | and at least 2 other recordings were made, that we know of |
| 04:49.06 | Corydon76-dig | So yes, massively redundant |
| 04:49.27 | Corydon76-dig | 'night... Me too, I'm in HSV in the am |
| 04:55.58 | *** join/#asterisk Ng (n=cmsj@mairukipa.tenshu.net) |
| 04:56.06 | Ng | any Junction Networks users in the house? |
| 04:56.29 | Ng | if so, is their IAX termination service broken at the moment? |
| 04:59.23 | *** part/#asterisk franck (n=franck@tikiwiki/franck) |
| 05:00.24 | *** join/#asterisk mikex- (n=none@c-98-200-81-179.hsd1.tx.comcast.net) |
| 05:01.17 | *** join/#asterisk MoonlightTaxi (i=CFNY332@c-68-45-147-92.hsd1.pa.comcast.net) |
| 05:02.02 | MoonlightTaxi | need help configuring BYOD trunk. |
| 05:02.29 | MoonlightTaxi | Provider only provides proxy,username and password. |
| 05:43.12 | *** join/#asterisk J_5 (n=J_5@cpe-71-72-210-44.cinci.res.rr.com) |
| 05:47.22 | *** join/#asterisk Flauto (n=zhao@71.194.141.225) |
| 05:47.28 | Flauto | [Oct 29 00:46:12] WARNING[2814]: chan_sip.c:1938 retrans_pkt: Maximum retries exceeded on transmission 581b6de0-4f84-dc11-9a60-000347bf9f6e@encorenetwork for seqno 1 (Critical Response) |
| 05:47.32 | Flauto | what is this for |
| 05:47.43 | Flauto | i tried to config ipkall to work with my asterisk |
| 05:47.47 | Flauto | this is what i got |
| 05:48.22 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 05:48.27 | L|NUX | hello every one |
| 05:51.59 | Flauto | hello |
| 05:52.12 | Flauto | [Oct 29 00:46:12] WARNING[2814]: chan_sip.c:1938 retrans_pkt: Maximum retries exceeded on transmission 581b6de0-4f84-dc11-9a60-000347bf9f6e@encorenetwork for seqno 1 (Critical Response) |
| 05:52.12 | Flauto | <PROTECTED> |
| 05:52.12 | Flauto | <PROTECTED> |
| 05:52.12 | Flauto | <PROTECTED> |
| 05:53.29 | *** join/#asterisk craigk (n=ckowald@58.174.122.198) |
| 05:56.06 | L|NUX | Flauto: nating |
| 06:01.22 | Flauto | oh |
| 06:04.54 | *** join/#asterisk blq (i=Bl@dslb-088-065-171-128.pools.arcor-ip.net) |
| 06:08.30 | *** join/#asterisk katsuodo (n=musashi@ool-45776f4f.dyn.optonline.net) |
| 06:08.43 | katsuodo | hello |
| 06:10.40 | katsuodo | FXS SIGNALLING IS NOT PASSING TO FXO CHANNEL RESULT ANALOG PHONE DOES NOT RING FOR INBOUND CALLS Any suggestions? |
| 06:10.51 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 06:11.02 | jql | perhaps the GAIN is too high? |
| 06:11.21 | katsuodo | one moment let me check |
| 06:12.04 | katsuodo | rxgain=0.0 txgain=0.0 |
| 06:13.05 | katsuodo | jql: the rx/tx gain are set to zero |
| 06:14.08 | linxroute | ? |
| 06:14.19 | linxroute | what's your card katsoudo |
| 06:14.50 | katsuodo | tdm400p (1) FXO and (1) FXS |
| 06:14.56 | katsuodo | asterisk |
| 06:15.13 | linxroute | have you connected the power cable |
| 06:15.18 | linxroute | for the FXS ? |
| 06:15.24 | JT | katsuodo: PERHAPS HE WAS TALKING ABOUT THIS SORT OF GAIN |
| 06:15.31 | katsuodo | yes reseat twice |
| 06:15.54 | linxroute | when you take the phone offhook |
| 06:16.00 | katsuodo | Jt not follow |
| 06:16.04 | linxroute | do you get any dialtone |
| 06:16.08 | katsuodo | yes |
| 06:16.30 | katsuodo | I am able to make outbound call |
| 06:16.47 | linxroute | but inbound call |
| 06:16.54 | linxroute | it's does not ring ? |
| 06:17.05 | katsuodo | not ring |
| 06:17.12 | JT | katsuodo: ALL CAPS |
| 06:17.12 | *** join/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 06:17.26 | linxroute | i think you have miss config in the extensions |
| 06:18.24 | katsuodo | the actual exten => instruction |
| 06:18.47 | katsuodo | JT understand |
| 06:21.35 | katsuodo | exten=> s,1,Answer() |
| 06:21.36 | katsuodo | exten=> s,2,Background(enter-ext-of-person) exten=> 1238,1,Dial(${CHU},20) exten=> 1238,2,Playback(vm-nobodyavail) exten=> 1238,3,Hangup |
| 06:22.50 | katsuodo | This for incoming under context default as listed in zapata.conf |
| 06:23.03 | katsuodo | linxroute no difficult |
| 06:23.41 | katsuodo | suggestions? |
| 06:26.19 | linxroute | can you pastebin me |
| 06:26.26 | linxroute | the CLI of the call ? |
| 06:28.35 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 06:29.03 | katsuodo | yes one moment |
| 06:32.54 | Chris-NB | hi |
| 06:33.05 | Chris-NB | anyone using a thomson ST2030S phone? |
| 06:36.06 | katsuodo | linxroute posted log from CLI on pastebin |
| 06:36.41 | linxroute | send me the link |
| 06:36.43 | linxroute | please |
| 06:37.16 | katsuodo | the link |
| 06:37.21 | katsuodo | ? |
| 06:38.01 | linxroute | yes |
| 06:38.04 | katsuodo | http://pastebin.com/m1f6f2d87 |
| 06:38.57 | *** join/#asterisk shtoom (n=godson@59.93.120.93) |
| 06:39.19 | linxroute | oh |
| 06:40.05 | katsuodo | yes? |
| 06:40.42 | linxroute | <PROTECTED> |
| 06:40.46 | linxroute | first |
| 06:40.51 | linxroute | you dont have that file |
| 06:41.10 | katsuodo | yes |
| 06:41.27 | linxroute | i mean like |
| 06:41.32 | linxroute | hold on |
| 06:41.46 | linxroute | in the zapata.conf |
| 06:41.56 | linxroute | what's your default context ? |
| 06:42.27 | katsuodo | should I list at pastebin |
| 06:43.08 | linxroute | the best way is you send me the configuration of your, extensions.conf sip.conf and zapata.conf |
| 06:43.15 | linxroute | so i can edit it for you |
| 06:43.26 | katsuodo | understood |
| 06:44.08 | katsuodo | how to send? |
| 06:44.24 | linxroute | open it |
| 06:44.42 | linxroute | and copy and paste it to paste bin |
| 06:44.52 | katsuodo | okay |
| 06:44.57 | katsuodo | one moment |
| 06:45.36 | linxroute | are you japanese ? |
| 06:49.26 | katsuodo | http://pastebin.com/m4e303acc |
| 06:50.03 | katsuodo | no change to sip only analog phone |
| 06:51.38 | katsuodo | linxroute list link |
| 06:51.46 | *** join/#asterisk bantu (n=Miranda@rz-du-mbx-136-213.rz.uni-karlsruhe.de) |
| 06:55.06 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:55.10 | linxroute | zap4 is fxs ? |
| 06:55.47 | linxroute | ok |
| 06:58.37 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:01.13 | linxroute | http://pastebin.com/m711a6a8a |
| 07:01.15 | linxroute | try this |
| 07:01.20 | linxroute | it's extensions.conf |
| 07:02.43 | katsuodo | one moment |
| 07:09.56 | katsuodo | linuxroute made change to server remotely someone in office answer phone which mean it rings. What is difference? |
| 07:11.33 | linxroute | you have miss config it |
| 07:12.09 | linxroute | take your time and reading about asterisk |
| 07:12.14 | linxroute | and extensions configuration |
| 07:12.22 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 07:13.04 | katsuodo | linuxroute thank you |
| 07:13.51 | linxroute | no problem |
| 07:13.55 | linxroute | you r welcome |
| 07:17.06 | katsuodo | linuxroute besides OReiley book suggest other reading about asterisk and extensions.conf |
| 07:17.23 | linxroute | well |
| 07:17.31 | linxroute | there's alot of asterisk book |
| 07:17.38 | linxroute | you can find it on emule |
| 07:18.10 | katsuodo | what this emule |
| 07:18.22 | linxroute | it's a peer to peer program |
| 07:18.33 | linxroute | allows you to download and exchange files |
| 07:18.47 | linxroute | not sure if it's legal in your country |
| 07:19.01 | katsuodo | will check |
| 07:19.56 | katsuodo | this is very serious file share program no |
| 07:21.07 | linxroute | yes it is |
| 07:21.27 | linxroute | or you can find alot of infor about asterisk |
| 07:21.38 | linxroute | with " asterisk turtorial" on google |
| 07:22.31 | katsuodo | linuxroute you have most helpful |
| 07:23.15 | linxroute | u r welcome |
| 07:23.45 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 07:26.47 | *** join/#asterisk socken23 (n=socken@ip-213-189-154-029.fix.magnet.ch) |
| 07:27.17 | *** join/#asterisk jarod14 (n=jarod14@ns2.viatelecom.com) |
| 07:28.02 | katsuodo | linuxroute must leave it is 4:30 PM here |
| 07:36.34 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 07:41.12 | *** join/#asterisk jarod14 (n=jarod14@ns2.viatelecom.com) |
| 07:48.26 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 07:50.36 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 07:51.44 | *** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com) |
| 07:54.02 | Daviey | Hey.. looking for a way to get 4 pstn lines into *... a Bank or a PCI card i don't know of? |
| 07:56.05 | *** join/#asterisk TrentCreek (i=GeekBoy@cpe-70-117-207-168.rgv.res.rr.com) |
| 07:57.37 | TrentCreek | wake up |
| 08:01.14 | *** join/#asterisk saftsack (n=saftsack@pD9E06356.dip.t-dialin.net) |
| 08:12.38 | *** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 08:16.48 | *** join/#asterisk FlatFoot (n=bigflatf@80.88.192.113) |
| 08:23.50 | *** join/#asterisk ronr (n=ron@ip51cdd509.speed.planet.nl) |
| 08:32.41 | ronr | hi, could anyone please advise on the following configuration for a production asterisk server? http://pastebin.ca/753468 (do the mentioned ISDN cards have echo cancellation?) |
| 08:35.04 | *** join/#asterisk BBHoss (n=hoss@146.229.191.117) |
| 08:35.53 | TrentCreek | ISDN cards are for digital communications not analog to digital conversion |
| 08:39.45 | ronr | TrentCreek: meaning echo cancellation is not an issue with an ISDN line and card? |
| 08:40.21 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-69-66.dsl.irvnca.pacbell.net) |
| 08:40.23 | TrentCreek | it could be if there is latency and jitter issues |
| 08:40.46 | TrentCreek | meaning ISDN is really not all that fast |
| 08:41.00 | TrentCreek | but should be able to hold, what 2 calls? |
| 08:41.07 | TrentCreek | maybe? |
| 08:41.21 | ronr | and if it is the solution is in cancellation in software, not hardware? |
| 08:41.51 | ronr | what do you mean with hold 2 calls? it'll be a E1 line giving me 15 ISDN channels for incoming and outgoing calls |
| 08:42.06 | TrentCreek | it is hardware issue if you use hardware for you AD/DA conversion |
| 08:43.06 | TrentCreek | okay then I see no problem...why would you have ISDN card? Why not E1 termination? |
| 08:43.35 | ronr | my plan is to ISDN -> Asterisk -> VoIP for most phones and ISDN -> Asterisk -> VoIP -> ATA -> Analog (dect) for some phones we already have laying around |
| 08:43.50 | ronr | what's E1 termination? |
| 08:44.09 | TrentCreek | a box that terminates the E1 so you can plug your network into |
| 08:44.23 | ronr | (I'm new to asterisk, just read the first 7 chapters of the o'reailly book last weekend, so that's my current state of knowledge) |
| 08:44.24 | TrentCreek | only makes sense |
| 08:44.53 | TrentCreek | That has nothing to do with Astrisk. It has to do with typical network communication |
| 08:45.28 | TrentCreek | nobody with E1 would connect to it using 15 IDSN cards |
| 08:45.37 | TrentCreek | *ISDn |
| 08:45.42 | peanut- | yarg. anyone use the xten client? trying to get it to work from behind nat, but asterisk box is still trying to send data to it's private IP instead of public one |
| 08:45.43 | peanut- | 03:41:32.321948 IP 10.0.4.6.10002 > 192.168.0.102.62026: UDP, length 172 |
| 08:45.59 | peanut- | 10.0.4.6 being asterisk, 192.168.0.102 being the SIP client on another NAT'd network |
| 08:46.09 | peanut- | nat=yes for the client in sip.conf |
| 08:46.26 | ronr | TrentCreek: that's not what I intented, I mentioned ISDN cards that connect to the E1 |
| 08:46.50 | linxroute | ISDN cards connect to E1 ? |
| 08:46.53 | linxroute | strange huh |
| 08:46.58 | peanut- | and 192.168 isn't in a localnet definition |
| 08:47.03 | TrentCreek | yes, but true |
| 08:47.17 | peanut- | and it damn well knows its public ip: Received Address: 69.148.18.126:63103 |
| 08:47.27 | peanut- | anyone know why it's trying to send to its private? |
| 08:47.50 | TrentCreek | Just terminate with E1 termination and plug your LAN up to it via a single cable. Don;t make it more difficult thanit is |
| 08:48.17 | ronr | TrentCreek: sounds good, what kind of device would that require? |
| 08:48.40 | linxroute | digium T102P |
| 08:48.44 | linxroute | single E1 |
| 08:48.46 | linxroute | card |
| 08:49.17 | TrentCreek | You need to ask your ISP about getting a E1 DSU |
| 08:49.23 | TrentCreek | or try eBay |
| 08:49.24 | ronr | 102P doesn't exist (at least according to google) |
| 08:50.53 | linxroute | http://www.digium.com/en/products/hardware/te120p.php |
| 08:50.57 | linxroute | TE102B |
| 08:51.07 | linxroute | 30B+D for you |
| 08:52.02 | TrentCreek | DSu would be a lot cheaper than that digium styff |
| 08:52.04 | TrentCreek | stuff |
| 08:53.01 | ronr | ah, TE120, that was the one I was considering, but now I'm also going to read up on the DSU stuff (I want to know what it is / does before making the decision) |
| 08:53.33 | TrentCreek | Look here. One on eBay only $79 US http://cgi.ebay.com/Eastern-Research-DNS-3000-E-E1-CSU-DSU-PBX-MUX_W0QQitemZ160172244677QQihZ006QQcategoryZ11175QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 08:53.54 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 08:54.25 | linxroute | how do you connect it |
| 08:54.27 | linxroute | to asterisk ? |
| 08:54.31 | linxroute | by LAN ? |
| 08:54.33 | TrentCreek | oops..its too old..who uses 10base T? |
| 08:54.36 | TrentCreek | YES |
| 08:55.45 | ronr | you happen to know a site with basic info on those devices / technology? |
| 08:56.20 | linxroute | yeap that would be nice |
| 08:56.34 | linxroute | because i dont see any way i can connect a lan to that device |
| 08:56.43 | TrentCreek | google would probably list a could of billion |
| 08:56.52 | TrentCreek | couple of billion |
| 08:58.23 | ronr | DSU = data service unit? |
| 08:59.37 | TrentCreek | http://en.wikipedia.org/wiki/Data_service_unit |
| 08:59.42 | TrentCreek | yeppers |
| 08:59.49 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:00.56 | linxroute | sorry if i may ask you TrentCreek |
| 09:01.00 | linxroute | if i buy this |
| 09:01.06 | linxroute | how can i let say |
| 09:01.07 | TrentCreek | but broadband is becoming so cheap, who wants to pay for high priced Dinos of T1/E1? |
| 09:01.18 | linxroute | register this device |
| 09:01.18 | *** join/#asterisk blq (n=Bl@dslb-088-067-024-190.pools.arcor-ip.net) |
| 09:01.21 | linxroute | with asterisk |
| 09:01.34 | TrentCreek | you don't. |
| 09:01.58 | linxroute | so what's this device use for |
| 09:02.13 | TrentCreek | You just plug it into your LAN card which Asterisj should use |
| 09:02.31 | TrentCreek | that is how I got my connected |
| 09:03.08 | linxroute | like this E1 connection -> DSU device LAN -> Asterisk |
| 09:03.23 | TrentCreek | yes |
| 09:03.39 | linxroute | hihi |
| 09:03.40 | TrentCreek | in other words, just like anybody else connects to the internet |
| 09:03.51 | linxroute | oh you mean for the net |
| 09:03.55 | ronr | and how do you handle calls? does the DSU speak some voip protocol or something? |
| 09:03.59 | TrentCreek | it's the same thing |
| 09:04.04 | TrentCreek | no |
| 09:04.07 | linxroute | i thougt this use to hande voice |
| 09:04.09 | TrentCreek | its TCP/IP |
| 09:04.14 | TrentCreek | it does |
| 09:04.49 | TrentCreek | If you are asking these questions...sounds like you need to go study computing 101 |
| 09:05.23 | TrentCreek | ALL voice traffic in the telephone system is carried via digial network |
| 09:05.46 | linxroute | still not getting it |
| 09:06.02 | linxroute | i use cisco connect to my * box |
| 09:06.12 | linxroute | the cisco ack as trunk device |
| 09:06.16 | linxroute | so i can dial out |
| 09:06.28 | TrentCreek | it dont matter if you use a string and tin can. |
| 09:06.43 | linxroute | i mean |
| 09:06.46 | TrentCreek | all voice traffic is converted to digial |
| 09:06.58 | linxroute | yes i know that |
| 09:07.30 | TrentCreek | okay then how else would voice traffic be getting out of * Box? |
| 09:07.44 | linxroute | i said cisco because some how i need to config the cisco gateway to register |
| 09:07.52 | linxroute | with the * box |
| 09:08.02 | linxroute | if the IOS support SIP |
| 09:08.24 | TrentCreek | is it a ATA Box? |
| 09:08.41 | linxroute | i mean i connect my asterisk server |
| 09:08.46 | linxroute | to a cisco 1700 router |
| 09:09.00 | linxroute | with a bri card installed |
| 09:09.31 | linxroute | for this device from what you said |
| 09:09.35 | linxroute | it would be very nice |
| 09:09.45 | linxroute | because it's much cheaper |
| 09:09.55 | TrentCreek | then get one |
| 09:09.58 | linxroute | compare to those cisco gear |
| 09:10.21 | linxroute | but i dont know how to config it |
| 09:10.26 | TrentCreek | just plug them in. If Linux box has drivers for cards then * should be able to use them to communicate |
| 09:10.52 | *** join/#asterisk stimpie (n=michiel@84-104-5-115.cable.quicknet.nl) |
| 09:11.10 | linxroute | does asterisk need any E1/T1 card ? |
| 09:11.39 | TrentCreek | if you have a E1/T1 then yes |
| 09:11.43 | Maliuta | how the heck to you intend to get a cisco 1700 BRI to handle phonelines for asterisk? |
| 09:11.57 | linxroute | just for lab |
| 09:12.04 | linxroute | Malitu |
| 09:12.19 | linxroute | with VIC-FXO as well |
| 09:12.28 | TrentCreek | you just plug it in and off you go |
| 09:12.30 | Maliuta | I don't think you'll get any VoIP functionality out of the 1700 series |
| 09:13.49 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 09:13.49 | *** topic/#asterisk is Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.4.13 (Oct. 10, 2007) Addons 1.2.8, 1.4.4 (Oct. 16, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. -=- http://www.digium.com/en/company/switchvox-acquisition-faq.php |
| 09:13.58 | linxroute | a sample configuration of cisco 1700 connect to asterisk |
| 09:13.59 | *** join/#asterisk LH-euhost (n=LH-euhos@L69df.l.strato-dslnet.de) |
| 09:14.06 | linxroute | with VIc-2FXo |
| 09:14.14 | linxroute | maliuta ? |
| 09:14.20 | TrentCreek | anything that can be converted to digital signal can be sent over any digital medium |
| 09:14.48 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 09:15.38 | linxroute | just plug in ? how does * understand and direct the voice trafic to the LAN card ? |
| 09:15.45 | linxroute | i meant the DSU device |
| 09:16.17 | TrentCreek | it does not..it is between the LAN card and the DSU |
| 09:16.37 | TrentCreek | * just simply communicated with a software socket to the LAN cards |
| 09:17.39 | linxroute | exten => _88.,1,Dial,sip/${EXTEN}@192.168.1.90 |
| 09:17.43 | linxroute | just like that |
| 09:17.47 | linxroute | for example ? |
| 09:18.08 | linxroute | 192.168.1.90 is the DSU device |
| 09:18.13 | linxroute | ? |
| 09:18.14 | TrentCreek | no |
| 09:18.19 | TrentCreek | that is not a SIP device |
| 09:18.51 | linxroute | if i dont bother you then can you please explain more |
| 09:19.01 | TrentCreek | http://revision3.com/systm/asterisk/ watch this video..2 years old and it explains the basics |
| 09:19.20 | TrentCreek | can have you up and running in minutes |
| 09:21.06 | TrentCreek | even can download sample configs |
| 09:22.11 | linxroute | so what kind of hardware do i need to get to run with DSU unit ? Aterisk box without an E1 card and just LAN connection to the DSU device ? |
| 09:22.53 | TrentCreek | E1 card or E1 DSU is about the same thing |
| 09:23.43 | TrentCreek | did you get the Second Edition book? |
| 09:24.33 | *** join/#asterisk af_ (n=getsmart@81-174-44-189.dynamic.ngi.it) |
| 09:24.44 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:25.08 | linxroute | yes but E1 card has driver for it, so i can config it to provide dial in-out function, how do you dialout |
| 09:25.14 | linxroute | with this device |
| 09:25.40 | linxroute | would you be so kind such as to provide a sample configuration |
| 09:26.16 | *** join/#asterisk carrello (n=salvator@81-174-56-54.static.ngi.it) |
| 09:26.21 | TrentCreek | you dont dial out and there is no configuartion..it is a transport and communication between networks |
| 09:26.42 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:26.55 | *** join/#asterisk phearless (n=phear@host217-34-75-65.in-addr.btopenworld.com) |
| 09:27.17 | TrentCreek | sounds like you should ne studying network communcation before you dive into Asterisk |
| 09:28.13 | linxroute | well i dont know much about telecom service stuff |
| 09:29.11 | TrentCreek | yes, I figured that out, but I am not refering to telecom stuff persay but a typical computer networking |
| 09:29.15 | linxroute | i just thought that, with this DSU device, i can rent a E1 line for 30 telephone number |
| 09:29.21 | TrentCreek | so do study that stuff. |
| 09:30.12 | linxroute | connect it to an asterisk box and provide telephone service |
| 09:30.17 | TrentCreek | you are thinking old ways |
| 09:31.03 | penguinFunk | why use CSU/DSU ? |
| 09:31.15 | TrentCreek | the E1 /30 telephone lines was just a way to describe the way it works in simple terms before computers were dominate and people were ignorant |
| 09:31.24 | penguinFunk | if you are planning to rent an E1 line, just get an E1 card |
| 09:31.43 | linxroute | from what Trent said |
| 09:32.08 | linxroute | i thought it could be cheaper so we can deploy it for one of our blind people call center |
| 09:32.29 | linxroute | i'm from vietnam so we dont have alot of money to do charity stuff |
| 09:32.30 | *** join/#asterisk yxa (n=lonari@58.185.90.101) |
| 09:32.37 | yxa | <PROTECTED> |
| 09:32.38 | TrentCreek | no matter what it is E1 card /E1 DSU simple translates from TCP/IP to another communicaton protocol to connect to the E1 network |
| 09:32.58 | penguinFunk | well when you have a DSU on one end of a line you need a CSU on the other? |
| 09:33.40 | linxroute | the how do you config stuff like DID number ? |
| 09:33.49 | penguinFunk | yxa, make sure your in the right directory |
| 09:33.57 | TrentCreek | that is why the Internet exists. An internet brings different protocols together to communicate on the same network |
| 09:34.20 | penguinFunk | linxroute: in /etc/asterisk/extensions.conf |
| 09:34.29 | linxroute | yes i know |
| 09:34.30 | linxroute | i mean |
| 09:34.36 | linxroute | with DSU device |
| 09:34.40 | TrentCreek | due |
| 09:34.42 | TrentCreek | dude |
| 09:34.49 | TrentCreek | i just told you |
| 09:34.57 | TrentCreek | go read up on networking |
| 09:35.06 | TrentCreek | it has NOTHING to do with Asterisk |
| 09:35.51 | linxroute | okay |
| 09:36.16 | TrentCreek | and check out that video |
| 09:36.45 | TrentCreek | http://revision3.com/systm/asterisk/ |
| 09:37.37 | linxroute | just wonder how it's communicate with asterisk so i can set up an call center for those blind people since the price is very good |
| 09:37.54 | TrentCreek | I alredy mentioned |
| 09:38.03 | penguinFunk | linxroute: as far as i can see, asterisk doesn't support DSU |
| 09:38.17 | k31th | Morning |
| 09:38.31 | TrentCreek | correct, because Asterisk does not communicate with it |
| 09:38.44 | penguinFunk | linxroute: you will need an E1 card if your planning to get an E1 line |
| 09:38.53 | penguinFunk | and use asterisk |
| 09:38.59 | linxroute | sure |
| 09:39.00 | TrentCreek | it sends packets to a software socket which communicated with the interface |
| 09:39.11 | TrentCreek | there are E1 DSUs |
| 09:39.12 | linxroute | i've been using * for 3 years |
| 09:39.19 | linxroute | hi K31th |
| 09:42.05 | TrentCreek | And you dont config DIDs in Asterisk |
| 09:42.33 | TrentCreek | you setup with your provider to point it to your box |
| 09:42.57 | TrentCreek | however you can reroute that number within Asterisk |
| 09:43.07 | *** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 09:43.16 | *** join/#asterisk munmun (n=mun_mun@202.122.127.252) |
| 09:43.19 | jeremy_g | morning |
| 09:43.25 | TrentCreek | just point it to the box and asterisk should answer |
| 09:44.09 | neax | linxroute |
| 09:44.34 | linxroute | yeap neax |
| 09:46.02 | neax | I can understand the desire of your organisation to spend as little as is earthly possible on its telephony infrastructure, but the reality is, telephony isn't cheap.. if they require 30 lines, their cheapest option is with an E1, however there is no avoiding the initial equipment cost which is quite high |
| 09:46.16 | neax | bite the bullet and buy an E1 card |
| 09:46.22 | penguinFunk | agreed |
| 09:46.57 | TrentCreek | Their cheapest option is not E1..Broadband getting quite cheap |
| 09:47.14 | linxroute | well i'm from vietnam |
| 09:47.30 | linxroute | it's not the same in the us or major developed countries |
| 09:47.48 | linxroute | an E1 line here ( just voice ) cost 120$ |
| 09:47.52 | linxroute | for a month |
| 09:48.03 | TrentCreek | that is cheap |
| 09:48.11 | linxroute | sure i can get an E1 card |
| 09:48.16 | jeremy_g | can asterisk 1.2.10 route anonymous incoming calls to some other number |
| 09:48.24 | linxroute | for as low as 300$ |
| 09:48.27 | neax | hell yes.. here in New Zealand, an E1 is around NZ$780 per month |
| 09:48.43 | linxroute | woops |
| 09:49.07 | TrentCreek | And in US major city..no less than $250 |
| 09:49.22 | TrentCreek | and we got T1, which is slower than E1 |
| 09:50.24 | TrentCreek | i am sure you can |
| 09:50.31 | ronr | E1 (15 lines) in the netherlands is free, however they do require you to make calls worth at least 270 euro a month |
| 09:50.50 | TrentCreek | that's not "free" |
| 09:50.54 | neax | jeremy_g: pretty sure you can.. calls can be routed based on their CID; i don't see why you couldn't route calls without CID differently, however I have never tried it |
| 09:51.48 | TrentCreek | you would have to look up in the book on how the scripting works |
| 09:51.49 | neax | isn't it strange how the telcos in other countries handle their billing in such different manners, yet generally end up with the same amount of money |
| 09:51.53 | ronr | feels free if you're coming from 3 ISDN BRI boxes (about 150 a month) + 500 euro worth of calls |
| 09:51.55 | linxroute | i meant 120 is just the rental cost , and when you call , pay on price per min |
| 09:51.56 | BBHoss | heh a full t1 in north alabama will run you at least 1k a month |
| 09:52.08 | neax | all the while convincing end users that they're getting an excellent deal :) |
| 09:52.16 | TrentCreek | that's out in the sticks! |
| 09:52.31 | BBHoss | still bull though |
| 09:52.38 | TrentCreek | cheaper to run your own optical cable to the phone company |
| 09:52.48 | BBHoss | well thats funny |
| 09:53.00 | BBHoss | they have fiber in the ground, but won't sell it to you |
| 09:53.32 | BBHoss | and the local cable company charges 10k+ a month for a simple 100mbit |
| 09:53.55 | TrentCreek | just run one to their central office and you can get super fast internet for dirt cheap |
| 09:54.19 | BBHoss | heh i wish it was that easy |
| 09:54.26 | TrentCreek | in Utah AT&T has fiber 10/10MB for $40 a month |
| 09:54.45 | BBHoss | you cant get fiber here for under 20k a month |
| 09:54.59 | BBHoss | probably because the cost of t1 is so high |
| 09:55.00 | TrentCreek | sure you can..i just told you |
| 09:55.10 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 09:55.24 | BBHoss | i spoke with the head of operations for the whole state and he said they couldnt do it |
| 09:55.42 | BBHoss | or wouldnt do it i guess |
| 09:55.55 | TrentCreek | http://www.usa.att.com/fiber/compare/index.jsp |
| 09:56.08 | TrentCreek | and full of shit |
| 09:56.26 | TrentCreek | would not do it...cuts into profits |
| 09:56.45 | BBHoss | yep |
| 09:57.02 | BBHoss | they haven't even lost the bellsouth branding yet |
| 09:57.50 | TrentCreek | Well...PacBell did not either before they changed again |
| 09:57.56 | BBHoss | § AT&T Direct Internet Access is not available in all areas; requires UTOPIA fiber installed to customer's property line. |
| 09:58.11 | BBHoss | i think the closest area this is avaliable is atlanta |
| 09:58.14 | TrentCreek | that is an example.. |
| 09:58.29 | TrentCreek | THE phone company in Hawaii is doing it also |
| 09:58.40 | TrentCreek | all those mountains and rock????? |
| 10:03.31 | Chris-NB | hi |
| 10:03.41 | Chris-NB | anyone had this message on cli: The previous reload command didn't finish yet |
| 10:07.09 | *** join/#asterisk HarryR`Work (n=harryr@77.240.56.17) |
| 10:09.21 | TrentCreek | `book |
| 10:09.27 | TrentCreek | ~book |
| 10:09.28 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 10:12.22 | jeremy_g | TrentCreek:does this contain info on configuring asterisk for anonymous calls |
| 10:16.49 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 10:18.37 | marl | anyone tell me why the following dial command wont timeout? exten => 01415351234,n,dial(ZAP/2/1470w0123456,20,gr) ? it never seems to timeout :( |
| 10:20.50 | neax | goodnight troops |
| 10:21.02 | jeremy_g | good nigh ne[a]x |
| 10:23.00 | *** join/#asterisk yannj_fr (n=yannj_fr@APuteaux-152-1-44-172.w82-120.abo.wanadoo.fr) |
| 10:24.15 | marl | can anyone tell me were the template is for * sending voicemail emails? |
| 10:24.25 | marl | or is it hard coded into the voicemail app? |
| 10:25.45 | tzafrir | marl, you asked this before and got no answer. Perhaps give more details, such as a trace? |
| 10:26.06 | tzafrir | see the sample voicemail.conf |
| 10:27.39 | marl | the trace apears to be normal, this is the strange thing, the dial command is not timeing out, everything else is working fine, but as far as i can tell that dial command should try the number its calling for 20 seconds, and if not answered in that time then should go onto the next command, but it doesnt time out |
| 10:27.54 | marl | even without the voicemail bit in it! |
| 10:28.46 | tzafrir | marl, are you sure that the dial command was reached? |
| 10:28.50 | marl | wat would be the best info to pastebin? got in trouble last night for pasting a log with verb set at 2! |
| 10:29.21 | tzafrir | what problems? |
| 10:31.12 | marl | yup dial command executes, and dials the number (my mobile) but instead of cutting off after 20 seconds, it just keeps ringing until my mobile provider cuts it off with there is no one answering your call, which is about 50 seconds! |
| 10:32.05 | marl | if i answer my mobile the call i connected as normal, and works, its just the timeout part that is failing :( |
| 10:35.55 | *** join/#asterisk Tourinho (n=gtourinh@cm-virtua-poa-C8B0F83B.dynamic.brdterra.com.br) |
| 10:36.21 | Tourinho | good day guys, where can I find information about programign extensions to asterisk? |
| 10:36.42 | Tourinho | by the way.. can I write programs in python and run it from asterisk? thanks |
| 10:36.47 | tzafrir | Tourinho, start with the sample extensions.conf |
| 10:37.08 | tzafrir | To use a different programming language, use AGI |
| 10:37.14 | Tourinho | tzafrir: I want to write ans external program |
| 10:37.27 | Tourinho | humm AGI? Right thanks |
| 10:37.55 | tzafrir | But generally try first using the internal dialplan logic. It's powerful enough for quite a few tasks. AGI has a performance penalty |
| 10:38.47 | tzafrir | ~pyasterisk |
| 10:38.48 | jbot | i guess pyasterisk is somewhat similar to res_perl. Allows you to call Asterisk API's from Python. See http://vox.groovy.net/moin/PyAsterisk |
| 10:38.51 | Tourinho | tzafrir: but if I need something that is not in dialplan applications, in that case I need to use AGI only? |
| 10:39.05 | Tourinho | great |
| 10:39.06 | *** join/#asterisk FreezeS (n=bla@82.208.157.125) |
| 10:39.11 | FreezeS | hello |
| 10:39.16 | HarryR`Work | and FastAGI should be considerably less overhead than local AGI |
| 10:39.19 | FreezeS | I have an AGI problem |
| 10:39.27 | tzafrir | You may also use the manager interface to have a remote program control Asterisk |
| 10:39.33 | Tourinho | link is donw :( |
| 10:39.49 | FreezeS | is there a way to send a SIP message through AGI ? |
| 10:40.23 | HarryR`Work | unless a dialplan application provides a way, no |
| 10:40.28 | agx | FreezeS, yes, you can invoke sipsak :) |
| 10:40.32 | FreezeS | I need to close the line without the BUSY signal, but to send a message that will close a cellphone gracefully |
| 10:41.53 | agx | FreezeS, i do not understand, but instead of Busy() you can use Congestion() i suppose |
| 10:43.03 | FreezeS | agx: thanls |
| 10:43.06 | FreezeS | thanks |
| 10:46.24 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
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| 11:00.54 | FreezeS | agx: seems Congestion is not exactly what I need, I still get the busy signal. Is there another command that will close the call transparently ? So the user wouldn't hear a busy tone and wouldn't need to press any buttons... |
| 11:02.52 | *** join/#asterisk blq (n=Bl@dslb-088-066-244-156.pools.arcor-ip.net) |
| 11:04.20 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
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| 11:04.55 | *** join/#asterisk myiagy (n=myiagy@189.34.7.113) |
| 11:05.46 | *** join/#asterisk spaghetty (n=spaghett@lugbari/people/spaghetty) |
| 11:07.07 | agx | anyone has a daylight saving time rule for GXP2000 in europe? |
| 11:08.37 | spaghetty | mm hi |
| 11:09.02 | agx | FreezeS, you want to transfer your user to another extension when you hungup? |
| 11:10.57 | *** join/#asterisk loca|host (n=tux@196.203.53.221) |
| 11:12.13 | loca|host | anyone can advise me a good Linux sipphone like Ekiga and supporting conference-calling ? |
| 11:12.17 | spaghetty | i'v some trouble into agi... when i call Dial from agi i got -1 in result even if the call is done |
| 11:16.20 | spaghetty | is -1 the code for error ? |
| 11:17.20 | FreezeS | agx: I have a call that is from a cellphone. I need to hangup that call, do some database queries, then call that number back. However, I want the hangup to be transparent for the user so he wouldn't hear the busy tone |
| 11:19.38 | agx | FreezeS, i think this is not possible if you're going to do a callback |
| 11:21.35 | FreezeS | the callback will be from another platform, I will launch a http request for callback. However, if the user doesn't press the "Cancel" button fast enough, that callback could fail |
| 11:23.04 | FreezeS | also, I just noticed something strange |
| 11:23.46 | FreezeS | although I removed all the "hangup" lines from my phpagi, the next priorities in the context are not executed after it exists |
| 11:31.55 | *** join/#asterisk yassaccan (n=yassacca@admin189.hgo.se) |
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| 11:40.18 | cy3o3 | sup |
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| 12:17.00 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:21.27 | *** join/#asterisk phix (i=threat@123-243-44-131.tpgi.com.au) |
| 12:21.29 | phix | :D |
| 12:23.00 | phix | Weird errors, a client has issues (distorted quality) phoning ppl on landlines via VoIP provider except for me, etf |
| 12:23.03 | phix | wtf |
| 12:23.37 | *** join/#asterisk coppice (n=chatzill@39.192.17.210.dyn.pacific.net.hk) |
| 12:24.58 | k31th | morning |
| 12:25.27 | k31th | phix: is there phone using the right codec, is there network congested? |
| 12:25.27 | *** join/#asterisk disa (n=disa8@shpd-78-36-169-5.vologda.ru) |
| 12:25.32 | disa | hi, all |
| 12:26.07 | disa | voip gateway register in * as 2222/2222 78.36.169.5 D N 10848 Unmonitored |
| 12:26.39 | disa | but when i dialing from this, asterisk write me: |
| 12:26.48 | disa | [Oct 29 15:41:05] WARNING[44129]: chan_sip.c:2262 get_in_brackets: No closing bracket found in '"2222" <sip:2222@78.36.169.5tag=2cd2749a-685799' |
| 12:26.48 | disa | [Oct 29 15:41:05] NOTICE[44129]: chan_sip.c:9026 check_user_full: From address missing 'sip:', using it anyway |
| 12:26.48 | disa | [Oct 29 15:41:05] WARNING[44129]: chan_sip.c:2262 get_in_brackets: No closing bracket found in '"2222" <sip:2222@78.36.169.5tag=2cd2749a-685799' |
| 12:26.54 | disa | what is problem ? |
| 12:27.37 | *** join/#asterisk bantu (n=Miranda@rz-du-mvx-142-16.rz.uni-karlsruhe.de) |
| 12:27.50 | agx | disa, missing ">" i suppose |
| 12:28.53 | disa | in sip.conf or in another place ? |
| 12:29.26 | *** join/#asterisk michael-i (n=michael-@141.41.40.55) |
| 12:30.15 | agx | disa, into the sip message; i don't know who generate it |
| 12:30.36 | disa | my connection is: VoIPGW->adsl_modem(nat)->asterisk |
| 12:31.11 | phix | k31th: evening :) |
| 12:32.18 | phix | NAT is fun |
| 12:32.42 | [TK]D-Fender | Your gateway is sending mangled SIP headers. |
| 12:33.00 | tzanger | sip's fucking fun even without nat |
| 12:34.09 | agx | disa, if this is the problem: try to put your asterisk on 5062 port instead of 5060 or check you bugged-gateway for "SIP ALG" mangling option... |
| 12:34.32 | tzanger | file: any idea why an ACL'd peer set (box a and box b) would owrk fine for months, then wehen I upgrade box a to svn trunk NOT CHANGING any sip.conf entries on either side, and not even reloading box b, they break? box a is not maching its peer entry on box b anymore and I can't figure out why. |
| 12:34.38 | tzanger | no nat, static ips.. it's infuriating |
| 12:35.37 | tzanger | svn-79915 -> 86264 |
| 12:36.22 | *** join/#asterisk saftsack (n=saftsack@pD9E06356.dip.t-dialin.net) |
| 12:36.24 | *** join/#asterisk lukketto (n=lukketto@host249-175-dynamic.3-87-r.retail.telecomitalia.it) |
| 12:37.53 | coppice | Nat Traversal is public enemy number 1 :-) |
| 12:38.17 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:38.17 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 12:40.16 | tzanger | coppice: I'm not traversing nat... that's the kicker |
| 12:40.34 | tzanger | now agx is... he may be in for more fun than i< am |
| 12:40.45 | coppice | see how far his evil reaches |
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| 12:42.06 | tzanger | :-) |
| 12:42.09 | [TK]D-Fender | Let the hate flow through you..... |
| 12:42.23 | agx | tzafrir uh? |
| 12:43.14 | coppice | fries [TK]D-Fender with courgettes and onions |
| 12:43.17 | agx | tzanger, do not understand :) |
| 12:43.42 | *** join/#asterisk ivrc (n=chatzill@adsl-074-228-054-168.sip.bct.bellsouth.net) |
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| 12:59.29 | *** join/#asterisk saftsack (n=saftsack@pD9E06356.dip.t-dialin.net) |
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| 13:10.03 | *** join/#asterisk roxlu (n=Roxlu@84-107-142-180.dsl.quicknet.nl) |
| 13:10.04 | roxlu | hi |
| 13:10.27 | roxlu | I'm trying to add some verbose logging to my php-agi script. I wrtie to the stderr, but the messages arent shown in the CLI... ? |
| 13:12.15 | FreezeS | roxlu: $agi->verbose("text"); |
| 13:12.25 | roxlu | FreezeS: i'm nog using phpagi |
| 13:12.33 | *** join/#asterisk mocker (n=user@198.247.173.227) |
| 13:12.37 | roxlu | I do a write to stderr |
| 13:13.03 | FreezeS | core set verbose 32 ? |
| 13:13.08 | roxlu | ah |
| 13:18.29 | destructure | I'm trying to make an outbound call, and ask the callee to press 1 to accept the call before bridging |
| 13:18.39 | destructure | I know I can play a message with dial, but it doesn't seem to accept dtmf |
| 13:21.51 | tzanger | ok svn upgrade mystery solved |
| 13:21.55 | agx | roxlu, i believe you see message sent to stderr only when you run asterisk as a non forking daemon "-f" |
| 13:22.06 | tzanger | older svn trunk incorrectly handled nat=no both on transmission and reception |
| 13:22.18 | tzanger | new svn trunk doesn't, but breaks interaction with older trunk |
| 13:22.24 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 13:22.29 | tzanger | solution: on older trunk, tell asterisk nat=yes so it accepts rport |
| 13:23.49 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 13:23.56 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
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| 13:26.27 | _x86_ | how do i setup hook-switch transfers? |
| 13:26.44 | [TK]D-Fender | tzanger: So now your * can both "taste great" AND be "less filling" :p |
| 13:26.48 | _x86_ | where an analog FXS station can hook flash and transfer to another extension? |
| 13:27.02 | tzanger | [TK]D-Fender: haha |
| 13:29.10 | file | tzanger: trying to talk to me that early doesn't work :D |
| 13:29.23 | *** join/#asterisk Teeli (n=tili@153.Red-80-38-134.staticIP.rima-tde.net) |
| 13:29.45 | tzanger | file: no, I am actually thanking you for fixing the sip nat stuff |
| 13:29.56 | roxlu | agx: thanks! I found out that the asterisk -r doesn't show output.. I need: asterisk -c |
| 13:30.08 | [TK]D-Fender | _x86_: "transfer=yes" in zapata.conf |
| 13:30.13 | file | random thanks? I'll accept that! |
| 13:30.18 | tzanger | file: :-) |
| 13:30.26 | _x86_ | [TK]D-Fender: then what do i have to setup in features.conf? |
| 13:30.36 | tzanger | I do not, however, thank you for making me learn about rport at 0130 :-p |
| 13:30.47 | [TK]D-Fender | _x86_: You don't. |
| 13:30.47 | _x86_ | [TK]D-Fender: and do I have to put anything in the dialplan to activate that feature? |
| 13:31.08 | [TK]D-Fender | _x86_: features.conf is more for things like loser SIP phones and analog Zap LINES. |
| 13:31.19 | _x86_ | gotcha |
| 13:31.25 | _x86_ | nifty :) |
| 13:31.35 | _x86_ | I'm so used to just using Polycom phones... |
| 13:31.41 | [TK]D-Fender | _x86_: Oh.. and Zaptel FXS = ASS :p |
| 13:31.45 | _x86_ | crappy ass analog phones suck |
| 13:32.08 | [TK]D-Fender | _x86_: I'd rather have a decent analog + Linksys ATA than many IP phones... |
| 13:32.09 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:32.14 | [TK]D-Fender | (hard) |
| 13:32.32 | _x86_ | [TK]D-Fender: transfer = yes is set already on the T1 going to the channel bank in zapata.conf |
| 13:32.41 | _x86_ | [TK]D-Fender: do i also need threewaycalling = yes? |
| 13:32.49 | [TK]D-Fender | _x86_: Probably a good idea. |
| 13:33.19 | puzzled | hi |
| 13:33.23 | _x86_ | but then when they hang up and try to pick it up to dial, they freak out and claim it's "never hanging up calls" |
| 13:34.00 | [TK]D-Fender | _x86_: ell them to grow up and realize that a quick trip to the hook = FLASH. |
| 13:34.27 | [TK]D-Fender | _x86_: I mean its not like this isn't a HOME grade service as it is... |
| 13:34.40 | *** join/#asterisk ManxPower (n=manxpowe@45.sub-70-221-163.myvzw.com) |
| 13:35.44 | puzzled | _x86_: you can adjust a setting in the source so that a quick hangup (which normally is a hook flash) is seen as a hangup by asterisk resulting in a new call with dialtone |
| 13:38.57 | blitzrage | tzanger: fixing the SIP NAT stuff? |
| 13:39.08 | *** join/#asterisk tru_`z24 (n=truz_`24@74-129-166-232.dhcp.insightbb.com) |
| 13:39.45 | *** join/#asterisk felix_da_catz (n=felix@PixNationalGroup.phonoscope.com) |
| 13:39.48 | puzzled | _x86_: in zaptel src play with SHORT_FLASH_TIME and ZT_DEFAULT_RXFLASHTIME |
| 13:40.03 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-fecd04509537605b) |
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| 13:49.41 | blitzrage | tzanger: you still in Toronto? Or back in the K-dub? |
| 13:51.04 | *** part/#asterisk BBHoss (n=hoss@146.229.191.117) |
| 13:57.18 | *** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it) |
| 13:57.52 | Mimmus | hi, what is the scope of setting a var in sip.conf by setvar=VAR=value? |
| 13:57.54 | *** join/#asterisk anonymiss (n=travesty@ool-435275b2.dyn.optonline.net) |
| 13:58.31 | ManxPower | Mimmus: channels for calls from that device will have that variable set in the dialplan |
| 13:58.38 | blitzrage | Mimmus: it's a channel variable |
| 13:58.49 | blitzrage | (what ManxPower said) |
| 13:59.03 | anonymiss | hey, does anyone have a working patch or solution to the SLTA/SLTM chan_ss7 with siemens switch problem? |
| 13:59.04 | [TK]D-Fender | (what blitzrage said about what ManxPower said) |
| 13:59.05 | Mimmus | mmmm.... |
| 13:59.17 | blitzrage | it's not global, if that's what you're asking |
| 13:59.30 | blitzrage | it's associated to the channel created by the device |
| 13:59.32 | Mimmus | blitzrage: probably you understand what was my problem |
| 13:59.41 | blitzrage | I do? I doubt it |
| 13:59.55 | blitzrage | maybe if you explained the problem I might, but at this point in time, I doubt it :) |
| 14:00.03 | [TK]D-Fender | Mimmus: You're hoping to lookup a value "about" a SIP peer that you wanted to DIAL I |
| 14:00.07 | [TK]D-Fender | 'm guessing |
| 14:00.24 | Mimmus | [TK]D-Fender: yes, for instance device associated to its number |
| 14:00.37 | Mimmus | [TK]D-Fender: DEV_232=SIP/232 |
| 14:00.49 | [TK]D-Fender | Mimmus: Sorry, you'd use something like AstDB or another outside resource for that... |
| 14:00.52 | *** join/#asterisk socken23 (n=socken@ip-213-189-154-029.fix.magnet.ch) |
| 14:01.03 | anonymiss | i'm commenting out code but i don't think i'm smart enough to fool mtp.c in to thinking it got the response |
| 14:01.08 | _x86_ | [TK]D-Fender: I've got three groups in zapata.conf, T1 to stations, T1 to PSTN, and (4) POTS lines |
| 14:01.23 | *** join/#asterisk Darthclue (n=e054502@fw149.nisd.net) |
| 14:01.41 | Mimmus | [TK]D-Fender: ah, I still need my awful globals_additional.conf |
| 14:01.52 | _x86_ | [TK]D-Fender: POTS lines are for inbound calls only... someone calls in on POTS, and talks to someone at an analog station... analog stations wants to transfer to SIP or to another analog station |
| 14:01.52 | socken23 | hi all! I tried to install a ISDN card from junghanns and now I can't start asterisk anymore (pbx.c:2902 ast_register_application: Already have an application 'PickUp'). Any idea where I should start looking?? |
| 14:02.03 | puzzled | anonymiss: don't know but you can ask on the asterisk ss7 list: http://lists.digium.com/mailman/listinfo/asterisk-ss7 |
| 14:02.17 | _x86_ | [TK]D-Fender: do the POTS lines _and_ the station T1 groups both have to be transfer=yes and threewaycalling=yes? |
| 14:02.23 | anonymiss | puzzled sounds good |
| 14:02.29 | [TK]D-Fender | _x86_: only FXS. |
| 14:03.12 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.165.58) |
| 14:04.17 | puzzled | socken23: add noload => <the filename of the pickup app> to modules.conf. just cures the symptom not the cause: you seem to have 2 applications that overlap. that should be fixed |
| 14:04.41 | socken23 | thanks! I'll try that!! |
| 14:05.40 | *** join/#asterisk javb (n=javb@190.80.231.205) |
| 14:06.16 | _x86_ | [TK]D-Fender: would the T option in my Dial command help? :) |
| 14:06.36 | socken23 | puzzled: Thanks! That worked so far! Have to dig deeper into the problem later on.. |
| 14:06.42 | [TK]D-Fender | _x86_: I don't think you should need it, but you could always try. |
| 14:06.59 | _x86_ | [TK]D-Fender: right now i just use "t" |
| 14:07.25 | _x86_ | [TK]D-Fender: transfer=yes, threewaycalling=yes also... transfers no workie |
| 14:07.27 | [TK]D-Fender | _x86_: Depends. if you're dialing TO your FXS, then "T", from = "t" |
| 14:07.55 | puzzled | socken23: good. have fun |
| 14:08.03 | _x86_ | yeah, from POTS to FSX channel bank station |
| 14:08.10 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:08.49 | [TK]D-Fender | _x86_: and need I repeat... Zaptel FXS = ASS :p |
| 14:10.41 | Mimmus | to attach residual analog devices, do you suggest a channel-bank or a fxs/sip gateway? |
| 14:11.53 | *** join/#asterisk ccaron (n=cedric@85-218-16-115.dclient.lsne.ch) |
| 14:12.33 | javb | What would the best sip phones out there? (branch?) ... I know it depends of individual expirience, but, there is always a generalize opinion |
| 14:12.35 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:13.03 | penguinFunk | polycom or snom |
| 14:13.24 | [TK]D-Fender | Mimmus: ATA or gateway. |
| 14:13.39 | [TK]D-Fender | javb: Polycom, pretty much hands down. |
| 14:13.48 | Mimmus | [TK]D-Fender: even if I have fax devices? |
| 14:14.08 | [TK]D-Fender | Mimmus: For those... stand-alone analog line kept as far away from * as humanly possible |
| 14:14.32 | javb | Ok. How well does Fax Modules / Detection on Asterisk work? |
| 14:14.43 | Mimmus | [TK]D-Fender: :-) in a site, I have a Rhino channel-bank and it is OK |
| 14:15.15 | Mimmus | [TK]D-Fender: for this reason, I I'm scaried to use an ATA/gw |
| 14:16.55 | [TK]D-Fender | Mimmus: Then you go on being scared, seems to work for you :) |
| 14:17.23 | Mimmus | [TK]D-Fender: for this, I need a channel-bank! |
| 14:21.03 | *** join/#asterisk l2trace99 (n=asd@fl-67-76-209-28.sta.embarqhsd.net) |
| 14:21.21 | Darthclue | Anybody have suggestions on where to begin with cleaning up tts using cepstral? The voice quality is somewhat iffy. |
| 14:21.51 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 14:22.27 | ManxPower | Darthclue: start by using it outside of asterisk and see how the sound quality is then |
| 14:25.32 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:25.32 | Darthclue | it sounds fine outside of asterisk. it has various quality issues when used inside of the system that may be more related to the voip factor but it's something i would like to try and resolve even if it leads to a dead end |
| 14:26.26 | *** join/#asterisk shtoom (n=godson@59.93.125.37) |
| 14:27.45 | k31th | How do i go about setting up call recording for asterisk? |
| 14:27.51 | *** join/#asterisk saftsack (n=saftsack@pD9E06356.dip.t-dialin.net) |
| 14:28.13 | k31th | I want to divert all marketing calls to the monkeys, and hear there reaction |
| 14:28.25 | ai-a | k31th: read up monitor |
| 14:30.49 | tzafrir | k31th, you know "tt" stands for "telmarketing torture" (or something similar) |
| 14:31.13 | l2trace99 | anyone know if you can change call info via a stun server ? |
| 14:31.25 | l2trace99 | like caller id |
| 14:31.30 | k31th | tzafrir: hahahaha no i didn't |
| 14:32.28 | *** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 14:33.11 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 14:33.30 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 14:33.38 | [TK]D-Fender | l2trace99: No, all STUN does is help tell the client what kind of NAT they are behind. |
| 14:34.13 | anonymouz666 | [TK]D-Fender: and sometimes tell you the wrong type. |
| 14:34.30 | *** join/#asterisk wick2o (n=wick2o@72.25.0.101.static.dejazzd.com) |
| 14:34.32 | wick2o | hello |
| 14:34.37 | Daviey | anonymouz666: O RLY |
| 14:34.45 | Daviey | anonymouz666: never found that myself |
| 14:35.05 | [TK]D-Fender | anonymouz666: Thats just a free bonus bundled with the software :p |
| 14:35.22 | [TK]D-Fender | (Failure is NOT an option) :p |
| 14:35.22 | wick2o | I've been reading Building telephony systems with asterisk and Asterisk the future of telephony, and about to start asterisk hacking |
| 14:35.35 | [TK]D-Fender | wick2o: Yee-haw. More power to you. |
| 14:35.36 | [TK]D-Fender | ~book |
| 14:35.37 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 14:35.40 | [TK]D-Fender | ^^^ more for you then... |
| 14:35.51 | wick2o | I have a merlin lengend communications system 2.0 i want to expand using asterisk |
| 14:38.28 | wick2o | nice |
| 14:42.52 | k31th | monitor files should go to /var/spool/asterisk/monitor ? |
| 14:45.34 | *** join/#asterisk BadPacket (n=John@unaffiliated/badpacket) |
| 14:46.37 | *** join/#asterisk nightrid3r (n=nnscript@d54C0303C.access.telenet.be) |
| 14:46.41 | *** part/#asterisk lukketto (n=lukketto@host249-175-dynamic.3-87-r.retail.telecomitalia.it) |
| 14:47.25 | *** join/#asterisk ussrback (n=MAX@80.92.183.84) |
| 14:47.33 | *** join/#asterisk af_ (n=getsmart@88-149-230-121.dynamic.ngi.it) |
| 14:49.28 | ussrback | Please, help me with realtime voicemail. I have configured it but i receive a lot of errors when executing it. http://pastebin.ca/753749 |
| 14:49.34 | ussrback | How can i fix that? |
| 14:49.52 | *** join/#asterisk gardo (n=gardo@121.97.138.233) |
| 14:51.06 | ai-a | ussrback: tried READING IT ? |
| 14:51.07 | ai-a | Unknown column 'dir' in 'where clause' (78) |
| 14:51.28 | ussrback | yes |
| 14:51.31 | ussrback | i see this |
| 14:51.37 | ussrback | so what u suggest? |
| 14:51.38 | ai-a | so.. fix your db. |
| 14:51.45 | ai-a | you need a column called 'dir' |
| 14:52.46 | ussrback | ohhh common i know databases. but i dont need to store voicemail messages. i just need realtime voicemail. to store only context and mailbox number |
| 14:53.16 | ussrback | http://www.voip-info.org/wiki-Asterisk+RealTime+Voicemail |
| 14:53.19 | ussrback | read this |
| 14:53.52 | ussrback | and this tooo http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage |
| 14:54.04 | ussrback | if you see they are quite different |
| 14:54.14 | ussrback | so your answer is not help for me |
| 14:54.58 | destructure | how about parked calls. anyone work with callparking? is it possible to continue in the dialplan after a parked call is answered and then hung up? |
| 14:55.41 | Darthclue | destructure, explain what you mean cause once a call is hung up it no longer exists |
| 14:56.27 | destructure | Darthclue: scenario- call "a" was parked. call "b" picks up, talks, hangs up, call "a" should then continue to the next step in the dialplan |
| 14:56.45 | destructure | in my testing, it "a" gets hung up |
| 14:57.24 | Darthclue | is b picking up and talking to a? |
| 14:57.40 | destructure | yes, b is retrieving the parked call |
| 14:57.47 | destructure | so, no timeout |
| 14:58.46 | Darthclue | ok, so unless b transfers a back into the dialplan, as far as * is concerned, when b hangs up, the call is over, so it terminates a. a would have to transfer b back into the dialplan |
| 14:59.18 | destructure | ok, thanks. That's too bad. not what I would expect |
| 14:59.18 | Darthclue | er, i mean b would have transfer a back into the dialplan |
| 14:59.22 | [TK]D-Fender | destructure: There are ways. You want to pick up a parked call... then when the call you picked up decides to hangup (not YOU), what would you want to do? |
| 15:00.01 | destructure | [TK]D-Fender: for example, a completes a survey about the call |
| 15:00.31 | destructure | if I wanted it to hangup, I would execute hangup after park ends |
| 15:00.52 | [TK]D-Fender | destructure: Ok, that would be fairly tricky to do and involve a lot of use of local channels and possibly call-files. |
| 15:01.19 | [TK]D-Fender | destructure: But I think its doable without excessive trickery |
| 15:03.00 | k31th | exten => 101,n,Monitor(wav,lol-monkeys,mb) should work right and mux the audio after the call ? |
| 15:03.27 | k31th | but it doesnt seem to be doing it? if i dont us ,mb it records to different files one for in and one for out. |
| 15:03.46 | destructure | yeah, I already have some local channels before this, for whisper tones |
| 15:04.06 | destructure | this is already part of a several thousand line ruby agi application |
| 15:04.28 | destructure | the problem is that it's difficult with asterisk to break and rejoin bridged calls |
| 15:04.34 | [TK]D-Fender | k31th: "show application mixmonitor" |
| 15:06.06 | [TK]D-Fender | destructure: To pickup a call I'd dial an exten that generates a call-file with a local channel that calls the pickup inside of a SECOND local channel. When that pickup answers, it will dial the person who INITIATED the dial with "g" so that they can hang up and let the caller continue on. |
| 15:06.23 | [TK]D-Fender | destructure: So 2 Local channels, and 1 call-file. |
| 15:06.53 | [TK]D-Fender | destructure: BTW... tracking billing/ etc for that call would kinda SUCK :p |
| 15:07.02 | destructure | interesting idea. I'll check that out. So when the first local channel hangs up, the wrapping channel goes to the next line in the dialplan |
| 15:07.26 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 15:08.00 | destructure | I have a separate asterisk server acting as a switch for that part |
| 15:08.14 | destructure | that part being billing |
| 15:10.24 | [TK]D-Fender | destructure: At least now you know what the HARD part will be. |
| 15:10.52 | [TK]D-Fender | destructure: And yeah, if you pass all of that through a border * server, that might do it... |
| 15:12.12 | destructure | this is looking good so far |
| 15:16.02 | *** join/#asterisk irule (n=irule@201.151.9.227) |
| 15:16.40 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:17.49 | *** join/#asterisk c0rnflake (n=c0rnflak@38.112.4.210) |
| 15:20.40 | *** join/#asterisk ctaloi (n=ctaloi@nat-66-218-1-87.usadatanet.com) |
| 15:21.40 | k31th | when i try and dial into my server it errors with: [Oct 29 16:20:43] NOTICE[9866]: chan_sip.c:13669 handle_request_invite: Call from '' to extension '101' rejected because extension not found. |
| 15:21.57 | k31th | however this ext works fine internally |
| 15:23.07 | [TK]D-Fender | k31th: O RLY. |
| 15:23.22 | destructure | [TK]D-Fender: I had to add /n to the local dial, but it worked. Thanks, you really saved me some time. I didn't need the call file, although that is happening earlier with the outbound portion |
| 15:23.23 | [TK]D-Fender | k31th: pastebin another attempt with SIP DEBUG enabled then... |
| 15:23.32 | [TK]D-Fender | destructure: Oh yes,... thats almost a given. |
| 15:23.45 | [TK]D-Fender | destructure: "/n" = sanity |
| 15:24.34 | destructure | what's the mnemonic there? "nsane" doesn't work, heh |
| 15:24.43 | [TK]D-Fender | destructure: I though the extra call filoe would be needed because the PARKED guy needs to be executing dialplan in order to continue. How'd you work around that? |
| 15:24.57 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:24.57 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:25.33 | destructure | the parked guy is already in a dialplan (technically, I will have to migrade this to agi). So after the local call ends, it just goes to the next line |
| 15:25.50 | destructure | the next priority that is |
| 15:26.13 | k31th | [TK]D-Fender: http://pastebin.ca/753782 |
| 15:27.31 | [TK]D-Fender | k31th: Found no matching peer or user for '193.111.200.11:5082' |
| 15:27.39 | [TK]D-Fender | k31th: Looking for 101 in default (domain 195.112.25.179) |
| 15:27.45 | [TK]D-Fender | k31th: SIP/2.0 404 Not Found |
| 15:27.49 | [TK]D-Fender | k31th: hmmmmm <----------- |
| 15:27.57 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 15:28.09 | [TK]D-Fender | k31th: Gee I guess calls aren't landing in the context you THINK they should be <-- |
| 15:29.27 | *** join/#asterisk Tili (n=tili@153.Red-80-38-134.staticIP.rima-tde.net) |
| 15:30.38 | destructure | [TK]D-Fender: here's the proof of concept if you are curious http://pastebin.com/m5227a1fe |
| 15:31.03 | destructure | really it's not even too hard to follow |
| 15:31.32 | destructure | I am only recently discovering how useful local channels are. |
| 15:32.08 | [TK]D-Fender | destructure exten => 9685,1,ParkedCall(9684) <- Didin't know you could PICK a lot like that.... |
| 15:32.15 | s34n | I have calls coming in from a SIP trunk that are ringing the correct extension, but... |
| 15:32.31 | destructure | yeah! I'd be dead in the water if I hadn't found that |
| 15:32.40 | destructure | actually, I'd be using meetme, which I really want to avoid |
| 15:32.50 | [TK]D-Fender | destructure: Oh that would jsut be FUGLY... |
| 15:32.50 | s34n | When the SIP extension picks up, asterisk has an INVITE failure on the trunk |
| 15:33.27 | [TK]D-Fender | s34n: PASTEBIN the failed attempt with SIP DEBUG enabled and verbose 10 |
| 15:33.30 | destructure | exactly. I have a prototype which worked by kicking the person out of the meetme, playing audio, and then rejoining them. it worked, but so convoluted |
| 15:33.46 | [TK]D-Fender | destructure: Well glad to hear its working out.... |
| 15:33.54 | destructure | yeah, thanks again |
| 15:34.54 | [TK]D-Fender | destructure: NP |
| 15:37.45 | mocker | Is there a way to have asterisk play a sound file only after the user has picked up the phone? |
| 15:37.53 | s34n | [TK]D-Fender: their is too much to capture on my scrollback. How to I redirect it to a file? |
| 15:37.56 | mocker | Right now I'm just looping it, but was wondering if there's a better way. |
| 15:38.09 | [TK]D-Fender | s34n: Get a bigger scrollback and stop copping out. |
| 15:38.15 | s34n | :) |
| 15:42.34 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 15:46.06 | Yourname`` | Myth or fact: AMD detection is affected when the system uses more channels than it can handle, and also when the load averages are more than normal, like more than 1.0 |
| 15:46.27 | k31th | [TK]D-Fender: it's in [internal] |
| 15:46.38 | russellb | isn't "AMD detection" redundant? :-p |
| 15:46.46 | tzafrir | Yourname``, single-cpu single core system? |
| 15:47.05 | coppice | Fact: AMD is just plain flaky, as it is so dependant on the behavour of the answering party |
| 15:47.09 | *** join/#asterisk cpina (n=carles@ip23498.bcn.altecom.net) |
| 15:47.18 | Yourname`` | tzafrir: Yes sir. |
| 15:47.44 | Yourname`` | coppice: Wouldn't that be the flakiness of the varied human responses, rather than the flakiness of the app itself? |
| 15:48.18 | cpina | hi |
| 15:48.19 | coppice | the algorithm is flaky, and there is no existing algorithm which is not |
| 15:48.25 | tzafrir | if load avarage > 1, chances are a channel is starved for CPU time. This may eventually affect the quality of audio |
| 15:48.26 | [TK]D-Fender | k31th: I saw your pastebin. Its coming in under [general] as un-authed. |
| 15:48.39 | Yourname`` | tzafrir: But the performance of an app like amd? |
| 15:48.56 | cpina | Is it possible to call to two extensions (Dial (ext1&ext2)) but, if one of both is busy, it jumps as busy? By default, if one is busy it rings other one |
| 15:48.56 | tzafrir | I don't really know AMD |
| 15:49.01 | Yourname`` | True enough, coppice |
| 15:49.07 | Yourname`` | russellb: It is? :P |
| 15:49.56 | [TK]D-Fender | cpina: Do a ChanIsAvail check on both before calling. |
| 15:50.19 | cpina | ok [TK]D-Fender, looks fine :-) |
| 15:50.25 | k31th | what does un-authed mean in this context? how an an un-authed call be coming in? |
| 15:50.42 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 15:52.32 | [TK]D-Fender | k31th: ....... |
| 15:52.42 | *** join/#asterisk marc7 (n=marc@S0106001c100a3e7c.gv.shawcable.net) |
| 15:53.00 | marc7 | if the credentials in sip.conf are correct, what would be causing asterisk to return a SIP/2.0 401 Unauthorized error? |
| 15:53.09 | [TK]D-Fender | k31th: it mean WTF is this guy calling in? I don't know who this is so just shove him wherever [general] says. |
| 15:53.24 | [TK]D-Fender | marc7: The OTHER side si wrong then. |
| 15:53.53 | [TK]D-Fender | marc7: The 2 sides don't agree. Feel free to blame whichever side of this that you want, the fact is they don't match |
| 15:54.36 | marc7 | [TK]D-Fender: let's go on the assumption that the credentials on both ends are correct, but that a change to the syntax of my sip.conf has now blown everything up |
| 15:54.37 | marc7 | because several peers who have previously been connecting fine are now no longer able to |
| 15:55.01 | [TK]D-Fender | marc7: Guess you should go examine what those changes were.... |
| 15:55.02 | s34n | [TK]D-Fender: http://rafb.net/p/1fFAOq88.html |
| 15:55.46 | Yourname`` | tzafrir: Thank you... |
| 15:56.44 | marc7 | there's no command to validate sip.conf? if I `reload sip` from the console, it shows that it's parsing both files without error... |
| 15:56.44 | marc7 | i don't know if it ever *would* error out |
| 15:57.44 | [TK]D-Fender | s34n: From: <sip:dslpbx@sipdomain.mvnet.com>;tag=as04dc714b <--- ? |
| 15:57.57 | *** join/#asterisk dandan (n=dandan@yarde-GW.customer.alter.net) |
| 15:58.02 | dandan | hey :) |
| 15:58.06 | dandan | is digium.com down? |
| 15:58.07 | [TK]D-Fender | marc7: No, there is little validation there... |
| 15:58.21 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 15:58.30 | dandan | or asterisk.org? or digiumasteriskworld.com ? |
| 15:59.03 | *** join/#asterisk techie (n=techie@adsl-76-214-7-62.dsl.lsan03.sbcglobal.net) |
| 15:59.07 | wick2o | dandan: i was just at digium.com |
| 15:59.12 | *** join/#asterisk tc3driver (n=huh@rrcs-24-199-16-118.west.biz.rr.com) |
| 15:59.18 | dandan | hmmmm |
| 15:59.22 | wick2o | tring it again now |
| 15:59.38 | dandan | i was trying to check the speakers at digiumasteriskworld.com |
| 15:59.43 | dandan | and was unable to... |
| 15:59.45 | wick2o | weird, its not comming up |
| 15:59.51 | russellb | yeah ... digium is ... dows |
| 15:59.53 | russellb | er, down. |
| 15:59.55 | wick2o | i was on there no less then 30 mins ago looking at hardware |
| 15:59.58 | russellb | like, all of it |
| 16:00.16 | coppice | get it some counselling |
| 16:00.18 | wick2o | anyone have experience with merlin legend systems? |
| 16:00.26 | russellb | coppice: :-p |
| 16:00.36 | dandan | hey russ :) prolly don't remember me, i spoke with you at the party in az :) re: ipv6 and srtp :) |
| 16:00.38 | k31th | [TK]D-Fender: nothing is in [general] apart from autofallthrough=yes |
| 16:00.55 | [TK]D-Fender | k31th: I didn't say EXTENSIONS.CONF DID I? |
| 16:01.09 | [TK]D-Fender | k31th: sip.conf silly...... calls aren't AUTHED by your DIALPLAN. |
| 16:01.21 | k31th | damn, my bad |
| 16:01.27 | russellb | dandan: yeah, i remember, hey :) |
| 16:01.27 | s34n | [TK]D-Fender: line #? |
| 16:01.27 | *** join/#asterisk Tourinho (n=gtourinh@cm-virtua-poa-C8B0F83B.dynamic.brdterra.com.br) |
| 16:01.39 | [TK]D-Fender | s34n: "Search" :) |
| 16:01.40 | dandan | russ: coming to Boston? |
| 16:01.42 | russellb | Darthclue: not me! |
| 16:01.42 | Tourinho | hi again, How can I execute a perl script from asterisk? |
| 16:01.51 | russellb | dandan: nope, wish i was though |
| 16:01.53 | russellb | i love boston |
| 16:02.02 | dandan | yeah, it was 26F today in CT |
| 16:02.10 | dandan | you can freeze your a$$ off :) |
| 16:02.11 | russellb | Tourinho: look at either the system application or AGI |
| 16:02.36 | k31th | is it possible to create an extension on sip.conf that forwards to a dialplan app ? |
| 16:02.39 | [TK]D-Fender | s34n: Hrm.... actually... not all that sure right now... |
| 16:02.49 | Tourinho | russellb I only found exec -> exec Executes a given Application |
| 16:02.52 | dandan | k31: you mean in extensions.conf? |
| 16:03.02 | Tourinho | but can I treat a perl script as an application? |
| 16:03.11 | russellb | Tourinho: exec is for executing asterisk applications |
| 16:03.16 | [TK]D-Fender | k31th: No and your concept is totally cracked |
| 16:03.20 | russellb | Tourinho: you probably want AGI |
| 16:03.26 | k31th | [TK]D-Fender: ? |
| 16:03.29 | russellb | search for "perl agi" |
| 16:03.34 | russellb | ~perlagi |
| 16:03.34 | [TK]D-Fender | k31th: EXTENSIONS have nothing to do with sip.conf. |
| 16:03.48 | dandan | ~book |
| 16:03.49 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 16:03.58 | [TK]D-Fender | k31th: You seem to completely misunderstand how calls get processed. |
| 16:04.00 | dandan | k31th: start ^^ there |
| 16:04.02 | *** join/#asterisk Rawplayer (n=kevin@cp108757-a.landg1.lb.home.nl) |
| 16:04.10 | *** join/#asterisk juanjoc (n=juanjoc@200.69.219.113) |
| 16:04.13 | dandan | btw. is v.2 downloadable yet? |
| 16:04.17 | russellb | yes |
| 16:04.25 | dandan | from that link? |
| 16:04.42 | dandan | (i got mine in carefree anyway, but I prefer pdfs when searching |
| 16:05.03 | s34n | [TK]D-Fender: I'll admit that I am slow and stupid if you will give me a better hint. |
| 16:05.06 | Tourinho | russellb thanks.. Ill take a look at it |
| 16:05.10 | Tourinho | thanks |
| 16:05.29 | [TK]D-Fender | s34n: I retracted my previous guess... not sure on this one. |
| 16:05.34 | s34n | k |
| 16:05.35 | dandan | yeah! v.2 downloaded :) |
| 16:05.56 | dandan | russellb: so who's gonna be there? |
| 16:06.08 | *** join/#asterisk MacDeath (n=davidn@hobbit.tsol.co.za) |
| 16:06.23 | russellb | dandan: i have no idea ... a bunch of people, though |
| 16:06.52 | dandan | ah, cool |
| 16:06.58 | dandan | gotta go and say hi :) |
| 16:08.39 | MacDeath | if i have a diginum card in my asterisk box |
| 16:08.50 | MacDeath | and i want my out bound calls to cycle |
| 16:09.14 | MacDeath | using the 4 channels in a random / round robin order |
| 16:09.18 | MacDeath | is this possible? |
| 16:10.55 | s34n | [TK]D-Fender: Is it normal for * to send an invite back to the trunk when the extension picks up? |
| 16:11.15 | *** join/#asterisk MacWinner (n=chatzill@70-100-130-167.dsl1-fairport.roc.ny.frontiernet.net) |
| 16:12.11 | s34n | [TK]D-Fender: if so, that INVITE will need to be auth'ed. But I don't see a clean way to do that. |
| 16:13.37 | s34n | [TK]D-Fender: I don't know how to have incoming and outgoing calls on the trunk use different secrets |
| 16:14.57 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 16:14.57 | *** mode/#asterisk [+o angler] by ChanServ |
| 16:15.05 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-d6a1c45127ef99ae) |
| 16:15.05 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:15.39 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 16:16.40 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-70-240-164-157.dsl.hstntx.swbell.net) |
| 16:16.58 | *** part/#asterisk jarod14 (n=jarod14@ns2.viatelecom.com) |
| 16:17.41 | k31th | [TK]D-Fender: for example if i call a 101@myasteriskbox.com i would presume that asterisk would look at the sip.conf and see if the extension exist and route a call to that sip device? ? |
| 16:18.46 | ManxPower | k31th: no, it looks at extensions.conf to see if the extension exists. |
| 16:19.22 | ManxPower | it looks at sip.conf to see what extensions.conf [context] the destination extension is in as well as the auth info to decide if it even wants to accept the call. |
| 16:21.16 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 16:26.11 | [TK]D-Fender | k31th: NO. |
| 16:26.44 | [TK]D-Fender | k31th: Call comes in, * sees who its FROM, then routes the call to the appropriate CONTEXT in your DIALPLAN and THAT decides what happens. |
| 16:26.58 | [TK]D-Fender | k31th: SIP devices are NOT extensions! |
| 16:27.33 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.149) |
| 16:30.25 | k31th | so basically i need to make a context for unknown calls ? |
| 16:31.00 | ManxPower | k31th: You had better if you don't want zillion dollar phone bills. |
| 16:31.23 | ManxPower | the context= in [general] in sip.conf is used if the call is not authenticated. |
| 16:31.31 | destructure | think of asterisk as a mux/demux. sip.conf muxes inbound calls from a remote system, depositing them in a dialplan context. the dialplan demuxes them based on extension called |
| 16:31.59 | k31th | ManxPower: i am talking about incoming calls... into my * box |
| 16:32.54 | [TK]D-Fender | k31th: Calls get processed auth'd by the channel driverf they arrive on and the processed by your dialplan. If you cannot get a solid grasp on this, * is not for you.by |
| 16:33.59 | ManxPower | k31th: so am I |
| 16:34.30 | ManxPower | incoming, unauthenticated SIP calls will use the context= in [general] in sip.conf |
| 16:34.42 | ManxPower | what is that context? |
| 16:34.58 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 16:35.18 | destructure | a part of the confusion here is the meaning of context I think. [general] in sip.conf is not the same as [general] in extensions.conf |
| 16:35.25 | ManxPower | obviously if you Dial(SIP/101@myasteriskbox.com) then the call will NOT BE AUTHENTICATED |
| 16:35.50 | *** join/#asterisk sevard (n=sev@192.235.0.85) |
| 16:35.55 | sevard | did you guys notice that the google street view icon to click and drag is an animated witch on a broom? |
| 16:36.13 | *** join/#asterisk jimmysolis (n=jimmy@190.41.82.1) |
| 16:36.56 | ManxPower | sip.conf: [general] context=fnords extensions.conf [fnords]. When an unauth'd SIP call comes in the call will land in the extensions.conf section called [fnords] and look for the extension there. |
| 16:37.20 | ManxPower | if you have some other kind of incoming calls, I'd love to hear about it. |
| 16:37.36 | k31th | ManxPower: other kind? |
| 16:37.49 | [TK]D-Fender | k31th: thats called SARCASM |
| 16:38.08 | [TK]D-Fender | k31th: Call is either authed, nor NOT authed. |
| 16:38.12 | [TK]D-Fender | or* |
| 16:38.53 | Tourinho | does asterisk support send REFER Method? |
| 16:39.12 | k31th | Ok i can see my error here... |
| 16:40.43 | k31th | works now, had the wrong context in [general] in sip.conf |
| 16:41.31 | anonymouz666 | Tourinho: yes. it can handle refer requests. |
| 16:43.02 | Tourinho | anonymouz666 but it can generate refer requests? |
| 16:43.33 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 16:44.13 | anonymouz666 | Tourinho: generate? |
| 16:44.36 | anonymouz666 | why asterisk should generate by itself a SIP Refer msg? |
| 16:45.38 | *** part/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 16:45.44 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 16:46.12 | Assid | heya |
| 16:46.18 | Assid | anyone know whats up with asterlink? |
| 16:46.34 | Tourinho | anonymouz666 to instruct caller to make another call |
| 16:46.42 | Tourinho | is that possible? |
| 16:47.53 | ai-a | Assid: explain your question. |
| 16:50.06 | anonymouz666 | Tourinho: join the asteriskbrasil channel. I don't want to speak English anymore. |
| 16:51.26 | *** join/#asterisk ghento (n=ghento@64.180.85.230) |
| 16:51.42 | Tourinho | anonymouz666 Im sorry? |
| 16:52.04 | MacWinner | if you have a VMware VM created, should it work on both windows and linux versions of the player? |
| 16:52.25 | Qwell | MacWinner: wrong channel, but yes |
| 16:52.41 | [TK]D-Fender | MacWinner: PLAYER? what "player"? |
| 16:52.47 | anonymouz666 | Tourinho: Entra no canal asterisk brasil e explica teu problema, talvez a gente possa te ajudar. :) |
| 16:56.35 | MacWinner | [TK]D-Fender: that's what VMwayer calls the thing that "plays" the premade VMs |
| 16:56.45 | MacWinner | at least that what i think they call it |
| 16:57.19 | _x86_ | [TK]D-Fender: got analog FXS transfers working on one system |
| 16:57.44 | _x86_ | [TK]D-Fender: on another system, not working... would busydetect=yes be interfering? |
| 17:00.01 | *** join/#asterisk alrs (n=lars@pozug.com) |
| 17:00.08 | [TK]D-Fender | _x86_: no clue |
| 17:02.12 | tzafrir | wow, just noticed Asterisk actually has multi-line comments |
| 17:02.41 | *** join/#asterisk fskrotzki (n=fskrot@host.textwise.com) |
| 17:06.12 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-f4bec359df98ae33) |
| 17:06.21 | hmmhesays | [TK]D-Fender: help me |
| 17:06.27 | hmmhesays | lol |
| 17:06.31 | [TK]D-Fender | hmmhesays: #drphil |
| 17:06.52 | hmmhesays | I can't figure out how to disable call waiting on this 501 in the config file |
| 17:06.57 | hmmhesays | call-limit is asterisk is not working |
| 17:06.58 | blitzrage | tzafrir: :D |
| 17:07.21 | [TK]D-Fender | hmmhesays: callsPerLinekey="1" lineKeys="1" |
| 17:07.40 | hmmhesays | but then they can only make one total call right? |
| 17:07.45 | *** join/#asterisk RypPn2 (i=TuMbL@rosscom.demon.co.uk) |
| 17:07.53 | hmmhesays | I want to disable call waiting but allow multiple outbound calls |
| 17:09.15 | *** part/#asterisk Tourinho (n=gtourinh@cm-virtua-poa-C8B0F83B.dynamic.brdterra.com.br) |
| 17:10.20 | [TK]D-Fender | Rawplayer: pwned |
| 17:10.21 | MacWinner | is there an easy way to configure a callback version of the meetme conference. ie, if I call a conference and enter a PIN, the pbx will call me back and join me to the conference? |
| 17:10.51 | [TK]D-Fender | MacWinner: lookup "call files" and "AMI Originate" on the WIKI |
| 17:10.54 | [TK]D-Fender | ~wikis |
| 17:10.54 | jbot | i guess wikis is http://www.voip-info.org |
| 17:11.07 | MacWinner | cool, thanks |
| 17:12.33 | MacWinner | this maybe a very dumb questions, by why does "terminate" and "originate" seem backwards in their definitions.. seems like when you call the PBX it should be terminating the call.. and when the PBX dials out, it should be originating them |
| 17:12.51 | ussrback | is there any way to reply to the user who left voicemail message? |
| 17:13.16 | destructure | everything is backwards in telephony. ex: FXO and FXS |
| 17:13.21 | [TK]D-Fender | ussrback: go read the WIKI page on Voicemail |
| 17:14.06 | ussrback | [TK]D-Fender: give me the link plz |
| 17:14.20 | [TK]D-Fender | ussrback: look up about 5 lines |
| 17:15.23 | ussrback | ahh this wikis are old and i did not found any interesting thing for this |
| 17:17.00 | [TK]D-Fender | ussrback: it still has the answer for this because the feature is old. |
| 17:17.07 | *** join/#asterisk marc7 (n=marc@S0106001c100a3e7c.gv.shawcable.net) |
| 17:18.12 | ussrback | mmmmmmmm more usefull info :) so it means this is possible to reply ? :) |
| 17:18.38 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:18.59 | *** join/#asterisk marc7 (n=marc@S0106001c100a3e7c.gv.shawcable.net) |
| 17:20.24 | c0rnflake | unf, i knew i should have waited before buying the 2nd edition book |
| 17:23.07 | [TK]D-Fender | ussrback: Yes, there are ways depending on a certain point of view. |
| 17:23.18 | [TK]D-Fender | c0rnflake: And why is that? |
| 17:23.29 | hmmhesays | [TK]D-Fender is there any way to do what I described? |
| 17:23.48 | *** join/#asterisk techie (n=techie@adsl-76-214-7-62.dsl.lsan03.sbcglobal.net) |
| 17:24.19 | [TK]D-Fender | hmmhesays: Closest option is to "conference" but never complete. |
| 17:24.44 | [TK]D-Fender | hmmhesays: Only other option is to limit at the point where you dial the device in your dialplan. |
| 17:25.01 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 17:26.08 | jimmysolis | someone knows how to work adit 600 with asterisk ?? |
| 17:27.56 | hmmhesays | [TK]D-Fender: ok, I'll have to check in the dialplan if I can send a call to the device or not |
| 17:28.06 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:28.25 | generalhan | hey all ! |
| 17:28.30 | hmmhesays | theres trouble |
| 17:30.16 | generalhan | anyone know of a way to specify the recording filename when using recordagentcalls in agents.conf ?? |
| 17:32.32 | *** join/#asterisk Tourinho (n=gtourinh@cm-virtua-poa-C8B0F83B.dynamic.brdterra.com.br) |
| 17:33.20 | generalhan | for example, in extensions.conf i save my recorded calls in this format: CALLFILENAME=/asterisk/monitor/${EXTEN}/${DATE}/incoming/${CALLERID(number)}-${TIMESTAMP}; Monitor(wav,${CALLFILENAME}) but i dont really have that syntax option in agents.conf |
| 17:33.35 | Tourinho | I just put exten => 123,12,AGI(agi-test.agi) but asterisk complain about extension that does not exists "rejected because extension not found." |
| 17:33.49 | *** join/#asterisk methods (i=1000@c-68-36-237-152.hsd1.nj.comcast.net) |
| 17:34.07 | methods | anyone know of a client which supports recording ? |
| 17:34.32 | [TK]D-Fender | Tourinho: PASTEBIn the failed call in its entirety |
| 17:35.00 | Tourinho | [TK]D-Fender where can I past it? |
| 17:35.14 | generalhan | ~pb |
| 17:35.15 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:35.16 | Tourinho | its just one line |
| 17:35.35 | [TK]D-Fender | Tourinho: not jsut 1 LINE, I want the whole call with SIP DEBUG if applicable/ |
| 17:35.40 | Tourinho | oks |
| 17:36.18 | [TK]D-Fender | Tourinho: Or whatever debug is applicable to that channel as well as your full dialplan context. |
| 17:36.27 | Tourinho | one second |
| 17:36.29 | [TK]D-Fender | Rawplayer: Indeed I do :) |
| 17:37.03 | Rawplayer | :p |
| 17:37.21 | *** join/#asterisk macros73 (n=cs@dsl093-063-236.pit1.dsl.speakeasy.net) |
| 17:38.44 | [TK]D-Fender | Rawplayer: Thats for trying to call my bluff :p |
| 17:39.05 | Yourname`` | Very nice, I now try to use ramfs for /var/spool/asterisk/outgoing and /var/lib/asterisk/sounds. May the force be with me. |
| 17:39.34 | [TK]D-Fender | Yourname``: Getting a heavy hit on disk? |
| 17:40.05 | Yourname`` | [TK]D-Fender: Quite a lot. Too many call files, and ofcourse when the sound files are being played at large. :( |
| 17:40.30 | [TK]D-Fender | Yourname``: Thats what disk caching is disk caching is for |
| 17:40.45 | Yourname`` | [TK]D-Fender : What do you mean? |
| 17:41.23 | [TK]D-Fender | echocancel=yes |
| 17:41.42 | [TK]D-Fender | Yourname``: Yeah, um... good disk caching should spare you the hit on playback of the same files.. |
| 17:41.45 | Tourinho | [TK]D-Fender there it is: http://pastebin.com/d40399a02 |
| 17:42.12 | [TK]D-Fender | Tourinho: And your full dialplan context please.... |
| 17:42.21 | Yourname`` | [TK]D-Fender: So you suggest I use disk caching rather than RAMfs? |
| 17:42.26 | Tourinho | sorry |
| 17:42.28 | destructure | after a file is loaded in .../sounds it is cached in ram, unless you don't have enough ram, in which case, ramdisk won't help |
| 17:42.39 | destructure | spool file is different |
| 17:42.54 | s34n | [TK]D-Fender: any clues from that dump? |
| 17:42.58 | destructure | that is, after you play it once |
| 17:43.22 | [TK]D-Fender | s34n: Didn't see it, and I don't do Core dumps |
| 17:43.27 | Yourname`` | destructure: I do have enough ram though. And that file will be played way too many times.. |
| 17:43.40 | destructure | played often means it should stay cached |
| 17:43.45 | destructure | what OS is this? |
| 17:43.52 | Yourname`` | destructure: In the RAM? (CentOS5) |
| 17:43.56 | s34n | [TK]D-Fender: not a core dump. the log you asked for at verbosity=10 |
| 17:44.00 | Tourinho | [TK]D-Fender : put extension together |
| 17:44.01 | destructure | try running free |
| 17:44.04 | Tourinho | [TK]D-Fender http://pastebin.com/d1cdb1db |
| 17:44.27 | [TK]D-Fender | Tourinho: Looking for 668 in default (domain XXX.XXX.XXX.XXX) |
| 17:44.27 | destructure | see that buffers/cache: that's the OS trying to keep oft used files in ram |
| 17:44.35 | [TK]D-Fender | Tourinho: SIP/2.0 404 Not Found |
| 17:44.56 | [TK]D-Fender | Tourinho: Does "default" look like [sip_testing] to YOU? |
| 17:45.33 | Yourname`` | destructure: Ok.. so if says 1866704 in free, and 143824 in used.. total mem 2010528 |
| 17:45.59 | MacDeath | using the 4 channels in a random / round robin order |
| 17:46.04 | destructure | Yourname``: try running time cat /var/lib/asterisk/sounds/demo-instruct.gsm >/dev/null |
| 17:46.07 | destructure | then run it again |
| 17:46.08 | MacDeath | oops |
| 17:46.12 | destructure | you'll see it's faster the second time |
| 17:46.17 | methods | anybody ? |
| 17:46.19 | destructure | you could run that test with any large file |
| 17:46.24 | MacDeath | is there a way to have outgoing calls use your trunks in a round robin order? |
| 17:46.40 | ai-a | whats a Trunk ? |
| 17:46.45 | Tourinho | [TK]D-Fender : no, but why its happening... if i have context=sip_testing in sip.conf |
| 17:47.13 | s34n | [TK]D-Fender: http://rafb.net/p/1fFAOq88.html |
| 17:47.33 | [TK]D-Fender | Yourname``: Oh yeah? Show me... |
| 17:48.03 | xheliox | AGI Rx << EXEC Background female_birthday |
| 17:48.03 | xheliox | ) -- AGI Script Executing Application: (Background) Options: (female_birthday does not exist in any format19]: file.c:563 ast_openstream_full: File female_birthday |
| 17:48.03 | xheliox | <PROTECTED> |
| 17:48.04 | Yourname`` | destructure: You'rr right! It's faster the next time. So you're saying that a sound file that's used on EVERY CALL by Asterisk is automatically put in RAM? |
| 17:48.06 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 17:48.12 | xheliox | What is this directory_birthday stuff/ |
| 17:48.14 | xheliox | ? |
| 17:48.49 | xheliox | The file female_birthday.ulaw and .gsm are both in the /var/lib/asterisk/sounds dir.. it's almost like asterisk is misreading my file name? |
| 17:48.58 | [TK]D-Fender | Tourinho: Oh yeah? Show me... <- rather |
| 17:48.58 | Tourinho | [TK]D-Fender : now its right, but still doesnt worki |
| 17:49.04 | Tourinho | Looking for 668 in sip_testing (domain 200.229.119.88) |
| 17:49.14 | Tourinho | X) |
| 17:49.19 | destructure | Yourname``: yes, if you have enough ram, it will stay there |
| 17:49.30 | destructure | Yourname``:however, that doesn't help for dynamic files, such as callfiles |
| 17:49.41 | [TK]D-Fender | Tourinho: exten => _668.,1,Wait(1) <-- guess you need to re-read the page on PATTERN MATCHING. |
| 17:49.51 | destructure | although they may be cached, still, there's work flushing to disk, etc. |
| 17:49.59 | destructure | they're temporary anyway |
| 17:50.03 | Yourname`` | destructure: So there really is no need for running RAMfs for these sound files then! phew! |
| 17:50.15 | destructure | not for the static ones, definitely |
| 17:50.51 | Yourname`` | destructure: But you think it'd make sense for ramfs for the callfiles? |
| 17:50.57 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:51.08 | destructure | I don't know. Are you having a problem? I'm inclined not to pre-optimized |
| 17:51.16 | destructure | it's easy enough to mount a ramdisk if you are having trouble |
| 17:51.43 | Yourname`` | destructure: I don't know if I'm having a problem, just trying to minimize CPU usage when Asterisk is being used. |
| 17:51.56 | Tourinho | [TK]D-Fender Will do it, for sure... but for now, can u tell me what is the problem, please :) |
| 17:51.59 | Qwell | Yourname``: it's not so much CPU that you'd be saving - it's disk I/O |
| 17:52.14 | destructure | the best way to minimize cpu usage would be to minimize transcoding. for example, if you call is ulaw, transcode those gsm files |
| 17:52.17 | destructure | in advance |
| 17:52.23 | Yourname`` | Qwell: Sorry, that's what I meant. :S |
| 17:52.27 | destructure | esp if it's g.729 |
| 17:52.35 | Qwell | Yourname``: what some people do with a ramfs, is save things like voicemail/recordings there, and at night (off-peak), save to disk |
| 17:52.40 | Yourname`` | destructure: Transcoded all sound files to ulaw as career uses ulaw. |
| 17:52.54 | Yourname`` | Qwell: Ah.. |
| 17:53.27 | Yourname`` | destructure: Currently transcoding is cut off. All sound files are in ulaw.. |
| 17:53.40 | destructure | Yourname``:I suggest you run some tests. Automate a few hundred calls to your server and see how it performs. Call yourself during the test to see if the audio is ok |
| 17:54.12 | [TK]D-Fender | Tourinho: "." = 1 or MORE extra digits. |
| 17:54.31 | [TK]D-Fender | Tourinho: Go read THE BOOK. |
| 17:54.33 | [TK]D-Fender | ~book |
| 17:54.34 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 17:54.45 | Yourname`` | destructure: Done those tests. Just that sometimes when load avg goes above 1.. the AMD starts slowing down. |
| 17:55.17 | destructure | how many calls? how does it sound? how many cpus? |
| 17:55.22 | Tourinho | [TK]D-Fender : damn, Im too dumm.. Im sorry that stupid question. It is OK now.. thanks for your time :) |
| 17:55.27 | destructure | did you look at the voip-info dimensioning page? |
| 17:55.46 | destructure | also, how did you automate? I'm curious since I have to do that, soon, and I'm always looking for tools |
| 17:55.56 | Yourname`` | destructure: Yes, that's where I branched out from. |
| 17:55.59 | Yourname`` | Automate what? |
| 17:56.04 | destructure | your load testing |
| 17:56.17 | [TK]D-Fender | Tourinho: I think its more like "lazy" I tell you that your pattern is clearly wrong and you didn't just go look at it and what each of the symbols you used MEANT. Instead you jsut asked me to TELL you. |
| 17:56.26 | Yourname`` | Thankfully, so far every kind of load testing was manual. Except for calls, where I used SIPp |
| 17:56.32 | [TK]D-Fender | Tourinho: So I'll write that off as sloppy AND lazy. |
| 17:56.48 | destructure | Yourname``: ah. I use sipp with rtp echo |
| 17:56.50 | [TK]D-Fender | *sigh* |
| 17:57.41 | Yourname`` | destructure: Meanwhile, do you have any reference material on how CentOS does the disk caching? Or even if it does at all. As I didn't know about it, but I'd like to know more. |
| 17:57.53 | destructure | it's not really a centos thing, but a linux thing |
| 17:57.57 | Tourinho | All right, I deserve it |
| 17:58.01 | Tourinho | :) |
| 17:58.19 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 17:58.19 | *** mode/#asterisk [+o denon] by ChanServ |
| 17:58.46 | dandan | hm, time to leave :) |
| 17:58.49 | dandan | cu all :) |
| 17:59.02 | dandan | wednesday I am in Boston, gotta see what's going on there... |
| 17:59.33 | hmmhesays | i'm running into the strangest problem with voicemailmain |
| 17:59.42 | hmmhesays | when it goes to play the old message it just hangs up |
| 18:01.03 | Yourname`` | destructure: : Looking.. |
| 18:01.03 | destructure | Yourname``: http://tldp.org/LDP/tlk/fs/filesystem.html section 9.3 might give you a start |
| 18:01.59 | destructure | Yourname``: notice that acronym LRU, "least recently used". the kernel has algorithms for expiring stuff in ram to make room for new stuff |
| 18:03.03 | Yourname`` | yeah.. |
| 18:04.14 | *** join/#asterisk kkn088 (n=kikoun@77.204.108.68) |
| 18:04.16 | destructure | Yourname``: another comment, load average (basically) means the number of processes waiting for CPU, not disk. I'd run the sipp test again, and study it. find the bottlenecks. how many calls did you run at 1.0 load? |
| 18:05.25 | Yourname`` | destructure: That's what I came to a conclusion on. Every call above 300 was causing it. As long as it's <= 300 it's fine.. |
| 18:05.52 | Yourname`` | destructure: On the other hand, thanks for your help in the RAMfs understanding. |
| 18:06.07 | *** part/#asterisk methods (i=1000@c-68-36-237-152.hsd1.nj.comcast.net) |
| 18:06.39 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 18:06.49 | *** join/#asterisk GreggB (n=GreggB@66.206.86.107) |
| 18:07.11 | hmmhesays | yeah asterisk is just randomly hanging up |
| 18:08.12 | hmmhesays | http://www.pastebin.ca/753979 |
| 18:08.55 | destructure | 300 doesn't sound bad. It might be easier to get another two boxes and load balance between them |
| 18:09.03 | destructure | plus you get redundancy out of it |
| 18:09.09 | *** part/#asterisk Tourinho (n=gtourinh@cm-virtua-poa-C8B0F83B.dynamic.brdterra.com.br) |
| 18:10.43 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 18:11.03 | *** join/#asterisk Egonis (n=roman@207.245.216.9) |
| 18:11.08 | *** join/#asterisk s0x (n=lalalal@200.27.107.122) |
| 18:11.14 | *** join/#asterisk bantu (n=Miranda@p54A340DA.dip0.t-ipconnect.de) |
| 18:11.33 | Egonis | I am trying to place an outgoing call on a T1 PRI, and it is giving me Code 54 -- INCOMING_CALL_BARRED -- has anyone experienced this or know how to fix it? |
| 18:12.44 | Yourname`` | Maybe linux caches it's spool dirs too. :D |
| 18:14.08 | hmmhesays | Darthclue: how are you? |
| 18:14.31 | Yourname`` | Someone should remove the reference to RAMfs on http://www.voip-info.org/wiki/view/Asterisk%20sound%20files |
| 18:15.19 | Darthclue | meh, lost in the southwest according to jbot. I've been getting more and more calls from people who need help with asterisk systems so I figured it was time to start lurking and see what all has changed in 2 years. |
| 18:16.29 | *** part/#asterisk Egonis (n=roman@207.245.216.9) |
| 18:16.43 | destructure | Yourname``: I don't think so. It might be relevant in certain contexts. For example, the first time a file is loaded, it will take awhile, so if you are loading lots if different files, it is a relevant optimization |
| 18:16.52 | destructure | maybe another bullet would be good |
| 18:16.53 | destructure | heh |
| 18:17.09 | Yourname`` | Makes sense. |
| 18:18.39 | hmmhesays | can anyone look at that pastebin? |
| 18:20.07 | xheliox | If someone might be able to give me an idea on this AGI Playback problem.. it can't find the file name no matter how I present it... And I don't get what "directoryind..." means --- http://pastebin.ca/753989 |
| 18:20.12 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 18:20.48 | *** join/#asterisk beasty (n=beasty@about/apple/macbook/beasty) |
| 18:20.51 | beasty | morning |
| 18:21.59 | beasty | anyone knows how i can't connect to the office asterisk machine from my home ? |
| 18:22.21 | ai-a | beasty: as in console, or sip phone ? |
| 18:22.33 | beasty | sip phones |
| 18:22.54 | *** join/#asterisk mamep (i=fallen@helios.edu.uoc.gr) |
| 18:23.35 | mamep | hello, how can i connect my asterisk to cisco callmanager in order to route calls to callmanager |
| 18:23.36 | mamep | ? |
| 18:25.31 | [TK]D-Fender | mamep: Setup SIP on CCM & dial it from your * server. |
| 18:25.38 | [TK]D-Fender | mamep: Oh, and go read THE BOOK |
| 18:25.39 | [TK]D-Fender | ~book |
| 18:25.40 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 18:25.53 | ai-a | beasty: try http://www.aocomputing.net/?p=3 |
| 18:26.22 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 18:27.29 | mamep | i have callmanager version 3.x and doesnot support sip |
| 18:27.33 | mamep | only h323 |
| 18:27.34 | *** join/#asterisk callguy (n=callguy@pool-72-70-78-28.bstnma.east.verizon.net) |
| 18:28.02 | ManxPower | mamep: then your life will suck, as H323 is very difficult to make work with Asterisk |
| 18:28.20 | [TK]D-Fender | mamep: Then read up on how to se tup * with H.323, and good luck... |
| 18:28.34 | mamep | something to start? |
| 18:28.46 | *** part/#asterisk s34n (n=smcmurra@ip-208-76-93-125.mvdsl.com) |
| 18:28.55 | hmmhesays | this is just crazy |
| 18:29.23 | irule | is there some place in the internets where I may find a clear and simple explanation on how to use dundi to make 2 * server act as one? Ive seen some examples but none are in my language |
| 18:29.39 | Darthclue | hmmhesays, i looked at the pastebin, but I couldn't tell ya what it really means. I see where it disconnects, but unless it has something to do with the file locking I dunno. |
| 18:29.45 | hmmhesays | can you adjust zaptel gains while you are on the phone? |
| 18:29.59 | mamep | [TK]D-Fender i have already an * running...but i can't find any usefull guide in order to get work with h323 |
| 18:30.15 | Darthclue | ~book |
| 18:30.15 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 18:31.05 | *** join/#asterisk s34n (n=smcmurra@ip-208-76-93-125.mvdsl.com) |
| 18:31.13 | s34n | \quit |
| 18:31.45 | hmmhesays | after I change the gain I just ztcfg ? |
| 18:32.03 | *** join/#asterisk s34n (n=chatzill@ip-208-76-93-125.mvdsl.com) |
| 18:33.13 | GreggB | Short of unloading the wcte12xp module and reloading it, does anyone know how to manually force the entry shown in /proc/zaptel/1 from "1 WCT1/0/1 Clear Master (In use)" to "1 WCT1/0/1 Clear (In use)"? |
| 18:33.30 | hmmhesays | yes no maybe so? |
| 18:33.44 | mamep | [TK]D-Fender : something except the book? |
| 18:33.45 | *** join/#asterisk Shaun2222 (n=shaun@ip68-4-127-67.oc.oc.cox.net) |
| 18:36.08 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:37.00 | marc7 | I've figured out that when I comment out 'bindaddr' in the [general] section of my sip.conf, so that I can listen on more than one IP address (all)... I'm getting a ton of "SIP/2.0 401 Unauthorized" error messages... adding the line back in solves the problem. does this give you guys any ideas? I'm running 1.4.13 |
| 18:37.31 | marc7 | is there any syntax to bindaddr to have it listen specifically on two addresses? |
| 18:37.57 | *** join/#asterisk techie (n=techie@adsl-76-214-21-190.dsl.lsan03.sbcglobal.net) |
| 18:38.00 | *** part/#asterisk techie (n=techie@adsl-76-214-21-190.dsl.lsan03.sbcglobal.net) |
| 18:39.21 | ManxPower | is there a way to have verbose=1 in the CLI, but verbose=3 for /var/log/asterisk/messages ? |
| 18:39.37 | ManxPower | marc7: why do you want to specify the addresses? It almost always screws things up. |
| 18:40.12 | ManxPower | GreggB: no. sync cannot be changed without dropping all calls. |
| 18:40.26 | [TK]D-Fender | mamep: Go read the sample docs with *, and the WIKI. |
| 18:40.28 | [TK]D-Fender | ~wikis |
| 18:40.29 | jbot | i guess wikis is http://www.voip-info.org |
| 18:40.30 | marc7 | ManxPower: I want to listen on every interface, but in so doing, sip peers that try to register get a 401 unauthorized error... so how do you suggest I listen on more than one IP if I don't specify anything |
| 18:40.42 | tzanger | how does one override the desired codec for a specific call in the dialplan? |
| 18:40.52 | hmmhesays | do I have to do anything to change zaptel gain other than ztcfg after the config file change? |
| 18:40.53 | tzanger | I remember there was a way, I just don't remember what variable it is |
| 18:40.54 | ManxPower | marc7: by default Asterisk will listen on all interfaces. |
| 18:40.59 | GreggB | ManxPower: alright - could that explain why we'd hear noise (squealing, and random buzzing) on this channel, but not on others? |
| 18:41.07 | ManxPower | tzanger: __SIP_CODEC, I think |
| 18:41.32 | ManxPower | GreggB: I doubt it. |
| 18:42.03 | ManxPower | marc7: The Asterisk relies on the OS to route the packets out the correct interface. |
| 18:42.13 | marc7 | ManxPower: okay... so I have a sip.conf that works fine if there is a 'bindaddr = 123.45.67.8' line in there, but if it's commented out with a semicolon... our peers can't register. I don't understand why that is. |
| 18:42.38 | ManxPower | marc7: what address are your peers trying to register to? |
| 18:42.45 | mamep | [TK]D-Fender : http://www.voip-info.org ? |
| 18:42.52 | ManxPower | and where are those peers, local network or remote? |
| 18:43.04 | GreggB | ManxPower: we were getting all kinds of awful noise when dialing out via this channel, but the others seemed fine. The only change was to unload and reload this module, and things cleared up. |
| 18:43.18 | ManxPower | GreggB: then you have some other issue. |
| 18:43.23 | ManxPower | IRQ conflict would be my guess. |
| 18:43.27 | [TK]D-Fender | mamep: Yes |
| 18:43.32 | ManxPower | also that is a T-1 card, so you will have 24 channels |
| 18:43.34 | marc7 | half of the peers are remote, connecting to an outbound interface. half of them are trying to connect to a 10-space IP address that's routing packets from a VPN concentrator in our network |
| 18:43.42 | ManxPower | Well, T-1/E-1, of course. |
| 18:43.49 | *** join/#asterisk Boones (i=Boones@port-212-202-42-6.dynamic.qsc.de) |
| 18:44.07 | marc7 | ManxPower: i could show you a sip debug log if you'd like |
| 18:44.14 | ManxPower | marc7: the devices must connect to the correct address in the first place. SIP packets have INSIDE the DATA part, the IP addresses specified |
| 18:44.47 | ManxPower | marc7: so BOTH types of devices can't connect when you remove the bindaddr= option? |
| 18:44.53 | GreggB | ManxPower: IRQ conflict? Strange...this box had been working great for ~6 months. |
| 18:45.04 | ManxPower | GreggB: what changed? |
| 18:45.48 | GreggB | ManxPower: telco removed a bridge-tap about a month back, and we've been keeping up with the asterisk/libpri/zaptel releases |
| 18:46.21 | *** join/#asterisk Egonis (n=roman@207.245.216.9) |
| 18:46.23 | GreggB | ManxPower: other than that, nothing |
| 18:46.33 | generalhan | is there a way that i can pass a varaible from extensions.conf to agents.conf ? i cannot figure out how to get the agent recordings to label properlly and if i can just set a variable before i call the queue then maybe i can get this to work somehow |
| 18:46.55 | marc7 | ManxPower: correct, but hold on... something coming up |
| 18:46.57 | Egonis | When trying to dial via Zap/G1 (which is span 1 channels 1-23 in zapata.conf) I get the message: 'Channel 0/23, span 1 got hangup, cause 54' -- why would it try dialing channel 0? |
| 18:47.39 | *** join/#asterisk jimmysolis (n=jimmy@190.41.82.1) |
| 18:48.06 | *** join/#asterisk tru_`z24 (n=truz_`24@74-129-166-232.dhcp.insightbb.com) |
| 18:48.13 | jimmysolis | someone know how to work asterisk with ADIT 600(CMG)? |
| 18:50.28 | *** join/#asterisk anujsingh (n=anuj@59.94.129.244) |
| 18:50.39 | anujsingh | hello everyone |
| 18:52.22 | anujsingh | how do i save all conversations in asterisk+vicidialer setup, At once it worked , now i made some changes and it is not working , files were recorded in /var/spool/asterisk/monitor directory |
| 18:53.23 | anujsingh | I recreated whole database of vicidial asterisk , dialplan , |
| 18:53.53 | anujsingh | can anyone help me where to look at , i am new with asterisk |
| 18:53.58 | [TK]D-Fender | anujsingh: Its your dialplan, go shove Monitor & MixMonitor around however you like.... |
| 18:54.27 | [TK]D-Fender | anujsingh: "show application monitor" , "show application mixmonitor" |
| 18:54.37 | *** part/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 18:54.43 | _x86_ | [TK]D-Fender: trying to do a hook-flash transfer, and getting dialtone upon flash, but when i enter the extension to transfer to, it returns me to the first leg of the call, and puts this in the CLI: Dumping incomplete call on Zap/16-1 |
| 18:54.50 | hmmhesays | sounds like a routing issue |
| 18:54.53 | anujsingh | yes, last time i followed scratch install, |
| 18:55.05 | [TK]D-Fender | _x86_: Sorry, can't help from there... don't use Zaptel FXS... |
| 18:55.17 | anujsingh | it means i have to make proper changes in dialplan ? |
| 18:55.25 | marc7 | ManxPower: the problem is that when bindaddr is set, the handshake/negotiation never completes successfully. Typically, there's a SIP REGISTER sent by the phones, the asterisk server says "Unauthorized" and provides the realm for credentials to be passed back... the phone passes the credentials, and everybody's happy..... when we remove that last bindaddr, that last step (of the phones passing back credentials) never happens |
| 18:55.41 | marc7 | if you want, take a look at these error logs: http://samedwards.info/asterisk/ |
| 18:55.51 | hmmhesays | [TK]D-Fender: can I make the ip 501 save its handset volume position somehow? |
| 18:56.04 | [TK]D-Fender | hmmhesays: Lookup "persist" in sip.cfg |
| 18:56.16 | ManxPower | marc7: sounds like the phones are behind NAT or a firewall |
| 18:57.11 | marc7 | ManxPower: well, the phones connecting to the "outside" interface *are*.. but NAT is setup correctly. again, everything works fine when the 'bindaddr' is in the sip.conf file |
| 18:57.37 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 18:57.39 | marc7 | ManxPower: and the phones connecting on the "internal" interface are *not* using NAT/firewall... there's a router here that's passing traffic as if they were on localnet |
| 18:57.42 | *** join/#asterisk VijayG (i=VijayG@58.68.47.118) |
| 18:57.46 | _x86_ | anyone ever get "Dropping incomplete call on on Zap/16-1" or similar, when trying to do a hook-flash transfer? |
| 18:57.59 | _x86_ | [TK]D-Fender: is there a way i can do a blind transfer without attempting a 3-way? |
| 18:58.05 | VijayG | hello |
| 18:58.22 | [TK]D-Fender | _x86_: short of DTMF, no clue. |
| 18:58.52 | *** part/#asterisk doug (i=doug@zaxxon.telerama.com) |
| 18:59.13 | VijayG | how can i clear the logs of asterisk |
| 18:59.15 | VijayG | where does it maintain the logs |
| 18:59.33 | [TK]D-Fender | VijayG: /var/log/asterisk |
| 18:59.37 | marc7 | [TK]D-Fender: any thoughts on this issue that I've been talking to ManxPower about? it's as if the phones do a completely new register... it doesn't respond to the 401 unauthorized response |
| 18:59.58 | [TK]D-Fender | marc7: no clue |
| 19:00.52 | VijayG | can i delete everything over there? |
| 19:00.54 | VijayG | or few files? |
| 19:01.14 | [TK]D-Fender | VijayG: You can delete them all if you want. |
| 19:01.33 | VijayG | ok |
| 19:04.35 | *** join/#asterisk dlynes_laptop (n=dlynes@216.251.149.66) |
| 19:06.58 | s34n | I need a better understanding of how * connects a trunk call to an extension |
| 19:07.33 | s34n | Once the extension picks up the incoming call, it looks like * sends an INVITE back to the trunk |
| 19:08.28 | s34n | in my case, that INVITE fails for some reason (could be auth) |
| 19:11.27 | ManxPower | s34n: Sorry, but I don't help people that use the wrong terms. There is no such thing as a "SIP Trunk" in Asterisk. |
| 19:12.17 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:15.03 | [TK]D-Fender | s34n: When its a 401 it isn't "maybe" the auth, it IS the auth |
| 19:16.43 | *** join/#asterisk mountainm2k (n=mountain@165.236.183.1) |
| 19:16.56 | s34n | [TK]D-Fender: Well, I'm a bit confused on how to fix it. I'm registered fine. The call is incoming. But the auth seems to be my * server failing auth against the provider. |
| 19:17.06 | k31th | [TK]D-Fender: cheers for you help today |
| 19:17.21 | [TK]D-Fender | s34n: register has nothing to do with authing incoming calls. |
| 19:17.39 | [TK]D-Fender | k31th: Np, and grab the book, sit down, and get your head ouut of your ass :) |
| 19:17.58 | k31th | I can see your frustration looking back. |
| 19:18.36 | s34n | [TK]D-Fender: ok. but registration should be the only time the * server has to auth to the provider |
| 19:18.38 | [TK]D-Fender | k31th: well.... you seem to aspire to offering services yet fail to grasp the simplest and most important concepts of * for ANY kind of use. Scary stuff.... |
| 19:18.49 | [TK]D-Fender | s34n: LOL, no :) |
| 19:18.51 | [TK]D-Fender | ~sipregister |
| 19:18.51 | jbot | [~sipregister] SIP Registration is to tell your provider as to what IP address & EXTEN to send INCOMING calls to. Some ITSP's let you use a fixed IP or host rather that register. Registration is NOT normally needed to PLACE a call, as those are typically auth'd independently. Others accept UN-AUTH'D calls once you are registered (saves on negotiation BW) |
| 19:18.54 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^ |
| 19:19.17 | k31th | [TK]D-Fender: It's not just me who works here :p |
| 19:19.28 | [TK]D-Fender | s34n: It only informs them WHERE to send calls, not HOW |
| 19:19.45 | [TK]D-Fender | k31th: time for a roster change :p |
| 19:20.35 | k31th | lol, i have only been playing with asterisk for a few days, untill then i used freepbx... |
| 19:21.12 | Strom_M | freepbx: it rots the mind |
| 19:21.18 | mountainm2k | Q: How can I change Caller-ID for calls being *transferred*? Today the caller-ID shows that of the person doing the transfer, rather than (preferrably) that of the call they are transferring |
| 19:21.35 | Strom_M | mountainm2k: do a blind transfer rather than an attended transfer |
| 19:21.43 | s34n | [TK]D-Fender: calls from the ITSP are unauthed. If I'm reading it correctly, the failure is happening when * sends an INVITE back to the ITSP. |
| 19:22.01 | s34n | (not sure why * is sending an INVITE to the ITSP) |
| 19:22.01 | mountainm2k | Hrrrm, need to RTFM for Polycom phones on that then I'm guessing |
| 19:22.06 | mountainm2k | only one transfer key |
| 19:22.13 | Strom_M | mountainm2k: which phone? |
| 19:22.20 | mountainm2k | IP301 |
| 19:22.25 | [TK]D-Fender | k31th: s/days/weeks/ <- I've seen you here longer than I'd use "days" to represent.... |
| 19:22.36 | s34n | [TK]D-Fender: I'm unclear on how to configure incoming auth vs outgoing auth |
| 19:22.44 | [TK]D-Fender | mountainm2k: Do a BLIND transfer instead of an ATTENDED transfer |
| 19:22.47 | Strom_M | mountainm2k: yeah, id have to drag mine out and dust it off |
| 19:22.52 | Strom_M | [TK]D-Fender: thats what I just said |
| 19:22.54 | [TK]D-Fender | s34n: thats in your peer. |
| 19:22.57 | s34n | [TK]D-Fender: but this is incoming, so it should be unath'd |
| 19:23.17 | [TK]D-Fender | Strom_M: Did read for it, seemed faster to type that search. |
| 19:23.20 | [TK]D-Fender | didn't* |
| 19:23.24 | mountainm2k | [tk]: on Polycom, is there a setting for that? Or is it a different feature key, etc? |
| 19:23.24 | Strom_M | laaaaaaaaaaaaaaaame |
| 19:23.30 | k31th | [TK]D-Fender: been here yes, but not really attempted to learn * untill about a week ago. |
| 19:23.34 | s34n | [TK]D-Fender: 'm unclear on how to configure incoming auth vs outgoing auth in the peer |
| 19:23.54 | wick2o | anyone familer with spanning from a merlin legend system? |
| 19:24.05 | [TK]D-Fender | Strom_M: Yes, you are, but I'm used to your whining... go on some more about the raging hard-on VSC evangelizing gives you ;) |
| 19:24.10 | k31th | But im sure i'v been here asking stupid questions alot longer than that :D |
| 19:24.32 | *** join/#asterisk dioedu (n=dioedu@201.7.117.114) |
| 19:24.34 | Strom_M | [TK]D-Fender: go play with a nut |
| 19:24.59 | dioedu | hello, i have a gateway that use the protocol h248. |
| 19:25.03 | [TK]D-Fender | Strom_M: This channel is full of them as it is... |
| 19:25.35 | dioedu | This protocol is known as megaco, that is a implementation of MGCP protocol. |
| 19:25.55 | s34n | dioedu: true |
| 19:26.05 | dioedu | chan_mgcp works with this protocol ? |
| 19:26.05 | [TK]D-Fender | dioedu: * supports MGCP only to support PHONES IIRC |
| 19:26.12 | _x86_ | gah |
| 19:26.15 | [TK]D-Fender | dioedu: Yes as stated above |
| 19:26.28 | _x86_ | i cant figure out how to make this transfer work |
| 19:26.52 | anujsingh | [TK]D-Fender is it possible to record all conversation by vicidial web admin interface without altering dialplan? |
| 19:27.12 | [TK]D-Fender | anujsingh: I wouldn't know and this isn't the place to ask. GUI's are NOT supported here. |
| 19:27.33 | dioedu | [TK]D-Fender, ok, than is more easy give up of this implementation with this gateway... |
| 19:27.35 | dioedu | right ? |
| 19:27.36 | dioedu | :P |
| 19:27.46 | [TK]D-Fender | dioedu: Pretty much... |
| 19:27.55 | dioedu | :( |
| 19:27.57 | dioedu | thanks |
| 19:28.02 | anujsingh | GUI's can I have the address for GUI's? |
| 19:28.05 | [TK]D-Fender | dioedu: time to decide on "Plan B" |
| 19:28.21 | [TK]D-Fender | anujsingh: HUH? |
| 19:28.41 | Nugget | I think that translates to "what irc channel can I join to sk about my GUI" |
| 19:28.44 | anujsingh | I mean any other channel ? |
| 19:28.46 | Nugget | er, ask |
| 19:29.13 | *** part/#asterisk mountainm2k (n=mountain@165.236.183.1) |
| 19:29.17 | anujsingh | Nugget is right, sorry i asked wrong . |
| 19:30.03 | anujsingh | what irc channel can i join to ask about GUI's? |
| 19:30.10 | [TK]D-Fender | anujsingh: No idea if they HAVE a channel. Go check hteir home-page to see what resources they offer |
| 19:30.32 | s34n | [TK]D-Fender: for instance, |secret| in the peer config will be used for incoming and outgoing auth? |
| 19:30.41 | anujsingh | ok |
| 19:31.00 | [TK]D-Fender | s34n: if that peer gets CALLED for incoming & outgoing, yes. |
| 19:31.41 | *** join/#asterisk kaigoh (n=kaigoh@82.133.70.150) |
| 19:31.50 | kaigoh | hi guys, need some advice |
| 19:31.54 | *** join/#asterisk rantsh (n=rantsh@190.36.185.139) |
| 19:32.03 | rantsh | hello everyone |
| 19:32.06 | dioedu | does nobody needs a megaco implementation ? |
| 19:32.27 | kaigoh | not sure if it can be done, but I am creating a web interface for voicemail and one of the features I want to implement is a call back facility |
| 19:32.33 | rantsh | hey anyone knows where I can get the latest (final) change history for asterisk 1.2 branch? |
| 19:32.37 | [TK]D-Fender | dioedu: MGCP is pretty much DEAD as far as * is concerned. |
| 19:32.42 | [TK]D-Fender | dioedu: You should avoid it. |
| 19:33.04 | [TK]D-Fender | kaigoh: Yes, its possible |
| 19:33.34 | kaigoh | i.e. you click a link on the webpage and your extension will ring and connect you to the person who called you. Any solutions? |
| 19:33.54 | dioedu | [TK]D-Fender, agree, but here in my country, telcos wanna use this to their implementations.... |
| 19:34.00 | [TK]D-Fender | kaigoh: Nothing "pre-build" that I know of, you'd have to design it yourself. |
| 19:34.25 | [TK]D-Fender | dioedu: You may be screwed then. Go find an MGCP gateway that can convert to something usable. |
| 19:34.28 | kaigoh | yeah, thats not an issue, I'm just not quite sure of the dial plan commands to use? |
| 19:35.09 | tzanger | CDR registered backend: Adaptive ODBC |
| 19:35.14 | tzanger | how do I disable that? |
| 19:35.17 | tzanger | cdr.conf has no mention of it |
| 19:35.20 | tzanger | res_odbc has no mention of it |
| 19:35.48 | [TK]D-Fender | kaigoh: Get reading : |
| 19:35.50 | [TK]D-Fender | ~book |
| 19:35.51 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 19:35.52 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^ |
| 19:37.26 | J4k3 | my trixbox either got semi-owned, or owned itself |
| 19:37.28 | J4k3 | ;) |
| 19:37.50 | J4k3 | of course, its trixbox, who'dathunkit |
| 19:37.54 | peanut- | maybe it's because you're a filthy socialist. |
| 19:38.12 | J4k3 | oh really, I'm not the one that sits in here begging for help |
| 19:38.27 | J4k3 | :P |
| 19:38.41 | peanut- | oh speaking of, I think I hate xten's soft phone |
| 19:38.44 | kaigoh | thanks for that D-Fender, thats the first route I always take, but i'm not to sure on the correct termiology, is it called call bridging? |
| 19:38.45 | J4k3 | HELPING IS A SOCIALIST ACTIVITY |
| 19:38.59 | J4k3 | its a softphone, theres a lot to hate. |
| 19:39.22 | peanut- | one-way audio, it's munging up RP ports it uses |
| 19:39.28 | peanut- | *RTP |
| 19:39.48 | [TK]D-Fender | kaigoh: nO, AND EVERYTHING DEPENDS ON how YOU SET THINGS UP. |
| 19:39.51 | J4k3 | eh, I just dislike softphones because Dell sucks at the IRQ |
| 19:39.58 | [TK]D-Fender | And my caps-lock NEVER gets stuck... |
| 19:40.15 | J4k3 | everything in this laptop is stuck on irq 11... usb, minipci wifi, ethernet, internal sound chip |
| 19:40.26 | *** join/#asterisk kaigoh (n=kaigoh@82.133.70.150) |
| 19:40.28 | J4k3 | all on 11, on an apic-capable laptop... dell rides the dumbass bus. |
| 19:40.37 | kaigoh | sorry d-fender, please repeat |
| 19:40.42 | mocker | [TK]D-Fender: Removed Trixbox *and* 10 Cisco phones from a site over the weekend. :P |
| 19:40.43 | [TK]D-Fender | peanut-: 1-way audio is a networking issue I've never pinned to the soft-phone before, and I use X-Lite / eyeBeam. |
| 19:40.50 | mocker | Replaced w/ Polycom and straight asterisk. |
| 19:40.55 | [TK]D-Fender | mocker: And then.....? :) |
| 19:41.00 | [TK]D-Fender | mocker: heh |
| 19:41.09 | peanut- | [TK]D-Fender: well I'm blaming the softphone, because all networking is good thus far. |
| 19:41.17 | mocker | Soon to replace TDM400P w/ A200D |
| 19:41.43 | peanut- | actually I haven't sniffed the client side yet, it's time consuming and I have work to do, but bitching is easy |
| 19:42.11 | [TK]D-Fender | peanut-: Yes, you mastered that part very quickly. |
| 19:42.25 | [TK]D-Fender | mocker: Wow, talk about doing everything wrong in sequence. |
| 19:42.36 | mocker | [TK]D-Fender: ? |
| 19:42.46 | mocker | All dependent on when things arrive from the shipping place. |
| 19:42.51 | [TK]D-Fender | mocker: Trixbox, Cisco phones, TDM400.... |
| 19:43.03 | mocker | Oh, Trixbox/Cisco/TDM400 was before me. |
| 19:43.11 | mocker | That's what I walked into. |
| 19:43.11 | peanut- | [TK]D-Fender: initially stun wasn't working on it, I didn't change anything and restarted the client a few times and it started working, hence my initial blame to the softphone |
| 19:43.24 | [TK]D-Fender | mocker: they should win a plushie or something for hitting a "consecutive mistake quota" in 1-shot |
| 19:43.29 | peanut- | not that just one restarting didn't do anything |
| 19:43.31 | mocker | I said, "You know, that has an IRC server on it?" |
| 19:43.40 | [TK]D-Fender | peanut-: don't NEED stun for a softphone behind NAT with * |
| 19:43.55 | generalhan | ok, i have been trying to get my queue calls to record into a certain directory for ease of labeling and i cant seem to get it to work. what i would like to do is specify that calls be placed into a directory reflecting the Agent that answers the queue. can anyone think of a way to make this happen ? apparently i cannot use variables like ${Agent} or ${EXTEN} in agents.conf |
| 19:44.13 | kaigoh | D-Fender, is the thing I am looking for called "call bridging"? |
| 19:44.18 | peanut- | [TK]D-Fender: until I enabled stun, asterisk server was sending RTP to 192.168.0.102 instead of its public IP. |
| 19:44.33 | peanut- | [TK]D-Fender: and nat=yes in it's sip.conf declaration |
| 19:44.43 | [TK]D-Fender | kaigoh: You are throwing dangerously vague terms around. Stop now, and get reading. |
| 19:45.21 | [TK]D-Fender | peanut-: "thats nice", you need to set a number of things. you'd have to show your full setup for proper commentary. |
| 19:45.52 | peanut- | [TK]D-Fender: ok, what configs do you want to see? |
| 19:46.15 | [TK]D-Fender | peanut-: sip.conf. [general] and [your-phone] masking only passwords |
| 19:46.17 | [TK]D-Fender | ~pb |
| 19:46.17 | jbot | A Pastebin is a place to paste your stuff 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 |
| 19:46.27 | peanut- | I know pastebin |
| 19:46.45 | Darthclue | generalhan, if you are using chan_agent, i don't think you can do that. |
| 19:46.51 | kaigoh | thanks |
| 19:46.57 | [TK]D-Fender | peanut-: I also know you're lazy so I figured give you a link you could likely click on would save one last bit of whining :p |
| 19:47.22 | *** join/#asterisk shtoom (n=godson@59.93.114.32) |
| 19:47.39 | [TK]D-Fender | generalhan: If you aren't on "ringall" then when you call an agent, you are inthe dialplan, you should be able to pick it up there. |
| 19:47.48 | [TK]D-Fender | generalhan: ${exten} that is. |
| 19:49.03 | *** part/#asterisk Egonis (n=roman@207.245.216.9) |
| 19:49.06 | generalhan | [TK]D-Fender: i AM using ringall ! how does that affect the logic ? |
| 19:49.18 | *** join/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 19:49.22 | *** part/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 19:49.32 | [TK]D-Fender | generalhan: well... hrm... actually... I suppose it really WOULDN'T affect... as long as ALL your members are using chan_agent. |
| 19:50.00 | generalhan | [TK]D-Fender: http://generalhan.pastebin.ca/754093 |
| 19:50.07 | generalhan | this is what i am working with currently |
| 19:50.58 | generalhan | i read several forum posts that said just by sepcifying the MONITOR_FILENAME before i enter the queue that, that would set the recording file name ... but it doesnt. The CLI shows the variable being set, but the calls do not go there |
| 19:51.18 | peanut- | [TK]D-Fender: messaged you link |
| 19:51.26 | *** join/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 19:51.37 | [TK]D-Fender | generalhan: -- agent_call, call to agent '7050' call on 'SIP/7010-9e820330' <- do the monitor inside your AGENT dialing dialplan. |
| 19:52.03 | generalhan | hmm |
| 19:52.48 | [TK]D-Fender | generalhan: Don't do monitoring on the QUEUE, do it in the local call to the agent. |
| 19:53.03 | *** join/#asterisk darkskiez (i=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 19:53.47 | generalhan | [TK]D-Fender: i appologize for my ignorance ... where is that handeled ? i thought that the Agent dialing was handled in the queues.conf file ?! |
| 19:53.48 | [TK]D-Fender | generalhan: Should look at the call as it comes in. |
| 19:54.03 | [TK]D-Fender | generalhan: strike that.. |
| 19:54.11 | [TK]D-Fender | blarg, window cross-over errors |
| 19:54.25 | [TK]D-Fender | generalhan: Agents get called through the DIALPLAN. |
| 19:54.46 | [TK]D-Fender | generalhan: Thats why you LOG IN <---- you should already know exactly where this is done in your setup... |
| 19:55.40 | generalhan | exten => 5557050,1,AgentLogin(7050) <-- thats is the only line in my entire dialplan that has any mention of an Agent |
| 19:56.17 | generalhan | so how would i tell the dial plan that whenever a call is answered by Agent 7050 that it should monitor ? |
| 19:56.46 | generalhan | right now i have it working without agents using a MASSIVE ammount of code in the dial plan ... i was hoping to get away from that using Agents ! |
| 19:57.37 | [TK]D-Fender | generalhan: OMG EW! |
| 19:57.48 | [TK]D-Fender | generalhan: You must be like the 2nd only person to do that. |
| 19:58.07 | generalhan | [TK]D-Fender: lol "ew" ? 2nd person to do what ? |
| 19:58.41 | ManxPower | generalhan: most people want their agents to be called by asterisk, not be forced to stay on the line to accept calls. |
| 19:58.48 | [TK]D-Fender | generalhan: "AgentLogin" <- this method of running queues |
| 19:59.00 | [TK]D-Fender | generalhan: as ManxPower said |
| 19:59.19 | generalhan | well yes, after i get this all worked out i want to change to agentcallback |
| 19:59.39 | ManxPower | Hell, we have enough trouble just keeping the damn receptionist at her desk instead of running around the office doing non-phone stuff. |
| 19:59.43 | [TK]D-Fender | generalhan: I'm guessing you WON'T get this to work without changin it NOW. |
| 20:00.00 | [TK]D-Fender | ManxPower: Time to "whip" out the BDSM gear :p |
| 20:00.07 | generalhan | ok ! |
| 20:00.11 | ManxPower | [TK]D-Fender: don't tempt me. |
| 20:00.30 | ManxPower | I have a pair of handcuffs that would work just fine to chain her leg to the desk. |
| 20:00.35 | generalhan | ok i will switch it over using the agentcallback first then ! |
| 20:00.53 | ManxPower | The better solution would be to fire her sorry ass, but that is not politically possible at tis time. |
| 20:01.16 | J4k3 | handcuffs at the desk? kinky |
| 20:01.19 | generalhan | the code that im using now to make this work is pretty crazy and i would really like to do this a different way ! http://generalhan.pastebin.ca/754105 |
| 20:01.30 | J4k3 | I've never used handcuffs, I'm more a rope or ribbon man |
| 20:01.35 | J4k3 | :D |
| 20:01.58 | [TK]D-Fender | J4k3: silk ties, les friction burn :) |
| 20:03.50 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 20:07.10 | mocker | ManxPower: Get her a DECT phone. |
| 20:07.11 | mocker | :P |
| 20:07.49 | *** join/#asterisk trippss (n=ss@ASA-ParksLuttrell.phonoscope.com) |
| 20:09.22 | *** join/#asterisk dlynes_ (n=dlynes@216.251.149.66) |
| 20:09.39 | _x86_ | mmm handcuffs |
| 20:09.59 | _x86_ | [TK]D-Fender: got it to work... helps if i use the correct feature code ;) |
| 20:10.13 | _x86_ | [TK]D-Fender: *2 did the trick, no hook flash trickery involved |
| 20:10.28 | J4k3 | DECT is hot. |
| 20:10.34 | [TK]D-Fender | _x86_: Well DTMF is a LAME way to do a transfer. |
| 20:10.46 | [TK]D-Fender | _x86_: Stop using channel-banks! |
| 20:10.53 | _x86_ | [TK]D-Fender: no! :) |
| 20:11.00 | J4k3 | ugh, channel banks |
| 20:11.02 | _x86_ | i agree it's ugly... |
| 20:11.11 | _x86_ | but the company i work for is fucking cheap |
| 20:11.15 | ManxPower | Only non-cool people use DTMF transfers |
| 20:11.21 | [TK]D-Fender | _x86_: CB + T1 card = PRICEY. |
| 20:11.22 | _x86_ | the sales people dont even have computers at their desks |
| 20:11.40 | [TK]D-Fender | _x86_: SIP ATA's & gateways are far cheaper |
| 20:11.48 | [TK]D-Fender | _x86_: And would spare you this BS |
| 20:12.17 | _x86_ | 24 port ATA cheaper than Rhino CB24-FXS? |
| 20:12.50 | ManxPower | _x86_: I think he means that 12 2-port adapters is cheaper than a CB + T-1 card for it. |
| 20:13.39 | _x86_ | but then you need a bigger data switch |
| 20:13.59 | [TK]D-Fender | _x86_: $1400 for 24 port Audiocodes / Mediatrix |
| 20:14.20 | [TK]D-Fender | _x86_: Cheaper if you use 3 X SPA-8000 |
| 20:14.33 | ManxPower | + $10,000 worth of time to try to figure out how to configure it. |
| 20:15.15 | _x86_ | ManxPower: that's like a whole month of my pay almost... i clearly didn't spend that much time on this ;) |
| 20:15.19 | generalhan | [TK]D-Fender: ok trying to setup AgentCallbackLogin but i need to beable to have the Agent dial the new extension but it will look to ${EXTEN}@default how can i have them dial the new extension and have it look to a different context ? cause i cant use ${CALLERID(number)} because that puts in an actual local phone number |
| 20:15.20 | [TK]D-Fender | _x86_: 780$ for the SPA solution. And far more versatile |
| 20:15.22 | _x86_ | perhaps.... 3 hours off and on? |
| 20:15.38 | _x86_ | [TK]D-Fender: how does it mount in a rack? |
| 20:15.57 | _x86_ | voipsupply.com doesn't have any spa-8000 |
| 20:16.08 | [TK]D-Fender | _x86_: http://www.8774e4voip.com/ProductDetails.asp?ProductCode=Linksys+SPA8000 |
| 20:16.50 | *** join/#asterisk jfitzgibbon (n=chatzill@64.72.237.187) |
| 20:16.56 | [TK]D-Fender | _x86_: plenty of other places offering it |
| 20:17.24 | [TK]D-Fender | generalhan: Read the instructions for that app.... |
| 20:17.56 | _x86_ | [TK]D-Fender: doesnt look like it mounts in a rack easily |
| 20:17.58 | mocker | So w/ that, does each port show up as a SIP user? |
| 20:18.21 | [TK]D-Fender | _x86_: Nope, for that you'd spend a few odd bucks on a TRAY |
| 20:18.33 | [TK]D-Fender | mocker: Yup |
| 20:18.39 | mocker | [TK]D-Fender: hot. |
| 20:18.47 | _x86_ | [TK]D-Fender: ugly, not a solution... |
| 20:19.02 | mocker | _x86_: It's a solution, but maybe not for you. :) |
| 20:19.03 | [TK]D-Fender | mocker: VERY. Its a real winner in the "economic deployement" category |
| 20:19.21 | k31th | cheap ass solution? |
| 20:19.34 | [TK]D-Fender | _x86_: Then $1400 it is. how much is a CB-24-FXS for you? |
| 20:19.49 | [TK]D-Fender | k31th: No... this isn't GRANDSTREAM :p |
| 20:19.57 | _x86_ | [TK]D-Fender: less than $1400 ;) |
| 20:20.10 | [TK]D-Fender | _x86_: how MUCH? Add the T1 costs on top.... |
| 20:20.20 | _x86_ | [TK]D-Fender: my cost was somewhere around $1100 each |
| 20:20.59 | _x86_ | I bought 20 of them (well, about 16 FXS, bought a couple FXO banks as well) |
| 20:21.03 | [TK]D-Fender | _x86_: And I''m sure you could get one of those 2 gateways for that easily as well in wholesale. I quoted RETAIL. Lets keep the apples away from the oranges, ok? |
| 20:21.10 | _x86_ | the cards were next to nothing too, because i bought bulk |
| 20:21.13 | generalhan | [TK]D-Fender: it doesnt specify how to change the context, without having a variable there. if the CID was setup that way in sip.conf i could use ${CALLERID(number)}@whatever_i_Want but without it i cannot put in (7050| @whatever_i_want so that they are prompted and then it goes to that context |
| 20:21.29 | _x86_ | [TK]D-Fender: dont attack my working solution, ok? :) |
| 20:21.35 | [TK]D-Fender | generalhan: yes, you CAN put the context into your app... |
| 20:21.54 | generalhan | with nothing in front of it so that it needs to be entered by the agent > |
| 20:22.01 | [TK]D-Fender | _x86_: Sure, you keep scraping the bottom of that barrel, and will cram you in with a pile of monkeys too... |
| 20:23.47 | ManxPower | generalhan: you will need to search the mailing list archives and the wiki for example of what you want to do |
| 20:24.08 | generalhan | ive already read up and down the wiki, so ... to the archives i go ! |
| 20:24.55 | ManxPower | we gave up on queues. |
| 20:25.11 | generalhan | lol, why's that ? |
| 20:25.33 | *** join/#asterisk bakermd (n=none@204.10.20.30) |
| 20:25.40 | ManxPower | far, far, far too complicated to make work the way we wanted. |
| 20:25.41 | *** join/#asterisk theron (n=theron@65.198.151.150) |
| 20:25.56 | bakermd | I am trying to get an * box going with realtime through odbc, but when I use isql to test my connection I am getting [unixODBC][TCX][MyODBC]Lost connection to MySQL server during query - any ideas |
| 20:26.01 | mocker | [TK]D-Fender: Sent the link to my boss. |
| 20:26.07 | mocker | Here's hoping I get a new toy to play w/! |
| 20:26.32 | [TK]D-Fender | mocker: Fun toys. impressive price point given SPA's WORK. |
| 20:26.49 | mocker | Yeah, I do wish they were rack mountable though. |
| 20:26.53 | mocker | But for the $$$ |
| 20:26.54 | [TK]D-Fender | mocker: Cost is on par with almost their cheapest ATA / port |
| 20:27.09 | [TK]D-Fender | mocker: Its a question of market segment... can't blame them... |
| 20:27.10 | Yourname`` | Holy crap! In the Aastra 480i, where do I put the Asterisk server's IP? Outbound proxy? Registrar proxy? or which one! |
| 20:27.17 | [TK]D-Fender | anyways... checkout time here, later all... |
| 20:27.19 | Yourname`` | So hard to even understand their UI. |
| 20:27.21 | mocker | later [TK]D-Fender |
| 20:27.31 | ManxPower | Yourname``: Yes! |
| 20:28.03 | Yourname`` | ManxPower: There is Proxy Server, then Backup proxy server, then outbound proxy server, and then registrar server!! lol |
| 20:28.12 | generalhan | well all my phones are defined in [extensions] i guess i could just use an include => default and move them all there ! lol then i wouldnt have to deal with this anymore ! |
| 20:28.39 | ManxPower | Yourname``: what happens when you put the ip address of your asterisk in all of them? |
| 20:28.45 | ManxPower | then what happens if you remove it from one of them? |
| 20:29.06 | ManxPower | Yourname``: You are empowered to try these things for yourself. |
| 20:29.34 | Yourname`` | ManxPower: I tried it, and nothing happened! :( |
| 20:30.39 | *** join/#asterisk geparkt (n=Werner@lx10-hrz.uni-duisburg.de) |
| 20:33.31 | *** join/#asterisk shawdog22 (n=shaw@pacman.oaklandcorp.com) |
| 20:33.55 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 20:35.10 | shawdog22 | Was wondering if anyone had a couple minutes for a question about changing the configs for a different ISDN? |
| 20:35.28 | Strom_M | ~ask |
| 20:35.29 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 20:36.08 | *** join/#asterisk Geijin (i=reaper@wbs-196-2-123-107.wbs.co.za) |
| 20:36.30 | bakermd | anyone had issues with isql to test the unixodbc? Getting [unixODBC][TCX][MyODBC]Lost connection to MySQL server during query |
| 20:36.49 | shawdog22 | I'm looking at moving from a NON-IDSN line to an ISDN line, to get ANI. I'm not sure what all I need to change to get the new T1 working. |
| 20:37.10 | Qwell | shawdog22: what was it before, regular old T1? |
| 20:37.13 | Strom_M | shawdog22: i assume you mean 'get caller ID on inbound calls' right? |
| 20:37.23 | shawdog22 | Correct. |
| 20:37.37 | ManxPower | ANI and Callerid are not the same |
| 20:37.52 | Strom_M | shawdog22: its generally just a matter of changing a few settings in zaptel and zapata.conf |
| 20:38.00 | ManxPower | shawdog22: callerid number arrives automatically normally. |
| 20:38.52 | Geijin | <---New to the asterisk thing and waiting for it to finish with compile to start using it |
| 20:39.13 | Qwell | ~welcome |
| 20:39.14 | jbot | It's great to be here! |
| 20:39.37 | Geijin | thanx Qwell |
| 20:39.43 | Qwell | ~book |
| 20:39.44 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 20:39.45 | Qwell | ~wikis |
| 20:39.45 | jbot | wikis is, like, http://www.voip-info.org |
| 20:39.57 | Qwell | Geijin: ^^ for you |
| 20:40.14 | Rawplayer | is ldap documented in that book? |
| 20:40.18 | Geijin | the last one i use on a daily basis but thanx :) |
| 20:40.21 | shawdog22 | I've got the general info on the new ISDN line, such as framing, encoding. But a little confused on the switch type and the signaling. |
| 20:40.35 | ManxPower | Rawplayer: since asterisk does not support LDAP, I suspect it is not in the Asterisk Book |
| 20:40.38 | Qwell | jbot: no, book is <reply> Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 20:40.39 | jbot | okay, Qwell |
| 20:40.45 | Qwell | Free == good |
| 20:40.47 | Rawplayer | ManxPower: excuse me? |
| 20:40.48 | shawdog22 | Info I have on the new line says NI2 Protocol and and 5E switch. |
| 20:40.52 | Rawplayer | ManxPower: its supported |
| 20:41.00 | ManxPower | Rawplayer: cite your source. |
| 20:41.03 | Rawplayer | or is it third party?:) |
| 20:41.26 | ManxPower | shawdog22: switchtype=national |
| 20:41.37 | jfitzgibbon | Rawplayer: it's not part of core or CVS - you have to download / compile / install it as a module |
| 20:41.41 | generalhan | ok SWEET ! got the recordings to work with the AgentCallbackLogin cmd. but this presented ANOTHER issue ... you think there is a way to set up my dialplan so that if an agent has logged into an extension and someone directly dials that extension that it wont ring? |
| 20:42.06 | shawdog22 | ManxPower: Do I need to change anything dealing with the 5E switch? |
| 20:42.13 | ManxPower | generalhan: um, that is usually a phone issue. |
| 20:42.16 | generalhan | not that this would ever happen, but lets say an agent logs into the bosses phone for a bit, we dont want that agent getting the bosses calls while he is on that phone |
| 20:42.19 | Strom_M | shawdog22: no, NI2 is NI2 |
| 20:42.29 | ManxPower | shawdog22: NI2 means you don't have to worry about the switch type. |
| 20:43.00 | jfitzgibbon | generalhan: stick a hint in ASTdb when AgentCallbackLogin returns, then check that hint before dialing the boss's phone? |
| 20:43.07 | Rawplayer | the word "ldap" is mentioned one time in the book |
| 20:43.15 | shawdog22 | ManxPower: How do I know what needs to be set for the signaling? |
| 20:43.28 | Strom_M | shawdog22: we told you already... switchtype=national |
| 20:43.33 | Strom_M | signalling=pri_cpe |
| 20:43.34 | generalhan | hmm, seems a bit complicated for this application, but if thats the best anyone can come up with ! |
| 20:43.57 | shawdog22 | Sorry didn't see it. |
| 20:44.19 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 20:44.28 | jfitzgibbon | generalhan: well, any simpler way of doing it would involve chan_agents sticking it's dirty hands into the guts of Dial(), which doesn't sound any less complex to me |
| 20:44.39 | shawdog22 | Thanks |
| 20:44.51 | generalhan | interesting way to put that ! lol |
| 20:45.16 | jfitzgibbon | generalhan: it just moves the complexity inside of the * code instead of the dialplan, which is not the direction things seem to be going (and this is a good thing) |
| 20:45.37 | ManxPower | shawdog22: http://www.fnords.org/~eric/etc.asterisk.zapata.conf |
| 20:46.14 | *** join/#asterisk dlynes_laptop (n=dlynes@216.251.149.66) |
| 20:46.19 | ManxPower | and http://www.fnords.org/~eric/etc.zaptel.conf |
| 20:46.31 | generalhan | jfitzgibbon: show agents displays the extensions that agents are logged into ... it just seems to me like this should be stored somewhere already that i can reference before doing a DIAL command |
| 20:46.31 | ManxPower | ignore the info about sangoma. The PRI in these configs is 8 channels |
| 20:46.47 | ManxPower | these are direct copies of the 2 config files of a production PBX |
| 20:47.13 | peanut- | anyone capable ot looking at my sip debug output and telling me why I only have one-way audio? |
| 20:47.20 | peanut- | http://crypto.ponybite.com/debug1.txt |
| 20:48.08 | jfitzgibbon | generalhan: well, if does stick it somewhere - chan_agent puts stuff in ASTdb as well if you have persistent agents enabled. But then you're parsing someone else's data which could (though probably won't, given the deprecation of AgentCallbackLogin) change |
| 20:48.09 | *** join/#asterisk docelmo (n=vircuser@c-76-99-153-242.hsd1.de.comcast.net) |
| 20:48.39 | generalhan | why is it being depreciated anyway ? |
| 20:48.47 | shawdog22 | ManxPower: Okay, thanks. Mine looks like that, just using channels 1-23. |
| 20:48.53 | generalhan | im just starting to see the bennefits |
| 20:49.14 | ManxPower | peanut-: audio is RTP |
| 20:49.32 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 20:49.40 | jfitzgibbon | generalhan: it seems that the direction * is taking is to move away from large apps that encapsulate lots of logic inside them and moving to lots of small apps that you glue together using the dialplan |
| 20:49.59 | ManxPower | peanut-: looks like a NAT issue. [Oct 29 15:28:40] VERBOSE[66027] logger.c: Peer audio RTP is at port 10.0.4.2:51000 |
| 20:50.13 | ManxPower | and most everything else in there is public routable IP addresses |
| 20:50.16 | shawdog22 | ManxPower: Is the timing source on the span variable usually set to 1? |
| 20:50.38 | ManxPower | shawdog22: you must have at least 1 span with a timing (sync) source set to 1 |
| 20:50.40 | peanut- | WIP300 is on 10.0.4.2, asterisk is on 10.0.4.6 |
| 20:50.42 | jfitzgibbon | generalhan: however, the migration from AgentCallbackLogin to dynamic queue members has been difficult for some because the documented example of how to do it was a) done in AEL and b) only one queue model, so other situations had to figure out how to translate their systems over |
| 20:50.59 | peanut- | and I have sip debug up there because it sets up RTP |
| 20:51.09 | Shaun2222 | can the microbrowser on the polycom 550's load images? |
| 20:51.13 | shawdog22 | ManxPower: Thanks for your help |
| 20:51.14 | ManxPower | peanut-: then were are all this 70.113.100.193 and 69.148.18.126 coming from? |
| 20:51.22 | generalhan | jfitzgibbon: well thats true too, i assume when they all together get rid of agentcallbacklogin that there will be more extensive documentation on the other forms ! |
| 20:51.38 | peanut- | the 70 address is outside NAT for asterisk/. the 69 address is client casey |
| 20:51.44 | peanut- | casey can only hear and not speak |
| 20:52.04 | mocker | ~grandstream |
| 20:52.05 | jbot | well, grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 20:52.07 | jfitzgibbon | generalhan: for all intents and purposes, they have gotten rid of it. The replacement is in the release version and the old code will just not be in 1.6 |
| 20:52.16 | mocker | Anyone have any web review of grandstreams? |
| 20:52.18 | ManxPower | peanut-: so exactly what device is not working? I suspect you used bindaddr= that usually screws up audio |
| 20:52.33 | jfitzgibbon | generalhan: so I wouldn't do a new deployment using it, nor would I hold my breath waiting for someone to come up with more docs |
| 20:52.35 | peanut- | bindaddr=0.0.0.0 I think |
| 20:52.47 | [hC] | peanut-: asterisk box natted? |
| 20:52.49 | peanut- | http://crypto.ponybite.com/sip.conf |
| 20:52.50 | ManxPower | you have debug from multiple devices. |
| 20:52.59 | ManxPower | Sorry, but *you* are supposed to do the work, not me. |
| 20:53.01 | peanut- | [hC]: yes, both asterisk and client are NAT'd |
| 20:53.15 | [hC] | peanut-: have you specified in sip.conf externip= and localnet= ? |
| 20:53.37 | generalhan | jfitzgibbon: well i cant ditch the queues, and i need to find better ways to keep tabs on them, so i really need this. i dont have much of a choice |
| 20:54.05 | jfitzgibbon | if this is a new part of your system, why don't you just do it with the non-deprecated bits? |
| 20:54.13 | ManxPower | peanut-: internal phones should connect to the internal address, external phones should connect to the external address. |
| 20:54.18 | jfitzgibbon | generalhan: why invest in something that limits your upgrade path? |
| 20:54.22 | peanut- | ManxPower: it does |
| 20:54.22 | generalhan | like what ? |
| 20:54.34 | *** part/#asterisk shawdog22 (n=shaw@pacman.oaklandcorp.com) |
| 20:54.35 | peanut- | ManxPower: internal phone connects to 10.0.4.6, external to the 70. address |
| 20:54.39 | ManxPower | peanut-: as I said, your pastebin is so confusing it is too much work. |
| 20:54.42 | jfitzgibbon | generalhan: dynamic queues using AddQueueMember / RemoveQueueMember. They didn't just say "queues are gone |
| 20:54.47 | [hC] | peanut-: have you specified externip and localnet in sip.conf? if you havent, this is your problem |
| 20:55.05 | mocker | Anyone recognize this phone? http://www.hybsys.bg/Products/VoIP/IP/Phones/5000/ |
| 20:55.20 | jfitzgibbon | generalhan: they provided modular replacements with (too little) docs and now you can make queues do all sorts of interesting things, though it's more work on the designer's end than the old way |
| 20:55.35 | Kwakwa | mocker, looks grandstreamy |
| 20:55.42 | generalhan | jfitzgibbon: actually if i dont go for a cig right now, i may start setting things on fire in here ! lol. if you have the time when i get back (3 min) i would love to hear your insights about dynamic queues. |
| 20:56.03 | peanut- | [hC]: I have specified it |
| 20:56.06 | jfitzgibbon | generalhan: if by insights, you mean "wild rants about how they suck", sure |
| 20:56.16 | jfitzgibbon | generalhan: not that I'm not using them mind you - it was just painful |
| 20:56.23 | peanut- | ManxPower: it's not confusing, there's the sip.conf and the sip debug |
| 20:56.23 | [hC] | peanut-: both, though? it wont work if you do only one. |
| 20:56.29 | generalhan | jfitzgibbon: any insight is better than no insight ! brb |
| 20:56.55 | peanut- | [hC]: http://crypto.ponybite.com/sip.conf and debug1.txt |
| 20:56.58 | ManxPower | peanut-: you have sip debug for two totally different phones |
| 20:57.10 | peanut- | it's the NAT phone calling the non-nat phone |
| 20:57.22 | ManxPower | canreinvite=no |
| 20:57.25 | peanut- | it is |
| 20:57.26 | ManxPower | that should solve the issue |
| 20:57.33 | peanut- | in [general] and client |
| 20:57.44 | *** join/#asterisk blq (n=Bl@dslb-088-067-042-033.pools.arcor-ip.net) |
| 20:57.58 | ManxPower | Is your Asterisk box also your NAT box? |
| 20:58.00 | peanut- | yes |
| 20:58.05 | *** join/#asterisk RailsAddict (n=scottbau@38.114.107.1) |
| 20:58.07 | [hC] | peanut-: pretty sure you just gave us your voicepulse username and pass... not that anyone cares. |
| 20:58.07 | peanut- | ports are forwarded |
| 20:58.09 | ManxPower | Then your asterisk box is not "behind nat" |
| 20:58.18 | ManxPower | don't forward the ports. |
| 20:58.19 | peanut- | [hC]: I commented it out |
| 20:58.26 | ManxPower | also, what ports are forwarded? |
| 20:58.53 | [hC] | doesnt look like it to me |
| 20:58.53 | [hC] | register => JNK45RpH45:UQX82Kry95@connect03.voicepulse.com |
| 20:59.04 | ManxPower | [hC]: it does to me |
| 20:59.05 | peanut- | ohnice. |
| 20:59.07 | [hC] | anyhow, do nat=yes on all the clients |
| 20:59.14 | peanut- | thanks for pasting that in the channel as well. |
| 20:59.26 | ManxPower | [hC]: is always helpful |
| 20:59.28 | [hC] | just change your password, you did it already by giving everyone the url. |
| 20:59.34 | peanut- | indeed |
| 20:59.52 | peanut- | it should be a goddamn variable there... |
| 20:59.54 | *** join/#asterisk agile (n=mike@38.114.107.1) |
| 20:59.58 | peanut- | why is it even hard coded |
| 21:00.00 | peanut- | abstards |
| 21:00.06 | agile | abstards! |
| 21:00.24 | peanut- | haha |
| 21:00.38 | [hC] | all ive ever had to do with people behind nat is nat=yes, externip, localnet |
| 21:00.53 | [hC] | and of course if a client is behind nat, make sure their phone/client supports nat properly |
| 21:00.56 | Qwell | peanut-: next time, play it as though that isn't your password, and quietly change it :) |
| 21:01.49 | *** join/#asterisk punkgode (n=punkgde@rev-200-40-119-222.netgate.com.uy) |
| 21:01.58 | ManxPower | [hC]: and forward ports, of course. |
| 21:02.13 | ManxPower | assuming your asterisk and your NAT box are NOT the same. |
| 21:02.26 | [hC] | Yep |
| 21:02.46 | ManxPower | I can see how forwarding ports and using localnet= and externip= when the asterisk box is also the NAT router could cause all sorts of audio issues. |
| 21:02.53 | [hC] | and the ports to forward are not limited to UDP 5060, you need UDP 10000-20000 as well |
| 21:03.01 | [hC] | it can, yeah. |
| 21:03.21 | ManxPower | [hC]: you need what ever is in /etc/asterisk/rtp.conf. If you don't have an rtp.conf, then asterisk defaults to 10000-20000 |
| 21:03.40 | *** join/#asterisk zydrunas (n=Miranda@216.49.228-ip-184.ckt.net) |
| 21:03.42 | [hC] | ManxPower, with the more specific answer ftw :) |
| 21:04.02 | ManxPower | in the case of Asterisk = NAT Box, then Asterisk is not actually "behind nat" |
| 21:04.28 | peanut- | Qwell: oh no, someone is gonna run up my long distance! |
| 21:04.50 | RailsAddict | Ay idea why files in /var/spool/asterisk/outgoing wounldn't get picked up? This was working before a reboot. |
| 21:05.04 | ManxPower | peanut-: don't worry too much. There are too many unsecure Asterisk boxes for you to worry about being targeted. |
| 21:05.20 | ManxPower | RailsAddict: a timestamp in the future would do it. |
| 21:05.24 | peanut- | I'm not concerned. |
| 21:05.29 | *** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-195-124.tx.res.rr.com) |
| 21:05.40 | peanut- | how long does it take for voicepulse to sync the new passwords across their servers? |
| 21:05.43 | ManxPower | if you rebooted and your box sync'd with NTP and the date/time was set back... |
| 21:05.53 | IOscanner | anyone else having problems with Cisco phones jacking up DST. |
| 21:06.35 | ManxPower | RailsAddict: date && ls -l /var/spool/asterisk/outgoing/* && daye |
| 21:06.38 | Shaun2222 | with the polycoms the microbrowser, how can it make it display a page while the phone is being used on a call? |
| 21:06.38 | ManxPower | date too |
| 21:08.07 | RailsAddict | ManxPower: ya, an hour off... |
| 21:08.09 | RailsAddict | thx |
| 21:09.58 | Darthclue | anyone use iprimus? good, bad, awful? |
| 21:10.10 | IOscanner | Is there a new image to fix it. |
| 21:11.44 | Qwell | IOscanner: move to AZ |
| 21:11.59 | IOscanner | sure packing now |
| 21:12.44 | generalhan | everyone should move to AZ !!! best state EVER ! |
| 21:12.50 | generalhan | lol |
| 21:14.24 | *** join/#asterisk duckz (n=duckz@81-31-157-49.vm.dnshosting.it) |
| 21:14.31 | bakermd | I have installed Asterisk --with-odbc and configured the ODBC driver - I want to run realtime over odbc (1.4.4) - but I cant find the script to create the db tables |
| 21:15.50 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 21:16.13 | Qwell | bakermd: there isn't one. look for examples on the wiki |
| 21:16.16 | Qwell | ~wikis |
| 21:16.16 | jbot | i guess wikis is http://www.voip-info.org |
| 21:16.30 | bakermd | cool - thx |
| 21:16.40 | WilliamK | whois mmichelson (nickname) ? |
| 21:16.48 | Qwell | WilliamK: putnopvut |
| 21:17.07 | Qwell | putnopvut: You should register that nick btw |
| 21:17.10 | Qwell | you can link it to yours |
| 21:17.38 | WilliamK | ah okie - I have a question... if removing this is really needed (Removing a completely unnecessary quota check from IMAP code. |
| 21:17.46 | WilliamK | ), what happens if the mbox is full? |
| 21:18.05 | WilliamK | and how does * handle it or does it cause bad affects? |
| 21:20.38 | putnopvut | As far as I can tell, the quota is checked elsewhere. That one spot was just a "test" according to the comment above it. |
| 21:20.47 | JT | peanut- is hilarious |
| 21:20.47 | JT | "i'm not concerned" |
| 21:20.59 | JT | "how long until voicepulse sync their password... guys?!?" |
| 21:21.04 | putnopvut | And if the mbox is full, then Asterisk will play the "vm-full" file and not allow messages to be left. |
| 21:22.49 | *** part/#asterisk fskrotzki (n=fskrot@host.textwise.com) |
| 21:23.19 | *** join/#asterisk Pons (n=pons@unaffiliated/pons) |
| 21:24.41 | *** join/#asterisk Pagautas (n=bigman@83.171.14.250) |
| 21:24.41 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 21:25.31 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 21:25.49 | *** join/#asterisk nroej (n=joern@heaven.cyphertext.de) |
| 21:25.49 | *** join/#asterisk TrentCreek (i=GeekBoy@cpe-70-117-207-168.rgv.res.rr.com) |
| 21:25.50 | nroej | hi |
| 21:26.06 | Pons | guys, i compiled trunk and installed chan_mobile.. I successfully received a call from the phone but when I make a call from a SIP softphone through the mobile phone i don't hear anything in both sides, but the call is successfuly connected. I also tried the softphone by making a call to a demo extension i made for this, and it works. I've also tried 2 different cellphones, no success. Any suggestions or advices on this? |
| 21:26.10 | nroej | tzafrir: hey |
| 21:26.19 | tzafrir | hi |
| 21:26.26 | nroej | tzafrir: you remember my problem yesterday |
| 21:26.27 | nroej | ? |
| 21:26.39 | tzafrir | not really |
| 21:26.44 | nroej | regarding meetme with ztdummy loaded |
| 21:26.55 | nroej | ... |
| 21:26.56 | tzafrir | on sparc? |
| 21:26.59 | nroej | yepp |
| 21:27.06 | tzafrir | ok |
| 21:27.42 | nroej | read the docs carefully again, the problem might be that my kernel is compiled with a 250hz rtc |
| 21:28.06 | nroej | hopefully it works then |
| 21:28.27 | tzafrir | rtc works on sparc? |
| 21:28.42 | nroej | it has the kernel option |
| 21:28.49 | tzafrir | kernel was built with 250Hz, and there's no RTC (probably) |
| 21:29.01 | nroej | hmpf |
| 21:29.05 | nroej | damnit |
| 21:29.23 | tzafrir | 1000 should work anyway |
| 21:29.32 | marl | can someone tell me which var would contain the called number within the dialplan outbound route? |
| 21:30.01 | nroej | crw-rw---- 1 root audio 10, 135 2007-10-27 21:47 /dev/rtc |
| 21:30.10 | nroej | the device is there... |
| 21:30.12 | JT | nroej: bit unreasonable to expect people here to remember what everyone else's problem was |
| 21:30.16 | JT | anyway |
| 21:30.18 | JT | give up |
| 21:30.20 | JT | it's sparc |
| 21:30.34 | JT | asterisk was designed to work properly on linux x86, that is all |
| 21:30.42 | nroej | JT: :P |
| 21:31.18 | nroej | JT: hey, come on he remembered after some hints |
| 21:33.09 | marl | i have an outbound dial command of : exten => _0[1-9].,n,Dial(IAX2/7875324@web-voip/44${EXTEN:1}), which works fine, but i cant seem to find a varable that contains just the outbound number |
| 21:34.12 | ManxPower | marl: in the example above what would the outbound number me? |
| 21:34.22 | ManxPower | ..er..be |
| 21:34.53 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:35.04 | _x86_ | re [TK]D-Fender |
| 21:35.39 | marl | when i initate a call via a softfone, say dialing 07432675475, the call goes through correctly (dial via web-voip as 447432675475) |
| 21:36.05 | ManxPower | because you are building the actual number INSIDE the Dial like 44${EXTEN:1} there is no variable that holds it. |
| 21:36.23 | marl | wat about the original number that was dialed? |
| 21:36.24 | [Blacky] | any1 knows how to enable the MESSAGE method to work in asterisk ? |
| 21:36.27 | ManxPower | ${EXTEN} is what was dialed to get the call into Asterisk. |
| 21:36.33 | ManxPower | the original number is ${EXTEN} |
| 21:36.40 | [Blacky] | to be used with sip softphones like eyebeam/x-lite ? |
| 21:37.00 | ManxPower | [Blacky]: thousands of people use X-lite without special "MESSAGE" configuration |
| 21:40.18 | [Blacky] | there's a built-in message you can send between your contacts in x-lite/eyebeam |
| 21:40.36 | [Blacky] | when i try to send such a message while connecting to an asterisk server, it gives an error |
| 21:40.42 | [Blacky] | did you try this before ? |
| 21:41.08 | marl | thanks ManxPower, was missreading the noop output from the * log :( |
| 21:41.08 | [Blacky] | "Instant Message" option, when you right click on a contact |
| 21:41.22 | JT | no, most people use instant messaging programs to send instant messages |
| 21:41.22 | ManxPower | Ah. I don't believe Asterisk supports that. |
| 21:41.38 | [Blacky] | i get "Error: Method Not Allowed." in the message box when i try to send a message |
| 21:41.39 | [Blacky] | oh |
| 21:41.41 | [TK]D-Fender | [Blacky], * does not support SIP IM's |
| 21:41.52 | ManxPower | [TK]D-Fender: not even in TRUNK? |
| 21:42.29 | [TK]D-Fender | ManxPower, maybe at most... maybe as a forward. |
| 21:42.30 | [Blacky] | ain't SMS messages works with the same method ? |
| 21:42.46 | ManxPower | [Blacky]: SMS and SIP IM are totally different. |
| 21:43.07 | ManxPower | that is like saying Saturn and Earth are the same. They are both planets, afterall. |
| 21:43.27 | [Blacky] | well, i tought i might be using the same method from the softphone client to the asterisk server |
| 21:43.37 | [Blacky] | but i understand it isn't now.. |
| 21:43.41 | ManxPower | SMS uses FSK modem tones to connect to the SMS Control Center to deliver the message. |
| 21:43.46 | *** join/#asterisk perd (i=[U2FsdGV@207.44.158.6) |
| 21:43.51 | JT | err |
| 21:43.59 | JT | modem to sms gateway might do that |
| 21:44.01 | JT | nothing else does |
| 21:44.20 | ManxPower | JT: um, what about the SMS application in Asterisk? |
| 21:44.24 | perd | anyone know the name of the log analyzer that makes pretty graphs and stuff for asterisk.. i forgot the name |
| 21:44.38 | JT | yes, that's modem to sms as far as i'm aware |
| 21:44.46 | [Blacky] | if i look at the OpenSER platform, they mention their a support for SMS, is it through a sip trunk? |
| 21:44.48 | JT | certainly not how the majority of smses are sent |
| 21:44.57 | JT | sip is not a trunk |
| 21:45.18 | [Blacky] | perd: Munin ? |
| 21:45.47 | ManxPower | [Blacky]: cite your source |
| 21:45.57 | perd | that's not the one im thinking of... but i'll check it out. need something flashy for the boss to look at heh |
| 21:46.06 | JT | most smses are sent with digital messages |
| 21:46.07 | ManxPower | I see an SMS module for OpenSER, but I didn't see a conversion module to convert SIP MESSAGE into SMS. |
| 21:46.17 | ManxPower | it requires a modem, of course. |
| 21:46.28 | [Blacky] | ok, i get it all connected now |
| 21:46.32 | [Blacky] | thanks for the info. |
| 21:47.05 | k31th | Blacky? |
| 21:47.10 | [Blacky] | k31th? |
| 21:47.15 | [Blacky] | EFnet ? |
| 21:47.15 | ManxPower | From OpenSER: This module provides a way of communication between SIP network (via SIP MESSAGE) and GSM networks (via ShortMessageService). Communication is possible from SIP to SMS and vice versa. |
| 21:47.29 | ManxPower | seems like a silly thing for a "PROXY" to do, but. |
| 21:47.40 | k31th | are you the same blacky i know from #debian ? |
| 21:48.00 | [Blacky] | hmm, nope.. |
| 21:48.07 | [Blacky] | but i know your nickname from somewhere as well |
| 21:48.08 | [Blacky] | so dunno |
| 21:48.23 | [Blacky] | basicbeats maybe |
| 21:48.27 | k31th | efnet is possible |
| 21:48.44 | k31th | basicbeat dj / music related? |
| 21:48.48 | [Blacky] | yeah |
| 21:48.48 | *** part/#asterisk agile (n=mike@38.114.107.1) |
| 21:48.51 | [Blacky] | i was running it |
| 21:49.04 | k31th | most likely then |
| 21:49.09 | [Blacky] | the radio station |
| 21:49.18 | [Blacky] | small world ;) |
| 21:49.54 | k31th | i'm interested, why the nick ? |
| 21:50.12 | [Blacky] | ManxPower: with their state, looks like it's possible to link between those two technics |
| 21:50.22 | [Blacky] | dunno, from BBS times |
| 21:50.31 | k31th | BBS? |
| 21:50.46 | JT | [Blacky]: technics? too much djing for you |
| 21:50.47 | ManxPower | [Blacky]: correct. Asterisk does not do this for SIP MESSAGE |
| 21:50.51 | [Blacky] | lol |
| 21:50.52 | k31th | lol |
| 21:51.00 | k31th | thats what i was thinking JT |
| 21:51.12 | [Blacky] | haha |
| 21:51.23 | peanut- | does asterisk cache passwords anywhere? changed my iax2 and sip passwords for voicepulse and I'm getting 'no authority' still |
| 21:51.26 | peanut- | can't login with them |
| 21:51.36 | k31th | link two CDJ's via SIP ? |
| 21:51.47 | dlynes_laptop | peanut-: did you do a sip reload and an iax2 reload? |
| 21:51.47 | marl | anyone know how i can get more info on the followme command and followme.conf files? only thing i can find so far is : http://www.voip-info.org/wiki/index.php?page=Asterisk%20cmd%20FollowMe |
| 21:51.50 | [Blacky] | rofl :D |
| 21:52.06 | peanut- | dlynes_laptop: yes |
| 21:52.13 | k31th | "show current djs" |
| 21:52.29 | [Blacky] | sip play cdj1 |
| 21:53.02 | k31th | haha |
| 21:54.41 | dlynes_laptop | peanut-: are you in control of both endpoints? |
| 21:55.07 | s34n | asterisk is sending a 407 out with To and From headers that seem reversed to me. From: Caller; To: * |
| 21:55.13 | peanut- | dlynes_laptop: it's a connection to voicepulse |
| 21:55.17 | ManxPower | marl: You did not check the CLI first! |
| 21:55.25 | ManxPower | marl: "show application followme" |
| 21:55.33 | ManxPower | THAT is how you should find application docs. |
| 21:55.37 | s34n | Aren't those headers backwards? |
| 21:55.46 | dlynes_laptop | peanut-: are you using the 'switch =>' statement to send calls? |
| 21:55.52 | *** join/#asterisk Bob_LobLaw (n=michaelc@fwsdo.projectdesign.com) |
| 21:56.50 | *** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au) |
| 21:59.35 | peanut- | dlynes_laptop: no |
| 22:00.48 | ManxPower | peanut-: removing the port forwarding and the localnet= and externip= did not help? |
| 22:02.33 | dlynes_laptop | peanut-: make sure that the extension you're sending it into is correct |
| 22:03.08 | dlynes_laptop | peanut-: what your understanding of voicepulse's dialplan is, and what reality is, might not match |
| 22:03.09 | peanut- | ManxPower: I'm trying to get my calls to work again after changing passwords.. |
| 22:07.48 | *** join/#asterisk PaulAviles (n=Miranda@dsl-7-36.cofs.net) |
| 22:08.10 | PaulAviles | can anyone assist compiling meetme? |
| 22:08.53 | ManxPower | PaulAviles: there is no assistance needed. MeetMe will compile when you build Asterisk -- IF -- it sees Zaptel already installed. |
| 22:09.38 | ManxPower | peanut-: try your old password. |
| 22:10.02 | PaulAviles | Max: that is the think., I guess I did not have it. so.. downloaded, did a make and make install and still nothing when I recompile asterisk |
| 22:10.22 | PaulAviles | I get no errors when I compile zaptel at all |
| 22:10.35 | ManxPower | and what about when you install zaptel? |
| 22:10.37 | ManxPower | "make install" |
| 22:10.46 | PaulAviles | done.. |
| 22:11.11 | ManxPower | if you built asterisk before zaptel was installed and you now have zaptel installed and ./configure does not find it, then you need to talk to a 1.4 person. |
| 22:11.46 | ManxPower | IIRC the configure script or the menuconfig does not pick up the new zaptel. |
| 22:12.06 | ManxPower | PaulAviles: what version of zaptel and what version of asterisk? |
| 22:12.59 | PaulAviles | I know why.. it recompiled zaptel but not in smp mode, let me try that |
| 22:13.12 | rantsh | Hi people |
| 22:13.54 | rantsh | I'm trying to monitor (record) agent calls using mixmonitor on asterisk 1.2.24, for some reason it's producing 2 files as if it was monitor |
| 22:14.25 | *** join/#asterisk techie (n=techie@76.214.23.171) |
| 22:14.27 | marl | ok, anyone used the followme app? i cant find a way to make it use a context other than Local for dialing outbound numbers, have found a couple of posts on it, but no solutions |
| 22:14.38 | rantsh | any clues on why could this be happening? |
| 22:14.51 | bakermd | if I do 'odbc show' I see 'Connected: yes' and I have an extensions table that it sees apparently 'Binding extensions to odbc/MySQL-asterisk/extensions' however it cannot find my extensions - ideas? |
| 22:15.11 | *** join/#asterisk doug (i=doug@zaxxon.telerama.com) |
| 22:15.17 | [TK]D-Fender | rantsh, first guess is a lack of SOX |
| 22:15.42 | doug | anyone know of a liveCD as a standalone voip client? |
| 22:16.15 | doug | barring that, what's the best Linux or BSD voip client? |
| 22:16.19 | doug | doesn't have to be graphical |
| 22:16.20 | rantsh | [TK]D-Fender: yup, there's no sox installed in this machine... didn't know mixmonitor used sox though |
| 22:16.50 | rantsh | [TK]D-Fender, very well, that explains it... I'll have to install it, thanks |
| 22:17.52 | [TK]D-Fender | doug, I think Ubuntu comes with Ekiga. |
| 22:18.07 | doug | ekiga? |
| 22:18.28 | doug | is that highly rated? |
| 22:18.44 | doug | like on xubuntu? which i'm told has the best livecd's.. |
| 22:20.15 | shtoom | Hi iam trying to compile zaptel 1.2.17 on ubuntu7.10 i am getting the following error |
| 22:20.16 | shtoom | /usr/share/vicidial_final/zaptel-1.2.16/ztd-eth.c: In function ‘ztdeth_rcv’: |
| 22:20.16 | shtoom | /usr/share/vicidial_final/zaptel-1.2.16/ztd-eth.c:95: error: ‘struct sk_buff’ has no member named ‘nh’ |
| 22:20.16 | shtoom | /usr/share/vicidial_final/zaptel-1.2.16/ztd-eth.c: In function ‘ztdeth_transmit’: |
| 22:20.16 | shtoom | /usr/share/vicidial_final/zaptel-1.2.16/ztd-eth.c:174: error: ‘struct sk_buff’ has no member named ‘nh’ |
| 22:20.18 | shtoom | make[2]: *** [/usr/share/vicidial_final/zaptel-1.2.16/ztd-eth.o] Error 1 |
| 22:20.20 | shtoom | make[1]: *** [_module_/usr/share/vicidial_final/zaptel-1.2.16] Error 2 |
| 22:20.22 | shtoom | make[1]: Leaving directory `/usr/src/linux-headers-2.6.22-14-server' |
| 22:20.23 | fujin_ | ~pb |
| 22:20.24 | jbot | A Pastebin is a place to paste your stuff 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 |
| 22:20.24 | [TK]D-Fender | doug, this is that magical time where you get off your ass and TRY THEM :p |
| 22:20.24 | shtoom | make: *** [all] Error 2 |
| 22:20.26 | shtoom | any ideas? |
| 22:20.32 | fujin_ | here's an idea |
| 22:20.33 | fujin_ | die in a fire |
| 22:20.37 | [TK]D-Fender | shtoom, Yeah, NEVER spam like that in here again |
| 22:20.47 | shtoom | I am sorry guys |
| 22:20.58 | doug | use pastebin shtoom |
| 22:21.03 | shtoom | accidently pasted more lines |
| 22:21.11 | [TK]D-Fender | shtoom, And that sure doesn't LOOK like 1.2.17 to me. |
| 22:21.54 | shtoom | oh 1.2.17 is the version of asterisk I am going to install |
| 22:22.45 | *** join/#asterisk techie (n=techie@76.214.18.225) |
| 22:23.29 | [TK]D-Fender | shtoom> Hi iam trying to compile zaptel 1.2.17 on ubuntu7.10 i am getting the following error |
| 22:23.30 | [TK]D-Fender | ^^^ |
| 22:23.39 | *** part/#asterisk zerohalo (n=zeroHalo@pool-72-70-78-28.bstnma.east.verizon.net) |
| 22:23.41 | *** join/#asterisk Edwin_Quijada (n=macaruch@190.94.11.95) |
| 22:23.43 | Edwin_Quijada | hI |
| 22:24.05 | *** join/#asterisk marl (n=marl@78.144.49.39) |
| 22:24.14 | Edwin_Quijada | I am a newbie with asterisk i have 4 ip phone working with * |
| 22:24.28 | Edwin_Quijada | internaly and 4 softphone |
| 22:24.49 | Edwin_Quijada | now we want add to go out calls to PSTN |
| 22:24.51 | marl | ok, sorry folks system crash :( did anyone reply to my followme dialing context question? sorry to ask again, but didnt see if there was any replys! |
| 22:25.04 | *** join/#asterisk mohsen (n=chatzill@213.233.160.50) |
| 22:25.10 | shtoom | D-Fender:ya that was a typo :( i am using combination of asterisk 1.2.17 and zaptel 1.2.16 |
| 22:25.26 | Edwin_Quijada | i have a voicepulse accouunt for this but i dont know how to do a trunk with it |
| 22:26.17 | [TK]D-Fender | shtoom, Well vicidial isn't supported here, and Ubuntu is going to make things that extra bit more difficult. Your odds are shinking by the minute |
| 22:26.57 | [TK]D-Fender | Edwin_Quijada, They have guide on their site. Go read them. They are one of the most popular providers for which tons of people have made guides. Google is your friend |
| 22:28.37 | shtoom | oh what a bad day for me :( #vicidial says asterisk 1.4 is not supported thats y I am in a process of replacing 1.4 with 1.2 im not a perl hacker other wise i would have modified vicidial files which #vicidial guys are reluctant to do |
| 22:28.57 | *** join/#asterisk marc7 (n=marc@S0106001c100a3e7c.gv.shawcable.net) |
| 22:29.49 | Edwin_Quijada | [TK]D-Fender: i downloaded the files from voicepulse but i dont know |
| 22:30.05 | ManxPower | It just sounds like a porn line. Talk to Vici for only $1.99/min. Call 1-800-VICI-DIAL |
| 22:30.06 | Edwin_Quijada | how to setup the trunk |
| 22:30.24 | [TK]D-Fender | Edwin_Quijada, they give you samples, TRY THEM. |
| 22:30.28 | ManxPower | Edwin_Quijada: we do not use the term "trunk" around here./ |
| 22:30.41 | Edwin_Quijada | ManxPower: why not? |
| 22:30.47 | Edwin_Quijada | it is wrong? |
| 22:30.58 | ManxPower | A trunk is a multiplexed UDP stream containing IAX2 packets from more than 1 call. |
| 22:31.16 | ManxPower | In telco terms "trunk" is "a single voice channel" |
| 22:31.37 | mohsen | How can one get the calling party peername and/or ip address? |
| 22:31.37 | Edwin_Quijada | so how is the correct term? |
| 22:31.37 | ManxPower | In #asterisk "trunk" means "I'm a retard." So we don't use that term around here. |
| 22:32.02 | ManxPower | If you are talking about sip.conf entries, the term is usually "SIP peer". |
| 22:32.08 | ManxPower | or SIP device |
| 22:32.29 | trippss | anyone successfully config a mediant to mediant T38 fax relay? |
| 22:32.45 | Edwin_Quijada | and if i want stream containing iax2 packets from more than 1 channel? |
| 22:32.53 | ManxPower | "IAX2 trunking" |
| 22:32.58 | Edwin_Quijada | how must i say? |
| 22:33.03 | Edwin_Quijada | ok |
| 22:33.06 | Edwin_Quijada | thks |
| 22:33.24 | rantsh | [TK]D-Fender do I need to get soxmix too? |
| 22:33.30 | ManxPower | Edwin_Quijada: Voicepulse's sample configs for Asterisk are reported to work. |
| 22:33.44 | *** part/#asterisk RailsAddict (n=scottbau@38.114.107.1) |
| 22:33.44 | ManxPower | I would assume you would get them from Voicepulse's website. |
| 22:33.50 | *** part/#asterisk Bob_LobLaw (n=michaelc@fwsdo.projectdesign.com) |
| 22:34.02 | *** join/#asterisk Bob_LobLaw (n=michaelc@fwsdo.projectdesign.com) |
| 22:38.03 | mohsen | I am working on a project which involves wring an AGI script for billing. I wonder how can I know the username of the calling party in the script. Any hint? |
| 22:38.53 | mohsen | in the prototype I am using ${CALLERID(num)}, but that must not be the right one |
| 22:39.19 | [TK]D-Fender | mohsen, perhaps you should look at the CHANNEL name... |
| 22:39.36 | Edwin_Quijada | ManxPower: maybe i dont know how |
| 22:39.44 | mohsen | [TK]D-Fender: but channel name differs from username (e.g. the sip username) |
| 22:39.58 | mohsen | or do you mean I should extract it from there? |
| 22:40.06 | ManxPower | ~trunk |
| 22:40.06 | jbot | i heard trunk is In Asterisk a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call", in telecom a trunk is a "single voice channel connecting to the CO." There is no such thing as a "SIP Trunk". Don't use the term. |
| 22:40.06 | [TK]D-Fender | mohsen, Do the math :) |
| 22:40.27 | ManxPower | now it is there for future reference, Qwell [TK]D-Fender |
| 22:40.49 | mohsen | [TK]D-Fender: Doing the math to get the user id does not feel like a genuine solution to me :) |
| 22:41.45 | ManxPower | mohsen: read README.variables (or whatever they call it in 1.4) |
| 22:42.01 | [TK]D-Fender | mohsen, sorry, I'm not here to support your illusions of "genuine" :) |
| 22:42.30 | mohsen | [TK]D-Fender: So take your time on anything you like :) |
| 22:43.17 | mohsen | ManxPower: I have checked them here http://www.voip-info.org/wiki/view/Asterisk+variables and could not find a good candidate |
| 22:43.28 | ManxPower | mohsen: the Wiki is always very out of date. |
| 22:43.34 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 22:43.51 | ManxPower | mohsen: in fact, I believe that page is for Asterisk 1.0 |
| 22:44.00 | fujin_ | awesome |
| 22:44.43 | ManxPower | mohsen: they may have named the file something else, but it should be fairly obvious. |
| 22:45.09 | mohsen | okay. |
| 22:45.17 | Edwin_Quijada | i need a card with 2 analog lines |
| 22:45.19 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 22:45.19 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 22:45.31 | Edwin_Quijada | which card do you recommend me? |
| 22:45.34 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 22:45.44 | mohsen | channelvariables.txt |
| 22:46.17 | ManxPower | mohsen: also "show applications like sip" in the Asterisk CLI |
| 22:47.46 | [TK]D-Fender | ManxPower, getting warmer ;) |
| 22:48.04 | ManxPower | [TK]D-Fender: it should whine and tell him the correct command. |
| 22:48.28 | [TK]D-Fender | ManxPower, lol.... * nowhere near so kind :) |
| 22:48.47 | ManxPower | it does for some of them |
| 22:49.27 | [TK]D-Fender | mohsen, "show functions like SIP |
| 22:49.28 | [TK]D-Fender | " <---- welcome to 1.2+ |
| 22:50.52 | ManxPower | that matches all 39 functions in 1.2.latest |
| 22:51.13 | [TK]D-Fender | ManxPower, matches 4 in 1.4 |
| 22:51.48 | [TK]D-Fender | of course "show function CHANNEL" might be a hint as well :) |
| 22:53.38 | ManxPower | [TK]D-Fender: are the sorted in 1.4 too? |
| 22:53.43 | mohsen | Alright, there is SIPCHANINFO function which does it |
| 22:54.33 | mohsen | another solution is ${ACCOUNTCODE} which I guess is technology independent |
| 22:54.44 | ManxPower | Yup. |
| 22:54.52 | mohsen | Thank you :) |
| 22:58.05 | *** join/#asterisk felix_da_catz (n=felix@c-76-31-59-80.hsd1.tx.comcast.net) |
| 23:01.33 | *** join/#asterisk NovceGuru (i=shelby@ballmung.easymac.org) |
| 23:01.38 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 23:07.18 | marl | can anyone tell me, if i have 2 contexts listed in extensions.conf that are named the same, will * combine them together? am combining conf files together, and wanted to know if this would work? |
| 23:07.37 | PaulAviles | hey guys, when I compile zaptel it creates the zaptel.ko inside /lib/modules/2.6.9-55.0.9.EL/extra |
| 23:08.10 | PaulAviles | and it is supposed to be inside 2.6.9-55.9.9.ELsmp |
| 23:08.29 | Qwell | PaulAviles: Did you install the correct kernel sources package? |
| 23:08.32 | PaulAviles | yes |
| 23:09.32 | PaulAviles | stock centos so I did a yum install kernel-devel |
| 23:10.03 | Qwell | centos/rh likes to mess up a lot with that |
| 23:10.11 | [hC] | marl: im not sure, but you can try then do a show dialplan and see what asterisk interpreted |
| 23:10.24 | *** join/#asterisk saftsack (n=saftsack@pD9E06356.dip.t-dialin.net) |
| 23:11.18 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 23:11.50 | *** join/#asterisk trippsss (n=ss@ASA-ParksLuttrell.phonoscope.com) |
| 23:12.10 | grandpapadot | Hi all, for the HPEC, how does /usr/sbin/zaphpec_enable initially get called by zaptel? On module load? What kind of permissions does that file need? Owned by root or my asterisk user/group? |
| 23:12.13 | *** join/#asterisk craigk (n=ckowald@58.174.122.198) |
| 23:13.05 | mohsen | h323, the asterisk's stock one, seems to only support plain text secrets. Is that correct? |
| 23:13.56 | *** join/#asterisk Freman (n=freman@brdr-gw-01.benon.com) |
| 23:14.03 | peanut- | [TK]D-Fender: you around? |
| 23:14.24 | Freman | hey guys, how do I for a call placed on a zap channel to use a different codec? |
| 23:17.33 | [TK]D-Fender | peanut-, barely |
| 23:17.47 | [TK]D-Fender | Freman, Zap doesn't use codecs. it isn't voip. |
| 23:18.03 | peanut- | [TK]D-Fender: if you have a chance, http://crypto.ponybite.com/debug1.txt |
| 23:18.31 | Freman | yes, but I've got a phone plugged into a zap interface, and when I dial out over it it sends to my provider as ulaw I want gsm |
| 23:19.37 | [TK]D-Fender | Freman, you need to fix your SIP PEER <---- |
| 23:20.18 | *** join/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 23:20.26 | rhombus | Any Aastra users out there? |
| 23:20.35 | Freman | but that effects the other phones doesn't it? |
| 23:20.37 | rhombus | I can't get my 480i to register. |
| 23:20.42 | grandpapadot | rhombus: Some, what's ur q? |
| 23:20.56 | grandpapadot | rhombus: pastebin your sip.conf |
| 23:21.02 | rhombus | okay |
| 23:21.50 | rhombus | grandpapadot: how about sip debug output instead? |
| 23:22.00 | grandpapadot | Both? |
| 23:22.04 | rhombus | sure, okay |
| 23:23.16 | [TK]D-Fender | Freman, Your peer determines how you call your provider. |
| 23:23.17 | *** join/#asterisk knarfly (n=vladimir@c-75-74-155-198.hsd1.fl.comcast.net) |
| 23:24.17 | grandpapadot | Freman: SIP or IAX? |
| 23:24.45 | Freman | I want the zap to make calls in gsm, I want all my other phones to use g729 |
| 23:24.48 | marl | [hc] it would appear that it does combine the contexts :) thanks |
| 23:24.50 | Freman | peer is iax |
| 23:24.54 | [hC] | marl: good to know :) |
| 23:25.09 | grandpapadot | pastebin your iax.conf |
| 23:25.10 | JT | Freman: most ITSPs don't support GSM |
| 23:25.20 | Freman | most don't |
| 23:25.22 | Freman | mine does |
| 23:25.25 | grandpapadot | And does your provider support gsm? |
| 23:25.29 | *** join/#asterisk Darthclue (n=Darthclu@adsl-75-50-243-110.dsl.snantx.sbcglobal.net) |
| 23:25.31 | grandpapadot | Oh, what JT said. |
| 23:26.08 | JT | Freman: what do your allow= and disallow= lines say in iax.conf? |
| 23:26.26 | s34n | when a peer is set with a username and secret, any calls coming from it must be authenticated, right? |
| 23:26.44 | Freman | disallow=all allow=g729,gsm,ulaw,alaw |
| 23:26.46 | s34n | so * sends it a 407 in response to an INVITE, right? |
| 23:27.07 | JT | Freman: disallow=all |
| 23:27.10 | JT | allow=gsm |
| 23:27.44 | rhombus | grandpapadot: here's my sip.conf http://pastebin.ca/754395 |
| 23:28.04 | Freman | but doesn't that stop my ip headsets from using g729? |
| 23:28.05 | rhombus | and here's my sip debug ip output: http://pastebin.ca/754389 |
| 23:28.19 | grandpapadot | aastra_test? |
| 23:28.34 | rhombus | yes |
| 23:28.38 | s34n | ManxPower: (is 'peer' a better word than 'trunk'?) |
| 23:28.51 | [TK]D-Fender | Freman, ok, last time : Zap has NOTHING to do with chosing the CODEC of your VoIP call! |
| 23:29.07 | rhombus | TK]D-Fender |
| 23:29.10 | rhombus | whoops. |
| 23:29.11 | rhombus | sorry. |
| 23:29.12 | [TK]D-Fender | Freman, That is determined by your IAX or SIP peer as appropriate. |
| 23:29.47 | peanut- | peer 'casey' doesn't have the address 70.113.100.193, but instead it's 69.148.18.126, why does it show this way on 'sip show channels'? 10.0.4.2 WIP300 0d136b240bd 00102/00000 ulaw No Tx: ACK |
| 23:29.48 | [TK]D-Fender | s34n, Not as complete a definition. |
| 23:29.48 | peanut- | 70.113.100.193 casey NzMxZGZiMGQ 00101/00002 ulaw No Rx: ACK |
| 23:30.20 | peanut- | is it the NAT my asterisk box is behind breaking? |
| 23:30.45 | grandpapadot | rhombus: Anything in between the phone and the asterisk server? firewall, proxy, etc? |
| 23:30.46 | s34n | [TK]D-Fender: I've simplified my problem for now to 3 SIP messages |
| 23:31.12 | Freman | the zap is doing the dialing, the asterisk is converting that call to voip and sending it to my vsp, I want my digital handsets to use g729 (because they can and the vsp supports it, so passthru works) and my analog to have it's calls through asterisk passed as gsm |
| 23:31.13 | rhombus | grandpapadot: no, these are on the same LAN. |
| 23:31.14 | s34n | [TK]D-Fender: An INVITE, a 407, and an ACK <--- that is the complete session |
| 23:31.51 | grandpapadot | rhombus: Looks like a simple auth issue to me. |
| 23:32.07 | grandpapadot | Pastebin your <mac>.cfg for that phone. |
| 23:32.38 | s34n | [TK]D-Fender: So my problem has to be in the 407, or in the ITSP handling of the 407 |
| 23:32.42 | [TK]D-Fender | Freman, "converting" BLAH! You just don't get it. If your provider uses G.729 as a preference that peer will ONLY choose G.729 and you're SCREWED because thats your FIRST CHOICE. |
| 23:33.04 | peanut- | shouldn't Theoretical address and Reported address be the same? |
| 23:33.10 | [TK]D-Fender | Freman, Make. Another. PEER <--------------------- |
| 23:33.12 | peanut- | s/reported/received |
| 23:33.22 | grandpapadot | rhombus: or are you manually configuring? |
| 23:33.54 | rhombus | grandpapadot: yeah, first via the web UI then via the phone UI |
| 23:34.03 | rhombus | I normally use Polycom sets |
| 23:34.08 | rhombus | this is my first Aastra |
| 23:34.41 | grandpapadot | .cfg files are braindead simple on the aastra, much easier than on polycom (I still prefer polycom, though) |
| 23:34.46 | s34n | [TK]D-Fender: the 407 looks strange to me |
| 23:34.58 | grandpapadot | rhombus: Check your phone config, looks like an auth issue. |
| 23:35.14 | rhombus | okay. |
| 23:35.15 | s34n | [TK]D-Fender: the headers say it is From: itsp; To: * |
| 23:35.22 | rhombus | maybe I should use a name without an underscore? |
| 23:36.01 | grandpapadot | I don't think that would matter, I use something.ext, i.e., foo.801 as my peer names. |
| 23:37.13 | grandpapadot | Did you configure through the aastra http server or through the phone menu directly? |
| 23:39.30 | grandpapadot | Anyone know of a 1.2.24 patch that will update voicemail passwords in static real-time before I go hacking up the source with my shitty C skills? |
| 23:39.50 | rhombus | grandpapadot: through the http server |
| 23:40.07 | grandpapadot | rhombus: open me up a port to that phone's http server and let me take a look |
| 23:40.17 | rhombus | grandpapadot: okay |
| 23:42.06 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:44.40 | peanut- | anyone know what the "Theoretical Address" in "sip show channel xxxx" is for? |
| 23:46.15 | [TK]D-Fender | ok, stepping out for a while. |
| 23:46.24 | peanut- | yarg. |
| 23:47.08 | *** part/#asterisk Pons (n=pons@unaffiliated/pons) |
| 23:49.10 | *** part/#asterisk doug (i=doug@zaxxon.telerama.com) |
| 23:52.28 | rhombus | grandpapadot: |
| 23:52.30 | rhombus | oh damn. |
| 23:52.44 | rhombus | Any other Aastra users out there? I just got the port opened. |