| 00:01.06 | beek | If I plug in an analog phone instead of the * box to the VM port I can easily do the transfer. Asterisk does pick the call up and play the greeting. |
| 00:01.13 | fujin_ | dijungal: tshark |
| 00:02.42 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 00:03.02 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 00:03.34 | *** join/#asterisk chad (n=chad@c-67-174-112-139.hsd1.co.comcast.net) |
| 00:04.17 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) |
| 00:04.21 | jim9119 | Packet sniffer: http://www.wireshark.org/ |
| 00:04.21 | chad | I'm reading through the internals trying to figure out how Asterisk knows when an incoming AGI command is done since it doesn't seem to be terminated in any way. Any pointers to where in the code to look for that? |
| 00:06.19 | jim9119 | or if you want overkill setup http://www.snort.org/ |
| 00:06.35 | *** join/#asterisk mindCrime_ (i=chatzill@nat/redhat/x-6ed25639506a4d40) |
| 00:07.50 | CrazyTux | Hey guys say I want to match: Host: test.somehost.com, I could also do just Host: *.somehost.com ? |
| 00:09.48 | *** join/#asterisk craigk (n=ckowald@58.174.122.198) |
| 00:10.29 | jim9119 | does anyone here have a cluster setup with UltraMonkey or Linux-HA? |
| 00:12.35 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 00:12.35 | *** mode/#asterisk [+o angler] by ChanServ |
| 00:15.12 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:20.05 | MacWinner | is there a standard way that I can configure a phone number on my cellphone that automatically includes sending DTMF after the call is connected? ie, like dialing 5551212##1234 |
| 00:20.40 | beek | My cell phone accepts commas (,) as a pause. So I'd use: 5551212,,,,1234 |
| 00:20.59 | *** join/#asterisk jmesquita (n=jmesquit@201.20.243.195.user.ajato.com.br) |
| 00:23.37 | MacWinner | beek, cool, is that normal across cellphones? |
| 00:24.14 | MacWinner | beek: and does the pause including the time the phone is ringing? or do the pauses only take effect if the phone is picked up |
| 00:28.19 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-ea6b33ae92da3b0d) |
| 00:31.46 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 00:39.09 | *** join/#asterisk tristanbob (n=tristan@oalug/member/tristanbob) |
| 00:40.43 | *** join/#asterisk Thazza (n=me@eth767.nsw.adsl.internode.on.net) |
| 00:41.03 | Thazza | Hey all. |
| 00:41.44 | saint_ | hey |
| 00:41.53 | saint_ | does anyonw know how the DIGITMAP on Polycom works ? |
| 00:42.15 | beek | MacWinner: It has been that way on my Nokia and Motorola phones. The "," was used in modem dialing strings and I think that the cell phone mfgs simply copied that standard. |
| 00:43.49 | J4k3 | , means a half second pause iirc. |
| 00:43.51 | J4k3 | or 1 second |
| 00:43.52 | J4k3 | thats it |
| 00:44.05 | Thazza | Is is possible to cause to sip channels to ring and bridge from the CLI? |
| 00:46.03 | MacWinner | beek: cool.. another question, if I call my PBX from my cellphone and then the PBX initiates a call to a 3rd party, can i make my PBX spoof the outgoing call's CID to look like my cell phone's CID? |
| 00:51.00 | *** join/#asterisk tristanbob (n=tristan@oalug/member/tristanbob) |
| 00:52.01 | *** join/#asterisk techie (n=techie@adsl-76-240-178-179.dsl.lsan03.sbcglobal.net) |
| 00:52.13 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:55.25 | beek | MacWinner: That depends on your phone company. |
| 00:59.20 | *** join/#asterisk BiG^DoG (n=BiG^DoG@c-67-162-233-20.hsd1.de.comcast.net) |
| 00:59.34 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-227-239.dsl.irvnca.pacbell.net) |
| 00:59.36 | BiG^DoG | anyone successfully gotten call waiting to work with asterisk and an SPA-3102? |
| 00:59.52 | BiG^DoG | I've read every page I can find about passing hook flash to the PSTN but I just can't get it to work |
| 01:01.59 | *** join/#asterisk Mercestes (n=Merceste@c-76-30-153-176.hsd1.tx.comcast.net) |
| 01:11.20 | phix | oh there are problems with the SPA 3102? I am about to buy one :) |
| 01:12.48 | phix | I hate waiting in queues when ringing up tech support, etc, I would like to transfer a call to some extension which will repeat a message saying "Press 1 when I am no longer in the queue", and get it to ring my extension back and pass the call to me |
| 01:15.17 | phix | :) |
| 01:16.56 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:20.18 | J4k3 | phix: I wish they'd just take your number and *really* call you back |
| 01:21.05 | phix | J4k3: true, some places do that no |
| 01:21.05 | phix | now |
| 01:21.27 | phix | if the queue is uber they say they will ring back, but most dont do that so I need to transfer them |
| 01:21.33 | phix | transfering I don't have a problem with |
| 01:21.59 | phix | getting asterisk to ring my extension again I do have a problem with, how would this be done? using a queue or something? |
| 01:22.07 | phix | is it even possible? |
| 01:22.16 | phix | or do I need to do some AGI stuff |
| 01:22.48 | JT | phix: i wouldn't call it a problem |
| 01:23.12 | JT | BiG^DoG: don't expect hook flash to work over sip |
| 01:24.45 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 01:25.07 | phix | JT: awww |
| 01:25.30 | phix | JT: that is good |
| 01:25.34 | phix | JT: any suggestsions/. |
| 01:25.35 | phix | ? |
| 01:28.11 | *** join/#asterisk jmacz (n=jmacz@201.244.174.187) |
| 01:29.22 | phix | JT: ! |
| 01:29.35 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 01:29.47 | puzzled | hi |
| 01:35.11 | phix | puzzled: hi |
| 01:37.13 | BiG^DoG | JT: Are you saying I can't transfer a hook flash from a FXS port on the SPA3102 to the FXO port? |
| 01:38.49 | Mercestes | BiG^DoG, I think the SPA intercepts that hookflash and does it's own magic. Is there a setting for "hook flash" under the web admin? |
| 01:39.21 | BiG^DoG | there is a setting but I'll be damned if I can get it to do anything |
| 01:41.53 | Mercestes | Might want to try app Flash() |
| 01:42.03 | JT | BiG^DoG: probably not |
| 01:42.07 | JT | Mercestes: isn't that zap only? |
| 01:42.20 | Mercestes | Yea. |
| 01:42.23 | Mercestes | He's on an FXO. |
| 01:42.32 | Mercestes | I guess if he's trying to answer call waiting it won't help him much. |
| 01:42.41 | Mercestes | But if he wants to hookflash a call out he can use Flash to do a transfer. |
| 01:42.50 | BiG^DoG | no, it was for call waiting |
| 01:42.55 | BiG^DoG | I'm trying to handle call waiting on my analog line |
| 01:42.56 | Mercestes | OH, then flash wont' help you. |
| 01:43.53 | Mercestes | I think "call waiting" should be one of the options under hookflash tho. |
| 01:44.05 | Mercestes | but it's kind of up to the SPA to handle that correctly. |
| 01:44.45 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 01:44.59 | BiG^DoG | should I attempt linksys support? |
| 01:46.44 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 01:47.06 | xpot | anyone know if IAXtel is down for good or what? |
| 01:48.00 | *** join/#asterisk pcooper (n=phatlip@60-242-220-197.static.tpgi.com.au) |
| 01:49.30 | phix | who? |
| 01:49.39 | phix | JT! |
| 01:55.03 | JT | Mercestes: he's on an SPA |
| 01:55.06 | JT | which is SIP |
| 01:55.11 | JT | Not zap |
| 01:55.15 | JT | to asterisk |
| 01:55.35 | Mercestes | <BiG^DoG> JT: Are you saying I can't transfer a hook flash from a FXS port on the SPA3102 to the FXO port? |
| 01:55.49 | Mercestes | sorry, I thought he was running from FXS -> FXO. |
| 02:01.44 | phix | JT: |
| 02:02.36 | phix | Is it possible to handle incomming calls differently based on caller id? |
| 02:02.53 | phix | actually, nm, I think I remember reading that in "The Book" |
| 02:03.38 | puzzled | phix, sure. in english: if callerid is 12345 do something |
| 02:04.08 | phix | http://newd2event.net/img/hacks/PseudoResolution.jpg |
| 02:04.17 | phix | oops |
| 02:04.18 | phix | exten => 123,1,GotoIf($[${CALLERIDNUM} = 8885551212]?20:10) |
| 02:04.21 | phix | I meant that :) |
| 02:04.37 | phix | stupid windows, not copying things to clipboard when I highlight them |
| 02:04.56 | *** part/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 02:05.54 | JT | phix: that's the dumb way to do it :) |
| 02:06.03 | JT | use callerid extension matches |
| 02:06.40 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 02:06.41 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 02:06.57 | tzanger | hmm |
| 02:07.10 | tzanger | I want my ztdeth spans positioned before my wctdm span |
| 02:07.25 | tzanger | but I can't run ztcfg because it will bomb out my rc.local script before I can load wctdm |
| 02:07.29 | tzanger | wtf |
| 02:09.00 | *** join/#asterisk VJFROMGT (n=vjfromgt@user-387g9sf.cable.mindspring.com) |
| 02:09.00 | VJFROMGT | client is trying to dial a number but gest a busy |
| 02:09.00 | VJFROMGT | http://pastebin.ca/766705 |
| 02:09.00 | VJFROMGT | can anyone tell me hwat is goign |
| 02:10.38 | [hC] | anyone have any thoughts why an IAX2 based call would sound bad, whereas simply changing it over to SIP makes the call sound perfectly fine? |
| 02:10.57 | VJFROMGT | iax2 between what? |
| 02:11.01 | puzzled | [hC], maybe no ztdummy or other zaptel driver loaded |
| 02:11.13 | [hC] | VJFROMGT: two asterisk boxes? |
| 02:11.18 | [hC] | puzzled: what does that have to do with it? |
| 02:11.22 | Mercestes | VJFROMGT, busy, or reorder? |
| 02:11.38 | VJFROMGT | merce,, beep, beep , beep |
| 02:11.48 | puzzled | [hC], for some iax stuff you need to have a zaptel driver loaded for it to work properly |
| 02:11.58 | [hC] | puzzled: i dont think that is correct. |
| 02:12.06 | puzzled | [hC], iirc with trunking or meetme |
| 02:12.09 | VJFROMGT | hc,, i use iax2 to link boxes all the time, no zaptel involved |
| 02:12.35 | puzzled | obviously meetme has nothing to do with iax but does not a zaptel driver for timing |
| 02:12.38 | [hC] | puzzled: not using trunking, and thats because meetme transodes to slin, im not talking about meetme though, im talking a point to point call, simply traversing IAX as opposed to SIP |
| 02:12.39 | VJFROMGT | hc, in ure iax trunk, what codec is specified |
| 02:13.03 | [hC] | g729, for both |
| 02:13.06 | puzzled | [hC], maybe your jitterbuffer is going nuts? |
| 02:13.19 | VJFROMGT | how about iax.conf file, is codec enabled? |
| 02:13.29 | [hC] | guys thanks for the effort but im looking for some more lower level reason as to how IAX transmits packets different than SIP, specifically in relation to say, latency spikes or packet loss |
| 02:13.46 | [hC] | VJFROMGT: what does that mean, is codec enabled? of course there's a codec. |
| 02:14.07 | phix | JT: really? where is that? |
| 02:14.10 | puzzled | [hC], maybe your router has sip QoS built-in and not for iax? |
| 02:14.25 | VJFROMGT | what i mean is sometimes you get cases where users enable a codec in sip.conf but not iax.conf |
| 02:14.26 | phix | JT: in "The Book" still ? |
| 02:14.49 | puzzled | phix, or <exten> => do something |
| 02:14.52 | [hC] | okay we're way beyond all of this stuff. |
| 02:15.10 | phix | puzzled: ? |
| 02:15.47 | phix | callerif extension matches, where is that documentated? |
| 02:15.51 | puzzled | phix, let me rephrase: <callerid> => 1,bla |
| 02:16.26 | *** part/#asterisk dijungal (n=kdaniel@209.59.110.30) |
| 02:16.29 | phix | puzzled: umm I replace <callerid> with something? or I type that in as you typed it? |
| 02:16.37 | phix | puzzled: I need documentation :) |
| 02:16.44 | *** join/#asterisk gerphimum (n=trekkie@70.125.148.108) |
| 02:16.59 | puzzled | replace with the number that you want to do special stuff for |
| 02:17.56 | phix | I need <> around it? |
| 02:18.03 | puzzled | no |
| 02:18.34 | phix | can you give me a better example with number sd:) |
| 02:18.57 | JT | phix: it's in the book |
| 02:19.00 | puzzled | phix, 12345678 => 1,Answer |
| 02:19.14 | JT | 1234567/7373489 => |
| 02:19.28 | phix | oh, so the number instead of exten? |
| 02:19.32 | puzzled | phix, http://www.asteriskdocs.org/ |
| 02:19.39 | phix | JT: ok :) what chapter? :) |
| 02:19.41 | phix | I have the book |
| 02:19.54 | JT | the ones about extension matching |
| 02:19.58 | puzzled | search the index for extensions I guess |
| 02:20.38 | simond | is there some way to enable a module without going through 'make menuconfig'? |
| 02:20.57 | phix | ok |
| 02:21.57 | puzzled | simond, iirc you can enable it in (?) menuselect.opts and do make |
| 02:22.07 | *** part/#asterisk beek (n=klinebl@65.211.106.243) |
| 02:22.31 | phix | can't find it |
| 02:22.34 | phix | page number? |
| 02:22.47 | puzzled | simond, but that's only after make menu... has already been done |
| 02:23.03 | phix | chapter at least |
| 02:23.40 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 02:24.18 | puzzled | phix, http://voip-info.tr3ss.com/wiki/view/Asterisk+config+extensions.html |
| 02:24.29 | puzzled | phix, search for ex-girlfriend in the page |
| 02:24.34 | VJFROMGT | user is getting a fast busy when he dials http://pastebin.ca/766715 |
| 02:24.43 | sahafeez | question. i have a box with a digum pri card and a 4 port analog card. i am trying the asterisknow b6 on it. it only seems to see the 4 port and misses the t1 all together. |
| 02:24.47 | sahafeez | know issues? |
| 02:24.52 | phix | puzzled: thank you |
| 02:25.04 | JT | phix: damn you're lazy |
| 02:25.46 | puzzled | sahafeez, try loading the modules for the T1 card first, then for the analog card |
| 02:26.05 | sahafeez | hum, it does it automagicly on boot. |
| 02:26.21 | puzzled | then unautomagic it and do it yourself :) |
| 02:26.22 | sahafeez | i will have to find out how. however lsmod does not even seem to be on the system :) |
| 02:26.37 | puzzled | lol |
| 02:27.08 | phix | ok I undertand now |
| 02:27.18 | phix | JT: no, just impatient :) |
| 02:27.22 | VJFROMGT | user is getting a fast busy when he dials 18686243211 http://pastebin.ca/766715 |
| 02:28.17 | *** join/#asterisk obnauticus (n=obnautic@c-71-236-181-11.hsd1.or.comcast.net) |
| 02:28.32 | JT | phix: same thing |
| 02:28.42 | obnauticus | hey JT |
| 02:28.50 | obnauticus | Is a cisco 7940 any good> |
| 02:28.51 | sahafeez | there is no lsmod on this!!! |
| 02:29.37 | puzzled | obnauticus, I have Cisco & Polycom phones. prefer the Polycoms |
| 02:29.47 | puzzled | sahafeez, you have been hax0red |
| 02:29.48 | obnauticus | Why is that? |
| 02:29.51 | obnauticus | The screens look awesome. |
| 02:30.03 | JT | obnauticus: polycoms are far better |
| 02:30.04 | sahafeez | it is a new install that is 30 secs old so no |
| 02:30.05 | puzzled | obnauticus, the polycom screens are even more leet |
| 02:30.08 | JT | ciscos are overpriced junk |
| 02:30.15 | JT | their sip firmware sucks too |
| 02:30.19 | obnauticus | Affermative. |
| 02:30.33 | puzzled | indeed, their SIP sucks in capitals, including donkey balls |
| 02:30.40 | obnauticus | MMM |
| 02:30.41 | obnauticus | tastyl. |
| 02:30.44 | sahafeez | ah, bad pathing. |
| 02:31.03 | sahafeez | :) |
| 02:31.27 | sahafeez | god, it has been so long since i did the 1st box. what is the mod for the t1 called |
| 02:32.15 | puzzled | depends on the card but I don' recall |
| 02:32.19 | puzzled | wct1xx? |
| 02:32.33 | sahafeez | zaptel 177956 19 zttranscode,wcusb,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wct4xxp,tor2 |
| 02:33.24 | puzzled | I guess one of wctdm or wcte11xp ot wct1xxp |
| 02:33.39 | sahafeez | hum so it is loaded but asterisk does not think it is there |
| 02:34.42 | sahafeez | hum...just do it man. i guess |
| 02:40.33 | sahafeez | hum, reboot and it undoes my changes. nice. |
| 02:41.29 | *** join/#asterisk puzzled_ (n=patrick@puzzled.xs4all.nl) |
| 02:45.08 | sahafeez | Asterisk Now currently does not support digital cards, only analog. You will need to manually configure zaptel.conf and zapata.conf and probably disable the 'zapscan' utility from running on boot up a ... |
| 02:45.12 | sahafeez | okay..the you go.. |
| 02:57.27 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 02:57.44 | *** join/#asterisk lokkju_wrk_ (n=lokkju@unaffiliated/lokkju) |
| 02:57.54 | *** join/#asterisk BillBinko (n=BillBink@rrcs-24-73-75-102.se.biz.rr.com) |
| 02:58.59 | BillBinko | hello everyone |
| 02:59.24 | BillBinko | I am having odd performance issues and was wondering if anyone could help |
| 03:01.43 | *** join/#asterisk speekac (n=alwin@60.51.217.61) |
| 03:01.59 | speekac | anyone familiar with gs config generator ? |
| 03:14.19 | *** part/#asterisk BillBinko (n=BillBink@rrcs-24-73-75-102.se.biz.rr.com) |
| 03:14.38 | *** join/#asterisk techie (n=techie@adsl-76-240-176-149.dsl.lsan03.sbcglobal.net) |
| 03:17.21 | *** part/#asterisk techie (n=techie@adsl-76-240-176-149.dsl.lsan03.sbcglobal.net) |
| 03:27.35 | *** join/#asterisk Thazza (n=me@eth767.nsw.adsl.internode.on.net) |
| 03:28.00 | Thazza | Hey All.. Is it possible to join to sip channels via the CLI? |
| 03:30.11 | *** join/#asterisk mindCrime__ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 03:30.38 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:35.58 | *** join/#asterisk stybba (n=stybba@190.10.0.136) |
| 03:36.52 | *** join/#asterisk mindCrime__ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 03:42.40 | *** join/#asterisk JRsup1 (n=chatzill@12-207-206-43.client.mchsi.com) |
| 03:43.49 | JRsup1 | HALP...I haven't used asterisk in ages and I am setting up a system. I have a trunk set up and 1 extension so far. I can make oubound calls but inbound calls are just being picked up and getting a "goodbye" from the system and hanging up |
| 03:46.06 | JRsup1 | pls? |
| 03:50.20 | fujin_ | have you defined a context for incoming calls to be placed into |
| 03:50.30 | fujin_ | have you defined an extension that matchines calls to that incoming context |
| 03:51.04 | Thazza | Is it possible to join to sip channels via the CLI? |
| 03:54.02 | JRsup1 | context is...let me look that up |
| 03:54.26 | Strom_M | Thazza: not that i'm aware of |
| 03:54.59 | JRsup1 | context=incoming-mobile |
| 03:55.07 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 03:55.11 | JRsup1 | I don't know how to define an extension to match a context |
| 03:55.27 | Strom_M | JRsup1: pastebin your incoming-mobile context |
| 03:56.52 | Thazza | Strom_M, Is it possible instead to create a new called between 2 SIP extentions, via the CLI? |
| 03:57.22 | JRsup1 | :) um...well, I just have that line in the mobile.conf file. last time I tried asterisk I don't remember dealing with contexts. Where would I set that up...maybe that's my problem. |
| 03:58.26 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177582290.dsl.bell.ca) |
| 04:01.52 | Strom_M | JRsup1: when was the last time you used asterisk? |
| 04:01.56 | Strom_M | Thazza: generally, no |
| 04:03.37 | JRsup1 | um...sometime back when asterisk@home was asterisk@home not freepbx. Early/mid 2006? |
| 04:03.50 | Strom_M | that's not asterisk, you realize |
| 04:04.01 | JRsup1 | well, yeah...that's just the interface |
| 04:04.09 | JRsup1 | and technically it would be trixbox |
| 04:04.09 | Strom_M | are you using plain vanilla asterisk, or are you using a gui on top of it? |
| 04:05.03 | JRsup1 | I do have freepbx installed and I'm using it to add extensions and trunks (which has worked except incoming apparently) |
| 04:05.14 | JRsup1 | but I can access the .configs too.... |
| 04:05.35 | Strom_M | freepbx takes over the dialplan and makes it incredibly difficult to debug |
| 04:05.56 | JRsup1 | infact I'll probably skip out on using freepbx since it isn't quite working right. |
| 04:05.59 | [TK]D-Fender | ~freepbx |
| 04:06.00 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 04:06.02 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^ |
| 04:06.04 | Strom_M | i recommend you either start with vanilla asterisk on a linux distro of your choice, or go to #freepbx and inquire there |
| 04:06.26 | Thazza | Strom_M: Yeah i know, i just wanted to freak out a couple of friends plugged into my astisk install, by causing the system to call them both, and then bridge the connection. |
| 04:06.36 | *** join/#asterisk implicit (n=implicit@c-67-191-24-188.hsd1.fl.comcast.net) |
| 04:06.38 | Strom_M | Thazza: that's doable, but not from the CLI |
| 04:07.00 | JRsup1 | ok, will check that out, thx |
| 04:07.55 | Thazza | Strom_M: Call files? |
| 04:07.56 | Strom_M | Thazza: yes |
| 04:08.31 | Thazza | Strom_M: Thats pretty much the only way right? |
| 04:08.44 | *** join/#asterisk hesco (n=hesco@c-76-17-99-50.hsd1.ga.comcast.net) |
| 04:08.49 | Strom_M | well, you could write an AMI program to do it also |
| 04:08.56 | Strom_M | but call files are a bit simpler |
| 04:09.30 | hesco | <PROTECTED> |
| 04:10.20 | hesco | Any clues where I might start looking? |
| 04:10.26 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-ea6b33ae92da3b0d) |
| 04:14.05 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-94731a95fa45b8bb) |
| 04:15.12 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:18.08 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-823048d19b9d5f88) |
| 04:22.58 | [TK]D-Fender | hesco, Where is your phone relative to your * server? |
| 04:23.25 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 04:27.58 | hesco | its in an adjacent machine. different IP on the same subnet |
| 04:28.47 | [TK]D-Fender | hesco, and what do you mean playback is not getting YOUR audio? |
| 04:29.32 | hesco | Echo() works as expected. I say 'test' into the dialed local phone, I hear 'test' from my soft phone's speakers. |
| 04:30.29 | hesco | But this: exten => 600,3,Playback(demo-echotest) never gets played back through the softphone. |
| 04:31.09 | [TK]D-Fender | hesco, pastebin a call attempt |
| 04:31.18 | hesco | I'm wonderin if that is a path issue? or if something else might be at play. |
| 04:31.21 | [TK]D-Fender | hesco, along with your dialplan |
| 04:31.24 | [TK]D-Fender | ~pb |
| 04:31.25 | 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 |
| 04:31.27 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 04:32.51 | hesco | how would I observe a call attempt? I have a console running at -vvvc, is that where I need to look? |
| 04:33.21 | [TK]D-Fender | yes. Do "set verbose 10" and pastebin the complete call attempt and your associated dialplan |
| 04:37.43 | hesco | well I seem to have messed things up in my tinkering. Now I can't even raise an answer. I'll be back with you on this, hopefully in a moment. Thanks. |
| 04:40.37 | hesco | That's weird. The softphone at .105 is ringing, but its not getting an answer from .106, now. My first line of this extension reads: exten => 600,1,Answer() |
| 04:41.09 | [TK]D-Fender | hesco, pastebin the whole mess. |
| 04:43.21 | hesco | [TK]D-Fender: Is this whole mess enough? Or do you want to see all of extensions.conf? http://paste.debian.net/41884 |
| 04:45.04 | [TK]D-Fender | hesco, whats at the other end of this? : exten => 600,2,Dial(IAX2/diamondcard/17707551543) |
| 04:45.20 | hesco | So I see now that Echo() was in fact commented out. Do I need to restart my server every time I change the dialplan? |
| 04:45.28 | hesco | that's my landline |
| 04:45.43 | [TK]D-Fender | hesco, for dialplan changes a simple "reload" or "extensions reload" would do |
| 04:46.19 | [TK]D-Fender | hesco, So far nothing I see in there loks liek it dials a "phone" per se (discounting that IAX provider dial) |
| 04:48.04 | hesco | I just reloaded and then updated the pastebin. |
| 04:48.39 | hesco | doesn't this pass off the incoming call to my landline? |
| 04:49.18 | [TK]D-Fender | hesco, pastebin the call attempt |
| 04:49.40 | [TK]D-Fender | hesco, user pastebin.com please |
| 04:49.51 | [TK]D-Fender | considerably better & faster |
| 04:51.31 | hesco | I just updated the paste.debian.net, but I'll switch if you'd prefer. |
| 04:53.43 | hesco | OK, here you go: http://pastebin.com/d69a5df88 |
| 04:55.19 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 04:59.41 | [TK]D-Fender | hesco, ok, what are you dialing, and what seems to be happening? |
| 05:03.04 | hesco | I just restored the default sample extensions.conf, and started testing it and its working fine, now. |
| 05:04.00 | hesco | Thanks for your helpfulness. I think I'm going to start over and hack on what works, instead of trying to figure out what that was about. |
| 05:04.49 | hesco | After a week of interruptions frustrating successful tests, the possibilities I'm seeing here with a few successful tests are pretty exciting. |
| 05:05.32 | hesco | Thanks to all who helped build this. |
| 05:05.32 | *** join/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 05:05.32 | *** join/#asterisk Kirko (n=kirkalle@dsl093-224-026.slc1.dsl.speakeasy.net) |
| 05:05.55 | [T]ank | all of my phones are configured exactly the same. However I am getting an error on just one phone: [Nov 8 22:03:04] NOTICE[4388]: chan_sip.c:14474 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1204. What causes this? |
| 05:06.17 | Kirko | i keep getting this error on my asterisk system: |
| 05:06.19 | Kirko | [Nov 8 20:40:07] WARNING[4088] res_monitor.c: Execute of ( nice -n 19 soxmix "//dev/shm/1194579607.38441-in.wav" "//dev/shm/1194579607.38441-out.wav" "//dev/shm/1194579607.38441.wav" && rm -f "//dev/shm/1194579607.38441-"* ) & failed. |
| 05:06.28 | Kirko | anyone know why? |
| 05:08.26 | ManxPower | [T]ank: does [1204] in sip.conf have a mailbox= line? |
| 05:08.44 | [T]ank | yeah, that would do it ;-) duh |
| 05:09.52 | *** join/#asterisk bmg505 (n=leon@196.209.183.44) |
| 05:10.28 | [TK]D-Fender | ManxPower, Doctor, Doctor... it hurts when I raise my arm like this! |
| 05:10.58 | [TK]D-Fender | Kirko, Go confirm that you have "nice" and "soxmix" installed. |
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| 05:19.02 | Kirko | [TK]D-Fender, both are installed |
| 05:19.29 | *** join/#asterisk lemanal (n=lemanal@wifi-224-227.sc07.org) |
| 05:19.30 | Kirko | [TK]D-Fender, the commmand is executed successfully, but i always get that error back. |
| 05:28.26 | Freman | http://en.wikipedia.org/wiki/+61 <- jesus... that complicates my dial plan )c: |
| 05:30.45 | *** join/#asterisk hillct (n=H@cpe-024-211-242-232.nc.res.rr.com) |
| 05:31.06 | hillct | Good evening all |
| 05:31.48 | hillct | has anyone here worked with the PBXpress switch? From what I can tell, it's an asterisk fork but it's not clear from the documentation |
| 05:56.49 | *** join/#asterisk implicit (n=implicit@c-67-191-24-188.hsd1.fl.comcast.net) |
| 06:12.31 | *** part/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
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| 06:13.14 | *** part/#asterisk jim9119 (n=jandrews@ppp245-163.static.internode.on.net) |
| 06:14.13 | [T]ank | setting up a new set of t1s and getting an error on the cli when i start asterisk. I have added all of the configs and the error to pastebin: http://pastebin.ca/766853 I could use some help identifying what I have done wrong. |
| 06:14.50 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 06:17.38 | [T]ank | anyone? |
| 06:17.39 | *** join/#asterisk bintut (n=chatzill@202.92.128.20) |
| 06:20.35 | Strom_M | [T]ank: channels 48 and 96 are also d-channels |
| 06:20.46 | Strom_M | you miiiight want to not assign those at the bottom of zapata.conf... |
| 06:21.08 | [T]ank | that is where i got confused. this is how the guys at digium told me to do it. |
| 06:21.16 | Strom_M | well, what does your telco say? |
| 06:21.18 | [T]ank | the d chans are 24 and 72. |
| 06:21.29 | Strom_M | ok, so it's a single NFAS group? |
| 06:21.40 | [T]ank | the guy at digium said that the t1s without dchans would be backup. |
| 06:21.48 | [T]ank | that did not make sense to me. |
| 06:21.50 | Strom_M | ugh, no no no . |
| 06:21.53 | Strom_M | who told you that? |
| 06:22.00 | [T]ank | i thought a back up dchan would have to be exactly that... a dchan |
| 06:22.07 | [T]ank | patrick |
| 06:22.20 | Strom_M | i will have to yell at patrick |
| 06:22.32 | [T]ank | thats right... you are there to, right? |
| 06:23.01 | Strom_M | no |
| 06:23.04 | Strom_M | i'm in california |
| 06:23.25 | [T]ank | oh, ok |
| 06:23.31 | [T]ank | let me update and have you double check me. |
| 06:24.19 | [T]ank | so if i have 4 pri with dchan on 24 and 72 would this be correct? http://pastebin.ca/766861 |
| 06:25.16 | Strom_M | is it a single NFAS group? |
| 06:25.22 | Strom_M | or is it two NFAS groups? |
| 06:26.34 | [T]ank | well... all i really want is 4 t1s as one group, but they only have 2 dchannels |
| 06:26.43 | Strom_M | thats not what i'm asking |
| 06:26.49 | Strom_M | how has the telco configured the PRIs? |
| 06:26.59 | Strom_M | is it a single NFAS group with a primary and backup d-channel? |
| 06:27.25 | [T]ank | well... I have to guess at that until I can talk to them in the morning. I would assume that it is a single with a backup. |
| 06:27.51 | Strom_M | ok...in that case, you want to set up a single trunkgroup |
| 06:28.55 | *** join/#asterisk BeeBuu (n=chatzill@125.95.250.63) |
| 06:29.58 | [T]ank | Strom_M: http://pastebin.ca/766864? |
| 06:30.20 | Strom_M | no |
| 06:30.34 | Strom_M | trunkgroup => 1,24,72 |
| 06:31.04 | Strom_M | spanmap=1,1 |
| 06:31.09 | Strom_M | spanmap=2,1 |
| 06:31.12 | Strom_M | spanmap=3,1 |
| 06:31.14 | Strom_M | spanmap=4,1 |
| 06:31.50 | Strom_M | but make sure your logical spans match up with what your telco has too |
| 06:31.55 | Strom_M | try 1,2,3,4 |
| 06:32.41 | [T]ank | should i be concerned about this when i restart asterisk: |
| 06:32.42 | [T]ank | [Nov 8 23:31:51] WARNING[10691]: chan_zap.c:8550 pri_dchannel: Restart requested on odd/unavailable channel number 3/24 on span 1 |
| 06:33.18 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 06:33.31 | Strom_M | i would suspect it means that your settings and the telco's settings don't match up |
| 06:33.38 | [T]ank | makes sense. |
| 06:34.23 | *** join/#asterisk ozus (n=ozus@ip72-205-206-86.sb.sd.cox.net) |
| 06:34.35 | Strom_M | don't you have a copy of the order? |
| 06:35.58 | [T]ank | no... i am calling them now to see if i can get anything. |
| 06:36.19 | Strom_M | who is the telco? |
| 06:36.29 | [T]ank | I think I understand now how this is supposed to work now. |
| 06:36.32 | [T]ank | global crossing |
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| 06:41.59 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 06:43.27 | J4k3 | they're sending you 23Ds and 1 B per PRI :D |
| 06:46.07 | Strom_M | J4k3: nope |
| 06:46.12 | Strom_M | it's an NFAS group |
| 06:48.34 | J4k3 | I'm talking noise anyways |
| 06:49.25 | J4k3 | also, does anyone know of a chan_bluetooth, chan_mobile whatever to connect to a CDMA phone that will do automated dialouts? |
| 06:49.30 | J4k3 | I only need to dial one number |
| 06:50.57 | J4k3 | hmm, I guess I could use the voicedial capabilities of the phone |
| 06:51.25 | J4k3 | have * click the bluetooth, then say "dial someone [pause] jackass [pause] yes [connect call]" |
| 07:01.58 | [T]ank | Strom_M: reading over these freaking orders from the phone company... It has one trunk group number. Would that mean that it is a single nfas group? |
| 07:03.31 | Strom_M | yes |
| 07:06.08 | [T]ank | ok, so that is how i have it set up i think now... |
| 07:06.18 | [T]ank | i did a ztcfg -vv and restarted asterisk... |
| 07:06.56 | [T]ank | seemed to go ok. however checking pri show spans it shows: |
| 07:06.56 | [T]ank | slc-gbx-01*CLI> pri show spans |
| 07:06.56 | [T]ank | PRI span 1/0: Provisioned, Up, Active |
| 07:06.56 | [T]ank | PRI span 1/1: Provisioned, Down, Standby |
| 07:06.56 | *** join/#asterisk CtRiX (n=CtRiX@aretha.navynet.it) |
| 07:07.16 | [T]ank | if i had it configured correctly, would both of those show as up and active? |
| 07:10.22 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 07:15.29 | Strom_M | i'm not sure |
| 07:16.01 | [T]ank | ok.. i am getting no errors anymore, but calling in i am not getting any audio. errrrr. i hate pstn |
| 07:16.25 | Strom_M | the PSTN is easy |
| 07:16.35 | Strom_M | are you sure the circuit is supervising? |
| 07:17.36 | [T]ank | I dont know what that means. |
| 07:18.26 | Strom_M | is the call being "answered" by the called party? |
| 07:18.41 | [T]ank | checking |
| 07:18.56 | [T]ank | yes it is |
| 07:19.02 | Strom_M | how can you tell? |
| 07:19.53 | Strom_M | also, what are you callig from, and what specifically is answering the call? |
| 07:19.53 | *** join/#asterisk dlynes_laptop (i=dlynes@dhcp64-134-24-23.wnc.chi.wayport.net) |
| 07:20.12 | [T]ank | ok.. here is the setup. |
| 07:20.32 | [T]ank | the pstn goes into server 1. where it routes the call to server 2. This is where the dialplan answers the call. |
| 07:20.48 | [T]ank | specifically calling and Answer() in the dialplan for this number. |
| 07:20.51 | Strom_M | the entire PSTN goes into your server? :) |
| 07:20.52 | [T]ank | cli output verifies. |
| 07:20.56 | [T]ank | well... |
| 07:20.59 | [T]ank | you know what i mean |
| 07:21.03 | [T]ank | i have 4 pri |
| 07:21.08 | Strom_M | no |
| 07:21.08 | [T]ank | going into the server 1 |
| 07:21.10 | Strom_M | you have one PRI |
| 07:21.23 | Strom_M | which consists of four spans |
| 07:21.27 | [T]ank | understood |
| 07:21.30 | [T]ank | k |
| 07:21.52 | Strom_M | why two servers? |
| 07:22.02 | Strom_M | and how do the two connect to each other? |
| 07:23.33 | [T]ank | the two servers because server 1 is at the datacenter plugged into the pstn. these calls are routed to multiple locations based on dialed number. connected via point to point ds3 using iax2 |
| 07:23.42 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 07:23.50 | Strom_M | ok |
| 07:23.59 | Strom_M | you're plugged into the PRI, not into the PSTN |
| 07:23.59 | Strom_M | :) |
| 07:24.08 | Strom_M | let's try this |
| 07:24.25 | [T]ank | btw... thanks for taking this time to teach me |
| 07:24.28 | Strom_M | have the datacenter server answer the call, play a sound file of reasonable length, and then hang up |
| 07:24.43 | [T]ank | ok. and not send it to the other server... right? |
| 07:24.49 | Strom_M | correct |
| 07:26.41 | [T]ank | ok, done, reloaded, called and did not hear anything., |
| 07:26.49 | [T]ank | i played the demo-congrats. |
| 07:27.12 | Strom_M | show me the relevant part of the dialplan |
| 07:27.18 | Strom_M | and also tell me what you're calling /from/ |
| 07:27.42 | [T]ank | may have had a stray keystroke... hang on... |
| 07:28.15 | [T]ank | nope... ok copying the info |
| 07:29.04 | *** join/#asterisk jarod14 (n=jarod14@viatel-nautilsiege-52-218.cnt.nerim.net) |
| 07:30.26 | [T]ank | http://pastebin.ca/766894 |
| 07:30.56 | Strom_M | and also tell me what you're calling /from/ |
| 07:31.06 | [T]ank | my nmber? |
| 07:31.08 | [T]ank | number? |
| 07:31.30 | Strom_M | no |
| 07:31.35 | [T]ank | salt lake city, utah |
| 07:31.37 | Strom_M | what kind of phone equipment |
| 07:31.39 | [T]ank | cell phone |
| 07:31.44 | Strom_M | ok |
| 07:32.12 | Strom_M | is that XXXX masking yours? |
| 07:32.15 | [T]ank | yes |
| 07:32.21 | Strom_M | or is that what it actually looks like in the dialplan |
| 07:32.26 | [T]ank | no |
| 07:32.32 | [T]ank | it has the actual number |
| 07:32.37 | Strom_M | ok |
| 07:33.00 | Strom_M | try throwing a Progress() in there before you Answer() |
| 07:33.07 | [T]ank | ok... hang tight |
| 07:33.42 | Strom_M | in that example call, did you hang the call up from your cell phone? |
| 07:34.07 | [T]ank | yes |
| 07:34.14 | Strom_M | ok |
| 07:34.15 | [T]ank | <PROTECTED> |
| 07:34.41 | Strom_M | you did answer() after that, right? |
| 07:34.44 | [T]ank | yes |
| 07:34.51 | Strom_M | and it still doesnt work? |
| 07:34.54 | [T]ank | no |
| 07:35.04 | Strom_M | hm |
| 07:35.31 | Strom_M | your cellphone does show the call as having answered, right? |
| 07:35.37 | [T]ank | yes |
| 07:35.49 | [T]ank | i hear that the call connects. |
| 07:35.52 | Strom_M | let the call stay up and see if the disconnect works properly |
| 07:35.56 | Strom_M | you "hear" it? |
| 07:35.58 | [T]ank | i just do not ever hear the playback |
| 07:36.03 | [T]ank | light static... |
| 07:36.09 | Strom_M | ok |
| 07:36.13 | Strom_M | don't listen for it |
| 07:36.16 | Strom_M | look on your display |
| 07:36.21 | [T]ank | doing that also. |
| 07:36.23 | Strom_M | does the call timer start going? |
| 07:36.33 | [T]ank | on my cell? |
| 07:36.35 | Strom_M | yes |
| 07:36.37 | [T]ank | yes |
| 07:36.39 | Strom_M | ok |
| 07:36.46 | Strom_M | let the asterisk box hang up the call |
| 07:36.51 | Strom_M | see if that works properly |
| 07:36.57 | [T]ank | ok |
| 07:37.34 | [T]ank | let me pm you my output so i dont have to edit it |
| 07:37.45 | Strom_M | sure |
| 07:37.54 | Strom_M | pastebin it |
| 07:37.59 | Strom_M | and pm me the url |
| 07:38.33 | Strom_M | nice pastebinning :/ |
| 07:38.43 | [T]ank | sorry... did that before i saw your post here ;-) |
| 07:39.30 | Strom_M | i'm not sure. I /think/ what's probably going on is a mistake in the NFAS configuration where both switches think they're supposed to be using different b-channels |
| 07:39.54 | Strom_M | so i'd call global crossing and confirm exactly how the circuits are supposed to be set up |
| 07:39.56 | [T]ank | i think you are right. |
| 07:40.09 | [T]ank | i have a call into them. should hear from them in the next hour. |
| 07:40.17 | Strom_M | cool. |
| 07:40.20 | [T]ank | you gonna be 'round? or you going to bed? |
| 07:40.24 | Strom_M | i'll be here |
| 07:40.57 | [T]ank | ok... questions I should ask? single group? single dchan with backup? or two dchan? anything else? |
| 07:41.13 | Strom_M | get all the details you can |
| 07:41.29 | Strom_M | span numbering, span assignments, switchtype, etc etc etc etc etc etc |
| 07:41.36 | [T]ank | ok... will do |
| 07:41.48 | Strom_M | see if they can e-mail it to you |
| 07:41.59 | [T]ank | i appreciate your help |
| 07:42.16 | [T]ank | I am learning a ton, just not fast enought ;-) |
| 07:42.35 | [T]ank | gotta have this back up and running before the office opens in the morning. sigh |
| 07:43.03 | Strom_M | ugh. |
| 07:43.06 | Strom_M | PLANNING! |
| 07:43.27 | [T]ank | well... been having errors, so patrick was helping me get that resolved. |
| 07:43.44 | [T]ank | unfortunatly i cant make changes till after 10pm. |
| 07:44.28 | Strom_M | still...for any mission-critical system, there must always be a maintenance window and a way to take part of the system out of service while leaving the other part up |
| 07:45.12 | [T]ank | yeah... I agree. I wish I had the ability to make that happen. I would sleep more |
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| 08:11.51 | FlatFoot | morning all |
| 08:11.56 | obnauticus | hai2u |
| 08:22.14 | [T]ank | Strom_M: with dchans on span 1 and 3 do i specify a dchan on span 2 and 4 in the /etc/zaptel.conf? Or do i leave it blank? trying to fix this still while i wait for the telco. Here is my /etc/zaptel.conf. I know that it does not match my setup. That is what I still need to do. This is my starting point http://pastebin.ca/766908 |
| 08:23.17 | *** join/#asterisk af_ (n=getsmart@88-149-240-124.dynamic.ngi.it) |
| 08:24.32 | Strom_M | [T]ank: no, you don't specify d-channels on the other spans, because spans 2 and 4 are all b-channels. |
| 08:24.41 | [T]ank | just making sure. |
| 08:24.42 | *** join/#asterisk Schumie (i=SteveWri@cpc2-rdng2-0-0-cust382.winn.cable.ntl.com) |
| 08:24.47 | Strom_M | [T]ank: just sit tight until you talk to the telco |
| 08:24.52 | Strom_M | relax, have some tea |
| 08:24.58 | [T]ank | :-D |
| 08:27.14 | FlatFoot | TEA good thinking Strom_M |
| 08:37.24 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 08:37.55 | obnauticus | What hardphone do you guys suggest? |
| 08:39.02 | *** join/#asterisk ApEtc (n=apetc@ip70-162-218-46.ph.ph.cox.net) |
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| 08:39.26 | *** join/#asterisk slowshutt (n=d@196.211.34.2) |
| 08:40.16 | slowshutt | how does music on hold work? |
| 08:40.28 | obnauticus | it plays music through a media player |
| 08:40.38 | obnauticus | and sets the output device as a zaptel device. |
| 08:40.40 | obnauticus | i think |
| 08:40.45 | obnauticus | or something of the sort. |
| 08:41.54 | *** join/#asterisk blq (n=Bl@dslb-088-066-228-074.pools.arcor-ip.net) |
| 08:42.07 | obnauticus | slowshutt does that answer your question? |
| 08:46.30 | *** join/#asterisk Dirk- (n=a@oaktyres.force9.co.uk) |
| 08:47.14 | *** join/#asterisk jarod14 (n=jarod14@ns2.viatelecom.com) |
| 08:47.41 | Dirk- | help! My Asterisk installation seems to have no zaptel support. Zaptel is installed and asterisk is compiled to support it, but 'core show channeltypes' does not show Zap and the help command does not show anything related to zaptel either |
| 08:48.01 | Dirk- | I've tried recompiling everything and I'm at something of a loss here |
| 08:49.40 | kaldemar | have you loaded zaptel modules? |
| 08:50.14 | Dirk- | as far as I know, yes. lsmod shows an entry for zaptel and typing module load chan_zap in the console does nothing |
| 08:50.18 | tzafrir_home | [T]ank, next time just use genzaptelconf / zapconf |
| 08:50.51 | [T]ank | not familiar with that |
| 08:51.52 | tzafrir_home | Dirk-, do you have any zaptel hardware? |
| 08:52.37 | *** join/#asterisk parag0n (n=parag0n@87-194-9-117.bethere.co.uk) |
| 08:53.01 | Dirk- | yes, two Sangoma cards, A200 8 port analogue and A101 PRI |
| 08:53.26 | Dirk- | I have a standard phone here connected to one of the FXS ports, there is power to it but no dial tone |
| 08:54.35 | *** join/#asterisk J4zen (n=Jvan4zen@a213-84-139-244.adsl.xs4all.nl) |
| 08:56.57 | *** join/#asterisk darkskiez (i=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 08:58.34 | *** join/#asterisk friedrich| (i=friedric@trem-servers.com) |
| 08:59.25 | obnauticus | JT, are you there? |
| 09:00.23 | JT | just |
| 09:00.32 | obnauticus | What polycom phones do you run? |
| 09:00.47 | obnauticus | I'm looking for a good hardphone solution to my problem that includes a lack of hardphnes. |
| 09:00.47 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 09:01.34 | JT | Dirk-: is wanpipe installed and running? |
| 09:01.34 | JT | obnauticus: i have a few, what are your needs? |
| 09:01.34 | obnauticus | i dunno |
| 09:01.37 | obnauticus | I just want a hardphone, because I don't have one |
| 09:01.47 | obnauticus | I'm only 16 and I mess with asterisk in my free time and i've been using soft phones. |
| 09:01.52 | Dirk- | JT, it is, I've just ran the uninstall and I'm going to reboot and reinstall zaptel-libpri-asterisk-wanpipe and see what happens |
| 09:03.51 | JT | obnauticus: i guess the cheapest option is the IP320, probably is it's PoE only |
| 09:03.57 | JT | otherwise there's the IP430 |
| 09:03.59 | obnauticus | I got PoE, and money. |
| 09:03.59 | JT | and up |
| 09:04.02 | JT | ok |
| 09:04.02 | obnauticus | k |
| 09:04.07 | Strom_M | JT: no, the 320 also has a DC IN jack |
| 09:04.21 | obnauticus | JT, what kinds of features does it have? |
| 09:04.22 | Strom_M | it just doesnt ship with an adapter; you have to buy it separately |
| 09:04.25 | obnauticus | well |
| 09:04.37 | JT | Strom_M: well, that too |
| 09:04.37 | obnauticus | the good ones (ie. >=IP430 ) |
| 09:05.04 | Strom_M | obnauticus: they're all equally good; the difference comes down to footprint, screen real estate, and number of line appearances |
| 09:05.26 | obnauticus | That brings me to a funny point, i liked the cisco ones because they have a bigass screen., |
| 09:05.42 | obnauticus | Which i know is irrelevant to the functionallity, but hell, i think it looks cool. |
| 09:05.42 | obnauticus | lol |
| 09:05.46 | Strom_M | heh |
| 09:05.50 | Strom_M | then get a 650 |
| 09:05.55 | Strom_M | bigass BACKLIT screen! |
| 09:05.58 | Strom_M | oooOOOOooooo |
| 09:06.01 | obnauticus | yaaa |
| 09:06.08 | Strom_M | aaaaAAAAAAaaaa |
| 09:06.15 | obnauticus | I'll take 5 |
| 09:06.28 | obnauticus | lol. |
| 09:06.41 | obnauticus | What is so bad about the cisco phones, other than having to pay for firmware which is retarded imo. |
| 09:07.01 | Strom_M | obnauticus: my cisco phones have proven to be a touch on the flaky side |
| 09:07.17 | obnauticus | Everything on my net is flakey :/ |
| 09:07.25 | tzafrir_home | Dirk-, chan_zap probably failed to load due to mismatch between its configuration and the actual configured Zaptel channels |
| 09:08.26 | Dirk- | I tailed /var/log/asterisk/full during an asterisk -c and it seems you are right, there was an issue loading chan_zap that looks isdn related, I'm looking into it now |
| 09:08.34 | Dirk- | this thing has been driving me crazy! |
| 09:09.31 | obnauticus | Can anyone here tell me why uhh |
| 09:09.51 | obnauticus | I'm not receiving DTMF tones, or anything for that matter on my server via ipkall.. |
| 09:09.58 | obnauticus | i know IpKall's total crap, but im cheap you see. |
| 09:10.15 | Strom_M | you just said you had money |
| 09:10.24 | obnauticus | for a hardphone. |
| 09:10.30 | obnauticus | Well you got a point |
| 09:10.33 | obnauticus | i should get my priorites right. |
| 09:10.35 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 09:10.39 | obnauticus | priorities* |
| 09:11.12 | obnauticus | http://pastebin.ca/766928 <-- there's what i got :/ |
| 09:12.45 | J4zen | At present time, what would be 'the' best (price vs. quality) SIP-phone available? |
| 09:13.43 | Strom_M | J4zen: polycom ip320 is cheap and well-built and reliable |
| 09:13.45 | J4zen | I currently have 2 SNOM320 SIP phones, but they show some instability when left powered-on for longer periodes of time |
| 09:14.06 | J4zen | the whole thing will just stall |
| 09:14.16 | J4zen | polycom ip320? Thanks, ill look into that one |
| 09:14.25 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 09:14.49 | *** join/#asterisk Woifi1988 (n=anon@M1505P013.adsl.highway.telekom.at) |
| 09:15.03 | J4zen | Would the polycom 320 be superior to the snom320 or similar? |
| 09:15.09 | J4zen | it is fairly low-budget |
| 09:15.30 | Dirk- | both are good, snom is more 'high class' |
| 09:15.55 | FlatFoot | J4zen: i only use snom's never had a prob with them , what firmware are you using ? |
| 09:16.40 | FlatFoot | J4zen: and what version of phone ? |
| 09:16.53 | *** join/#asterisk linxroute (n=linxrout@117.0.26.118) |
| 09:22.42 | J4zen | Let me see |
| 09:23.07 | J4zen | ah my phones aren't connected atm due to some testing, so i can't check exactly |
| 09:23.23 | J4zen | straight out of the box firmware, phones ordered about 2 months ago |
| 09:24.05 | J4zen | Occasionally when i get back to my office the following day, my SNOM320 will be frozen. Not accepting any interaction what so ever |
| 09:24.09 | J4zen | only fix is restart it |
| 09:24.19 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 09:24.51 | J4zen | It could be just that specific phones hardware or so, as i havn't been able to test out the other SNOM320 yet |
| 09:25.06 | FlatFoot | J4zen: if your phone is Version 7 which at it's age i suspect it is make sure that you use snom320-7.1.19-SIP-f.bin |
| 09:25.16 | FlatFoot | .24 does not work correctly |
| 09:25.21 | J4zen | i see |
| 09:25.41 | J4zen | will do :), so in general you could say the 320's are actually rather stable ? |
| 09:25.43 | FlatFoot | BLF and all manner of other stuff seems broke |
| 09:26.05 | FlatFoot | yeah we have just upgraded from 190's to 320' and they perform fine for us |
| 09:26.25 | FlatFoot | each phone has 6 dept's and 2 direct dial numbers working perfectly |
| 09:26.58 | J4zen | nice |
| 09:27.01 | FlatFoot | i get my firmaware from |
| 09:27.12 | FlatFoot | http://snom.provu.co.uk/sw/snom320-7.1.19-SIP-f.bin |
| 09:27.13 | J4zen | did you configure them to retrieve their config from tftp? |
| 09:27.38 | FlatFoot | NO as we only have 12 phones i do them by hand at the web gui |
| 09:27.58 | J4zen | did you ever look into it? I had some issues configuring them for such a job :\ |
| 09:28.09 | J4zen | the settings seemed fine, on both client and server side |
| 09:28.19 | J4zen | but they refused to "see" the update on the server |
| 09:28.37 | FlatFoot | no but my provider provu will set them up for me so they connect t the net then download config per MAC |
| 09:28.48 | *** part/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 09:28.59 | J4zen | yeah so did mine |
| 09:29.05 | J4zen | but i'd rather have them stored on my local server |
| 09:29.09 | FlatFoot | this will be handy for whn we do the next install of 380 snom's |
| 09:29.26 | J4zen | provisioning.snom or so i believe it was |
| 09:29.33 | FlatFoot | thats due for the summer break of a school we are dealing with |
| 09:29.49 | J4zen | thats quite far away :p |
| 09:30.14 | FlatFoot | yeah well it's a private school that will only perform major upgrades etc through summer |
| 09:30.41 | J4zen | i see, well. Thanks for your input FlatFoot :) |
| 09:30.50 | FlatFoot | np J4zen |
| 09:30.56 | J4zen | ill check into that new firmware, hopefully that fixes the instability |
| 09:31.04 | J4zen | oh yeah |
| 09:31.23 | J4zen | you wouldn't happen to have any expierence with Outlook integration of asterisk right? |
| 09:31.42 | J4zen | or anyone in the channel for that matter |
| 09:31.46 | FlatFoot | no sorry not looked at that yet |
| 09:31.52 | J4zen | np |
| 09:31.57 | *** join/#asterisk Uatec (n=uatecuk@adsl.ntsols.com) |
| 09:32.22 | Uatec | Hi there |
| 09:34.35 | Uatec | Has anybody had any experience with VoiceMail() simply not recording messages? |
| 09:35.25 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:36.40 | Uatec | I have a customer who called up and left a message, but the message was never received by email and it's not stored on disk. |
| 09:36.56 | Uatec | but I have a MixMonitor recording of the message being left |
| 09:37.19 | Uatec | I also have a CDR entry in my DB of the call being as long as the recording i have and ending up in VoiceMail() |
| 09:39.30 | BeeBuu | when i set waitexten(10),how can i get the caller input and dial what caller input? |
| 09:46.50 | tzafrir_home | Dirk-, pastebin cat /proc/zaptel/* and /etc/asterisk/zapata.conf |
| 09:47.43 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 10:00.45 | *** join/#asterisk eserra (i=nobody@89-96-52-24.ip10.fastwebnet.it) |
| 10:01.21 | eserra | hi all |
| 10:02.04 | *** join/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 10:02.23 | eserra | I'm having a problem with SIP registration |
| 10:02.35 | *** join/#asterisk yidiyuehan (n=yidiyueh@58.185.151.162) |
| 10:02.57 | *** join/#asterisk saftsack (n=saftsack@pD9E044AE.dip.t-dialin.net) |
| 10:03.10 | eserra | asterisk stopped registering to my sip provider after a reboot |
| 10:03.38 | eserra | I see asterisk is not matching their "proxy auth required" with the REGISTER it just sent |
| 10:03.41 | [T]ank | seems like it never ends. i figured out my nfas issue and everything was working perfectly. Well... I felt like it was time to upgrade from 1.4.10 to 1.4.13 to resolve an iax2 bug. Well, after the upgrade I cannot even get my zap stuff to start. I am seeing an error: Unable to get span status: Inappropriate ioctl for device then it says it is unable to register my channels 1-23 and so on.... is there something that has changed betwee |
| 10:04.37 | yidiyuehan | hi, any one knows is it possible to reduce the bandwidth used further for remote phone calls? |
| 10:04.43 | eserra | I have my sip debug full of |
| 10:04.44 | eserra | Nov 9 10:57:09 DEBUG[3558]: chan_sip.c:3244 find_call: = No match Their Call ID: 77d54e84165dfda103b12ac343482bf8@127.0.1.1 Their Tag Our tag: as0c6051e2 |
| 10:09.37 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 10:11.33 | oej | Someone is sending SIP messages that your Asterisk doesn't recognize |
| 10:11.56 | oej | Might be stuff to clear up calls that happened before a reboot |
| 10:14.58 | *** join/#asterisk yannj_fr (n=yannj_fr@APuteaux-152-1-42-89.w82-120.abo.wanadoo.fr) |
| 10:18.04 | *** join/#asterisk yassaccan (n=yassacca@admin189.hgo.se) |
| 10:19.17 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 10:19.34 | MacWinner | have any of you used voipjet? is the quality good? |
| 10:19.56 | MacWinner | and reliability |
| 10:23.06 | *** join/#asterisk ming_zym (n=ming_zym@221.219.129.230) |
| 10:25.27 | *** join/#asterisk Derky (n=derky@dsl-083-247-065-012.solcon.nl) |
| 10:26.33 | [T]ank | has anyone else run into this: http://bugs.digium.com/view.php?id=11030 |
| 10:29.56 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-38-156.lns10.syd7.internode.on.net) |
| 10:31.52 | awk | please can somebody please shed some light on this |
| 10:31.53 | awk | http://www.pastebin.ca/766970 |
| 10:32.17 | MacWinner | could someone point me to a good document on implement High availability with asterisk? ie, i want to have a 2 servers (maybe in different datacenters) that back each other up on the fly |
| 10:32.28 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 10:33.12 | awk | back each other up? just rsync each other |
| 10:33.24 | awk | if you want fail over use, something like mysql realtime |
| 10:36.59 | [T]ank | anyone here using the svn trunk of asterisk? |
| 10:37.13 | [T]ank | any issues to be aware of? |
| 10:41.40 | awk | anyone writing billing software that can aid in my quest? |
| 10:41.59 | awk | I really need to understand how u bill for this.. http://www.pastebin.ca/766970 |
| 10:42.23 | slowshutt | hi there i dont understand how the music on hold ties into your dial plan. |
| 10:43.34 | slowshutt | if i have a incoming call and the user is busy how do i play the music to him? |
| 10:43.53 | slowshutt | to the caller that is? |
| 10:45.58 | MacWinner | awk: i was thinking more like if one box was hosting a DID, is there a config option to have a backup box take over automatically if the first one fails |
| 10:46.37 | MacWinner | or maybe the question should be, how do you load balance 1 DID across multiple asterisk boxes |
| 10:49.27 | slowshutt | how many users do you have MacWinner? |
| 10:49.34 | sergee | awk: is this call generated with callfiles? |
| 10:49.37 | awk | well you could use a quintum or something nad have 2 voip accounts, so the pri comes into the quintum and if the 1 box isn't up it goes to fall over trust |
| 10:49.50 | MacWinner | slowshutt: about 1000 pretty soon |
| 10:50.01 | awk | MacWinner read what i said |
| 10:50.07 | MacWinner | slowshutt: just closing a deal and doing architecture planning |
| 10:50.20 | awk | sergee: /var/log/asterisk/cdr/Master.csv |
| 10:51.03 | awk | I don't want to get events from the manager as if my software drops for a second I lose cal data, this way I lose nothing it just quiries the last entry in the cdr (csv) |
| 10:53.13 | dandre | Hello |
| 10:54.21 | dandre | I have put this: |
| 10:54.21 | dandre | member = Local/6014 |
| 10:54.21 | dandre | in my queue.conf to ring a user extension from a queue. |
| 10:54.21 | dandre | But this doesn't work. What should I do? |
| 10:55.01 | awk | clear |
| 10:55.26 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 10:55.41 | *** join/#asterisk Woifi1988 (n=anon@M1379P026.adsl.highway.telekom.at) |
| 10:57.51 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 10:58.57 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 10:59.41 | slowshutt | and you have only 1 did? MacWinner |
| 10:59.55 | *** join/#asterisk bantu (n=Miranda@rz-du-fgx-146-148.rz.uni-karlsruhe.de) |
| 11:00.31 | slowshutt | do you need to do something special to get the moh to play wav files? |
| 11:01.01 | sergee | awk: try to se NoCDR() or something like that in your dialplan, befor dialing backup route, or you can set userfield to some value that will indicate to your billing to skip that record... |
| 11:01.09 | slowshutt | moves some wave files from one asterisk box to another works on the one but not the other any help welcome? |
| 11:01.41 | sergee | s/se/use/ |
| 11:03.33 | sergee | jbot: i love you! |
| 11:03.36 | jbot | You love you!? |
| 11:08.33 | DarkFlib | jbot: I love aardvarks! |
| 11:08.34 | jbot | You love aardvarks!? |
| 11:09.53 | *** join/#asterisk _ys (i=ys@91.151.196.254) |
| 11:11.37 | awk | sergee I don't want to ignore the field I want that field |
| 11:11.45 | awk | sergee as you can see the full call is split over 2 lines |
| 11:11.54 | awk | I want to bill that correctly, from start address to end ... |
| 11:12.12 | awk | so I have to bind those 2 records, why on earth would asterisk dev create 2 unique ids for 1 call? |
| 11:12.19 | awk | or else I could then match the unique id |
| 11:13.10 | slowshutt | anyone help me with music on hold please |
| 11:14.33 | sergee | awk: i suppose you call DIAL 2 times, so that is 2 calls, not 1 |
| 11:14.44 | *** join/#asterisk Tebi (n=rantis@gw.aller.fi) |
| 11:15.02 | slowshutt | using MusicOnHold command after answering the channel, can see asterisk console says playing but can not hear anything |
| 11:15.24 | slowshutt | where does one start? |
| 11:19.45 | *** join/#asterisk kkjoe (n=kkjoe@p57A6AB3F.dip0.t-ipconnect.de) |
| 11:20.22 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 11:21.50 | kkjoe | im having trouble while try dial on multible channels, how could i save the DIALEDPEERNUMBER Variable in Cdr, im currently try it with Dial(mISDN/3/20&mISDN/3/21,60,TtM(getpeernumber)) and set the cdr user file in these context but this isn`t working |
| 11:22.33 | *** join/#asterisk MicW (n=michael@dslb-088-074-146-060.pools.arcor-ip.net) |
| 11:22.46 | MicW | hi |
| 11:22.56 | kkjoe | hi |
| 11:23.00 | slowshutt | try Dial(mISDN/3/20&21,60,TtM(getpeernumber)) |
| 11:23.12 | *** join/#asterisk parag0n (n=parag0n@87-194-9-117.bethere.co.uk) |
| 11:23.23 | MicW | is there a way to group some sip extensions together (like an alias)? e.g. shen i have Dial(Sip/1&SIP/2...) i'd like to use Dial(GROUP1) |
| 11:24.24 | kkjoe | you could use an variable called GROUP1 and set the DIal Parameter in these variable |
| 11:25.52 | kkjoe | http://www.voip-info.org/wiki/view/Asterisk+func+group These seems like the informations your looking for |
| 11:28.31 | *** join/#asterisk thewiizle (n=nick@87.127.85.42) |
| 11:28.33 | MicW | thanks |
| 11:28.34 | thewiizle | yo |
| 11:28.41 | thewiizle | if audio playback is erratically fast |
| 11:28.44 | thewiizle | thats kernel timing isnt it |
| 11:28.49 | thewiizle | or hardware sufferage |
| 11:30.09 | *** part/#asterisk Dirk- (n=a@oaktyres.force9.co.uk) |
| 11:37.22 | FlatFoot | ok daft question cdr_mysql.conf ... hostname= ... can this be an IP address or does it have to be a hostname ???? |
| 11:37.44 | ai-a[afk] | lookup of a ip returns the ip. |
| 11:39.25 | FlatFoot | basically trying to dump cdr across the network but the server its going to has ip only , just tried ip only did a reload and the * froze up |
| 11:39.36 | Uatec | hey, does anybody know any way of getting asterisk to authenticate sip clients using radius other than portaone, the portaone thing looks like such a nasty hack... Why is it doing authenticaiton of SIP clients in extensions.conf for starters...? |
| 11:39.43 | FlatFoot | had to comment out the details in file and reboot server |
| 11:39.51 | Uatec | FlatFoot, SQL server? |
| 11:39.55 | *** join/#asterisk ekimus (n=mm@xover.htu.tuwien.ac.at) |
| 11:40.08 | FlatFoot | Uatec: mysql server yes |
| 11:40.21 | Uatec | FlatFoot, i log my CDR data to an MSSQL server and refer to that by IP. |
| 11:40.33 | FlatFoot | MSSQL tell me more |
| 11:40.47 | Uatec | i'm sure that using an IP wouldn't freeze asterisk |
| 11:40.51 | Uatec | there must be somethign else wrong |
| 11:40.58 | Uatec | tell you more? |
| 11:41.18 | Uatec | i use odbc and freetds to log to a remote mssql database, it works very well |
| 11:41.21 | FlatFoot | yeah could be a coincedence we have had some tunnel probs today |
| 11:41.34 | Uatec | except for the fact that asterisk mashes up cdr data |
| 11:41.35 | Uatec | *sigh* |
| 11:41.57 | FlatFoot | Uatec: thats a bugger then |
| 11:43.09 | Uatec | yeah well |
| 11:44.10 | *** part/#asterisk Derky (n=derky@dsl-083-247-065-012.solcon.nl) |
| 11:46.49 | FlatFoot | Uatec: can i be cheeky , have you got an example of your cdr setup anywhere ? |
| 11:48.01 | Uatec | i basically followed the advice that i found on the wiki and googl |
| 11:48.02 | Uatec | e |
| 11:48.17 | FlatFoot | ta just found some interesting stuff |
| 11:50.45 | Uatec | come on |
| 11:51.00 | Uatec | i can't believe that nobody knows anything about radius stuff |
| 11:53.21 | *** join/#asterisk wmurailbfinance (n=wmurail@242.136-14-84.ripe.coltfrance.com) |
| 11:54.12 | *** join/#asterisk BeeBuu (n=chatzill@125.95.248.8) |
| 11:54.29 | BeeBuu | anyone here? |
| 11:54.39 | wmurailbfinance | yes |
| 11:54.44 | FlatFoot | nobody here but us chickens |
| 11:55.27 | BeeBuu | i had _333,1,waitexten(10) |
| 11:56.14 | BeeBuu | and user input some number,i want to dial that number |
| 11:56.22 | BeeBuu | what i need to do? |
| 11:59.24 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 11:59.35 | Mavvie | http://www.voip-info.org/wiki-Asterisk+cmd+WaitExten |
| 12:01.04 | FlatFoot | ~lunch |
| 12:01.04 | jbot | well, lunch is a nightmare in the Mensa |
| 12:01.12 | BeeBuu | _X.n,Dail(Zap/1/${EXTEN}? |
| 12:01.57 | Uatec | i've looked in to it more |
| 12:02.01 | Uatec | portaone is no good |
| 12:02.05 | Uatec | it's all about billing via radius |
| 12:02.07 | Uatec | not authentication |
| 12:02.42 | FlatFoot | Uatec: we use nttac on windoze for our radius ( PPPoE ) etc |
| 12:03.24 | FlatFoot | http://www.nttacplus.com/home.asp |
| 12:03.44 | *** join/#asterisk ivanfm (n=ivanfm@c906b486.virtua.com.br) |
| 12:04.01 | FlatFoot | you might i suppose be able to route radius requests through to it . we have used this in various guises for the last ten years |
| 12:04.54 | *** part/#asterisk simond (n=simon@208.68.95.5) |
| 12:05.06 | *** join/#asterisk simond (n=simon@208.68.95.5) |
| 12:06.01 | simond | ah, at last, found it. you can create a file called ~/.asterisk.makeopts, or /etc/asterisk.makeopts to get around having to run 'make menuconfig' |
| 12:06.04 | Uatec | FlatFoot, but you don't know how to persuade asterisk to use radius for sip authentication? |
| 12:06.22 | Uatec | i'm not interested in the radius server software at this point |
| 12:06.26 | Uatec | just the asterisk client side bit |
| 12:06.37 | BeeBuu | Mavvie: are you still there? |
| 12:06.41 | FlatFoot | ahh sorry m8 misunderstood |
| 12:07.30 | Uatec | i mean, sip is completely open to brute force attacks on asterisk |
| 12:07.56 | Uatec | radius would not only make usermanagement either, but it would protect against brute force |
| 12:08.29 | Uatec | and potentially even centralise authentication on a distributed asterisk implementation |
| 12:14.20 | *** join/#asterisk casix (n=casix@edifici-pub.adam.es) |
| 12:14.23 | casix | hello |
| 12:14.46 | casix | I'm having this error in my asterisk: chan_zap.c: Write returned -1 (Resource temporarily unavailable) on channel 53 |
| 12:15.02 | casix | I've been looking for information but I didn't find it |
| 12:15.10 | casix | anyone knows what can it be? |
| 12:18.53 | *** join/#asterisk allankardec (n=root@20150099019.user.veloxzone.com.br) |
| 12:19.46 | *** join/#asterisk blq (n=Bl@dslb-088-066-234-233.pools.arcor-ip.net) |
| 12:33.51 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 12:37.18 | tzafrir_home | casix, what is zap channel 53? what device? |
| 12:37.27 | *** join/#asterisk xonico (n=root@host30.190-31-73.telecom.net.ar) |
| 12:38.18 | casix | t4xxp with a pri module |
| 12:46.58 | *** join/#asterisk docelmo (n=vircuser@c-76-99-153-242.hsd1.de.comcast.net) |
| 12:53.22 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 13:00.38 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 13:01.16 | lirakis | good mornign |
| 13:01.33 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 13:04.04 | *** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 13:04.19 | *** join/#asterisk lukketto (n=lukketto@host158-5-dynamic.7-87-r.retail.telecomitalia.it) |
| 13:06.59 | *** join/#asterisk irule (n=irule@200.53.61.4) |
| 13:08.22 | *** join/#asterisk bantu (n=Miranda@rz-du-mvx-142-44.rz.uni-karlsruhe.de) |
| 13:12.06 | *** join/#asterisk michael-i (n=michael-@141.41.40.55) |
| 13:12.48 | kkjoe | @slowshutt: Dial(mISDN/3/20&21 istn`t working " Dial argument takes format" |
| 13:20.12 | kkjoe | is there a way to save the peer which accept the call in the cdr entry while using this command:Dial(mISDN/3/20&mISDN/3/21&Sip/23,60,Tt) ? |
| 13:25.47 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:28.08 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 13:31.30 | *** join/#asterisk cypherdelic (n=cypher@p5B27D908.dip.t-dialin.net) |
| 13:33.18 | tzanger | fun |
| 13:33.42 | phix | hi |
| 13:34.05 | tzanger | I get to figure out a way to deinterleave 256 bits |
| 13:34.05 | *** part/#asterisk xonico (n=root@host30.190-31-73.telecom.net.ar) |
| 13:34.15 | *** join/#asterisk dijungal (n=kdaniel@209.59.110.30) |
| 13:35.30 | dijungal | hello |
| 13:36.06 | dijungal | anyone knows of any good packet sniffer i can use to capture my raw rtp streams between myself and the asterisk server ? |
| 13:36.19 | dijungal | or a packet analyser |
| 13:36.26 | [TK]D-Fender | dijungal: Wireshark |
| 13:36.36 | dijungal | console base |
| 13:36.48 | *** join/#asterisk zerohalo (n=zeroHalo@pool-71-255-162-167.bstnma.east.verizon.net) |
| 13:37.12 | ai-a[afk] | dijungal: easy :) |
| 13:37.23 | ai-a[afk] | tcpdump |
| 13:37.31 | dijungal | k |
| 13:38.07 | dijungal | my service provider told me he uses some thing called usnif or something like that |
| 13:38.21 | ai-a[afk] | thats nice.. |
| 13:38.26 | ai-a[afk] | and we want to know that why ? |
| 13:38.42 | ai-a[afk] | tcpdump -s2000 -w /var/tmp/capture.pcap 'host xxx.xxx.xxx.xxx' |
| 13:38.49 | dijungal | he uses wireshack to capture the packets and usniff or something like to analyse it |
| 13:38.52 | ai-a[afk] | capture your full audio call. |
| 13:39.05 | dijungal | ok.. thanks |
| 13:39.05 | ai-a[afk] | wireshark (gui) can open the .pcap file and view it. |
| 13:39.31 | dijungal | k |
| 13:39.48 | dijungal | ai-a[afk]: i'm about to make a call i'll try that command |
| 13:40.07 | dijungal | can i run it in the background???? |
| 13:40.16 | dijungal | like use the & at the end or something |
| 13:40.32 | ai-a[afk] | man tcpdump |
| 13:40.39 | ai-a[afk] | ffs, get yourself a manual. |
| 13:40.46 | dijungal | irie |
| 13:42.33 | thewiizle | anyone tried internal_timing=yes over SIP? |
| 13:45.07 | wmurailbfinance | Hello |
| 13:45.49 | wmurailbfinance | I want to change the number menu of voicemail, are thay possible to change that with voicemail.conf ? |
| 13:46.03 | wmurailbfinance | or i need to change source code asterisk 1.4 ? |
| 13:46.25 | wmurailbfinance | i want to select the same menu of orange telecom |
| 13:49.39 | *** join/#asterisk adminguru (n=atze@p508A76A2.dip.t-dialin.net) |
| 13:50.52 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 13:50.58 | *** part/#asterisk adminguru (n=atze@p508A76A2.dip.t-dialin.net) |
| 13:51.15 | docelmo | Say does anyone know why a polycom 601 would show a SIP URI for the callerid number? |
| 13:51.36 | MrTelephone | asterisk rewrties the From: header in the sip message with what you specify in callerid="". will remote party ID overrid the from header? |
| 13:51.43 | docelmo | and or how to make it show just the number? |
| 13:52.03 | MrTelephone | docelmo, the new cisco 7960 firmware does the same |
| 13:52.14 | docelmo | MrTelephone yes if the accepting party trusts RPID |
| 13:52.49 | docelmo | actually let me rephrase.. yes if you decide to trust RPID |
| 13:52.50 | MrTelephone | becase im having trust issues between asterisk and openser |
| 13:53.13 | MrTelephone | a use with eyebeam can change his username and display name to anything and asterisk will accept it |
| 13:53.22 | docelmo | ya simple enough.. If you get RPID then tell your OpenSER Peer in asterisk to trustrpid |
| 13:53.40 | wmurailbfinance | hum, what's version firmware cisco 7960 ? |
| 13:54.31 | thewiizle | 7.5 ftw! |
| 13:55.44 | MrTelephone | im using 8.2 |
| 13:56.00 | *** join/#asterisk Faustov (n=faustov@unaffiliated/faustov) |
| 13:56.12 | docelmo | I would just like to know how to stop it. |
| 13:56.17 | docelmo | Its never done it before |
| 13:56.23 | Faustov | hi, i'm trying to get call recording on demand (*1), should i be able to see anything in the console log when someone wants to start recording? |
| 13:56.31 | MrTelephone | docelmo, what has changed? did you do a sip image upgrade? |
| 13:56.37 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:56.37 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 13:56.38 | lirakis | Faustov: yes. |
| 13:56.51 | Faustov | hmmm |
| 13:57.01 | lirakis | Faustov: you need to enable the feature code, AND you need to pass the Ww parameter to your Dial application |
| 13:57.04 | docelmo | I just set it up again and now it shows it. So I am assuming my config is skewed somewhere.. yes.. I setup provisioning cause web configuration wasnt working |
| 13:57.12 | lirakis | Faustov: without the Ww it wont record |
| 13:57.28 | docelmo | and instead of getting number for my callerid I get sip:number@ip |
| 13:59.00 | MrTelephone | docelmo, your best bet is to find out the sip version and download the administrators guide and search for call display |
| 13:59.08 | Faustov | lirakis: like this? : |
| 13:59.10 | Katty | herro. |
| 13:59.11 | MrTelephone | i havn't had that problem with my 501s yet |
| 13:59.14 | Faustov | [trunklocal] |
| 13:59.15 | Faustov | exten => _9NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}},wW) |
| 13:59.17 | MrTelephone | if I did I'd scream :( |
| 13:59.21 | MrTelephone | i don't like uri |
| 13:59.59 | *** join/#asterisk freezey (n=freezey@maher.mercy.edu) |
| 14:00.01 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 14:00.38 | MrTelephone | im having a uri problem right now |
| 14:00.55 | lirakis | Faustov: uh ... yeah sure.. ha ha.. i mean the wW just enables you to record on demand... assuming the rest of that dial string works fine ;) |
| 14:01.19 | MrTelephone | does anyone use openser here? |
| 14:01.39 | Faustov | lirakis: well i can dial but nothing happens when i press *1 |
| 14:02.06 | Faustov | lirakis: do i have to press *1 before the call? |
| 14:02.29 | dandre | Is there any way to change de default extension for local channels to something else than 'default' ? |
| 14:02.34 | docelmo | Yes |
| 14:02.46 | docelmo | MrTelephone I have setup Asterisk and OpenSER many times |
| 14:02.49 | dandre | I have seen local.conf file in /etc/asterisk |
| 14:03.15 | lirakis | Faustov: no .. during the call |
| 14:03.41 | Faustov | lirakis: hmm nah, still doesn't start recording |
| 14:04.02 | Faustov | i've added wW to each Dial() function in extensions.conf... |
| 14:04.19 | Faustov | DYNAMIC_FEATURES => automon as well |
| 14:04.30 | Faustov | and uncommented the automon line in features.conf |
| 14:04.36 | Faustov | maybe there's something else? |
| 14:04.38 | wmurailbfinance | Faustov: Yes press quickly *1 for recording and re-press that when you whant to stop |
| 14:04.49 | Faustov | ic |
| 14:05.12 | wmurailbfinance | Faustov: and your files wav are in /var/spool/asterisk/monitor |
| 14:05.42 | MrTelephone | docelmo, im using append_rpid_hf() and it doesnt appear in the sip message? any idea? |
| 14:06.04 | lirakis | Faustov: you should see some thing like this " User hit '*7' to record call. filename: wav|auto-1194617139-6468625191-2034792949|m" |
| 14:06.19 | lirakis | Faustov: what does your features.conf look like? |
| 14:06.24 | *** join/#asterisk zydrunas (n=Miranda@216.49.228-ip-184.ckt.net) |
| 14:08.41 | *** join/#asterisk salzh (n=salzh@124.77.5.180) |
| 14:09.31 | Faustov | lirakis: i dont see that line and /var/spool/asterisk/monitor remains empty |
| 14:10.03 | salzh | hi, all.when i start asterisk, it reports that "Ouch ...error while writing audio data: : Broken pipe". how can i repair it? |
| 14:10.56 | MrTelephone | rewriting from: is supposedly forbidden |
| 14:10.57 | MrTelephone | what a joke |
| 14:10.59 | Faustov | lirakis: http://www.pastebin.ca/767132 |
| 14:11.08 | Faustov | linagee: ^ is my features.conf |
| 14:11.40 | *** join/#asterisk felix_da_catz (n=felix@PixNationalGroup.phonoscope.com) |
| 14:11.57 | Faustov | sorry, not linagee, lirakis :> |
| 14:14.12 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:14.12 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 14:14.14 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:14.25 | lirakis | Faustov: features.conf looks "okay" |
| 14:14.31 | BeeBuu | hi,MrTelephone |
| 14:15.23 | Faustov | lirakis: yeah, well, all i did there was uncomment that line for *1 |
| 14:15.47 | Faustov | lirakis: there must be something more with the extensions.conf |
| 14:15.52 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:16.28 | MrTelephone | hi beebuu |
| 14:16.43 | BeeBuu | could you help again? |
| 14:17.20 | *** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
| 14:17.33 | iratik | Where can I get the non-beta version of asterisk now ? |
| 14:17.35 | dmz | hello, does anyone know how to turn manager debugging levels up? i'm trying to debug a script and all I get in the asterisk log is the login / logout, and the action command but not why it's erroring |
| 14:17.43 | lirakis | Faustov: .. you said you have DYNAMIC_FEATURES=>automon in extensions.conf right? |
| 14:18.03 | wmurailbfinance | yes |
| 14:18.15 | BeeBuu | i want a extension: dial 333 to a menu,press 1 to dial local( need press another number),press 2 to dial outlines;( need press another number) |
| 14:18.30 | BeeBuu | i don't know how to make that |
| 14:19.04 | Faustov | lirakis: yes |
| 14:19.20 | Faustov | lirakis: with spaces around => to be exact |
| 14:19.30 | MrTelephone | docelmo? |
| 14:20.05 | iratik | The current beta of asterisknow is completely broken ... it doesn't work in any browser -- it has javascript problems... where can i get the stable - non beta release? |
| 14:21.01 | lirakis | Faustov: also i think your dial string is wrong ... you need to pass a blank param ... exten => s,n,Dial(SIP/mytrunk/${EXTEN},,wW) |
| 14:21.13 | lirakis | Faustov: note 2 commas |
| 14:21.19 | *** part/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 14:21.20 | *** join/#asterisk l0verb0y (n=l0verb0y@210.1.137.41) |
| 14:21.24 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 14:22.38 | lirakis | Faustov: .. because right now you are passing wW as the timeout param... ;p |
| 14:23.04 | Faustov | right |
| 14:23.10 | Faustov | lets see if it fixes my prob |
| 14:23.16 | l0verb0y | does anyone know how to use callforwarding in trixbox? I dial *72 and punch in the number i want to forward to? |
| 14:23.34 | [TK]D-Fender | l0verb0y: You're in the wrong channel. Please read the topic. |
| 14:24.01 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 14:24.01 | iratik | anyone? |
| 14:24.15 | [TK]D-Fender | iratik: you too :p |
| 14:24.23 | iratik | I'm trying to get started with asterisk-now and the web interface doesn't work in any browser |
| 14:24.35 | l0verb0y | thx |
| 14:24.35 | iratik | is there a non beta version? |
| 14:24.38 | iratik | or more stable release? |
| 14:24.40 | [TK]D-Fender | iratik: You're in the wrong channel. Please read the topic.<-- |
| 14:25.22 | iratik | sorry.. i was trying #asterisk-now |
| 14:27.06 | BeeBuu | please check http://pastebin.com/d653d75df , what's the right extension? |
| 14:27.08 | iratik | oh welll... what other options are there besides asterisk now ? |
| 14:27.15 | BeeBuu | anyone help me please? |
| 14:27.18 | *** join/#asterisk ManxPower (n=manxpowe@179.sub-75-203-131.myvzw.com) |
| 14:27.37 | *** join/#asterisk Darthclue (n=e054502@fw149.northside.isd.tenet.edu) |
| 14:28.35 | lirakis | BeeBuu: wtf are you talking about.. lol |
| 14:28.51 | *** join/#asterisk zydrunas (n=Miranda@216.49.228-ip-184.ckt.net) |
| 14:29.39 | [TK]D-Fender | BeeBuu: go read the BOOK, you're going to have to use mutlple CONTEXTS that contain the separate menus & extens you want to dial. |
| 14:30.26 | *** join/#asterisk jetlagmk2 (n=jetlag@151.204.7.155) |
| 14:30.43 | Faustov | lirakis: http://www.pastebin.ca/767155 <-- this is what i get in the log, no info about the recording tho... anyways based on this, can you tell in which part of extensions conf should i modify the Dial() functions? |
| 14:30.58 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:30.58 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:31.14 | kkjoe | hi there, is there a way to save the peer which accept the call in the cdr entry while using this command:Dial(mISDN/3/20&mISDN/3/21&Sip/23,60,Tt) ? |
| 14:32.03 | [TK]D-Fender | Faustov: Executing [0015@default:1] Dial("SIP/192.168.127.253-08215410", "SIP/0015") in new stack <--- this has no wW! |
| 14:32.18 | codefreeze | kkjoe: the cdr **should** record the src/dst info... why, are you having probs? |
| 14:32.22 | [TK]D-Fender | Faustov: You have to add it EVERYWHERE you want to be able to record |
| 14:33.00 | BeeBuu | [TK]D-Fender: i read the book,but it's beyond me....so please help me,give me a sample |
| 14:33.22 | BeeBuu | if you get a free time... |
| 14:34.02 | destructure | how about an expensive time |
| 14:34.46 | lirakis | Faustov: you need to modify ANY Dial() call that you want to be able to record from .. in that example .. the wW was not passed , so the recording wont work. |
| 14:35.01 | kkjoe | codefreeze: yes thats true but it do not save the entry who answered the call |
| 14:35.08 | [TK]D-Fender | BeeBuu: http://pastebin.com/m4260e901 |
| 14:35.28 | Faustov | ffs |
| 14:35.34 | Faustov | but it is everywhere |
| 14:35.39 | [TK]D-Fender | lirakis: Could have sworn I jsut said that ;) |
| 14:35.40 | Faustov | or i'm blind and i need more coffee |
| 14:35.44 | Faustov | which i'm having now :> |
| 14:35.47 | [TK]D-Fender | Faustov: everywhere BUT there :p |
| 14:36.03 | BeeBuu | thanks again. |
| 14:36.04 | lirakis | [TK]D-Fender: yeah sorry didnt see it till i already posted |
| 14:36.25 | [TK]D-Fender | lirakis: Took you over 2 minutes to type yours? ;) |
| 14:36.42 | lirakis | [TK]D-Fender: i dont multi task as well as you :P |
| 14:36.45 | BeeBuu | [TK]D-Fender: if i press 1,and how to dial out a num? |
| 14:36.58 | lirakis | sweet jeebus |
| 14:37.01 | [TK]D-Fender | BeeBuu: put other extens in those menu's contexts <----- |
| 14:37.58 | BeeBuu | put dial(${exten})? in [menu2] |
| 14:38.13 | lirakis | BeeBuu: this is covered in THE BOOK starting on page 121 |
| 14:38.15 | lirakis | !the book |
| 14:38.44 | lirakis | err.. whatever the jbot command is |
| 14:38.59 | *** join/#asterisk icewaterman (n=immagine@i538743D2.versanet.de) |
| 14:39.28 | BeeBuu | thanks,all. |
| 14:39.36 | Faustov | lirakis: http://rzadzins.info/extensions.conf |
| 14:39.40 | Faustov | [TK]D-Fender: ^ |
| 14:39.41 | [TK]D-Fender | BeeBuu: if you don't know how to deal with the dialplan you are completely screwed. Go read chapter 5 over and over again until you get it, or your eyes bleed, whichever comes first. |
| 14:39.44 | Faustov | it is everywhere... |
| 14:39.44 | icewaterman | hi, i have a problem with loading the misdn driver for my card: http://rafb.net/p/rxFCeh50.html |
| 14:39.45 | [TK]D-Fender | ~osmosis |
| 14:39.46 | jbot | i heard osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 14:39.47 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 14:39.55 | _x86_ | Strom_M: ugh... that office that keeps getting reorder tones after dialing, they are threatening me of switching to a used (circa 1983) mitel switch, and pointing the incompetence finger at me... this sucks |
| 14:40.32 | _x86_ | hahaha until your unconsciousness restores peace to the channel |
| 14:40.48 | [TK]D-Fender | Faustov: permanently remove EVERY comment that you did not personally write and all code that is not actively being used. |
| 14:42.42 | [TK]D-Fender | Faustov: Except in the place called by that line you pasted. Maybe fixed now... |
| 14:42.55 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-d429e46753ac8bcc) |
| 14:43.22 | Faustov | just did reload and retried |
| 14:43.25 | Faustov | no good |
| 14:43.39 | [TK]D-Fender | Faustov: turn up the verbose and pastebin it. |
| 14:43.40 | *** join/#asterisk stybba (n=stybba@190.10.0.136) |
| 14:43.41 | lirakis | Faustov: seriously .. that dialplan is a nightmare to read.. and .. maintain too probably |
| 14:43.51 | [TK]D-Fender | Faustov: and clean the crap out loike I asked :) |
| 14:43.58 | ManxPower | _x86_: I've found that most dialing problems that newbies have are issues with not checking the value of DIALSTATUS/HANGUPCAUSE after a Dial. |
| 14:44.02 | Faustov | yes i agree :> |
| 14:44.17 | lirakis | "Automatically generated configuration file" .. ugh |
| 14:44.24 | Faustov | [TK]D-Fender: i'd love to clean it up but i'm afraid of removing something i actually use :> |
| 14:44.29 | ManxPower | _x86_: I thought you were using wwww to fix the reorder problem |
| 14:44.41 | [TK]D-Fender | Faustov: Start with EVERY commented out line :) |
| 14:44.41 | Faustov | lirakis: nevermind that line... :) |
| 14:44.49 | Faustov | ok :) |
| 14:45.17 | ManxPower | Faustov: just exactly how did you get auto-generated configs? |
| 14:45.57 | *** join/#asterisk khronos (n=khronos@c-66-229-159-201.hsd1.fl.comcast.net) |
| 14:46.20 | ManxPower | [TK]D-Fender: based on the mailing lists, to day is going to be "one of those days" |
| 14:46.54 | [TK]D-Fender | ManxPower: not generated, just the stock stuff lightly modded |
| 14:47.06 | [TK]D-Fender | ManxPower: And how so for the "that kind of day" comment? |
| 14:47.07 | ManxPower | [TK]D-Fender: oh, yeah, THAT will work well. |
| 14:47.27 | ManxPower | [TK]D-Fender: I already had to correct two totally wrong things people said on the -users |
| 14:47.28 | [TK]D-Fender | ManxPower: it can. al you have to do is not use an existing context ;) |
| 14:47.41 | [TK]D-Fender | ManxPower: Yeah we've had some of that already today.. |
| 14:47.55 | ManxPower | Some moron said that if you don't specify a port number when you dial SIP by IP, then it will go to port 0 |
| 14:47.55 | Faustov | ManxPower: by the gui, which i swore never to use again! :) |
| 14:48.06 | hesco | What does this mean, and should I be concerned with it? |
| 14:48.11 | hesco | NOTICE[16486]: res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?! |
| 14:48.12 | *** join/#asterisk af_ (n=getsmart@88-149-240-124.dynamic.ngi.it) |
| 14:48.17 | ManxPower | Faustov: if you are still using the config files generated by the GUI, then you are still using the GUI. |
| 14:48.44 | lirakis | hesco: possibly that your system time is wrong, and yes because it can cause other problems especially when you recompile a kernel. |
| 14:48.44 | ManxPower | hesco: that should be a HARMLESS message, indicating that your system is a little slower than it needs to be to make everything happen that needs to happen when it needs to happen. |
| 14:48.57 | ManxPower | lirakis: that has NOTHING to do with the system time. |
| 14:49.10 | lirakis | ManxPower: okay.. thats why i said possibly |
| 14:49.55 | ManxPower | [TK]D-Fender: can you think of ANY issue with Asterisk that would be caused by an incorrect system time? |
| 14:50.31 | ManxPower | I guess maybe the time read back by Voicemail when it tells you when the message was left. |
| 14:50.58 | ManxPower | lirakis: for the most part, Asterisk does not depend on the system time for anything important. |
| 14:51.03 | [TK]D-Fender | NTP skew <- |
| 14:51.10 | [TK]D-Fender | in RTP |
| 14:51.22 | ManxPower | [TK]D-Fender: Huh? |
| 14:51.22 | Faustov | [TK]D-Fender: can i remove the [demo] section? Or sections that i dont set anything in? |
| 14:51.29 | robl^ | CDR logs would be wrong -- so that could cause billing issues ;-) |
| 14:51.38 | ManxPower | robl^: Ah! Yes, that is correct. |
| 14:51.38 | *** join/#asterisk billybongo (n=rich@82.153.23.79) |
| 14:51.42 | [TK]D-Fender | ManxPower: Of course a clock that screwed would have other dire consequences no doubt |
| 14:51.58 | ManxPower | [TK]D-Fender: yeah, but not for asterisk. |
| 14:52.10 | [TK]D-Fender | Faustov: I love hearing my statements spat back as rearranged questions :) |
| 14:52.14 | _x86_ | ManxPower: i tried that, and I also tried relaxdtmf=yes, but it still is not recognizing each DTMF digit correct every time (seems semi-random), and they're still getting reorder tones after dialing (and asterisk's CLI doesn't say anything except starting simple switch, and then hungup) |
| 14:52.19 | *** join/#asterisk Tili (n=tili@203.Red-83-53-146.dynamicIP.rima-tde.net) |
| 14:52.35 | ManxPower | Do people really think that all clocks have to be syncronized across all devices talking on all RTP connections? |
| 14:52.35 | _x86_ | ManxPower: also, I'm checking hangupcause, and it's always "normal clearing" |
| 14:52.47 | Faustov | [TK]D-Fender: heheh just making sure :> |
| 14:53.16 | ManxPower | _x86_: Huh? relaxdtmf seldom solves anything, and frequently causes issues with repeated digits (like 504-551-2211) |
| 14:53.36 | ManxPower | _x86_: What specific issue are you having? |
| 14:53.40 | _x86_ | ManxPower: Mr. Carlson told me to do that |
| 14:53.48 | _x86_ | (relaxdtmf) |
| 14:53.55 | ManxPower | _x86_: it's a suggestion of desperation |
| 14:54.21 | tzanger | I've never ever ever recommended relaxdtmf, it causes more problems than it solves |
| 14:54.21 | _x86_ | ok, well i'm there ;) |
| 14:54.35 | ManxPower | _x86_: So, to ask again. What specific issue are you having? |
| 14:55.09 | ManxPower | "reorder" is pretty generic. I can thing of at least 3 totally different situations where you could get a reorder tone. No, 4, I just thought of another one. |
| 14:55.34 | _x86_ | DTMF sometimes being recognized incorrectly (for example, had a few reports of "847" being dialed to the PSTN as "815") |
| 14:55.46 | Faustov | [TK]D-Fender, lirakis: brand new version (still not working): http://rzadzins.info/extensions.conf |
| 14:55.47 | *** join/#asterisk TheAndichrist (n=andy@CPE-72-128-95-237.wi.res.rr.com) |
| 14:55.57 | destructure | lol andichrist |
| 14:56.12 | [TK]D-Fender | Faustov: and the CLI output of your failed call please.... |
| 14:56.15 | robl^ | maybe users are pressing the wrong buttons? ;-) |
| 14:56.17 | _x86_ | also, after dialing a number, they get a reorder (fast "busy") tone 9 times out of 10 |
| 14:56.23 | ManxPower | Ok, DTMF not being recognized when a call does into an IVR, when a call goes to VoicemailMain, when a call comes in from the PSTN, when a call goes out to the PSTN, when a call comes from a SIP devcice???? |
| 14:56.35 | FlatFoot | just wanna check is 1.4.13 stable before i go ahead wiv dis new box |
| 14:56.45 | _x86_ | ManxPower: everything here is Zap channels |
| 14:56.57 | Faustov | [TK]D-Fender: http://www.pastebin.ca/767177 |
| 14:57.09 | Faustov | looks exactly the same as before (yes i did 'reload') |
| 14:57.11 | ManxPower | Based on the lack of virtually any information, I assume this: Analog phone -> somerandomcard -> Asterisk -> somerandomcard -> PRI PSTN |
| 14:57.32 | ManxPower | now, start providing some details. |
| 14:57.36 | _x86_ | ManxPower: i've got all the stations terminated to a patch panel, that comes out RJ21 (amphenol 25-pair) to a rhino channel bank, rhino feeds into asterisk over a CAS T1 interface to a Sangoma A102D-x |
| 14:58.06 | ManxPower | OK. now we know how the phones are connected to Asterisk. How is the PSTN connected to Asterisk? |
| 14:58.09 | _x86_ | ManxPower: your assumption was damn close :) |
| 14:58.14 | [TK]D-Fender | Faustov: -- Executing [0015@default:1] Dial("SIP/192.168.127.253-08219fd8", "SIP/0015") in new stack <-- see this? no "wW" |
| 14:58.48 | ManxPower | Sp we have Analog Phone -> Rhino Channel Bank -> Sangoma T-1 card -> Asterisk -> ???? -> ???? |
| 14:58.50 | [TK]D-Fender | Faustov: Now what you'll be noticing is that first you HAVE NO EXTEN FOR 0015! |
| 14:58.55 | _x86_ | ManxPower: asterisk dials out to the PSTN over another CAS T1 interface on the same sangoma A102D-x, which goes into a rhino FXO channel bank with 18 CO lines |
| 14:59.11 | [TK]D-Fender | Faustov: That is autogenerated by the FLAMING PIECE OF SHIT known as "user.conf". |
| 14:59.12 | *** part/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 14:59.19 | ManxPower | _x86_: Did you really design this system to make sure it fails? |
| 14:59.29 | [TK]D-Fender | Faustov: now TRASH that file and use sip.conf like a SANE person! :p |
| 14:59.36 | _x86_ | ManxPower: I've got a PSTN T1 on order ;) |
| 14:59.41 | ManxPower | _x86_: now, you need to watch the CLI to see what the Dial line for going to the PSTN is. Paste it. |
| 14:59.59 | [TK]D-Fender | Faustov: *-GUI & users.conf = GARBAGE |
| 15:00.04 | _x86_ | ManxPower: when they get the reorder tone, there is never a dial in CLI |
| 15:00.05 | ManxPower | not the CONFIGURED Dial() line, the ACTUAL dial line as shown on the CLI. |
| 15:00.20 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:00.22 | _x86_ | ManxPower: the only thing CLI shows me is starting simple switch, and then when the user hangs up, "hungup" |
| 15:00.26 | ManxPower | _x86_: so it looks like the issue is NOT the PSTN. |
| 15:00.28 | Faustov | [TK]D-Fender: wow, you sound scary :P Now, if i understood you correctly, * picks some stuff from users.conf instead of extensions.conf? |
| 15:00.33 | _x86_ | ManxPower: right |
| 15:00.36 | ManxPower | _x86_: what version of Asterisk? |
| 15:00.48 | _x86_ | 1.2.21.1 |
| 15:00.57 | [TK]D-Fender | Faustov: Yes... that config file gets "creative" with your dialplan, etc. Hence its garbage :) |
| 15:01.08 | [TK]D-Fender | Faustov: I think the param was "hasexten" <-- |
| 15:01.33 | [TK]D-Fender | Faustov: remake your accounts int he PROPER channel driver file and FLUSH users.conf. |
| 15:01.50 | _x86_ | ManxPower: if you have any suggestions, i'd be more than happy to hear them :) |
| 15:02.00 | ManxPower | _x86_: We have eliminated several possible issues. What are the rxgain / txgain settings in zapata.conf? |
| 15:02.06 | _x86_ | default |
| 15:02.24 | ManxPower | _x86_: what channels are analog phones and what channels are PSTN lines? |
| 15:02.41 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:02.41 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 15:02.54 | [TK]D-Fender | Faustov: the problem became apparent as soon as I saw [voicemenu-custom-1] called and traced that there was no exten that could match 0015. |
| 15:03.08 | _x86_ | 1-24 are stations; 25-35,37,39,41 are PSTN |
| 15:03.20 | Faustov | [TK]D-Fender: so where should i define all my accounts? |
| 15:03.26 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:03.28 | file | turning on dtmf logging in logger.conf will show the DTMF being dialed on the analog phones so you know what Asterisk sees... |
| 15:03.33 | _x86_ | ManxPower: 36, 38, and 40 are also PSTN, but we use those only for an inbound hunt group |
| 15:03.38 | [TK]D-Fender | Faustov: SIP in sip.onf, IAX2 in iax.conf |
| 15:03.38 | Faustov | [TK]D-Fender: if users.conf is not the right place? |
| 15:03.41 | Faustov | ok |
| 15:03.45 | [TK]D-Fender | Faustov: ... |
| 15:03.49 | Faustov | heh |
| 15:03.50 | [TK]D-Fender | ~osmosis |
| 15:03.50 | jbot | osmosis is, like, the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 15:03.52 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 15:03.54 | [TK]D-Fender | :D |
| 15:03.57 | ManxPower | OK, BEFORE the channel => 1-24 line, put rxgain=2, then before the channel => 25-35 put rxgain=0. reload chan_zap.so and try dialing. |
| 15:04.05 | icewaterman | |
| 15:04.06 | Faustov | [TK]D-Fender: plz dont hurt yourself :> |
| 15:04.16 | [TK]D-Fender | Faustov: thats for YOU ;) |
| 15:04.26 | ManxPower | [TK]D-Fender: I think Faustov is beyond our help. |
| 15:04.58 | Faustov | no way, it's been very helpful so far |
| 15:04.59 | [TK]D-Fender | ManxPower: No, not yet.... |
| 15:05.05 | _x86_ | file: but we already know what asterisk sees is not correct |
| 15:05.05 | ManxPower | _x86_: it is not well known, but sometimes DTMF issues are really audio gain issues. |
| 15:05.15 | *** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 15:05.18 | Faustov | i'm beginning to hate my boss for not letting me reinstall this from scratch without any gui |
| 15:05.25 | [TK]D-Fender | ManxPower: I'm pounding through with heavy-hitting knowledge and the simple stuff :) |
| 15:05.28 | ManxPower | Faustov: until he does, you are screwed. |
| 15:05.28 | _x86_ | ManxPower: ah cool... so would you suggest i increase the gain or decrease it? |
| 15:05.36 | *** join/#asterisk Seldon75 (n=chatzill@69.77.161.3) |
| 15:05.44 | ManxPower | _x86_: what did I tell you to set rxgain to? |
| 15:05.59 | Darthclue | Faustov, it would take less time to start over than it would to fix the existing system. |
| 15:06.00 | _x86_ | ah right, sorry didnt see that |
| 15:06.00 | ManxPower | ManxPower: OK, BEFORE the channel => 1-24 line, put rxgain=2, then before the channel => 25-35 put rxgain=0. reload chan_zap.so and try dialing. |
| 15:06.34 | Seldon75 | we are moving from Zap channels (on a Sangoma card) to using only IP connections from a Voip provider - can someone please tell me briefly what steps are involved to make this transition? |
| 15:06.58 | ManxPower | Seldon75: now why would you want to make your system less reliable? |
| 15:07.14 | Seldon75 | ManxPower: we've had so much trouble with Copper |
| 15:07.21 | Seldon75 | you wouldnt believe |
| 15:07.25 | _x86_ | ManxPower: also, i'm only dialing two 'ww's before the number, should i increase that to 4? |
| 15:07.27 | [TK]D-Fender | Seldon75: setup your peer and change all your dials. The End. |
| 15:07.35 | ManxPower | Seldon75: if you can't make copper work, you will have major issues with making voip work. |
| 15:07.42 | [TK]D-Fender | _x86_: 4 should do nicely |
| 15:07.48 | [TK]D-Fender | _x86_: at .5s/ea |
| 15:07.50 | Faustov | Darthclue: most probably, but if i strt from scratch the company wont have a working asterisk, which is not acceptable |
| 15:07.51 | Seldon75 | ManxPower: the copper issues were not under our control |
| 15:07.54 | ManxPower | _x86_: since the calls are not even getting to Asterisk, that would be pretty pointless at this time. |
| 15:08.01 | _x86_ | ManxPower: true |
| 15:08.06 | ManxPower | Seldon75: and the voip issues won't be under your control either. |
| 15:08.32 | ManxPower | At least with copper you can talk to your provider. With Voip you would have to try to talk to every provider between you and your carrier. |
| 15:08.42 | Darthclue | Faustov, do they not have a backup system that could be used as a dev? or downtime on the weekends? (yeah i know, nobody likes to work on the weekends) |
| 15:09.28 | Seldon75 | ManxPower: we've been having dropped calls due to attenuation on our copper lines; at least that's theoreticaly impossible with IP |
| 15:09.46 | Faustov | Darthclue: i do have to do stuff on the weekends :< Also, no, atm they're short in boxes |
| 15:09.49 | ManxPower | Seldon75: dropped calls are not caused by attenuation. |
| 15:10.05 | *** join/#asterisk ReD-MaN (i=daemon@172-220.static.golden.net) |
| 15:10.05 | Faustov | Darthclue: as ManxPower said, i'm pretty much screwed :> |
| 15:10.24 | Darthclue | Faustov, yeah, pretty much. |
| 15:10.45 | Seldon75 | ManxPower: regardless of the terminology; the copper lines were dropping calls |
| 15:10.50 | ManxPower | Seldon75: dropped calls can be caused by many things, but not attenuation. |
| 15:11.00 | ManxPower | Seldon75: well good luck with this. |
| 15:11.05 | Seldon75 | thanks |
| 15:11.06 | ManxPower | _x86_: I'm waiting. |
| 15:11.23 | Seldon75 | so maybe a little bit more detail: where is the peer setting done? |
| 15:11.24 | Faustov | heheh |
| 15:11.32 | ManxPower | Seldon75: sip.conf if you are using sip. |
| 15:11.33 | Faustov | [TK]D-Fender: good that you didn't see my sip.conf :D |
| 15:11.36 | Seldon75 | thx |
| 15:11.45 | ManxPower | Seldon75: but you need to READ THE BOOK |
| 15:11.47 | ManxPower | ~book |
| 15:11.48 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 15:11.57 | Faustov | [TK]D-Fender: it is 32kB for some reason :D |
| 15:12.10 | Seldon75 | I read the book two years ago and have forgotten everything ;S |
| 15:12.35 | Faustov | [TK]D-Fender: in summary, i moved all sip accounts to sip.conf, in users.conf i only got extensions now. Should i modify them all to "hasextension = yes"? |
| 15:13.15 | ManxPower | Faustov: how about mv /etc/asterisk/users.conf /tmp |
| 15:13.20 | [TK]D-Fender | Faustov: No, you should COMPLETELY flush that file and prevent its module from even LAODING |
| 15:13.23 | [TK]D-Fender | LOADING* |
| 15:13.28 | Faustov | wow |
| 15:13.35 | *** join/#asterisk bcnx (n=root@d54C0E204.access.telenet.be) |
| 15:13.43 | ManxPower | Faustov: users.conf was designed mainly to make GUIs easier to write. |
| 15:13.47 | robl^ | users.conf is... "icky" |
| 15:14.28 | [TK]D-Fender | When solar climate is simply too kind ;) |
| 15:14.41 | robl^ | my users.conf is one line -- " ; #Ignore this darn file!" |
| 15:14.42 | bcnx | Hi all - hope you're doing fine ... I'm trying to get security added to the IAX channel, but as soon as I add the "secret" parameter to iax.conf, my softphone can't register anymore |
| 15:15.04 | bcnx | not with md5, not with plaintext |
| 15:15.12 | Faustov | heh |
| 15:15.12 | bcnx | anyone experienced this before? |
| 15:15.47 | Faustov | [TK]D-Fender: but i think i need all the extension entries from users.conf somewhere? |
| 15:16.17 | [TK]D-Fender | Faustov: extensions = extensions.conf |
| 15:16.26 | Faustov | ok |
| 15:16.29 | docelmo | [TK]D-Fender any experience with the Polycom 601 displaying the From: SIP URI header and not just the number? |
| 15:16.37 | ManxPower | Faustov: if it is not in extensions.conf then it is not an extension |
| 15:16.39 | [TK]D-Fender | Faustov: Time to write your ENTIRE dialplan yourself and stop depending on that junk to do it for you (poorly) |
| 15:16.44 | robl^ | Faustov: users.conf stuff should be split up into iax.conf, sip.conf, extensions.conf, voicemail.conf... |
| 15:16.50 | ManxPower | docelmo: there is a SIP get header function. |
| 15:17.02 | [TK]D-Fender | docelmo: Think I saw ti ONCE, but that was long ago and I don't remember any of the circumstances. |
| 15:17.20 | ManxPower | "show function SIP_GETHEADER" I think. |
| 15:17.43 | docelmo | ManxPower I know.. Im getting the From: SIP URI in the callerid number area of my Polycom 601 |
| 15:17.46 | docelmo | Not sure why |
| 15:18.07 | ManxPower | docelmo: sip debug might help you. |
| 15:18.08 | docelmo | I have looked over the configs and such and nothing jumps out at me saying this is the problem |
| 15:18.46 | Faustov | ok... |
| 15:18.48 | docelmo | did.. Debug looks normal. I believe this to be localized to the polycom. I know this is #asterisk but I figured someone might have seen this before and could point me in the right direction |
| 15:19.04 | Faustov | [TK]D-Fender: extensions moved to extensions.conf, sip data moved to sip.conf |
| 15:19.13 | Faustov | [TK]D-Fender: i did 'reload' and now i get this: |
| 15:19.15 | Faustov | [Nov 9 16:17:52] NOTICE[3265]: chan_sip.c:14758 handle_request_register: Registration from '<sip:0014@192.168.127.253>' failed for '192.168.127.252' - Device does not match ACL |
| 15:19.19 | ManxPower | docelmo: uh, why are you not setting callerid= in sip.conf for those things |
| 15:19.27 | ManxPower | Faustov: don't use ACLs |
| 15:19.38 | [TK]D-Fender | Faustov: time to learn how to set up an account for a SIP phone PROPERLY. |
| 15:20.11 | ManxPower | Faustov: you are the POSTER BOY for why you should not use GUIs |
| 15:20.15 | *** join/#asterisk devel (n=devel@wiggum.digitalcoven.com) |
| 15:20.27 | Faustov | [TK]D-Fender: pattern "acl" in extensions.conf not found |
| 15:20.34 | docelmo | ManxPower cause in theory callerid= would still build a SIP URI for the From: header like this sip:0000000000@ip(host) well the SIP URI is whats showing up. Not the 000000000 |
| 15:20.35 | ManxPower | Every single issue I've seen you talk about was caused by the GUI fucking up your config files. |
| 15:20.52 | ManxPower | docelmo: screw theory. try it. |
| 15:20.59 | _x86_ | ManxPower: i've done the rxgain thing you recommended, and told the office that I'll check back in within an hour to see if the situation is better / worse |
| 15:21.00 | Faustov | ManxPower: at first i thought ur just a hater, but i've begun to agree with your point :< |
| 15:21.10 | docelmo | hehe GUI's suck |
| 15:21.12 | devel | greetings all. if i'm not autoloading modules, which module(s) do i need to load to enable Playback() of *.ulaw files? |
| 15:21.17 | docelmo | brb kid screaming |
| 15:21.18 | ManxPower | _x86_: the problem is not happening often enough for you to test it? |
| 15:21.23 | robl^ | Faustov: he's a GUI hater, but its a well justified hatred. |
| 15:21.40 | _x86_ | ManxPower: *sigh* i'm roughly 100 miles away ;) |
| 15:21.55 | [TK]D-Fender | Faustov: those are not EXTENSIONS |
| 15:22.04 | ManxPower | _x86_: Exactly how did you install a system and not see the issue before you left? |
| 15:22.05 | _x86_ | ManxPower: i have to do stuff, have the office up there test it, and get a report back to check status |
| 15:22.17 | [TK]D-Fender | Faustov: strike that. Scoll back error |
| 15:22.24 | [TK]D-Fender | Faustov: Here, read this a bit : |
| 15:22.27 | [TK]D-Fender | ~jerjerguide |
| 15:22.28 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 15:22.36 | _x86_ | ManxPower: well, I only tried one number (the same number; my cell phone) from each station, and it worked fine |
| 15:22.38 | *** join/#asterisk adrin_ (n=adrin@chello084010032216.chello.pl) |
| 15:22.41 | adrin_ | hello |
| 15:22.45 | _x86_ | ManxPower: that was a 10-digit number |
| 15:23.00 | PepOSX | google is down? |
| 15:23.01 | ManxPower | _x86_: increase the rxgain by 2 until the problem disappears, or until you get to 12. Then start at -2 and keep going down to -12. |
| 15:23.16 | [TK]D-Fender | Darthclue: He's close enough right now... just need to tweak out the BS. |
| 15:23.33 | Faustov | Darthclue: i think this is a very good idea... |
| 15:23.34 | adrin_ | I have a lame VoIP related question - can i make free calls between SIP accounts made by two different providers? |
| 15:23.35 | _x86_ | ManxPower: now they are dialing a plethora of numbers, all 15 digit (6XXX6NXXNXXXXXX) and 16 digit (6XXX61NXXNXXXXXX) |
| 15:23.46 | _x86_ | ManxPower: ah ok |
| 15:23.47 | ManxPower | _x86_: chances are you'll see results fairly quickly. |
| 15:23.49 | adrin_ | like num@a.com and num2@b.com |
| 15:23.51 | Darthclue | ah, ok, well then, scratch that |
| 15:24.07 | ManxPower | _x86_: you also need to figure out how to turn on dtmf debugging to see what Asterisk is seeing. |
| 15:24.21 | _x86_ | ManxPower: i did that in logger.conf |
| 15:24.29 | Darthclue | adrin_, not likely |
| 15:24.40 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 15:24.42 | ManxPower | _x86_: and set debug 5 or something like that in the CLI? |
| 15:25.12 | *** join/#asterisk MicW (n=michael@dslb-088-074-146-060.pools.arcor-ip.net) |
| 15:25.15 | MicW | hi |
| 15:25.29 | *** join/#asterisk Zefk (n=Zefk@wsc-fo.b.astral.ro) |
| 15:25.32 | robl^ | out of curiosity -- is there a simple way to dump an astdb "family" into a text file that can be easily re-imported? |
| 15:25.45 | ManxPower | _x86_: so where are digits being missed? |
| 15:25.48 | *** part/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 15:26.00 | Zefk | hi! what is the best solution to record all conversations over a trunk (asterisk 1.4.13) ? thx |
| 15:26.05 | [TK]D-Fender | adrin_: calling through your providers costs whatever they charge you. |
| 15:26.31 | adrin_ | ok thanks D-Fender |
| 15:26.32 | [TK]D-Fender | Zefk: Putting Monitor or MixMonitor before every Dial that goes out it |
| 15:26.34 | ManxPower | _x86_: is the reorder (congestion) done heard before the user is done dialing, right when the user is done dialing, or is it delayed after the user is finished dialing? |
| 15:26.35 | MicW | is anyone here using terrasip? is it required to pre-pay even for free outbound calls? |
| 15:26.42 | adrin_ | co nothing like free p2p |
| 15:26.45 | adrin_ | ? |
| 15:27.09 | _x86_ | ManxPower: sometimes it's right after they dial, sometimes it's up to 2 seconds after dialing |
| 15:27.10 | [TK]D-Fender | adrin_: they are not P2P, they are companies that charge you for their use. |
| 15:27.17 | ManxPower | adrin_: Sure, but you can't just force random voip calls flying across the internet to route to your box. |
| 15:27.34 | [TK]D-Fender | adrin_: If you want to do VoIP between say you and family members who are all connected to the net, thats what FWD is for. |
| 15:27.48 | dmz | i use vitelity and think their rates are quite good and service has been extremely good |
| 15:27.55 | adrin_ | D-Fender: ok, so what is FWD? |
| 15:27.59 | dmz | and FWD for everyone who "knows" :) |
| 15:28.06 | ManxPower | adrin_: but as [TK]D-Fender says, that is only ONE company, not TWO companies. |
| 15:28.21 | *** join/#asterisk tobias (n=tobias@cpe-069-134-226-227.nc.res.rr.com) |
| 15:28.29 | ManxPower | _x86_: but never BEFORE they are done dialing? |
| 15:28.37 | _x86_ | ManxPower: reportedly not |
| 15:28.37 | dmz | it's your pbx, use as many different companies as gets you the best calling rates for where you call to/from :) |
| 15:28.53 | ManxPower | _x86_: you owe me a big contribution to the Manx Power Beer Fund if this fixes the problem. Paypal eric@fnords.org |
| 15:29.58 | [TK]D-Fender | adrin_: here |
| 15:30.00 | [TK]D-Fender | ~fwd |
| 15:30.01 | jbot | [~fwd] Free World Dialup, created by Jeff Pulver, is a free SIP server for P2P style that does not involve the PSTN (there is a charged option for this as well though). http://www.freeworlddialup.com/ |
| 15:30.14 | devel | greetings all. if i'm not autoloading modules, which module(s) do i need to load to enable Playback() of *.ulaw files? i'm trying to use logging/debugging, but it seems to have changed and i'm not seeing what i did in 1.2.x |
| 15:30.23 | _x86_ | ManxPower: looks like it's cutting off the last 2 digits or so |
| 15:30.42 | [TK]D-Fender | devel: add them one by one till it stops complaining :) |
| 15:30.48 | _x86_ | ManxPower: i'm tailing the dtmf log file, grepping for a specific channel (otherwise there's so much data it's impossible) |
| 15:30.50 | Faustov | [TK]D-Fender: ok, i've decided... another weekend goes for asterisk :< |
| 15:30.57 | Faustov | thanks for all the help guys |
| 15:31.00 | adrin_ | ok and what if i want to call lets say number@free.fr directly from my pc? |
| 15:31.06 | *** join/#asterisk fiXXXerMet (n=kjohnson@69.85.26.2) |
| 15:31.08 | _x86_ | ManxPower: the last 2 calls this station has attempted, it's been 2 digits short |
| 15:31.09 | adrin_ | is ist possible |
| 15:31.16 | devel | oh, [TK]D-Fender, i was just trying to avoid that. but if it's the only way... |
| 15:31.17 | ManxPower | adrin_: then you would put that in the softphone you have on your PC. |
| 15:31.19 | adrin_ | or directly from asterisk |
| 15:31.31 | ManxPower | then you would Dial(number@free.fr) in Asterisk |
| 15:31.32 | adrin_ | put that what do you mean? |
| 15:31.35 | adrin_ | oooh |
| 15:31.46 | adrin_ | lol |
| 15:31.51 | adrin_ | and how it routes it? |
| 15:31.59 | adrin_ | connects to free.fr |
| 15:32.04 | ManxPower | adrin_: to dial from the softphone on your pc you would tell the softphone to dial number@free.fr. |
| 15:32.04 | adrin_ | and calles numbers? |
| 15:32.12 | adrin_ | ok guys thanks |
| 15:32.21 | ManxPower | no asterisk required in that situation |
| 15:32.57 | adrin_ | yeah i know but i have 3 hardphones |
| 15:33.04 | adrin_ | using asterisk |
| 15:33.05 | ManxPower | adrin_: Asterisk is a PBX toolkit. expect to spend 1 - 4 weeks getting an Asterisk server setup and working OK. If you are unusually dense, then expect 1 - 4 months. |
| 15:33.11 | [TK]D-Fender | adrin_: Who are these people you are planning to talkwith? |
| 15:33.27 | adrin_ | i have a working asterisk setup |
| 15:33.37 | adrin_ | and i am using widevoip as my provider |
| 15:33.44 | ManxPower | adrin_: then it should be pretty obvious what you need to do. |
| 15:33.50 | adrin_ | i just wanted to know if i can call other sip accounts directly |
| 15:33.57 | adrin_ | i am a noob rather in voip :-) |
| 15:33.59 | ManxPower | adrin_: yes, but it is not recommended. |
| 15:34.11 | iCEBrkr | Dude. |
| 15:34.14 | adrin_ | ManxPower: not recommended why? |
| 15:34.18 | ManxPower | If you "dial by ip", which is what you are talking about then you have very little control over the call. |
| 15:34.25 | iCEBrkr | Someone just send me a URL to an old .MOD file I made back in the day for a colleage hacker radio show. |
| 15:34.28 | iCEBrkr | lol |
| 15:34.31 | iCEBrkr | I'm laughing my ass off |
| 15:34.31 | ManxPower | if you dial by sip.conf entry, then you have full control over the call. |
| 15:34.33 | iCEBrkr | http://members.usvoicedata.com/~pyster/audio/31337.MOD |
| 15:35.28 | devel | also, i was under the impression that if my channels are all ulaw (sip) that using the ulaw audio files would be the lowest cpu usage option. is that wrong thinking? |
| 15:35.31 | [TK]D-Fender | adrin_: Please don't use generic terms like "sip accounts" ok? EXACTLY who are these "accounts"? |
| 15:35.40 | ManxPower | devel: that is correct. |
| 15:36.04 | devel | well thank the gods i'm at least on the right track. :) thanks ManxPower |
| 15:36.55 | ManxPower | devel: show translation recalc 10 will give you a GENERAL idea of CPU costs for converting from one codec to another codec. |
| 15:36.55 | [TK]D-Fender | devel: Watch out for oncoming trains... |
| 15:37.01 | Darthclue | which is better voice quality, gsm or ulaw? |
| 15:37.15 | ManxPower | Darthclue: ulaw is what the PSTN in USA and Canada uses. |
| 15:37.29 | devel | yes, that's what i thought ManxPower, i just wanted to make sure that disk file translated to audio channel is the "same" |
| 15:37.38 | *** part/#asterisk datachomper (n=russ@75.146.194.61) |
| 15:37.47 | ManxPower | ulaw and alaw are the BEST sounding codecs other than the newfangled "wide band" codecs and nobody supports those. |
| 15:38.08 | ManxPower | devel: actually it is more like, read from disk, stuff in a packet. |
| 15:38.18 | adrin_ | [TK]D-Fender: a free.fr account, btw how do i make a call to number@free.fr using asterisk? ,exten => 791,1,Dial(SIP/number@free.fr,(...)) would work? |
| 15:38.37 | ManxPower | adrin_: what happens when you try that? |
| 15:38.57 | devel | yes, so no translation overhead, ManxPower. that's what i likes. now back to my original "which module to load to read ulaw files" |
| 15:39.13 | Darthclue | ok, so if a call is purely voip, then what? i ask because i've got a situation where a pure voip gsm connection sounds fine, but pstn -> voip provider -> pbx (all ulaw) sounds like crap (pops, static, crackles) |
| 15:39.14 | [TK]D-Fender | adrin_: Ah, calling to an existing proxy service. Yes, that will work. Dial(SIP/number@free.fr,15) for example |
| 15:39.16 | ManxPower | adrin_: when you dial like that, you have no control over the codec used, the dtmf mode used, the NAT translation, etc. |
| 15:39.28 | adrin_ | ManxPower: dunno didnt try yet, i thought that is incorrect |
| 15:39.50 | [TK]D-Fender | adrin_: But you should set up a proper peer entry in sip.conf to use rather than using a full URL like that as ManxPower was suggesting. |
| 15:39.51 | adrin_ | ManxPower: but i dont have to pay for the call? |
| 15:40.04 | ManxPower | adrin_: that would be up to the company free.fr |
| 15:40.05 | [TK]D-Fender | adrin_: Correct |
| 15:40.14 | destructure | wow, any idea how to grab the hostname that is running asterisk from inside the dialplan or agi? |
| 15:40.18 | [TK]D-Fender | adrin_: If THEY are setup to allow you to call their members that way <--- |
| 15:40.19 | adrin_ | [TK]D-Fender: oooh a peer entry for free.fr? |
| 15:40.20 | ManxPower | if free.fr is not free, then you would have to pay for the call. |
| 15:40.33 | ManxPower | destructure: what version of asterisk |
| 15:40.37 | destructure | 1.4 |
| 15:40.57 | _x86_ | ManxPower: did you say increment / decrement by 2 each time? |
| 15:40.59 | De_Mon | TrySystem() could do it, not sure if theres another way |
| 15:41.05 | [TK]D-Fender | adrin_: Yes. in sip.conf something like [dialtofreefr] type=peer host=free.fr disallow=all allow=alaw , etc |
| 15:41.06 | _x86_ | ManxPower: so if 2 doesnt work, go to 4, etc? |
| 15:41.18 | ManxPower | destructure: read /path/to/src/asterisk-1.4/doc/channelvariables.txt pay special attention the the ${ENV()} variable. |
| 15:41.19 | adrin_ | ManxPower: pay for incoming call in free.fr or for outgoing? i call someone who has account on free.fr but personally i dont have an accoiunt there so how can they charge me |
| 15:41.30 | destructure | hmm, I guess I could exec system from agi |
| 15:41.32 | ManxPower | _x86_: yes. |
| 15:41.36 | [TK]D-Fender | adrin_: and then Dial(SIP/dialtofreefr/12345) |
| 15:41.46 | destructure | the issue is of course that AGI is servicing a bunch of asterisks |
| 15:41.49 | De_Mon | I'd go with channel variables |
| 15:41.57 | _x86_ | ManxPower: and a simple reload chan_zap.so from CLI will re-parse that? |
| 15:41.58 | adrin_ | [TK]D-Fender: thanks! hmm not @dialtofreefr? |
| 15:42.00 | ManxPower | adrin_: if you don't have an account with them they can't charge you, so if they charge for that call they would prolly reject the call. |
| 15:42.04 | _x86_ | ManxPower: don't have to restart asterisk? |
| 15:42.14 | [TK]D-Fender | adrin_: you seem to be ASSUMING that free.fr will allow you to call their memeber this way.... |
| 15:42.14 | destructure | that should be doable, thanks for the tip |
| 15:42.17 | ManxPower | _x86_: correct. rxgain is something that can be changed on a reload |
| 15:42.35 | ManxPower | adrin_: VoIP does not magically let you make free calls. |
| 15:42.52 | ManxPower | Whoever you are connecting to must allow you to connect to them. |
| 15:42.56 | adrin_ | [TK]D-Fender: hehe right i forgot that they may not like the idea of free calls |
| 15:43.34 | [TK]D-Fender | adrin_: Since you haven't actually seen this option anywhree, the odds are REMARKABLY LOW that this is possible and you are coming up with crack-head ideas that jsut won't work. |
| 15:43.44 | [TK]D-Fender | :) |
| 15:44.00 | adrin_ | ManxPower: but there are no middlemen involved and i have two endpoins connected to a free worldwide compiuter network |
| 15:44.12 | *** join/#asterisk IPetrov2 (i=IPetrov2@ppp85-140-235-115.pppoe.mtu-net.ru) |
| 15:44.22 | adrin_ | ok i understand they charge fotr the fact that i have account on their server |
| 15:44.23 | ManxPower | adrin_: which free world wide computer network? |
| 15:44.24 | adrin_ | lol |
| 15:44.29 | adrin_ | internet |
| 15:44.35 | *** join/#asterisk seanbright (n=elixer@65.207.74.18) |
| 15:44.38 | adrin_ | ;-) |
| 15:44.45 | ManxPower | that just transports data, it has nothing to do with phone calls and does not know what the data is. |
| 15:45.04 | ManxPower | You still have to get your data to a phone. |
| 15:45.23 | adrin_ | but it is a computer that understands ip traffic |
| 15:45.25 | ManxPower | and there ARE middlemen involved. in your example that middleman would be free.fr |
| 15:45.50 | [TK]D-Fender | adrin_: Ok, you clearly hae NO CLUE about how calls are authed or how your provider works. |
| 15:45.53 | adrin_ | so why use providers :-) they just steal my money |
| 15:46.11 | adrin_ | [TK]D-Fender: i am afraid you may be right |
| 15:46.24 | ManxPower | adrin_: providers connect you to phones. |
| 15:47.10 | devel | ok, to read the ulaw files, format_pcm.so was the "key" |
| 15:47.26 | adrin_ | ManxPower: to PSTN |
| 15:47.35 | ManxPower | adrin_: not just the PSTN. |
| 15:47.59 | ManxPower | For the most part people don't just have phones laying around waiting to accept calls from random people. |
| 15:48.13 | ManxPower | They have phones connected to a provider. |
| 15:48.41 | ManxPower | None of my customers accept random unauthenticated calls, even if the calls could get thru the NAT and firewall. |
| 15:49.28 | ManxPower | adrin_: some of the peer-to-peer providers like FWD allow their users to call other providers for free, but they all have to be configured to do that. |
| 15:49.41 | ManxPower | and the provider has to accept the call. |
| 15:49.56 | destructure | it wasn't in the environment, but I exported it. hopefully I remember to do on additional boxes, heh |
| 15:50.01 | adrin_ | ManxPower: hmm ok |
| 15:50.21 | ManxPower | For example, there is a "gateway" to allow IAXTel users and FWD users to call each other without having to have an account on both networks, but that is a fairly unusual thnig. |
| 15:50.45 | ManxPower | destructure: on my systems HOSTNAME is the ENV var, it would vary from distro to distro. |
| 15:51.16 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 15:51.20 | *** join/#asterisk PepOSX (n=pepOSX@190.72.153.45) |
| 15:52.06 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 15:52.16 | *** join/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 15:52.26 | ManxPower | BTW, such inter-provider gateways usually suck. |
| 15:52.35 | [T]ank | anyone else running 1.4.13 with cdr_mysql? |
| 15:52.48 | [TK]D-Fender | adrin_: Va-t'ens! Merci la visite! ;) |
| 15:52.54 | [T]ank | i upgraded from 1.4.10 where cdr_mysql was running fine. now i cannot get it to run. |
| 15:52.56 | ManxPower | ~zeeek |
| 15:52.57 | jbot | well, zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 15:53.13 | Zeeek | good evening |
| 15:53.17 | [T]ank | wondering if anyone else is having issues with 1.4.13 |
| 15:53.49 | Zeeek | anyone ever use Wengo? THey seem to be having issues. Like going out of business |
| 15:54.22 | _x86_ | ManxPower: http://pastebin.ca/767217 |
| 15:54.57 | _x86_ | ManxPower: this is an example of someone getting a reorder... there were two 6's in the number next to each other, and it looks like it missed the second one |
| 15:55.00 | [TK]D-Fender | Zeeek: Wengo-ing, going, GONE! :p |
| 15:55.00 | Zeeek | I hate when companie go out of business with my money |
| 15:55.17 | Zeeek | EBay is the devil: wish tey'd go out of business |
| 15:55.30 | Zeeek | where's my meds? |
| 15:55.31 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 15:55.49 | ManxPower | _x86_: that is a CLASSIC relaxdtmf symptom |
| 15:55.53 | Katty | Zeeek: you took them 3 hours ago |
| 15:55.59 | [TK]D-Fender | Zeeek: Hate to say that if you have to come up with your own retarded client and give yourself a faggoty name (cue Trixbox!) expect the impending demise :p |
| 15:56.02 | _x86_ | ManxPower: i've disabled relaxdtmf though |
| 15:56.06 | Zeeek | damn. I'll some more then |
| 15:56.10 | _x86_ | ManxPower: took it completely out of zapata.conf |
| 15:56.22 | _x86_ | ManxPower: unless that doesn't take affect on a reload chan_zap.so |
| 15:56.39 | Zeeek | Ebay is pretty lame, and they make millions |
| 15:56.45 | ManxPower | _x86_: It is also a symptom of gain issues. chances are you actually need to do rxgain=-2 |
| 15:56.56 | Zeeek | but we dogress |
| 15:57.01 | ManxPower | when the audio level is too high, asterisk can miss repeated digits. |
| 15:57.29 | ManxPower | if it is too low, then it can just miss digits (not just repeated ones). |
| 15:57.32 | _x86_ | ManxPower: are we sure that relaxdtmf takes affect on a reload chan_zap.so? |
| 15:57.51 | *** join/#asterisk coppice (n=chatzill@39.192.17.210.dyn.pacific.net.hk) |
| 15:58.05 | ManxPower | _x86_: I would have to read the source, but when you do a reload chan_zap.so the CLI will SHOW YOU what options were not changed. (ignoring signalling), etc. |
| 15:58.30 | Katty | what's everyone's opinion of the polycom 301? |
| 15:58.41 | _x86_ | ManxPower: yeah the only thing it showed me was the signalling |
| 15:58.51 | _x86_ | Katty: it's only got half-duplex speakerphone |
| 15:58.57 | Katty | is it okay, or something you start the weekend's bbq fire with? |
| 15:58.58 | Katty | ewwwwww |
| 15:59.00 | _x86_ | Katty: you'll be MUCH happier with the 501 |
| 15:59.01 | ManxPower | Katty: It's the worst polycom, there are Polycoms with a better screen and microphone for aonly a little more. |
| 15:59.08 | Katty | _x86_: oh yes, we use the 501s (= |
| 15:59.15 | Katty | so the 320 is the.. |
| 15:59.18 | Katty | most resonable cheapy |
| 15:59.19 | Zeeek | DO NOT look for them on EBAY, please! |
| 15:59.20 | ManxPower | there are also the 320 and 340s as well |
| 15:59.35 | billybongo | polycom 301 - crap screen |
| 15:59.40 | _x86_ | Katty: i've heard good things about 330's, 430's, and 650's... though i've not used them |
| 16:00.07 | billybongo | 430s are OK |
| 16:00.13 | _x86_ | I only have like (2) 301's in production, the rest are all 501's and 601's |
| 16:00.15 | billybongo | 501 and 601 are cool |
| 16:00.20 | Katty | 501s are nice. |
| 16:00.23 | Katty | real nice. |
| 16:00.26 | _x86_ | i've got (3) 601's at my house :P |
| 16:00.26 | ManxPower | _x86_: ask the users if they think the audio levels are out of whack. i.e. too soft, too loud, etc, and what DIRECTION the issue is. |
| 16:00.35 | _x86_ | ManxPower: ok |
| 16:00.43 | Katty | i think the 320 would do... we don't need the extra network jack. |
| 16:00.46 | billybongo | _x86_: trying to persuade my SO that we need such things here too |
| 16:00.56 | ManxPower | i.e. if they say people they call say they are unusually loud or soft, that is a clue. |
| 16:01.06 | Zeeek | I know someone who watches West Wing reruns just to see th Cisco phones! Looks are everything |
| 16:01.20 | billybongo | is there a way to crank up the volume on polycoms? |
| 16:01.28 | billybongo | I had one customer turn them down purely on that |
| 16:01.38 | billybongo | this is with them turned up full on the front panel |
| 16:01.42 | Katty | any real advantage of the 550 over the 501? |
| 16:01.44 | ManxPower | Zeeek: After katrina I was in a feed and farm supply store in a tiny town in texas. they had a computer department and the tech was playing with cisco phones and asterisk. |
| 16:01.51 | Katty | power options, mayhaps? |
| 16:01.59 | _x86_ | billybongo: you turn the volume up during a call ;) |
| 16:01.59 | ManxPower | I thought I was having an LSD trip and got out of there fast. |
| 16:02.08 | Zeeek | ManxPower :) |
| 16:02.19 | billybongo | _x86_: yeah, they didn't like that ide |
| 16:02.20 | billybongo | a |
| 16:02.27 | billybongo | they wanted it loud all the time |
| 16:02.39 | _x86_ | ManxPower: yeah looks like i'm missing digits, not just repeated ones either |
| 16:02.40 | billybongo | also they weren't impressed with its speakerphone capabilities |
| 16:02.47 | ManxPower | billybongo: It really depends on what calls are too soft. polycom<->polycom or polycom<-> PSTN? |
| 16:02.55 | billybongo | I've yet to find a SIP phone that excels as a speakerphone |
| 16:03.10 | billybongo | ManxPower: both |
| 16:03.15 | robl^ | Polycom 650 is a GREAT speakerphone |
| 16:03.16 | _x86_ | ManxPower: and I've got the rxgain at 4 now |
| 16:03.24 | Katty | ManxPower: you know if the ip320s have a little microbrowser thing for logo display? |
| 16:03.26 | ManxPower | billybongo: *shrug* Pretty easy to fix in the polycom configs. |
| 16:03.31 | Zeeek | The Poly's are very good spkphns |
| 16:03.43 | _x86_ | ManxPower: scratch that, rxgain was at 6 when that happened |
| 16:03.44 | ManxPower | _x86_: Didn't I just suggest you try rxgain=-2? |
| 16:03.44 | ManxPower | \ |
| 16:04.07 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
| 16:04.09 | _x86_ | ManxPower: but you also said "too low and it will miss digits (not just repeated ones)" |
| 16:04.13 | Katty | [TK]D-Fender: i bet you'd know! |
| 16:04.26 | ManxPower | billybongo: in fact, low volume is a classic symptom of using config files for older phones with newer phones. |
| 16:04.43 | ManxPower | _x86_: yes. |
| 16:04.50 | [TK]D-Fender | Katty: Mew. |
| 16:04.58 | ManxPower | _x86_: DTMF issues on zap are a bitch to fix. |
| 16:04.59 | billybongo | ManxPower: ok, will look at that |
| 16:05.00 | [TK]D-Fender | Katty: 550 = waste of money |
| 16:05.15 | [TK]D-Fender | Katty: Price of a 601 with no expansion, and 2 less line-keys |
| 16:05.19 | _x86_ | ManxPower: ok, set it at -2, and i was able to restart asterisk completely because it's break time and no one is on the phone |
| 16:05.28 | *** part/#asterisk wmurailbfinance (n=wmurail@242.136-14-84.ripe.coltfrance.com) |
| 16:05.37 | [TK]D-Fender | Katty: Basically paying for snazzy (to some) colourscheme and a backlit screen. |
| 16:05.55 | ManxPower | _x86_: Asterisk's DTMF detection has significant bugs. I don't think there is a fix for 1.2, but I've not had major DTMF issues. |
| 16:06.07 | [TK]D-Fender | Katty: And yes, every phone except the 301 & 4000 can have an idle screen. |
| 16:06.17 | _x86_ | [TK]D-Fender: i'd be tickled if I could get a backlit screen on the 501 for roughly the same price ;) |
| 16:06.41 | [TK]D-Fender | Katty: Oh, and the 550 is native PoE for which you need to buy the brick seperately. |
| 16:06.43 | _x86_ | ManxPower: this weekend I was planning on upgrading to 1.4.12, like i've already done in other offices |
| 16:06.53 | _x86_ | ManxPower: you think that will solve the issues perhaps? |
| 16:07.22 | _x86_ | [TK]D-Fender: unless you have a PoE switch :) |
| 16:07.22 | ManxPower | _x86_: let me check a couple of things before I answer that. |
| 16:07.22 | FlatFoot | Zeeek: mines a guiness |
| 16:07.45 | Katty | [TK]D-Fender: uber. |
| 16:07.49 | Zeeek | FlatFoot he left |
| 16:07.56 | Zeeek | this is the bot |
| 16:08.09 | Katty | [TK]D-Fender: so "idle screen" is what the logo thing is... it's not a microbrowser? |
| 16:08.18 | Katty | [TK]D-Fender: or is the microbrowser what displays idle screen |
| 16:08.25 | [TK]D-Fender | Katty: Yuo can have either. |
| 16:08.29 | Katty | [TK]D-Fender: nice. |
| 16:08.32 | Zeeek | What phone can you put Youtube vids on when it's not in use? |
| 16:08.44 | [TK]D-Fender | Katty: Static image or idel browser. I prefer the latter as you can do nifty stuff on that. |
| 16:08.44 | Zeeek | iPhone for asterisk? |
| 16:09.06 | Katty | Zeeek: i thought it was gphone |
| 16:09.12 | Katty | Zeeek: i bet google will do something |
| 16:09.37 | robl^ | is switchvox supposed to replace AsteriskNOW? |
| 16:09.52 | Katty | robl^: i think trixbox did already |
| 16:09.56 | Katty | oh wait, no, that's asterisk@home |
| 16:09.58 | Katty | nevermind |
| 16:10.01 | _x86_ | ManxPower: oh man, -2 might have done the trick |
| 16:10.09 | Zeeek | You could do worse than to join us : http://voipusersconference.org/ning |
| 16:10.18 | Zeeek | ^^^^^^^ to anyone ^^^^^^^^^^^ |
| 16:10.27 | robl^ | Katty: Switchvox and AsteriskNOW are official digium products |
| 16:10.36 | ManxPower | _x86_: don't count your raptors before they are hatched. |
| 16:10.54 | Zeeek | so AsteriskTHEN is depricated? |
| 16:11.09 | pepse | AsteriskTWOWEEKSAGO |
| 16:13.23 | Katty | robl^: oh ah. |
| 16:13.28 | unixdog | come over to 1.4 |
| 16:13.35 | Katty | robl^: i can't say i've worked with them. |
| 16:13.36 | unixdog | come over to the dark side |
| 16:14.20 | casix | hello |
| 16:14.24 | casix | I'm having this error in my asterisk: chan_zap.c: Write returned -1 (Resource temporarily unavailable) on channel 53 |
| 16:14.27 | casix | I've been looking for information but I didn't find it |
| 16:14.30 | casix | anyone knows what can it be? |
| 16:14.43 | robl^ | Katty: I haven't either -- I used the old fashioned Asterisk -- bust saw the switchvox advert on Digium's home page... |
| 16:15.46 | Qwell | Zeeek: As bmd puts it - if you want something to hack up, use AsteriskNOW. Give switchvox to your grandmother. |
| 16:17.47 | *** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 16:17.50 | jameswf | switchvox is locked up tighter than an alter boy at a priest convention as far as hardware goes. so if you use it you either have to use pure voip or digium, AsteriskNow is a better bet if you need hardware support outside of digium products. |
| 16:18.21 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 16:18.44 | _x86_ | ManxPower: well in the last 7 dial attempts, 6 have worked just fine |
| 16:19.04 | _x86_ | ManxPower: which is a hell of a better ratio than the 15:1 they were claiming earlier |
| 16:19.19 | jameswf | its like mac before they gave you a shell |
| 16:19.24 | _x86_ | [TK]D-Fender: where is that place you buy hardware from? |
| 16:19.34 | tzanger | *sigh* |
| 16:19.47 | tzanger | is it simply not possible to have tdmoe spans installed BEFORE T1 spans? |
| 16:19.53 | tzanger | ztcfg won't even bring htem up now |
| 16:20.14 | jameswf | agrees with tzanger |
| 16:20.33 | ManxPower | _x86_: is it better than when rxgain was 2, 4, or 6? |
| 16:20.33 | _x86_ | anyone ever buy from telephony depot? |
| 16:20.48 | _x86_ | ManxPower: this is by far the best it's been |
| 16:20.55 | jameswf | I control module loading in rc.local to ensure analogs load last |
| 16:21.00 | ManxPower | try rxgain=-3 then |
| 16:21.07 | _x86_ | ManxPower: keep in mind, i'm grepping for a single channel that i'm using as a test case |
| 16:21.36 | tzanger | jameswf: won't work |
| 16:21.43 | _x86_ | where does everyone get their Sangoma hardware from? |
| 16:21.51 | tzanger | ztcfg fails since the hardware module isn't installed |
| 16:21.53 | _x86_ | looking for price comparisons to voipsupply.com |
| 16:21.55 | tzanger | and that *ABORTS* rc.local |
| 16:22.03 | tzanger | _x86_: I buy it direct |
| 16:22.58 | [TK]D-Fender | _x86_: Recently : www.telephonydepot.com |
| 16:22.58 | _x86_ | [TK]D-Fender: they seem to have the best price so far, it seems |
| 16:23.01 | [TK]D-Fender | _x86_: they average very well. Not the best in any one thing, but a good overall value compared to the majority |
| 16:23.24 | jameswf | tzanger, does work, blacklist modules, udev, modprobe, ztcfg |
| 16:23.34 | jameswf | do it all the time |
| 16:23.35 | tzanger | jameswf: you're not understanding |
| 16:23.41 | tzanger | I have wctdm blacklisted |
| 16:23.43 | tzanger | so it does not load |
| 16:23.51 | tzanger | that works |
| 16:24.09 | tzanger | ztcfg fails because 4 channels aren't there (wctdm isn't loaded, which is what I want) |
| 16:24.15 | tzanger | that failure causes rc.local to STOP |
| 16:24.22 | tzanger | so I can't, for example |
| 16:24.24 | tzanger | ztcfg |
| 16:24.25 | tzanger | modprobe wctdm |
| 16:24.26 | tzanger | ztcfg |
| 16:24.31 | jameswf | are you modprobing before ztcfg |
| 16:24.35 | tzanger | nope |
| 16:24.36 | tzanger | I can't |
| 16:24.41 | tzanger | otherwise wctdm has zap channels 1-4 |
| 16:24.47 | tzanger | I want 1-576 to be the 4 tdmoe spans |
| 16:24.54 | tzanger | and 577-580 to be tdm400 |
| 16:25.01 | jameswf | black list your digital too |
| 16:25.06 | tzanger | jameswf: I have no digital in this setup |
| 16:25.19 | tzanger | you don't understand, a failure of ztcfg ABORTS the rst of rc.local |
| 16:25.58 | tzanger | HAHAHAHAHAHAHHA |
| 16:25.58 | tzanger | I'm listening to a country station |
| 16:25.58 | jameswf | so throw together a bash script that runs last... |
| 16:25.58 | tzanger | this song is saying how he'd like to kiss this girl in the back stix, how he'd like to check her for ticks |
| 16:26.57 | tzanger | jameswf: fuck it |
| 16:27.03 | tzanger | I'm gonna hve two ztcfg scripts |
| 16:27.05 | tzanger | this is ridiculous |
| 16:27.10 | tzanger | why is ztcfg aborting rc.local on failure |
| 16:27.29 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 16:30.03 | fskrotzki | Looking for a way to hook asterisk so I know when it crashes and restarts. Having a issue with the current trixbox distro and faxing. sometimes when a fax is coming in it crashes when calling rxfax. So I'd like to find a way that it can notify me it is happening. |
| 16:30.22 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 16:31.11 | ManxPower | _x86_: see http://bugs.digium.com/view.php?id=10535 You might have to apply the 1.2 version of the patch. |
| 16:31.52 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 16:32.36 | tzanger | ztcfg -c /etc/zaptel.conf.fuck.you.ztcfg |
| 16:32.38 | tzanger | let's try that |
| 16:37.54 | Qwell | ~book |
| 16:38.58 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 16:38.58 | ManxPower | Qwell: I think jbot is having some quiet time with janebot |
| 16:38.58 | Qwell | somebody link me to the pdf? :p |
| 16:39.04 | *** part/#asterisk michael-i (n=michael-@141.41.40.55) |
| 16:39.09 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 16:39.18 | *** part/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 16:40.12 | ManxPower | fskrotzki: we cannot help you. |
| 16:41.05 | fskrotzki | ManxPower: thx |
| 16:42.53 | ManxPower | fskrotzki: but I'll bet #trixbox can help you. |
| 16:43.02 | ManxPower | ~zeeek |
| 16:43.20 | jbot | methinks zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 16:43.34 | hesco | Sorry, I asked this earlier, but then got distracted by the boss, let me try this again, What does this mean, and should I be concerned with it? |
| 16:43.36 | hesco | NOTICE[16486]: res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?! |
| 16:43.40 | _x86_ | ManxPower: is that patch committed to 1.4.12 already? |
| 16:44.07 | _x86_ | hesco: you need a clock source, like real TDM hardware |
| 16:44.12 | _x86_ | hesco: or, ztdummy |
| 16:44.12 | ManxPower | _x86_: I have no idea. check the commit date, if it is before 1.4.12 was released...... |
| 16:44.18 | ManxPower | _x86_: no he does not. |
| 16:44.29 | ManxPower | a clock source MIGHT help with that harmless message, but I doubt it. |
| 16:44.47 | ManxPower | hesco: if you want help here, we expect you to pay attention. |
| 16:44.48 | hesco | _X86_: might that be available with apt-get? |
| 16:44.51 | *** join/#asterisk FlatFoot (n=bigflatf@80.88.192.113) |
| 16:45.13 | *** part/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 16:45.29 | ManxPower | hesco: that is a harmless message, indicating that your system is not quite fast enough to handle everything without some extra latency. |
| 16:45.30 | hesco | ManxPower: understood. My boss isn't pointed headed, but she's still the boss |
| 16:46.09 | tzanger | ManxPower: I get hesco's message from time to time if something blocked the processor and caused it to not be able to send something out on time |
| 16:46.23 | ManxPower | BTW, I prefer the term "sync source" rather than "clock source", because everyone seems to thing "clock" means "system clock" and this is not the case. |
| 16:46.32 | ManxPower | tzanger: exactly. |
| 16:48.14 | hesco | so where again can I find ztdummy? |
| 16:48.36 | ai-a[afk] | hesco: google.com |
| 16:48.47 | _x86_ | ManxPower: dude... EVERY CALL in my scrollback buffer on the tail -f terminal monitoring the dtmf debug for the test case channel, has worked just fine since changing rxgain to -3 |
| 16:49.19 | ManxPower | _x86_: Damn I'm good. |
| 16:49.46 | ManxPower | _x86_: you MIGHT want to put a txgain=3 before the PSTN channels to compensate for the audio level on PSTN calls. |
| 16:49.58 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 16:50.49 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:50.54 | *** part/#asterisk fiXXXerMet (n=kjohnson@69.85.26.2) |
| 16:52.16 | unixdog | ok |
| 16:52.16 | Zeeek | In about 30 minutes, the VOIP Users Conference goes live for your enjoyment, edification or just if you want to waste your employer's time and bandwidth listening. http://voipusersconference.org |
| 16:52.16 | unixdog | its time things be said |
| 16:52.16 | *** join/#asterisk pepo-- (n=pepOSX@201.248.215.16) |
| 16:52.22 | Zeeek | IRC #voip-users-conference right here on Freenode |
| 16:52.41 | _x86_ | ManxPower: http://pastebin.ca/767265 |
| 16:52.51 | _x86_ | ManxPower: 11 out of 11 calls went through just fine now! |
| 16:52.59 | _x86_ | ManxPower: thanks SO MUCH :) :) :) |
| 16:53.34 | ManxPower | _x86_: You should send a large donation to the ManxPower Beer Fund PayPal eric@fnords.org |
| 16:53.40 | _x86_ | Zeeek: you're anti-shy enough as it is... damn |
| 16:53.48 | _x86_ | ManxPower: hehe |
| 16:54.23 | ManxPower | For some reason people always thing that's funny. |
| 16:54.33 | unixdog | well I want to see some major clen up done |
| 16:54.41 | unixdog | len/clean up |
| 16:54.44 | unixdog | in the code |
| 16:54.45 | _x86_ | ManxPower: wait, 12th call went just fine, then 13th failed |
| 16:54.54 | unixdog | all the mutex issues need to go |
| 16:54.58 | [TK]D-Fender | _x86_: lucky #13 strikes again. |
| 16:55.01 | unixdog | and the color in the console |
| 16:55.03 | _x86_ | ManxPower: 14th went ok, 15th failed |
| 16:55.13 | [TK]D-Fender | _x86_: Meatloaf said 2 out of 3 ain't bad, and you have 12/13! |
| 16:55.23 | Zeeek | mutex is the most obscene technical term after.... "intermittent" |
| 16:55.37 | ManxPower | _x86_: I think the dtmf fix is your real solution |
| 16:56.14 | _x86_ | ManxPower: yeah, but this will get me off the hook while i do the migration to 1.4 this weekend |
| 16:56.18 | unixdog | and I think that the show users is a security hole |
| 16:56.23 | ManxPower | _x86_: download the 1.2 version of the patch. |
| 16:57.50 | unixdog | because it shows passwords |
| 16:57.50 | ManxPower | unixdog: no, it is not. If you are allowed to use the CLI, you are an admin. |
| 16:58.08 | unixdog | well its a risk if you run any of the guis out there |
| 16:58.10 | *** join/#asterisk c0rnflake (n=c0rnflak@38.112.4.210) |
| 16:58.16 | unixdog | and have access to it on the web |
| 16:58.31 | unixdog | like the digium asterisk gui and thirdlane and freepbx |
| 16:59.20 | ManxPower | If you are using a GUI, you have much bigger problems than seeing the secrets |
| 16:59.20 | unixdog | it should not show passwords |
| 16:59.20 | unixdog | this is not going to be a gui war |
| 16:59.20 | ManxPower | unixdog: you poor naive thing. |
| 16:59.32 | tzanger | asterisk-gui is eventually being replaced with switchvox though isn't it? |
| 16:59.38 | unixdog | < far from niive |
| 16:59.45 | [TK]D-Fender | unixdog: more of a GUI massacre really ;) |
| 16:59.50 | unixdog | and I look at all security issues |
| 17:00.04 | unixdog | and its a security hole |
| 17:00.10 | ManxPower | unixdog: you would be better served by complaining about using plaintext passwords. |
| 17:00.16 | unixdog | it should never show the opasswords |
| 17:00.39 | unixdog | I yell about that all the time to providers but they wont listen |
| 17:00.40 | ManxPower | Well, good luck with getting it changed. |
| 17:00.58 | unixdog | they should be using md5 but they are to stupid |
| 17:01.20 | unixdog | and they say its just easier for us to do plain text |
| 17:01.33 | coppice | why are they stupid? do they stand to loose if something goes wrong? |
| 17:01.37 | unixdog | and letting the user choose a password and not encode it in md5 |
| 17:01.49 | unixdog | yes they do |
| 17:02.02 | unixdog | plaintext passwords are stupid |
| 17:02.07 | ManxPower | unixdog: Why do you think the password listed in the GUIs are taken from "sip show users" rather than directly from sip.conf??? |
| 17:02.09 | unixdog | I agree 100000% |
| 17:02.09 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 17:02.12 | Zeeek | If everyone always had fixed ip addresses... |
| 17:02.15 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:02.15 | coppice | like what, for instance? |
| 17:02.55 | ManxPower | unixdog: you are under the misconception that you can force users to act in a way that is good for security. |
| 17:03.01 | ManxPower | You can't. |
| 17:03.27 | unixdog | then it come to the software to be more secure |
| 17:03.39 | ManxPower | ManxPower: unixdog: Why do you think the password listed in the GUIs are taken from "sip show users" rather than directly from sip.conf??? |
| 17:03.44 | unixdog | wich means things like sip show users not showing password. |
| 17:04.23 | unixdog | they may be pulled from sip.conf. but they should not be displayed |
| 17:04.30 | unixdog | your missing the poing |
| 17:04.40 | unixdog | point |
| 17:05.30 | unixdog | the point is having a way to display what users are on the system is fine |
| 17:05.38 | ManxPower | unixdog: no, you are missing the point. It is up to the GUI to hide the password. The GUI has admin privs, so the GUI is responsible for being secure and masking passwords. |
| 17:05.41 | unixdog | but displaying the passwoerd is not secure |
| 17:05.49 | coppice | no. i think you are missing the point. who has the potential to seriously loose by what they do today, who also has the competance to assess and fix it? |
| 17:05.52 | *** join/#asterisk bantu (n=Miranda@p54A32C89.dip0.t-ipconnect.de) |
| 17:06.47 | unixdog | I am told its a sinple finx in chan_sip.c |
| 17:06.55 | ManxPower | unixdog: and chan_h323.c and chan_mgcp.c, and chan_sccp.c and chan_skinny.c |
| 17:07.41 | unixdog | ok |
| 17:07.41 | unixdog | so its in all of them |
| 17:07.41 | unixdog | but it should not be |
| 17:07.41 | ManxPower | but feel free to submit a patch to bugs.digium.com and you can argue with them about it. |
| 17:07.41 | unixdog | displaying all the other info is fine |
| 17:07.41 | ManxPower | unixdog: well and manager.conf, of course too./ |
| 17:07.41 | Zeeek | boyz |
| 17:07.41 | unixdog | but it should never display passwords |
| 17:07.43 | ManxPower | or complain on asterisk-dev |
| 17:07.51 | unixdog | thats a security hole |
| 17:07.55 | ManxPower | (which is the correct place to discuss code changes) |
| 17:08.14 | unixdog | every time I have gone there no one responds |
| 17:08.20 | unixdog | so I come here |
| 17:08.25 | ManxPower | that is because nobody cares. |
| 17:08.28 | *** join/#asterisk irule (n=irule@200.53.61.4) |
| 17:08.33 | unixdog | thats sad |
| 17:08.44 | Zeeek | too many windows for one thing! |
| 17:08.47 | ManxPower | go use a different PBX if it is so important to you. |
| 17:09.10 | [TK]D-Fender | unixdog: Since when would someone with CLI access not have access to the raw config files anyways? |
| 17:09.27 | unixdog | asterisk is the only one I have working on bsd and I am active in keeping it ported |
| 17:09.37 | ManxPower | [TK]D-Fender: he does not want to listen. |
| 17:09.46 | unixdog | I am listening |
| 17:09.48 | jennyw | hey everyone. I'm looking for a VoIP provider. I notice that a lot of them charge per phone number. Do any of them sell blocks of DIDs? |
| 17:10.16 | ManxPower | jennyw: blocks of DIDs is more of a wholesale thing for most companies. |
| 17:10.21 | unixdog | but even endusers for the most part can go look at the files if they have access to the box |
| 17:10.42 | jennyw | ManxPower: So is this something I'm not likely to get through a voip provider? |
| 17:10.48 | ManxPower | unixdog: THAT sounds like a permissions problem to me. |
| 17:11.18 | [TK]D-Fender | unixdog: there is a saying "Why worry about your hair when they are about to take your head?" |
| 17:11.28 | ManxPower | jennyw: not likely to get from a VoIP provider than only sells retail type of accounts. A VoIP provider that does wholesale or larger corporate type of accounts might have that service. |
| 17:12.16 | [TK]D-Fender | unixdog: Anyone with CLI access can do far worse damage than just passwords anyways, so who cares? |
| 17:12.18 | ManxPower | jennyw: most places that are large enough to need a block of DIDs is large enough to realize that betting the company's phone service on an internet connection is a bad idea. |
| 17:12.20 | jennyw | ManxPower: Is there a list of such providers somewhere? I looked at a few that offer business accounts (VoicePulse, Race.com) but I didn't see any info on larger blocks of numbers. |
| 17:12.44 | ManxPower | jennyw: e-mail them, I doubt they would have much info on their web sites. |
| 17:13.22 | jennyw | ManxPower: thanks. Yeah, I could see that most companies wouldn't want to depend on VoIP, but this is an experimental proof of concept type of project that they want to do cheaply (not for production). |
| 17:13.35 | ManxPower | ~cheep |
| 17:14.23 | ManxPower | ~ygwypf |
| 17:15.07 | jbot | methinks ygwypf is You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 17:15.07 | Zeeek | Oh I thought you were there |
| 17:15.07 | Zeeek | SHIIIIIT |
| 17:15.07 | ManxPower | jennyw: for proof of concept you really don't need a block of DIDs |
| 17:15.07 | c0rnflake | hey guys, quick question. i recently upgraded to asterisk 1.4 and updated my polycoms to sip 2.2 (at the same time, silly i know) |
| 17:15.10 | ManxPower | c0rnflake: did you read the UPGRADE.txt and CHANGES documents that are included in the 1.4 source code? |
| 17:15.10 | c0rnflake | so now incoming calls show up as 2125555555@192.168.0.10 instead of 2125555555, and when my users try to re-dial from missed calls, the call fails |
| 17:15.10 | c0rnflake | well, i should say i inherited the administration of the box, i didnt install it. i'll go read those over now. |
| 17:15.24 | dandre | hELLO? |
| 17:15.41 | Zeeek | hEllo |
| 17:15.50 | dandre | sorry ;-) |
| 17:17.14 | dandre | is there any possibility to test from the manager interface weather an extension is present in on context? |
| 17:18.30 | *** join/#asterisk Bladerunner05 (n=feelme@81-174-56-54.static.ngi.it) |
| 17:18.46 | Zeeek | Phone in your questions: http://x2z.eu |
| 17:18.53 | Bladerunner05 | Hola; I need to know, how to setup gpx2000 to (pressing msg button) login into mailbox automatically |
| 17:18.55 | davevg-btwtech | dandre, yes |
| 17:18.56 | [TK]D-Fender | dandre: Why would you? |
| 17:19.07 | *** join/#asterisk pepo--- (n=pepOSX@190.72.153.45) |
| 17:19.28 | De_Mon | ManxPower what if that company IS an internet provider? if their internet goes down they are out of business anyways... |
| 17:19.50 | dandre | I want to know if an extension number is available before essigning it |
| 17:20.39 | ManxPower | De_Mon: the provider does not have to go down for your service to fail. |
| 17:20.39 | dandre | davevg-btwtech: how? |
| 17:20.39 | ManxPower | your ISP and any ISP between you and the provider could go down. |
| 17:20.39 | De_Mon | oh, fine! be that way |
| 17:20.47 | De_Mon | unixdog hey, what did you say about colors? |
| 17:20.53 | De_Mon | don't be messin with my CLI colors |
| 17:21.07 | GreggB | ManxPower: my situation appears to be the inverse of jennyw - our T1 has failed so often in the past couple months, and the telco has been unable to resolve the issues, that we're considering a 100% migration to VoIP now (w/ 122 DIDs too). |
| 17:21.15 | davevg-btwtech | dandre: Command: dialplan show exten@context |
| 17:21.17 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 17:21.37 | ManxPower | GreggB: Best of luck with that. |
| 17:22.20 | De_Mon | its like anything else, its great, till it stops working. |
| 17:22.48 | dandre | ok thanks |
| 17:22.54 | coppice | some things are better when they stop working |
| 17:22.59 | ManxPower | I trust T-1s over internet connections. |
| 17:23.01 | unixdog | conf time |
| 17:23.06 | ManxPower | coppice: like Windows ME? |
| 17:23.25 | coppice | or a phone bringing me complaints |
| 17:23.27 | ManxPower | or Rev Phelps heart? |
| 17:25.02 | GreggB | ManxPower: As do I...normally... though in this case, our internet connect has been rock-solid (as have our backup VoIP providers), while the T1 has been down for far more than 80-90% of the time for the past couple months. It's not like we're in a "bad" area either; a few blocks from the city square, in Portland OR... |
| 17:25.54 | Bladerunner05 | Hola; I need to know, how to setup gpx2000 to (pressing msg button) login into mailbox automatically |
| 17:26.07 | ManxPower | GreggB: sounds like time for a new T-1 provider. |
| 17:26.19 | ManxPower | Bladerunner05: nobody here uses grandstream for obvious reasons. |
| 17:27.08 | GreggB | You'd think Integra would be good enough...I guess not. |
| 17:27.59 | ManxPower | We usually have less than 1 day of downtime per year on our T-1s |
| 17:28.16 | ManxPower | and we have something like 20 T-1s in various places. |
| 17:28.49 | ManxPower | I should have said "8 hours", not "day" |
| 17:29.17 | GreggB | In my last job, I managed upwards of 15 T1 (or T1-like circuits - ie: E1), mostly US-based.... I rarely saw downtime. |
| 17:31.08 | aiksa[LV] | Good evening everybody |
| 17:38.58 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 17:39.10 | teknoprep | does asterisk 1.4 have t.38 gateway or is it pass only? |
| 17:40.27 | puzzled | afaik it's pass only |
| 17:40.34 | *** join/#asterisk ricko73 (n=dhartman@wilug/newlug/ricko73) |
| 17:41.32 | Darthclue | GreggB, have they given you any explanation as to why it is down? When hurricane Ivan took out I10 it also took out some circuits that a t1 provider of ours (at the time) was using. This caused mass failure on their entire network because they didn't have a redundancy in place. The POTS system is notorious for this type of stuff. A good provider will be honest and tell you about it. |
| 17:42.07 | teknoprep | hmm switchvox looks nice |
| 17:42.26 | _x86_ | GreggB: you do know that you can't get commercial E1 circuits in the US right? |
| 17:42.40 | _x86_ | GreggB: E1 is NOT the same as T1 or J1 |
| 17:42.54 | *** join/#asterisk kv0s (n=kv0s@p4FD249E1.dip.t-dialin.net) |
| 17:42.57 | kv0s | Hi! |
| 17:43.17 | kv0s | I've a running asterisk, which works great. But one thing ... |
| 17:43.44 | _x86_ | GreggB: E1 is a European standard with 32 channels, T1 is a US standard with 24 channels, and J1 is a Japanese standard that uses chicken scratches instead of channels ;) |
| 17:43.56 | kv0s | ... my outgoing MSN on my HFC-ISDN Card isn't set corret on outgoing calls. Any where i must search? Zapata.conf? |
| 17:45.16 | *** join/#asterisk |omni| (n=rob@net82.allied-security.com) |
| 17:45.55 | GreggB | _x86_: yea, I'm aware of the technical differences... but they are "close enough" that I just slip into calling them "T1s" when talking to most folks...instead of explaining X number of T1s, X number of E1s, X number of... It's not like I called the T1s, an iDSL circuit :-) |
| 17:47.10 | *** join/#asterisk Gunirus (n=Gunirus@unaffiliated/Gunirus) |
| 17:48.00 | _x86_ | GreggB: it is exactly like you called a T1 an iDSL circuit |
| 17:48.09 | _x86_ | GreggB: T1 != E1 != J1 |
| 17:48.21 | _x86_ | T1 == T1, E1 == E1, J1 == J1 |
| 17:48.23 | GreggB | _x86_: alright...thanks |
| 17:49.42 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 17:49.52 | GreggB | Darthclue: their original blame was a bridge tap (which had been in place at our DMARC for ~6 months prior), then it was 50-pair riser bundle, then it was the card at the CO, then the termination equipment in our basement...then it was the Adtran they provided (splitting 4 data channels out, leaving the rest for voice). |
| 17:51.02 | _x86_ | is DSP-based EC better than octasic? |
| 17:51.10 | _x86_ | nvm, dumb question |
| 17:52.24 | [TK]D-Fender | _x86_: First step is admitting you have a problem ;) |
| 17:52.42 | _x86_ | tzanger: were you the one talking about the USB channel banks? |
| 17:52.50 | _x86_ | tzanger: they any good with Asterisk? |
| 17:52.56 | GreggB | Darthclue: The issues are still randomly reoccurring. I've already run a hard-loop on our Digium card, and all VoIP calls go through the Asterisk box fine - so my attention keeps returning to the circuit itself. |
| 17:52.58 | tzanger | _x86_: you'd have to as tzafrir |
| 17:53.11 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:53.11 | _x86_ | ah, get yall confused |
| 17:53.29 | *** join/#asterisk steve (i=steve@bouncer.stephen.marsh.name) |
| 17:53.45 | steve | hi all |
| 17:53.53 | steve | is there a difference between asterisk and asterisknow? |
| 17:54.17 | [TK]D-Fender | _x86_: yay, moving up from T1 CB's to something even LESS standard. Why don't you just go Grandstream 100% end be done with it? ;) |
| 17:54.32 | [TK]D-Fender | steve: AsteriskNOW is a distro including * and a shitty GUI |
| 17:54.46 | steve | [TK]D-Fender: including *? |
| 17:54.53 | [TK]D-Fender | steve: Yes |
| 17:54.57 | steve | * = ? |
| 17:55.00 | steve | everything? |
| 17:55.54 | J4k3 | ps - [TK]D-Fender needs to get laid. Unless its something he's configured or used previously its automagically "shit", and he'll be quite willing to stomp his foot repeatedly to ensure everyone around him knows it |
| 17:56.05 | GreggB | steve: The * is an asterisk...think about the channel you're on... |
| 17:56.18 | steve | oh, LOL |
| 17:56.24 | steve | yes.. I get it |
| 17:56.41 | [TK]D-Fender | steve: * = whats the name of that symbol again? Star? No wait, that wasn't it... Oh I know, ASTERISK <--- |
| 17:57.26 | [TK]D-Fender | J4k3: Nice try, but not quite. |
| 17:57.44 | steve | heh, I hadn't mentally linked the symbol and the name of the software |
| 17:57.46 | steve | my bad :) |
| 18:00.30 | steve | what's the verdict on trixbox? is it easier to deploy than the asterisk tarballs on an existing system? |
| 18:01.00 | [TK]D-Fender | steve: Sure, if you don't mind selling your soul to the lowest bidder :) |
| 18:01.20 | J4k3 | do you have anything technically useful to add, [TK]D-Fender ? |
| 18:01.21 | J4k3 | really |
| 18:01.26 | steve | interesting answer |
| 18:01.28 | J4k3 | every fucking time I join here all I read is your bitching |
| 18:01.31 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 18:01.37 | J4k3 | really, you need a new hobby |
| 18:01.40 | J4k3 | that doesn't involve #asterisk |
| 18:02.26 | Nugget | Hey. [TK]D-Fender contributes a lot more here than I do. I'm just here to whinge about how awful mysql is. :) |
| 18:02.36 | J4k3 | steve: trixbox requires no clue to get working, thats its only real advantage. its only disadvantage is listening to [TK]D-Fender piss and moan profusely about it. |
| 18:02.45 | _x86_ | who would think that two people in some random IRC channel would have nicks starting with "tza" |
| 18:02.46 | [TK]D-Fender | J4k3: And that's followed about your bitching about my bitching in greater proportions :) He asked an opinion and he got one. You just don't like them so ignore them and move on. |
| 18:03.17 | _x86_ | [TK]D-Fender: you think USB channel banks are bad? |
| 18:03.17 | _x86_ | [TK]D-Fender: you've used them? |
| 18:03.51 | Seldon75 | J4k3: [TK]D-Fender is knowlegable, but jaded |
| 18:04.10 | [TK]D-Fender | _x86_: It may work, but here's the downside (I do VALIDATE my opinions at least) : 1st directly tied to your * server in terms of failure risk and wiring restrictions. 2 : ONLY works with *. 3 : Resale / reusability value? Meh. |
| 18:05.08 | Seldon75 | I'd probably be jaded to if I had a million people asking me things contained in a well-distributed book because they couldnt be bothered reading it |
| 18:05.19 | _x86_ | [TK]D-Fender: ah, i was originally questioning it versus spending more money on another T1 card |
| 18:05.40 | [TK]D-Fender | _x86_: These factors make them a very dead end choice. Also Zaptel places a heavier load on your system, requires timing (can cause issues), and forces you to use Zaptel FXS handling (ICK!!!!). a SIP gateway takes all the load OFF of * and handles call feautres FOR YOU, can have phisical internal failover as well as extra routing options, somprehensive dialplan handling, etc. |
| 18:05.50 | _x86_ | [TK]D-Fender: was thinking that the astribank 24-port FXS solution would be cheaper than another dual-port T1 card, which does not seem to be the case |
| 18:07.03 | GreggB | It appears sRTP is under active development for * (http://bugs.digium.com/view.php?id=5413) has anyone tried it out yet? |
| 18:07.49 | [TK]D-Fender | Seldon75: Jaded? Not towards the few products I truely am (don't speak about in public here). I warn off where the greater pool of experience agrees anyways. Thing is that I am here constantly and AM the one you may see repeating it more often just because of my presence. |
| 18:07.58 | [TK]D-Fender | GreggB: Should end up in 1.6 |
| 18:08.20 | [TK]D-Fender | GreggB: But few people would run trunk in production so your odds are somewhat low... |
| 18:08.38 | [TK]D-Fender | GreggB: This is worth following up via the mailing-list or -dev |
| 18:08.52 | Seldon75 | fair enough |
| 18:09.03 | GreggB | [TK]D-Fender: great, thanks :-) |
| 18:09.29 | [TK]D-Fender | GreggB: Check the -dev channel and maybe you'll find someone to test with |
| 18:11.51 | *** join/#asterisk alexhopper (n=Alex@mctnnbsa24w-142167041234.pppoe-dynamic.nb.aliant.net) |
| 18:12.09 | *** join/#asterisk brea (n=brea@67.42.21.177) |
| 18:14.42 | jameswf | I would rather debug windows then boot bsd..... |
| 18:14.45 | jameswf | narf |
| 18:19.04 | Katty | herro file |
| 18:19.08 | Katty | and mister fender. |
| 18:20.42 | *** join/#asterisk tobias (n=tobias@cpe-069-134-226-227.nc.res.rr.com) |
| 18:21.03 | file | Katty: how are 'chu? |
| 18:22.04 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 18:24.11 | robl^ | hrmm.. are we on that oat bran muffin thing again? |
| 18:25.51 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:25.51 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 18:35.22 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 18:35.44 | *** join/#asterisk dijungal (n=kdaniel@209.59.110.30) |
| 18:36.24 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:36.24 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 18:38.23 | jameswf | would like to reference pouncing as the reason females fear computer people.... |
| 18:39.17 | steve | does asterisk come with a web GUI? |
| 18:39.23 | steve | or is that only asterisknow/trixbox |
| 18:39.45 | Qwell | steve: nope, but there is the "Asterisk GUI" which can be downloaded separately. |
| 18:39.48 | steve | ah |
| 18:39.53 | Qwell | (which is what comes with asterisknow) |
| 18:40.01 | steve | right |
| 18:40.04 | Qwell | (and trixbox, because they liked it so much) |
| 18:40.41 | steve | so if asterisk has this functionality available, and asterisknow is a distro in itself, what's the purpose of trixbox? |
| 18:42.32 | Qwell | [TK]D-Fender: ? |
| 18:42.47 | steve | lol, I guess you have a rather low opinion of it? :P |
| 18:43.00 | Qwell | that would be an understatement |
| 18:43.03 | jameswf | steve, trixbox has no purpose... download freepbx, or asterisk gui and avoid the fluff |
| 18:43.33 | steve | what does freepbx offer? |
| 18:43.34 | jameswf | and avoid the flame war in here lol |
| 18:43.42 | Qwell | steve: freepbx is the standard gui trixbox uses |
| 18:43.55 | jameswf | steve, frepbx is 95% of trixbox |
| 18:43.56 | Qwell | my opinion of that is rather low too though :p |
| 18:43.58 | [TK]D-Fender | steve: Trixbox used to be Asterisk@Home which was a distro like AsteriskNOW except that that project has been out for YEARS <-- |
| 18:44.07 | [TK]D-Fender | steve: AsteriskNOW is *new* |
| 18:44.10 | steve | does freepbx have any advantages over asterisk and asterisk GUI? |
| 18:44.23 | Qwell | steve: despite the name, freepbx is just a gui - for asterisk |
| 18:44.46 | steve | ah, is it a whole distro on its own? I'm not keen on that idea |
| 18:44.50 | Qwell | no |
| 18:44.56 | Qwell | it's just a gui |
| 18:44.59 | [TK]D-Fender | steve: FreePBX is a COMPLETE end-to-end solution, AsteriskGUI requires a little bit of hand made work. |
| 18:45.00 | lirakis | steve: asterisk qui is a framework for developing front ends to asterisk... it just happens to come with a pre built front end |
| 18:45.04 | jameswf | steve, all guis are a down fall. asterisk is so much more powerfull than what you can fit in a gui, pick up a book and go linux style on it |
| 18:45.13 | steve | aha |
| 18:45.26 | Qwell | lirakis: yeah, that sums it up pretty well |
| 18:45.32 | lirakis | steve: freepbx .. is a steaming pile of.. errr .. he he |
| 18:45.40 | Qwell | lirakis: You must work in our marketing dept :P |
| 18:45.50 | [TK]D-Fender | steve: You don't buy a professional painters kit to build a "Colour By Numbers" clidren's book :) |
| 18:46.02 | lirakis | Qwell: no but i had lunch with jsmith at von/asteriskworld .. maybe its rubbing off |
| 18:46.09 | lirakis | ;) |
| 18:46.12 | Qwell | [TK]D-Fender: You do when you're a hardcore color by numbers fanatic |
| 18:46.17 | Qwell | lirakis: that'd do it |
| 18:46.17 | steve | heh |
| 18:46.49 | [TK]D-Fender | <beavis> hehehe hard-core hehehe </beavis> |
| 18:46.52 | jameswf | i found a cook by numbers website, you enter your fridge and pantry contents, it spits out recipies |
| 18:47.15 | lirakis | <PROTECTED> |
| 18:47.21 | lirakis | ok end sophomoric humor |
| 18:48.00 | jameswf | its sad kids now a days dont get to have quality wholesome cartoons like bevis & butthead |
| 18:48.27 | jameswf | or ren and stimpy |
| 18:48.34 | *** join/#asterisk LivedType (n=LivedTyp@202.172.97.35) |
| 18:48.54 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:48.54 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 18:48.54 | stybba | they have pokemon |
| 18:49.00 | LivedType | If anyone wants to join our mini-conference we are holding about Asterisk (and maybe some off topic) you can call sip:999@202.172.97.34 |
| 18:49.01 | jameswf | bleh |
| 18:49.02 | steve | does asterisk GUI require a webserver or is one built in? |
| 18:49.14 | Qwell | steve: asterisk is the webserver |
| 18:49.22 | stybba | built in |
| 18:51.33 | *** join/#asterisk mrtelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 18:51.42 | mrtelephone | can I get asterisk to use Remote ID for caller name? |
| 18:52.17 | file | you mean remote party id? |
| 18:52.22 | *** part/#asterisk Bladerunner05 (n=feelme@81-174-56-54.static.ngi.it) |
| 18:52.23 | lmadsen | rpidtrust=yes |
| 18:52.28 | file | lmadsen: backwards |
| 18:52.30 | file | trustrpid=yes |
| 18:52.32 | lmadsen | doh! |
| 18:52.38 | lmadsen | I've forgotten :) |
| 18:52.45 | lmadsen | sendrpid=yes |
| 18:52.53 | lmadsen | as well |
| 18:53.06 | steve | http://sourceforge.net/projects/astguiclient << is that asterisk gui? |
| 18:53.44 | lmadsen | nope |
| 18:53.57 | steve | oh, got a download url? :) |
| 18:54.02 | steve | can't seem to find it |
| 18:55.20 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 18:56.31 | mrtelephone | its using the number from rpid but the name from the from header |
| 18:57.18 | *** join/#asterisk Eter4 (n=eter4@pop.nakinasystems.com) |
| 18:58.41 | mrtelephone | in the log anyways |
| 19:01.05 | *** join/#asterisk ghento (n=ghento@75.155.241.7) |
| 19:03.58 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 19:05.00 | [TK]D-Fender | steve: This isn't event he channel to really be asking about such thing. Read the channel topic fo links, and check out asterisk.org for links to AsteriskGUI/NOW |
| 19:05.14 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 19:06.27 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 19:06.50 | *** join/#asterisk agx (n=badpengu@81-174-44-16.dynamic.ngi.it) |
| 19:07.12 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:08.12 | Eter4 | Is there a way to use Asterisk as a Long Distance server. I would like to call into the PBX, punch in an ext and then get a dialtone to dial to call out to another number |
| 19:08.23 | hmmhesays | show application disa |
| 19:08.44 | hmmhesays | does jbot know about it? |
| 19:08.48 | hmmhesays | ~disa |
| 19:08.48 | jbot | from memory, disa is direct inward system access. show application disa |
| 19:08.49 | Nugget | Any question that begins "Is there a way to use Asterisk as..." has the answer "yes" |
| 19:08.58 | Nugget | it's simply a matter of how much time and effort you're willing to expend |
| 19:10.26 | hmmhesays | I need to get a compressor but i'm not sure which one |
| 19:10.35 | hmmhesays | ouch, d@mnit |
| 19:10.43 | Eter4 | Nugget: I guess I should of asked if someone has already built something like this beforehand? |
| 19:10.45 | Darthclue | what kind of compressor? |
| 19:10.52 | hmmhesays | for guitar |
| 19:11.03 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:11.03 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 19:11.13 | Darthclue | Eter4, look at DISA |
| 19:11.22 | Nugget | what's guitar? some sort of x11 front end to tar? ;) |
| 19:11.32 | _x86_ | lol |
| 19:11.36 | Darthclue | yeah, I don't have guitar installed, can't help ya there. |
| 19:12.22 | [TK]D-Fender | hmmhesays: Boss GT-8 ;) |
| 19:12.55 | Eter4 | Darthclue: Thanks |
| 19:13.02 | Eter4 | THat's exacly what I wanted! |
| 19:14.54 | hmmhesays | [TK]D-Fender: overkill |
| 19:15.00 | hmmhesays | I think i'm just going to get a dynacomp |
| 19:15.21 | [TK]D-Fender | hmmhesays: "just enough kill" IMO and means I haven't needed a seperate amp or anything for years now :) |
| 19:15.44 | hmmhesays | I like my pedals |
| 19:15.53 | [TK]D-Fender | hmmhesays: While you fiddle around with a huge pedal rack batteries, retweaking settings between songs, etc... |
| 19:16.10 | [TK]D-Fender | hmmhesays: Set once, use fast |
| 19:18.43 | [TK]D-Fender | hmmhesays: Good sample : http://www.youtube.com/watch?v=GVKT5JBnIaw |
| 19:18.48 | hmmhesays | I don't use batteries |
| 19:20.20 | *** join/#asterisk Teln12100 (i=hello123@bas2-toronto12-1128663474.dsl.bell.ca) |
| 19:20.44 | shido6 | neither do I |
| 19:20.58 | shido6 | not until im at least 80 |
| 19:21.14 | shido6 | if i keep the ginseng going i might not need them then either |
| 19:22.37 | hmmhesays | everything that guy played sounded very processed |
| 19:23.27 | JerJer | has anyone seen / figured out a way to setup different languages for the in-queue announcements based on the queue itself ? |
| 19:24.39 | *** join/#asterisk Schumie (n=Steve@212.183.134.66) |
| 19:25.11 | JerJer | i see that various ->lanquage elements getting passed around |
| 19:26.12 | JerJer | looks like just set the language via dialplan |
| 19:27.11 | *** join/#asterisk sacitec (n=tobi@189.149.99.172) |
| 19:27.28 | sacitec | hello |
| 19:27.42 | blitzrage | ya, I just think you set the language then have the /var/lib/asterisk/sounds/XX directory for the language... |
| 19:27.53 | JerJer | yeah - sounds logical now |
| 19:28.03 | sacitec | i'm looking for opinions about SIP/IAX trunks to LA, what company do u recommend ? |
| 19:28.31 | JerJer | didn't want to say yes we can do different language prompts without verifying :) |
| 19:28.53 | blitzrage | JerJer: I wouldn't say it until you try it :D |
| 19:29.28 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:29.30 | JerJer | oh i will - just haven't done different languages at all within asterisk |
| 19:29.53 | blitzrage | ya, me either |
| 19:30.37 | blitzrage | until I was wondering why my prompts weren't working with trunk after I upgraded a test box from 1.4 (they look in /var/lib/asterisk/sounds/en/ for english files now, but I didn't install the prompts again, so they only existed in /var/lib/asterisk/sounds/) |
| 19:30.51 | mrtelephone | I'll sip trunk you one in the balls :P |
| 19:33.36 | *** join/#asterisk Cherebrum (n=jgarland@pdpc/supporter/base/Cherebrum) |
| 19:34.46 | Cherebrum | FreeSwitch just added mod_dialplan_asterisk if anyone is interested. It means you can now also have the stupidity of the asterisk dialplan on FreeSwitch. :) http://fisheye.freeswitch.org/changelog/FreeSWITCH/?cs=6205 |
| 19:34.53 | *** join/#asterisk nestAr (i=nester@makes.all.the.girlies.go.wewt.wewt.net) |
| 19:35.31 | *** join/#asterisk Voicemeup (n=VoiceMeU@modemcable132.108-83-70.mc.videotron.ca) |
| 19:35.32 | nestAr | anyone mess with manager through Net::Telnet? |
| 19:35.32 | Nugget | telnet is eeeeeeevil! |
| 19:35.35 | nestAr | lol |
| 19:35.39 | nestAr | telnet? |
| 19:35.41 | *** part/#asterisk Cherebrum (n=jgarland@pdpc/supporter/base/Cherebrum) |
| 19:35.58 | *** join/#asterisk [intra]lanman (n=lanman@va-76-6-209-153.dhcp.embarqhsd.net) |
| 19:36.15 | tzafrir_home | Nugget, telnet is used ot convert \n to \r\n |
| 19:36.43 | *** join/#asterisk icewaterman (n=immagine@i53875C3B.versanet.de) |
| 19:36.49 | icewaterman | hi |
| 19:37.03 | tzafrir_home | nestAr, there's a Asterisk::Manager module as part of the Asterisk modules (which are now in CPAN) |
| 19:37.04 | icewaterman | octvqe_linux.c:1: error: CPU you selected does not support x86-64 instruction set <-- why is that? and why the hell is it in line 1? |
| 19:37.16 | icewaterman | i am crosscompiling on x86_64 for i386 |
| 19:37.22 | icewaterman | the rest of the kernel compiles fine |
| 19:37.29 | *** join/#asterisk flujan (n=flujan@201-27-90-218.dsl.telesp.net.br) |
| 19:37.33 | flujan | hi all. |
| 19:37.35 | destructure | the asterisk kernel? lol |
| 19:37.42 | flujan | I am having a problem with asterisk . |
| 19:37.47 | tzafrir_home | icewaterman, what card do you have? |
| 19:37.48 | nestAr | tzafrir_home: the one with no documentation? i played with it for a second, but the lack of docs scared me off. |
| 19:37.57 | icewaterman | destructure: hfcsusb |
| 19:38.06 | flujan | It is restarting sometimes during the day. It looks like someone is logged in the cli and typed restart now. |
| 19:38.08 | nestAr | i have the telnet thing working, except for the line breaks aren't what i'm expecting. |
| 19:38.10 | icewaterman | tzafrir_home: but i suspect this to be card independent |
| 19:38.11 | *** part/#asterisk [intra]lanman (n=lanman@va-76-6-209-153.dhcp.embarqhsd.net) |
| 19:38.23 | flujan | I enabled the full log but no hint about what is happening... |
| 19:38.25 | JerJer | I think Barry Bonds is having issues with Asterisk as well :D |
| 19:38.32 | JerJer | well 'an' Asterisk |
| 19:38.35 | tzafrir_home | hfcusb? with misdn? visdn? |
| 19:38.50 | flujan | I am using 1.4.13 |
| 19:38.52 | icewaterman | tzafrir_home: with misdn? what is visdn? |
| 19:39.20 | tzafrir_home | icewaterman, where are you getting this error from? building what, exactly? |
| 19:39.36 | *** join/#asterisk kambei (n=kambei@unaffiliated/kambei) |
| 19:39.43 | icewaterman | tzafrir_home: i am building the misdn modules |
| 19:39.44 | tzafrir_home | the kernel? |
| 19:39.53 | icewaterman | tzafrir_home: yes, i used std2kern for that |
| 19:40.32 | tzafrir_home | well, I don't know misdn well enough |
| 19:40.34 | icewaterman | tzafrir_home: CONFIG_MISDN_DSP is the reason, if i skip that it works |
| 19:40.34 | kambei | [TK]D-Fender, I came back to thank you for the help yesterday. I forget the name of the other guy that helped me. It was regarding call files. You pointed me to THE BOOK. I wanted to express my appreciation. |
| 19:40.50 | [TK]D-Fender | kambei: Glad to help |
| 19:41.39 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 19:41.51 | kambei | [TK]D-Fender: Alright. I'll be back to thank the other guy. Take care. |
| 19:42.05 | icewaterman | tzafrir_home: maybe it is better i use a stable version and do not build misdn from git |
| 19:42.39 | *** join/#asterisk moemoe (i=moemoe@kuschelhoelle.netzhure.de) |
| 19:43.01 | flujan | [TK]D-Fender: do you have problems where asterisk 1.4.13 restarting? |
| 19:43.22 | moemoe | hi guys. i want to use asterisk with my hfc on *bsd. can you tell me which *bsd has matching drivers for that card? |
| 19:43.54 | *** part/#asterisk khronos (n=khronos@c-66-229-159-201.hsd1.fl.comcast.net) |
| 19:46.17 | moemoe | okay, just found http://www.voip-info.org/wiki/index.php?page=FreeBSD+zaptel - i think ill try freebsd |
| 19:48.08 | icewaterman | btw, i want asterisk to behave as a voip -> isdn telephony gateway. i assume it can do that, can it? |
| 19:48.23 | icewaterman | to act would be more appropriate. |
| 19:49.08 | stybba | hi... what means this warnings??? http://pastebin.com/d1dd26e89 |
| 19:50.54 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:50.54 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 19:51.15 | tzafrir_home | moemoe, is that included in standard BSD ports? |
| 19:51.24 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 19:51.29 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582580.dsl.bell.ca) |
| 19:51.33 | tzafrir_home | FreeBSD ports, that is |
| 19:51.36 | moemoe | tzafrir_home: Its also in the ports misc/zaptel (Check http://www.pbxpress.com/~gonzo/ to download file.) |
| 19:52.14 | tzafrir_home | I get 403 on that page |
| 19:54.47 | moemoe | oh, /me 2 |
| 19:55.10 | moemoe | but i think freebsd is the better choice on a p90 than linux. |
| 19:55.27 | moemoe | this box just has to convert isdn to sip, without changing codecs or anything |
| 19:55.32 | moemoe | max. 2 lines |
| 19:55.39 | [TK]D-Fender | I think that a real computer is a better choice than a P90. |
| 19:56.04 | moemoe | hey. this still is a real computer. only somewhat outdated ;) |
| 19:56.07 | [TK]D-Fender | moemoe: Considering most cards REQUIRE PCI2.2 compliance |
| 19:56.11 | moemoe | and all my other boxes need too much powers |
| 19:56.17 | [TK]D-Fender | moemoe: Like by a DECADE |
| 19:56.47 | moemoe | [TK]D-Fender: i know. but i also still like my nes and atari :) |
| 19:57.08 | lirakis | moemoe: nes 8 bit is the bomb |
| 19:57.38 | [TK]D-Fender | lirakis: NES rocks on the Motorola Q :) |
| 19:58.00 | hmmhesays | I was thinking about getting a Q |
| 19:58.07 | hmmhesays | what winmo version you have? |
| 19:58.19 | tzafrir_home | moemoe, how much memory do you have? |
| 19:58.26 | moemoe | lirakis: i went to rgb2r.de last weekend, only hardware older than 10 years is allowed. playing nintendo world cup on a beamer at 3 a clock in the morning with big speakers and loud music rocks :D |
| 19:58.45 | lirakis | moemoe: he he.. |
| 19:59.21 | lirakis | [TK]D-Fender: hrmm... not a fan of the Q really (kinda unrelated) ... it seems any time i touch one it freezes |
| 20:00.07 | [TK]D-Fender | lirakis: Bell is scheduled to release the HTC Touch (possibly Duo) mid month, and is rumoured to be going for a 7$ unlimited data plan. |
| 20:00.19 | lirakis | [TK]D-Fender: huzzah!? |
| 20:00.31 | [TK]D-Fender | lirakis: Indeed |
| 20:00.35 | _x86_ | ugh |
| 20:00.40 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 20:00.44 | icewaterman | tzafrir_home: yes, stable version builds perfectly |
| 20:00.45 | [TK]D-Fender | hmmhesays: Wait for the touch... it rocks. |
| 20:00.48 | _x86_ | one of my analog phones somehow put his extension in DND mode |
| 20:00.55 | icewaterman | icewaterman: so i am going to stay with that |
| 20:00.58 | lirakis | [TK]D-Fender: im not a windows mobile fan.. frankly ... |
| 20:01.00 | _x86_ | how do i disable DND completely? |
| 20:01.01 | [TK]D-Fender | hmmhesays: Q = WM5 "smart phone", not "PDA" |
| 20:01.09 | *** join/#asterisk _omer (i=omer@203.128.20.222) |
| 20:01.13 | lirakis | [TK]D-Fender: neo1973 here i come! |
| 20:01.24 | lirakis | im ill never crash that lol |
| 20:01.33 | [TK]D-Fender | lirakis: Palm, or iPhone? We are rumoured to get that through our GSM carriers for Christmas at extortion pricing. |
| 20:01.46 | *** part/#asterisk Voicemeup (n=VoiceMeU@modemcable132.108-83-70.mc.videotron.ca) |
| 20:01.52 | [TK]D-Fender | lirakis: Oh, MOW we're talking... but thats once they get it to a sane level of operation. |
| 20:02.05 | [TK]D-Fender | NOW* |
| 20:02.10 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 20:02.28 | lirakis | [TK]D-Fender: i have a BB now... i got funambol on it so i can sync finally |
| 20:02.59 | lirakis | [TK]D-Fender: i want to read the LJ interview.. it just came in the mail yesterday |
| 20:03.57 | tobias | _x86_: i'm having a problem where i none of my phones ring, they just go to voicemail straight away |
| 20:04.05 | tobias | _x86_: is that the same thing you're talking about? |
| 20:04.20 | _x86_ | tobias: i'm having that problem with a single zap channel... user dialed something to put the channel in DND |
| 20:04.22 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 20:04.49 | tobias | how annoying |
| 20:05.06 | tzafrir_home | is this one of chan_zap's special "features"? |
| 20:05.36 | Strom_M | _x86_: so have them dial *79 to turn it off :) |
| 20:06.20 | *** join/#asterisk implicit (n=implicit@m5d5e36d0.tmodns.net) |
| 20:06.21 | tzafrir_home | _x86_, make an extension *78 that does something else :-( |
| 20:06.47 | tzafrir_home | And please file a bug report that thus this is a bad idea |
| 20:07.23 | tzafrir_home | I have already suggested removeing this pointless code and got fried on asterisk-dev |
| 20:07.23 | _x86_ | Strom_M: is there a way to remove that feature completely? |
| 20:07.24 | _x86_ | features.conf does not explicitly enable it |
| 20:07.28 | tzafrir_home | patch chan_zap.c and rebuild asterisk |
| 20:07.50 | tzafrir_home | look for "*78" (with the quotes) |
| 20:08.14 | _x86_ | i don't even have anything in my dialplan for *78 / *79 |
| 20:08.29 | tzafrir_home | Right. It is implemented inside chan_zap.c |
| 20:08.35 | _x86_ | how is it that it's not explicitly defined in features.conf, nor in extensions.conf, and it's still able to work? |
| 20:08.44 | _x86_ | oh man that's weak :( |
| 20:09.12 | tzafrir_home | Hence the workaround of having the extension *78 :-( |
| 20:09.30 | [TK]D-Fender | _x86_: Thats right, keep on ignoring me and using Zap FXS :p |
| 20:09.47 | moemoe | oh, it's even a k6-2 350mhz |
| 20:10.46 | *** join/#asterisk cypherdelic (n=cypher@p5B27CA85.dip.t-dialin.net) |
| 20:12.17 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 20:13.04 | *** join/#asterisk Darthclue (n=e054502@fw149.nisd.net) |
| 20:13.48 | _x86_ | Strom_M: had them dial *79, got this: |
| 20:13.49 | _x86_ | [2007-11-09 14:12:59] -- Starting simple switch on 'Zap/9-1' |
| 20:13.50 | _x86_ | [2007-11-09 14:13:01] -- Disabled DND on channel 9 |
| 20:14.21 | _x86_ | then, tried to call that extension, and got this: |
| 20:14.22 | _x86_ | [2007-11-09 14:13:11] -- Executing [s@macro-stdexten:1] Dial("IAX2/rpc-pbx-peo-02-711", "Zap/9|20|tT") in new stack |
| 20:14.25 | _x86_ | [2007-11-09 14:13:11] == Everyone is busy/congested at this time (1:1/0/0) |
| 20:15.09 | _x86_ | i made sure he was off the phone when i tried calling that extension also |
| 20:15.12 | lmadsen | _x86_: in the future, please use pastebin |
| 20:15.24 | _x86_ | lmadsen: oh man you use IRC? :) |
| 20:15.31 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.133.162) |
| 20:15.32 | lmadsen | of course not |
| 20:15.43 | _x86_ | lmadsen: never seen you around here I guess |
| 20:15.52 | _x86_ | ah! |
| 20:15.59 | _x86_ | you are leif right? |
| 20:16.11 | blitzrage | that's what the rumours say |
| 20:16.18 | _x86_ | I've got you on my LinkedIn :) |
| 20:16.23 | blitzrage | :) |
| 20:16.53 | *** join/#asterisk implicit_ (n=implicit@68.156.43.202) |
| 20:17.26 | _x86_ | so question for you... I've got a channel that I can't dial... I made sure the channel is NOT in DND by having the user dial *79, and getting "Disabled DND on channel 9" in CLI |
| 20:17.49 | _x86_ | any ideas why asterisk would still be deflecting calls to that channel? |
| 20:18.43 | *** join/#asterisk mercutioviz (n=chatzill@66-17-33-47.biz.visl.arrival.net) |
| 20:21.02 | tzafrir_home | _x86_, any chance it is simply busy? |
| 20:21.12 | tzafrir_home | zap show channel 9 |
| 20:21.24 | tzafrir_home | See if it is off-hook |
| 20:22.27 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:22.27 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 20:23.01 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 20:23.50 | Katty | file: am goodly. howsre you? |
| 20:26.44 | *** part/#asterisk agx (n=badpengu@81-174-44-16.dynamic.ngi.it) |
| 20:26.58 | _x86_ | Hookstate (FXS only): Onhook |
| 20:27.53 | _x86_ | tzafrir_home: no, it's hung up |
| 20:29.29 | tzafrir_home | _x86_, plese enable debug, and pastebin what you see in debug / full log when you try to call |
| 20:29.56 | Seldon75 | hi, can I reset an individual Zap channel without affecting the others, which are tied up with calls? |
| 20:30.03 | Seldon75 | help zap |
| 20:30.30 | [TK]D-Fender | Seldon75: "soft hangup [channel]" |
| 20:30.34 | *** join/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 20:30.36 | Seldon75 | i tried: zap destroy channel 7 bu it killed _all_ the channels |
| 20:30.47 | Seldon75 | aha |
| 20:30.52 | hesco | I'm trying to get the * server to record a prompt, and the console comes up saying: Incoming call: Got SIP response 400 "BadRequest" back from 192.168.0.105. And I get a pop-up messages from my soft phone on the desktop saying my audio is not full duplex. What does it mean and what must I do to address this? |
| 20:31.26 | *** join/#asterisk fskrotzki (n=fskrot@host.textwise.com) |
| 20:32.26 | *** join/#asterisk Greek-Boy (n=Greek-Bo@41.221.58.2) |
| 20:33.07 | tzafrir_home | Seldon75, 'zap destroy channel' kills a "phisical" channel for Asterisk. Asterisk cannot re-add it without restarting |
| 20:33.30 | tzafrir_home | (or 'zap restart or whatever if you have just analog channels) |
| 20:33.51 | Seldon75 | i see, that was my mistake |
| 20:34.52 | *** join/#asterisk techie (n=techie@adsl-76-240-176-149.dsl.lsan03.sbcglobal.net) |
| 20:34.54 | *** join/#asterisk jstew (n=jstewart@fw-ext.fusionary.com) |
| 20:35.06 | _x86_ | http://pastebin.ca/767522 |
| 20:35.16 | *** part/#asterisk simond (n=simon@208.68.95.5) |
| 20:35.23 | hesco | Here are my full set of errors: http://paste.debian.net/41942 |
| 20:35.54 | jstew | Greetings. Anyone know what the best firmware to use for polycom ip501 phones and asterisk 1.4.13 is? |
| 20:37.23 | [TK]D-Fender | Seldon75: Zap destroy really wrecks your system. You may as well restart * cold. |
| 20:37.44 | Seldon75 | ok, gotcha |
| 20:37.51 | Seldon75 | thx |
| 20:37.56 | [TK]D-Fender | _x86_: "show channels" please... |
| 20:38.14 | _x86_ | hold on, i did zap destroy channel 9 |
| 20:38.23 | _x86_ | reload chan_zap.so didn't bring it back hehe |
| 20:38.35 | _x86_ | so i have to get everyone off the phone to do a zap restart |
| 20:39.44 | [TK]D-Fender | _x86_: Don't you just LOVE Zaptel FXS? ;) |
| 20:40.06 | *** join/#asterisk Rhinoo_ah (n=ahonea@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 20:40.30 | _x86_ | the only thing i dont like is the forced use of DND |
| 20:41.45 | Kobaz | anyone know any good docs on getting h323 phones going |
| 20:42.12 | _x86_ | interestingly, restarting asterisk fixed the DND issue ;) |
| 20:42.18 | _x86_ | now i can call that channel again |
| 20:45.31 | [TK]D-Fender | Kobaz: Like I told you before... go ask for your money back ;) |
| 20:47.00 | _x86_ | Kobaz: dont waste your money on h323 phones, they are crap |
| 20:47.21 | *** join/#asterisk fiXXXerMet (n=kjohnson@69.85.26.2) |
| 20:48.17 | fiXXXerMet | Are there any soft phones that can act as a front-desk device (where the receptionist would normally sit)? |
| 20:48.45 | fiXXXerMet | Would be neat to use a large touchscreen lcd with a soft phone for the front desk. |
| 20:49.13 | Greek-Boy | I want to configure my IP phones via web interface but I'm running my voice network on a seperate network so I'm trying to configure my asterisk box which is running debian to route traffic between two interfaces. no luck :( |
| 20:50.58 | [TK]D-Fender | Greek-Boy: try ##linux |
| 20:51.14 | [TK]D-Fender | Greek-Boy: this is Linux Networking 101 to setup forwarding. |
| 20:52.33 | robl^ | That is like asking: My asterisk box has a keyboard where the "z" key sticks. How do I clean it? |
| 20:54.04 | tzafrir_home | _x86_, have you set debug to a decent value? |
| 20:54.13 | tzafrir_home | e.g: 5 or so? |
| 20:54.41 | [TK]D-Fender | robl^: Run it throught the dishwasher |
| 20:54.49 | *** part/#asterisk mercutioviz (n=chatzill@66-17-33-47.biz.visl.arrival.net) |
| 20:54.49 | *** join/#asterisk grantm (n=grant@kolob.wingateservices.com) |
| 20:54.49 | tzafrir_home | #debian , rather |
| 20:55.39 | *** join/#asterisk grantm (n=grant@kolob.wingateservices.com) |
| 20:55.46 | robl^ | [TK]D-Fender: ahh! I tried a pressure washer and sand blasting, followed a sulfuric acid dip |
| 20:56.22 | [TK]D-Fender | robl^: "John was here but is no more, for what he thought was H2O was H2SO4" :D |
| 20:56.32 | Greek-Boy | lol |
| 20:56.33 | Greek-Boy | guys |
| 20:56.36 | Greek-Boy | i did all the stuff |
| 20:56.45 | Greek-Boy | there's something wrong on the routers |
| 20:56.59 | Greek-Boy | most importantly, I did echo 1 > /proc/sys/net/ipv4/ip_forward |
| 20:57.00 | tzafrir_home | what? the dishwasher? |
| 20:57.09 | tzafrir_home | or /etc/network/options? |
| 20:57.23 | Greek-Boy | thats only for boot time : |
| 20:58.56 | Greek-Boy | tcpdump shows 23:56:08.205078 IP 172.16.77.7.1101 > 192.168.211.46.80: S 2424727961:2424727961(0) win 65535 <mss 1460,nop,nop,sackOK> |
| 20:59.05 | Greek-Boy | thats when I try to browse to the phone |
| 20:59.21 | *** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 20:59.55 | tzafrir_home | and you have set up the routing properly on both sides, right? |
| 21:00.42 | tzafrir_home | that system is the default router for the phone? |
| 21:01.09 | Greek-Boy | yip |
| 21:01.43 | tzafrir_home | can you ping the phone from the router? |
| 21:01.53 | Greek-Boy | yes I can |
| 21:03.50 | tzafrir_home | obviously the phone can't route back packets to 172.16.77.7 |
| 21:06.48 | De_Mon | why not? |
| 21:06.57 | [TK]D-Fender | question is is the * system the default gteway on BOTH ends, or linked as a gatway to each end's respective subnets? |
| 21:12.09 | Greek-Boy | how would I force dhcpd to renew leases? |
| 21:19.54 | Katty | [TK]D-Fender: which one is it that you plug an analog line into... |
| 21:19.56 | Katty | [TK]D-Fender: fxo? |
| 21:20.23 | Katty | [TK]D-Fender: and if it is fxo, then what do they use fxs for? |
| 21:22.20 | [TK]D-Fender | Katty: FXO is for plugging in telco lines. FXS is for plugging in regular PHONES. |
| 21:22.35 | [TK]D-Fender | Katty: as opposed to using SIP phones like you Polycoms |
| 21:22.36 | Katty | [TK]D-Fender: analog phones? |
| 21:22.44 | [TK]D-Fender | Katty: Yes |
| 21:22.55 | Katty | <bkw> NEXT! |
| 21:23.02 | Katty | [TK]D-Fender: also, thank you |
| 21:23.27 | [TK]D-Fender | Katty: np |
| 21:24.37 | Mw3 | what is this annoying spam on www.voip-info.org? |
| 21:24.56 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 21:25.29 | Mw3 | oh its gone. i just needed a reload :) |
| 21:26.55 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 21:27.20 | *** part/#asterisk fiXXXerMet (n=kjohnson@69.85.26.2) |
| 21:32.21 | *** join/#asterisk seanmh (i=fiber0pt@216.31.101.41) |
| 21:33.01 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:34.47 | *** join/#asterisk metfan2007 (n=root@189.135.178.159) |
| 21:34.51 | metfan2007 | hi all |
| 21:35.56 | *** join/#asterisk |dennis| (n=dennis@200.32.236.18) |
| 21:46.50 | _x86_ | Mw3: it's called "information" |
| 21:47.18 | _x86_ | people "read" it |
| 21:48.43 | _x86_ | as a result of such "reading," people become "knowledgeable" |
| 21:48.57 | _x86_ | ;) |
| 21:49.08 | *** join/#asterisk |dennis| (n=dennis@200.32.236.18) |
| 21:50.18 | *** join/#asterisk steve (i=steve@bouncer.stephen.marsh.name) |
| 21:50.54 | steve | hmm, are X100P cards recognised automatically by generic kernels or are special modules needed? |
| 21:51.12 | steve | I'm using a generic centos kernel, kudzu found it during boot but asterisk can't find it |
| 21:51.40 | *** join/#asterisk dijungal (n=kdaniel@63.175.159.171) |
| 21:56.37 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 21:57.09 | *** join/#asterisk WindBack (n=Administ@host72.190-31-68.telecom.net.ar) |
| 21:57.28 | tzafrir_home | It's called "spam". As a result of rading it, people remove it from voip-info.org |
| 21:57.50 | tzafrir_home | steve, you need Zaptel |
| 21:58.02 | tzafrir_home | It's not in mainline kernel |
| 21:58.07 | WindBack | I have a TDM400p How I can avoid to run ztcf every time I restart my server?? |
| 21:58.39 | tzafrir_home | WilliamK, use /etc/init.d/zaptel at startup before asterisk is started |
| 21:58.45 | *** join/#asterisk rantsh (n=rantsh@190.36.185.139) |
| 21:58.53 | rantsh | hi everyone |
| 21:58.58 | tzafrir_home | How is asterisk started? |
| 21:59.00 | tzafrir_home | what distro is it? |
| 21:59.22 | Rhinoo_ah | generally anyways |
| 21:59.45 | WindBack | tzafrir_home, Debian |
| 21:59.51 | WindBack | tzafrir_home, ETCH |
| 22:00.19 | rantsh | anyone ever implemented some sort of secure VoIP |
| 22:00.22 | Rhinoo_ah | is asterisk in your sbin? |
| 22:00.35 | WindBack | tzafrir_home, * is started with the script from the sources |
| 22:00.41 | rantsh | not only TLS-ing SIP messages but encrypting voice too? |
| 22:00.42 | *** part/#asterisk zerohalo (n=zeroHalo@pool-71-255-162-167.bstnma.east.verizon.net) |
| 22:00.45 | dmz | if i don't have SHA1 function, where would i find it to build/install? |
| 22:00.49 | Rhinoo_ah | usually its in /etc/init.d/asterisk start |
| 22:01.04 | dmz | rantish, closed network (router based vpn) |
| 22:01.05 | WindBack | Rhinoo_ah, yes, it is |
| 22:01.18 | tzafrir_home | WindBack, ls /etc/rc2.d # do you have both Asterisk and Zaptel there? Zaptel before Asterisk? |
| 22:02.01 | tzafrir_home | rantsh, I think you can do that on IAX between two Asterisk servers |
| 22:02.24 | WindBack | tzafrir_home, and where is the zaptel script? |
| 22:02.45 | WindBack | tzafrir_home, in the sources? |
| 22:02.52 | tzafrir_home | yes |
| 22:02.58 | rantsh | tzafrir_home, how about phone - Asterisk??? |
| 22:03.32 | tzafrir_home | rantsh, are you limited to using a specific phone? |
| 22:04.25 | rantsh | tzafrir_home, not really |
| 22:07.50 | dmz | show function SHA1....nothing shown :( |
| 22:08.27 | *** join/#asterisk asdx (n=diego@67-207-128-81.slicehost.net) |
| 22:09.23 | asdx | ~book |
| 22:09.23 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 22:11.42 | *** join/#asterisk WindBack (n=Administ@host208.190-30-185.telecom.net.ar) |
| 22:12.19 | WindBack | tzafrir_home, sorry, I lost the conection |
| 22:12.26 | WindBack | tzafrir_home, tzafrir_home, For the asterisk daemon I use the script from /astriskSources/contrib/init.d/rc.debian.asterisk, but Where is the script for zaptel?? |
| 22:13.17 | tzafrir_home | zaptel.init in the zaptel sources |
| 22:13.38 | tzafrir_home | wget http://svn.digium.com/svn/zaptel/branches/1.4/zaptel.init |
| 22:13.48 | tzafrir_home | chmod +x zaptel.init |
| 22:14.10 | WindBack | tzafrir_home, I have to copy it to /etc/init.d and the update-rc.d |
| 22:14.11 | WindBack | ? |
| 22:15.13 | WindBack | tzafrir_home, update-rc.d is the command used in debian to add services to init.d |
| 22:15.41 | tzafrir_home | WilliamK, make sure it is run before asterisk . You know z > a |
| 22:16.04 | tzafrir_home | yes, use updat-erc.d |
| 22:16.09 | tzafrir_home | yes, use update-rc.d |
| 22:17.15 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-227-239.dsl.irvnca.pacbell.net) |
| 22:18.31 | *** join/#asterisk Windback (n=Administ@host110.190-30-196.telecom.net.ar) |
| 22:19.02 | Windback | tzafrir_home, Does this script work well in debian?? |
| 22:19.10 | *** join/#asterisk [TK]D-Fender (n=joe_blow@64.235.216.2) |
| 22:19.14 | tzafrir_home | טקד |
| 22:19.17 | tzafrir_home | yes |
| 22:19.37 | [hC] | love how cisco 7940's seem to like to ignore the factory reset key sequence whenever i try. |
| 22:19.39 | [hC] | ugh. |
| 22:20.26 | Windback | tzafrir_home, I saw that the function restart from the script asterisk doesn't work well in debian |
| 22:21.31 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.pa.comcast.net) |
| 22:21.38 | tzafrir_home | What do you mean? |
| 22:21.52 | tzafrir_home | what did you expect? what happened? |
| 22:27.00 | *** join/#asterisk techie (n=techie@adsl-76-240-176-149.dsl.lsan03.sbcglobal.net) |
| 22:27.59 | *** join/#asterisk yannj_fr (n=yannj_fr@vpn.intelunix.fr) |
| 22:30.17 | *** join/#asterisk Windback (n=Administ@host60.190-138-93.telecom.net.ar) |
| 22:30.41 | Windback | tzafrir_home, sorry, my connection is not working well |
| 22:31.17 | tzafrir_home | so again: what do you mean by "not working"? what did you expect? what happened? |
| 22:32.40 | Windback | tzafrir_home, for example, if I made changes in sip.conf, if I use /etc/init.d/asterisk restart, the changes dont take effect |
| 22:32.55 | hmmhesays | does the moto q have wifi capabilities? |
| 22:33.22 | Windback | tzafrir_home, so, I have to do /etc/init.d/asterisk stop and then start |
| 22:33.25 | tzafrir_home | WilliamK, this is not related to Zaptel |
| 22:33.41 | Windback | tzafrir_home, of course |
| 22:33.48 | tzafrir_home | Windback, not WilliamK , sorry |
| 22:33.59 | yannj_fr | hello, as anyone a dsp ccard from Digium? |
| 22:34.16 | tzafrir_home | a transcoder card? |
| 22:35.17 | tzafrir_home | Windback, it seems to use safe_asterisk by default, which I don't really like |
| 22:35.29 | tzafrir_home | The debian debs have a more polished script |
| 22:35.51 | yannj_fr | yes |
| 22:36.02 | yannj_fr | a transcoder |
| 22:36.13 | yannj_fr | I bougth one |
| 22:36.27 | Windback | tzafrir_home, ok |
| 22:36.30 | yannj_fr | but when booting, loading the driver take a really long time |
| 22:37.46 | Windback | tzafrir_home, coming back to the zaptel theme. I put the scrip in init.d before the asterisk script, but it does't start the zap interface when I restart the server :( |
| 22:38.32 | yannj_fr | and I have the same pb with my E1 card |
| 22:38.52 | Siya | which is best to follow from svn for *? |
| 22:38.57 | Siya | branch or trunk? |
| 22:39.24 | tzafrir_home | if you don't explicitly run ztcfg or anything, what do you get in /proc/zaptel/* ? |
| 22:40.28 | yannj_fr | Siya : branch is stable version |
| 22:41.01 | tzafrir_home | Siya, branches/1.4 if you don't follow development closely |
| 22:41.29 | Siya | cool thanks |
| 22:43.40 | yannj_fr | tzafrir : do you have any idea about my problem when loading module? |
| 22:45.39 | Windback | t |
| 22:45.49 | Windback | tzafrir_home, http://www.pastebin.ca/767656 |
| 22:47.29 | Rhinoo_ah | in /proc/zaptel/* that reads the info from the zaptel card...the channels that are provisioned by zaptel |
| 22:47.42 | tzafrir_home | Windback, so it's indeed not configured |
| 22:48.11 | tzafrir_home | gone again... |
| 22:48.13 | Rhinoo_ah | when it is configured it will say the signalling and then (in use) |
| 22:48.42 | tzafrir_home | Rhinoo_ah, Windback has some connectivity problems... |
| 22:49.17 | Rhinoo_ah | im noticin... |
| 22:49.42 | Rhinoo_ah | is it connectivity or configuration? |
| 22:50.12 | *** join/#asterisk BillBinko (n=BillBink@65.210.151.194) |
| 22:50.13 | tzafrir_home | And generally he complained that he needed to explicitly run ztcfg at boot time |
| 22:50.27 | *** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 22:51.25 | *** join/#asterisk gicode (n=gicode@scorn.csh.rit.edu) |
| 22:51.42 | BillBinko | Hi everyone, having choppy sound on SIP connections now that I've switched to a new multi-processor server... am I missing something obvious? |
| 22:51.57 | BillBinko | disabled acpi/apic for testing (no change) |
| 22:52.16 | Katty | Wooohoo 5pm!!!! |
| 22:54.42 | BillBinko | Any pointers to timing/ztdummy issues? |
| 22:55.24 | BillBinko | (I have tried to post this to asterisk-users but my posts have not appeared -- I have a request in with the list admin for help) |
| 22:55.25 | *** part/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 22:56.20 | *** join/#asterisk CVirus (n=GoD@196.205.192.246) |
| 22:57.01 | jameswf | I just installed a viagra module on my server, now it ha fast reaction and long up times |
| 22:57.22 | BillBinko | congratulations |
| 22:57.28 | Mavvie | see what happens when it dumps it core. |
| 22:57.43 | gicode | Hello, I bought a x100p fxo card and got the asterisk call-in demo working, but now it is holding the line off-hook all the time. Since then I have tried the card in two other machines and both just hold the line off-hook. Anyone know what might be up? |
| 22:58.30 | BillBinko | Calls that last longer than 4 hours are *not normal* |
| 22:58.37 | BillBinko | (sorry, running gag) |
| 22:59.01 | jameswf | gicode, does your telco have far end disconnect supervision |
| 23:01.34 | TimGroe | oh damn, BillBinko, I have a call that is reaching 4:00:00 now |
| 23:01.55 | BillBinko | Ok, perhaps something more specific: I am getting these zttest results with ztdummy loaded |
| 23:01.56 | BillBinko | --- Results after 15 passes --- |
| 23:01.56 | BillBinko | best: 99.963379 -- Worst: 99.938965 -- Average: 99.953613 |
| 23:01.56 | BillBinko | <PROTECTED> |
| 23:02.43 | gicode | jameswf: not really sure. I am in a US College dorm room. It was disconnecting alright last night, but even after a machine restart it won't accept even one call. |
| 23:03.44 | gicode | What I mean is, last night I was able to call multiple times. Now I can't call once, even after a reboot. |
| 23:07.08 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 23:07.43 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-68-90-41-25.dsl.hstntx.swbell.net) |
| 23:08.10 | gicode | Is there a way to manually tell the channel to hang up the line? |
| 23:08.20 | gicode | like, in the console |
| 23:08.39 | Strom_M | gicode: soft hangup |
| 23:09.03 | Strom_M | gicode: check for disconnect supervision on the phone line |
| 23:10.46 | *** join/#asterisk ManxPower (n=manxpowe@179.sub-75-203-131.myvzw.com) |
| 23:13.08 | gicode | Strom_M: soft hangup claims that Zap/1 is not a known channel, but zap show channels shows a channel 1. |
| 23:13.23 | gicode | Strom_M: what is the proper way to check for disconnect supervision on the phone line? |
| 23:13.46 | *** join/#asterisk Mercestes (n=Merceste@c-76-30-153-176.hsd1.tx.comcast.net) |
| 23:14.08 | jameswf | a telco disconnect in most cases is a battery pull, simply unplug the wire for 1/2 second |
| 23:14.25 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 23:15.40 | Strom_M | well, it's called a "battery drop", not a battery pull |
| 23:15.49 | Strom_M | gicode: tab-complete is your friend |
| 23:16.36 | gicode | jameswf: unplugging the wire resets the line on the telco side, but the x100p doesn't hang-up |
| 23:16.58 | Strom_M | gicode: no no no. |
| 23:16.59 | jameswf | so you have a hardware issue |
| 23:17.15 | Strom_M | gicode: what kind of signaling did you put the card in? |
| 23:17.27 | gicode | Strom_M: fxs_ks |
| 23:17.33 | Strom_M | ok |
| 23:17.55 | Strom_M | now plug a regular phone directly into that phone line and check for disconnect supervision |
| 23:17.57 | jameswf | gicode, check your firmware |
| 23:18.28 | jameswf | Strom_M, if the card doesnt hang up on battery removal it doesnt ,matter |
| 23:19.18 | Strom_M | jameswf: i understand that, but knowing whether the telco is also working properly is important too |
| 23:19.23 | jameswf | gicode, you may also look in zttool and see if zaptel is signalling the hangup, if so then asterisk is pooched |
| 23:20.04 | Mercestes | pooched? |
| 23:23.20 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 23:23.33 | gicode | If I had asterisk in the wrong signalling mode for a bit would that kill the hardware? |
| 23:24.26 | gicode | jameswf: I am not sure if this card has firmware... It was only $20 |
| 23:25.37 | ManxPower | jameswf: X100Ps and clones do not have user modifiable firmware (if they even have firmware at all) |
| 23:25.50 | ManxPower | gicode: what country are you in? |
| 23:26.03 | gicode | ManxPower: US |
| 23:26.30 | ManxPower | gicode: chances are you have ha compatibility issue between the card and the motherboard. It's not that uncommon with that card. |
| 23:27.01 | ManxPower | gicode: I assume you are DIRECTLY connected to the telco and that telco is one of the original Bell companies (even if their name changed)? |
| 23:27.20 | ManxPower | gicode: has anyone told you to check the IRQs? |
| 23:28.14 | gicode | ManxPower: I am in a College Dormitory; I figure there is probably a pbx in the building here as we have to dial 9 to get out |
| 23:28.36 | gicode | ManxPower: I ran lspci and I didn't see any IRQ sharing |
| 23:28.37 | ManxPower | gicode: *nod* Most PBXs do not provide battery drop on their analog lines. |
| 23:28.52 | ManxPower | gicode: lspci shows you BEFORE the IRQ reassignment the kernel does. |
| 23:29.12 | gicode | ManxPower: hmm |
| 23:29.31 | ManxPower | cat /proc/interrupts is what you want |
| 23:29.55 | ManxPower | gicode: if unplugging the phone line does not make asterisk hangup the call, then the problem is almost certinally hardware. |
| 23:30.08 | ManxPower | the X100P is the only analog card that actually looks for voltage on the port. |
| 23:30.17 | ManxPower | (from Digium, at least) |
| 23:31.16 | *** join/#asterisk killfill_ (n=killfill@pc-164-134-45-190.cm.vtr.net) |
| 23:31.44 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-227-239.dsl.irvnca.pacbell.net) |
| 23:32.07 | gicode | ManxPower: doesn't look like it is sharing IRQ's; plus I have tried it in 3 different machines with different equipment |
| 23:33.04 | gicode | ManxPower: probably a hardware issue |
| 23:33.22 | gicode | I guess I should just pay some real cash for something that will work |
| 23:33.33 | ManxPower | *nod* Those X100P clones can have "issues". |
| 23:34.58 | ManxPower | gicode: your best bet is a SIPura SPA-3000. They are a bitch to get the FXO port working well with Asterisk, but they are the cheapest solution. |
| 23:35.37 | ManxPower | http://www.sipura.com/products/spa3000.htm |
| 23:35.55 | ManxPower | I would not use them in a production business enviroment, but they are fine for playing around with. |
| 23:36.49 | ManxPower | gicode: I assume the "three different machines" were different models/brands? |
| 23:37.52 | gicode | ManxPower: Yea, two custom built 8 years apart and one dell poweredge 1400 |
| 23:38.09 | tzanger | http://www.ldc.upenn.edu/myl/llog/OpusMindLick.gif |
| 23:39.22 | ManxPower | Oh! Wait! I hate the copyright police! Nevermind. |
| 23:39.37 | tzanger | hahaha |
| 23:39.41 | gicode | Thanks everyone for your help |
| 23:40.25 | gicode | Probably the most responsive irc channel I have been on |
| 23:41.26 | *** part/#asterisk gicode (n=gicode@scorn.csh.rit.edu) |
| 23:41.54 | ai-a[afk] | my girlfriend is responsive, but not in the way i would like. |
| 23:43.16 | Mercestes | your girlfriend is an irc channel? |
| 23:43.37 | tzanger | I was just going to ask the same thing |
| 23:43.40 | Mercestes | ascii porn anyone? |
| 23:44.37 | `Sauron | libaa |
| 23:45.57 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 23:46.07 | ai-a[afk] | she might as well be. |
| 23:46.49 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 23:46.59 | [TK]D-Fender | ai-a[afk]: Does she split like 20 times a day like #asterisk does? ;) |
| 23:47.12 | Mercestes | lol |
| 23:47.13 | ai-a[afk] | lol |
| 23:47.15 | [TK]D-Fender | ok, that was bad.....even for me... |
| 23:47.23 | [TK]D-Fender | :D |
| 23:48.24 | Mercestes | yea, it's pretty bad when you say something that even I don't have a response to. |
| 23:48.32 | ai-a[afk] | but she gets a slitting headache every night.. so i end up sleeping on the sofa, while the cat gets to share the bed instead. |
| 23:48.35 | *** join/#asterisk duxy786 (n=duxy786@host81-155-227-21.range81-155.btcentralplus.com) |
| 23:48.39 | duxy786 | hi all |
| 23:48.43 | ai-a[afk] | *splitting. |
| 23:49.04 | duxy786 | Anyone used Asterisk in conjuction with OpenSER |
| 23:49.19 | Mercestes | ai-a[afk], so she disses you for another kitty? |
| 23:49.26 | ai-a[afk] | duxy786: Operser is a port forwardingin. |
| 23:49.29 | ai-a[afk] | and asterisk is a service. |
| 23:50.11 | Mercestes | duxy786, see if Clona is around in #ser |
| 23:50.23 | duxy786 | thanks |
| 23:50.36 | Mercestes | I think google asterisk-ha might have some tips too but, I'm not even close to proficient in SER. |
| 23:51.03 | Mercestes | Or asterisk really, if you believe half of what D-Fender says about me. >.> |
| 23:52.26 | [TK]D-Fender | All of it is true... ESPECIALLY the lies ;) |
| 23:54.22 | Mercestes | indeed. |
| 23:54.26 | *** join/#asterisk fskrotzki (n=fskrot@cpe-74-74-245-250.rochester.res.rr.com) |
| 23:54.56 | *** join/#asterisk blq (i=Bl@dslb-088-064-146-217.pools.arcor-ip.net) |
| 23:56.38 | duxy786 | basiacally, I have now set up 4 production servers with * which have 4 E1 circuits each |
| 23:58.28 | duxy786 | The problem I have is now when somene comes in to server 2 to join a conferenece on server 1 it causes grief for me! |
| 23:59.16 | Mercestes | you would likely need a dedicated conference server with all the other servers pointing the meetme extensions back to yoru conferencing server. |
| 23:59.18 | duxy786 | am having to put each conferecne deails on to a DA and pull that off, its releven server and thenn place a sip call and DTMF in to the conference |
| 23:59.59 | duxy786 | if I could find a resolve for this, I'd be over the moon, but I duabt it will be a simple straight forward thing |
