| 00:00.50 | *** join/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 00:01.05 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:01.08 | rhombus | Are Asterisk variable names case sensitive? |
| 00:01.37 | uwe | rhombus, you mean those used in agi ? |
| 00:01.51 | rhombus | no, I just mean those used in the dialplan |
| 00:02.01 | *** join/#asterisk BB|AtWork (n=karl@38.99.18.98) |
| 00:02.26 | rhombus | if I set a custom variable, is it case sensitive? Will ${goat} be the same as ${GOAT} or are they handled as separate variables? |
| 00:02.32 | km- | uwe: yeah, I'm seriously bummed. I tried it by sox'ng the wav beforehand and it works fine |
| 00:03.02 | BB|AtWork | i have some users complaining of echo problems, but unfortunatly i can't think of a good way to test this. any time i use my mobile phone i don't get anything. Does any one know of any test numbers i can call? |
| 00:03.06 | *** join/#asterisk DrukenHME (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 00:03.15 | rhombus | Aha -- I guess they are not. |
| 00:03.19 | rhombus | thanks anyway! |
| 00:03.22 | *** part/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 00:03.28 | BB|AtWork | i've been searching for web pages with related info but have not come up with anything so far besides sip addresses |
| 00:05.03 | uwe | BB|AtWork, maybe you have issues with irq sharing or inturrupting too much |
| 00:05.13 | BB|AtWork | hrm |
| 00:06.04 | uwe | or network issues |
| 00:06.04 | uwe | maybe ... |
| 00:06.08 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 00:06.08 | BB|AtWork | Best: 100.000000 -- Worst: 98.437500 -- Average: 99.943441 |
| 00:06.19 | BB|AtWork | its a local physically seperate 10/100 network |
| 00:06.30 | BB|AtWork | using a pri t1 |
| 00:06.42 | BB|AtWork | interupt issue sounds plausible |
| 00:06.45 | BB|AtWork | it only happens some times |
| 00:06.58 | BB|AtWork | what are bad numbers for zttest? |
| 00:07.06 | Alex_20 | storna, i can't see it for now |
| 00:07.30 | Alex_20 | but after i see the 223 in the sip.conf |
| 00:07.36 | Alex_20 | what i have to do |
| 00:07.40 | *** join/#asterisk colinm_ (n=colol@VDSL-130-13-104-63.PHNX.QWEST.NET) |
| 00:08.20 | km- | Sucks, I thought I could do this project without having to make asterisk source code changes, but unfortunately AGI just doesn't do the magic I need |
| 00:08.50 | uwe | BB|AtWork, how does your /proc/interrupts look like |
| 00:09.42 | uwe | ive just had similar problems for the last 3 weeks :) |
| 00:09.57 | *** join/#asterisk kiscokid (n=xxx@208.106.33.66) |
| 00:10.46 | BB|AtWork | 10: 1719405387 0 XT-PIC 3w-9xxx, uhci_hcd, uhci_hcd, wcte11xp, wctdm |
| 00:11.18 | BB|AtWork | http://rafb.net/p/3JFv9H73.html |
| 00:11.30 | kiscokid | Anyone know why adding after adding a service provider it won't show up as a choice when adding a new calling rule? |
| 00:11.48 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 00:12.06 | kiscokid | sorry, I meant to ask that in asterisknow |
| 00:12.20 | uwe | i think your wcte11xp, wctdm are sharing inturrupt with many other stuff |
| 00:12.35 | storna | alex20_dunno |
| 00:12.47 | uwe | and all is hitting the same cpu BB|AtWork |
| 00:12.55 | BB|AtWork | uwe, looks like its sharing with the 3ware card |
| 00:13.01 | BB|AtWork | how would i go about fixing this? |
| 00:13.11 | uwe | wait a minute |
| 00:13.30 | uwe | there is the only article i actually found useful |
| 00:13.44 | uwe | http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html |
| 00:14.26 | uwe | BB|AtWork, i had to disable some hardware from the bios to get the digium on an inturrupt that is not shared |
| 00:14.40 | uwe | and ah, use lspci -vb to check inturrupts too |
| 00:14.45 | BB|AtWork | think it would be ok if 2 digium cards shared? |
| 00:14.54 | BB|AtWork | i think i can get the 3ware card onto something else |
| 00:14.55 | uwe | no ... i dont think so |
| 00:15.02 | BB|AtWork | damn this might be hard |
| 00:15.07 | uwe | yes :) |
| 00:16.25 | uwe | try first with smp_affinity for that inturrupt |
| 00:16.54 | *** join/#asterisk Alex_20 (n=Alex_20@139-62.al.cgocable.ca) |
| 00:17.12 | uwe | because the eth1 is making also a lot of inturruption |
| 00:17.52 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 00:18.26 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 00:18.48 | JT | BB|AtWork: definitely not ok for them to share |
| 00:19.12 | BB|AtWork | hurk |
| 00:19.23 | JT | "hurk"? |
| 00:19.30 | BB|AtWork | are there any problems with using apic and asterisk? |
| 00:19.43 | JT | BB|AtWork: not that i've found |
| 00:19.51 | BB|AtWork | for somereason it seems to be off |
| 00:20.01 | BB|AtWork | i'll have to see what caused that |
| 00:20.50 | BB|AtWork | yeah tahts probably what the problem is |
| 00:21.43 | BB|AtWork | we arn't using the other telephony card yet (its a 4 port fxo) but the raid controller and the pri on the same irq is probably causing trouble |
| 00:21.43 | uwe | BB|AtWork, you have to make sure they are not sharing them by checking the output of lspci -vb too |
| 00:22.10 | BB|AtWork | so i can solve this by either turning off enough devices that they all end up on there own. or turn on apic? |
| 00:22.18 | uwe | because having them on diffrent irq when using apic doesnt mean the are not sharing the same inturrupt |
| 00:22.26 | BB|AtWork | ah |
| 00:22.42 | Alex_20 | so now, what should i do? |
| 00:22.50 | BB|AtWork | this is going to be painful |
| 00:23.03 | uwe | thats what it understood in the last two days |
| 00:23.24 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 00:23.35 | uwe | um , not really, simply waiting between reboots :) |
| 00:23.45 | BB|AtWork | have to do it out of office hours |
| 00:23.53 | BB|AtWork | and there are 3 cards sharing irq's :) |
| 00:24.15 | uwe | BB|AtWork, ive had this exact experience 2 days ago :D |
| 00:24.28 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 00:24.29 | BB|AtWork | going to come in way late tomorrow |
| 00:24.30 | BB|AtWork | :) |
| 00:25.24 | uwe | hahahaaa ! i swear to the devils this must be dejavu |
| 00:26.18 | *** join/#asterisk docelmo (n=vircuser@c-76-99-157-112.hsd1.de.comcast.net) |
| 00:26.25 | BB|AtWork | :) |
| 00:26.28 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 00:26.37 | BB|AtWork | so.... come in late... or go home for a chunk of the day |
| 00:26.41 | BB|AtWork | tough decision |
| 00:27.15 | uwe | id choose come in late |
| 00:27.24 | BB|AtWork | heh i already come in at 9 :) |
| 00:27.27 | BB|AtWork | i'm going to be bored |
| 00:27.42 | BB|AtWork | but i think i'll chose that and just stay up till 4 or something |
| 00:28.08 | uwe | 4am or 4pm ? |
| 00:29.10 | *** join/#asterisk Insanity5 (i=Insanity@66-225-36-14.dynamic.tbc.net) |
| 00:29.12 | Insanity5 | Does asterisk offer any built in debugging to show jitter/out of order packet information/etc or anything else that may help show a poor call quality problem? |
| 00:30.34 | BB|AtWork | did you try set verbosity 99 set debug 99 and pri intense debug (if you have a pri line) |
| 00:31.34 | kiscokid | Is there a way to trace calls going through the dialplan? |
| 00:31.53 | *** join/#asterisk wolferine (n=profx@unaffiliated/wolferine) |
| 00:32.39 | Insanity5 | BB|AtWork - Verbo is at like 4. |
| 00:32.42 | Insanity5 | BB|AtWork - IT's SIP |
| 00:32.53 | Insanity5 | BB|AtWork - internet is suspect |
| 00:33.11 | BB|AtWork | tcpdump can show you out of order packets i think |
| 00:33.56 | Insanity5 | BB|AtWork - reconstructing might be a PITA |
| 00:33.57 | Insanity5 | :( |
| 00:35.12 | *** part/#asterisk kiscokid (n=xxx@208.106.33.66) |
| 00:37.59 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 00:38.09 | *** join/#asterisk defswork (n=andy@mailgate2.3gcomms.co.uk) |
| 00:40.05 | *** join/#asterisk fx0 (n=fx0@cypher.punk.net) |
| 00:42.59 | Alex_20 | sorry i don't remember :S |
| 00:43.48 | *** join/#asterisk Avochelm (n=damo@gw-morphett.koalatelecom.com.au) |
| 00:44.11 | JT | BB|AtWork: never share any irq with a digium card |
| 00:44.17 | JT | let alone a raid controller |
| 00:44.28 | JT | that and ethernet are about the worse things to share with |
| 00:46.38 | *** join/#asterisk Alex_20 (n=Aex_20@139-62.al.cgocable.ca) |
| 00:47.25 | red9012 | I have problem with dtmf recognition while in a call trasfer mode (ex press 1 accept/refuse call). DTMF works in all other cases. |
| 00:49.31 | Alex_20 | i have zap config and asterisk 1.4.2 |
| 00:49.51 | Alex_20 | i want to change a ext number of a phone to 223 for 229. how can i do that? |
| 00:50.03 | Alex_20 | help me pls |
| 00:53.58 | *** join/#asterisk FastFeet (n=FastFeet@CPE0013109fd25b-CM000f9fa60d7a.cpe.net.cable.rogers.com) |
| 00:56.59 | *** part/#asterisk Alex_20 (n=Aex_20@139-62.al.cgocable.ca) |
| 01:00.10 | blitzrage | anyone know how to make Asterisk not load a module again after you tell it to preload in modules.conf? |
| 01:02.08 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 01:02.26 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:07.20 | *** join/#asterisk AndrewGearhart (n=chatzill@h122.183.88.75.ip.alltel.net) |
| 01:08.21 | AndrewGearhart | is it possible to change what extension a call is forwarded to based upon the callerID information? |
| 01:08.50 | toombaloomba | yea, theres a variety of ways you can do it |
| 01:08.54 | toombaloomba | depending on what exactly you want to do |
| 01:09.22 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 01:10.12 | AndrewGearhart | toombaloomba: I was thinking along the lines of if it's one of these numbers: a, b, c... forward to extension X |
| 01:11.40 | toombaloomba | on incoming calls? |
| 01:12.29 | toombaloomba | you could just do If callerid(num) = 123456789 |
| 01:12.33 | toombaloomba | or whatever, its easy |
| 01:12.45 | toombaloomba | look on the wiki for syntax |
| 01:13.40 | AndrewGearhart | toombaloomba: k, sorry if I am asking an overly simple question that is well documented. :( was trying to answer a question for somebody over in MythTV |
| 01:13.41 | toombaloomba | or exten => 500/123 |
| 01:13.50 | JT | err that sets callerid |
| 01:14.02 | JT | that's more like it |
| 01:14.16 | toombaloomba | yea JT its not proper syntax |
| 01:14.19 | toombaloomba | its just an idea |
| 01:14.33 | JT | 500/123 would work |
| 01:14.43 | AndrewGearhart | but, yes, the idea is that it would be callerid from the outside line(s) |
| 01:15.56 | *** join/#asterisk mike38533 (n=omar@dsl092-214-151.atl1.dsl.speakeasy.net) |
| 01:16.12 | *** join/#asterisk burt75 (n=burt@189.157.128.236) |
| 01:16.18 | burt75 | hello guys |
| 01:16.40 | burt75 | I need help (payed) with FXO reverse polarity issue , wonder who can help me please :D |
| 01:18.15 | mike38533 | Hi all, I am trying to use T.38 faxing with an ATA, by receiving the call with the ATA and sending it over to * with IAXModem. Can anybody give me any pointers on how to configure this? My tests haven't been successful as the fax tone isn't recognized. |
| 01:18.52 | *** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 01:19.27 | burt75 | Looking for a consultant specialized in FXO ports |
| 01:29.33 | FuriousGeorge | im taking a poll. i have two identical systems. one has two tdm400p the other has only one. They have identical OS and kernel and everything else. the former deadlocks, the latter does not. what are the odds replacing the tdm400p (with one sangoma a200 series) will resolve the problem? |
| 01:30.10 | danp | have you tried a memory test? |
| 01:30.16 | JT | tried swapping the cards? |
| 01:30.24 | danp | that too |
| 01:31.03 | SwK | checked for shared interupts |
| 01:31.17 | SwK | other stuff running on the box that is starving it for CPU time |
| 01:31.47 | shido6 | :) |
| 01:33.48 | blitzrage | SwK: y0! |
| 01:37.58 | shido6 | no other res_odbc.so lines in there no autoload yes crap? |
| 01:39.25 | SwK | y0 blitzrage |
| 01:44.09 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 01:47.22 | *** join/#asterisk tomcontr3 (n=tomcontr@101-77-246-201.adsl.terra.cl) |
| 01:47.38 | tomcontr3 | hi, can anyone help me with the G729 codec? |
| 01:48.15 | tomcontr3 | I just got that codec, and a cofigure my extentions with allow=g729 and disallow=all |
| 01:48.28 | tomcontr3 | but every time I try to call an extention, I get Service Unavailable |
| 01:48.46 | blitzrage | SwK: have you ever done a preload, then saw that the module loads a 2nd time during the standard module load? |
| 01:49.07 | tomcontr3 | how can I do that? |
| 01:49.24 | *** join/#asterisk hansin321 (n=hansin32@c-71-196-138-170.hsd1.co.comcast.net) |
| 01:49.45 | shido6 | hehe |
| 01:50.03 | blitzrage | shido6: we should have been having our conversation out here :) |
| 01:50.10 | shido6 | yeah |
| 01:50.48 | blitzrage | maybe just paste the last part of the conversation.... ? |
| 01:50.50 | shido6 | i think it would be simpler to add the donotloadagainmodule => module.so but the 2nd version should have a weighted module loading system. |
| 01:50.51 | blitzrage | anyone interested? :) |
| 01:51.14 | tomcontr3 | anyone? |
| 01:51.27 | shido6 | parts of the system, necessary parts weigh more or are more relevant to the core than others |
| 01:51.48 | shido6 | then u have the option to FORCE the order that the modules load |
| 01:52.19 | shido6 | that sounds like what you need in loader.c |
| 01:52.32 | shido6 | force the order but have the system bitch at you about its relevance. |
| 01:54.52 | tomcontr3 | ? |
| 01:58.01 | *** join/#asterisk ccole (n=cole@cpe-24-166-57-30.neo.res.rr.com) |
| 02:02.06 | LeddyHM | hrm |
| 02:02.30 | LeddyHM | anyone know which settings change the date/time format on a polycom phone? |
| 02:02.47 | LeddyHM | I'm guessing it's sip.cfg, but can't make heads or tails of which does what |
| 02:02.55 | *** join/#asterisk l2cache (n=admin@62.180.8.67.cfl.res.rr.com) |
| 02:03.56 | l2cache | I have a box that can make calls successfully using g729. Because i have the format_g729 module in usr/lib/asterisk/modules. Why isn't background(filename.g729) playing back? |
| 02:03.58 | justdave | look for the SNTP tag |
| 02:04.10 | shido6 | ipmid.cfg |
| 02:04.14 | LeddyHM | that changes location on the screen? |
| 02:04.20 | shido6 | LEddyHM i think its in ipmid.cfg |
| 02:04.35 | justdave | oh, sorry, I thought you meant setting the time, didn't realize you meant how to display it |
| 02:04.59 | shido6 | dattime |
| 02:05.04 | shido6 | yes has "Top" |
| 02:05.06 | shido6 | in there |
| 02:05.16 | shido6 | lcl.datetime.date.dateTop="1" |
| 02:05.32 | shido6 | what does lcl.datetime.date.dateBottom="1" do? |
| 02:05.33 | shido6 | :) |
| 02:06.33 | l2cache | any takers? |
| 02:06.53 | justdave | g729 is a licensed codec, isn't it? do you have a license? |
| 02:06.54 | shido6 | ? |
| 02:07.06 | shido6 | ".g729 ? |
| 02:07.07 | l2cache | not the open source on...i am only using passthrough |
| 02:07.09 | LeddyHM | I have myt date at the bottom, and datetop says 1 |
| 02:07.11 | LeddyHM | weird |
| 02:07.14 | shido6 | is that the suffix? |
| 02:07.16 | l2cache | one* |
| 02:07.20 | l2cache | yes |
| 02:07.23 | justdave | passthrough works without a license. |
| 02:07.28 | shido6 | dont include the suffix |
| 02:07.40 | l2cache | good tip..(trying now) :) |
| 02:07.57 | shido6 | the original filename should have one |
| 02:08.07 | l2cache | gotcah |
| 02:08.10 | shido6 | but in the dialplan u dont use it with background or playback |
| 02:08.49 | justdave | LeddyHM: if you have the manual for the Polycom phones it on page 83 |
| 02:08.57 | justdave | section 4.6.1.3.2 |
| 02:09.17 | justdave | there is no dateBottom |
| 02:09.34 | justdave | dateTop is 1 for date above the time, 0 for time above the date |
| 02:10.58 | justdave | supposed I should clarify, I'm looking at the manual for IP501 :) not sure they're all the same |
| 02:11.41 | justdave | guess it is, it says "SoundPoint®/SoundStation® IP SIP" on the front cover, no model number |
| 02:12.45 | l2cache | everytime i try to do a playback it says unable to open "filename" does not exist in any format |
| 02:12.50 | l2cache | and i know its in the right dir |
| 02:12.59 | l2cache | because i can playback gsm fine |
| 02:13.22 | justdave | l2cache: show formats |
| 02:13.30 | justdave | see if it's listed |
| 02:14.07 | l2cache | i dont have that command... |
| 02:14.15 | justdave | hmm, maybe that's 1.4 only |
| 02:14.23 | l2cache | im using 1.4 |
| 02:14.40 | l2cache | show codecs? |
| 02:14.46 | justdave | show codecs seems to work |
| 02:14.51 | l2cache | yeah |
| 02:14.51 | justdave | but that's not quite the same thing |
| 02:15.34 | l2cache | i have the format_g729 module..and can talk fine...why wont it playback a .729 file then? |
| 02:15.42 | justdave | show modules like format |
| 02:15.44 | l2cache | says does not exist in any format.....maybe its a diff g729 |
| 02:15.54 | justdave | .729 or .g729 ? |
| 02:15.55 | LeddyHM | I'm wondering about 501 specifically |
| 02:16.02 | lee_is_me | Question: ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 02:16.03 | l2cache | ahhh..good point |
| 02:16.06 | LeddyHM | is the doc available for dl? |
| 02:16.12 | *** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 02:16.15 | lee_is_me | i have zaptel.conf and zapata.conf configured |
| 02:16.16 | l2cache | .g279 |
| 02:16.24 | l2cache | i need .729 |
| 02:16.40 | l2cache | correct? |
| 02:17.19 | lee_is_me | but keep getting the above error with /sbin/ztcfg -vv |
| 02:17.32 | justdave | LeddyHM: http://www.polycom.com/resource_center/0,,pw-16595-16595,FF.html |
| 02:18.25 | l2cache | i think my problem is im playing back the wrong sound files....wrong g729 files :) |
| 02:18.32 | justdave | LeddyHM: the Admin Guide is the one you're looking for |
| 02:24.16 | ccole | What is a quality, reputable place to get IAX service through? |
| 02:24.42 | ccole | So far, I am pleased with NuFone, but they do not have any 330 area codes :~( |
| 02:24.45 | LeddyHM | found it thanks |
| 02:24.54 | LeddyHM | even if my config is contradictory :| |
| 02:37.19 | *** join/#asterisk sione (i=sione@208-46-202-201.dia.static.qwest.net) |
| 02:38.59 | sione | can anyone help me trouble shoot a SIP loop problem? |
| 02:39.32 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 02:41.25 | sione | everyone is asleep? |
| 02:41.32 | Putzz | zzzzzzzz |
| 02:41.42 | sione | bummer |
| 02:41.53 | Putzz | whats your problem? |
| 02:42.26 | sione | trying to get a phone behind asterisk-A to call a phone behind astrisk-b via SIP all the way |
| 02:42.48 | Putzz | ok |
| 02:43.03 | sione | 37.786956 192.168.0.1 -> 192.168.3.129 SIP/SDP Request: INVITE sip:6001@192.168.3.129, with session description |
| 02:43.12 | sione | tahts the 1st |
| 02:43.17 | sione | <PROTECTED> |
| 02:43.25 | sione | <PROTECTED> |
| 02:43.27 | Putzz | dont flood channel |
| 02:43.38 | Putzz | ~pb |
| 02:43.50 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 02:43.50 | sione | trying to post slowly sorry |
| 02:43.50 | Putzz | postbin.ca |
| 02:43.51 | shido6 | pastebin.ca your post |
| 02:43.51 | Putzz | or whatver u prefer |
| 02:44.03 | Putzz | *pastebin.ca oops |
| 02:44.05 | sione | let me check it out |
| 02:45.06 | sione | phone (2222) calls (6004) when when 6004 picks up Astersk sends an invite to 2222 with 6004 then asterisk reportd loop and kills the channel |
| 02:46.29 | *** join/#asterisk bluelinq (n=bluelinq@dsl-7-36.cofs.net) |
| 02:47.03 | bluelinq | hey guys, anyway to make a conference call using a 7960 with chan_sccp wiouth using the meetme stuff? |
| 02:47.35 | *** join/#asterisk raptorra1 (n=rathomps@cpe-66-25-25-138.houston.res.rr.com) |
| 02:48.17 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 02:50.15 | *** join/#asterisk tenzind (n=tenzind@202.144.144.77) |
| 02:52.15 | *** join/#asterisk NovceGuru (n=asdf@oh-71-50-248-25.dhcp.embarqhsd.net) |
| 02:52.24 | l2cache | i need a g729 translation codec for a wrt54g...anyone ever heard of this |
| 02:52.36 | l2cache | i know its processor specific so it'd be an awesome find! |
| 02:53.37 | wunderkin | i really doubt it would have enough juice for g729 |
| 02:53.48 | l2cache | i know..but i need it for testing reasons |
| 02:54.00 | l2cache | any ideas if anyone's tried it? |
| 02:54.06 | wunderkin | what good is testing if it will not even work |
| 02:54.11 | l2cache | it will |
| 02:54.16 | l2cache | is a 266mhz box |
| 02:54.16 | wunderkin | ha |
| 02:54.24 | l2cache | and i have a similar box doing that fine |
| 02:54.36 | raptorra1 | wunderkin: with out testing one will never know |
| 02:54.37 | l2cache | but you've never heard of anyone trying then? |
| 02:54.56 | *** join/#asterisk thoughtpolice (n=austin@c75-111-138-216.plaicmtc01.tx.dh.suddenlink.net) |
| 02:55.27 | l2cache | ? |
| 02:55.44 | raptorra1 | http://forums.whirlpool.net.au/forum-replies-archive.cfm/356705.html |
| 02:55.48 | raptorra1 | might be of interest |
| 02:56.03 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 02:56.11 | l2cache | yeah ive seen that one...one guy mentions 729..but no others did |
| 02:56.47 | raptorra1 | what processor is in the wrt54g |
| 02:56.57 | mitcheloc | a p4 |
| 02:57.00 | l2cache | i can passthrough fine..but since the module is processor specific i can't find a translation codec module |
| 02:57.07 | l2cache | mitcheloc you are no help |
| 02:57.13 | mitcheloc | :) |
| 02:57.41 | l2cache | 32-bit MIPS architecture processors by broadcom |
| 02:57.50 | raptorra1 | I would guess an arm, but I'm using my wrt54g and really don't feel like cracking it |
| 02:58.28 | l2cache | http://en.wikipedia.org/wiki/MIPS_architecture |
| 03:00.19 | l2cache | i really wish some one knew about this |
| 03:00.34 | NormanAthol | hey guys i have just set up an IAX trunk but every where i have read that the sending server need type=user and the reciving server need type =peer this dosent work for me i need to have both of the set as the same type could they be guides based on 1.2 or somethign |
| 03:00.50 | mosty | we are intermittently getting some dropouts on iax calls, with trunk_queue: Maximum trunk data space exceeded to <host> messages in the logs. looking at the code there seems to be a fixed upper limit for the amount of iax trunking you can do. is there any way around this? |
| 03:01.00 | l2cache | if they are talking both ways use type=friend |
| 03:03.30 | NormanAthol | yeah i will be using type friend but then that raises another question for me the sleave sever connects to the master server is ther anything different i need to do in the dial plan if the master server wants to call someone on the slave server or once its registered then it should all work fine |
| 03:03.45 | l2cache | should work find both ways |
| 03:03.52 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:04.28 | NormanAthol | ok thanks |
| 03:07.48 | sione | anyone know why i cant select crc_ccitt in my make menuconfig for kernel 2.6.16? |
| 03:08.51 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 03:10.08 | Insanity5 | Does anyone know a good method for SIP call quality issues? Out of order packets/jitter? An easy method to see these things? |
| 03:10.09 | sione | never mind, i just answerd my own quetion on that |
| 03:11.20 | *** part/#asterisk l2cache (n=admin@62.180.8.67.cfl.res.rr.com) |
| 03:12.32 | sione | weird i just found something out with my SIP problem |
| 03:12.35 | emirdamadi | I need to ask a question guys |
| 03:12.42 | emirdamadi | can anyone help me? |
| 03:13.03 | hansin321 | Go ahead and just ask. If someone knows, they will most likely answer. |
| 03:13.04 | Strom_M | is it still "how do I patch asterisk"> |
| 03:13.05 | Strom_M | ? |
| 03:13.05 | JunK-Y | emirdamadi: asking a specific question is always much better |
| 03:13.45 | sione | I find using softphones my looping problem goes away |
| 03:14.43 | *** join/#asterisk jmacz (n=jmacz@190.24.103.194) |
| 03:15.41 | *** join/#asterisk hacim (n=micah@debian/developer/micah) |
| 03:16.50 | hacim | can anyone give me an idea of where I can get a cheap DID in sao paolo, brasil? |
| 03:17.21 | hansin321 | emirdamadi: The reason it is best just to ask the specific question as opposed to asking if anyone can help, is that no one can really know if they can help unless they know what the question is. Therefore it is just better to ask. You can't be insured of a response, but that's just the way IRC works... |
| 03:18.27 | Insanity5 | Do certain codecs deal better with jitter than others? IS there an easy way to at the cost of delivery delay, buffer two second of audio each way on the asterisk box or something and have it resort to correct for jitter? |
| 03:20.55 | mosty | Insanity5, jitterbuffer |
| 03:22.05 | Strom_M | TWO SECONDS?! |
| 03:22.05 | Strom_M | are you completely mad?? |
| 03:22.08 | Strom_M | you can't do two seconds of jitter buffering - no one will be able to converse naturally |
| 03:22.22 | blitzrage | Strom_M: over |
| 03:22.34 | blitzrage | Strom_M: the bird flies at midnight. over. |
| 03:23.03 | blitzrage | damnit.... Jays just lost :( |
| 03:24.11 | wunderkin | the rooster crowed at 12:01, over |
| 03:24.11 | Strom_M | three nine seven zero six |
| 03:24.39 | Insanity5 | Strom_M - Well ok, what is normal on a cell phone? It has to be at least 400 ms at each end. |
| 03:24.59 | Strom_M | maximum total allowable call latency is 400ms |
| 03:25.21 | Strom_M | so if you want to have 400ms of jitterbuffer, that means you have to have 0ms latency between endpoints |
| 03:25.40 | blitzrage | I use 30ms |
| 03:25.48 | blitzrage | might even be 20ms |
| 03:25.52 | Strom_M | now that's sanity |
| 03:32.27 | Insanity5 | Strom_M - What's the latency on a cell-to-cell call? |
| 03:33.27 | *** join/#asterisk dlynes_laptop (n=dlynes@d207-216-161-56.bchsia.telus.net) |
| 03:37.13 | sione | whats the differance from "Packet2Packet Bridging" and "Native bridging" ? |
| 03:38.11 | Strom_M | Insanity5: depends...likely no more than 40-50ms assuming it's within the same metro area |
| 03:38.45 | Insanity5 | Strom_M - Ok, will I always wondered why the data cards have 300ms latency to the first hop, minimum. Cross country calls must be a little nasty then :) |
| 03:38.52 | sione | mysoft phones are showing up as packet2packet bridging and my Linksys PAP2 shows up as Native |
| 03:38.54 | *** join/#asterisk IguanaNed (n=you@CPE000625db3f84-CM00111ae43f1e.cpe.net.cable.rogers.com) |
| 03:39.15 | Insanity5 | Strom_M - WOould updating my ages-old asterisk version help any, or realistically, a perfectly working installation and a crappy internet connection can't be improved much? |
| 03:41.19 | Insanity5 | Or even a way to debug jitter and see how bad it is with tethereal/tcpdump would be nice. I can't find a usable time stamp/ordering determination mechanism, just the time it arrived. |
| 03:41.41 | sione | wireshark can do it |
| 03:42.10 | *** join/#asterisk Plecebo (n=larry@D-128-208-60-80.dhcp4.washington.edu) |
| 03:42.11 | Insanity5 | sione - How so? I got the packet capture... |
| 03:42.31 | sione | Wireshark has a good SIP/RTP analist<sp> |
| 03:42.44 | Insanity5 | Take it and bring it to a win32 machine then? |
| 03:42.50 | Insanity5 | No gui on my server, just tethereal |
| 03:43.04 | sione | ya so you have to bring it to a gui computer |
| 03:43.38 | Insanity5 | Ok, did not know any such tool exists :) |
| 03:43.39 | NOT_guru | anyone mind my asking a trixbox 2.0 question? |
| 03:44.08 | sione | wireshark is ethereal |
| 03:44.31 | sione | erthereal is no longer been develop as the name ethereal they moved to wireshark |
| 03:45.18 | NOT_guru | should I allow kudzu to touch my tdm804p? |
| 03:45.30 | Insanity5 | sione - They still call the binary tethereal, even off the new wireshark compile. |
| 03:45.44 | NOT_guru | should I configure it through kudzu there by creating eth1 |
| 03:45.52 | Insanity5 | I don't know why... ethereal is wireshark, but tehereal is not twireshark yet :) |
| 03:46.45 | sione | heh im so use to using ethereal so im glad they didnt change that name |
| 03:49.59 | FuriousGeorge | anyway, to those people who answered me last time, sorry to repeat myself, but right after i asked the question my roommate pulled me away to look at his failed psu fan |
| 03:50.31 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:50.32 | FuriousGeorge | i have two identical systems. one has two tdm400p the other has only one. They have identical OS and kernel and everything else. the former deadlocks, the latter does not. what are the odds replacing the tdm400p (with one sangoma a200 series) will resolve the problem? |
| 03:50.49 | FuriousGeorge | i have not tried swapping the cards (dont have spare modules/cards) |
| 03:50.59 | FuriousGeorge | im not sharing interrupts |
| 03:51.15 | FuriousGeorge | and i ran memtest about 5 months ago |
| 03:51.22 | FuriousGeorge | i guess i should run it again |
| 03:51.26 | dlynes_laptop | FuriousGeorge: sounds like you might have some kind of a hardware issue, possibly with the cpu |
| 03:51.38 | NOT_guru | or check the caps on the board |
| 03:51.44 | dlynes_laptop | FuriousGeorge: are they identical cpus? |
| 03:51.51 | *** join/#asterisk bmg505 (n=leon@196.209.180.128) |
| 03:51.56 | dlynes_laptop | FuriousGeorge: and are they configured on the bios identically? |
| 03:52.00 | FuriousGeorge | dlynes_laptop: its been my experience that cpus either work or they dont. generally dont cause instability |
| 03:52.15 | dlynes_laptop | FuriousGeorge: that's not what I'm saying |
| 03:52.16 | FuriousGeorge | dlynes_laptop: bios has to be pretty close to identical, i change few defaults |
| 03:52.21 | FuriousGeorge | software raid |
| 03:52.27 | FuriousGeorge | oh i get you now |
| 03:52.28 | NOT_guru | bad power in the board ( bad caps ) could cause it |
| 03:52.37 | FuriousGeorge | NOT_guru: how do i diagnose that? |
| 03:52.46 | NOT_guru | just look at them |
| 03:52.58 | dlynes_laptop | FuriousGeorge: i'm thinking more like hyperthreading vs non-hyperthreading, smp vs non-smp, and what NOT_guru said, too |
| 03:52.59 | FuriousGeorge | what does a bad cap vs a good cap look like? |
| 03:53.01 | NOT_guru | look for bulging ( at the top mainly ) or even worse leaking |
| 03:53.28 | NOT_guru | I always check caps stamped with a + on top |
| 03:53.31 | FuriousGeorge | both boxes have identical mb, identicap processor, non-identical 600 watt psus |
| 03:53.49 | NOT_guru | I would check caps first |
| 03:53.57 | dlynes_laptop | FuriousGeorge: caps == capacitors |
| 03:53.59 | FuriousGeorge | non-identical not-the-cheapest-generic-memory (patriot and ocz ddr400 non-ecc) |
| 03:54.12 | FuriousGeorge | what does a bad cap look like tho? |
| 03:54.12 | NOT_guru | after that I would pull the second tdm from system A and add it to system B and see if system B starts acting up |
| 03:54.22 | dlynes_laptop | FuriousGeorge: he said it'll be leaking or bulging |
| 03:54.23 | FuriousGeorge | oh bulging |
| 03:54.47 | dlynes_laptop | FuriousGeorge: the capacitors he's referring to will be the electrolytic capacitors, not the ceramic capacitors |
| 03:54.50 | NOT_guru | thanks dlynes_laptop I made an assumption |
| 03:54.52 | NOT_guru | my bad |
| 03:55.14 | FuriousGeorge | electrolytic = cylindrical |
| 03:55.31 | dlynes_laptop | FuriousGeorge: correct |
| 03:55.31 | NOT_guru | ussually metal top with a colored wrap on it |
| 03:55.36 | dlynes_laptop | FuriousGeorge: and they'll be marked with a '-' sign on one side with a white stripe usually |
| 03:56.12 | FuriousGeorge | NOT_guru: these cards were in a mb that seemed to have fried on the job (didnt check the caps that time), do you guys think the tdm400p were damadged |
| 03:57.01 | NOT_guru | possible but I don't think likely unless it was an electrical surge that killed it ( lightning ) |
| 03:57.16 | FuriousGeorge | come to think of it i put these in another system and it deadlocked, but at the time (months ago), i assumed it was that the platform there was unreliable (vai platform asus desktop mb 32 bit) |
| 03:57.42 | NOT_guru | yah I would goto the system thats fine |
| 03:57.52 | NOT_guru | and roll through the boards 1 at a time |
| 03:57.58 | NOT_guru | and find the bad apple |
| 03:58.15 | NOT_guru | time consuming yes... thorough yes |
| 03:58.18 | FuriousGeorge | isnt that kinda unethical, i mean i could do it but they arent the same business :) |
| 03:58.26 | NOT_guru | oh |
| 03:58.37 | NOT_guru | my bad sorry thought these were on a shelf systems |
| 03:59.15 | NOT_guru | well I would still pull the boards 1 at a time at the place with a bad one after checking the board |
| 03:59.19 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:59.28 | NOT_guru | see if its just one or the other |
| 03:59.50 | FuriousGeorge | NOT_guru: tbh, im leaning toward selling those and getting a sangoma a200 |
| 04:00.07 | NOT_guru | sure... but only after eliminating the board |
| 04:00.18 | NOT_guru | new mobo cost less |
| 04:00.19 | FuriousGeorge | u mean mb right |
| 04:00.24 | NOT_guru | yes |
| 04:00.41 | NOT_guru | sorry for not having best IRC etiquette |
| 04:00.47 | FuriousGeorge | i guess your right |
| 04:00.51 | NOT_guru | I don't IRC often |
| 04:01.04 | FuriousGeorge | i do, and i still do stuff like that all the time |
| 04:01.14 | NOT_guru | I am only here as I have been having fun with a tdm804p |
| 04:01.18 | FuriousGeorge | NOT_guru: the only thing is that this mb is brand spanking new |
| 04:01.30 | FuriousGeorge | literally installed maybe three weeks |
| 04:01.33 | FuriousGeorge | the old one fried |
| 04:01.36 | NOT_guru | yeah thats a touphie |
| 04:01.45 | NOT_guru | but bad boards do ship from time to time |
| 04:01.53 | NOT_guru | even from the best manufacturers |
| 04:02.00 | NOT_guru | mobos that is |
| 04:02.11 | FuriousGeorge | f* me |
| 04:02.16 | NOT_guru | what motherboard you using? |
| 04:02.24 | NOT_guru | well I will say |
| 04:02.32 | NOT_guru | generally caps don't go that fast |
| 04:02.40 | FuriousGeorge | tyan s2865 series socket 939 nforce with opteron 165 |
| 04:02.41 | NOT_guru | if this system is less than 6 months old |
| 04:02.47 | NOT_guru | i would lean not board |
| 04:02.52 | FuriousGeorge | me too |
| 04:02.53 | NOT_guru | my |
| 04:02.57 | NOT_guru | thats a nice board |
| 04:03.03 | FuriousGeorge | the tdm400 are a couple of years old otoh |
| 04:03.05 | NOT_guru | and tyan is a good company in my book |
| 04:03.18 | FuriousGeorge | i like'em for my servers |
| 04:03.26 | NOT_guru | so lets just guess its the suspect tdms |
| 04:03.37 | IguanaNed | anyone here have experience integrating * with legacy PBX? |
| 04:03.38 | NOT_guru | as you mentioned these were in a system that lost the magic smoike |
| 04:04.04 | FuriousGeorge | i was using an asus a8n as a temp when the original one died (yes, the magic smoke was blue ;) |
| 04:04.14 | NOT_guru | LOL |
| 04:04.38 | *** join/#asterisk k-man (n=jason@unaffiliated/k-man) |
| 04:04.40 | FuriousGeorge | anyway, i put these tdm400p in that mb and it behaved poorly but i blamed the platform at the time |
| 04:04.43 | FuriousGeorge | maybe i should rethink |
| 04:04.44 | NOT_guru | another nice board that an8 |
| 04:04.44 | k-man | hey guys |
| 04:04.50 | FuriousGeorge | IguanaNed: yes and it sucked |
| 04:04.52 | k-man | anyone been to cebit? |
| 04:05.04 | IguanaNed | FG.. thanks |
| 04:05.17 | FuriousGeorge | although looking back on it, i dont know if it was the tdm400p which was my bridge |
| 04:05.39 | NOT_guru | iguana most from what I understand get a t1 card for the lagacy system and the * box |
| 04:05.45 | FuriousGeorge | if ur legacy pbx supports a t1 interface type job it may be a different story |
| 04:06.03 | IguanaNed | my * box is off site |
| 04:06.10 | NOT_guru | heh |
| 04:06.20 | NOT_guru | too much for my |
| 04:06.24 | Insanity5 | Does ICMP being blocked cause any registration problems? For some reason the host pings back when I try to register. |
| 04:06.27 | IguanaNed | I want to be able to press 8 and have the pbx call my DID over PSTN |
| 04:06.45 | NOT_guru | oh |
| 04:07.01 | IguanaNed | similar to how you press 9 for an outside trunk |
| 04:07.17 | FuriousGeorge | IguanaNed: simle dialplan digit strip |
| 04:07.18 | NOT_guru | so press 8 on lagacy and have it dial into your * or other way around? |
| 04:07.29 | FuriousGeorge | ~s/simle/simple |
| 04:07.42 | IguanaNed | press 8 on legacy |
| 04:08.16 | IguanaNed | problem is my comapny7 does not have broadband inet connection |
| 04:08.29 | NOT_guru | then its all on your lagacy at that point from what I can tell as the * box you can have do whatever after the handoff |
| 04:08.32 | IguanaNed | so I have a hosted box with a nice 100 mbit port |
| 04:08.42 | NOT_guru | but you have did's on the * box? |
| 04:08.48 | IguanaNed | yeah I got the * part down pat |
| 04:08.50 | FuriousGeorge | exten => _8X.,1,dial(${TRUNK_OUT}) or exten => _9X.,1,dial(${VOIP_OUT}) or something |
| 04:09.11 | IguanaNed | Furious.. no i need to prgram the legacy PBX |
| 04:09.17 | FuriousGeorge | oh |
| 04:09.18 | NOT_guru | over thinking furious |
| 04:09.19 | FuriousGeorge | :) |
| 04:09.24 | NOT_guru | =) |
| 04:09.32 | NOT_guru | at least your there to help |
| 04:09.43 | FuriousGeorge | trying to work on my karma |
| 04:09.49 | NOT_guru | but yah same thing just on your lagacy system |
| 04:09.51 | FuriousGeorge | deadlocks will make you para-religious |
| 04:09.52 | IguanaNed | I have the 900 page programming manual for the PBX but dont know where to start |
| 04:09.55 | NOT_guru | what is your lagacy system |
| 04:10.06 | IguanaNed | NEC Elektra IPK |
| 04:10.08 | NOT_guru | first place |
| 04:10.20 | FuriousGeorge | IguanaNed: replacing it is out of the question, huh |
| 04:10.27 | IguanaNed | unfortunately |
| 04:10.34 | IguanaNed | ohterwise I would put * in there |
| 04:10.42 | NOT_guru | look for a consultant... I am not trying to be a jerk... but he may be able to do it in 10 minutes and it could take you days to find the answer |
| 04:10.55 | IguanaNed | NOT_gur... I hear ya |
| 04:10.59 | Insanity5 | Every registration request looks like this and the phone won't register. Sometimes it does this. I wait a few hours and it starts working again. Any ideas? IT displays this many times. |
| 04:11.00 | NOT_guru | and 10 should be relatively cheap |
| 04:11.00 | Insanity5 | 284.907180 66.233.151.86 -> 66.225.32.67 SIP Request: REGISTER sip:sip.domain.com |
| 04:11.00 | Insanity5 | 284.907368 66.225.32.67 -> 66.233.151.86 ICMP Destination unreachable (Port unreachable) |
| 04:11.02 | IguanaNed | i know it is prbbly simple |
| 04:11.22 | NOT_guru | I agree we have a merlin at work |
| 04:11.24 | NOT_guru | I hate the ting |
| 04:11.28 | Insanity5 | That is a ethereal capture from the SIP server side |
| 04:11.44 | NOT_guru | I am putting a * box at a remote office as a demo |
| 04:12.05 | NOT_guru | what is your phone system model |
| 04:12.15 | NOT_guru | might be able to find an answer |
| 04:12.27 | IguanaNed | IPK Elektra Eliant IPK |
| 04:13.14 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 04:14.58 | FuriousGeorge | Insanity5: did you change the domain to sip.domain.com |
| 04:15.15 | Insanity5 | yes |
| 04:15.16 | Insanity5 | :) |
| 04:15.20 | NOT_guru | iguanna thats uhm |
| 04:15.32 | NOT_guru | I can't find a single doc on it |
| 04:15.40 | Insanity5 | It does that over and over furious. I can't figure it out. |
| 04:16.25 | FuriousGeorge | is there nat? |
| 04:16.55 | FuriousGeorge | Insanity5: ? |
| 04:17.03 | Insanity5 | yes |
| 04:17.09 | Insanity5 | On client side only. |
| 04:17.26 | FuriousGeorge | shouldnt matter then |
| 04:18.01 | FuriousGeorge | whats the client? |
| 04:18.37 | Insanity5 | hacked vonage box |
| 04:19.07 | Insanity5 | User-Agent: Linksys/PAP2-3.1.3(LS) |
| 04:19.38 | FuriousGeorge | ur sure its registering w/ the right port, right? |
| 04:19.43 | FuriousGeorge | 5060 |
| 04:20.04 | Insanity5 | yes |
| 04:20.13 | Insanity5 | eterheal cap shows that |
| 04:20.28 | FuriousGeorge | have you tried x-lite or something just to be sure? |
| 04:21.59 | Insanity5 | nope, but could |
| 04:22.07 | Insanity5 | It'll work kind of randomly if I just wait :) |
| 04:22.20 | FuriousGeorge | oh, thats always grat |
| 04:22.23 | FuriousGeorge | great* |
| 04:22.39 | FuriousGeorge | got qualify = yes in the sip.conf peer section |
| 04:22.47 | Insanity5 | gah, it's working again |
| 04:23.22 | FuriousGeorge | i would try qualify first then i would try x-lite |
| 04:23.29 | FuriousGeorge | im gonna make some eggs brb |
| 04:24.48 | Insanity5 | Where can you get a reasonably priced DID? |
| 04:25.02 | Insanity5 | 208 area code :) |
| 04:25.13 | Insanity5 | That's the whole state of Idaho... |
| 04:26.19 | *** join/#asterisk b00gz (n=b00gz@d233-124-245.col.wideopenwest.com) |
| 04:27.29 | *** join/#asterisk antlers (n=antlers@ip70-173-89-173.lv.lv.cox.net) |
| 04:32.29 | Insanity5 | What kind of jitter can I make work in a call? This has near zero jitter inbound, but outbound from SIP device to server it's high -- 20-40 ms. |
| 04:32.29 | Insanity5 | http://img501.imageshack.us/img501/8897/jitteryj7.jpg |
| 04:33.35 | Strom_M | you have to jitterbuffer on the receiving end |
| 04:34.26 | Insanity5 | Why is there no jitter on the one leg but there is on the other? I need jitter buffer at server? |
| 04:34.54 | Insanity5 | And even if a jitter buffer was present -- would having the jitter buffer change the tethereal capture at all? Or it would just make that jitter tolerable, even though it would stay. |
| 04:36.24 | NOT_guru | I have just upgraded my zaptel drivers so my system could see the tdm804p... kudzu wants to make it eth1 |
| 04:36.42 | NOT_guru | I am new to digium cards larger than the 100p and suggestions |
| 04:37.03 | NOT_guru | oh I am mainly a BSD guy to so the kudzu thing is new to me |
| 04:38.46 | Insanity5 | I really need a packet guru :(. |
| 04:39.32 | *** join/#asterisk sione (i=sione@208-46-202-201.dia.static.qwest.net) |
| 04:39.52 | sione | how you limit a SIP user to have only one channel |
| 04:40.21 | sione | so they cant do 3 way calling or forward calls |
| 04:43.29 | *** join/#asterisk kuku5 (n=kuku5@c-71-201-219-72.hsd1.il.comcast.net) |
| 04:43.32 | Juggie | http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf |
| 04:43.44 | kuku5 | I'm getting massive errors trying to compile rxfax - any takers ? |
| 04:44.00 | Insanity5 | Strom_M - You still here? Can you offer any advice on my packet delay problem? |
| 04:44.11 | Insanity5 | Strom_M - you said receiving end but I'm not sure what you meant :) |
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| 05:04.26 | *** join/#asterisk cgb1911 (n=cgb1911@kim.netcraft.com.au) |
| 05:05.15 | cgb1911 | Anyone around able to assist with a SIP peer "Maximum retries exceeded on transmission" problem? |
| 05:06.24 | cgb1911 | as in , during an active call that's been operating for minutes, a reinvite occurs from the external peer, Asterisk sends 200 OK, extnernal peer ACK's the OK, asterisk appears to ignore it, retransmitting the 200 OK 6 times before dropping the call? |
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| 05:21.10 | *** part/#asterisk dasuberdavid (n=root@user-69-1-11-117.knology.net) |
| 05:22.36 | kuku5 | If I load up 1.4, can I use the 1.2 configs ? |
| 05:27.32 | CunningPike | kuku5: Sort of - but quite a bit changed, so you're best bet is to read UPGRADE.txt |
| 05:27.43 | CunningPike | s/you're/your/ |
| 05:31.09 | *** part/#asterisk burt75 (n=burt@189.157.128.236) |
| 05:32.31 | *** join/#asterisk Zipper_32 (n=None@d154-5-87-181.bchsia.telus.net) |
| 05:34.07 | Zipper_32 | Has anybody had experience, or could anybody point me in the direction of an asterisk setup with an intercom / paging system which plays music on a continuous basis until interrupted by pages? I currently have a paging system enabled through the soundcard on my asterisk server, but I now need music playing at all times in the mean time. |
| 05:35.10 | pipwerk | hmmm, intresting |
| 05:35.32 | Zipper_32 | Quite interesting. |
| 05:36.48 | Zipper_32 | I suppose I could have it route through some other device which will play the music from any source until the page causes an interrupt and takes over... but I was hoping Asterisk would be capable of doing the trick. |
| 05:36.56 | *** join/#asterisk yidiyuehan (n=yidiyueh@58.185.253.70) |
| 05:37.23 | yidiyuehan | hi, anyone can explain the callgroup= and pickupgroup= ? i am a bit confused with these two lines. |
| 05:37.46 | pipwerk | maybe you could have some script pause or mute the playing music from your dialplan and unmute after? |
| 05:38.55 | yidiyuehan | oh no, these two lines are related to remote call pickup |
| 05:39.24 | yidiyuehan | but i am just not sure what exactly they are meaning |
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| 05:53.10 | irule | hii |
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| 06:47.48 | dejandinic | scan |
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| 07:11.20 | ghenry | where's the best place to get res_config_pgsql.c for * 1.2? Running 1.2.18 here |
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| 07:19.37 | *** join/#asterisk clofilo (n=eSe@87.218.150.72) |
| 07:19.49 | clofilo | hi |
| 07:19.59 | clofilo | i have a wireless sip phone |
| 07:20.26 | clofilo | when i recieved a incoming call i can hear but i cant speak |
| 07:20.36 | clofilo | when i call all works fine |
| 07:21.23 | clofilo | in the asterisk CLI when i recieved incoming call i can only see this: |
| 07:21.32 | clofilo | SIP/2300-00834e30 answered SIP/0000-00820800 |
| 07:21.59 | clofilo | but when i call i can see one more line |
| 07:22.17 | clofilo | Native bridging SIP/2300-00820800 and SIP/0000-00834e30 |
| 07:22.30 | clofilo | anybody have idea? |
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| 08:20.52 | Uatec | What do you call a man covered in leaves? |
| 08:23.18 | sergee | Adam? |
| 08:23.51 | hads | Neville? |
| 08:31.35 | Uatec | Russel |
| 08:32.43 | hads | hehe |
| 08:33.06 | *** join/#asterisk ptblank (n=MURDER1@cpe-76-175-68-176.socal.res.rr.com) |
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| 08:55.15 | Polis_ttt | how do i get timestamp in cli-console |
| 08:55.50 | hads | NoOp |
| 08:57.21 | *** join/#asterisk darkmug (n=Dennis@200.103.148.166) |
| 08:59.17 | ghenry | how can you get * to compile against a different locatin of openssl? |
| 08:59.23 | ghenry | oen in /usr/local/ssl |
| 09:05.42 | DarKnesS_WolF | why asterisk website in the downloads ponts to 1.4.00 beta4 !? |
| 09:05.50 | *** join/#asterisk vics (n=vics@Brylant.iit.pwr.wroc.pl) |
| 09:06.20 | DarKnesS_WolF | and old 1.2.14 |
| 09:14.18 | *** join/#asterisk sevard (i=chuck-th@adsl-71-129-115-242.dsl.irvnca.pacbell.net) |
| 09:20.58 | *** join/#asterisk penguinFunk (n=penguin@87.224.86.46) |
| 09:25.11 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:26.12 | *** join/#asterisk pseudor (n=kvirc@161-118-207-82.ip.ukrtel.net) |
| 09:31.40 | sergee | ghenry: ./configure --help | grep -i ssl |
| 09:31.49 | pseudor | sergee: excuse me, I've got an interesting problem calling from H323 to SIP |
| 09:32.11 | sergee | pseudor: and? :) |
| 09:32.18 | hads | heh |
| 09:32.34 | ghenry | sergee: not on 1.2 |
| 09:32.43 | pseudor | sergee: and want to solve that :) with your help |
| 09:32.51 | ghenry | I hacked the Makefiles in / and res |
| 09:32.55 | ghenry | got it sorted |
| 09:33.00 | sergee | oh yes that's an issue.. |
| 09:33.09 | ghenry | yeah, sucks |
| 09:33.51 | pseudor | ghenry: have you ever called from h323 to sip? |
| 09:34.03 | ghenry | not yet, sorry pseudor |
| 09:34.12 | ghenry | ask away though |
| 09:34.58 | pseudor | ghenry: do you know who did? |
| 09:35.06 | ghenry | just ask |
| 09:35.09 | *** join/#asterisk tutt9876 (n=tut123@cvl92-2-82-228-144-230.fbx.proxad.net) |
| 09:35.11 | ghenry | many people listening |
| 09:35.55 | tutt9876 | hi: got a problem to find the exact syntax to Dial a user connected with a Sip phone in my dialplan |
| 09:36.18 | pseudor | chan_h323.c:977 ooh323_indicate: Don't know how to indicate condition 17 on ooh323c_o_1 |
| 09:36.19 | pseudor | This message is caused by the request of indication of the state 17: |
| 09:36.19 | pseudor | frame.h: |
| 09:36.19 | pseudor | AST_CONTROL_UNHOLD = 17, /*!< Indicate call is left from hold*/ |
| 09:36.19 | pseudor | while calling from SIP to H323. |
| 09:36.19 | pseudor | While calling from H323 to SIP the sound os absent at the H323 side. Instaed I get the message by chan_h323 module: |
| 09:36.22 | pseudor | Don't know how to deal with mode 0x40 (slin). |
| 09:36.24 | pseudor | And after this anothe one by the same module: |
| 09:36.26 | pseudor | Don't know how to indicate condition -1 on ooh323c_2 |
| 09:37.05 | pseudor | that is the problem... |
| 09:37.40 | tutt9876 | Dial(SIP/${EXTEN}) return "no Such Host" |
| 09:38.04 | pseudor | the sound is absent at the side of h323 |
| 09:38.13 | tutt9876 | So the local sipphone is not dialed |
| 09:38.46 | tutt9876 | Must a i put a "localhost" somewhere? |
| 09:39.25 | pseudor | the call is created but there is no sound at the h323. I can hear from the h323 at the sip side but not in the contrary |
| 09:40.08 | *** part/#asterisk uppal (n=uppal@2001:618:400:888c:218:deff:fe9f:a77f) |
| 09:41.11 | JT | tutt9876: Dial(SIP/sip.confentry/number) |
| 09:41.31 | pseudor | JT: what about my problem? |
| 09:41.49 | hads | Pushy |
| 09:42.42 | tutt9876 | JT: what is "sip.confentry" ? |
| 09:43.38 | JT | tutt9876: the name of the relevant entry in sip.conf |
| 09:44.05 | hads | Or just Dial(SIP/801) if 801 is registered. |
| 09:44.25 | JT | depends if he needs to dial a number or not |
| 09:44.58 | tutt9876 | 801 is not registred, and don't really know what to get in sip.conf |
| 09:45.21 | JT | tutt9876: you need to read the book |
| 09:45.22 | JT | ~thebook |
| 09:45.32 | jbot | well, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 09:46.12 | tutt9876 | I have no section in sip.conf |
| 09:46.25 | JT | add one, would be the logical idea |
| 09:46.56 | hads | heh |
| 09:47.05 | tutt9876 | With what data? |
| 09:47.11 | tutt9876 | which |
| 09:47.13 | pseudor | jbot: do you have any idea about my problem? |
| 09:47.17 | jbot | yes, I have any idea about my problem. |
| 09:47.22 | hads | tutt9876: Read the book |
| 09:48.00 | tutt9876 | ok will make some tries, thanks |
| 09:48.05 | *** part/#asterisk tutt9876 (n=tut123@cvl92-2-82-228-144-230.fbx.proxad.net) |
| 09:48.24 | pseudor | JT: need help |
| 09:48.40 | JT | pseudor: and? |
| 09:48.58 | pseudor | JT: the problem with SIP and H323 |
| 09:49.22 | JT | pseudor: i don't like people constantly soliciting me for help in here when i've never spoken to them before |
| 09:49.30 | JT | i don't have the necessary h.323 experience |
| 09:49.42 | JT | most people don't use h.323 with asterisk |
| 09:50.12 | penguinFunk | is h.323 better than sip ? |
| 09:50.16 | penguinFunk | we use sip everywhere |
| 09:50.34 | JT | debatable, but probably not, especially for an end user/pbx perspective |
| 09:51.25 | pseudor | penguinFunk: I just want to provide full compatibility, that is why I check calls from H323 to SIP |
| 09:51.41 | penguinFunk | cant say we have any trouble with sip, if it's not broken don't fix it i guess. |
| 09:51.50 | hads | Quite |
| 09:51.54 | JT | pseudor: asterisk may not be the best choice then |
| 09:52.09 | JT | there are at least 3 different h.323 channel drivers available for asterisk also |
| 09:52.31 | pseudor | JT: I use OOH323 from addons-1.4.1 |
| 09:53.38 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 09:54.22 | *** join/#asterisk clona (n=Atle@82.196.214.14) |
| 09:55.20 | clona | Yo Guys, In extentions conf, you have ${CALLERID(num)} wich is the caller's number.. What is the variable for the Called party ? I cant use ${EXTEN) because it's changed before I get where I need it |
| 09:55.44 | JT | incoming call from what technology? |
| 09:55.47 | clona | sip |
| 09:55.52 | hads | Assign it to something before it changes |
| 09:56.07 | clona | I tried doing |
| 09:56.11 | clona | exten => _X.,1, Answer() |
| 09:56.11 | clona | exten => _X.,2, set(${ruri} = ${EXTEN}) |
| 09:56.11 | clona | exten => _X.,3, Goto(xMenu,xMenu,1) |
| 09:56.18 | clona | but.. in xMenu I cant get out $ruri |
| 09:56.24 | clona | or ${ruri} |
| 09:57.00 | JT | incorrect Set syntax |
| 09:57.06 | clona | oh |
| 09:57.15 | clona | how should it look like ? :$ |
| 09:57.32 | JT | well you're setting the variable, not trying to read the resukt if ut |
| 09:57.52 | hads | show application set |
| 09:57.57 | clona | I try in [xMenu] |
| 09:58.56 | JT | clona: also be consistant |
| 09:59.11 | JT | applications start with capitals normalls |
| 09:59.29 | clona | Hmm okey |
| 09:59.30 | hads | Also watch your whitespace |
| 09:59.33 | clona | <- I'm new to asterisk :$ |
| 09:59.38 | JT | no spaces between priority and application command |
| 09:59.41 | JT | ~thebook |
| 09:59.42 | jbot | thebook is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 10:00.22 | hads | And in the Set command, you |
| 10:00.39 | hads | 'll likely get whitespace in your variable or something |
| 10:01.24 | clona | Hmm.. I now swaped it so there is no whitespaces there |
| 10:01.56 | clona | in xMenu (will change it) I have exten => xMenu,1, agi(test.agi,${ruri}) |
| 10:02.12 | clona | but, I'll try somewhat more:D |
| 10:03.07 | clona | ah.. Now I see what I did wrong :$ |
| 10:03.11 | clona | STUPID me |
| 10:03.28 | clona | set(ruri=${EXTEN}) :S |
| 10:08.25 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
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| 10:12.47 | skirmisha | hello guys |
| 10:14.05 | skirmisha | any ideas if type=friend i can send more than 1 call |
| 10:14.47 | hads | Yes |
| 10:14.53 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 10:15.22 | skirmisha | this is default |
| 10:16.36 | skirmisha | then why i get Everyone is busy/congested at this time |
| 10:16.54 | skirmisha | and when i change type=peer all is working fine |
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| 11:11.22 | Vec | I need to convert a wav (I think a-law) file to a wav49 file using sox, not sure how to ?, must U convert it to gsm and then rename it, that does not sound right ? |
| 11:11.29 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
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| 11:12.43 | penguinFunk | what's wrong with a standard wav file ? |
| 11:13.07 | penguinFunk | type: man sox |
| 11:13.33 | *** join/#asterisk coppice (n=chatzill@94.143.17.210.dyn.pacific.net.hk) |
| 11:14.52 | Vec | penguinFunk : its already a pcm wav file, seems like I must convert it to gsm then rename it to WAV, according to http://lists.digium.com/pipermail/asterisk-users/2007-January/176577.html |
| 11:16.20 | JT | that sounds wrong |
| 11:16.22 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 11:16.25 | JT | it must be a mono wave of 16kHz |
| 11:16.31 | JT | for optimum performance |
| 11:16.32 | DrukenHME | what are you trying to accomplish? |
| 11:17.06 | Zeeek | ladies and gentlemen of earth |
| 11:17.37 | Zeeek | is it possible for asterisk to bow out, i.e., leave a call by connecting two SIP channels? |
| 11:17.40 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 11:17.59 | JT | you can reinvite media, but it still does signalling |
| 11:18.00 | DrukenHME | of course.... |
| 11:19.02 | Zeeek | the setup is, I receive the incoming SIP call, I dial a SIP number and send some DTMF. Then, I want to leave the media stream because I am too far from each end |
| 11:19.11 | Zeeek | I don't see how to do this |
| 11:19.24 | Vec | I am trying to overite a voicemail greeting which is stored in WAV49 format, I have a wave file which was recorded using the record app ? |
| 11:20.14 | Zeeek | JT I don't control the two endpoints so I don't see how to reinvite? |
| 11:20.38 | JT | Zeeek: canreinvite=yes |
| 11:21.47 | Zeeek | added to sip.conf in the two peer entries? |
| 11:22.00 | Zeeek | wouldn't that happen before the DTMF from dial() |
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| 12:06.19 | redax | hi, |
| 12:06.33 | redax | I'm having serious problem with mISDN + asterisk latest. |
| 12:07.32 | redax | getting these lines in asterisk/full: channel.c: Avoiding deadlock for 'mISDN/1-1' |
| 12:07.50 | redax | meanwhile the kernel log floods: mISDN_rdata: rport queue overflow 256/256 [addr:52010201 prim:120282 dinfo:ffffffff] |
| 12:07.59 | DrukenLPY | uhg... finally... anyone want an AP? i'm about ready to throw it out in the street infront of a passing dumptruck |
| 12:08.24 | redax | no incoming calls at all, restarting asterisk fix the problem |
| 12:09.01 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:09.39 | redax | same effect with hfcmulti (HFC4S card) and with 4x HFC-PCI (singleport) card |
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| 12:43.27 | Formater | hi |
| 12:45.56 | nasls_lsa | I have a beronet BNS04 ISDN card .. can I use it with zapata.conf or I have to install misdn ? |
| 12:49.46 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 12:53.20 | Formater | i'm using asterisk's Read command to get PIN number...... when i call the number from one trunk, it works.. when i call it from another provider, I see that asterisk recevies the DTMFs but at the end it says, it is empty :( but before that it says Sending dtmf: 57 (9), at 87.116.148.176 |
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| 13:06.21 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 13:08.05 | naitram_ | anyone know of a sip command line client for windows ce, mobile 2003. |
| 13:09.08 | naitram_ | help |
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| 13:13.18 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:18.08 | shido6 | whats the dtmfmode set to for the other provider? |
| 13:18.25 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
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| 13:23.30 | LeddyHM | YAY! |
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| 13:44.49 | brad_mssw | is asterisk 1.4 stabilizing yet? |
| 13:47.07 | irule | good morning vietnam! |
| 13:47.18 | mosty | brad_mssw, it's still quite buggy |
| 13:47.50 | brad_mssw | if I'm on 1.4.2, 1.4.4 should at least be less buggy, right? |
| 13:48.02 | mosty | maybe, maybe not |
| 13:48.04 | irule | does that mean that a production system should not be 1.4? |
| 13:48.19 | mosty | irule, for now, i wouldn't use 1.4 in production |
| 13:48.36 | brad_mssw | (seem to get random issues where with 1.4.2, we get calls in where the caller can hear us, but we cannot hear them ... didn't start happening until we went from 1.2 -> 1.4) |
| 13:48.37 | xheliox | It really also depends on what you're doing. |
| 13:49.02 | xheliox | brad_mssw: it's certainly not going to hurt you to upgrade to 1.4.4 if you're already on 1.4.2. |
| 13:49.19 | brad_mssw | ... that said, 1.4 seemed to resolve some other issues with sip for us, especially related to quality |
| 13:49.30 | brad_mssw | (over high-latency extensions) |
| 13:49.42 | *** join/#asterisk ToyMan (n=Stuart@72.168.167.241) |
| 13:49.51 | xheliox | I'm using 1.4 on servers that I can stay on top of and where my ass isn't on the line (too much). |
| 13:50.01 | ctaloi | hey all - question; when using a virtual IP on an Asterisk server, I am having trouble getting sip user to register to the ViP. They are able to register with the true IP, just not the virtual. It seems Asterisk is rejecting the SIP invite. I've added all the IP's to the domain listing in sip.conf and in the Asterisk console a "sip show domains" shows both the virtual and the physical IP. Am I missing something? I have Asterisk bound |
| 13:50.35 | ctaloi | note: 1.4.1 |
| 13:51.23 | *** join/#asterisk champster (n=asterisk@AH.tescogroup.com) |
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| 13:54.10 | *** part/#asterisk _Vile (n=vile@bc182112.bendcable.com) |
| 13:55.12 | irule | xheliox is there really a big difference? |
| 13:55.37 | *** join/#asterisk ToyMan (n=Stuart@72.168.167.241) |
| 13:56.58 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 13:57.01 | xheliox | http://ftp.digium.com/pub/asterisk/ChangeLog-1.4.4 - lots of changes made between 1.4.2 and 1.4.4. Though I have a bunch it wouldn't fix your issue.. . |
| 13:57.01 | *** join/#asterisk ToyMan (n=Stuart@72.168.167.241) |
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| 13:59.14 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:59.35 | *** join/#asterisk [Airwolf] (n=airwolf@martijn.lico.nl) |
| 14:04.14 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:04.26 | *** join/#asterisk yenno (i=dfgdfgdg@84-72-188-41.dclient.hispeed.ch) |
| 14:04.40 | yenno | hi, where can i download a ulaw encoded file? |
| 14:04.58 | shido6 | u can make one |
| 14:05.01 | shido6 | with Asterisk |
| 14:05.37 | yenno | ok thanks, and do you know what command? |
| 14:05.42 | shido6 | show application Record |
| 14:05.49 | yenno | thx |
| 14:10.22 | *** join/#asterisk agile (n=mike@63.98.55.146) |
| 14:12.33 | *** join/#asterisk walhala (i=cisco@resonix.fr) |
| 14:12.37 | walhala | hi |
| 14:13.20 | walhala | i have a problem with sccp and 7960 |
| 14:13.42 | walhala | i received digit on my server but nothing else |
| 14:13.52 | shido6 | 1 digit? |
| 14:13.59 | walhala | yes |
| 14:14.03 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 14:14.05 | shido6 | :) |
| 14:14.26 | shido6 | are you able to dial all the digits or does the 7960 complain after 1 digit? |
| 14:14.55 | walhala | i can dial more than 1 digit if i dial and hook on |
| 14:15.38 | walhala | do you want a log sample ? |
| 14:16.10 | walhala | http://pastebin.ca/469009 |
| 14:16.37 | shido6 | what is that from? |
| 14:17.05 | shido6 | what is collecting digits? |
| 14:17.12 | *** join/#asterisk Gpl_Source (n=The_natu@unaffiliated/gplsource) |
| 14:17.37 | walhala | i don't know i just get this with skinny debug |
| 14:18.08 | walhala | here an other example : http://pastebin.ca/469010 |
| 14:18.08 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:18.13 | shido6 | are you forced to stick with skinny? you can put a sip load on the 7960 |
| 14:18.31 | Qwell | except cisco sip sucks |
| 14:18.35 | walhala | i must stay in sccp yes :( |
| 14:18.50 | drako | how i can disable the "transfer" behavior from "flash" or pressing the hang up key on the phone? |
| 14:19.12 | drako | and just hang up the phone call |
| 14:19.18 | shido6 | zapata.conf and features.conf |
| 14:19.27 | walhala | in the last example i dialed 132 and just press the dial key on cisco |
| 14:19.52 | drako | shido6, but i still want transfer with # |
| 14:20.29 | walhala | if i call my cisco from a SIP phone there is no problem |
| 14:21.07 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 14:21.52 | *** join/#asterisk ToyMan (n=Stuart@72.168.167.241) |
| 14:22.11 | walhala | nobody see any solution ? |
| 14:23.44 | shido6 | whats your dialplan look like? |
| 14:24.59 | walhala | shido6: very simple http://pastebin.ca/469017 |
| 14:26.36 | walhala | shido6: nothing difficult isn't it ? :) |
| 14:27.57 | *** join/#asterisk af_ (n=getsmart@81-174-46-10.f5.ngi.it) |
| 14:33.17 | *** join/#asterisk [Airwolf] (n=airwolf@martijn.lico.nl) |
| 14:33.49 | *** part/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 14:35.01 | CrazyTux | If I want an incoming call to ring two devices connected to asterisk at the same time, would that be nesc a que, or just extensions.conf setup to dial twice? |
| 14:36.43 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 14:37.57 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:38.10 | penguinFunk | CrazyTux: just dial twice like this: exten => s,2,Dial(SIP/101&SIP/102&SIP/103&SIP/104&SIP/105&SIP/106&SIP/107,20,tr) |
| 14:41.39 | *** join/#asterisk hwt (n=hwt@curb.thorkildssen.com) |
| 14:41.52 | hwt | hi, which module provides SIPAddHeader()? |
| 14:41.59 | hwt | or where can i find out which module provides what. |
| 14:42.09 | *** join/#asterisk VJFROMGT (n=vjfromgt@pool-72-80-126-195.nycmny.east.verizon.net) |
| 14:42.17 | hwt | is SIPAddHeader() at all available in 1.0? |
| 14:42.26 | VJFROMGT | is there a way to rotate my trunks so they all get used evenly? |
| 14:42.30 | hwt | or is there another way to achieve the same in 1.0? |
| 14:43.32 | Mercestes | <PROTECTED> |
| 14:43.58 | neverblue | ~thebook |
| 14:44.00 | jbot | i guess thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 14:44.35 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 14:44.58 | Mercestes | UJFROMT: Did you use a roll over cable? |
| 14:45.05 | *** join/#asterisk jm|laptop (n=jm@sentry.flags.co.uk) |
| 14:45.50 | Mercestes | UJFROMGT: You have to get the trunk channels started rotating from the SmartJack to yoru PBX and inertia will take care of it from there. |
| 14:46.25 | neverblue | morning |
| 14:46.57 | hwt | apparently chan_sip provides SIPAddHeader. |
| 14:47.01 | hwt | hmphf. |
| 14:47.06 | hwt | i guess my * version is too old, then |
| 14:47.25 | hwt | Mercestes: i can't. the version is heavily patched. |
| 14:48.03 | Mercestes | morning, Neverblue. :) |
| 14:48.10 | Mercestes | hwt: Then patch it some more. |
| 14:48.11 | neverblue | how are you Mercestes ? |
| 14:48.35 | Mercestes | hwt: If it's that heavily patched then you can backport SipAddheader() :D get a dev machine and download the latest source and diff it. |
| 14:48.48 | Mercestes | neverblue: Ok, I guess. Yourslef? |
| 14:48.59 | neverblue | Mercestes, today is a new day :) |
| 14:49.18 | Mercestes | VJFROMGT: Dial(Zap/G1/#######) should accomplish what you wish to do if you setup your groups correctly in zapata.conf. |
| 14:49.27 | Mercestes | It is indeed. |
| 14:50.28 | hwt | Mercestes: i might. |
| 14:50.48 | hwt | Mercestes: but are you saying there were no way to add/del headers in asterisk <1.2-CVS2004something? |
| 14:51.37 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 14:51.37 | Mercestes | hwt: Is that when SipAddHeader() was added? |
| 14:51.40 | ZaVoid | morning all |
| 14:51.45 | Mercestes | morning, Void. |
| 14:52.15 | ZaVoid | anyone ever see somthing where IAX2 call processing just kinda dies with no messages in any logs. and when i do a iax2 show channels it gives a huge list(that appears to grow) |
| 14:52.19 | ZaVoid | so Mercestes |
| 14:52.24 | ZaVoid | er so =sup |
| 14:53.26 | Mercestes | ZaVoid: I haven't. What are you connecting with IAX2? |
| 14:53.33 | *** join/#asterisk unixlike (n=spid3r@31.67.modemcable.oricom.ca) |
| 14:54.09 | ZaVoid | a custom iax2 dialer |
| 14:54.13 | ZaVoid | its worked fine for 10 months |
| 14:54.28 | ZaVoid | then we built a new asterisk box and replaced(same build, new hardware) and now it seems to be crashing |
| 14:54.29 | ZaVoid | its very strange |
| 14:55.41 | Mercestes | Weird. |
| 14:57.49 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 14:58.04 | unixlike | anyone can help me with a softphone issue ? |
| 15:00.02 | ZaVoid | yeah it doesn't really make any sense ya know? |
| 15:00.03 | *** join/#asterisk ToyMan (n=Stuart@72.168.167.241) |
| 15:00.26 | ZaVoid | and i didn't capture any debugs when i confirmed it was all buggered |
| 15:03.05 | unixlike | i am using mitel hardphone and ifedisk softphone both LAN, both same SIP config. mitel phones works great, but i get voice lost with ifedisk only when i dial outside through tdm400p digium cards, not when i dial other softphone in lan... any ideas ? |
| 15:03.28 | *** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com) |
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| 15:03.37 | *** part/#asterisk emiquelito (n=evandro@200-170-201-155.core01.spo.ifx.net.br) |
| 15:07.07 | *** join/#asterisk prevail20 (n=prevail2@wan-hts-26.highertech.net) |
| 15:08.19 | *** join/#asterisk jusMe (n=info@cuscon27948.tstt.net.tt) |
| 15:08.54 | jusMe | Good day to all from the Caribbean! |
| 15:09.31 | agile | bleh |
| 15:09.59 | Strom_M | catsex |
| 15:10.01 | unixlike | good day to u from Quebec, Canada |
| 15:11.09 | jusMe | THANK YOU! |
| 15:11.25 | *** join/#asterisk ChkDigit (n=mrw@static24-72-71-175.regina.accesscomm.ca) |
| 15:12.37 | Mercestes | Good day to you from texas |
| 15:12.46 | Mercestes | The United State of Texas. |
| 15:13.30 | jusMe | Anyone working with asterisknow? |
| 15:14.01 | Mercestes | Try #asterisknow |
| 15:14.44 | prevail20 | hello, I am trying to use the playback app in extensions.conf and when I call I get this error: file.c:512 ast_openstream_full: File /playback/Test.mp3 does not exist in any format |
| 15:14.44 | prevail20 | the file does exist and is located in that directory. Here is my playback line in extensions.conf: exten => s,2,Playback(/playback/Test.mp3) |
| 15:14.44 | prevail20 | Any advice would be helpful |
| 15:14.46 | jusMe | need some help with lighttpd |
| 15:14.52 | jusMe | Already in there! |
| 15:15.06 | *** join/#asterisk Fieldy (i=8dfErHGS@gentoo/contributor/Fieldy) |
| 15:15.37 | Mercestes | prevail20, output of ls -l of /playback/Test.mp3 please? |
| 15:15.40 | *** join/#asterisk hansin321 (n=eric@c-71-196-138-170.hsd1.co.comcast.net) |
| 15:16.14 | *** join/#asterisk explidous (n=explidou@rrcs-24-173-134-222.se.biz.rr.com) |
| 15:16.51 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 15:17.30 | brodiem | prevail20 don't put the file extension on the name. Playback(playback/Test) assuming it was in /var/lib/asterisk/sounds/playback/Test.mp3. I didn't think Playback would support MP3 but I geuss I've never tried it.. |
| 15:17.33 | [TK]D-Fender | prevail20: You cannot specify the file extension in the Playback app... |
| 15:17.47 | carrar | prevail20, put Test.mp3 in /var/lib/asterisk/sounds, then just use Test.mp3? |
| 15:17.48 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 15:18.03 | carrar | ah |
| 15:18.05 | [TK]D-Fender | carrar: nope ;) see above |
| 15:18.10 | explidous | Hi |
| 15:19.35 | wunderkin | you would need format_mp3 |
| 15:19.42 | wunderkin | or use mp3playback |
| 15:21.34 | prevail20 | Mercestes: -rwxrwxrwx 1 root root 6603772 May 3 09:55 Test.mp3 |
| 15:22.05 | Mercestes | prevail20, I think [TK]D-fender got it. remove the .mp3 |
| 15:22.14 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 15:22.40 | jusMe | trying to use php in lighttpd for asterisk now, already intalled PHP but can't get it to work. |
| 15:23.00 | [TK]D-Fender | wunderkin: I'll let him take all the rope he wants..... 1 yard at a time ;) |
| 15:23.20 | prevail20 | [TK]D-Fender: I removed the .mp3 from the extensions.conf file and reload extensions in asterisk but I still get the same error only now it says /playback/Test does not exist in any format |
| 15:23.41 | [TK]D-Fender | jusMe: Not an * question for sure, and this isn't a GUI or lighttpd support channel. |
| 15:24.05 | [TK]D-Fender | prevail20: Ok, now you need to have format_mp3 installed, which is part of the asterisk-addons package |
| 15:24.16 | jusMe | I understand, just trying to do something with asterisknow, and not gething help in #asterisknow!!! no problem, thank you! |
| 15:24.48 | prevail20 | [TK]D-Fender: Addons should be already installed. Where is the format_mp3 located? |
| 15:24.49 | jusMe | if anyone can direct me in the right direction! I apreciate it! |
| 15:25.49 | [TK]D-Fender | prevail20: What do you mean "should"? |
| 15:26.30 | prevail20 | We installed the Asterisk business edition and included the Asterisk-addons |
| 15:27.25 | [TK]D-Fender | prevail20: look at "show modules like format" and see if its there |
| 15:28.29 | prevail20 | [TK]D-Fender: That must be the problem, it is not. I only have format_tiff and format_wav |
| 15:28.35 | *** join/#asterisk NOT_guru (n=NOT_wiza@24-241-103-142.static.stls.mo.charter.com) |
| 15:29.18 | prevail20 | [TK]D-Fender: I will install the addons and verify that the format_mp3 is installed, hopefully that will solve the problem |
| 15:29.29 | prevail20 | [TK]D-Fender: and Marcestes: Thank you |
| 15:29.34 | prevail20 | Mercestes |
| 15:29.57 | *** join/#asterisk saftsack (n=saftsack@pD9E04468.dip.t-dialin.net) |
| 15:29.58 | Mercestes | I've answered to worse. But your welcome. |
| 15:30.23 | *** join/#asterisk queuetue (n=scott@70.54.254.134) |
| 15:31.09 | *** join/#asterisk AnThOnYhO (i=AnThOnYh@218.104.248.250) |
| 15:31.39 | ZaVoid | anyone else play with the new digium tcb400 cards yet? mine don't work and digium cant' figure out why yet :( |
| 15:31.39 | kuku5 | Has anyone lately compile rxfax or valetparking ? |
| 15:31.53 | kuku5 | ZaVoid: the 8 por t? |
| 15:32.10 | NOT_guru | I am in a similar situation I think zavoid |
| 15:32.16 | NOT_guru | I have the tdm804p |
| 15:32.18 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 15:32.21 | NOT_guru | does odd stuff |
| 15:32.27 | generalhan | hey all |
| 15:32.39 | NOT_guru | kuku5 I have the 8 port |
| 15:32.49 | queuetue | I'm trying to connect a vonage d-link adapter into a tdm port (to do outgoing over vonage using zap). When I connect the phone cord, the vonage adapter flashes the phone indicator - as though it's "phone" was offhook - and zap does not work over it. Can anyone explain this? |
| 15:33.05 | dlynes_laptop | [TK]D-Fender: I found out the problem with the BLF light staying on, seems to be an asterisk problem |
| 15:33.18 | Qwell[] | queuetue: what type of port are you connecting it to? |
| 15:33.22 | Qwell[] | It would need to be an fxo |
| 15:33.29 | *** join/#asterisk WeBRainstorm (n=ask@81-174-12-48.f5.ngi.it) |
| 15:33.35 | generalhan | so im still having this ridiculous error that i was posting yesterday:: http://generalhan.pastebin.ca/468038 and im just wondering if i could get all the users off the phone so i can reboot the machine if things could actually get worse !? |
| 15:33.44 | dlynes_laptop | [TK]D-Fender: The blf light stays on, the show hints says the phone's in use, but show channels doesn't show it in use |
| 15:33.55 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-212159.home.otenet.gr) |
| 15:33.59 | queuetue | Qwell: It's a tdm port with fxs kewlstart signalling - the same I use for my POTS lines. |
| 15:34.02 | *** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 15:34.24 | queuetue | (In fact, the same port works as my POTS connection) |
| 15:34.24 | dlynes_laptop | [TK]D-Fender: I'm going to file a bug report on it |
| 15:36.08 | WeBRainstorm | hi guys, I'm having a problem with a queue and ast 1.2.18: some members are not being called |
| 15:36.21 | WeBRainstorm | it only calls a subset of the queue members |
| 15:36.24 | kuku5 | I'm trying to compile rxfax for almost a week now. Tried different spandsp versions, tiff versions, asterisk version. This is the least amount of errors that I got so far: http://pastebin.ca/469115 Thank you. |
| 15:36.41 | WeBRainstorm | even if the corresponding channels seems to be available |
| 15:36.56 | queuetue | Qwell: Any clue why the tdm would appear any different than an extension phone to the vonage adapter? |
| 15:37.06 | *** join/#asterisk rene- (n=rene@200.34.66.137) |
| 15:37.27 | Mercestes | WeBRainstorm, Are you using the priorities? |
| 15:37.39 | WeBRainstorm | Mercestes, no, they are at 0 |
| 15:38.20 | Mercestes | WeBRainstorm, Always the same subset? |
| 15:38.46 | WeBRainstorm | Mercestes: no, it changes from call to call, I'm preparing a pastebin with some output |
| 15:40.24 | *** join/#asterisk phobus (n=phobus@crlspr-69.65.75.232.myacc.net) |
| 15:40.44 | generalhan | Mercestes: do you remember my "Unable to find our posistion" error that you were helping me with yesterday ? |
| 15:41.00 | Qwell[] | Mercestes: sign up for trial |
| 15:42.24 | kuku5 | Qwell: Can you take a look at the errors I'm getting when compiling rxfax ? |
| 15:42.28 | Qwell[] | no |
| 15:42.57 | Mercestes | Qwell[] A trial for what? |
| 15:43.01 | Qwell[] | WoW :p |
| 15:43.06 | Mercestes | Oh! |
| 15:43.07 | Mercestes | lol |
| 15:43.11 | Qwell[] | I sent you one, heh |
| 15:43.17 | Mercestes | generalhan, yes. |
| 15:43.29 | Mercestes | I saw, thanks. |
| 15:43.39 | Mercestes | I'll do it this weekend |
| 15:44.04 | queuetue | A TDM port with fxs signalling is an fxo port, isn't it? |
| 15:44.16 | Qwell[] | queuetue: is it a digium card? |
| 15:44.20 | NOT_guru | queuetue: as I understand it yes |
| 15:44.22 | queuetue | Qwell: Yes. |
| 15:44.27 | Qwell[] | what color is the module? |
| 15:44.53 | queuetue | Qwell: I think red, will have to pop open the server to check. |
| 15:44.53 | generalhan | Mercestes: could restarting the server make this issue any worse ? cause things are getting real bad now, after a call is answered it takes a good 10 -15 seconds for it to actually connect the callers. And i cant afford for this things to just stop working completely |
| 15:44.53 | Qwell[] | red is fxo |
| 15:45.25 | *** join/#asterisk ToyMan (n=Stuart@72.168.167.241) |
| 15:45.27 | NOT_guru | <PROTECTED> |
| 15:45.39 | generalhan | so i would rather have everyone be unhappy with the phones while i work on getting a replacement, rather than shutting it down and not being able to get it back up. |
| 15:46.03 | Qwell[] | generalhan: having problems getting it back up? |
| 15:46.44 | Mercestes | Qwell[] he hasn't tried yet, he's afraid he wont' be able to get it back up |
| 15:46.51 | *** join/#asterisk ToyMan (n=Stuart@72.168.167.241) |
| 15:47.05 | *** join/#asterisk santibiotico (n=santi@ip23498.bcn.altecom.net) |
| 15:47.06 | santibiotico | hi |
| 15:47.07 | queuetue | Yes, red daughter card. |
| 15:47.13 | generalhan | Qwell[]: yeah, what Mercestes said ! |
| 15:47.34 | santibiotico | i'd like to have n-way calling feature under asterisk 1.2.14 |
| 15:47.37 | queuetue | So, if the port is an FXO, then why does the vonage adapter see it as "off hook"? |
| 15:47.41 | NOT_guru | queuetue: yes red daughter card for a digium is an FXO card |
| 15:47.46 | generalhan | and i went home and looked all night for any documentation on this kind of error, but found nothing |
| 15:47.47 | santibiotico | right now i am using the meetme feature |
| 15:47.56 | Mercestes | generalhan, I'd say it'd be relatively safe. |
| 15:47.57 | ZaVoid | not guru your using the card for g.723 transcodings? |
| 15:48.00 | queuetue | NOT_guru: Thanks - I thought it was, but good to be reassured. :) |
| 15:48.01 | NOT_guru | queuetue that I can't answer sorry |
| 15:48.11 | Mercestes | generalhan, but, jbot thinks I'm a nub. |
| 15:48.17 | NOT_guru | queuetue: I help where I can |
| 15:48.24 | generalhan | ~Mercestes |
| 15:48.35 | jbot | somebody said mercestes was almost a total nub |
| 15:48.35 | Mercestes | I think more importantly you need to figure out why that error is happening. |
| 15:48.37 | Mercestes | It could be a Hdd failure. |
| 15:49.00 | generalhan | Mercestes: hmmm |
| 15:49.01 | ZaVoid | NOT_guru: the tcb400b your using? |
| 15:49.02 | santibiotico | i've seen a howto for having n-way calling, but it uses channelredirect |
| 15:49.12 | santibiotico | which is not an available app under 1.2.14 |
| 15:49.15 | santibiotico | any help? |
| 15:49.37 | generalhan | Mercestes: im really torn because i told my boss (the owner) that i was seeing issues with the asterisk server 3 weeks ago. and i told him we need another server so that i can have redundant phone servers, and he didnt listen. |
| 15:49.59 | generalhan | So i dont know if i should care ! lol. other than having 50 pissed off employees at my door. |
| 15:50.19 | queuetue | Does anyone know how to get the IP address from a Dlink VTA-CV? |
| 15:50.29 | Mercestes | generalhan, It may be time for an "I told you so" speech then. |
| 15:50.45 | NOT_guru | sorry phone brb |
| 15:51.16 | [TK]D-Fender | Mercestes: Yeah.... VERY smart...... thats an ECM (Ending Career Move) |
| 15:52.01 | Mercestes | [TK]D-Fender, Otherwise known as a "fire hazard" |
| 15:52.28 | Mercestes | generalhan, I'd check your Hdds. |
| 15:52.44 | Mercestes | generalhan, And isn't the choppy audio in your IVRs? |
| 15:53.44 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 15:54.16 | WeBRainstorm | Mercestes: http://www.pastebin.ca/469137 |
| 15:54.52 | WeBRainstorm | Mercestes: here is queues.conf - sip show peers - show queues - and a log of an incoming call |
| 15:56.27 | generalhan | Mercestes: no its only in the voicemail system |
| 15:56.28 | WeBRainstorm | as you can see only some queue members are called |
| 15:56.41 | Mercestes | Aye. Why are some of these peers offline? |
| 15:56.43 | generalhan | my recorded wavs that i use for the IVR, they all play perfectly |
| 15:58.25 | Mercestes | generalhan, I ran into a similiar problem with another customer. On hold music sounded like *crap* |
| 15:58.25 | lesouvage | I'm working on a server and suddenly the sound quality is like ztdummy isn't properly working any more. Is there a way to check if zaptel and ztdummy is doing what it is supposed to do? |
| 15:58.30 | Mercestes | mp3s. Sounded like it was playing uberfast but it was just skipping audio lik emad and playing every 5th frame or something. |
| 15:58.49 | Mercestes | generalhan, only in voicemail. Gah. Nice. |
| 15:59.09 | Mercestes | WeBRainstorm, Are these all the same phones or are they different types? |
| 15:59.10 | *** join/#asterisk FluxIRCd (i=Prif@cpe-069-132-040-093.carolina.res.rr.com) |
| 15:59.24 | generalhan | its funny though, for her to finish saying you have 10 new messages and 1 old message, takes about 45 seconds ! lol |
| 15:59.42 | FluxIRCd | so is there a windows client to connect to the server or? |
| 15:59.46 | WeBRainstorm | Mercestes: Xlite |
| 16:00.20 | WeBRainstorm | Mercestes: a tethereal shows that no net trafic is sent to peer 327 (as example) |
| 16:00.32 | Mercestes | WeBRainstorm, My problem is that you have random ping times on all these phones, and some of them show an "unknown" status and others show an actual status, and some of them are randomly offline. |
| 16:00.59 | Mercestes | s/problem/concern/ |
| 16:01.04 | Mercestes | ~botsnack |
| 16:01.05 | jbot | Mercestes: :) |
| 16:01.28 | WeBRainstorm | :) |
| 16:01.29 | Mercestes | WeBRainstorm, 327 shows "unknown" as a status, but it *has* taken calls. |
| 16:02.08 | WeBRainstorm | Mercestes: some of them are not logged, infact sip show peers says that, and that's ok |
| 16:02.43 | Mercestes | Yea, but I just checked and every phone that showed an actual status was notified. |
| 16:02.52 | WeBRainstorm | Mercestes: but 327 has a "OK (49 ms)" and in the queue is marked as unknown |
| 16:03.08 | WeBRainstorm | Mercestes: that's incoherent, isnt'it ?? |
| 16:03.15 | Mercestes | not at all. |
| 16:03.21 | Mercestes | unknown means it's state is unknown |
| 16:03.25 | *** join/#asterisk [Airwolf] (n=airwolf@martijn.lico.nl) |
| 16:03.32 | Mercestes | Asterisk doesn't know if it's inuse/available/offline, etc. |
| 16:03.57 | Mercestes | only two unknown phones were noified in yoru queue, so I think that's your symptom. |
| 16:04.06 | Mercestes | Or one of yoru symptoms. |
| 16:04.13 | Mercestes | are all the xlite phones the same version? |
| 16:04.18 | FluxIRCd | poing |
| 16:04.23 | FluxIRCd | s/poing/test |
| 16:04.31 | FluxIRCd | blah |
| 16:04.54 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 16:05.02 | Mercestes | ACK! |
| 16:05.19 | WeBRainstorm | Mercestes: you're right... it should try unknown members |
| 16:05.27 | WeBRainstorm | Mercestes: yes, xlite same version |
| 16:05.36 | *** join/#asterisk Qwell[] (i=qwell@pdpc/sponsor/digium/Qwell) |
| 16:05.36 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 16:05.44 | Mercestes | it either should or should not, it bugs me that it only tries them sometimes/half the time. |
| 16:06.08 | WeBRainstorm | Mercestes: do you think that a queue with SIP and IAX members mixed could cause probles ? |
| 16:06.25 | WeBRainstorm | Mercestes: it should me deterministic... but it isn't.. :( |
| 16:06.44 | Mercestes | Nah, should be ok to mix sip/iax2 |
| 16:06.55 | Mercestes | Are ppl using DND in the queues? |
| 16:07.07 | WeBRainstorm | you mean on the xlite ??? |
| 16:07.10 | Mercestes | yea |
| 16:07.36 | Mercestes | DND + queues = random bad things. |
| 16:07.47 | WeBRainstorm | they say they are not... but shouldn't DND justt causing a BUSY to be sent back to asterisk ?? |
| 16:08.08 | Mercestes | Well, sorta. |
| 16:08.39 | Mercestes | Asterisk checks the phone to see if it's able to sen da call to it and the phone answers that it can recieve a call. So asterisk sends the call, then the phone goes "oops, sorry, DND." and asterisk is all "WTF?" because asterisk already checked if tha tphone was in use |
| 16:09.25 | Mercestes | It's a little different on a call because asterisk tries the call first, then error handles, in a queue I think it error handles first, and then tries the call and gets pissy if there are errors after that. |
| 16:09.34 | Mercestes | but that's a *guess* at this point. |
| 16:09.55 | Mercestes | If they say they're not using DND then I wouldn't blame DND at this point. |
| 16:09.57 | *** join/#asterisk zm23 (n=chatzill@dyn-160-39-251-228.dyn.columbia.edu) |
| 16:10.49 | NOT_guru | ZaVoid: I am back now sorry my sister called me and went on a whine rant |
| 16:11.03 | zm23 | hello all. A quick question about asterisk voicemail app. How can i delete my temp greeting using the phone access to voicemail |
| 16:11.03 | ZaVoid | no worries |
| 16:11.13 | Mercestes | NOT_guru, is she hot? |
| 16:11.13 | NOT_guru | ZaVoid : I am using a Wildcard TDM800P |
| 16:11.14 | [TK]D-Fender | NOT_guru: Its clearly genetic ;) |
| 16:11.20 | ZaVoid | ahh different |
| 16:11.23 | NOT_guru | lol she's 41 now |
| 16:11.26 | ZaVoid | thats not for g723 transcoding right? |
| 16:11.29 | WeBRainstorm | Mercestes: no DND and then i don't see network trafic to that host... |
| 16:11.44 | NOT_guru | now now fender |
| 16:11.53 | Mercestes | ...so asterisk is clearly refusing to send traffic to that device. hrm. |
| 16:11.57 | Mercestes | atleast sometimes. |
| 16:12.02 | NOT_guru | and yes my buddies all still wanna "be with" her |
| 16:12.12 | Mercestes | I'm not interested in a 41 year old. |
| 16:12.25 | NOT_guru | oh and Fender were you the one helping me yesterday with this card? |
| 16:12.47 | NOT_guru | and asked why I configured all 8 ports when its only got 1 daughter card in it |
| 16:12.56 | NOT_guru | well genzaptelconf did that |
| 16:12.59 | WeBRainstorm | Mercestes: yes, it's not considering it, let's get a look at full log |
| 16:13.03 | Mercestes | WeBRainstorm, Well I would pose it to the smart guys and see if they have an answer and then bug report it. |
| 16:13.56 | [TK]D-Fender | zm23: its in the menu in VoicemailMain |
| 16:13.57 | Mercestes | I can't really say whether asterisk should or should not send traffic to an unknown device but I think it should pick one. Could be bad network connections but, I'm giving the network the benefit of the doubt. |
| 16:14.12 | [TK]D-Fender | NOT_guru: Yes, I was |
| 16:14.20 | WeBRainstorm | Mercestes: here i am... May 3 18:52:21 ERROR[27150] chan_sip.c: Unable to build sip pvt data for '327' (Out of memory) |
| 16:14.33 | oej | That's unusual |
| 16:14.40 | oej | small or large system? |
| 16:14.57 | NOT_guru | DFender: well I am building a paste for when tzafrir gets back to me |
| 16:15.05 | NOT_guru | I will link it in a minute |
| 16:15.43 | Mercestes | ew |
| 16:15.46 | Mercestes | googld that. |
| 16:15.50 | Mercestes | s/googld/google/ |
| 16:16.31 | *** join/#asterisk kikoafonso (n=rafonso@cronopio.rits.org.br) |
| 16:16.45 | *** join/#asterisk Zeeek (n=randulo@pdpc/supporter/active/Zeeek) |
| 16:17.08 | WeBRainstorm | Mercestes: and I also have a 'Too many open files' |
| 16:17.15 | Mercestes | oh! |
| 16:17.17 | NOT_guru | Fender here is the aste I have of information for tzafrir http://generalhan.pastebin.ca/469164 |
| 16:17.19 | Mercestes | There is a fix for that. |
| 16:17.25 | Mercestes | google asterisk too many open files |
| 16:17.37 | Mercestes | or linux too many open file |
| 16:18.19 | [TK]D-Fender | NOT_guru: fix your channels in zapata.conf too... |
| 16:18.29 | NOT_guru | well look now |
| 16:18.34 | NOT_guru | err will look now |
| 16:18.36 | NOT_guru | thank you |
| 16:18.58 | Wvirtual | Hi, i have problems when i try to start my asterisk, after install and register the g729 codec. I´m using FreeBSD 6.2 Stable |
| 16:19.00 | [TK]D-Fender | NOT_guru: Its probably still trying to call channels that are no longer defined at all |
| 16:19.01 | Wvirtual | log: WARNING[82799] loader.c: /usr/local/lib/asterisk/modules/codec_g729a.so: Undefined symbol "__errno_location" |
| 16:19.40 | [TK]D-Fender | GTG, back in a few |
| 16:23.04 | WeBRainstorm | Mercestes: I don't think it's a problem of increasing ulimt... the server has 3 active calls but it has 900 UDP open ports... |
| 16:23.04 | queuetue | Has anyone else ever had a problem trying to connect an FXO port to an ATA adapter? This should work, shouldn't it? |
| 16:23.04 | WeBRainstorm | # lsof |grep asterisk |grep UDP |wc -l says 968 |
| 16:23.20 | WeBRainstorm | Mercestes: asterisk 5654 asterisk 1007u IPv4 91811 UDP *:19311 |
| 16:24.16 | explidous | queuetue: you always have to connect FXO to FXS |
| 16:25.04 | queuetue | explidous: yes, i meant the FXS port of an ATA adapter. |
| 16:25.10 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 16:25.13 | explidous | queuetue: However there might be electrical problems depending on the manufacturer |
| 16:25.48 | queuetue | explidous: The FXO is a TDM card from digium, the FSX is a d-link VTA-CV from vonage... |
| 16:25.49 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 16:25.53 | crochat | Hello |
| 16:26.01 | *** join/#asterisk qdk (n=qdk@213.150.62.33) |
| 16:26.09 | crochat | In indications.conf, what is stutter used for ? |
| 16:26.15 | explidous | queuetue: I think I get the idea ;-) |
| 16:26.20 | Mercestes | gtg. BBL, Sorry WeBRainstorm. |
| 16:26.28 | WeBRainstorm | ok... here I am |
| 16:26.29 | WeBRainstorm | http://bugs.digium.com/view.php?id=9235 |
| 16:26.34 | WeBRainstorm | fixed today !!! |
| 16:26.45 | crochat | I mean, in which case you will hear the stutter tone ? |
| 16:26.53 | explidous | queuetue: should work, do you have problems? |
| 16:27.57 | queuetue | explidous: yes, the vonage adapter flashes as though it is off-hook, and asterisk can not call out on it. (console reports connection, but then a click, and then dead air forever.) |
| 16:28.07 | queuetue | No call is actually made. |
| 16:28.52 | explidous | There are a number of parameters for call signaling, however I think vonage does not let you change them... |
| 16:29.21 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 16:29.41 | Zeeek | tomorrow at this time |
| 16:29.48 | explidous | I wonder if one could sniff out the sip connection of their softphone since that is Eyebeam... |
| 16:30.00 | *** join/#asterisk menfin (n=cray@AMontpellier-152-1-57-98.w83-197.abo.wanadoo.fr) |
| 16:32.23 | *** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 16:34.36 | *** part/#asterisk Wvirtual (n=marcos@euro.wvirtual.com.br) |
| 16:36.05 | *** join/#asterisk DrCool (i=DrCool@drcool.hungamacable.com) |
| 16:37.10 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 16:37.59 | WeBRainstorm | Mercestes: tnx for help |
| 16:38.04 | *** join/#asterisk neverblue (n=profx@unaffiliated/neverblue) |
| 16:39.04 | neverblue | what do I need to install to play mp3 files ? |
| 16:40.41 | *** join/#asterisk qdk_ (n=qdk@213.150.62.32) |
| 16:40.51 | Putzz | what do I need to make a sandwich? |
| 16:40.53 | Putzz | heh |
| 16:43.05 | Qwell[] | bread... all others are optional |
| 16:43.58 | anonymouz666 | tzanger: do you use chan_cellphone without problems? |
| 16:44.51 | *** join/#asterisk dzlabing (n=dzlabing@wan-gw.wien.zlabinger.at) |
| 16:51.54 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 16:53.54 | pif | hi, has the '.' atom changed in 1.4.x 's dialplan ? |
| 16:54.14 | pif | is it still an optional character? |
| 16:54.56 | *** part/#asterisk Zeeek (n=randulo@pdpc/supporter/active/Zeeek) |
| 16:57.10 | FuriousGeorge | whats the difference between a wanpipe and a crackpipe? |
| 16:57.46 | FuriousGeorge | seriously, im trying to figure out what this wanpipe that sangoma keeps advertising is |
| 16:58.03 | Corydon-w | FuriousGeorge: you look less retarded when you're smoking crack |
| 16:59.06 | FuriousGeorge | This package installs configuration tools and firmware modules for the Sangoma S508 and S514 router cards. You may use this software to build a stable and flexible WAN router for frame-relay, PPP, or Cisco HDLC leased-line links based on these cards. |
| 16:59.25 | FuriousGeorge | Corydon-w: i'd hit the wanpipe anyway |
| 16:59.57 | FuriousGeorge | i dont understand what a sangoma a200 with a few fxs has to do with my wan |
| 17:00.07 | errr | when I have iax2 debug turned on I see that when an incoming call has: CALLING NUMBER: 231 when I call it from my extension. How do I get this 231 value?? |
| 17:00.25 | *** part/#asterisk pavlicek (n=jpavlice@mail.genevakc.com) |
| 17:01.38 | *** join/#asterisk qdk_ (n=qdk@213.150.62.32) |
| 17:01.57 | errr | basicly I have another server running only voicemail and from the main pbx server you dial 4000 to trunk you to the VM server via iax2 and it goes straight into the main voicemail, but I would like for you to dial 4000 and take you straight to your box so all you enter is your vm password |
| 17:05.53 | lesouvage | should ztdummy be loaded before or after zaptel? |
| 17:07.42 | NOT_guru | lesouvage as I understand it... ztdummy should be loaded after your physical cards |
| 17:08.09 | NOT_guru | I have 1 card and its span1 and ztdummy is span 2.. but thats just my understanding.. I could be wrong |
| 17:08.48 | *** join/#asterisk saftsack (n=saftsack@pD9E04468.dip.t-dialin.net) |
| 17:09.32 | drako | how i can disable the "transfer" behavior from "flash" or pressing the hang up key on the phone? |
| 17:10.18 | lesouvage | NOT_guru: I don't have cards at all, I need it for the timing. |
| 17:11.39 | NOT_guru | lesouvage: lemme look at one of my systems for ref one moment |
| 17:12.43 | NOT_guru | lesouvage I assume we are looking at /etc/zaptel.conf |
| 17:13.51 | queuetue | Since interfacing with the vonage adapter is off the table, can anyone recommend a decent internet-refillable online phone card merchant? :) |
| 17:14.20 | DrCool | queuetue: binfone.com |
| 17:15.46 | queuetue | DrCool: They are a VOIP carrier, aren't they? |
| 17:17.42 | DrCool | yes.. you want to use pstn dialing? |
| 17:17.42 | WeBRainstorm | FuriousGeorge: wanpipe is the unified sangoma driver |
| 17:17.42 | queuetue | DrCool: I think we're done trying to actually make VOIP work... every carrier has just been unusable call quality to date - at signup, or soon after. |
| 17:17.43 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 17:17.43 | queuetue | I think we've got to go pstn from here on out. |
| 17:17.43 | DrCool | queuetue: :) I've been using binfone for the past 3 years and I've hardly (if ever) had probs with quality with them |
| 17:18.03 | *** join/#asterisk shinao1 (n=shinao1@80.89.187.214) |
| 17:18.07 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 17:19.31 | dlynes_laptop | [TK]D-Fender: btw...found out what the issue was with the blf, thanks to oej |
| 17:19.42 | dlynes_laptop | [TK]D-Fender: type=friend is incompatible with blf |
| 17:20.15 | [TK]D-Fender | dlynes_laptop: Apprently peer works BOTH ways now... |
| 17:20.34 | dlynes_laptop | [TK]D-Fender: yeah, that's what oej was telling me |
| 17:24.41 | lesouvage | NOT_guru : Yes that was my guess also. |
| 17:24.45 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 17:26.20 | dlynes_laptop | [TK]D-Fender: that was the whole reason I was using friend, because peer/user was very confusing |
| 17:26.41 | *** join/#asterisk _Turbo_ (n=Turbo@firewall.turbolink.net) |
| 17:26.42 | lesouvage | NOT_guru: the only line uncommented is defaultzone=nl |
| 17:27.04 | _Turbo_ | hey all |
| 17:27.50 | _Turbo_ | does anybody have or know of any example code to join 2 inbound call legs together? |
| 17:28.28 | _Turbo_ | there will only ever be at most 2, so conferencing will have too much overhead, I'd like to keep rtp unmolested if possible |
| 17:30.02 | *** part/#asterisk jpablo (n=jpablo@linuxuanl.org) |
| 17:30.20 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 17:32.17 | NOT_guru | I am sorry I was away hacking abuot with thsi tdm804p |
| 17:32.49 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 17:38.58 | *** join/#asterisk clona (n=Atle@82.196.214.14) |
| 17:39.12 | NOT_guru | tzafrir: you have a bit of time today or still busy? |
| 17:39.25 | clona | Hey, Does anybody know if app_mp4 has it's own "channel" ? |
| 17:39.55 | Qwell[] | why would mp4 be an app? |
| 17:40.19 | clona | Qwell[]: it saves down/plays h26x streams to mp4 |
| 17:40.26 | clona | to/from :-p |
| 17:40.34 | Qwell[] | yeah, why would it be an app? |
| 17:40.40 | clona | I dunno |
| 17:40.43 | clona | some guys made it as a app |
| 17:40.59 | clona | http://sip.fontventa.com/contant/view/15/44 |
| 17:41.05 | clona | there you see :) |
| 17:42.48 | *** join/#asterisk qdk_ (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 17:48.49 | irule | I press 9111 and _91XX dials 111, is 9 treated in a special way? I would like to dial 8111 to have Dial dial 111 |
| 17:49.23 | VJFROMGT | anyone here use polycom phones? |
| 17:49.31 | Mercestes | VJFROMGT, yes |
| 17:49.36 | *** join/#asterisk zm23 (n=chatzill@zaara.cuit.columbia.edu) |
| 17:49.38 | Mercestes | irule, Um, only if you program it that way |
| 17:49.50 | VJFROMGT | is there a way to configure the phone to use 2 different sip providers? |
| 17:49.56 | VJFROMGT | (multi line_) |
| 17:50.59 | redax | good evening, |
| 17:51.09 | VJFROMGT | you still there mercester? |
| 17:51.32 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 17:52.00 | redax | if I dont have any ResetCDR / NoCDR in my dialplan, where, and when CDR record is dumped to csv/mysql or whatver ? |
| 17:52.14 | redax | at hangup? |
| 17:52.55 | Mercestes | VJFROMGT, yes. |
| 17:53.24 | Mercestes | redax, yes. |
| 17:53.59 | VJFROMGT | how do i configure |
| 17:55.31 | codefreeze | redax: usually; |
| 17:55.49 | [TK]D-Fender | irule>I press 9111 and _91XX dials 111, is 9 treated in a special way? I would like to dial 8111 to have Dial dial 111 |
| 17:56.09 | [TK]D-Fender | irule: 9111 dial what you told it to. |
| 17:56.28 | [TK]D-Fender | irule: 9111 could jsut as easily done nothing productive at all. |
| 17:56.38 | hacim | can anyone point me to places to get DiDs for Sao Paolo? |
| 17:57.20 | [TK]D-Fender | VJFROMGT: Pretty mcuh every polycom supports 2 or more registrations. Go read the admin guide to set it up. |
| 17:57.51 | dacter | hacim: doesn't the telco provide the dids? |
| 17:58.36 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:58.40 | redax | aha, so if I want to do anything with the actual CDR, I must put my stuff to the actual context's extension 'h' |
| 17:58.45 | hacim | dacter: well, I guess I need a voip gateway to the telco, because all I have right now is an asterisk server in a colo |
| 17:59.26 | redax | is that correct if I do a `last chance' context with only extension `h' and I include this extension all over the contextes? |
| 18:00.03 | redax | if the context has ext `h' then the the context's `h' executed, if it missing, then the included ext => h,... |
| 18:00.08 | redax | is that correct? |
| 18:00.40 | [TK]D-Fender | redax: Go read up on extensions sort/include priority on the WIKI |
| 18:00.41 | irule | [TK]D-Fender what I am interested in understanding is the treatment of 9 because a regular extesnion is 333 for local sip phones and 333 is dialed, not 33 |
| 18:01.30 | redax | [TK]D-Fender: reading, just confused my mind... |
| 18:01.32 | [TK]D-Fender | irule: "9" is not treated any different. what you dial, and what it does have NOTHING to do with each other. |
| 18:01.42 | redax | sorry for the dump Q. |
| 18:02.07 | [TK]D-Fender | irule: exten => 88,1,Dial(SIP/5000) |
| 18:02.23 | [TK]D-Fender | irule: See? What does 88 have to do with 5000? Nothing. |
| 18:03.33 | [TK]D-Fender | irule: You should be looking at the actual CONTENT of what that extension is doing to see how the pattern matters. it doesn't even HAVE to matter. |
| 18:03.48 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 18:04.18 | [TK]D-Fender | irule: exten => _91XX,1,Dial(SIP/5000) <- Again, 100 possible ways to ring a SINGLE SIP device. |
| 18:08.04 | Mercestes | both 5000 and 88 are divisible by 2, 4, and 8 evenly. |
| 18:08.18 | Mercestes | See, it all makes perfect sense |
| 18:09.08 | [TK]D-Fender | Mercestes: And we all just silently pray that you never multiply ;) |
| 18:11.00 | Mercestes | [TK]D-Fender, nah, only divide. |
| 18:11.32 | [TK]D-Fender | Mercestes: My blades can divide VERY well ;) |
| 18:11.54 | Mercestes | I heard you divide very well. |
| 18:12.05 | [TK]D-Fender | I should take a pic or two of the whole new mounting I've done... |
| 18:12.26 | [TK]D-Fender | </contextabuse> :D |
| 18:12.47 | Mercestes | I don't want pictures of your mountings. |
| 18:14.02 | Mercestes | I am sure they are masterful though. :D |
| 18:15.00 | irule | is it possible to create a group of trunks? |
| 18:18.30 | Mercestes | irule, yes. |
| 18:18.54 | Mercestes | irule, with group- |
| 18:18.57 | Mercestes | irule, with group= even. |
| 18:19.32 | [TK]D-Fender | irule: Your term is dangerously vague. "trunks" is not a word to use around here. |
| 18:19.48 | [TK]D-Fender | irule: What kind of channels (and mix of them) are you referring to? |
| 18:20.25 | irule | heh ok sorry [TK]D-Fender I ment I have a couple zap's, ie 2 x x100p |
| 18:21.00 | irule | what should I search for in voip-info? |
| 18:21.15 | [TK]D-Fender | irule: "group=[number]" for those channels |
| 18:21.40 | [TK]D-Fender | irule: and Zap/g[number]/somethinghere |
| 18:22.49 | *** join/#asterisk lokkju_wrk (n=lokkju@unaffiliated/lokkju) |
| 18:23.27 | PioneerVM | I'm trying to run an AGI script that uses "say_alpha", but it just passed by it like it was never called -- say_number works but say_alpha gives a console error about "iax_read: i should never be called!" |
| 18:23.48 | irule | thanks |
| 18:24.19 | PioneerVM | actually now i dont get that error at all, but the system just passed it and never says anything |
| 18:24.25 | PioneerVM | why would say_number work but not say_alpha? |
| 18:24.37 | PioneerVM | score($AGI->say_alpha('John Smith',1)); |
| 18:24.37 | PioneerVM | score($AGI->say_number('1')); |
| 18:25.15 | *** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com) |
| 18:27.06 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 18:28.10 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-212159.home.otenet.gr) |
| 18:28.22 | PioneerVM | anyone have any thoughts? |
| 18:28.31 | l3jj | Hi everyone, I am looking for a recomendation on a voip provider, I need DID's at a low rate, (like .99 a month, and then a per min charge) I want a provider, that will take a credit card number, and has a clue to what they are doing |
| 18:28.40 | l3jj | Am I dreaming? |
| 18:30.02 | *** join/#asterisk crich1999 (n=crich@port-212-202-210-130.dynamic.qsc.de) |
| 18:30.27 | *** join/#asterisk ToyMan (n=Stuart@12.23.30.130) |
| 18:30.48 | irule | you are referring to group= for my zap channels within zaptel.conf? this is not very informative http://www.voip-info.org/wiki/view/Channels+and+Groups |
| 18:32.43 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 18:33.12 | [TK]D-Fender | irule: On the contrary, that explains it PERFECTLY |
| 18:33.37 | *** join/#asterisk beeew (n=chatzill@66-240-27-5.isp.comcastbusiness.net) |
| 18:34.41 | *** join/#asterisk bkw_ (i=brian@adsl-70-143-36-78.dsl.tul2ok.sbcglobal.net) |
| 18:34.56 | irule | which file pleas? |
| 18:35.19 | *** join/#asterisk saftsack (n=saftsack@pD9E04468.dip.t-dialin.net) |
| 18:37.09 | [TK]D-Fender | irule: What do you mean which file? You define the group in zapata.conf, and you USE it when choosing what channel to dial in extensions.conf |
| 18:37.13 | *** join/#asterisk iEatBabies (n=DudeMan@204.26.87.226) |
| 18:37.13 | dacter | hm. regarding that "channels and groups" page, I think it proves that hackers can't write. Note that the three types of groups are never explicitly listed. |
| 18:38.05 | irule | [TK]D-Fender that is exactly what I needed to know and it is not on that doc, thanks a lot |
| 18:39.05 | [TK]D-Fender | irule: It sort of implied. You are talking about Zap channels, and it says that "group=" is a parameter to set in them. You need to actually stretch those brain muscles of yours a bit more :) |
| 18:39.39 | *** part/#asterisk iEatBabies (n=DudeMan@204.26.87.226) |
| 18:40.05 | [TK]D-Fender | dacter: re-read the entire 2nd section. |
| 18:44.05 | irule | [TK]D-Fender I am, and I will NOT fail my job, I will become an * masta thanks to or despite you guys lol |
| 18:45.16 | [TK]D-Fender | irule: Be like Bush... and just redefine "success" ;) |
| 18:45.47 | irule | lol |
| 18:47.28 | drako | hey can you use variables on sip.conf ? |
| 18:47.41 | drako | so you don't have to repeat the same all the time for each entry |
| 18:48.44 | pipwerk | Set(VARNAME=value) |
| 18:48.57 | [TK]D-Fender | drako: No. |
| 18:49.00 | pipwerk | ow, sip.conf, dri |
| 18:49.22 | etfonhomey | ~wifisip |
| 18:49.27 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
| 18:49.53 | drako | hmm |
| 18:52.01 | drako | is a good idea store sip users on the database rather than on the plain sip.conf file? |
| 18:52.11 | drako | if we have like near 100 extensions ? |
| 18:52.43 | [TK]D-Fender | drako: One more thing to break/support. depending on the number of entries, potentially more or less work. |
| 18:52.44 | pipwerk | I like * realtime |
| 18:53.15 | pipwerk | I know big configs that are entirely generated out of a database |
| 18:53.48 | drako | in my house im using sip.conf since they are only like 17 extensions |
| 18:53.51 | drako | erm |
| 18:53.55 | drako | 10 extensions i mean. |
| 18:54.07 | drako | but in work its growing near 80 so far |
| 18:54.37 | drako | and is getting annoying with the sip.conf |
| 18:54.46 | PioneerVM | go to a script |
| 18:55.13 | PioneerVM | seems like you dont have to worry about the priority # changes too |
| 18:55.14 | drako | PioneerVM, i was thinking about that, but seem for me easier to script it if the config is stored on a database. |
| 18:55.29 | PioneerVM | yea i mean use a script that accesses the dbase |
| 18:55.41 | PioneerVM | store info in dbase and use a agi script to access it and generate config on fly |
| 18:55.47 | drako | thats why im thinking about migrate the sip accounts to a db. |
| 18:55.58 | drako | aight |
| 18:56.00 | PioneerVM | anything over a handful it seems thats the way to go |
| 18:56.03 | drako | is it stable? |
| 18:56.08 | PioneerVM | no idea :) |
| 18:56.17 | PioneerVM | im new to all this but im a programmer and just starting to write a script now |
| 18:56.25 | PioneerVM | actually as i write you im working in another window |
| 18:56.30 | drako | [TK]D-Fender, what did you mean with "One more thing to break/support" |
| 18:56.35 | PioneerVM | i have to do dynamic stuff for potentially thousands of #'s and mailboxes |
| 18:57.03 | drako | PioneerVM, what are you coding? |
| 18:57.10 | PioneerVM | new biz model |
| 18:57.18 | PioneerVM | i will have customers setting up lines dynamicall |
| 18:57.19 | PioneerVM | y |
| 18:57.28 | PioneerVM | and also changing menu dynamically |
| 18:57.35 | PioneerVM | so i need software/dbase type interface |
| 18:57.50 | PioneerVM | plus it seems if you do more than a handful of things its easier in code |
| 18:57.57 | PioneerVM | as you dont have to worry about these priority #'s etc. |
| 18:58.07 | PioneerVM | if you forget or skip a priority # it seems to f' everything up from what ive seen so far |
| 18:58.24 | PioneerVM | and if you want to insert you renumber etc. and have to keep creating contexts, its a pain for larger stuff |
| 18:58.42 | PioneerVM | i installed the perl interface, came with tons of samples |
| 18:58.48 | [TK]D-Fender | drako: Think about it, more effort ot install, configure, SECURE, and integrate a DB rather than a flat file. How often are you going to make changes? How sweeping are those changes ever likely to be? |
| 18:58.49 | PioneerVM | super easy |
| 18:59.00 | wunderkin | use the n priority, that is what it is for |
| 18:59.06 | PioneerVM | yea i saw that |
| 18:59.19 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:59.21 | PioneerVM | but i have dynamic stuff |
| 18:59.27 | PioneerVM | IE: users will be able to change things on a web page |
| 18:59.33 | PioneerVM | each voice mail will work different, etc. |
| 18:59.44 | PioneerVM | (and i love perl) |
| 19:02.20 | irule | I have one sort of confusion, I can group zap1 and zap2, BUT I dial out differently because 1 is a direct line and 2 is through a regular pbx to get to the phone line, so I am using _9XXXXXXX for 1 on the direct line and _99XXXXXXX to get to the phone line through the other PBX |
| 19:03.11 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 19:03.24 | irule | is there a way to GoTo dial _99. if using Zap2 or GoTo _9. if on Zap1? |
| 19:04.52 | [TK]D-Fender | irule: put your channels in different CONTEXTS |
| 19:05.42 | irule | [TK]D-Fender arent I missing on the possibility of doing a round robin search for available channels? |
| 19:06.03 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 19:06.05 | [TK]D-Fender | irule: You are completely mixing yourself up. |
| 19:06.10 | irule | indeed |
| 19:06.21 | Dr-Linux | any perl guy around? |
| 19:06.21 | [TK]D-Fender | irule: Let me make an example for you. |
| 19:06.30 | [TK]D-Fender | Dr-Linux: try asking in #perl |
| 19:06.32 | irule | thanks a lot, I gfreatelly appreciate it |
| 19:06.58 | Dr-Linux | [TK]D-Fender: it's about astman |
| 19:08.01 | *** join/#asterisk chefrs (n=joe@c-24-8-226-145.hsd1.co.comcast.net) |
| 19:08.21 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 19:08.24 | chefrs | Any idea why when someone faxes me it starts to come through, gets a bit and then says "Receive Failed" or when it goes to start it immediately drops to "Receive Failed" ? |
| 19:08.59 | [TK]D-Fender | http://www.pastebin.ca/469426 |
| 19:09.44 | [TK]D-Fender | irule: In there, we define your 2 zap channels. Each one has a differen INBOUND context. the Dial stament I show you in there says that it will dial out the first available line BETWEEN those 2. |
| 19:10.28 | *** join/#asterisk Exhar (n=Roy@213-73-139-87.cable.quicknet.nl) |
| 19:10.38 | [TK]D-Fender | chefrs: * doesn't do faxing, so be precise about what software / tech is involved. |
| 19:10.41 | *** join/#asterisk _DAW (n=chatzill@adsl-157-55-195.msy.bellsouth.net) |
| 19:10.47 | _DAW | Anyone here a polycom guru that can tell me if it is possible to disable the annoying call waiting beep on soundpoint? |
| 19:10.55 | chefrs | [TK]D-Fender: T1 into box. FXS out to an analog fax machine. |
| 19:11.25 | Corydon-w | _DAW: even better, you can CHANGE the call waiting beep |
| 19:11.31 | [TK]D-Fender | chefrs: could be gain/echo, zap timing, etc. |
| 19:11.51 | chefrs | [TK]D-Fender: Hmm. I tested and we have multiple fax machines. #1 can fax to #2 and visa versa perfectly. |
| 19:11.54 | _DAW | I would just like it to go away. |
| 19:11.57 | chefrs | And when they do, it goes out to the PSTN |
| 19:11.59 | [TK]D-Fender | _DAW: set the type to "silent" in sip.cfg |
| 19:12.14 | Corydon-w | <CALLWAITING se.pat.callProg.6.name="call waiting" se.pat.callProg.6.inst.1.type="silence" se.pat.callProg.6.inst.1.value="6"/> |
| 19:12.19 | [TK]D-Fender | chefrs: could be the far side |
| 19:12.31 | _DAW | Thanks.. will give it a try. |
| 19:12.32 | chefrs | [TK]D-Fender: Yeah, try convincing these people of that. |
| 19:12.47 | chefrs | It's happened too many times though for it to be a conincedence, IMHO |
| 19:13.02 | Corydon-w | chefrs: TDM400 or TDM800? |
| 19:13.07 | chefrs | Neither. |
| 19:13.13 | chefrs | Rhino Dual FXS Module |
| 19:13.24 | Corydon-w | Oh, then it's a Rhino issue for them to deal with |
| 19:13.51 | chefrs | So you believe there's nothing I could adjust in * to help? It's just outright Rhino's issue? |
| 19:14.06 | Corydon-w | Correct |
| 19:14.36 | Corydon-w | The T1 card works fine with a good channel bank |
| 19:14.48 | Corydon-w | especially for faxing |
| 19:15.56 | Corydon-w | It's the analog cards that have issues |
| 19:15.57 | chefrs | Hmm. Could it be echo cancellation or such? |
| 19:15.57 | Corydon-w | Highly unlikely, unless your wiring is bad |
| 19:15.58 | *** join/#asterisk red9012 (n=marc3234@206-248-160-251.dsl.teksavvy.com) |
| 19:15.59 | chefrs | That's entirely possible. |
| 19:16.00 | Corydon-w | Wiring between your Asterisk box and your fax machines is bad? |
| 19:16.10 | chefrs | I wouldn't doubt it. |
| 19:16.17 | Corydon-w | That's the only possible cause of echo in your situation |
| 19:16.17 | red9012 | while in privacy I get: press 1 to accept call. press 2.... but keypress sometime dont work. why? |
| 19:18.00 | Corydon-w | chefrs: besides, echocan is automatically turned off when a fax tone is detected |
| 19:18.09 | chefrs | hmm |
| 19:18.14 | chefrs | But it's immediately routed to that zap channel.. |
| 19:18.15 | [TK]D-Fender | chefrs: Grab an analog phone, slap it on the line and see how it feels... |
| 19:18.20 | chefrs | It works fine. |
| 19:18.38 | Corydon-w | Any echo with an analog phone? |
| 19:18.43 | chefrs | Not really |
| 19:18.50 | chefrs | But I was directly at the box, not where the fax machine is |
| 19:18.52 | Corydon-w | Then it's probably the analog card |
| 19:18.59 | Corydon-w | Oh |
| 19:19.17 | Corydon-w | Yeah, you could check that |
| 19:20.03 | chefrs | Maybe I should boost are Tx gain? |
| 19:20.20 | Corydon-w | Grabbing at straws, are we? |
| 19:20.29 | chefrs | All I got are straws to grab |
| 19:20.36 | Corydon-w | Anything to avoid calling Rhino tech support? |
| 19:20.45 | chefrs | Meh, Im reading their forums first. |
| 19:21.05 | Corydon-w | I'd call them and ask for a fix |
| 19:21.26 | chefrs | I may here in a bit |
| 19:21.58 | Corydon-w | In that exact situation, I've used a CAC-1 channel bank with a PRI line going out, and fax works perfectly |
| 19:22.21 | Corydon-w | legacy wiring to the fax, and all |
| 19:22.22 | chefrs | Channel bank isn't an option. |
| 19:24.32 | *** join/#asterisk tuan_modulis (n=chatzill@3-82-252-216-static.enter-net.com) |
| 19:25.13 | tuan_modulis | Hi, I'm a little bit stumped... need to install NVFaxdetect, but i seem to find outdated instructions so far... such as http://www.aussievoip.com/wiki/Asterisk+With+NVFaxDetect |
| 19:25.43 | chefrs | My install came withit |
| 19:26.23 | *** join/#asterisk tuan_modulis (n=chatzill@3-82-252-216-static.enter-net.com) |
| 19:27.27 | *** join/#asterisk mocker (n=mocker@198.247.173.227) |
| 19:29.16 | *** join/#asterisk jmacz (n=jmacz@190.24.97.247) |
| 19:31.06 | tuan_modulis | anyone know if http://www.newmantelecom.com/asterisk/faxdetect/ is down indefinitely? OR is it just down? |
| 19:31.19 | tuan_modulis | that's where I can download the fax detect app |
| 19:32.57 | *** join/#asterisk hfb (n=hfb@pool-72-67-156-130.lsanca.dsl-w.verizon.net) |
| 19:35.05 | *** join/#asterisk saftsack (n=saftsack@pD9E04468.dip.t-dialin.net) |
| 19:35.16 | *** join/#asterisk oej (n=olle@cust-IP-12.data.tre.se) |
| 19:35.35 | etfonhomey | Just sat through a Nortel/Windstream sales pitch and it was pretty laughable. |
| 19:36.41 | tuan_modulis | what did they sell? |
| 19:36.58 | pipwerk | or try to sell? |
| 19:37.04 | tuan_modulis | ya |
| 19:37.19 | Mercestes | pot |
| 19:37.23 | etfonhomey | Well, it turns out our Meridian PBX is still running software from 1996. |
| 19:38.03 | pipwerk | so why not? 'never fuck with a running system' |
| 19:38.06 | tuan_modulis | i say, out with the old |
| 19:38.09 | etfonhomey | So, our CFO signed a contract to update it and my manager thought it might be a good idea to see if VoIP might be a good idea and how it would fit in with the Meridian PBX. |
| 19:38.35 | etfonhomey | I work at university and as you can tell, that's not where I do my * work. |
| 19:39.09 | pipwerk | :) |
| 19:39.15 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 19:39.23 | etfonhomey | 600 phones on campus. |
| 19:39.29 | etfonhomey | Easily doable with *. |
| 19:40.36 | pipwerk | 600? small uni |
| 19:40.46 | etfonhomey | But, they'd rather pay ridiculous $$ for the Meridian "IP" license and $1K per 8 IP phone licenses. |
| 19:40.55 | etfonhomey | Yes, 1200 students + 300 fac/staff |
| 19:41.15 | sevard | Tell them that you bought the Meridian, put in *, pocket the difference. |
| 19:41.19 | Qwell[] | etfonhomey: what school is this? |
| 19:41.21 | pipwerk | my boss is thinking of going all-voip |
| 19:42.04 | etfonhomey | Qwell[], http://www.transy.edu |
| 19:42.19 | Qwell[] | yikes, that's easy to misread |
| 19:42.31 | pipwerk | uhuh |
| 19:42.45 | etfonhomey | Qwell[], which is why we've purchased as many domains close to ours as possible... |
| 19:43.04 | Qwell[] | I'm afraid to check |
| 19:43.26 | etfonhomey | Qwll[], they're committed to the meridian PBX, but are entertaining VoIP solutions, especially for a new building. |
| 19:43.33 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-7be1d10820e13a01) |
| 19:43.50 | *** join/#asterisk zapp-branigan (n=zapp-bra@81.202.214.78.dyn.user.ono.com) |
| 19:44.00 | etfonhomey | * can be easily integrated with a Meridian switch, right? |
| 19:44.08 | tuan_modulis | well... what was that other famous school that changed everything to * with cisco phones |
| 19:44.13 | tuan_modulis | Princeton? |
| 19:44.17 | tuan_modulis | MIT? |
| 19:44.21 | tuan_modulis | i think MIT |
| 19:44.41 | etfonhomey | Cisco is coming next week to talk Call Manager... |
| 19:44.58 | etfonhomey | Cisco + Nortel Meridian switch... |
| 19:45.02 | etfonhomey | :) |
| 19:46.12 | *** join/#asterisk NOT_guru (n=NOT_wiza@24-241-103-142.static.stls.mo.charter.com) |
| 19:46.26 | NOT_guru | ah much better |
| 19:46.28 | pipwerk | hmm, how much would a t1 for the meridian be? |
| 19:46.38 | NOT_guru | dunno what happened to my connection |
| 19:46.53 | NOT_guru | and I officially hate kudzu |
| 19:48.07 | NOT_guru | lesouvage you still around |
| 19:48.30 | etfonhomey | pipwerk, the meridian is already on a t1 or two. |
| 19:49.05 | pipwerk | so, you get one t1 to go into your * and some smart dailplan |
| 19:49.39 | pipwerk | as soon as everyting has walked over to * you tell the management... owww, that huge support bill, just don't pay :) |
| 19:51.23 | etfonhomey | I can't believe how expensive Nortel equipment is, but I bet it's comparable to Cisco. |
| 19:51.49 | pipwerk | hmmm, cisco can be pushed to huge discounts |
| 19:51.53 | etfonhomey | Base model IP phone they quoted $250 + cost of a license. (And they use H323) |
| 19:52.08 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 19:52.30 | pipwerk | for $250 you can get everybody on campus a dect handset |
| 19:52.52 | pipwerk | and have spare change for a pabx :) |
| 19:53.04 | etfonhomey | That model was comparable to the Polycom 330 which retails for $129 or so. |
| 19:53.33 | pipwerk | the linksys phones are cool, so are the snom units |
| 19:53.58 | etfonhomey | Polycom is goooooood. |
| 19:54.32 | sevard | etfonhomey: So $55 dollar 79xx models aren't considered base models (and are SIP)? |
| 19:55.01 | pipwerk | SIP is to easy and open |
| 19:55.06 | etfonhomey | sevard: I have no idea what you're talking about. |
| 19:55.08 | pipwerk | not cisco |
| 19:55.13 | sevard | etfonhomey: http://www.horizondatacom.com/eCart/browse/productinfo.php?manpartnum=CP-7910&flash=0&dram=0&price=55&condition=Refurb&voltage=#overview |
| 19:55.33 | sevard | etfonhomey: base cisco phone, $55, not H323 |
| 19:55.37 | LeddyHM | I relaly like our polycom's |
| 19:55.40 | etfonhomey | refurbished without a license? |
| 19:55.56 | sevard | I believe so |
| 19:56.33 | Qwell[] | 7910 doesn't run SIP... |
| 19:56.38 | etfonhomey | That doesn't really compare to a new Polycom 330 for $129 that plugs in and works. |
| 19:57.38 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 19:57.46 | etfonhomey | I haven't dealt with Cisco. How much is a license for one of their phones to run on their system? |
| 19:57.55 | sevard | a license is what, 7 freaking dollars? |
| 19:58.14 | *** join/#asterisk IPmonger (n=ipmonger@63.139.176.1) |
| 19:58.42 | sevard | Qwell[]: I thought there were SCCP and SIP images for those phones |
| 19:58.43 | Qwell[] | sevard: license? $100+ |
| 19:58.45 | pipwerk | a license is free if they want you bad enough |
| 19:58.49 | Qwell[] | I don't think so - not for the 7910 |
| 19:59.00 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 19:59.37 | etfonhomey | Gotta jet. |
| 19:59.41 | etfonhomey | exit |
| 19:59.48 | sevard | last time I got a license for one phone I swear it was $7.00 |
| 20:00.24 | sevard | and the license didn't give me any valueable tools that I needed |
| 20:00.37 | Qwell[] | license != support contract |
| 20:01.00 | Qwell[] | and there is only skinny firmware for 7910 |
| 20:01.11 | Qwell[] | 7911 can do sip |
| 20:01.58 | ZaVoid | no one should buy the tcb400 card if they have asterisk 1.4.2 apparently |
| 20:02.13 | Qwell[] | ZaVoid: why is that? |
| 20:02.18 | ZaVoid | bec ause it doesn't work |
| 20:02.22 | ZaVoid | and digium for 3 days can't find out why |
| 20:02.36 | Qwell[] | well, have you tried upgrading? |
| 20:02.41 | ZaVoid | whenver aq g.723 call is made it sounds metallicy |
| 20:02.42 | Qwell[] | 1.4.4 is the latest... |
| 20:02.46 | ZaVoid | they have upgarded my kernel |
| 20:02.50 | ZaVoid | upgarted to 1.4.4 |
| 20:02.53 | ZaVoid | downgraded to 1.2.x |
| 20:03.16 | Qwell[] | and did it work on 1.4.4? |
| 20:03.19 | ZaVoid | g.729 works fine.. just not g.723 |
| 20:03.20 | ZaVoid | no |
| 20:03.27 | ZaVoid | if it worked on 1.4.4 i woulda kept it there ) |
| 20:06.12 | ZaVoid | i should just get a real media gw i guess |
| 20:06.30 | sevard | return it and get one from digium? |
| 20:06.54 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 20:07.14 | sevard | oh, heh, i'm mis reading again |
| 20:07.55 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 20:12.16 | *** join/#asterisk neolynx (n=u@165-123-204-62-pool.dsl.fcom.ch) |
| 20:12.30 | neolynx | hello |
| 20:12.42 | neolynx | some swiss people around ? |
| 20:12.46 | neolynx | ich have a weird problem... |
| 20:13.03 | neolynx | i'm afraid swiss DSL providers start to block IAX traffic |
| 20:13.19 | neolynx | had it first with cablecom.ch, now with green.ch |
| 20:13.27 | neolynx | sucks bigtime |
| 20:15.33 | *** join/#asterisk KuJaX (n=kuj@customtrading.dsl.xmission.com) |
| 20:16.17 | neolynx | 305 users and noone says hello |
| 20:16.24 | neolynx | :-) |
| 20:16.30 | KuJaX | hi hi |
| 20:16.36 | Corydon-w | ~ask |
| 20:16.45 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily. |
| 20:16.45 | neolynx | :) |
| 20:16.47 | file | telnet |
| 20:16.47 | Nugget | telnet is eeeeeeevil! |
| 20:17.26 | neolynx | Corydon-w: tell me something I don't know :) |
| 20:18.39 | Strom_M | what Corydon-w was going to say was "---------- --- --------- - - ----------- - ----------" |
| 20:19.42 | neolynx | ok :) |
| 20:19.55 | neolynx | hoping it was somethign about *** and DSL providers |
| 20:20.01 | Corydon-w | There's undoubtedly quite a bit of that, but I don't talk of such things during work hours. |
| 20:20.16 | *** join/#asterisk AndrewGearhart (n=chatzill@h1.39.213.151.ip.alltel.net) |
| 20:20.18 | pipwerk | neolynx: so ask your provider |
| 20:20.19 | Corydon-w | in this channel, anyway |
| 20:20.22 | neolynx | 10pm here... should stop working... |
| 20:20.26 | *** part/#asterisk goozbach (n=goozbach@brooks.netradius.com) |
| 20:20.42 | Corydon-w | 15:20 here |
| 20:21.16 | Mercestes | 15:21 here |
| 20:21.19 | AndrewGearhart | what's the difference between point to point T1 and T1 for internet use? |
| 20:21.21 | *** join/#asterisk Aphelion (n=lk@unaffiliated/lv) |
| 20:21.38 | Corydon-w | AndrewGearhart: nothing |
| 20:22.07 | neolynx | AndrewGearhart: 2mbit link to your provider, or to your buddy |
| 20:22.09 | Corydon-w | the distinction is on what the other side is connected to, not anything about the T1 itself |
| 20:23.09 | *** join/#asterisk pfn (n=pfnguyen@netblock-66-245-252-239.dslextreme.com) |
| 20:23.10 | AndrewGearhart | neolynx: so, essentially, I'm choosing who the T1 will be connected to... the T1 provider... or some other ISP? |
| 20:23.24 | Corydon-w | or not an ISP at all, but simply another customer premise |
| 20:23.32 | neolynx | AndrewGearhart: at least in europe with our E1 it's like that |
| 20:23.54 | AndrewGearhart | neolynx & Corydon-w : that make sense |
| 20:23.54 | neolynx | AndrewGearhart: T1 are 2 copper lines to *somewhere* |
| 20:24.06 | AndrewGearhart | hehe |
| 20:24.09 | neolynx | AndrewGearhart: put some modem on both sides, and do what you want |
| 20:25.04 | neolynx | AndrewGearhart: use a HSDL modem for telefony, and you'll get a G.703 primary rate interface |
| 20:25.28 | AndrewGearhart | thanks. :) |
| 20:25.39 | AndrewGearhart | Just got my quote back on PRI pricing |
| 20:25.55 | neolynx | AndrewGearhart: I can sell you some PCI hardware to use it with the asterisk server I'll sell you afterwards |
| 20:25.56 | neolynx | :-> |
| 20:26.30 | Strom_M | G.703 doesn't actually specify ISDN PRI |
| 20:26.45 | Strom_M | it's merely the T1 or E1 circuit itself |
| 20:27.07 | explidous | neolynx how do you block IAX? you can use any port and TCP you should be able to find a way around that... |
| 20:27.28 | Qwell[] | Strom_M: what's the official time you have to wait between pressing the hookswitch and releasing, before it's considered a "new" call? |
| 20:27.35 | neolynx | I *dont* want to block it, my internet provider is blocking it |
| 20:27.37 | neolynx | ! |
| 20:27.47 | Strom_M | Qwell: I believe it's two seconds |
| 20:27.53 | neolynx | Strom_M: ah, an expert :) |
| 20:27.55 | Qwell[] | http://bugs.digium.com/view.php?id=9660 |
| 20:28.06 | Qwell[] | I'm kinda mehish about that |
| 20:28.26 | Aphelion | i've just been asked to create a rails interface to "an asterisk install using mysql". that is seriously -all- i have to go on atm... if i understand correctly, is the info on asterisk realtime architecture what i need to understand in order to accomplish this? |
| 20:28.28 | neolynx | Strom_M: you don't get G.703 directely out of your copper lines |
| 20:28.30 | Strom_M | Qwell: it should be two seconds IIRC |
| 20:28.30 | AndrewGearhart | Qwell: for POTS... it's 9 seconds |
| 20:28.37 | Qwell[] | AndrewGearhart: eh? |
| 20:28.39 | Strom_M | AndrewGearhart: uh, no |
| 20:29.10 | neolynx | AndrewGearhart: yes, but only if you receive the call |
| 20:29.31 | Strom_M | Qwell: if any services which require the use |
| 20:29.32 | Strom_M | er |
| 20:29.35 | AndrewGearhart | neolynx: ah, sorry... were we talking about a dialed call? |
| 20:29.37 | Qwell[] | hmm, that is true... it could be different on incoming vs outgoing |
| 20:29.59 | Qwell[] | he didn't specify I don't think |
| 20:30.16 | Qwell[] | oh, yes he did.. outgoing |
| 20:30.18 | Strom_M | if all services which need a recall dial tone to operate are turned off, then in theory you can merely bump the hookswitch and get a new dial tone |
| 20:30.38 | Strom_M | otherwise, two seconds |
| 20:30.46 | Qwell[] | this is gonna suck to fix |
| 20:30.57 | Strom_M | por que |
| 20:31.23 | *** join/#asterisk phocus (n=phocus@67.32.20.66) |
| 20:31.58 | phocus | hey guys , do those cheep vonage usb phones at walmart work |
| 20:31.59 | phocus | ? |
| 20:32.03 | Qwell[] | no |
| 20:32.14 | Strom_M | no |
| 20:32.20 | AndrewGearhart | vonage doesn't use sip... does it? |
| 20:32.34 | *** part/#asterisk NOT_guru (n=NOT_wiza@24-241-103-142.static.stls.mo.charter.com) |
| 20:32.57 | AndrewGearhart | well... to rephrase my level of uncertainty... does vonage use sip? |
| 20:33.23 | *** join/#asterisk vykarian (n=stefano@200.212.169.2) |
| 20:33.30 | Qwell[] | sure it does |
| 20:33.56 | AndrewGearhart | am I thinking of Skype then? (that doesn't use sip) |
| 20:34.01 | Corydon-w | Correct |
| 20:34.02 | Strom_M | Qwell's answer could also apply to the question "does vonage blow dead yaks?" |
| 20:34.02 | vykarian | hi all |
| 20:34.10 | vykarian | has someone already seen that? |
| 20:34.11 | vykarian | May 3 17:28:45 ERROR[7246] chan_unicall.c: Unable to open channel 1: Success here = 0, tmp->channel = 0, channel = 1 |
| 20:34.18 | Qwell[] | Strom_M: the "no", or "sure it does"? |
| 20:34.21 | Qwell[] | hopefully the latter |
| 20:34.24 | Strom_M | the latter |
| 20:34.31 | Qwell[] | good |
| 20:35.04 | phocus | so the usb skype phones wont work with asterisk |
| 20:35.08 | Qwell[] | no |
| 20:35.27 | Strom_M | phocus: just shell out the money for a real phone or a real line interface card :) |
| 20:35.39 | explidous | neolynx: sorry for the missunderstanding, it is very difficult blocking IAX, how does your provider block it, did you try to connect via udp or tcp? |
| 20:36.31 | *** join/#asterisk csaba (i=HydraIRC@adsl5-204.ptt.yu) |
| 20:37.47 | explidous | phocus: I suppose they wont, as most of the vonage boxes don't as well... beside of the ones that can be cracked... |
| 20:38.12 | csaba | Hello, I have talked to a person from this channel about a week ago regarding a solution, but I forgot your user name. If you remember me could you message me? |
| 20:38.52 | Sweeper | good cell phones with a SIP client available? |
| 20:39.23 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:39.40 | pipwerk | Sweeper: nokia e60 etc |
| 20:39.58 | KuJaX | Hello. I recently created a temporary vacation message. I cannot get my normal voicemail away message to come up and there doesn't seem to be an option when dialing *98 to use the other message over the vacation message. Any suggestions? |
| 20:40.53 | AndrewGearhart | can somebody help me understand what DID service is? My understanding was that it was phone numbers to go with your service (VoIP ITSP/PRI) ... I'm beginning to think I was wrong |
| 20:41.45 | pipwerk | http://en.wikipedia.org/wiki/Direct_Inward_Dialing |
| 20:41.50 | pipwerk | i guess |
| 20:42.06 | *** join/#asterisk Maxxed (i=foobar@65.59.245.122) |
| 20:42.14 | Maxxed | heya fellas, i gota quickie.. |
| 20:42.23 | Maxxed | i have a asterisk box with a t1 line card |
| 20:42.45 | Maxxed | when a did is dialed, asterisk is only getting the last 4 digits |
| 20:42.55 | Maxxed | like the telco is only sending the last 4 of the did |
| 20:43.03 | Sweeper | pipwerk: mmm |
| 20:43.03 | Qwell[] | Maxxed: yeah, tell them to send all of them |
| 20:43.07 | Sweeper | e60 is massive |
| 20:43.12 | Qwell[] | Maxxed: that's precisely what's happening |
| 20:43.16 | Sweeper | I'm wanting something like the e70 |
| 20:43.17 | Maxxed | and im trying to send, say for example 000-000-1234 to exten 200 for example |
| 20:43.18 | Sweeper | actually |
| 20:43.24 | Sweeper | I just want a freaking e70-2 |
| 20:43.28 | Sweeper | unlocked D: |
| 20:43.31 | Maxxed | qwell, tell them? |
| 20:43.36 | Qwell[] | your telco |
| 20:43.48 | Qwell[] | you usually have a choice of how many digits you receive |
| 20:43.49 | Maxxed | call them and say, hey bitches, send the whole number? |
| 20:43.52 | Qwell[] | yes |
| 20:43.53 | Maxxed | ah! |
| 20:43.54 | Strom_M | well, unless your telephone company is Cox, in which case you'll ask for DNIS remapping and they'll go "What's 'genus'?" |
| 20:44.01 | Qwell[] | tell them you want 10 digits |
| 20:44.07 | Maxxed | cool cool |
| 20:44.11 | Maxxed | well hey thanks :) |
| 20:44.17 | Qwell[] | ~thanks |
| 20:44.17 | jbot | de rien, Qwell[] |
| 20:44.38 | Qwell[] | jbot: no, thanks is <reply> Just send money... |
| 20:44.40 | jbot | okay, Qwell[] |
| 20:44.50 | Strom_M | ~thanks |
| 20:44.50 | jbot | Strom_M: de nada |
| 20:45.05 | Mercestes | ~mercestes |
| 20:45.07 | jbot | you are probably almost a total nub |
| 20:45.12 | Strom_M | ~strom |
| 20:45.15 | jbot | rumour has it, strom is Southern California's only residential ISDN BRI customer |
| 20:45.23 | Qwell[] | nice |
| 20:45.26 | serotonin|work | lol |
| 20:45.32 | Qwell[] | how's that working out, anyhow? |
| 20:45.40 | Strom_M | ive been in England for two weeks |
| 20:45.44 | Strom_M | so, it's still here |
| 20:46.01 | Qwell[] | BRI? In Europe? Are you mad? |
| 20:46.07 | Strom_M | no, i'm in los angeles |
| 20:46.15 | Strom_M | i /was/ in england |
| 20:46.16 | Qwell[] | England, LA? |
| 20:46.17 | Strom_M | got back tuesday |
| 20:46.20 | Qwell[] | oh |
| 20:46.58 | Strom_M | want to see the best of my vacation snaps? |
| 20:47.17 | Mercestes | no, jbot, mercestes is the dark overlord of #asterisk. |
| 20:47.26 | Mercestes | ..>.< |
| 20:47.28 | Qwell[] | you fail |
| 20:47.29 | Maxxed | so when i call the telco, what should i say? |
| 20:47.32 | Mercestes | I fail |
| 20:47.40 | Maxxed | send all the crap? |
| 20:47.45 | Qwell[] | jbot: no, Mercestes is a total nub |
| 20:47.47 | jbot | Qwell[]: okay |
| 20:47.47 | Strom_M | Maxxed: "I want 10 digit DNIS" |
| 20:47.47 | Mercestes | Maxxed, "Screw you AT&T! I went to Vonage!" |
| 20:47.52 | Maxxed | hah |
| 20:47.58 | Maxxed | ok ok, cool cool :) |
| 20:48.12 | Strom_M | and remember, it's pronounced "DEE-nis" |
| 20:48.21 | AndrewGearhart | well folks... I'm outta here... catch ya tomorrow. |
| 20:48.25 | Qwell[] | ~dnis |
| 20:48.26 | jbot | i guess dnis is Dialed Number Identification Service, wherein the telephone company delivers the called number as part of the call setup information. This service is useful when multiple numbers terminate on the same trunkgroup. |
| 20:48.26 | Mercestes | DEE IN EYE ESS |
| 20:48.47 | Qwell[] | I should make a Preferred Number Identification Service |
| 20:48.53 | Strom_M | bahahahhaha |
| 20:48.58 | Mercestes | .... |
| 20:48.59 | Mercestes | lol |
| 20:49.03 | file | Strom_M: vacation snaps? where! |
| 20:49.19 | Strom_M | http://www.stromcarlson.com/photos/london-processed/ |
| 20:49.23 | Strom_M | these are the best fourteen |
| 20:49.34 | Maxxed | oh! heres one |
| 20:49.34 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000e08cb2a29-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 20:49.41 | Maxxed | say for example if the caller |
| 20:49.47 | Maxxed | well, its in san antonio |
| 20:49.57 | Maxxed | and they dont have to dial the area code when calling localy |
| 20:50.03 | Maxxed | *i thik* |
| 20:50.06 | Strom_M | it should still send 210 |
| 20:50.12 | Maxxed | ok cool :) |
| 20:50.12 | Strom_M | 210-555-2368 |
| 20:50.26 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 20:50.31 | Strom_M | if dialing seven digits, the originating switch assumes 210 |
| 20:52.00 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 20:52.34 | syzygyBSD | can someone suggest a fax software to use on 1.2.17? |
| 20:53.59 | [hC] | Qwell: I recently implemented a Basic Analog Line Level Syncronizer on my analog lines, it works great |
| 20:54.31 | Qwell[] | nice |
| 20:54.48 | Strom_M | for digital circuits, you're going to need a Circuit Order Controller Kit |
| 20:54.53 | Qwell[] | is it the Automatic Conductor type? |
| 20:54.56 | explidous | hC: that sounds interesting... |
| 20:56.04 | [hC] | Oh yeah, its fantastic. I even wrote support for it into my Command Line Interface Tools |
| 20:56.34 | Qwell[] | you scared him off |
| 20:56.55 | Strom_M | what about the Automatic Noise Uncertainty Smoother? |
| 20:58.04 | Strom_M | file: do you like my photos? |
| 20:59.59 | file | yessssss |
| 21:00.23 | Strom_M | yay |
| 21:02.22 | *** join/#asterisk alexpe (n=alex@cev75-1-81-57-14-91.fbx.proxad.net) |
| 21:05.07 | Hmmhesays | heh |
| 21:05.09 | Hmmhesays | theres file |
| 21:05.50 | Hmmhesays | haha |
| 21:05.52 | Hmmhesays | how you |
| 21:06.11 | file | do you mean... how ARE you? |
| 21:06.29 | Hmmhesays | what are you the freaking grammar police? |
| 21:06.29 | *** join/#asterisk he11e (n=h@p549dbf51.dip0.t-ipconnect.de) |
| 21:06.47 | file | I will neither confirm or deny that |
| 21:07.45 | Hmmhesays | heh |
| 21:07.58 | Hmmhesays | i actually bought a netgear wireless router |
| 21:07.59 | Hmmhesays | crazy |
| 21:08.31 | he11e | hi all. on my ISDN TE line (mISDN) i get such "errors" -> P[ 2] * RELEASING CHANNEL pid:3 ctx:from-pstn dad:222}�� oad:43100812p�912. the extensions are totaly broken. can someone give me an hint what i can do to solve this? |
| 21:09.46 | shido6 | ... |
| 21:09.53 | shido6 | ... (..lint) |
| 21:11.34 | *** join/#asterisk Gpl_Source (n=The_natu@unaffiliated/gplsource) |
| 21:13.21 | *** join/#asterisk digus (n=digus@206.222.110.30) |
| 21:17.22 | shido6 | what kind of headset does bill buchanan wear in 24? |
| 21:17.52 | Strom_M | do you really want to take telephone advice from a TV show that uses an old AT&T Merlin ringer on a Cisco telephone? |
| 21:17.54 | *** join/#asterisk toot (n=toot@84.19.255.123) |
| 21:18.12 | Qwell[] | yes, yes I do |
| 21:18.21 | Mercestes | lol |
| 21:23.32 | shido6 | :) |
| 21:24.26 | shido6 | kinda looks like the nokia BH-900 but with a thinner boom and its almost invisible |
| 21:24.41 | Qwell[] | it's a prop |
| 21:24.53 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:24.55 | shido6 | dont ruin my dreams |
| 21:24.57 | Strom_M | no! it's TV! therefore it MUSt BE REAL |
| 21:25.18 | *** join/#asterisk dasenjo_ (n=dasenjo@190.24.176.78) |
| 21:25.23 | Qwell[] | if it were real, it would have a big logo |
| 21:25.55 | mog | heh |
| 21:25.58 | shido6 | looks the jawbone is going to win |
| 21:26.01 | mog | i liked the giant cisco add in the last one |
| 21:26.02 | vykarian | can somebody help me with unicall?? |
| 21:26.07 | shido6 | eww |
| 21:26.09 | shido6 | unicall |
| 21:26.14 | shido6 | chan unicall |
| 21:26.19 | shido6 | e1? |
| 21:26.32 | vykarian | r2 |
| 21:26.36 | shido6 | good lord |
| 21:26.45 | shido6 | mexico? or middle east? where ar? |
| 21:26.52 | shido6 | where at? |
| 21:26.55 | vykarian | brazil |
| 21:27.12 | shido6 | whats the prob? |
| 21:27.30 | vykarian | chan_unicall.c: Unable to open channel 1: Success |
| 21:27.36 | vykarian | here = 0, tmp->channel = 0, channel = 1 |
| 21:27.39 | vykarian | weird error msg |
| 21:27.58 | vykarian | thats when start asterisk, right after, it exits |
| 21:28.09 | vykarian | May 3 18:23:35 ERROR[11777] chan_unicall.c: Unable to register channel '1-15' |
| 21:28.55 | vykarian | already recompiled the zaptel driver and unicall patches =\ |
| 21:28.57 | vykarian | nothing |
| 21:29.30 | vykarian | searched at unicall.c source for that error message and nothing |
| 21:33.14 | *** join/#asterisk Cinen (n=Cinen@208.70.20.33) |
| 21:34.12 | shido6 | ok...... |
| 21:36.53 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-133936.home.otenet.gr) |
| 21:37.55 | *** join/#asterisk NOT_guru (n=chatzill@209.145.181.55) |
| 21:43.13 | *** join/#asterisk KuJaX (n=kuj@customtrading.dsl.xmission.com) |
| 21:43.27 | KuJaX | Where are the pre-recorded voicemail box messages stored? |
| 21:44.01 | shido6 | :) |
| 21:44.53 | shido6 | <PROTECTED> |
| 21:45.12 | shido6 | looking to change the sounds? |
| 21:45.26 | KuJaX | yes, right now i have a vacation sound and can't get my normal "away" message to prompt people. |
| 21:45.28 | *** join/#asterisk umay (n=chris@71-208-167-161.hlrn.qwest.net) |
| 21:49.11 | *** join/#asterisk erousse (n=chatzill@207.253.203.34) |
| 21:51.24 | erousse | hello, anyone here with some experience with VoiceGenie and Asterisk ? |
| 21:51.52 | Corydon-w | Never heard of it |
| 21:53.36 | erousse | It's basically a IVR system, but it also support SIP. But I'm having so much problems trying to integrate it with Asterisk... |
| 21:55.50 | Corydon-w | And you're doing this, because Asterisk doesn't have an IVR? |
| 21:56.06 | shido6 | so you're the guy from QC |
| 21:56.11 | shido6 | boucherville ? |
| 21:57.11 | shido6 | why are y ou scewing with the supheaders? |
| 21:57.16 | shido6 | sipheaders |
| 21:57.21 | erousse | I'm doing this because we bought licenses for VoiceGenie... before even thinking about the IVR within Asterisk... |
| 21:57.21 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 21:57.21 | *** mode/#asterisk [+o angler] by ChanServ |
| 21:57.22 | Sweeper | Corydon-w: and because Asterisk doesnt support SIP |
| 21:57.41 | erousse | shido6, yep!: P you must be the co-woker of Francis ? heh |
| 21:57.56 | shido6 | no :) |
| 21:58.06 | erousse | hahah |
| 21:58.23 | shido6 | incoming call from your lady |
| 21:58.39 | *** join/#asterisk Mavvie (n=edwin@ppp175-226.lns4.syd6.internode.on.net) |
| 21:58.53 | shido6 | Greg from NuFone |
| 21:59.38 | shido6 | the music on hold is putting me to sleep |
| 22:00.19 | *** join/#asterisk pfn_cIc (n=pfnguyen@64.235.249.50) |
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| 22:17.41 | angler | grrr..... |
| 22:22.38 | *** join/#asterisk Gregabyte (i=wintermu@nat/digium/x-cf37045d62d442f0) |
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| 22:25.22 | *** join/#asterisk NOT_guru (n=chatzill@209.145.181.55) |
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| 22:41.54 | *** join/#asterisk kieranmullen2 (n=kieranmu@71.245.97.59) |
| 22:43.20 | kieranmullen2 | hello all, I was wondering if I could setup an ivr as an extension that can be called via a sip uri form the outside? if so how? |
| 22:48.41 | NOT_guru | Gregabyte thanks again |
| 22:49.43 | Gregabyte | no problem |
| 22:50.08 | kieranmullen2 | what did you have done?> |
| 22:50.36 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 22:50.44 | NOT_guru | got some sense smacked into me |
| 22:51.23 | NOT_guru | nah greg was awsome |
| 22:51.43 | NOT_guru | I was having troubles with my brand new tdm804p |
| 22:51.50 | NOT_guru | and greg aced it |
| 22:52.13 | NOT_guru | nothing better than strong support people |
| 22:52.15 | file | what was the issue? |
| 22:52.36 | Corydon-w | Greg's a very nice guy |
| 22:52.37 | NOT_guru | heh 1st please don't spit on me.... I use trixbox 2.0 |
| 22:52.59 | NOT_guru | but I had already updated the zap drivers |
| 22:53.11 | NOT_guru | to 1.2.17.1 |
| 22:53.12 | Corydon-w | angler, otoh... ;-) |
| 22:53.21 | NOT_guru | and I just had some conf issues and |
| 22:53.25 | file | ah |
| 22:53.25 | NOT_guru | you know |
| 22:53.36 | kieranmullen2 | did you end up dumping trixbox? |
| 22:53.38 | NOT_guru | trixbox tricks that needed tricking |
| 22:53.46 | NOT_guru | oh I plan on it |
| 22:53.48 | kieranmullen2 | I am using freepbx & asterisk... |
| 22:53.54 | kieranmullen2 | no trix |
| 22:53.56 | NOT_guru | but I am not much in the linux world |
| 22:53.58 | NOT_guru | I am a BSD guy |
| 22:54.07 | NOT_guru | and the zaptel drivers for bsd are lagging |
| 22:54.24 | kieranmullen2 | centos |
| 22:54.27 | NOT_guru | so I will eaither learn more about this "linux" thing |
| 22:54.39 | NOT_guru | yes centos as thats what I am exposed to now |
| 22:54.56 | kieranmullen2 | are you the computer guy for your office or something? (reason why you got that line card) |
| 22:55.03 | NOT_guru | actually ... in all honesty I have another centos box I run VMserver on |
| 22:55.10 | NOT_guru | yes I am |
| 22:55.11 | angler | centos still has the spin lock issue i think |
| 22:55.23 | NOT_guru | I patched that as well prior to the call |
| 22:55.28 | NOT_guru | but thankyou for the heads up |
| 22:55.44 | kieranmullen2 | Get spare box and put cent freepbx or trix on it or *now |
| 22:55.50 | JT | kieranmullen2: trixbox uses freepbx |
| 22:55.56 | JT | kieranmullen2: don't recommend that in here :P |
| 22:56.00 | JT | ~trixbox |
| 22:56.02 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org |
| 22:56.05 | NOT_guru | correct |
| 22:56.14 | NOT_guru | is freepbx taboo here? |
| 22:56.25 | JT | we don't support it |
| 22:56.29 | NOT_guru | oh I know that |
| 22:56.37 | hardwire | JT: liar |
| 22:56.41 | NOT_guru | I just also understand trixbox is frowned upon |
| 22:56.43 | kieranmullen2 | join the #freepbx irc channel actually I am on both :-) |
| 22:56.48 | JT | hardwire: ? |
| 22:56.58 | hardwire | JT: you know what I'm talking about, I have pictures. |
| 22:57.12 | kieranmullen2 | for home use those little 1 line analog line cards are $15 on ebay :-) |
| 22:57.20 | JT | hardwire: err ok, you must be talking about someone else |
| 22:57.56 | hardwire | JT: you just keep thinking that |
| 22:58.08 | JT | hardwire: ... |
| 22:58.14 | NOT_guru | mind if I pop a link in here? I want to know if this howto for a real asterisk box on centos is.... acceptible to you all |
| 22:58.16 | kieranmullen2 | in asterisk how could I setup an extension to go straight to ivr? I have the ivr setup already... This would be for sip direct uri calling ext@somdina.com] |
| 22:58.18 | hardwire | JT: ... |
| 22:58.28 | JT | hardwire: dude, what's up? seriously |
| 22:58.39 | kieranmullen2 | appreciate any information |
| 22:58.56 | hardwire | JT: absolutely nothing, just being a punk at random. |
| 22:59.02 | JT | kieranmullen2: those $15 cards are really poor quality |
| 22:59.03 | hardwire | dude, you've been punked. |
| 22:59.18 | JT | the funny thing is, no-one's laughing |
| 22:59.20 | angler | kieranmullen2, exten => 100,1,Goto(mainmenu,s,1) ? |
| 22:59.21 | kieranmullen2 | jt - yes hence the words "play" and "home" use :-) |
| 22:59.24 | JT | live in your deluded mtv world ;) |
| 22:59.40 | hardwire | JT: I'm not laughing either.. |
| 22:59.53 | JT | kieranmullen2: my idea of "play and home use" is T1 channel bank + polycom sip phones |
| 23:00.02 | hardwire | it actually kind of sucked, but since you kept responding I had to go on and on. |
| 23:00.11 | hardwire | yadda yadda yadda.. here we are. |
| 23:00.14 | kieranmullen2 | you have a t1 channel bank in your home? |
| 23:00.18 | JT | kieranmullen2: yes |
| 23:00.32 | kieranmullen2 | running a business out of it I would guess and it is not for play |
| 23:00.39 | JT | it is for play |
| 23:00.50 | JT | i avoid consumer rubbish IT/telephony gear when i can |
| 23:00.54 | JT | because it's trash :) |
| 23:00.58 | kieranmullen2 | is sagoma a naughty word? |
| 23:01.06 | JT | maybe you mean sangoma |
| 23:01.14 | kieranmullen2 | what is the deal the prices area about the same for normal ones |
| 23:01.25 | kieranmullen2 | normal= digi |
| 23:01.50 | JT | yes they're competing :P |
| 23:02.15 | kieranmullen2 | = no clear advantage to me |
| 23:02.22 | *** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com) |
| 23:03.02 | *** part/#asterisk Inode (n=Inode@modemcable114.59-70-69.static.videotron.ca) |
| 23:03.55 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:06.36 | *** join/#asterisk etfonhomey (n=etfonhom@74-140-213-69.dhcp.insightbb.com) |
| 23:07.22 | *** join/#asterisk votre (n=jordan@69-178-156-154.static-ip.telepacific.net) |
| 23:08.02 | votre | i have recently started working with a small business and am trying to convince our head of IT to switch to an open source telephony server (I am not very familiar with the technology myself) and am wondering if Asterisk is the right product to replace our current product (altigen) with |
| 23:08.25 | votre | any thoughts? |
| 23:08.51 | JT | what do you need it to do? |
| 23:09.50 | kuku5 | Can I park calls onto different lots ? |
| 23:10.30 | votre | well we have a T1 line (half for internet and half for phone lines). We will need it to be able to provide voicemail, pass through caller ID from external calls, be able to direct external calls to the right line either by direct lines and/or extensions, and provide auto attendants |
| 23:10.41 | votre | i know for sure those are our most used and needed functions |
| 23:11.23 | JT | votre: yes they can be done |
| 23:14.34 | votre | fantastic |
| 23:15.05 | votre | is there a prebuilt server I can purchase anywhere with all the compatible hardware or what is the most convenient way to go about that? |
| 23:16.20 | kieranmullen2 | you could put together a decent white box server, put an os on it and hire someone to telnet in and configure it for you |
| 23:17.04 | votre | well i am pretty linux savvy, i am more just concerned about hardware compatibility when it comes to telephony as I am not familiar with the technology much |
| 23:17.25 | kieranmullen2 | buy digium cards they like asterisk :-) |
| 23:17.35 | NOT_guru | pretty much |
| 23:17.39 | NOT_guru | and great support |
| 23:17.55 | votre | haha okay great, I will look into that. Thank you so much guys for your help! |
| 23:18.07 | kieranmullen2 | sip phones? should be cross compatible... people like the grandstream product line, polycomms |
| 23:18.09 | NOT_guru | seriously... I was having problems with my new tdm804p ( I am NOT linux savy ) |
| 23:18.13 | kieranmullen2 | I know nothing |
| 23:18.39 | NOT_guru | called support they asked if they could get into the box, of course I said sure |
| 23:18.49 | NOT_guru | he had it going in about 10 minutes |
| 23:18.51 | votre | well all I know is we are currently using h.323 (i think thats what it is?) and our IT is very interested in switching to SIP |
| 23:18.53 | NOT_guru | MAYBE 15 |
| 23:19.36 | NOT_guru | I like they cisco phones we had from buying a company |
| 23:19.43 | NOT_guru | but the are pricey |
| 23:19.45 | JT | kieranmullen2: dude |
| 23:19.49 | JT | DaveCanoe: are you on crack? |
| 23:19.52 | JT | err |
| 23:19.54 | JT | kieranmullen2: i meant |
| 23:20.03 | JT | no-one here recommends granstream |
| 23:20.08 | JT | let alone likes them |
| 23:20.10 | JT | :P |
| 23:20.24 | kieranmullen2 | sorry I like them.. they work for me and I put a disclaimer in there already |
| 23:20.31 | NOT_guru | I wanted to try a grandstream 2000 just had these cisco's already |
| 23:20.49 | kieranmullen2 | I know nothing |
| 23:20.50 | NOT_guru | now I am leaning aastra to play with |
| 23:21.04 | JT | polycom and aastra are good |
| 23:21.06 | JT | ~phones |
| 23:21.08 | jbot | somebody said phones was http://bani.anime.net/phones/. SIP Hardphones in order of quality/suggestability : Polycom (any), Aastra 480i, Cisco 7940(+), Linksys SPA-9XX, Snom |
| 23:21.26 | NOT_guru | I have 7940's and 7960's |
| 23:21.31 | NOT_guru | good tough phones |
| 23:21.51 | NOT_guru | as long as you don't bash the screen |
| 23:22.01 | NOT_guru | but thats for any phone I guess |
| 23:22.35 | JT | bad company though |
| 23:22.44 | NOT_guru | who cisco? |
| 23:22.51 | shido6 | crisco |
| 23:22.56 | JT | cisco are shocking with support if you don't spend zillions a year with them |
| 23:22.57 | NOT_guru | <PROTECTED> |
| 23:23.05 | NOT_guru | this is true |
| 23:23.07 | JT | real tight with their firmware upgrades |
| 23:23.12 | JT | stupid charging for firmware |
| 23:23.15 | NOT_guru | you have to feel your away around cisco stuff |
| 23:23.21 | NOT_guru | that too |
| 23:23.23 | JT | with a hammer |
| 23:23.33 | votre | I know that are phones are currently just plugged in with regular phone cords into your generic phone jacks. We have plain old phones that can be bought from any Best Buy or Wal-Mart or anything, will these run the risk of not working with what Asterisk provides? |
| 23:23.38 | NOT_guru | but really I think you can get the firware account for like $7 |
| 23:24.10 | NOT_guru | you would want FXS ports on your card for the phones |
| 23:24.32 | NOT_guru | and FXO ports if your lines going into the phone system are analog |
| 23:24.49 | NOT_guru | or a t1 / e1 card if the phone lines are digi |
| 23:24.50 | NovceGuru | Anybody using http://voicelift.com/ ? I had a pleasant conversation with their "support manager" today, just looking for some opinions |
| 23:25.20 | votre | oh wow. This is starting to get over my head. Maybe I will just report to our head IT what you guys have told me and see if I can persuade him |
| 23:25.22 | NOT_guru | votre how many extensions now? |
| 23:25.36 | JT | votre: |
| 23:25.38 | JT | ~thebook |
| 23:25.43 | jbot | extra, extra, read all about it, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:25.51 | votre | well we have quite a few extensions but 12 dedicated lines |
| 23:26.03 | JT | votre: are the lines analogue or digital? |
| 23:26.19 | votre | I am not sure. Is there a way to find out? |
| 23:26.35 | votre | I know it comes in as T1 |
| 23:26.58 | NOT_guru | do you have 12 phone lines plugging into your phone system |
| 23:26.59 | votre | But again, I am pretty darn clueless when it comes to this technology |
| 23:27.02 | JT | T1s aren't analogue |
| 23:27.09 | votre | NOT_guru: yes |
| 23:27.19 | NOT_guru | but it could be broken out into analog lines |
| 23:27.29 | JT | NOT_guru: well that'd be pretty stupid |
| 23:27.32 | NOT_guru | thats what we have done at work from previous system |
| 23:27.38 | JT | sounds like something "only in america" |
| 23:27.39 | NOT_guru | its lagacy |
| 23:27.44 | NOT_guru | yes it is |
| 23:28.04 | NOT_guru | but anyways |
| 23:28.20 | votre | yes, that is what I am not sure about. I know there are software clients for the phone system that run on our Windows comps and that you can call from those clients if you have a headset in your comp |
| 23:28.27 | NOT_guru | if your t-1 get broken down into analog lines and then plugged into your phone system |
| 23:29.40 | *** join/#asterisk pdx5k (n=casey@64.81.142.112) |
| 23:30.03 | pdx5k | i installed 1.4.4 and have it working fine except for meetme .. i can't get it to work |
| 23:30.19 | pdx5k | when i call the extension for the conference it says there's no meetme application for that extension |
| 23:30.22 | shido6 | got ztdummy? |
| 23:30.28 | pdx5k | yes |
| 23:30.37 | shido6 | 1/2 way tere |
| 23:30.59 | NOT_guru | probably votre |
| 23:31.17 | NOT_guru | as JT mentioned the book is probably a good place to see functions |
| 23:31.33 | votre | OK, well I will investigate further |
| 23:31.42 | NOT_guru | ~thebook |
| 23:31.43 | jbot | i heard thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:32.14 | pdx5k | what do you have to do besides install ztdummy? |
| 23:32.21 | PioneerVM | anyone here use the AGI library in perl? |
| 23:32.24 | NOT_guru | dinner is calling my name have a good night all |
| 23:32.31 | shido6 | me, too |
| 23:33.26 | *** join/#asterisk JT_ (n=jon@unaffiliated/jt) |
| 23:34.47 | PioneerVM | anyone here familiar with "SAY ALPHA" command |
| 23:35.03 | PioneerVM | i'm having problems calling it -- it does not say anyting but the SAY NUMBER works |
| 23:37.01 | pdx5k | pbx_extension_helper: No application 'MeetMe' for extension (inbound, 7, 2) |
| 23:37.17 | pdx5k | ztdummy and zaptel are loaded .. so why is it not working? |
| 23:38.25 | JT | maybe meetme is not loaded |
| 23:38.50 | pdx5k | how do i check on that? |
| 23:39.04 | JT | see if it loads when asterisk starts |
| 23:39.09 | kieranmullen2 | I am having an odd problem with 2 ivrs playing at the same time.. I want an ext to go to ivr |
| 23:39.09 | kieranmullen2 | extensions_additional.conf |
| 23:39.09 | kieranmullen2 | [ext-local] |
| 23:39.09 | kieranmullen2 | include => ext-local-custom |
| 23:39.10 | kieranmullen2 | exten => 1,1,Goto(ivr-2|s|1) |
| 23:39.10 | JT | you may need to use logging |
| 23:39.12 | kieranmullen2 | exten => 2,1,Goto(ivr-13|s|1) So the ext picks up but so does "im sorry that is not a valid extension" |
| 23:40.19 | *** join/#asterisk ManxPower (n=manxpowe@64.246.207.186) |
| 23:40.42 | kieranmullen2 | jt - me ? |
| 23:40.43 | kieranmullen2 | http://clip.drlinky.com/20 |
| 23:40.50 | kieranmullen2 | I pasted it there |
| 23:41.00 | kieranmullen2 | I am calling from fwd |
| 23:41.10 | JT | no, pdx5k |
| 23:41.16 | kieranmullen2 | sorry |
| 23:41.17 | pdx5k | when i type 'core show applications' there's no meetme in there |
| 23:41.22 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 23:44.11 | pdx5k | how do i get meetme to load? |
| 23:44.26 | JT | check modules.conf |
| 23:44.54 | JT | and if there's any errors upon asterisk starting |
| 23:45.23 | pdx5k | meetme isn't in the modules.conf file |
| 23:45.38 | JT | are modules set to autoload? |
| 23:45.41 | pdx5k | yes |
| 23:47.14 | pdx5k | i don't see any errors when it starts |
| 23:48.18 | pdx5k | what is the module name for meetme .. maybe i can load it via the console |
| 23:49.33 | pdx5k | when i do 'module show' meetme isn't there |
| 23:51.41 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:52.12 | *** join/#asterisk fab5freddy (n=vmware@bas1-montreal19-1177818910.dsl.bell.ca) |
| 23:52.29 | fab5freddy | What is a good soft client to use that can record phone calls? |
| 23:52.29 | pdx5k | why it's not there .. i have no idea .. app_meetme.c is in the apps/ dir .. so it should have built it |
| 23:53.09 | kieranmullen2 | i dont know about good but gizmoproject.com client also has a connect to asterisk feature and it records |
| 23:53.12 | *** join/#asterisk ctaloi (n=ctaloi@nat-66-218-1-160.usadatanet.com) |
| 23:53.32 | fab5freddy | kieranmullen2: is it in the debian repositories? |
| 23:53.45 | pdx5k | why would meetme not be compiled? |
| 23:53.48 | kieranmullen2 | they have a linux client on their site |
| 23:53.48 | JT | fab5freddy: asterisk can do recording |
| 23:54.01 | ctaloi | pdx5k - you need zaptel for meetme |
| 23:54.06 | JT | pdx5k: is the object file in the modules directory? |
| 23:54.14 | kieranmullen2 | Linux Packages Avaliable |
| 23:54.14 | kieranmullen2 | Linspire 5.0+ CNR |
| 23:54.14 | kieranmullen2 | Binary Tarball |
| 23:54.14 | kieranmullen2 | RPM Install Package |
| 23:54.14 | kieranmullen2 | DEB Install Package |
| 23:54.18 | ctaloi | anyone using multiple interfaces/ips and Asterisk ? |
| 23:54.25 | shido6 | debian repositories.... hrmm compile man |
| 23:54.29 | pdx5k | ctaloi: i have zaptel installed |
| 23:54.30 | shido6 | dont be scared to compile |
| 23:54.37 | shido6 | compile from source :) |
| 23:54.42 | kieranmullen2 | http://www.gizmoproject.com/download-linux.html |
| 23:55.39 | fab5freddy | kieranmullen2: thanks man |
| 23:56.06 | *** join/#asterisk n4ycw (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 23:56.16 | JT | fab5freddy: why client side recording? |
| 23:56.20 | pdx5k | JT .. the object's not there |
| 23:56.24 | kuku5 | How can I do multi lot parking ? |
| 23:57.05 | fab5freddy | jt: i want to reach a call centre who is recording my call so i want to record the call too |
| 23:58.07 | kieranmullen2 | asterisk does a good job of recording on its own anyway right? |
| 23:59.01 | fab5freddy | kieranmullen2: i am still not an expert with asterisk yet.. having some troubles getting it setup.. so i am using my soft client direct to the provider for the moment |
