| 00:00.01 | *** part/#asterisk IPmonger (n=ipmonger@c-71-224-31-246.hsd1.pa.comcast.net) |
| 00:02.51 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 00:05.32 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 00:05.32 | *** mode/#asterisk [+o mog] by ChanServ |
| 00:06.00 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 00:06.35 | *** join/#asterisk rob0 (n=rob0@cardinal.lizella.net) |
| 00:06.36 | *** join/#asterisk NOT_guru (n=chatzill@209.145.181.55) |
| 00:07.59 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 00:08.18 | Juggie | hmmm i think my cable modem is hosed |
| 00:08.21 | Juggie | it keeps crashing |
| 00:10.11 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 00:10.39 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 00:12.36 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 00:18.51 | shido6 | Pay your bill. |
| 00:19.05 | Juggie | heh |
| 00:19.08 | Juggie | i pay them plenty. |
| 00:19.47 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 00:27.21 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 00:32.52 | *** join/#asterisk tako-san (n=Tako-san@154.5.212.245) |
| 00:34.04 | *** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM0012c9213a06.cpe.net.cable.rogers.com) |
| 00:34.55 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 00:41.02 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 00:42.16 | *** join/#asterisk joobie (n=joobie@joobie.org) |
| 00:43.16 | joobie | Hey Guys.. I have a polycom conference phone, however the cables are all over the desk for the ethernet.... is there a solution anyone has to fix this? I mean like some sort of small wireless device, or sumthen? How is this normally handled??? |
| 00:45.49 | `Sean | Quick question how do you enable Colours in CLi? |
| 00:47.36 | snuff-work | joobie, i think.. polycom has a new wifi model.. but dont hold me to that |
| 00:49.11 | snuff-work | http://www.polycom.com/usa/en/products/voice/small_medium_conference_room/soundstation2w.html |
| 00:52.16 | joobie | damn |
| 00:52.24 | joobie | i just bought a wired one :/ |
| 00:52.28 | joobie | have to see if i can change it |
| 00:52.58 | snuff-work | i have no idea how it connects etc.. just knew they bought out a new wifi :) |
| 00:53.08 | joobie | when did it come out snuff? |
| 00:53.20 | joobie | it says they have a battery recall.. so i think it uses battery for power |
| 00:53.29 | snuff-work | think i got a promo email 1-2 months ago |
| 00:54.05 | joobie | ahh k |
| 00:54.09 | joobie | thanks snuff |
| 00:55.05 | [TK]D-Fender | SoundStation2W + ATA <-------- |
| 00:55.16 | [TK]D-Fender | the @w is a wireless analog unit. |
| 00:55.38 | [TK]D-Fender | 2W |
| 00:56.01 | snuff-work | yer but ata's arent always horrible |
| 00:56.05 | `Sean | [TK]D-Fender how do you enable colours for CLi?> |
| 00:56.17 | snuff-work | colours should be on by default |
| 00:56.27 | `Sean | snuff work there not.... |
| 00:56.30 | snuff-work | when u go asterisk -r.. should be coloured |
| 00:56.37 | `Sean | i go rasterisk -T |
| 00:56.41 | `Sean | and no colours... |
| 00:57.06 | snuff-work | mm.. no idea mines always worked :( |
| 00:57.17 | Juggie | put a hole in the desk? :P |
| 00:57.38 | snuff-work | u'd think more boardroom tables would have a hole in the middle |
| 00:57.53 | Juggie | most decent ones do |
| 00:58.20 | snuff-work | what we really need is a boardroom table with a hole covered by a holographic image so it doesn't appear like there is one |
| 00:58.26 | snuff-work | :P |
| 00:59.33 | *** join/#asterisk mtaht4 (n=m@74.9.53.154) |
| 00:59.47 | `Sean | i guess no one knows how to enable colours |
| 00:59.50 | `Sean | if i stop asterisk |
| 00:59.55 | `Sean | and restart with asterisk -vvvvvc |
| 01:00.04 | `Sean | v beinf for verbosity and c for colour |
| 01:00.04 | `Sean | then i get colour |
| 01:00.11 | `Sean | but say i exit cli and then reenter no colour |
| 01:00.44 | snuff-work | so asterisk -r when u get in.. no colour.. |
| 01:00.57 | `Sean | no colour |
| 01:08.31 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 01:10.00 | snuff-work | only thing i can think of is ur terminal isn't saying it can support ANSI colour |
| 01:12.42 | *** join/#asterisk infowolfe_ (n=infowolf@unaffiliated/infowolfe) |
| 01:13.08 | rob0 | Same here, I don't know how to enable color, but I use a terminal which supports it. |
| 01:13.23 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 01:16.20 | *** join/#asterisk kusznir (n=kusznir@66-233-138-60.lew.clearwire-dns.net) |
| 01:20.29 | *** join/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) |
| 01:22.00 | Cybertoy | hihi .. I have a problem with WaitMusicOnHold() ... I always get an error that asterisk is unable to find a translation from ulaw to "unknown" ... however MusicOnhold works e.g. for parked calls or when placing someone on hold. |
| 01:22.13 | Cybertoy | Or WaitExten ... |
| 01:22.25 | Cybertoy | but WaitMusicOnhold doesn't work |
| 01:23.27 | Cybertoy | <PROTECTED> |
| 01:23.30 | Cybertoy | I'll try that first. |
| 01:23.35 | Cybertoy | I'm on 1.4.7.... |
| 01:33.53 | *** join/#asterisk fx0 (n=fx0@cypher.punk.net) |
| 01:36.38 | *** join/#asterisk phocus (n=phocus@c-75-65-161-101.hsd1.ms.comcast.net) |
| 01:37.13 | phocus | has anyone used the Grandstream GS-286 adpater? http://www.voipsupply.com/product_info.php?products_id=35 i am thinking about getting one, it is the cheepest i can find that looks decent |
| 01:38.12 | Qwell | s/ that looks decent// |
| 01:38.29 | `Sean | Qwell can you help with the colour issue please? |
| 01:38.50 | phocus | Qwell: true, true, but i am a poor man |
| 01:39.55 | *** join/#asterisk joobie (n=joobie@joobie.org) |
| 01:50.32 | *** join/#asterisk [reed] (n=reed@firefox/gnu.webmaster.reed) |
| 01:57.12 | kusznir | Hi all: I'm curious as to what it would take to "properly" handle E911 calls "in-house" (without going through a company like 911etc.com). I'm assuming I already have an accurate extension to location database. |
| 01:58.08 | kusznir | I noticed on asterisk's features page, it lists E911 as one of its features, but I haven't found any details on implementation. |
| 02:06.20 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
| 02:07.03 | *** join/#asterisk mattwj2005 (n=matt@c-76-17-133-96.hsd1.mn.comcast.net) |
| 02:07.20 | mattwj2005 | any voip parties going on? |
| 02:13.22 | *** join/#asterisk Strom_M (n=strom@ip70-170-60-8.lv.lv.cox.net) |
| 02:13.50 | *** join/#asterisk MrMister2 (n=mrmister@89.181.118.96) |
| 02:20.18 | *** join/#asterisk [hC] (n=hardcore@rmtw.com) |
| 02:25.16 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 02:29.03 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 02:31.21 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 02:36.09 | *** join/#asterisk hohum (n=dcorbe@dhcp64-134-231-213.shs.nyc.wayport.net) |
| 02:42.51 | *** join/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 02:49.13 | *** join/#asterisk supjigator (n=sysgod@64.147.217.5) |
| 02:54.34 | *** part/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) |
| 02:58.31 | *** join/#asterisk irule (n=chatzill@189.162.94.215) |
| 03:02.24 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 03:10.44 | yakkop | trying to hangup on a channel via deadagi, it returns a value of 1 indicating success, but it doesn't actually hangup -- whats' the problem? |
| 03:14.53 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:31.41 | *** join/#asterisk mattwj2005 (n=matt@c-76-17-133-96.hsd1.mn.comcast.net) |
| 03:32.09 | mattwj2005 | sip:17475552663@proxy01.sipphone.com |
| 03:32.09 | mattwj2005 | 253286# |
| 03:32.09 | mattwj2005 | if anyone wants voice |
| 03:32.09 | mattwj2005 | :) |
| 03:32.44 | mattwj2005 | any takers? |
| 03:34.15 | yakkop | matt, any ideas? |
| 03:34.18 | yakkop | trying to hangup on a channel via deadagi, it returns a value of 1 indicating success, but it doesn't actually hangup -- whats' the problem? |
| 03:34.47 | mattwj2005 | hey yakkop :) |
| 03:35.06 | yakkop | hi |
| 03:35.10 | mattwj2005 | is it possible you should be getting 0? |
| 03:35.11 | mattwj2005 | :S |
| 03:35.31 | yakkop | I've got agi debug on and sip debug on |
| 03:35.32 | mattwj2005 | I am an asterisk beginner/hobbyst |
| 03:35.43 | yakkop | it shows receiving the hangup command |
| 03:35.47 | mattwj2005 | dump some stuff in the pastebin |
| 03:35.56 | yakkop | and shows a result of 1 (which means success for hangup) |
| 03:36.09 | mattwj2005 | what do you hear? |
| 03:36.09 | yakkop | but nothing happens -- no SIP BYE or anything |
| 03:36.21 | mattwj2005 | okay hmmm |
| 03:36.37 | mattwj2005 | what type of DTMF are you using? |
| 03:36.44 | mattwj2005 | there are four :D |
| 03:36.56 | yakkop | rfc2833 but I don't see why that would matter... |
| 03:37.05 | yakkop | not doing anything with dtmf here |
| 03:37.20 | mattwj2005 | yakkop I am not sure.... |
| 03:37.30 | mattwj2005 | i just got off a 36 hour work week :S |
| 03:37.41 | yakkop | ok, no problem |
| 03:37.44 | mattwj2005 | over the course of 3 24 hour days :) |
| 03:37.52 | mattwj2005 | if you can.... |
| 03:38.08 | mattwj2005 | join the voice conference..... |
| 03:39.32 | mattwj2005 | okay first off |
| 03:39.35 | mattwj2005 | what is agi? |
| 03:39.54 | mattwj2005 | application graphic interface? |
| 03:40.33 | *** join/#asterisk rkeels (n=rkeels@c-67-171-15-176.hsd1.wa.comcast.net) |
| 03:40.35 | *** join/#asterisk orbikitti (n=orbitn@68-119-111-162.dhcp.jcsn.tn.charter.com) |
| 03:41.10 | rkeels | so can any one here help with a WARNING error |
| 03:41.28 | mattwj2005 | what does it say? |
| 03:41.52 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 03:42.24 | rkeels | when I do a dialplan reload I get a WARNING[7720]: pbx.c:955 pbx_find_extension: Maximum PBX stack exceeded |
| 03:42.40 | mattwj2005 | *shocked* |
| 03:42.41 | *** part/#asterisk orbikitti (n=orbitn@68-119-111-162.dhcp.jcsn.tn.charter.com) |
| 03:42.48 | mattwj2005 | how big is your dialplan? |
| 03:42.52 | shmaltz | rkeels, any include files? |
| 03:43.10 | mattwj2005 | how many lines we talking? |
| 03:43.13 | rkeels | No include files but it is pretty long |
| 03:43.26 | rkeels | What is the max number of lines allowd |
| 03:43.32 | shmaltz | rkeels, do snow dialplan what does the last line say? |
| 03:43.35 | mattwj2005 | I am guessing you'll need to shrink it some how |
| 03:43.35 | rkeels | I get about 7 lines of the error upon reload |
| 03:43.42 | shmaltz | rkeels, we are about to find out |
| 03:43.45 | shmaltz | :P |
| 03:44.05 | mattwj2005 | *needs to read his Asterisk box some year* |
| 03:44.12 | rkeels | hold on it will take a minute to find out. |
| 03:44.23 | mattwj2005 | your not on the clock.... |
| 03:44.37 | mattwj2005 | at least I am not....*looks for Diet Mt Dew* |
| 03:45.19 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 03:45.48 | shmaltz | pbx.c has this line |
| 03:45.50 | shmaltz | #define AST_PBX_MAX_STACK 128 |
| 03:46.19 | shmaltz | but I know I have way more than 128 lines, but when it exceeds that that's when it gives you that error |
| 03:47.19 | shmaltz | the line on top says: |
| 03:47.21 | shmaltz | /* Go no deeper than this through includes (not counting loops) */ |
| 03:47.22 | shmaltz | #define AST_PBX_MAX_STACK 128 |
| 03:47.29 | shmaltz | which means that your loops are way toooooooooooooo nested |
| 03:47.48 | shmaltz | sorry that your includes not loops |
| 03:47.51 | rkeels | it doesn't say how many lines when I do a "dialplan show" instead it says "340 extensions (632 Priorities) in 33 contexts |
| 03:47.57 | rob0 | Froot Loops |
| 03:48.16 | shmaltz | rkeels, well you have more than 128 deep includes, is that possible? |
| 03:50.11 | rkeels | it shows a lot of Context "ael-yatayatayata' stuff but I don't use ael. infact I have extensions.ael renamed and I am using 1.4 |
| 03:50.41 | rkeels | I guess somehow somewhere ther is 128 deep includes like you say. |
| 03:50.46 | shmaltz | rkeels, then add in modules.conf a noload for ael |
| 03:51.08 | rkeels | aha I will try that. Hold on |
| 03:52.59 | *** join/#asterisk bmg505 (n=leon@196.209.176.153) |
| 03:53.04 | rkeels | what is the full syntax for that? is it = noload => ael or noload => extensions.ael |
| 03:56.54 | *** join/#asterisk z0rb4 (n=z0rb4@c-67-188-69-103.hsd1.ca.comcast.net) |
| 03:57.56 | rkeels | I think it is noload => pbx_ael.so |
| 03:57.59 | rkeels | Is that right |
| 03:59.13 | z0rb4 | do you know how to remove the voicemail directory when the extension has been deleted ? |
| 03:59.39 | justdave | rm -rf /var/spool/asterisk/voicemail/default/<exten> |
| 04:00.09 | justdave | you'd need to remove it from voicemail.conf, too |
| 04:00.44 | z0rb4 | ok nothing else like remove extension blah@context ? |
| 04:06.12 | z0rb4 | found a cool trick for cisco 7970 running sip, I use to press # to avoid delay but it sent it as part of the extension |
| 04:07.21 | z0rb4 | in you dialplan.xml do this |
| 04:07.27 | z0rb4 | <TEMPLATE MATCH="*#" Timeout="0" Rewrite="%1"/> |
| 04:07.48 | z0rb4 | this will strip the # out of the extension |
| 04:08.33 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 04:10.26 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 04:18.34 | *** join/#asterisk nuonguy (n=john@yona.ursus.net) |
| 04:25.09 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 04:25.28 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 04:35.39 | *** join/#asterisk LoneShadow (n=vshivash@59.92.141.159) |
| 04:35.40 | *** join/#asterisk friedrich| (n=friedric@e177247155.adsl.alicedsl.de) |
| 04:36.25 | LoneShadow | what debug commands will show if a iax2 register call is failing ? |
| 04:37.16 | LoneShadow | or is there a "sip set debug ip xxx.xxx.xxx.xxx" equivalent for IAX ? |
| 04:48.05 | justdave | iax2 set debug |
| 04:48.20 | justdave | I haven't found a way to specify a host to do debugging for |
| 04:48.24 | justdave | I was having that problem earlier |
| 04:48.36 | justdave | mine turned out to be a missing nat=yes on the iax.conf entry |
| 04:49.10 | justdave | iax2 set debug off <- to disable it when you're done |
| 05:10.19 | *** part/#asterisk kiwoneka (n=kiwoneka@KTNRON06-1168103823.sdsl.bell.ca) |
| 05:42.59 | *** join/#asterisk HaMYaI (i=HaMYaI@125-25-131-123.adsl.totbb.net) |
| 05:43.16 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 05:43.41 | HaMYaI | what player do we use for moh in "*.asf" format? |
| 05:48.32 | [TK]D-Fender | HaMYaI, That s M$ streaming format. Convert it or else..... |
| 05:49.08 | coppice | .... the bogey man will get you |
| 05:50.02 | HaMYaI | coppice: oh, you are here =) |
| 05:51.14 | HaMYaI | coppice: about the blocking line signal after clear forward it's actually a clear back, isn't it |
| 05:51.42 | coppice | yes |
| 05:52.46 | HaMYaI | coppice: I am sending a pulse of 150 msec every sixteenth seconds and then the calling end sends me the clear forward |
| 05:53.17 | HaMYaI | coppice: it should be normal then |
| 05:55.58 | *** join/#asterisk gardo (n=gardo@121.97.208.254) |
| 05:56.45 | coppice | do you expect the calling end to clear forward? |
| 05:57.27 | *** join/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 05:58.21 | HaMYaI | coppice: yeah, when the caller hangs up |
| 05:59.12 | coppice | that sounds OK then. 16s sounds rather fast for billing pulses |
| 06:00.09 | HaMYaI | coppice: the first pulse will be sent when the caller enters "1" to enter the service |
| 06:00.58 | HaMYaI | coppice: and after 16s the next one will be sent, does it sound OK? |
| 06:03.17 | coppice | How should I know? you should know what is required by your particular provider |
| 06:04.18 | HaMYaI | coppice: their requirements is just to have the metering pulse sent to them after the callers press "1" |
| 06:04.48 | coppice | so why do you keep sending them every 16s? |
| 06:05.46 | HaMYaI | coppice: I thought it's ITU Q400 series' requirement |
| 06:06.12 | coppice | where did you see that? |
| 06:07.07 | HaMYaI | coppice: from dialogic global call's log file and from the test equipment from telco side |
| 06:08.08 | HaMYaI | coppice: now they asked me while I just kept sending that pulse, but I don't actually know how I can remove that |
| 06:09.04 | HaMYaI | coppice: it's from the software we bought 10 years ago, so I am trying to get unicall to work with this telco instead |
| 06:10.20 | coppice | if you are trying to emulate what old equipment does, that's fine. billing pulses aren't really covered in the Q.400 specs, though. they are national variants |
| 06:12.24 | HaMYaI | coppice: in zaptel.conf you said that to put ":1111" after the assigned channels for thatland, but is it the same for all teco? |
| 06:13.01 | coppice | no. most countries require 1011 |
| 06:13.25 | coppice | I think china and thailand are the only places requiring 1111 |
| 06:14.12 | HaMYaI | coppice: I had " unicall_report: MFC/R2 UniCall/116 R2 prot. err. [2/ 2/Seize ack /Seize ack ] cause 32773 - Unexpected CAS bit pattern", should I change from "1111" to "1011"? |
| 06:14.21 | HaMYaI | as I am in Thailand |
| 06:16.43 | coppice | can you paste the log of the entire call? |
| 06:23.47 | *** join/#asterisk friedrich| (n=friedric@e177247155.adsl.alicedsl.de) |
| 06:23.49 | HaMYaI | coppice: http://en.pastebin.ca/620089 |
| 06:24.21 | HaMYaI | coppice: I didn't actually make a call just plugged it in |
| 06:31.19 | LoneShadow | iax2_register: Host 'my-voip-server.com' not found at line 202 |
| 06:31.48 | LoneShadow | what does asterisk need to resolve the host for iax register ? |
| 06:31.56 | LoneShadow | server* |
| 06:37.43 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 06:39.30 | HaMYaI | coppice: the channels concerned are between channel => 94-108 and channel => 110-124 |
| 06:44.08 | *** part/#asterisk rkeels (n=rkeels@c-67-171-15-176.hsd1.wa.comcast.net) |
| 07:04.15 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-17-59-222.pskn.east.verizon.net) |
| 07:10.39 | LoneShadow | anyone know if IAX register calls requires stun support ? |
| 07:10.53 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:14.04 | *** join/#asterisk sergey (n=sergey@gw4-171.iks.ru) |
| 07:19.51 | coppice | HaMYaL: if you just plugged in the E1, what you see it probably to be expected as the line settles down. protocol error recovery seems to work OK, and the circuits seem to end up idle |
| 07:42.40 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000e08cb2a29-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 07:42.48 | DrukenLPY | evening everyone |
| 07:43.59 | DrukenLPY | anyone know if i can use the h exten with cdr() to set a cost value on a call ? |
| 07:49.37 | snuff-work | mm.. just do like set(cdr(userfield)=cost) |
| 07:50.54 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 08:07.54 | sweeper | man this is so awsome. I've got a hobo rails app talking to asterisk, with the dialplan being run through adhearsion which in turn shares the ActiveRecord models of the rails app |
| 08:09.22 | *** join/#asterisk Jedi-Jiji (n=Jedi-Jij@82.249.68.143) |
| 08:28.00 | *** join/#asterisk friedrich| (n=friedric@e177251242.adsl.alicedsl.de) |
| 08:29.36 | *** join/#asterisk xtr (i=94752345@216.19.191.191.novuscom.net) |
| 08:32.30 | *** join/#asterisk xtr-II (i=01928375@216.19.191.191.novuscom.net) |
| 08:36.17 | *** join/#asterisk mitcheloc (n=mitchelo@adsl-68-120-226-30.dsl.irvnca.pacbell.net) |
| 08:39.32 | *** join/#asterisk xtr-II (i=94752345@216.19.191.191.novuscom.net) |
| 08:44.32 | *** join/#asterisk MrMister2 (n=mrmister@89-180-221-15.net.novis.pt) |
| 08:45.06 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 08:48.22 | *** join/#asterisk gardo (n=gardo@125.212.15.240) |
| 08:56.03 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:56.15 | *** join/#asterisk af_ (n=getsmart@81-174-44-88.dynamic.ngi.it) |
| 09:03.05 | *** join/#asterisk littleball (n=littleba@bb219-74-144-111.singnet.com.sg) |
| 09:03.52 | littleball | hello, i have x100p fxo card installed. But when i try to call this card, the operator always prompt me that the called number is engaged |
| 09:04.29 | littleball | but when i take out the cable from asterisk, then i can hear the dialing tone |
| 09:04.29 | littleball | who can help? |
| 09:11.49 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 09:12.01 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 09:15.24 | *** part/#asterisk mattwj2005 (n=matt@c-76-17-133-96.hsd1.mn.comcast.net) |
| 09:22.01 | *** join/#asterisk Strom_M (n=strom@ip70-170-60-8.lv.lv.cox.net) |
| 09:22.29 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:39.05 | *** join/#asterisk somethingbad (i=vladi@vip.arenabg.com) |
| 09:48.18 | *** join/#asterisk zapata (n=herbert@chello213047080026.4.14.vie.surfer.at) |
| 10:00.00 | *** join/#asterisk svenna_ (n=svenna@p548D06AB.dip0.t-ipconnect.de) |
| 10:05.15 | *** join/#asterisk littleball (n=littleba@bb220-255-93-20.singnet.com.sg) |
| 10:12.57 | littleball | hello, i have x100p fxo card installed, but when i started asterisk, the channel always be offhook. So, cannot call in and callout |
| 10:13.01 | littleball | who can help? |
| 10:17.09 | *** join/#asterisk anthonyc (i=ac@adsl-75-10-112-97.dsl.frs2ca.sbcglobal.net) |
| 10:25.37 | *** join/#asterisk xaphir (n=lx@nc-65-40-81-157.sta.embarqhsd.net) |
| 10:37.34 | *** part/#asterisk sergey (n=sergey@gw4-171.iks.ru) |
| 11:03.07 | *** join/#asterisk X-Rob (n=rob-x@58.163.162.7) |
| 11:09.20 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 11:21.32 | *** join/#asterisk gmorg (n=garymorg@adsl-75-0-42-196.dsl.covlil.sbcglobal.net) |
| 11:22.07 | *** join/#asterisk griels (n=Ellis@60.67-246-213.ippool.namesco.net) |
| 11:23.18 | griels | hi, anyone here know where to get cubix for personal use? |
| 11:27.24 | gmorg | anyone have experience with aastra phones? |
| 11:27.46 | gmorg | specifically a 57i CT w/2 560M expansion modules |
| 11:38.34 | *** join/#asterisk Op3r (n=op3r@121.97.214.210) |
| 11:41.43 | *** join/#asterisk haplodag (n=mwe@DSL217-132-36-200.bb.netvision.net.il) |
| 11:42.59 | *** join/#asterisk saftsack (n=saftsack@83-131-192-4.adsl.net.t-com.hr) |
| 11:43.11 | haplodag | can some one help me please? |
| 11:43.45 | _Raptor_ | haplodag: no if you dont ask a question |
| 11:44.11 | haplodag | i will ask.. just wanted to see if some one is online |
| 11:44.17 | haplodag | well my problem is.... |
| 11:44.33 | haplodag | i am trying to build a new gui for asterisk now |
| 11:45.07 | haplodag | and i want that users will be abale to login to the gui |
| 11:45.30 | haplodag | for now what i know is happening is that it takes the users and pass from the manager.conf |
| 11:45.41 | haplodag | i want it to take it from useres.conf |
| 11:45.51 | haplodag | and i dont know how to do it.. |
| 11:45.56 | haplodag | (users.conf) |
| 11:46.04 | haplodag | any suggestion? |
| 11:46.36 | saftsack | isnt there a devel channel? |
| 11:47.07 | haplodag | what?? what is devel channel? |
| 11:48.24 | saftsack | an irc devel channel |
| 11:49.18 | haplodag | i guess there is. |
| 11:50.10 | *** join/#asterisk [hC] (n=hardcore@rmtw.com) |
| 11:55.05 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:56.20 | *** join/#asterisk equex (n=skldhxr@ti0030a380-0298.bb.online.no) |
| 12:00.32 | *** part/#asterisk equex (n=skldhxr@ti0030a380-0298.bb.online.no) |
| 12:16.19 | sweeper | haha |
| 12:16.39 | sweeper | for a second there I thought you meant a channel for the dev of irc |
| 12:28.25 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000e08cb2a29-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 12:37.07 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:42.05 | saftsack | haplodag, did you find a channel? |
| 13:00.43 | haplodag | nop |
| 13:00.49 | haplodag | there is a develop one |
| 13:00.56 | haplodag | but thet didnt answer me there |
| 13:02.05 | haplodag | that = they |
| 13:02.47 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 13:04.16 | DrukenLPY | anyone know if i do additional processing using the ,h, does it tack onto the billable duration ? |
| 13:12.33 | *** join/#asterisk Op3r (n=op3r@121.97.214.210) |
| 13:14.07 | shido6 | hrmm |
| 13:21.45 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 13:23.46 | *** join/#asterisk shinao1 (n=shinao1@dial-pool26.lagos.starcomms.net) |
| 13:29.58 | *** join/#asterisk mtaht4 (n=m@74.9.53.154) |
| 13:32.00 | *** join/#asterisk kovger (n=kovger@217.27.219.154) |
| 13:33.39 | *** join/#asterisk tsurko (n=tsurko@77.70.15.52) |
| 13:34.22 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:34.23 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 13:34.33 | blitzrage | bcnl: ping |
| 13:36.28 | *** join/#asterisk _DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:36.54 | blitzrage | is this thing on? |
| 13:39.03 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 13:44.36 | *** join/#asterisk tsurko (n=tsurko@77.70.15.52) |
| 13:49.29 | *** join/#asterisk tsurko (n=tsurko@77.70.15.52) |
| 14:03.47 | *** join/#asterisk saftsack (n=saftsack@83-131-192-4.adsl.net.t-com.hr) |
| 14:06.34 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-243-142-135.buff.east.verizon.net) |
| 14:14.52 | *** join/#asterisk IPmonger (n=ipmonger@c-71-224-31-246.hsd1.pa.comcast.net) |
| 14:15.14 | *** part/#asterisk IPmonger (n=ipmonger@c-71-224-31-246.hsd1.pa.comcast.net) |
| 14:15.51 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 14:18.10 | *** join/#asterisk Avalone (n=Avalone_@217.118.82.37) |
| 14:18.44 | *** join/#asterisk wunderkin (i=wunderki@ip68-2-61-64.ph.ph.cox.net) |
| 14:23.00 | SuPrSluG | hello all, I have a user who notes that calls from <UNKNOWN> get cut off when leaving voicemail after a few seconds. Anyone heard of this? |
| 14:24.48 | wunderkin | because its a telemarketer? |
| 14:26.29 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 14:28.50 | *** join/#asterisk johnthewolf (n=jd@216.252.79.163) |
| 14:30.10 | SuPrSluG | no it |
| 14:30.15 | SuPrSluG | is not |
| 14:30.52 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 14:30.57 | SuPrSluG | does asterisk do that? has voicemail changed |
| 14:31.01 | Corydon76-home | sounds like a phantom ring |
| 14:31.12 | Corydon76-home | Are you on analog lines? |
| 14:31.31 | Corydon76-home | Does it happen consistently every day at a particular time? |
| 14:31.38 | johnthewolf | is it possible to have a multilingual voicemail with asterisk? |
| 14:31.49 | Corydon76-home | johnthewolf: yes |
| 14:31.56 | johnthewolf | complicated? |
| 14:32.02 | Corydon76-home | johnthewolf: although it's a little weird to set up |
| 14:32.17 | Corydon76-home | You mean the greetings or the message left? |
| 14:32.34 | Corydon76-home | For the message left, there is no way. For the greetings, it's possible |
| 14:32.41 | SuPrSluG | no, not phantom. the call goes normal, but after about 5-10 secs leaving voicemail the call gets dropped. |
| 14:33.14 | johnthewolf | do you have to go through an ivr or something? |
| 14:33.24 | SuPrSluG | yes |
| 14:33.45 | johnthewolf | so you need to parse the language in the extensions.conf? |
| 14:34.11 | Corydon76-home | johnthewolf: it basically consists of making sure the directories exist, then setting the LANGUAGE(), then going into voicemailmain |
| 14:34.31 | SuPrSluG | so it wouldn't be a setting in voicemail.conf? |
| 14:34.59 | johnthewolf | Cause I know you pick the language in sip.conf |
| 14:35.05 | Corydon76-home | SuPrSluG: maxmessage, perhaps |
| 14:35.21 | Corydon76-home | SuPrSluG: what's maxmessage set to now? |
| 14:35.58 | SuPrSluG | ; Minimum length of a voicemail message in seconds |
| 14:36.00 | SuPrSluG | minmessage=5 |
| 14:36.01 | SuPrSluG | maxsilence=4 |
| 14:36.10 | johnthewolf | Corydon76-home: would there be a tutorial somewhere by any chance that you are aware of? |
| 14:36.28 | Corydon76-home | johnthewolf: nope, just recalling to the best of my knowledge from the code |
| 14:36.42 | Corydon76-home | johnthewolf: like I said, it's a little odd to set up |
| 14:37.11 | johnthewolf | Corydon76-home: I'll see if I can figure this out! Thanks for the poiters |
| 14:37.23 | Corydon76-home | We don't support it by design, but it could be done based on the fact that language is supported by the audio files |
| 14:38.39 | Corydon76-home | or rather, we never wrote that support specifically into voicemail, but since the file selection routine supports language, it could be done |
| 14:42.18 | *** join/#asterisk littleball (n=littleba@bb220-255-155-161.singnet.com.sg) |
| 14:49.33 | *** join/#asterisk ez` (n=ez@c66.110.149-45.clta.globetrotter.net) |
| 14:49.35 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:59.14 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 15:04.04 | *** join/#asterisk Ebola (n=Ebola@host86-143-168-67.range86-143.btcentralplus.com) |
| 15:06.30 | littleball | hi |
| 15:11.02 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 15:13.09 | littleball | hi, did you solve fxo busy tone issue? i have exactly the same proble as you. |
| 15:13.09 | littleball | I am doing test at home. |
| 15:13.09 | littleball | I use my mobile phone call my fixed line number. |
| 15:13.09 | littleball | (1)If fixed line not connect to anything phone or fxo, i can hear the ring ton. |
| 15:13.09 | littleball | (2)If i connect the fixed line to x100p, even x100p is not plug into PCI slot, i hear busy tone and the operator told me the line is engaged |
| 15:13.11 | littleball | Who can help? |
| 15:22.18 | *** part/#asterisk littleball (n=littleba@bb220-255-155-161.singnet.com.sg) |
| 15:24.46 | *** part/#asterisk mtaht4 (n=m@74.9.53.154) |
| 15:28.35 | sweeper | lilalinux: sounds like a wiring problem |
| 15:28.45 | sweeper | is the plug two wires only, or three? |
| 15:29.05 | sweeper | I read that people in the UK have 3 wires in the plugs, and they get that problem |
| 15:29.10 | sweeper | err |
| 15:29.17 | *** join/#asterisk kippi (n=none@untrust-gct.equinoxit.net) |
| 15:29.18 | sweeper | s/lilalinux/littleball/ |
| 15:29.20 | kippi | hey |
| 15:29.34 | kippi | is there away to get asterisk to send a text message? |
| 15:29.40 | sweeper | kippi: lots |
| 15:29.47 | sweeper | easiest is via email |
| 15:30.06 | sweeper | 1234567890@smsgateway.mycarrier.com |
| 15:31.41 | kippi | see BT allow you to text from a house phone, so I was wondering if you can just get askerisk to send a text out, I just need it to text out very 15mins to say the system is up, can I text using call files? |
| 15:32.06 | sweeper | yep |
| 15:32.10 | sweeper | uh |
| 15:32.18 | sweeper | actually, maybe not |
| 15:32.39 | sweeper | honestly, wouldn't it be better to just use a cron job? >.> |
| 15:32.48 | kippi | whats the command? dial would ring, is there a text command or somthing along these lines? |
| 15:33.39 | sweeper | sendmail :P |
| 15:34.25 | kippi | how would that work? |
| 15:35.10 | sweeper | write a little bash script that pipes the output from whatever status commands you want into an email, and send the email to your mobile |
| 15:35.15 | *** join/#asterisk _DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:35.39 | kippi | really want it as a text message |
| 15:35.45 | sweeper | it will be :v |
| 15:36.00 | sweeper | the email address is just an sms gateway |
| 15:36.07 | sweeper | it'll still show up as a text message |
| 15:36.48 | kippi | there is a SMS command from what I have just seen online |
| 15:37.31 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 15:37.38 | florz | kippi: That's only for transmitting a message to an analog landline phone capable of receiving SMSes |
| 15:38.21 | kippi | ah, so you can't send it over zap? |
| 15:38.37 | florz | kippi: sure, why not? |
| 15:38.58 | kippi | so SMS command might work? |
| 15:39.16 | florz | kippi: Depends on what you want to do? |
| 15:39.39 | kippi | just want to get asterisk to text me saying phone system is up |
| 15:40.01 | florz | kippi: if you want it to "text", buy a printer |
| 15:40.13 | Strom_M | or just use e-mail |
| 15:40.20 | Strom_M | or install nagios |
| 15:40.45 | kippi | ah ok |
| 15:42.47 | rob0 | Strom_M: BTW you were right, RMA time. :( Thx for the help yesterday. |
| 15:44.36 | Strom_M | rob0: you're welcome |
| 15:45.58 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 15:50.34 | *** join/#asterisk littleball (n=littleba@bb220-255-155-161.singnet.com.sg) |
| 15:50.41 | littleball | sss |
| 15:51.00 | *** join/#asterisk toot (n=toot@84.19.255.123) |
| 15:52.59 | Strom_M | littleball: regarding your earlier problem, i'd blame the card |
| 15:53.04 | Strom_M | x100p == crap |
| 15:53.11 | littleball | why? |
| 15:53.21 | Strom_M | ~cheap |
| 15:53.22 | jbot | it has been said that cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 15:53.25 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-82-81-153-115.red.bezeqint.net) |
| 15:53.39 | mosty | littleball, because the original digium cards were crap to begin with, and the clones you will find now are even worse |
| 15:53.47 | *** join/#asterisk guillote_GNU (n=guillote@190.7.27.17) |
| 15:53.58 | littleball | i need analog fxo |
| 15:54.06 | littleball | which one is better? |
| 15:54.09 | Strom_M | then get a tdm400 |
| 15:54.20 | littleball | tdm400? does it work? |
| 15:54.27 | littleball | from digium? |
| 15:54.39 | *** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:54.48 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 15:55.03 | Strom_M | littleball: yes |
| 15:55.04 | sweeper | funny how often "don't go cheap" translates to "buy digium" >.> |
| 15:55.50 | littleball | Strom_M, Digium FXO Module (X100M) ? |
| 15:55.53 | littleball | how aobut Digium FXO Module (X100M) ? |
| 15:55.55 | Strom_M | uh, no |
| 15:55.59 | Strom_M | thats just a module |
| 15:56.06 | littleball | i only need one channel for my customer |
| 15:56.08 | Strom_M | you want a TDM01B |
| 15:56.09 | mosty | littleball, sangoma cards are better |
| 15:56.42 | littleball | mosty, i only need to buy one channel or two channel fxo ? |
| 15:56.43 | sweeper | sipura/linksys DO make fxo ATA's |
| 15:56.59 | littleball | what device i should buy? |
| 15:57.17 | Qwell | mosty: "better"? |
| 15:57.26 | littleball | i need to run on asterisk |
| 15:57.30 | mosty | littleball, the sangoma cards have fxo/fxs modules that have two channels per module |
| 15:57.33 | file | <PROTECTED> |
| 15:57.42 | mosty | qwell: more reliable, easier to debug. you know- better. |
| 15:57.42 | Qwell | file: You /clear too much :p |
| 15:57.51 | file | Qwell: only once a day! |
| 15:58.34 | Qwell | mosty: How is hardware with closed-source drivers easier to debug? |
| 15:58.35 | mosty | littleball, maybe you can get a linksys spa 3102 |
| 15:58.52 | file | I like my SPA3102 |
| 15:59.08 | mosty | Qwell, wanpipe provides debugging/diagnostic utils which dont exist (as far as i know) for the equivalent digium hardware |
| 15:59.10 | littleball | mosty, does it work on asterisk? |
| 15:59.32 | sweeper | littleball: you could always, you know, go look at the product page |
| 15:59.33 | littleball | how does spa 3102 work? |
| 15:59.40 | mosty | littleball, sure, it's a SIP device. you might not even need asterisk if you get one |
| 16:00.44 | littleball | mosty, do you mean that analog phone line -->spa 3102 -->internet -->my asterisk server? |
| 16:01.14 | littleball | i need to get caller info and pass to my center asterisk server |
| 16:01.59 | mosty | littleball, sounds like a bad way to do things. get a DID with your VOIP service provider |
| 16:02.39 | littleball | mosty, DID providers can only provide open market places |
| 16:02.40 | littleball | <PROTECTED> |
| 16:03.09 | SuPrSluG | how are the zapmirco cards |
| 16:03.16 | mosty | i have no idea what you're trying to say littleball |
| 16:03.19 | Qwell | ~cheap |
| 16:03.19 | jbot | well, cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 16:03.22 | littleball | in some countries, you cannot get DID from DID resellers |
| 16:03.51 | mosty | littleball, give us a specific problem that you're trying to solve |
| 16:04.01 | littleball | most, i mean in those place, you cannot get DID numbers from DID sellers. you had to get analog line and pass call info yourself |
| 16:04.25 | mosty | littleball, which country? |
| 16:04.39 | littleball | ok. In VN, Thailand etc... |
| 16:04.45 | littleball | South Aisa |
| 16:05.05 | Strom_M | Qwell: http://rudholm.com/gallery/phonetrip |
| 16:05.27 | Qwell | colorful |
| 16:05.48 | mosty | littleball, and why do you need to hookup this analogue line to asterisk? what goal are you trying to serve? |
| 16:06.28 | littleball | mosty, callback system |
| 16:06.35 | littleball | i sell callback system to customers |
| 16:06.49 | littleball | they need to pickup caller ID to make callback, right? |
| 16:07.26 | mosty | littleball, it sounds like you would need an asterisk server directly connected to each analogue line, in that case |
| 16:07.40 | littleball | so, my customers will install analog FXO on asterisk machine and then apply one analog line in those countries |
| 16:08.23 | littleball | these asterisks only collection caller ID info etc and pass to center asterisk server, which will handle billing and call back functions |
| 16:08.30 | littleball | yes |
| 16:08.31 | mosty | something like that, yes |
| 16:08.52 | littleball | so, i am looking for analog FXO devices |
| 16:09.02 | littleball | any good one? |
| 16:10.05 | mosty | the two main options are the digium tdm400 series and the sangoma a200 series |
| 16:10.08 | xaphir | Would you guys check this and see if I have this right. The FXS/DPO daugherboards are for supplying dialtone to analogue telephones, and the FXS modules are for interfacing with the telco |
| 16:10.54 | littleball | mosty, thanks. let me check. |
| 16:12.56 | littleball | the problem is that how knows those card works ? After troubleshooting on X100P card, i feel it is not easy to handle analog lines |
| 16:13.24 | Qwell | littleball: the x110p is junk |
| 16:13.29 | Qwell | erm, x100p |
| 16:14.54 | littleball | Qwell, how about a200 from sangama? |
| 16:15.27 | Qwell | I don't know, never used their stuff. I don't buy from vendors who are anti-asterisk |
| 16:15.43 | mosty | littleball, x100p's are horrible. the digium tdm400/sangoma a200 are better, but still a pain to use |
| 16:16.10 | littleball | actually i make x100p work at home sometime ago |
| 16:16.24 | littleball | but suddendly, today, it just busy. always busy |
| 16:16.43 | xaphir | Is anybody here using the newer tdm2400 series? |
| 16:17.00 | Qwell | xaphir: they work great |
| 16:17.15 | Strom_M | xaphir: I have a TDM800, which is the mini version of the 2400 |
| 16:17.15 | Qwell | disclaimer: I work for Digium |
| 16:17.17 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-148-108.satx.res.rr.com) |
| 16:17.18 | Strom_M | and I love it |
| 16:17.24 | Strom_M | disclaimer: I contract for digium |
| 16:17.27 | xaphir | qwell: how about the echo cancellation option that comes on some of them? |
| 16:17.27 | Qwell | heh |
| 16:17.35 | Qwell | xaphir: haven't used it.. |
| 16:17.46 | littleball | i have one question: x100p analog card (not install on computer yet), if i plug phone line into the card and call the phone line number, what is the result? |
| 16:17.50 | [TK]D-Fender | xaphir, A200d > All |
| 16:18.27 | littleball | the card is not plugined to computer yet. |
| 16:18.27 | [TK]D-Fender | Disclaimer : I contract for anybody who wants a working * system ;) |
| 16:19.14 | mosty | i would rather work with an a100d than an a200d ;) |
| 16:19.26 | xaphir | These 2400's accept daughterboard modules, I was wondering if I need to get a FXS/DPO for the analogue telephones |
| 16:19.29 | [TK]D-Fender | mosty, No such card... |
| 16:19.48 | Qwell | DPO? |
| 16:19.52 | mosty | 104, sorry |
| 16:19.53 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 16:20.09 | [TK]D-Fender | xaphir, I recommend ATA's / gateways instead of PCI based FXS for everything execpt analog fax machines. |
| 16:20.10 | xaphir | Qwell: the driver shows the boards like that |
| 16:20.24 | [TK]D-Fender | mosty, Better :) |
| 16:20.25 | Qwell | I don't know what a DPO is |
| 16:20.26 | mosty | xaphir, FXO ports connect to phone lines, FXS ports connect to phones |
| 16:20.27 | littleball | [TK]D-Fender, if i just plug the phone line into fxo analog card, which itself is not plugin to computer yet, then if i call the fixed line number, will it be engaged? |
| 16:20.49 | mosty | littleball, it will be engaged if the line is in use. like with a telephone |
| 16:20.55 | xaphir | mosty: thanks |
| 16:21.17 | [TK]D-Fender | littleball, that is completely insane. What is an unplugged, unpowered, and unconfigured PCI CARD GOING TO DO?! |
| 16:21.21 | littleball | most, of course, no one use the line. because the card is not plugin to computer yet |
| 16:21.50 | littleball | i ask this question, becasue i found it always engaged |
| 16:22.08 | xaphir | d-fender: thanks for the advice. Are you saying that the new version of asterisk handles faxes ok? |
| 16:22.19 | littleball | and if i take out the line from this unpowerd card, and call again, i can hear the ring and not engaged |
| 16:22.28 | mosty | littleball, engaged means that the line is in use |
| 16:22.37 | xaphir | d-fender: cause I heard it was a nightmare on asterisk 1.2 |
| 16:22.49 | littleball | mosty, i feel strange, because this line is not used :-) |
| 16:23.09 | littleball | the only reason is the card has problem |
| 16:23.19 | mosty | littleball, why are you connecting an unpowered card to the line anway? there is no way that will do anything useful |
| 16:23.28 | [TK]D-Fender | xaphir, If you value your faxes you'll leave them on a seperate line with a real fax machine on it and keep * as far away as possible |
| 16:23.36 | littleball | this is one phase of my troubleshooting |
| 16:23.39 | xaphir | d-fender: lol |
| 16:23.48 | littleball | because i always get engaged |
| 16:23.50 | xaphir | d-fender: that's exactly what I heard |
| 16:24.00 | littleball | so i just take out the card and try again , the result is the same |
| 16:24.10 | littleball | even i plugin the card into PCI slot |
| 16:24.20 | littleball | alwasys get engaged |
| 16:24.26 | Qwell | littleball: try sending a call to it while it isn't plugged in |
| 16:24.29 | xaphir | d-fender: have you done any asterisk 1.4 installations? |
| 16:24.29 | Qwell | That might help |
| 16:24.35 | Qwell | while the phone line isn't plugged in |
| 16:24.41 | Qwell | see if the card reacts |
| 16:24.44 | [TK]D-Fender | xaphir, very few, but yes |
| 16:24.58 | littleball | Qwell, yes, thsi is what i did. it is also engaged |
| 16:25.03 | [TK]D-Fender | xaphir, Honestly they seem to work jsut fine.... |
| 16:25.07 | xaphir | d-fender: did you bother with the fax stuff or was it not worth it? |
| 16:25.09 | Qwell | it'll send the signal over wireless analog to the card |
| 16:25.13 | littleball | this card just engaed all calls on the connected line |
| 16:25.51 | [TK]D-Fender | xaphir, I do have a working SpanDSP setup at the office which is good for DID faxing, but my primary departmental faxes are 100% seperate analog |
| 16:25.58 | littleball | [TK]D-Fender, what do you mean " ATA's / gateways instead of PCI based FXS for everything execpt analog fax machines." ? |
| 16:26.03 | mosty | littleball, take the x100p and throw it in the bin, you will save yourself a lot of annoyance |
| 16:26.23 | littleball | i will... you see i am looking for other solution now :-) |
| 16:26.46 | littleball | mosty, i am trying to find out whether TK's ATA/gatewy works for me |
| 16:26.48 | littleball | or not |
| 16:26.52 | xaphir | d-fender: has anyone done any experimentation with fax in the newer versions of asterisk 1.4? |
| 16:27.23 | mosty | littleball, no, not for the situation you have described |
| 16:27.40 | littleball | i also think so |
| 16:27.49 | littleball | maybe i must try a200 first |
| 16:27.52 | littleball | how aobut the price? |
| 16:28.39 | mosty | littleball, i would say that there is no good solution to the problem you have. in the civilised world you would just get a DID |
| 16:29.30 | littleball | mosty, i know definitely. this is part of my business ;-). other people pay for this |
| 16:29.54 | littleball | i even have gsm modem solution for them |
| 16:29.57 | [TK]D-Fender | xaphir, I've heard of some 1.4 ports for rxfax/txfax but haven't gotten to test them personally |
| 16:31.07 | xaphir | d-fender: have you done any installations where you are pushing analog phones? |
| 16:33.46 | xaphir | d-fender: or have all your installs been ip-style phones |
| 16:35.07 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-17-59-222.pskn.east.verizon.net) |
| 16:40.47 | *** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 16:47.03 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 16:48.06 | *** join/#asterisk moren88 (n=moreno@host186-176-dynamic.17-87-r.retail.telecomitalia.it) |
| 16:48.14 | moren88 | Hi! |
| 16:48.30 | moren88 | do you know asterisk-mini? |
| 16:49.50 | SuPrSluG | open-wrt? |
| 16:51.40 | moren88 | yes! |
| 16:52.36 | moren88 | there are the sources????? |
| 16:53.51 | SuPrSluG | yes i install on a router. |
| 16:54.18 | SuPrSluG | ipkg install asterisk-mini |
| 16:59.16 | *** join/#asterisk TheresaJayne (n=Theresa@5ac51c31.bb.sky.com) |
| 17:00.45 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-17-59-222.pskn.east.verizon.net) |
| 17:02.01 | TheresaJayne | i am looking for clarification on how something is working to see whether i can solve my current problem |
| 17:02.28 | TheresaJayne | i have an inbound connection from a sip provider |
| 17:02.54 | TheresaJayne | i want also an outbound connection to a different number at the same sip provider |
| 17:03.11 | TheresaJayne | the inbound goes to a specific extension |
| 17:03.23 | TheresaJayne | with just the register it works |
| 17:03.54 | [TK]D-Fender | TheresaJayne, use seperate SIP peers. |
| 17:03.58 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkit.com.au) |
| 17:03.58 | TheresaJayne | when i add the outbound block (peer) it gives an auth fail and then ignores all calls |
| 17:04.16 | [TK]D-Fender | TheresaJayne, if you have an auth fail then you did it wrong. fix the user/pass |
| 17:04.21 | TheresaJayne | i am using 2 seperate accounts |
| 17:04.57 | TheresaJayne | but the ack is going back different to the incoming |
| 17:05.08 | TheresaJayne | say incoming is sip.1.com |
| 17:05.16 | TheresaJayne | the local machine is sip.2.com |
| 17:05.29 | TheresaJayne | invite is for sip1.com |
| 17:05.42 | TheresaJayne | then the ack is for sip2.com |
| 17:06.12 | TheresaJayne | ~pb |
| 17:06.13 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org |
| 17:07.36 | moren88 | SuPrSluG, but the sources? |
| 17:07.45 | moren88 | i want to compile my copy of asterisk-mini |
| 17:07.58 | TheresaJayne | http://pastebin.ca/620627 |
| 17:08.01 | TheresaJayne | the log file |
| 17:08.01 | moren88 | because in the asterisk-mini package there isn't the application Dial!! |
| 17:08.39 | TheresaJayne | the asterisk server is on the 135 number and the sip provider is on the 85 |
| 17:08.40 | moren88 | SuPrSluG, ! |
| 17:08.57 | SuPrSluG | yes just a min |
| 17:09.03 | moren88 | :) |
| 17:12.11 | SuPrSluG | moren88: generally the openwrt packages are taken from stock source code and trimmed to the bare essentials and repackaged to fit on the router |
| 17:12.31 | wunderkin | moren88, you can use the non-mini package... |
| 17:12.43 | wunderkin | it is a real pita to compile on your own |
| 17:12.49 | moren88 | wunderkin, yes but i not have enoug space on the router! |
| 17:13.00 | TheresaJayne | the working call with no outbound block is as follows |
| 17:13.02 | TheresaJayne | http://pastebin.ca/620636 |
| 17:13.09 | wunderkin | moren88, search for zandbelt |
| 17:13.14 | wunderkin | on google |
| 17:13.50 | moren88 | i dont found anything! |
| 17:14.21 | wunderkin | ... |
| 17:14.56 | SuPrSluG | moren88: http://sarwiki.informatik.hu-berlin.de/Programming_the_Linksys_WRT54GS_Wireless_Broadband_Router |
| 17:14.57 | [TK]D-Fender | Looking for 7911259766 in INVALID (domain 135.196.99.98) |
| 17:15.10 | [TK]D-Fender | # |
| 17:15.10 | [TK]D-Fender | Found no matching peer or user for '85.90.225.100:5060' |
| 17:15.25 | *** join/#asterisk _DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:15.38 | [TK]D-Fender | TheresaJayne, And why are you only showing me CLI output when your CONFIGS are screwed? |
| 17:15.41 | TheresaJayne | can someone just tell me if i understand this right, a call comes in via an account using the register definition and forwards to the default context unless there is another context handling the connection |
| 17:15.47 | wunderkin | http://zandbelt.dyndns.org/asterisk.html link #4 |
| 17:15.58 | TheresaJayne | its the same config you did for me on Friday |
| 17:16.09 | [TK]D-Fender | TheresaJayne, APSTEBIN. |
| 17:16.11 | TheresaJayne | you set the default context to invalid |
| 17:16.39 | [TK]D-Fender | TheresaJayne, and that was ManxPower's handiwork, not MINE. |
| 17:17.38 | TheresaJayne | http://pastebin.ca/620641 |
| 17:17.43 | TheresaJayne | thats the sip.conf |
| 17:18.15 | [TK]D-Fender | TheresaJayne, you only have 1 peer account, so dont' expect to be able to dial against the OTHER one. |
| 17:18.47 | TheresaJayne | http://pastebin.ca/620643 |
| 17:18.55 | [TK]D-Fender | TheresaJayne, And for inbound.... you don't even HAVE a 'type=user' entry to receive against so everything falls under the [general] |
| 17:19.08 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 17:19.13 | TheresaJayne | thats what i am trying to understand |
| 17:19.26 | [TK]D-Fender | TheresaJayne, And from what you show me now you don't even have EXTENS to match ANY of them. |
| 17:19.32 | TheresaJayne | so i need an inbound block |
| 17:19.48 | TheresaJayne | as well as the register line |
| 17:19.54 | [TK]D-Fender | Looking for 7911259766 in INVALID (domain 135.196.99.98) |
| 17:19.58 | TheresaJayne | and the outbound peer block as well |
| 17:19.59 | [TK]D-Fender | See this? |
| 17:20.10 | [TK]D-Fender | they are dialing a NUMBER and you have no exten to match it! |
| 17:20.22 | [TK]D-Fender | they are telling you to your face what you need. |
| 17:20.28 | [TK]D-Fender | Go caffeinate |
| 17:20.38 | TheresaJayne | or rather no INVALID context to take the call |
| 17:20.53 | TheresaJayne | i thinnk i understand |
| 17:21.06 | [TK]D-Fender | TheresaJayne, where in here http://pastebin.ca/620643 do yuo have an exten for it? |
| 17:21.29 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 17:21.34 | TheresaJayne | as i said this is what i was given Friday and it frankly doesnt make sense to me |
| 17:21.42 | [TK]D-Fender | TheresaJayne, and in here : http://pastebin.ca/620641 All you did was set a contect in GENERAL for invalid. |
| 17:22.05 | TheresaJayne | i think its starting to make sense |
| 17:22.07 | [TK]D-Fender | TheresaJayne, makes no sense because you didn't make USER accounts for them to get answered against like told |
| 17:22.22 | TheresaJayne | is that in sip.conf or user.conf? |
| 17:23.12 | rob0 | type=user in sip.conf |
| 17:23.13 | [TK]D-Fender | TheresaJayne, sip.conf. |
| 17:23.21 | [TK]D-Fender | users.conf = flaming pile 'o' shit |
| 17:23.26 | rob0 | haha |
| 17:23.32 | TheresaJayne | ok |
| 17:24.01 | [TK]D-Fender | it was a gross mis-step IMO. |
| 17:24.16 | TheresaJayne | please dont get exasperated with me I am trying to learn a fairly complex system and came to where I know the experts to be |
| 17:24.49 | TheresaJayne | ok so for each inbound I need a user block that also sets a context for it to go to |
| 17:25.10 | TheresaJayne | then in that context i need the extens |
| 17:25.40 | TheresaJayne | and if I am dialling out i do as i already have but it should work if the inbound is working ok |
| 17:25.54 | TheresaJayne | is that the rough idea |
| 17:28.09 | [TK]D-Fender | TheresaJayne, you need 1 peer per ACCOUNT you are dialing out again, jsut like for INCOMING. |
| 17:30.45 | TheresaJayne | ok |
| 17:30.57 | TheresaJayne | does the order of the blocks in sip.conf matter? |
| 17:32.09 | [TK]D-Fender | TheresaJayne, [general], then your REGISTER's, then anything goes |
| 17:32.46 | *** part/#asterisk TheresaJayne (n=Theresa@5ac51c31.bb.sky.com) |
| 17:33.12 | rob0 | I need a [coffee] section in my sip.conf ... |
| 17:33.40 | rob0 | ... or cool it down enough and put it in gulp.conf |
| 17:34.25 | *** join/#asterisk TheresaJayne (n=Theresa@5ac51c31.bb.sky.com) |
| 17:35.08 | TheresaJayne | BTW where was the line saying the INVALID line |
| 17:35.28 | [TK]D-Fender | TheresaJayne, ? |
| 17:35.46 | TheresaJayne | in the log |
| 17:36.04 | TheresaJayne | i see the found peer tovo-out |
| 17:36.21 | [TK]D-Fender | TheresaJayne, check your EYES |
| 17:36.30 | [TK]D-Fender | TheresaJayne, you provided the pastebin |
| 17:36.59 | [TK]D-Fender | TheresaJayne, not a web-browser I know of that doesn't have a TESXT SEARCH. |
| 17:37.12 | TheresaJayne | gives me a javascript error |
| 17:37.41 | TheresaJayne | error 239 object doesnt support this property or method |
| 17:38.54 | [TK]D-Fender | http://pastebin.ca/620636 |
| 17:40.20 | TheresaJayne | i see the line on line 42 |
| 17:41.14 | [TK]D-Fender | TheresaJayne, 1st problemt hat points out that you have no user created for the account. CONTINUE |
| 17:41.26 | TheresaJayne | the problem is that on the incoming call its not finding the user but the peer and trying to redirect out |
| 17:41.41 | [TK]D-Fender | TheresaJayne, No, it is NOT trying to "redirect out". |
| 17:41.49 | [TK]D-Fender | TheresaJayne, sip can't redirect ANYTHING |
| 17:41.56 | [TK]D-Fender | sip.conf that is* |
| 17:42.06 | [TK]D-Fender | TheresaJayne, your DIALPLAN says what to DO with calls. |
| 17:43.14 | TheresaJayne | http://pastebin.ca/620676 |
| 17:43.20 | TheresaJayne | this is the sip.conf now |
| 17:43.52 | TheresaJayne | so why does it say it finds tovo-out and not sip |
| 17:45.15 | TheresaJayne | it finds the peer then tries to hand it to the outbound rather than the context [sip] |
| 17:47.03 | TheresaJayne | sorry if it sounds as if i am thick but i am trying everything I am asked and am getting frustrated with the lack of progress |
| 17:47.54 | TheresaJayne | the way i read the log - it receives an incoming invite request |
| 17:48.25 | TheresaJayne | then finds peer tovo-out and then tries to hand the call to it and not the internal extension requested |
| 17:48.37 | [TK]D-Fender | TheresaJayne, that [sip] wasn't in the first sip.conf pyou pastebin'd. |
| 17:48.47 | TheresaJayne | i know |
| 17:48.54 | TheresaJayne | i changed it after talking to you |
| 17:48.59 | TheresaJayne | but it still doesnt work |
| 17:49.18 | [TK]D-Fender | TheresaJayne, And di I see a NEW pastebin for the NEW failure? NO. |
| 17:49.42 | [TK]D-Fender | TheresaJayne, make a new SINGLE pastbin with everything |
| 17:50.52 | *** join/#asterisk ManxPower (n=manxpowe@015-840-323.area5.spcsdns.net) |
| 17:52.18 | TheresaJayne | http://pastebin.ca/620686 |
| 17:54.56 | LoneShadow | wunderkin: are you there ? |
| 17:55.29 | wunderkin | just ask in the channel, and no |
| 17:56.25 | LoneShadow | had a question about QOS settings for IAX on openwrt, will check there |
| 17:58.53 | [TK]D-Fender | TheresaJayne, name your [sip] entry [7911259766] and make one for your other register as well. Ditch the "username=" line |
| 17:59.18 | TheresaJayne | i am not using the other register |
| 17:59.22 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 17:59.35 | TheresaJayne | or do i need that to call out as well |
| 18:01.00 | TheresaJayne | as the other one is for the tovo-out |
| 18:03.18 | [TK]D-Fender | TheresaJayne, you should ahve 1 each |
| 18:03.32 | [TK]D-Fender | TheresaJayne, both can call either way you know. |
| 18:03.42 | TheresaJayne | so i need a register line for the outbound as well |
| 18:03.56 | [TK]D-Fender | TheresaJayne, you HAVE 2 register lines |
| 18:04.03 | TheresaJayne | one is commented out |
| 18:04.07 | [TK]D-Fender | TheresaJayne, you need a peer to dial ot of EACH |
| 18:04.10 | [TK]D-Fender | kjshdfhflasdjhlfsdasfd |
| 18:04.40 | TheresaJayne | fender - the setup - inbound on 259766 |
| 18:04.46 | TheresaJayne | outbound on 253545 |
| 18:04.55 | [TK]D-Fender | TheresaJayne, UGH!!!!!!!!!!!!! |
| 18:05.05 | [TK]D-Fender | TheresaJayne, these are 2 COMPLETE SERVICE connection! |
| 18:05.20 | [TK]D-Fender | TheresaJayne, EITHER can PLACE or RECIEVE calls! |
| 18:05.30 | [TK]D-Fender | TheresaJayne, EACh one needs a REGISTER, a PEER, and a USER. |
| 18:05.32 | TheresaJayne | in the end we need more than 30 lines |
| 18:05.40 | TheresaJayne | oh right |
| 18:05.43 | TheresaJayne | hmmm |
| 18:05.44 | [TK]D-Fender | TheresaJayne, Stop trying to do HALF THAE JOB EFOR EACH |
| 18:05.58 | TheresaJayne | oh! |
| 18:06.06 | TheresaJayne | why didnt someone say something earlier |
| 18:06.11 | [TK]D-Fender | TheresaJayne, I DID |
| 18:06.14 | [TK]D-Fender | 1000 times |
| 18:06.21 | TheresaJayne | sorry i didnt understand |
| 18:06.26 | *** join/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 18:06.49 | *** part/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 18:07.05 | TheresaJayne | can i half the config by making the blocks friends rather than peer and user? |
| 18:09.16 | [TK]D-Fender | TheresaJayne, maybe. try and see |
| 18:09.24 | [TK]D-Fender | TheresaJayne, many might work that way |
| 18:09.49 | [TK]D-Fender | TheresaJayne, but you shouldn't go witha company that makes you reg each account seperately. |
| 18:10.12 | [TK]D-Fender | TheresaJayne, Also you shouldn't run a company on pure VoIP (in most peoples opinions). 30 = a LOT. |
| 18:11.49 | TheresaJayne | its running on a 2 proc ibm eserver on a gigabit pipe |
| 18:11.59 | TheresaJayne | and is an IVR system |
| 18:12.20 | TheresaJayne | so its mainly playing messages getting input and rerouting as requested |
| 18:13.11 | [TK]D-Fender | TheresaJayne, rerouted to where? |
| 18:16.19 | TheresaJayne | http://pastebin.ca/620717 |
| 18:16.31 | TheresaJayne | this is the current configs do they look ok |
| 18:16.39 | TheresaJayne | sip.tovo.com |
| 18:17.18 | TheresaJayne | basically another 7911 number (csr , tech support etc) |
| 18:18.56 | [TK]D-Fender | register => 7911253545:xxxxxxx@sip.tovo.com:5060/1002 <- change the 1002 for the real number and change the dialplan accordingly |
| 18:19.27 | TheresaJayne | doh sorry missed that one |
| 18:19.31 | TheresaJayne | changing it |
| 18:20.24 | [TK]D-Fender | TheresaJayne, and you should NOT be doing # |
| 18:20.24 | [TK]D-Fender | exten => 7911259766,3,WaitExten |
| 18:20.26 | [TK]D-Fender | in there |
| 18:20.44 | [TK]D-Fender | they can dial from 1 "line" to another directly. Thats just hideous. |
| 18:20.53 | TheresaJayne | thats just for testing |
| 18:21.08 | TheresaJayne | just so it answers reads a message and waits |
| 18:21.10 | [TK]D-Fender | TheresaJayne, do a HANGUP then so things don't go in circles |
| 18:21.24 | TheresaJayne | but its not answering with that |
| 18:21.34 | TheresaJayne | dialling in on 07911259766 I get http://pastebin.ca/620720 |
| 18:22.15 | *** join/#asterisk MrMister2 (n=mrmister@89-180-221-15.net.novis.pt) |
| 18:22.26 | TheresaJayne | it finds the peer 253545 and then transmitting to sip.tovo.com |
| 18:23.07 | *** join/#asterisk MrMister2 (n=mrmister@89-180-221-15.net.novis.pt) |
| 18:23.38 | TheresaJayne | or is the server expecting a password from the incoming call? |
| 18:24.05 | [TK]D-Fender | TheresaJayne, I'm not sure... why am I seeing "sippy" & "SER" in there |
| 18:24.16 | TheresaJayne | thats the sip.tovo.com provider |
| 18:24.51 | [TK]D-Fender | and THIS : Call-ID: 3c268bad88b8-044dydkqlfej@snom360 |
| 18:24.55 | [TK]D-Fender | snom 360? |
| 18:24.59 | [TK]D-Fender | WTF? |
| 18:25.12 | TheresaJayne | thats the phone i am using to dial via the sip provider |
| 18:25.37 | TheresaJayne | i am dialling from my snom on 210290 to 259766 on sip.tovo.com |
| 18:25.47 | [TK]D-Fender | TheresaJayne, ok, that is making this wholse setup just too damn messy. |
| 18:25.57 | [TK]D-Fender | TheresaJayne, use * to do all the work. |
| 18:25.58 | TheresaJayne | 259766 is the number on the asterisk |
| 18:26.17 | TheresaJayne | ?? |
| 18:26.33 | [TK]D-Fender | TheresaJayne, don't dial DIRECT from the Snom to your ITSP, go THROUGH *. |
| 18:26.48 | TheresaJayne | itsp? |
| 18:27.05 | [TK]D-Fender | TheresaJayne, you have taken all these bits and pieces and have smashed your setup into so many seperate pieces it'll take forever to get your head out of your ass this way. |
| 18:27.10 | [TK]D-Fender | tovo = ITSP |
| 18:27.16 | TheresaJayne | what is * |
| 18:27.21 | [TK]D-Fender | ASTERISK. |
| 18:27.28 | TheresaJayne | i cant |
| 18:27.30 | [TK]D-Fender | ok, this is just too much. |
| 18:27.37 | TheresaJayne | our customers will be calling from itsp |
| 18:28.02 | TheresaJayne | dialling customer services on 444 |
| 18:28.11 | TheresaJayne | that routes at itsp to asterisk |
| 18:28.20 | TheresaJayne | which gives them the options |
| 18:28.33 | TheresaJayne | and redirects to the appropriate external number |
| 18:28.54 | [TK]D-Fender | TheresaJayne, Sorry there is just too much going on for me to do this all for you. |
| 18:29.12 | TheresaJayne | i just cant work out why its intercepting the inbound |
| 18:29.26 | TheresaJayne | if i just have an inbound register line |
| 18:29.30 | TheresaJayne | with no block |
| 18:29.45 | TheresaJayne | it works as expected without being able to reroute to itsp |
| 18:30.01 | TheresaJayne | ass the block to route to itsp and it stops getting inbound calls |
| 18:30.04 | TheresaJayne | add |
| 18:31.04 | TheresaJayne | its looking more and more at the moment that asterisk cant do the job which means blowing away the server and replacing with a windoze rubbish server to run prophecy |
| 18:32.01 | TheresaJayne | (not my choice though( |
| 18:34.10 | [TK]D-Fender | TheresaJayne, you could probably run the whole setup open, forget seperate users, and just fill in the extens in there. |
| 18:34.23 | TheresaJayne | ??? |
| 18:34.51 | [TK]D-Fender | TheresaJayne, You were almost there in the way you had it itn the beginning when calls were landing on [INVALID] under your general. |
| 18:35.11 | TheresaJayne | you mean have them land on sip |
| 18:35.18 | TheresaJayne | and have the exten in there |
| 18:35.38 | TheresaJayne | but then how do i call back ou |
| 18:35.40 | TheresaJayne | out |
| 18:36.47 | [TK]D-Fender | TheresaJayne, for that you should have seperate PEER entries. |
| 18:37.14 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 18:37.42 | TheresaJayne | ok well i have commented out the peers and set the default to sip |
| 18:37.44 | [TK]D-Fender | TheresaJayne, unless your ITSP allows you to use a single one to cover them all and set CID, etc accordingly (and don't have to seperater for billing, etc. |
| 18:37.51 | TheresaJayne | calling in plays the message as required |
| 18:38.12 | *** part/#asterisk DrRighteous (n=DrRighte@ool-44c7ad06.dyn.optonline.net) |
| 18:38.12 | *** join/#asterisk matsk (i=matsk@h110n2fls32o882.telia.com) |
| 18:39.14 | TheresaJayne | uncomment the peer entry and it doesnt answer and has the fail again |
| 18:40.03 | [TK]D-Fender | TheresaJayne, When you started, the only thing you needed to accept the call was to put an exten in the context it was falling under. |
| 18:40.13 | TheresaJayne | yes done |
| 18:40.15 | [TK]D-Fender | TheresaJayne, 2 lines could have had THAT answering your call. |
| 18:40.16 | TheresaJayne | worked ok |
| 18:40.30 | TheresaJayne | now i need to dial an itsp number |
| 18:40.47 | TheresaJayne | add the peer block for that and it fails to answer now |
| 18:41.00 | [TK]D-Fender | TheresaJayne, you never showed me it working and you haven't shown me it FAILING to call out after. |
| 18:41.07 | *** join/#asterisk andresmujica (n=andresmu@190.24.106.145) |
| 18:41.08 | TheresaJayne | the inbound call sees the outbound block |
| 18:41.17 | TheresaJayne | ok i will do the logs and configs |
| 18:41.21 | TheresaJayne | bear with me |
| 18:43.16 | rob0 | Ahhhh, sometimes dial tone is a Beautiful Thing |
| 18:43.39 | TheresaJayne | working http://pastebin.ca/620744 |
| 18:45.03 | *** join/#asterisk gardo (n=gardo@125.212.15.240) |
| 18:45.52 | TheresaJayne | nonworking http://pastebin.ca/620750 |
| 18:46.05 | TheresaJayne | only the changed file included in that second one |
| 18:47.38 | [TK]D-Fender | TheresaJayne, fine so it works. YAY |
| 18:48.39 | TheresaJayne | as soon as i add the peer it doesnt |
| 18:49.39 | [TK]D-Fender | TheresaJayne, Add a user first then a peer, then reverse if it doesn't work. |
| 18:50.43 | TheresaJayne | user added |
| 18:50.45 | TheresaJayne | works ok |
| 18:51.49 | TheresaJayne | in a dial is there a way to send a user/pass to the outbound |
| 18:52.05 | [TK]D-Fender | TheresaJayne, thats what your PEER is for. |
| 18:52.45 | TheresaJayne | change the user to a friend and it fails to answer |
| 18:53.13 | *** join/#asterisk Penggu (n=me@203-213-102-59-nme-ts7-2600.tpgi.com.au) |
| 18:53.20 | [TK]D-Fender | TheresaJayne, did I say *FRIEND* in there?!/!!?: |
| 18:53.28 | TheresaJayne | peer same thing |
| 18:53.31 | [TK]D-Fender | NO |
| 18:53.43 | Penggu | hi all.. how would i Page an extension, but play a chime, before allowing a caller to speak? |
| 18:53.47 | TheresaJayne | ie change user to friend or peer same result |
| 18:53.53 | Penggu | i know about the sip-add-heaer, and Page() part |
| 18:54.03 | Penggu | but how to fit it all in with a chime |
| 18:54.26 | [TK]D-Fender | Penggu, what kind of phones? |
| 18:54.32 | Penggu | um... |
| 18:54.37 | Penggu | i havent picked a softphone yet |
| 18:54.50 | Penggu | (which will be hooked up to a sound card -> PA system) |
| 18:55.09 | Penggu | windoze box.. any suggestions? |
| 18:55.22 | [TK]D-Fender | Penggu, with auto-answer capabilities? dunno. |
| 18:55.38 | [TK]D-Fender | Penggu, nothing I know of for free with that |
| 18:56.11 | *** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net) |
| 18:56.16 | Penggu | okay, let's assume it's a snom 320 |
| 18:56.18 | Penggu | (which we have) |
| 18:56.35 | rbd | anyone ever get the asterisk "ices" command to work with ices0 (which supports mp3 streaming), rather than ices2 (ogg streaming source) |
| 18:56.39 | Penggu | i could prolly mod a connector for theext-audio jack |
| 18:56.57 | TheresaJayne | ok i have finally made some progress |
| 18:57.10 | TheresaJayne | i have a peer with it still answering |
| 18:57.55 | Penggu | i've got: SIPAddHeader(Call-Info:<sip:asterisk>\;answer-after=0) and then: Page(SIP/bell)... do i just add Playback(somefile) after it? or...? |
| 18:58.25 | TheresaJayne | http://pastebin.ca/620765 |
| 18:58.32 | TheresaJayne | is the working sip.conf |
| 18:58.44 | TheresaJayne | i dropped the secret from the peer block |
| 19:00.04 | TheresaJayne | penguu thats something i need in the future as well, |
| 19:00.30 | TheresaJayne | i need to take the incoming call, put on hold, dial the new number play a message then transfer the call |
| 19:01.23 | *** join/#asterisk af_ (n=getsmart@81-174-44-88.dynamic.ngi.it) |
| 19:02.56 | Keltus | has anyone here ever tried making an AGI script? |
| 19:03.36 | Penggu | you;ll probably find plenty of those |
| 19:05.06 | TheresaJayne | [TK]D-Fender you have been very patient so far but an a little confused with the next part |
| 19:05.17 | [TK]D-Fender | Keltus, ask your REAL question |
| 19:05.19 | TheresaJayne | now when it tries to call out via the tovo-out peer |
| 19:05.52 | Keltus | when I try to call in, it runs the AGI script and I get this: |
| 19:05.53 | Keltus | <PROTECTED> |
| 19:05.59 | TheresaJayne | its trying to auth with the incoming details, is there a way to tell it to auth using a different set of credentials |
| 19:06.22 | Keltus | but it exists because I did an ls and it worked |
| 19:06.23 | TheresaJayne | ie its trying to auth outbound as 210290 rather than 259766 |
| 19:06.39 | Keltus | -rwxr-xr-x 1 root root 268 2007-07-15 11:58 /var/lib/asterisk/agi-bin/python-test.agi |
| 19:09.06 | [TK]D-Fender | Keltus, pastebin your dialplan, the full CLI output, and the contents of your AGI |
| 19:09.13 | Penggu | anyoen know of any softphones that can be run win windoze as a service? |
| 19:12.09 | Keltus | [TK]D-Fender: okay, but it's pretty simple |
| 19:12.47 | [TK]D-Fender | Keltus, PASTEBIN |
| 19:12.48 | [TK]D-Fender | ~pb |
| 19:12.49 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org |
| 19:15.02 | Keltus | [TK]D-Fender: http://pastebin.ca/620795 |
| 19:15.22 | Penggu | hmm.. looks like im going to have fun with this stuff... |
| 19:16.59 | *** join/#asterisk mitcheloc (n=mitchelo@adsl-68-120-226-30.dsl.irvnca.pacbell.net) |
| 19:17.32 | [TK]D-Fender | Keltus, what user is * running as? |
| 19:17.45 | [TK]D-Fender | Keltus, Set as 777 to be sure as well |
| 19:17.48 | Keltus | what is * |
| 19:18.03 | Keltus | set what to 777? |
| 19:18.05 | Keltus | the agi file? |
| 19:18.17 | [TK]D-Fender | yes |
| 19:18.23 | [TK]D-Fender | 8 = ASTERISk |
| 19:18.25 | [TK]D-Fender | * |
| 19:18.33 | Keltus | ahh |
| 19:18.39 | Keltus | I don't know what user it runs as |
| 19:18.41 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 19:19.02 | Keltus | I set to 777. same thing |
| 19:19.11 | Keltus | it runs the other agi in that directory without problems |
| 19:19.16 | Keltus | ie. the example perl ones |
| 19:19.40 | Penggu | ok i give up... anyone here have any experience with SIP APIs? anyting like that exist out there? |
| 19:19.57 | Penggu | .. for windoze |
| 19:20.35 | juuva | Penggu: isn't there tapi or something like that |
| 19:21.17 | Keltus | [TK]D-Fender: any other ideas how I could deug it? |
| 19:22.03 | [TK]D-Fender | Keltus, can you run it from *nix CLI? |
| 19:22.17 | Keltus | well I can run it. it doesn't do anything |
| 19:22.19 | *** join/#asterisk KovBal1981 (n=KovBal@217.27.219.152) |
| 19:22.22 | Keltus | it runs just fine |
| 19:22.40 | [TK]D-Fender | Keltus, so your path to python, etc is fine? |
| 19:22.51 | Keltus | yes |
| 19:22.52 | [TK]D-Fender | Keltus, What user are you running * as? |
| 19:23.02 | TheresaJayne | penggu I read somewhere that asterisk uses simple pipes for the api |
| 19:23.08 | Keltus | root |
| 19:24.03 | Keltus | does asterisk perhaps cache the available AGI scripts by any chance? |
| 19:24.33 | Penggu | hmm.. #openwengo |
| 19:24.40 | *** join/#asterisk Un1x_ (i=Un1x@CPE000c256d416d-CM0012c9213a06.cpe.net.cable.rogers.com) |
| 19:25.38 | [TK]D-Fender | Keltus, dunno... hrm |
| 19:26.30 | KovBal1981 | hello, I have a problem setting up outgoing CID on a TE110P I'm running asterisk 1.4.7.1, and I use the new form of set{callerid(num)} . can anyone help me please? |
| 19:28.31 | Keltus | will keep at it. thanks guys |
| 19:29.16 | [TK]D-Fender | KovBal1981, Pastebin what you're doing,a nd the CLI outputof the failed attempt |
| 19:29.17 | [TK]D-Fender | ~pb |
| 19:29.17 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org |
| 19:29.18 | *** join/#asterisk [[blah]asdf (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 19:29.28 | KovBal1981 | http://pastebin.ca/620822 |
| 19:29.37 | KovBal1981 | this is the part of the extensions conf |
| 19:29.48 | [[blah]asdf | how 'bout a non asterisk question... if someone in the UK is giving me measurements of 200 X 160 X 300, how would that work out in inches? |
| 19:30.04 | KovBal1981 | but no failed attempt at the cli, because it will dial out with the default pri number |
| 19:31.00 | TheresaJayne | depends |
| 19:31.09 | TheresaJayne | I am in the uk and use inches |
| 19:31.34 | TheresaJayne | and ir depends on the units |
| 19:31.41 | TheresaJayne | could be mm cm m miles |
| 19:31.47 | [[blah]asdf | i am guessing that this measurement is not in centimeters either... must be mm |
| 19:32.05 | TheresaJayne | which is 20cmx16cmx300cm |
| 19:32.13 | TheresaJayne | 30cm |
| 19:32.21 | TheresaJayne | and 30cm is about a foot |
| 19:32.26 | TheresaJayne | (12") |
| 19:32.41 | [[blah]asdf | what is the conversion for mm to inches? |
| 19:32.54 | Iamnacho | 2.54cm=1inch |
| 19:33.29 | TheresaJayne | which means 25.4mm = 1inch |
| 19:33.33 | *** join/#asterisk saftsack (n=saftsack@83-131-213-63.adsl.net.t-com.hr) |
| 19:34.10 | [[blah]asdf | that looks better... must be mm... ok, thanks.. .back to asterisk. (someone sending me a CSU/DSU... wondering its size. ;-) |
| 19:34.50 | *** part/#asterisk [[blah]asdf (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 19:48.40 | *** part/#asterisk KovBal1981 (n=KovBal@217.27.219.152) |
| 19:51.57 | SuPrSluG | KovBal1981:try putting the number in <#####> |
| 19:52.41 | *** join/#asterisk ivanfm (n=ivanfm@c934f322.virtua.com.br) |
| 19:54.05 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 19:59.35 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 20:07.38 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 20:16.10 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 20:22.45 | *** join/#asterisk Hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net) |
| 20:23.26 | crimethinker | anyone familiar with the tdm400p boards? |
| 20:26.02 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 20:26.42 | [TK]D-Fender | crimethinker, plenty of us, jsut come out and ask your real question |
| 20:26.48 | *** join/#asterisk visik7 (n=dksakd@unaffiliated/visik7) |
| 20:27.04 | visik7 | hi |
| 20:27.36 | visik7 | anyone know if is possible to use aterisk as an mcu for netmeeting ? |
| 20:28.32 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 20:31.59 | Hmmhesays | bah creating ring groups from a database sucks |
| 20:36.13 | [TK]D-Fender | Hmmhesays : think smarter, not harder |
| 20:36.57 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 20:45.07 | Hmmhesays | [TK]D-Fender: always |
| 20:45.10 | Hmmhesays | still kind of sucks |
| 20:45.24 | Hmmhesays | i'm using cmd mysql for this particular install |
| 20:46.21 | Hmmhesays | I heard something about multirow in func_odbc |
| 20:46.30 | Hmmhesays | cause with cmd mysql you can only return one row at a time |
| 20:50.38 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 21:17.53 | *** join/#asterisk vn (n=vn@modemcable083.222-80-70.mc.videotron.ca) |
| 21:22.58 | *** part/#asterisk HockeyInJune (n=HockeyIn@pool-70-107-176-9.ny325.east.verizon.net) |
| 21:35.08 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 21:35.26 | Hmmhesays | now this is just freaking weird |
| 21:42.48 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 21:54.48 | Hmmhesays | bah arrays would be cool in asterisk |
| 21:59.17 | [TK]D-Fender | Hmmhesays, You can fake them with vars most of the time |
| 22:03.49 | Hmmhesays | yeah I just figured that out |
| 22:03.57 | *** join/#asterisk KDan (n=KDan@87-194-122-30.bethere.co.uk) |
| 22:04.04 | Hmmhesays | I'm trying to keep my sql connections open for as short a time as possible |
| 22:07.21 | *** join/#asterisk jeffik (n=Valued@206-248-129-240.dsl.teksavvy.com) |
| 22:08.04 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 22:09.57 | Hmmhesays | is there any problems s,n,Return from inside a while loop? |
| 22:15.16 | *** part/#asterisk jeffik (n=Valued@206-248-129-240.dsl.teksavvy.com) |
| 22:21.09 | *** join/#asterisk sigmounte (n=sigmount@81.56.234.199) |
| 22:23.11 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 22:23.44 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-181-57.dsl.irvnca.pacbell.net) |
| 22:23.52 | BSD_Tech | ok menuselect has to go |
| 22:24.51 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 22:25.41 | brookshire | too complicated? |
| 22:26.14 | [TK]D-Fender | Screw that..... USERS.CONF has to go..... |
| 22:26.31 | brookshire | hah.. users.conf is my favorite feature of asterisk 1.4 |
| 22:26.56 | [TK]D-Fender | brookshire, BLARG |
| 22:27.10 | [TK]D-Fender | brookshire, over associated doggy-doo! |
| 22:27.42 | brookshire | :( |
| 22:27.44 | brookshire | but it's easy! |
| 22:28.09 | brookshire | and when you have to manage 100 voip geeks who all have their favorite protocol, it's great! |
| 22:29.01 | [TK]D-Fender | brookshire, what we really need is "type=register" for sip.conf, for "type=peer" and "type=user" to actually MEAN it again (not matching incoming calls against PEERS (ARRRGGGGHHH!!!!!!!!!!!!!!!)) |
| 22:32.08 | Juggie | [TK]D-Fender, submit patches :) |
| 22:32.13 | russellb | BSD_Tech: would you like to elaborate? |
| 22:32.45 | [TK]D-Fender | Juggie, screw the patch..... some people need to lay off "The Crack-Pipe" cold-turkey! |
| 22:33.04 | Juggie | [TK]D-Fender, i dont use sip soo i dont know your pain :) |
| 22:33.23 | [TK]D-Fender | Juggie, Really.. what DO you use? |
| 22:33.35 | russellb | voice over cans and string |
| 22:33.38 | Juggie | [TK]D-Fender, just zap & iax |
| 22:33.45 | brookshire | russell: voice under ip |
| 22:34.01 | Juggie | i dont use asterisk as a general office pbx |
| 22:34.13 | [TK]D-Fender | best non-answers ever! :) |
| 22:34.27 | Juggie | we use asterisk to deliver automated services |
| 22:34.42 | Juggie | ivr's. |
| 22:35.14 | [TK]D-Fender | Juggie, Ah, well that is a real use, but quite the minority. |
| 22:35.45 | Juggie | perhaps |
| 22:35.56 | Juggie | phones are boaring, anyone can do that |
| 22:36.28 | Juggie | working with services and agents etc. |
| 22:36.31 | Juggie | thats more of a challenge :) |
| 22:37.24 | BSD_Tech | in bsd you cant run the menuselect when you build from ports |
| 22:37.37 | BSD_Tech | and its a pain in the ass to have |
| 22:38.09 | BSD_Tech | there for when we try to create a pkg for pkg_add -r we cant |
| 22:38.52 | *** join/#asterisk sergey (n=sergey@sergey.iks.ru) |
| 22:39.17 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 22:39.55 | brookshire | BSD_Tech: i believe menuselect just generates makefiles.. |
| 22:40.16 | BSD_Tech | no |
| 22:40.25 | BSD_Tech | it set build options |
| 22:40.34 | BSD_Tech | not just makes make files |
| 22:40.35 | sergey | Hi. In 1.4 new SLA system. What for it is? What case I need use trunk layer? |
| 22:40.54 | BSD_Tech | SLA needs work |
| 22:41.00 | BSD_Tech | diff issue |
| 22:41.08 | BSD_Tech | let me finish with menuselect |
| 22:41.38 | brookshire | so.. it's just scripts, what's your problem? |
| 22:42.16 | russellb | BSD_Tech: you are making no sense |
| 22:42.31 | russellb | you haven't identified an actual problem yet |
| 22:42.45 | russellb | "you cant run the menuselect when you build from ports" |
| 22:43.05 | russellb | um, ok? what is the *problem* |
| 22:44.33 | sergey | well, when migrate from 1.2 to 1.4 is image to use trunks and sla right now? |
| 22:44.33 | BSD_Tech | when you try to create a package for later install like a ropm for redhar we call them pkgs |
| 22:44.48 | russellb | i know that. |
| 22:45.12 | BSD_Tech | if it was not for menuselect then it would |
| 22:45.17 | Innatech | is there something special I have to do to make poly501's pull PoE? I have the adapters and a PoE switch, but they don't turn on w/o power from their bricks. The 430s work fine. |
| 22:45.29 | BSD_Tech | 1.2.x had no issues its only 1.4 and manuselect |
| 22:45.41 | BSD_Tech | so it makes it crap now |
| 22:45.45 | russellb | you haven't identified any issues |
| 22:45.47 | BSD_Tech | I want a package |
| 22:45.52 | russellb | then make one |
| 22:45.56 | BSD_Tech | yes I have |
| 22:46.01 | russellb | no you haven't |
| 22:46.23 | brookshire | ./configure --help |
| 22:46.25 | brookshire | :D |
| 22:46.32 | BSD_Tech | the issue is if you stop a built in ports and manualy run the menuselect |
| 22:46.46 | BSD_Tech | then go back and run make package |
| 22:47.04 | BSD_Tech | it fials because it cant stat the extra options |
| 22:47.17 | BSD_Tech | we did not have this issue befor the menu select |
| 22:47.41 | BSD_Tech | it just name makes it a pain in the a$$ |
| 22:47.48 | BSD_Tech | not to be able to make a pkg |
| 22:48.00 | BSD_Tech | because menuselect breaks it |
| 22:48.18 | *** join/#asterisk alexhopper (n=a27386@mctnnbsa24w-142167040033.pppoe-dynamic.nb.aliant.net) |
| 22:48.25 | russellb | well i'm sorry that you're not able to figure out how it works |
| 22:48.35 | BSD_Tech | its not that |
| 22:48.45 | BSD_Tech | your not understanding the bnsd ports setup |
| 22:48.56 | BSD_Tech | and we generate a plist |
| 22:49.05 | BSD_Tech | for makeing a port package |
| 22:49.34 | russellb | you build asterisk with ./configure && make && make install, as with 1.2 you build with make && make install |
| 22:49.45 | russellb | i'm not sure what is so difficult |
| 22:49.45 | BSD_Tech | not on bsd |
| 22:49.53 | russellb | running on menuselect is an optional step |
| 22:49.57 | BSD_Tech | on bsd we use a ports tree |
| 22:50.15 | russellb | right .. |
| 22:50.20 | BSD_Tech | you have to run it to get the extra sounds |
| 22:50.25 | russellb | but i don't understand why building asterisk through ports is so much harder |
| 22:50.35 | russellb | you don't *have* to run it |
| 22:50.40 | russellb | it is completely optional to run menuselect |
| 22:50.57 | BSD_Tech | then how else would you get the extra options |
| 22:51.04 | BSD_Tech | thats the issue |
| 22:51.06 | russellb | go download them manually |
| 22:51.15 | russellb | this only adds features, not removes them |
| 22:51.35 | BSD_Tech | menuselect allow you to add and rm opotions |
| 22:51.41 | russellb | right |
| 22:51.44 | russellb | but you don't have to use it |
| 22:51.52 | russellb | and it doesn't make sense to use it in an automated build system |
| 22:58.28 | De_Mon | 0/j #snmp |
| 23:00.09 | De_Mon | voip-info's snmp documentation of asterisk needs work |
| 23:00.41 | file | you just volunteered to improve it! :D |
| 23:01.32 | De_Mon | file Deal!.. now i just have to figure out how it works |
| 23:01.51 | BSD_Tech | to ginish my iso setup |
| 23:01.59 | BSD_Tech | I have everything but asterisk |
| 23:02.08 | De_Mon | file so far I've figured out that res_snmp.conf has to be edited.. |
| 23:03.16 | De_Mon | but.. I'm not sure what community string its using or anything fancy like that |
| 23:04.03 | JT | Innatech: is it plugged in the right way around? |
| 23:04.27 | *** join/#asterisk [hC] (n=hardcore@rmtw.com) |
| 23:04.28 | BSD_Tech | Alex whats up |
| 23:04.41 | Innatech | JT: so far as I can tell. I've checked the included diagram twice |
| 23:05.01 | Innatech | switch--->patch cable--->adapter---->phone. |
| 23:05.33 | brookshire | db_mon: you could always try emailing the author :) |
| 23:05.54 | Innatech | I'm wondering if it's a problem with the PoE autosensing in the switch.... |
| 23:05.58 | JT | Innatech: is the adapter plugged in the correct way around? |
| 23:06.14 | wunderkin | there is a notch in the phone side so you can only plug it in one way |
| 23:06.46 | De_Mon | brookshire hmm? |
| 23:07.10 | wunderkin | sounds like a wiring or switch problem |
| 23:07.55 | wunderkin | Innatech, all of the pairs are connected, and it goes directly into the switch? |
| 23:08.38 | Innatech | wunderkin: yeah, and its two units from different vendors. PoE is working fine on the 430s attached to the same switch. |
| 23:09.41 | Innatech | Hey....wait a minute. Are the PoE adapter and the PoE injector two different items? As in, does the cable that takes external power not work w/PoE when the power is disconnected? If so, I think I was sold the wrong cables. |
| 23:09.49 | wunderkin | i guess you would determine that by putting a 501 on one of the working 430 wall ports |
| 23:10.15 | Innatech | No wall ports. Plugged directly into the switch, this is pre-deployment. |
| 23:10.21 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 23:10.35 | JT | Innatech: there are two different cables |
| 23:10.43 | JT | Innatech: does your cable have a round DC in socket? |
| 23:10.50 | Innatech | JT: aha. Yes it does. |
| 23:10.54 | JT | wrong one |
| 23:10.55 | wunderkin | yeah i forgot about that |
| 23:10.56 | Innatech | they sold me the wrong one... |
| 23:10.58 | [TK]D-Fender | Innatech, then its NOT the POE adapter cable |
| 23:11.03 | JT | get the ieee802.3af PoE cable |
| 23:11.15 | JT | there's also cisco PoE cable for it, forget about that one |
| 23:11.22 | Innatech | JT: funny. That's exactly what it says on the box for the thing, but it's definitely an injector. |
| 23:12.10 | Innatech | "ASSY, TOP, 802.3af, PWR CBL, SPIP P/N 2200-11077-002 |
| 23:14.38 | Innatech | That's the right part number, too. Strange. |
| 23:14.46 | Innatech | http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-38040166656.htm |
| 23:19.07 | BSD_Tech | grrr |
| 23:19.12 | BSD_Tech | I just want a package |
| 23:19.15 | BSD_Tech | grrrr |
| 23:21.39 | *** join/#asterisk nohop (n=root@cc501678-a.hgv1.dr.home.nl) |
| 23:21.54 | nohop | good evening, telephone peoples :) |
| 23:21.55 | brookshire | bsd: use debian! |
| 23:23.16 | nohop | debian ftw! |
| 23:24.03 | BSD_Tech | Icant |
| 23:24.11 | BSD_Tech | it has to be on freebsd |
| 23:29.18 | *** join/#asterisk Spanska (n=bluetoot@c207.134.185-18.clta.globetrotter.net) |
| 23:31.53 | Innatech | OK--so it appears that this IS the right cable--even though it's the same one I would plug the DC into. I tried a second cable and a second 501 and it worked. Then the second 501 worked w/the first cable, so it appears the first 501 has some kind of problem--strange, since the power is coming from the same place. Can PoE negotiation be disabled in firmware or something? |
| 23:45.51 | BSD_Tech | finally I got it |
| 23:45.55 | BSD_Tech | the pkg built |
| 23:46.02 | BSD_Tech | wow that was a pain |
| 23:46.10 | Hmmhesays | heh |
| 23:46.11 | Hmmhesays | nice |
| 23:46.34 | BSD_Tech | I can now install my image |
| 23:52.20 | BSD_Tech | bbiab have to install pkgs' |
| 23:52.28 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |