| 00:01.21 | dacs | lanning: exten => _XX.,1,NoOp(Call received from VoicePulse) |
| 00:01.47 | lanning | show the sip.conf entry |
| 00:02.04 | lanning | what context does it drop the call into? |
| 00:04.05 | *** join/#asterisk salzh (n=Administ@61.87.216.5) |
| 00:07.10 | *** join/#asterisk j3rr3t (n=chatzill@70-91-184-69-BusName-Oregon.hfc.comcastbusiness.net) |
| 00:07.34 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:09.29 | j3rr3t | Looking for a little advice please. I have a small business that I need to setup a phone system for. I need two phones in different locations to "work together". I was thinking that a hosted asterisk PBX and two regular (SIP based?) phones to connect to the PBX is what I need. If possible I would like to use Skype for the incoming and outgoing calls. Thoughts? First steps? |
| 00:10.24 | *** join/#asterisk seb- (n=seb@li30-51.members.linode.com) |
| 00:10.39 | seb- | how tell asterisk to listen on OTHER NIC if it defaults to wrong one? |
| 00:17.18 | drmessano | bindaddr |
| 00:19.32 | seb- | drmessano: silly me..i can just just set up a DNS subdomain to point to whatever i want |
| 00:19.36 | seb- | e.g. voip.mydomain.com |
| 00:21.06 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 00:21.52 | dacs | lanning: http://pastebin.ca/1474881 |
| 00:23.46 | *** join/#asterisk apdg (n=grey@ash.stopavoiding.us) |
| 00:25.24 | j_kroon | is there a good guide on isdn somewhere? |
| 00:25.31 | j_kroon | things like what is B3 etc ... |
| 00:27.35 | dacs | B3? what is that |
| 00:28.02 | dacs | you mean DS3? |
| 00:30.25 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-19642a327e2b791f) |
| 00:32.21 | coppice | B3 == B channel layer 3 |
| 00:41.41 | eppigy | what |
| 00:42.25 | *** join/#asterisk l2trace99 (n=jr@205.245.6.162) |
| 01:07.47 | j_kroon | hi guys, i'm remotely (different country) trying to figure out why inbound calls to an asterisk server isn't working, making use of divas4linux-melware-3.1.2-109.50-1, asterisk 1.6.1.1 on a Diva Server PRI. |
| 01:09.45 | *** join/#asterisk l2trace99 (n=jr@205.245.6.162) |
| 01:12.48 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 01:15.28 | *** part/#asterisk ruben23 (n=RPL@124.107.3.178) |
| 01:19.36 | leifmadsen | o.O |
| 01:22.13 | lanning | dacs: so a call comes in from voicepulse (via voicepulse-primary) and gets dropped into the context voicepulse-in. |
| 01:22.44 | lanning | so what is in the voicepulse-in context? |
| 01:24.11 | j_kroon | i can makes calls out from the box, but I'm unable to receive any. |
| 01:24.21 | j_kroon | how can i go about trouble-shooting that? |
| 01:25.06 | lanning | core set verbose 10 |
| 01:26.15 | *** join/#asterisk BeeBuu (n=beebuu@219.130.234.144) |
| 01:26.51 | BeeBuu | how to make a user only listen in a meet room? |
| 01:34.32 | Aiatek | with "M" |
| 01:34.39 | Aiatek | BeeBuu |
| 01:37.44 | BeeBuu | Aiatek: which command? |
| 01:38.23 | Aiatek | exten => 2500,1,MeetMeAdmin(2525,M,2012) |
| 01:38.36 | BeeBuu | "M" make the user can't listen |
| 01:39.10 | Aiatek | that sentences mutes the user 2012 in the conferenced 2525 |
| 01:39.21 | BeeBuu | i want to make a user can't talk, but can listen |
| 01:39.26 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:41.39 | Aiatek | BeeBuu did you tried it |
| 01:41.47 | Aiatek | like i wrote to you |
| 01:41.50 | Aiatek | ? |
| 01:43.32 | Aiatek | BeeBuu |
| 01:44.33 | BeeBuu | OK,i got it! thanks. Aiateck |
| 01:45.13 | Aiatek | BeeBuu |
| 01:45.20 | Aiatek | what was the sentences |
| 01:45.21 | Aiatek | ? |
| 01:45.32 | BeeBuu | i make a mistake~~~ |
| 01:45.39 | Aiatek | you can do it |
| 01:45.39 | BeeBuu | sorry |
| 01:45.43 | Aiatek | in realtime |
| 01:45.46 | Aiatek | by cli |
| 01:45.53 | Jumpie | does hud lite use some different ports than regular sip/rtp range? |
| 01:46.03 | Jumpie | im tryin to use it to connect to remote server and its faling but i nkow the info is right |
| 01:46.21 | Aiatek | MeetMe mute <confno> <user> |
| 01:46.25 | Aiatek | from cli |
| 01:46.28 | Aiatek | :) |
| 01:46.29 | *** join/#asterisk s14ck (n=s14ck@190-76-76-241.dyn.movilnet.com.ve) |
| 01:46.36 | BeeBuu | yeah~~~ |
| 01:48.38 | *** join/#asterisk Deeewayne (n=dwayne@76.29.245.9) |
| 01:48.38 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 01:49.24 | *** join/#asterisk chendy (n=chatzill@58.251.102.197) |
| 01:55.01 | *** join/#asterisk tobias (n=tobias@user-0ce2hp1.cable.mindspring.com) |
| 02:16.55 | dacs | Aiatek: sorry i had to leave for a few...so what do you thing my problem is with not ablt to dial ext 102 |
| 02:21.00 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 02:27.31 | Jumpie | ah im an idiot |
| 02:27.33 | Jumpie | 6600 isnto pen |
| 02:29.37 | *** join/#asterisk Deeewayne (n=dwayne@76.29.245.9) |
| 02:29.37 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 02:30.19 | *** join/#asterisk saint_ (n=templar@pdpc/supporter/base/saint) |
| 02:31.37 | *** join/#asterisk OrNix (n=ornix@78.40.81.34) |
| 02:40.19 | dacs | everyime to try to dial ext 102 or 101 it hung up and i get [Jun 25 02:31:31] WARNING[17327]: pbx.c:2518 __ast_pbx_run: Invalid extension '10', but no rule 'i' in context 'outgoing' |
| 02:40.21 | dacs | [Jun 25 02:33:03] NOTICE[3109]: chan_sip.c:15746 handle_request_subscribe: Received SIP subscribe for peer without mailbox: Jane http://pastebin.ca/1474999 |
| 02:49.19 | *** part/#asterisk dacs (n=chatzill@unaffiliated/dacs) |
| 02:55.46 | *** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net) |
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| 03:39.05 | thehar | russellb: ping |
| 03:43.25 | thehar | http://www.kitchensoap.com/ << highly suggest watching the O'Reilly Velocity slides |
| 03:44.29 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
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| 04:10.41 | Micc | I'm running a non optimized asterisk 1.6.1.1. Every time I reload it crashes, how can I see why it is crashing? |
| 04:11.14 | Micc | should I ask this in another channel maybe? |
| 04:14.51 | *** part/#asterisk cathederal (n=catheder@216.167.183.174) |
| 04:18.29 | rob0 | If you don't tell it to fork at startup (command line option and in asterisk.conf) you will see the errors at the console. |
| 04:18.50 | *** join/#asterisk CunningPike (n=CunningP@S01060014bf81366b.vc.shawcable.net) |
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| 04:39.47 | j3rr3t_ | Looking to find an inexpensive hosted asterisk solution, any recommendations or good forums I should check out? |
| 04:47.49 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust923.oxfd.cable.ntl.com) |
| 04:58.04 | *** part/#asterisk seb- (n=seb@li30-51.members.linode.com) |
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| 06:35.33 | *** join/#asterisk xrmx__ (n=rm@host176-254-dynamic.14-87-r.retail.telecomitalia.it) |
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| 06:45.24 | McL0VIN | guy i am having hard time to get zoiper working properly with *...not translation problem thu because x-lite works fine |
| 06:48.31 | AlmightyOatmeal | how far would a pII 300-400mhz get me? |
| 06:48.49 | McL0VIN | AlmightyOatmeal: next room :) |
| 06:48.52 | AlmightyOatmeal | my dual 800 is having hardware issues and i need to move * to another box |
| 06:49.06 | AlmightyOatmeal | lol McL0VIN |
| 06:49.26 | McL0VIN | AlmightyOatmeal: what issues you are having |
| 06:50.07 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 06:50.13 | AlmightyOatmeal | McL0VIN: irrelivant to my question |
| 06:50.55 | McL0VIN | AlmightyOatmeal: i mean the HW |
| 06:51.06 | AlmightyOatmeal | i will be playing back sln files and i don't imagine myself having more than 2-3 lines running at a given time |
| 06:51.23 | AlmightyOatmeal | just wondering of 300-400mhz would be enough |
| 06:51.54 | McL0VIN | i don't know |
| 06:51.59 | AlmightyOatmeal | McL0VIN: i know what you meant but my dual 800's issues are not pertinant, i know my servers very well |
| 06:52.24 | McL0VIN | I am having hard time to get zoiper working properly with *...not translation problem thu because x-lite works fine |
| 06:52.39 | AlmightyOatmeal | its most likely something has gotten knocked loose.. after moving the server i'm having random lockups |
| 06:52.50 | AlmightyOatmeal | i know its not thermal or defective ecc ram |
| 06:52.58 | McL0VIN | i don't hear sounds playing from * |
| 06:53.23 | AlmightyOatmeal | are you on the same network as the * box? |
| 06:53.38 | AlmightyOatmeal | or are you doing it remotely? |
| 06:53.45 | McL0VIN | AlmightyOatmeal: no, but even when i am samething |
| 06:54.04 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.216) |
| 06:54.06 | AlmightyOatmeal | what codec are you using? |
| 06:54.15 | AlmightyOatmeal | i use ulaw and that works with my x-lite softphone |
| 06:54.20 | McL0VIN | AlmightyOatmeal: am at work now , so the night shit is getting slow, so thought to learn something new with * |
| 06:54.25 | McL0VIN | hold on |
| 06:56.08 | McL0VIN | AlmightyOatmeal: am using all of them |
| 06:56.20 | AlmightyOatmeal | i only use ulaw |
| 06:56.20 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.216) |
| 06:56.45 | McL0VIN | a-law ,u-law,Speex,iLBC 30, GSM and iLBC 20 |
| 06:57.36 | *** join/#asterisk oej (n=olle@24.Red-81-32-25.dynamicIP.rima-tde.net) |
| 06:58.09 | *** join/#asterisk xbp (i=irc@packet.framed.net) |
| 06:58.24 | AlmightyOatmeal | McL0VIN: for your specific user's context that you are connecting as with x-lite, what do you have as an allow line? |
| 06:59.18 | McL0VIN | AlmightyOatmeal: all |
| 07:01.32 | AlmightyOatmeal | i only have "allow=ulaw" |
| 07:01.34 | AlmightyOatmeal | try that |
| 07:04.31 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 07:06.53 | McL0VIN | same thing |
| 07:07.31 | AlmightyOatmeal | oh |
| 07:07.36 | AlmightyOatmeal | something way different |
| 07:08.22 | AlmightyOatmeal | in that case i'm not really sure |
| 07:08.49 | AlmightyOatmeal | it doesn't look like your sip.conf is setup properly to support your network |
| 07:09.01 | AlmightyOatmeal | and/or you have a firewall blocking something |
| 07:10.36 | festr_ | anyone know this? : whan CONTACT header is changed from number@10.1.1.1 to number@200.1.1.1 does it conform to RFC? Someone says that we are changing CONTACT uri ip address and they cannot pare BYE to call leg |
| 07:16.59 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 07:17.13 | *** part/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 07:18.49 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 07:22.53 | kaldemar | festr_: changed where? changed how? changed when? elaborate please. |
| 07:28.27 | *** join/#asterisk EricDGlobeTroote (n=GlobeTro@201.218.90.155) |
| 07:29.22 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
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| 07:47.41 | *** join/#asterisk tokozedg (n=Administ@92.241.73.254) |
| 07:48.38 | tokozedg | hi, i want to make so that, if callerid number equals from 23 to 27 do somethig, i tryied " [23-27]" but dont work |
| 07:48.48 | tokozedg | can anyone help |
| 07:49.34 | *** join/#asterisk s0lid (n=s0lid@61.28.187.210) |
| 07:52.47 | *** join/#asterisk CodeWork (n=Miranda@p5083BC53.dip.t-dialin.net) |
| 07:57.02 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 07:57.47 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 07:59.08 | kaldemar | tokozedg: something inside [] matches just one digit. 2[3-7] will match from 23 to 27. [3-7] equals to [34567]. |
| 08:00.22 | kaldemar | tokozedg: and that is with patterns only, if you're comparing the value of the function CALLERID(num) for example, that doesn't necessarily work. show the line you're using and you'll get a more accurate answer. |
| 08:01.05 | tokozedg | kaldemar, exten => _X.,1,Gotoif($["${CALLERID(num)}" = "1100"] || $["${CALLERID(num)}" = "120[0-1]"]?rviani:shemowmeba) |
| 08:06.10 | wdoekes | tokozedg: use the regexp-operator instead of the equality operator |
| 08:06.39 | tokozedg | wdoekes, what does it can? can you give little example? |
| 08:07.14 | *** join/#asterisk decimalz (n=pbxk1064@203.171.199.211) |
| 08:08.58 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 08:15.42 | tokozedg | wdoekes, exten => _X.,1,Gotoif($["${CALLERID(num)}" = "1100"] || $["${CALLERID(num)}" = "100" : "([0-7]+)"]?rviani:shemowmeba) |
| 08:15.47 | tokozedg | what am i doing wrong |
| 08:21.11 | wdoekes | "=" is not the regexp operator |
| 08:21.27 | wdoekes | http://www.voip-info.org/wiki/view/Asterisk+Expressions |
| 08:22.13 | wdoekes | and the logical or is only one pipe |
| 08:23.19 | wdoekes | $["${CALLERID(num)}"="1100"] | $["${CALLERID(num)}":"120[01]"] |
| 08:23.41 | wdoekes | (for your former example.. I can't parse what you're last statement is supposed to mean) |
| 08:24.43 | wdoekes | ("120[01]$" to be exact.. there is an implicit ^ for the : operator) |
| 08:25.22 | kaldemar | GotoIf($[$["${CALLERID(num)}" = "1100"] | $["${CALLERID(num)}" =~ "120[01]"]]??rviani:shemowmeba) |
| 08:25.37 | kaldemar | oops, one extra ? slipped in there. |
| 08:25.48 | wdoekes | and you forgot the anchors |
| 08:26.09 | tokozedg | kaldemar, thanks, i`ll try |
| 08:28.24 | kaldemar | so with achors and without double ? - GotoIf($[$["${CALLERID(num)}" = "1100"] | $["${CALLERID(num)}" =~ "^120[01]$"]]?rviani:shemowmeba) |
| 08:28.47 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
| 08:28.53 | tokozedg | kaldemar, worked thanks alot |
| 08:28.54 | tokozedg | :) |
| 08:28.56 | kaldemar | if you want it to match 1200-1207 instead of 1200-1201, replace [01] with [0-7] |
| 08:29.15 | tokozedg | kaldemar, ok |
| 08:29.40 | kaldemar | and as wdoekes suggested might be better to add the anchors ^$ |
| 08:30.19 | tokozedg | thanks guys, buys |
| 08:37.40 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 08:44.18 | *** part/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 08:48.14 | McL0VIN | i know am still learning, but i have a question: if i want * to dial lets say 20 phone numbers and leave a message ? how can i do that |
| 08:48.40 | *** join/#asterisk awk (n=phillip@gw1.security.web.za) |
| 08:49.17 | McL0VIN | am still in chapter 5 and i saw the variables , but will i assign a 20 phone numbers as a value to that variable?! |
| 08:49.23 | awk | hi, any advice on the following asterisk 1.4.24 [Jun 25 14:59:48] WARNING[17881]: channel.c:3319 ast_channel_make_compatible: No path to translate from OOH323/Band-X-4656 to SIP/4001-097b2c18(256) && == Begin MixMonitor Recording SIP/4001-097b2c18 && [Jun 25 14:59:48] NOTICE[17881]: channel.c:2379 __ast_read: Dropping incompatible voice frame on OOH323/Band-X-4656 of format g729 since our native format has changed to 0x0 (nothing) |
| 08:49.42 | awk | Then as soon as somebody hangs up, asterisk crashes? What is this No path to translate from? |
| 08:49.57 | McL0VIN | awk: sound like a codec issue |
| 08:51.19 | awk | McL0VIN: I have put a 30 channel g729 license on, and disabled all codecs except that.. still same thing, then I tried enabling *.* codecs, same thing, tried all veriations. my first guess was a codec issue too. |
| 08:52.32 | McL0VIN | awk: then i don't know , i am still on my baby steps in * :) wait until one of these guys come in a help you |
| 08:53.39 | awk | hmm, ok just setup a test lab for this... [Jun 26 10:51:05] WARNING[29535]: channel.c:3426 ast_channel_make_compatible: No path to translate from OOH323/h323-mobile-2c54 to SIP/306-09f3a798(4) && -- SIP/339-09ec82f8 is ringing && voip01*CLI> && Disconnected from Asterisk server && Executing last minute cleanups... <---- another crash |
| 08:55.58 | AlmightyOatmeal | that has got to be the most graceful crash i've ever seen |
| 08:57.14 | awk | getting gdb info now... |
| 08:57.39 | madduck | is there some way to call Wait(2) and goto some other context if the user presses * during that time? |
| 08:57.51 | madduck | i don't want Authenticate() or an announcement |
| 08:57.59 | madduck | I have a single number for a conference setup |
| 08:58.10 | madduck | and I want to give people the option to create a room instead of joining one |
| 08:58.14 | madduck | by pressing * |
| 08:58.26 | AlmightyOatmeal | madduck: i dont think you can read input while using wait() |
| 08:58.45 | madduck | well, maybe not wait() but something similar |
| 08:58.52 | AlmightyOatmeal | you could have a null 2 second playback(file) |
| 08:59.04 | AlmightyOatmeal | s/playback/background |
| 08:59.09 | madduck | okay. what a hack! ;) |
| 08:59.14 | madduck | that could work thoug |
| 08:59.41 | madduck | it's a pain. i wish meetme() would give me the option to switch to creation mode upon press of * |
| 08:59.51 | AlmightyOatmeal | just record 2 seconds of nothing and have away with it.. i'm sure theres other methods but i'm far from a master |
| 08:59.58 | AlmightyOatmeal | anyway, off to do stuff |
| 09:00.10 | awk | here is the crash info http://pastebin.com/f12dc6f40 |
| 09:02.10 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 09:03.19 | kaldemar | madduck: use WaitExten |
| 09:04.09 | McL0VIN | am still in chapter 5 and i saw the variables , but will i assign a 20 phone numbers as a value to that variable?! |
| 09:04.41 | kaldemar | madduck: so, don't use a silence playback hack when there's an appropriate application for that. |
| 09:06.59 | kaldemar | awk: that very much sounds like a codec issue. are both the OOH323 endpoint and the SIP endpoint using G.729? when you allowed more codecs, did you allow codecs that the endpoints actually support? did you allow them right? some holes in the question. but it shouldn't crash, that's for sure. |
| 09:09.00 | McL0VIN | kaldemar: if i want * to dial lets say 20 phone numbers and leave a message ? how can i do that? |
| 09:11.22 | kaldemar | originate the calls with AMI originate or call files. |
| 09:17.01 | *** join/#asterisk oej (n=olle@203.Red-81-34-197.dynamicIP.rima-tde.net) |
| 09:24.03 | McL0VIN | kaldemar: what is AMI? |
| 09:24.52 | decimalz | McL0VIN - want to telespam? |
| 09:26.14 | McL0VIN | decimalz: No, it for my church |
| 09:27.46 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:27.51 | kaldemar | McL0VIN: http://www.lmgtfy.com/?q=asterisk+ami |
| 09:30.37 | McL0VIN | kaldemar: http://www.lmgtfy.com/?q=thank+you |
| 09:31.17 | McL0VIN | lol |
| 09:33.00 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-234.gibnet.gi) |
| 09:35.40 | awk | McL0VIN if I match the codec then I don't get a codec issue but I still get a crash |
| 09:37.38 | *** join/#asterisk Unkn0wn (n=Unkn0wn@C-59-100-163-75.per.connect.net.au) |
| 09:38.42 | awk | after anyone hangs up |
| 09:39.23 | AlmightyOatmeal | if i do playback() can i do playback(folder/file)? |
| 09:43.07 | *** join/#asterisk grEvenX (n=even@apb99b.ip.ssc.net) |
| 10:02.15 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
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| 10:06.32 | aces1up | anyone here familiar with installing wanpipe drivers to work with sangoma a200 card i am having issues, keep getting the following error when configuring wanpipe drivers: error: âstruct dahdi_spanâ has no member named âechocanâ |
| 10:10.00 | Zeeek | you may have better luck later in the day as more people come on from USA |
| 10:14.50 | *** join/#asterisk odenkos (n=feeds@85-135-133-212.adsl.slovanet.sk) |
| 10:18.55 | madduck | kaldemar: sweet, thanks! |
| 10:25.11 | *** join/#asterisk Selveste1 (n=Selveste@88.83.86.20) |
| 10:26.45 | aces1up | how can i uninstall my current version of dahdi? |
| 10:30.19 | kaldemar | aces1up: how did you install it? |
| 10:33.13 | *** join/#asterisk generalBordeaux_ (n=generalB@proxy.iiit.ac.in) |
| 10:41.41 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 10:53.36 | McL0VIN | ~API |
| 10:53.36 | infobot | somebody said api was (Application Programming Interface) APIs allow you to program to a pre-constructed interface (the API) instead of programming a device or piece of software directly. This allows for faster development, since programming to a device's API is designed to be easier than programming to a device directly. APIs allow you to program without having intimate knowledge of the device or software that you a |
| 10:53.56 | *** join/#asterisk dna_ (i=_imspect@89.249.218.19) |
| 10:54.16 | dna_ | hi, is asteriskNOW a trial/demo or full distro? |
| 10:55.10 | McL0VIN | where can i learn about API, |
| 10:56.40 | kaldemar | McL0VIN: what API? |
| 10:57.21 | McL0VIN | kaldemar: like in general, how do i practice it ! learn it ? |
| 10:57.50 | *** join/#asterisk [jmc] (n=John@93-45-211-153.ip104.fastwebnet.it) |
| 10:57.56 | kaldemar | API is an abstraction. are you referring to manager API? |
| 10:59.30 | *** join/#asterisk neilh23 (n=chatzill@host213-123-185-55.in-addr.btopenworld.com) |
| 10:59.53 | McL0VIN | kaldemar: i don't know if i am asking the right question or no, I thought that API is a programming language ?and i want guides and tutorials on it ! |
| 11:00.01 | McL0VIN | in general |
| 11:00.03 | tomodachi | dna_: is a distribution with a full working asterisk |
| 11:00.15 | dna_ | tomodachi: ok thanks :) |
| 11:00.24 | tomodachi | dna_: although i personally would say i'd use it as a demo |
| 11:00.39 | tomodachi | others may disagree |
| 11:01.19 | dna_ | its not very stable you mean? |
| 11:01.35 | tomodachi | i think its stable |
| 11:01.40 | dna_ | okay |
| 11:01.52 | tomodachi | dont know what asterisk version its using though |
| 11:01.56 | tomodachi | i still use 1.2 |
| 11:01.57 | kaldemar | McL0VIN: API is not a programming language. it stands for application programming interface, and like the name says, API's are interfaces. |
| 11:02.04 | dna_ | i see |
| 11:02.04 | tomodachi | wich is in my opinion very stable |
| 11:02.08 | dna_ | :D |
| 11:02.48 | dna_ | you do need PBX hardware right? |
| 11:02.55 | kaldemar | if you want to learn asterisk or have total control over your phone system, use plain asterisk without a GUI. |
| 11:02.56 | tomodachi | dna_: no |
| 11:03.08 | dna_ | depends on the phones used? |
| 11:03.09 | tomodachi | dna_: listen to kaldemar hes right |
| 11:03.10 | aces1up | what is the correct procedure to remove asterisk from my system? |
| 11:03.18 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:03.18 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:03.26 | kaldemar | aces1up: how did you install it? depends on that. |
| 11:03.27 | tomodachi | aces1up: depends on the system, and how you set up asterisk, there is no "generic tip" |
| 11:03.37 | aces1up | hrmm just compiled from tar |
| 11:03.38 | tomodachi | other than "try removing it" |
| 11:03.42 | tomodachi | make uninstall |
| 11:03.55 | kaldemar | aces1up: ^ that goes for dahdi too |
| 11:04.40 | aces1up | is the correct procedure for dahdi and wanpipe asterisk-->dahdi-->wanpipe? |
| 11:04.47 | aces1up | installation procedure that is? |
| 11:04.59 | [jmc] | hi |
| 11:05.36 | kaldemar | aces1up: the other way round. |
| 11:05.44 | dna_ | you need the PBX hardware cards for external lines right? |
| 11:05.58 | dna_ | if u want pure internal voip u can do with normal networking equipment and voip phones? |
| 11:06.39 | kaldemar | dna_: if you attach external lines directly to your box, yes. but there are also ITSP's who offer e.g. DID services over VoIP. |
| 11:07.00 | [jmc] | I have been given a Linksys SPA3102 |
| 11:07.04 | kaldemar | dna_: yes. |
| 11:07.07 | dna_ | I see, thanks alot mate :) |
| 11:07.13 | [jmc] | has someone tried it_ |
| 11:07.15 | [jmc] | ? |
| 11:07.24 | tomodachi | dna_: i recomend that you read the asterisk manual, it gives you a good insight into ip telephony and asterisk, especially the starting chapters |
| 11:07.28 | [jmc] | I'd like to use it with my Asterisk server |
| 11:07.40 | [jmc] | so that I can make outgoing calls from my phone line |
| 11:07.48 | [jmc] | and receive from my SIP account |
| 11:09.23 | [jmc] | when I try to make a call from my softphone it tells me that "SIP/linksys-089540c8 is circuit-busy" |
| 11:09.34 | [jmc] | I'm attaching my sip.conf and extensions.conf |
| 11:09.52 | [jmc] | sip.conf => http://pastebin.com/d835bd11 |
| 11:10.06 | [jmc] | extensions.conf => http://pastebin.com/m66999ba5 |
| 11:10.51 | [jmc] | I think that the * configuration is right though |
| 11:11.11 | [jmc] | do I have to change the default SPA3102 dialplan too? |
| 11:13.05 | kaldemar | [jmc]: pastebin a sip debug of the call |
| 11:13.17 | [jmc] | ok, just a minute |
| 11:15.16 | [jmc] | here it is: http://pastebin.com/d464fed76 |
| 11:15.51 | dna_ | tomodachi: I'll read it for sure, I just had to ask some questions before i start :) |
| 11:16.34 | [jmc] | hmm, I can read a "SIP/2.0 404 Not Found" |
| 11:16.40 | [jmc] | wtf |
| 11:21.30 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-234.gibnet.gi) |
| 11:23.39 | kaldemar | [jmc]: that comes from the linksys. you need to configure the device so that it knows what to do with numbers. |
| 11:24.29 | kaldemar | asterisk side should be fine for now. |
| 11:25.43 | Poincare | Is it possible to do something like 'Dial(SIP/user&LOCAL/1@context)' without loosing the already set variables when starting 1@context? |
| 11:26.12 | [jmc] | kaldemar, that's what I thought, actually |
| 11:26.51 | [jmc] | but I think I enabled that number to go on my telephone line |
| 11:27.13 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:9c74:137e:b483:755b) |
| 11:27.17 | [jmc] | at least, according to some guides I found on the web |
| 11:27.32 | [jmc] | because the Linksys manual says nothing about that -_-' |
| 11:27.38 | kaldemar | Poincare: set variables so that they're inherited, Set(__VAR=value) instead of Set(VAR=value) |
| 11:28.00 | kaldemar | Poincare: _ means that the variables are inherited once, __ means indefinitely. |
| 11:28.09 | Poincare | kaldemar: ok thanks |
| 11:29.49 | [jmc] | I'll continue trying... thanks for the support, kaldemar :) |
| 11:32.30 | *** join/#asterisk casix (n=casix@pbxedifici.adamvozip.es) |
| 11:34.59 | casix | I have this error: un 26 13:31:26 WARNING[19056]: chan_zap.c:8816 zt_pri_error: 2 !! Got reject for frame 7, retransmitting frame 7 now, updating n_r! on a Asterisk 1.2.29-BRIstuffed-0.3.0-PRE-1y-s I have been googling but nothing found. This machine have a Junghanns 8x rdsi card |
| 11:35.02 | casix | any idea? |
| 11:36.56 | *** join/#asterisk kaptengu (n=kaptengu@unaffiliated/kaptengu) |
| 11:40.08 | tomodachi | dna_: i think you should do the opposite, read first, ask later |
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| 11:59.08 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:12.12 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 12:12.16 | ariel_ | Morning |
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| 12:37.22 | dna_ | tomodachi: I'm not asking technical questions about installing and configuring asterisk, i'm asking if it can do what i want :) |
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| 12:39.41 | ariel_ | dna_: there is not much you can't do with Asterisk it really depends on your will, programming ability and money. |
| 12:40.21 | dna_ | just making sure mate, no offense to the might of the holy asterisk 8] |
| 12:41.29 | ariel_ | dna_: I was not around when you asked your original question |
| 12:41.44 | dna_ | k |
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| 13:00.31 | Pan3D | dna_: you might want to restate what you posted earlier |
| 13:00.58 | dna_ | you mean my question/ |
| 13:02.11 | Pan3D | whatever it was, yes |
| 13:02.25 | dna_ | never mind, it got answered |
| 13:02.35 | Pan3D | shrugs |
| 13:02.37 | Pan3D | OK |
| 13:02.43 | dna_ | thanks :) |
| 13:03.12 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
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| 13:11.56 | *** part/#asterisk RoyCrowder (n=roy@net-216-37-64-130.in-addr.worldspice.net) |
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| 13:13.06 | ReDNeQ- | sup |
| 13:19.05 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 13:20.56 | Zeeek | TGIF ! |
| 13:21.20 | leifmadsen | Zeeek: amen |
| 13:21.27 | creativx | thank god for tits! |
| 13:21.34 | Zeeek | I'll see your Amen and raise you 1000 |
| 13:21.42 | Maliuta | it's only friday for 40 more minutes |
| 13:21.47 | creativx | 39 |
| 13:21.49 | Zeeek | creativx: they are extremely rare around here |
| 13:21.53 | Maliuta | get it over with |
| 13:22.27 | Maliuta | Zeeek: I think we get more tits in here than I can handle |
| 13:22.42 | Maliuta | of course that depends on context |
| 13:22.54 | creativx | well |
| 13:23.00 | creativx | take something good, and then complain about it |
| 13:23.01 | creativx | GTFO! |
| 13:23.13 | *** join/#asterisk cfh (n=luca@rex.vps.budgetdedicated.com) |
| 13:23.31 | Zeeek | nods sadly |
| 13:23.41 | Maliuta | creativx: you should be aware of the more coloquial usage of "tits" |
| 13:23.44 | Zeeek | There is one nice thing about Friday |
| 13:24.03 | creativx | Maliuta: i know only one usage of tits |
| 13:24.06 | creativx | no wait.. several actual |
| 13:24.09 | Maliuta | Zeeek: my oncologist has a clinic? |
| 13:24.10 | creativx | the word, or the tits themselves? |
| 13:24.18 | dna_ | tits? where! |
| 13:24.20 | Zeeek | The VoIP Users Conference: http://tr.im/voip |
| 13:24.22 | coppice | blue tits |
| 13:24.26 | coppice | coal tits |
| 13:24.42 | cfh | hi all, is possibile tag with vlan id the traffic from the pc interface on the polycom phones firmware 3.1.3 ? |
| 13:24.43 | Maliuta | coppice is being a tit |
| 13:24.44 | Zeeek | tit for tat |
| 13:25.45 | Maliuta | Zeeek: don't. Conferences are already exacerbating my depression |
| 13:26.04 | Zeeek | there is a conference on depression right htis minute |
| 13:26.14 | Maliuta | although I did get sponsorship to Kernel Conference Australia from Sun thanks to a mate |
| 13:26.29 | Zeeek | Popcorn Kernel? |
| 13:26.34 | ReDNeQ- | dahdi sucks |
| 13:26.48 | Maliuta | http://au.sun.com/sunnews/events/2009/kernel/agenda.jsp |
| 13:26.49 | *** join/#asterisk aethelrick (n=richard@mail.mexuar.net) |
| 13:26.57 | Maliuta | ReDNeQ: I think you'll find it blows |
| 13:27.19 | Maliuta | there may or may not be goats involved |
| 13:27.35 | ReDNeQ- | is there a way to remove and go back to zaptel with the latest version |
| 13:27.49 | ReDNeQ- | versions of * and the like |
| 13:27.50 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 13:27.53 | Maliuta | sure ... go back to 1.4.21 |
| 13:28.19 | Maliuta | I must say though I have seen 0 diff between zaptel and dahdi |
| 13:28.48 | aethelrick | does anyone know if I have to rebuild asterisk on ubuntu 9.04 if I want to use DAHDI insteda of zaptel for a dummy driver? |
| 13:29.21 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 13:29.33 | aethelrick | *instead |
| 13:29.51 | *** join/#asterisk esaym (n=user@cpe-24-174-186-34.satx.res.rr.com) |
| 13:29.57 | Maliuta | aethelrick: there are 1.6 packages for Debian that you should be able to use on Debian for LUsers |
| 13:31.35 | aethelrick | I prefer to think of it as Debian for PEOPLE... I am running server edition after all |
| 13:31.55 | Maliuta | it still rips off the work of Debian developers |
| 13:32.52 | Maliuta | and it's still dumbed down for the masses |
| 13:33.25 | Maliuta | you say "people" I say "LUsers" |
| 13:33.30 | *** join/#asterisk VaGoNeTaS (n=debian@xen.datapartner.cl) |
| 13:33.33 | VaGoNeTaS | xD |
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| 13:42.22 | fubada | hey. I'm looking to use asterisk as a conferencing destination. Can someone recommend a provider for sip termination |
| 13:42.30 | fubada | havent used * in a few years :) |
| 13:44.13 | fubada | is MeetMe still the app to use for conferencing? |
| 13:47.56 | *** join/#asterisk nauticalthinker (n=mratliff@74.5.197.138) |
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| 13:49.14 | nautical | okay guys...I'll probably get smacked for asking, but here goes...what are the main differences between version 1.4.25 and 1.6.1.1? |
| 13:50.35 | fubada | nautical :) read release notes maybe |
| 13:51.01 | Maliuta | changelogs are a good thing |
| 13:51.02 | fubada | but nautical, can you recommend a sip provider? trying to set up a 1-800 conferencing solution |
| 13:51.33 | *** join/#asterisk SlipperyChicken (n=andrew@LONDON14-1168107385.sdsl.bell.ca) |
| 13:51.46 | Maliuta | fubada: exactly where do you want this "SIP provider" to be? |
| 13:52.05 | fubada | basically i have a high bandwidth box out there on the net, and I need to set up asterisk on it for conferencing only |
| 13:52.09 | fubada | people will be calling in |
| 13:52.15 | fubada | i need a SIP account with a 1800 number i guess |
| 13:52.25 | fubada | by where you mean geographically? |
| 13:52.27 | Maliuta | 1800 in .au? |
| 13:52.32 | fubada | oh...US |
| 13:52.41 | Maliuta | specifics he;[ |
| 13:52.45 | Maliuta | help even |
| 13:52.47 | fubada | east coast, ct |
| 13:52.59 | Maliuta | catscan? |
| 13:53.04 | fubada | connecticut ;P |
| 13:53.11 | Maliuta | fucking americans |
| 13:53.20 | fubada | russian |
| 13:53.36 | Maliuta | russians are worse |
| 13:53.42 | Maliuta | <--- ukrianian |
| 13:53.43 | [TK]D-Fender | is in no hurry |
| 13:53.45 | fubada | oh here we go |
| 13:53.50 | fubada | Ukranian bwahahah |
| 13:53.55 | fubada | well. i wont say anything bad |
| 13:53.56 | fubada | im half |
| 13:54.02 | fubada | of your kind |
| 13:54.28 | Maliuta | fuck _try_ an russify me ... my gradfather survied winter '21/'22 |
| 13:54.50 | Maliuta | ULA all the way! |
| 13:55.07 | *** join/#asterisk hi365 (n=hi365@94.159.177.154) |
| 13:55.08 | fubada | you and your silly orange revolution |
| 13:55.17 | *** part/#asterisk nautical (n=mratliff@74.5.197.138) |
| 13:55.33 | Maliuta | you know who the ULA were right? |
| 13:55.42 | fubada | anyways :) sip provider suitable for my needs as a 1-800 conferencing endpoint in US, east coast |
| 13:55.45 | Maliuta | the forgot war of WWII |
| 13:56.51 | fubada | i dont know who the ULA were |
| 13:57.02 | fubada | google yields little results regarding ULA and ww2 |
| 13:57.03 | fubada | http://www.ula.com.ua/index_angl.htm |
| 13:57.39 | fubada | i just know you speak a very funny language and eat onions like we eat apples |
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| 14:01.15 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) |
| 14:02.18 | mchou | well well. apparently google voice is accepting new members, according to this: http://www.pcworld.com/businesscenter/article/167384/google_begins_adding_new_customers_for_voice_service.html?tk=rel_news |
| 14:06.20 | *** join/#asterisk iratik (n=itariki@209.248.216.146.nw.nuvox.net) |
| 14:06.38 | iratik | Doesn't asterisk usually listen on port 5060 for incoming sip connections? |
| 14:08.37 | kaldemar | iratik: it does, by default. |
| 14:09.10 | iratik | # netstat -an | grep 5060 |
| 14:09.11 | iratik | <PROTECTED> |
| 14:10.45 | kaldemar | are you sending the packets to the right interface? is a firewall blocking them? |
| 14:10.52 | iratik | i turned that off |
| 14:12.51 | kaldemar | netstat -anp | grep 5060 will show you if it is asterisk that listens to 5060. if it is, you have a network or client configuration problem. |
| 14:13.14 | *** join/#asterisk esaym (n=user@cpe-24-174-186-34.satx.res.rr.com) |
| 14:13.21 | iratik | its asterisk listening |
| 14:15.13 | iratik | This pastie shows 2 commands that completely contradict each other... http://pastie.org/525465 |
| 14:15.55 | iratik | its because its using tcp, needed udp |
| 14:17.36 | [TK]D-Fender | iratik: So stop asking on TCP. |
| 14:18.06 | kaldemar | iratik: -u uses udp mode, if your implementation has it |
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| 14:50.16 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:50.52 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.103.107) |
| 14:50.55 | DelphiWorld | hello |
| 14:51.18 | DelphiWorld | please what is the recommanded method to run asterisk in a Small Micro Processor (P3/256MB Memory? |
| 14:51.29 | *** join/#asterisk aethelrick (n=richard@mail.mexuar.net) |
| 14:52.11 | *** join/#asterisk DaveCanoe (n=Dave@strike.dclg.ca) |
| 14:52.26 | aethelrick | howdy peeps, does anyone know if it is possible to transfer a call from an AGI script or similar |
| 14:52.39 | ariel_ | DelphiWorld: my home pc is a Celeron 800mhz, 256mg ram and 20gig hdd. Works great |
| 14:53.03 | *** join/#asterisk Chinorro (n=Chino@54.225.117.91.dynamic.mundo-r.com) |
| 14:53.04 | ariel_ | asterisk 1.4.25 I even have the box doing my Inet router with IPtables. |
| 14:54.29 | DelphiWorld | ariel_: each OS you are runing? |
| 14:54.39 | *** join/#asterisk mort_gib (n=mjensen@111.Red-83-43-83.dynamicIP.rima-tde.net) |
| 14:54.51 | ariel_ | CentOS on this box 4.3 I think. |
| 14:55.04 | ariel_ | I have not had to update or change it in over a year |
| 14:55.28 | ariel_ | just keeps on running, I have also extensions setup with X10 to turn on and off lights in the house. |
| 14:55.40 | DelphiWorld | ariel_: please do you know astlinux? |
| 14:56.01 | ariel_ | no |
| 14:56.36 | DelphiWorld | ariel_: a embedded Asterisk applience |
| 14:56.37 | SuPrSluG | aethelrick: exec Executes a given Application |
| 14:57.23 | ariel_ | DelphiWorld: I have some systems running Asterisk Biz with Rpath, Which I really don't like at all. |
| 14:57.59 | *** join/#asterisk Joel (n=jjshoe@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 14:58.20 | *** join/#asterisk thomas (i=tm@tm.muc.de) |
| 14:58.20 | thomas | hello |
| 14:58.24 | thomas | have a: exten => s,n,GotoIf($["${ZielNr}"]?UmleitungAus:UmleitungEin) |
| 14:58.32 | thomas | i would liek chef, if ZielNr has a value |
| 14:58.35 | thomas | is it posible? |
| 14:58.36 | DelphiWorld | ariel_: but me i need to run asterisk shortly using any free / open source / small applience |
| 14:58.39 | thomas | a function "strlen" ? |
| 14:58.41 | thomas | length? |
| 14:59.32 | *** join/#asterisk galeras (n=galeras@186.80.186.118) |
| 14:59.38 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
| 15:02.02 | Aiatek | thomas i think NoOp can do that |
| 15:02.21 | [TK]D-Fender | DelphiWorld: Like what? |
| 15:02.23 | thomas | Aiatek: ${LEN(${ZielNr})} > 0 ? |
| 15:02.44 | eppigy | allo |
| 15:02.49 | Aiatek | exten => s,n,NoOP(${ZielNr}) |
| 15:02.50 | eppigy | HOLA |
| 15:02.55 | [TK]D-Fender | CRAZY PEOPLE |
| 15:03.00 | thomas | [TK]D-Fender: hi |
| 15:03.32 | thomas | Aiatek: jep. i mean check if more as 0: GotoIf([${LEN(${ZielNr})}]?Bla ? |
| 15:03.34 | [TK]D-Fender | thomas: exten => s,n,GotoIf($["${ZielNr}"=""]?UmleitungAus:UmleitungEin) |
| 15:03.50 | Zeeek | in ONE Hour ther Voip Users Conference with Matt Florell in living wideband |
| 15:03.57 | [TK]D-Fender | thomas: Yours would also work for different reasons |
| 15:04.10 | thomas | [TK]D-Fender: hello. yes, but i mean when i would like minimum 1 ? |
| 15:04.30 | thomas | [TK]D-Fender: exten => s,n,GotoIf($["${LEN(${ZielNr})} > 0"]?UmleitungAus:UmleitungEin) |
| 15:04.33 | thomas | is it correct? |
| 15:05.24 | [TK]D-Fender | thomas: No, you aren't thinking about why you have " <- in there |
| 15:05.38 | DelphiWorld | [TK]D-Fender: exemple: astlinux |
| 15:05.40 | DelphiWorld | [TK]D-Fender: is small |
| 15:05.46 | coppice | I remember when half the people here called me an idiot for pushing wideband. now its sexy. does that mean I'm sexy too? |
| 15:05.50 | DelphiWorld | [TK]D-Fender: 48MB |
| 15:07.17 | engien | no, youre still an idiot |
| 15:07.21 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 15:07.40 | Aiatek | exten => s,n,GotoIf($["${LEN(${ZielNr})}" > "0"]?UmleitungAus:UmleitungEin) <-------------- its this right [TK]D-Fender ? |
| 15:07.51 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 15:07.55 | *** join/#asterisk comradeb14ck (n=comradeb@72.37.252.50) |
| 15:07.55 | andres833 | hi pals |
| 15:08.17 | tzafrir_laptop | coppice, as long as you don't listen to it through some NB line |
| 15:08.46 | [TK]D-Fender | Aiatek: You shouldn't need quotes because of the expected result of that function. |
| 15:09.25 | Aiatek | ok |
| 15:17.57 | *** join/#asterisk ikks (n=ikks@190.144.69.234) |
| 15:18.11 | ikks | ping galeras, please take a look at your private messages |
| 15:20.09 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
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| 15:21.59 | eppigy | DONES ESTA |
| 15:22.01 | eppigy | DONDE |
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| 15:51.45 | Zeeek | Voip Users Conference with Matt Florell now #voip-users-conference http://vuc.me |
| 15:53.01 | *** join/#asterisk Chinorro (n=Chino@54.225.117.91.dynamic.mundo-r.com) |
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| 15:57.33 | *** join/#asterisk datafirm (n=wprater@74-94-79-145-Seattle.hfc.comcastbusiness.net) |
| 15:57.36 | datafirm | hello. |
| 15:57.50 | datafirm | have any of you seen a version of asterisk higher than 1.2.7.1 for Solaris? |
| 15:58.07 | leifmadsen | o.O |
| 15:58.35 | eppigy | TRABAJO |
| 15:59.04 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
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| 16:01.41 | Qwell | datafirm: 1.4 or 1.6 should work just fine on Solaris |
| 16:01.50 | Qwell | 1.6 moreso |
| 16:01.54 | datafirm | Qwell: do you have any build tips? |
| 16:02.00 | Qwell | ./configure, make install |
| 16:02.06 | datafirm | ha |
| 16:02.10 | Qwell | make sure you have gnu utils installed |
| 16:02.18 | datafirm | i'll try but that has not worked in the past for me on solairs |
| 16:02.34 | Qwell | Solaris uses stupid paths for things. add sfw/csw |
| 16:02.50 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 16:02.59 | datafirm | Qwell: i'll try 1.4.4 now. |
| 16:03.05 | Qwell | 1.4.4 is very old |
| 16:03.08 | datafirm | from here: https://svn.sunlabs.com/svn/solaris-asterisk/asterisk-1.4/trunk/ |
| 16:03.13 | datafirm | darn.. really? |
| 16:03.16 | Qwell | very |
| 16:03.21 | Qwell | type /topic |
| 16:03.35 | Qwell | http://downloads.asterisk.org/pub/telephony/asterisk/ |
| 16:04.00 | russellb | there have been 2685 changes in Asterisk 1.4 since version 1.4.4 |
| 16:04.03 | datafirm | Qwell: im on sunlabs svn.. perhaps there are modification they've made to the srource to help build. |
| 16:05.42 | *** join/#asterisk serph (n=serph@CPE001f5b006e5e-CM001371144daa.cpe.net.cable.rogers.com) |
| 16:07.25 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 16:08.23 | datafirm | how can I tell which features there are to disable with ./configure? |
| 16:08.33 | Qwell | --help |
| 16:08.51 | datafirm | Qwell: it just says I can disable feature.. --disable-FEATURE |
| 16:08.58 | *** join/#asterisk kannan (n=kannan@121.246.242.95) |
| 16:09.29 | Qwell | option-checking, dev-mode, largefile? |
| 16:09.45 | datafirm | Qwell: yeah.. not sure. Im looking in configure now. |
| 16:11.26 | datafirm | Qwell: yeah, gmake and make both give me errors. |
| 16:11.42 | Qwell | in editline? fix your paths :D |
| 16:12.08 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:14.08 | *** join/#asterisk bijit (i=1000@190.241.15.48) |
| 16:26.59 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 16:27.29 | jameswf | on a pri anyway to tel asterisk not to reset the span on a t200 timeout |
| 16:28.37 | *** part/#asterisk datafirm (n=wprater@74-94-79-145-Seattle.hfc.comcastbusiness.net) |
| 16:32.37 | Zeeek | jameswf: where are you? We miss you |
| 16:32.44 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
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| 16:59.21 | jaytee | jameswf, try adding resetinterval=never for each span in your zapata.conf or chan_dahdi.conf |
| 16:59.49 | beek | 'afternoon jaytee |
| 17:00.05 | jaytee | yo beek! whatup? |
| 17:01.00 | beek | Remotely working on another installation right now. Smallish system, about 30 sets and a PRI |
| 17:01.38 | beek | What's up at the zoo? |
| 17:02.47 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
| 17:04.18 | *** join/#asterisk RoyCrowder (n=roy@net-216-37-64-130.in-addr.worldspice.net) |
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| 17:07.24 | *** join/#asterisk chendy (n=alex@58.61.118.77) |
| 17:07.58 | hmmhesays | does asterisk have a tone detection app? |
| 17:07.58 | *** join/#asterisk clintc (n=clintc@n128-227-127-75.xlate.ufl.edu) [NETSPLIT VICTIM] |
| 17:07.59 | *** join/#asterisk micols (n=mio@rlogin.dk) [NETSPLIT VICTIM] |
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| 17:08.57 | anonymouz666 | hmmhesays: what kind of tone do you want to detect? |
| 17:08.57 | hmmhesays | 1100 hz |
| 17:08.59 | *** join/#asterisk rob0 (i=rob0@sorry.nodns4.us) |
| 17:08.59 | *** join/#asterisk salzh (n=Administ@122.144.138.5) [NETSPLIT VICTIM] |
| 17:09.00 | *** join/#asterisk xpot (n=james@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 17:09.00 | rob0 | yikes |
| 17:09.00 | jaytee | beek, it's like 87F and feels like 95F. Not as many visitors as usual due to the heat |
| 17:09.26 | anonymouz666 | hmmhesays: chan_dahdi should detect that |
| 17:09.33 | anonymouz666 | (I think) |
| 17:09.35 | beek | Cripes! We're in the low 70s now and it's getting to rain (again). |
| 17:09.47 | hmmhesays | I need to detect it from an incoming sip call |
| 17:11.44 | *** join/#asterisk intralanman (n=lanman@67.76.163.226) |
| 17:12.15 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 17:16.08 | hmmhesays | basically I'm wondering if it can listen for a fax tone in early media |
| 17:17.00 | *** join/#asterisk dacs (n=chatzill@unaffiliated/dacs) |
| 17:19.30 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 17:20.02 | dacs | wtf is wrong with zoIPER |
| 17:20.17 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 17:23.55 | [TK]D-Fender | dacs: Perhaps the same thing that is wrong with your inability to actually provide any details about your problems. |
| 17:24.07 | *** join/#asterisk ingcomrbr (n=ingcomrb@189.161.174.10) |
| 17:24.12 | *** join/#asterisk dbrotman (n=nonet@66.212.158.102.nauticom.net) |
| 17:24.43 | dacs | here we go again :p |
| 17:25.05 | kaldemar | something, maybe. not sure though. |
| 17:25.13 | [TK]D-Fender | dacs: Yup, you whine aimlessly, and I call you out on it! My favouritest game...EVAR! |
| 17:25.15 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 17:25.17 | kaldemar | dacs: did that answer your question? |
| 17:25.26 | dacs | [TK]D-Fender: i see it register my sip accounts with *, but i don't hear sounds. e.g main menu. |
| 17:25.54 | [TK]D-Fender | dacs: maybe what you're connecting it to is configured wrong <- |
| 17:26.04 | [TK]D-Fender | dacs: and I'm not seeing any debug from your attempts |
| 17:26.15 | *** join/#asterisk _pepo_ (n=_pepo_@190.94.131.140) |
| 17:26.25 | dacs | [TK]D-Fender: x-lite works fine for the same sip accounts |
| 17:26.33 | kaldemar | zoiper can be misconfigured on it's own too. |
| 17:26.47 | [TK]D-Fender | dacs: That doesn't help. Show us the problem, not say "well XYZ works". |
| 17:26.58 | _pepo_ | hi friends |
| 17:27.16 | dacs | [TK]D-Fender: ok, let re create the problem |
| 17:28.01 | bijit | whats the recommended codec to use via iax2 trunks to connect 2 asterisk? low bandwidth |
| 17:28.54 | hmmhesays | is it possible for asterisk to listen for tones in early media? |
| 17:29.07 | dacs | [TK]D-Fender: what debug do you need , rtp or sip |
| 17:29.21 | [TK]D-Fender | dacs: SIP for the failed call |
| 17:29.22 | *** join/#asterisk Greek-B0y (n=greek@41.222.89.77) |
| 17:29.22 | kaldemar | bijit: gsm is fine if bandwidth is more important than audio quality. ~13 kbps per channel. |
| 17:29.26 | _pepo_ | is there some way to make a redirect SIP server with Asterisk? (Where do I can look up info about that?) |
| 17:29.53 | dbrotman | How do you handle analag multiplexing with Asterisk? |
| 17:29.59 | bijit | kaldemar: combination of both? |
| 17:30.43 | kaldemar | bijit: low bandwidth and high quality don't mix. try the gsm codec out and decide if it's good enough for you. |
| 17:30.45 | [TK]D-Fender | dbrotman: As in? |
| 17:31.14 | kaldemar | _pepo_: you can do SIP redirects with application Transfer in asterisk. |
| 17:31.32 | *** join/#asterisk odenkos (n=feeds@85-135-160-109.adsl.slovanet.sk) |
| 17:31.40 | dbrotman | As in, I'm just starting took look at using asterisk for our small company. Our current PBX is a comdial, with 3 ports that deliver POTS to 6 internal lines |
| 17:31.56 | dbrotman | 1st is marked clearly "lines 1+2" and so on |
| 17:32.15 | bijit | kaldemar: will try that |
| 17:32.33 | *** join/#asterisk ruben23 (n=RPL@124.107.3.178) |
| 17:32.50 | *** join/#asterisk propellerhead (n=yogurt2u@host107.190-136-119.telecom.net.ar) |
| 17:32.51 | dbrotman | those double lines goto a breakout panel, and then to all the phones |
| 17:32.57 | [TK]D-Fender | dbrotman: that isn't multiplexing, thats just an analog port with 2 pairs |
| 17:33.06 | dbrotman | Ideally, I don't to have them rewired |
| 17:33.19 | dbrotman | Ok... How Does asterisk handle such pairs? :) |
| 17:33.36 | [TK]D-Fender | dbrotman: Asterisk is SOFTWARE. You don't plug ANYTHING into it |
| 17:34.14 | [TK]D-Fender | dbrotman: And I don't know of any common hardware maker that does products that do this except for dumb analog phones |
| 17:34.15 | Linuturk | today I had problems with my Digium T1 card |
| 17:34.22 | [TK]D-Fender | dbrotman: Rewire it |
| 17:35.00 | hmmhesays | I guess not |
| 17:35.02 | *** join/#asterisk VoipForces (n=kvirc@office.privalodc.com) |
| 17:35.07 | Linuturk | today I had problems with my Digium T1 card. it apparently stopped sending signaling after a power outage this morning. I had to open my server and reseat the card to get it working again. Should I look at replacing said card/server? |
| 17:35.15 | dbrotman | You headed off my question, thanks! |
| 17:35.27 | Linuturk | or, was it just a fluke |
| 17:35.28 | VoipForces | Hi, anyone knows a way to identify a call is a transfer, dialplan wise? |
| 17:35.30 | e0n` | hmmm FreePBX is a PITA, it's more confusing than configuring my DID's and extensions manually |
| 17:35.36 | e0n` | Is trixbox any more direct? |
| 17:35.54 | VoipForces | e0n`: trixbox uses freePBX |
| 17:36.25 | VoipForces | e0n`: what do you find confusing in setting up extension and did in freepbx ? |
| 17:36.58 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 17:36.59 | VoipForces | Linuturk: I would replace it. |
| 17:37.05 | e0n` | VoipForces: Not extensions, just overall configuration, I can setup a trunk from command line add a phone and a route incoming outgoing in 1/8th the time it takes for freepbx to even get it to work heh |
| 17:37.39 | [TK]D-Fender | e0n`: sO GO DO IT ON YOUR OWN THEN |
| 17:37.46 | VoipForces | e0n`: Yeah, but think about day-to-day maintenance. Settingup extensions, did, ivr in freepbx is brainless |
| 17:38.13 | e0n` | [TK]D-Fender: cut the frustration out, was looking for recommendations, not explosions ;) |
| 17:39.26 | hmmhesays | so there is no way to listen for a fax tone in early media with asterisk? |
| 17:39.32 | [TK]D-Fender | e0n`: that was a recommendatio, and 1 clearly misplaced capslock key |
| 17:39.58 | e0n` | ah, understood |
| 17:40.04 | VoipForces | e0n`: I have looked at a few gui and freepbx is by far superior to anything else. And it still allows you to do custom cord. For exemple I just implemented Nortel SMDR serial port without so much breaking freepbx |
| 17:40.33 | VoipForces | e0n`: That was custom code, not custom cord LOL |
| 17:40.45 | e0n` | VoipForces: lol |
| 17:41.42 | VoipForces | Anyone knows of a variable that is set when someone does a blind or announced transfer? |
| 17:42.34 | kaldemar | VoipForces: BLINDTRANSFER, see channelvariables.tex in src |
| 17:42.39 | dacs | [TK]D-Fender: http://pastebin.ca/1475671 |
| 17:43.03 | e0n` | Weird though, I have the Dial Rules: in the trunk and the outbound routes, but it seems outbound routes are ignored some how |
| 17:43.05 | ehsjoar | VoipForces: What I never understood is that it seems FreePBX is still writing everything out as files. They already have everything stored in the database, why don't they use Asterisk Realtime Static? I guess they are just use to it |
| 17:43.12 | VoipForces | kaldemar: Yeah, but what about announced transfer (transfer+extension+dial, bla bla, transfer |
| 17:43.35 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:44.25 | VoipForces | ehsjoar: I find configuration files far more flexible that realtime configuration. It allows you to do custom code. Every context created by freepbx has an equivalent context-custom that included so you can override everything. |
| 17:44.26 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 17:44.26 | dacs | [TK]D-Fender: my issue is that i am not able to hear sounds |
| 17:46.04 | *** part/#asterisk simNIX (n=simNIX@156-60.bbned.dsl.internl.net) |
| 17:46.28 | ehsjoar | VoipForces: I agree with that the capability to enter custom code is essential. However, in a Realtime Static environment you can in the database add an include statement that will still allow you to create your on files in /etc/asterisk and they will be included. It is really not a big issue, but it forces them to maintain consistency between the file system and the database. Extra headache it seems to me |
| 17:47.48 | *** join/#asterisk wierdo (n=jimmy@wifi-traf5.networx-bg.com) |
| 17:47.53 | VoipForces | ehsjoar: Indeed, and freepbx is a lot of line of code and changing from database->confgeneration concept to realtime would be a big dev effort. |
| 17:48.32 | VoipForces | ehsjoar: The only file that I know freepbx handles directly (read/write to/from it) is the voicemail onec. |
| 17:48.37 | VoipForces | ones. |
| 17:48.45 | ehsjoar | VoipForces: I think this is the very reason. Still, FreePBX is a great product |
| 17:51.53 | *** join/#asterisk cheetr (n=tom@oh.you.can.do.such.a.lot.with.a.wompom.com) |
| 17:52.17 | VoipForces | kaldemar: ${BLINDTRANSFER} is empty on an announced transfer, but indeed contains the value on a blind transfer... |
| 17:52.23 | cheetr | PRI related question: What does PRI got event: HDLC Abort (6) on Primary D-channel of span 1 mean, and why am i getting it? |
| 17:52.32 | cheetr | i can provide files if people want to see them |
| 17:52.59 | dacs | when i dial 123 from zoiper i don't hear anything , but when i do from x-lite i can hear the 'enter-ext-of-person' |
| 17:53.13 | dacs | brb |
| 17:53.22 | VoipForces | cheetr: This thread might provide some answerd: http://lists.digium.com/pipermail/asterisk-users/2005-March/085849.html |
| 17:53.23 | cheetr | would a rt-kernel help? |
| 17:54.20 | cheetr | VoipForces: it's asterisk 1.6, latest dahdi, and a E110P openvox card |
| 17:54.26 | cheetr | so pretty new hardware and packages |
| 17:54.50 | cheetr | ubuntu jaunty 2.6.28-11-generic |
| 17:55.10 | VoipForces | cheetr: you sure you have the line coding and protocol right to what the provider is set up to? |
| 17:55.24 | cheetr | yeah, it matches the config on the working box |
| 17:55.54 | VoipForces | cheetr: Is the card sharing interrupt with an other device? A PRI card generated 1000 IRQ per seconds |
| 17:56.48 | kaldemar | VoipForces: i guess there's no such variable for attended transfers. maybe a features conf entry and an own variable could achieve that? |
| 17:56.56 | cheetr | i'm not sure how to test that |
| 17:58.04 | cheetr | in proc/interrupts : 25: 28387766 1539149635 IO-APIC-fasteoi wcte11xp |
| 17:58.20 | [TK]D-Fender | dacs: Stop masking your output |
| 17:59.14 | [TK]D-Fender | dacs: And include your sip.conf masking ONLY passwords |
| 18:00.42 | *** join/#asterisk tfrew (n=tfrew@75-145-244-108-Richmond.hfc.comcastbusiness.net) |
| 18:02.00 | VoipForces | cheetr: IO-APIC-fastio... sounds like a cheap chipset. I would try changing slots to see. |
| 18:03.02 | VoipForces | kaldemar: That means having peoples do announced transfer in a non standard way which throws my documentation away. I'll check the sip header to see if I could not spot something. |
| 18:03.07 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.103.107) |
| 18:03.13 | DelphiWorld | hello |
| 18:03.19 | cheetr | VoipForces: it's a HP dl145 G3 |
| 18:03.21 | DelphiWorld | please is it pocible to know the asterisk channel op? |
| 18:04.11 | VoipForces | cheetr: Ok, then it;s not a cheap chipset. |
| 18:04.33 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-419fea44ad1a98e5) |
| 18:04.40 | VoipForces | cheetr: when you say latest asterisk, you mean latest released one not trunk right? |
| 18:06.34 | VoipForces | kaldemar: I get message like "SIP transfer to extension 1000@from-internal-xfer by 223@192.168.2.83" so I guess I could patch asterisk for a new variable... |
| 18:07.11 | cheetr | VoipForces: yeah, latest release. |
| 18:07.56 | *** join/#asterisk jedaustin (n=chatzill@168.158.190.111) |
| 18:08.18 | VoipForces | cheetr: ok... here is an other interesing post: http://lists.digium.com/pipermail/asterisk-users/2005-June/107504.html |
| 18:08.30 | VoipForces | cheetr: Try installing the card in an other slot. |
| 18:09.08 | kaldemar | VoipForces: sure, but it's always nicer to do things with configuration than by patching the source. :P |
| 18:10.28 | VoipForces | kaldemar: I know. I'll look at the source code and see if I could not find an identifier. |
| 18:10.42 | DelphiWorld | #asterisk op: please send to me a message i need you |
| 18:10.55 | Qwell | DelphiWorld: what? |
| 18:12.20 | DelphiWorld | Qwell: i have a private question ok? |
| 18:12.44 | Qwell | fine, msg me |
| 18:12.53 | *** join/#asterisk mykhyggz (n=mykhyggz@evolone.org) |
| 18:13.04 | DelphiWorld | Qwell: thanks |
| 18:13.22 | VoipForces | cheetr: Here is an other link: http://www.ctunion.com/node/95 |
| 18:14.25 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.103.107) |
| 18:14.44 | cheetr | VoipForces: cool, i'll try that now i think |
| 18:15.01 | DelphiWorld | Qwell: see my PM |
| 18:15.02 | cheetr | i'm starting to wish my employer would just ditch the PRI nonsense and get a massive SIP trunk in |
| 18:15.49 | Qwell | okay? I see no question... |
| 18:16.00 | *** join/#asterisk davevg-btwtech (n=davevg__@67.76.177.147) |
| 18:16.22 | DelphiWorld | Qwell: ok |
| 18:16.48 | VoipForces | cheetr: And you will run into other probem like sip provider going down, route that sucks,... |
| 18:17.36 | Qwell | DelphiWorld: no and no. |
| 18:17.55 | Qwell | infobot: tell DelphiWorld about help |
| 18:18.49 | [TK]D-Fender | cheetr: PRI isn't nonsense..... |
| 18:19.11 | [TK]D-Fender | cheetr: If your card has issues you should call support. |
| 18:19.19 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 18:22.32 | carrar | voip is nonsense!! |
| 18:22.34 | Qwell | DelphiWorld: You can stop messaging me now. |
| 18:22.38 | *** join/#asterisk davevg-btwtech (n=davevg__@67.76.177.147) |
| 18:23.08 | carrar | What can you say over voip that you can't say with smoke signals! |
| 18:23.23 | Qwell | carrar: "Help, I don't have and firewood!" |
| 18:23.26 | DelphiWorld | Qwell: thanks |
| 18:23.26 | Qwell | any* |
| 18:23.28 | carrar | haha |
| 18:26.52 | cheetr | VoipForces: i'm leaving my current employer in 20 days, i have that long to get it "working" |
| 18:26.57 | cheetr | whatever's the quickest. |
| 18:27.52 | cheetr | it's stupid, if they wanted a really good working phone system, it's time to invest some serious money and stop trying to pull together old ip phones and old networking gear |
| 18:27.56 | cheetr | i can't even use VLANs |
| 18:28.07 | *** join/#asterisk alrs (n=lars@dsl211-216-141.lax1.dsl.speakeasy.net) |
| 18:28.13 | VoipForces | cheetr: Well, indeed they are shooting for trouble |
| 18:28.43 | cheetr | VoipForces: pick a part of infrastructure and i'd tell you how underspent it is |
| 18:28.51 | [TK]D-Fender | cheetr: Whosays you need VLANS? |
| 18:29.14 | cheetr | i say i want them. |
| 18:29.25 | VoipForces | cheetr: For teh future, go with Sangoma cards. |
| 18:29.27 | [TK]D-Fender | [14:23]<Qwell>carrar: "Help, I don't have and firewood!" <- brilliant |
| 18:30.04 | eppigy | VLAN YOUR FACE |
| 18:30.23 | cheetr | VoipForces: i would have, but they were 2x more expensive at least |
| 18:30.30 | cheetr | and.. really.. i mean it when i say this |
| 18:30.36 | cheetr | the cheapest option gets it. |
| 18:30.52 | eppigy | horrible employer |
| 18:30.58 | eppigy | I will quit the first time |
| 18:31.04 | eppigy | I see that behavior |
| 18:31.08 | cheetr | why do you think i'm leaving? |
| 18:32.54 | cheetr | i put up with it for a year |
| 18:32.54 | eppigy | yes |
| 18:33.06 | cheetr | when i presented them with a choice of buying some HP gear for webservers |
| 18:33.06 | eppigy | Like "What is the IT budget?" |
| 18:33.08 | eppigy | "ummmmmmmm" |
| 18:33.08 | eppigy | No thanks |
| 18:33.08 | cheetr | eppigy: like 8k/year or so |
| 18:33.08 | eppigy | haha |
| 18:33.09 | cheetr | should be like 80k |
| 18:33.09 | eppigy | that is upsetting |
| 18:33.10 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:33.14 | cheetr | yeah |
| 18:33.18 | cheetr | it's pretty frustrating |
| 18:33.30 | cheetr | i think instead of buying HP servers, they bought Acer |
| 18:34.06 | VoipForces | cheetr: and they expect the phone system to work 100% of the time 24 hours a day...??? |
| 18:34.42 | VoipForces | cheetr: Tell how they are so we do not send resume to them LOL |
| 18:35.22 | [TK]D-Fender | Acer.... makes servers? |
| 18:35.50 | *** join/#asterisk ircnickiuse (n=xxx@gw4.mycosmos.gr) |
| 18:35.59 | ircnickiuse | hey, anyone using pjsip or sipp or sipsak ? |
| 18:36.16 | *** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net) |
| 18:36.22 | ircnickiuse | want to know which is best for testing latency issues for sip control traffic from a console - with sane command line |
| 18:36.29 | cheetr | VoipForces: yeah, hehe |
| 18:36.36 | cheetr | VoipForces: i'll tell you after i leave :P |
| 18:36.53 | cheetr | [TK]D-Fender: yeah, they do, apparently. |
| 18:37.27 | cheetr | what's astcanary btw? |
| 18:37.58 | [TK]D-Fender | cheetr: Another bird-brained idea |
| 18:38.49 | cheetr | heheh what is it supposed to do? |
| 18:40.13 | *** join/#asterisk davevg (n=davevg-b@67.76.177.147) |
| 18:40.28 | ReDNeQ- | dahdi and sangoma not playing welll |
| 18:40.48 | VoipForces | ReDNeQ-: What do you mean? I did not have problems so far |
| 18:41.04 | ReDNeQ- | well im a moron and i cant get * to see card |
| 18:41.21 | VoipForces | ReDNeQ-: Ok, first do you see the card when you do lspci ? |
| 18:41.21 | ReDNeQ- | loaded / reloaded tried tarball wanpipe nad rpms |
| 18:41.43 | ReDNeQ- | lscpi? |
| 18:42.04 | VoipForces | ReDNeQ-: Yes from the linux cli it will list your pci devices |
| 18:42.15 | ReDNeQ- | no im sorry |
| 18:42.22 | ReDNeQ- | thought you were *cli |
| 18:42.26 | VoipForces | ReDNeQ-: You should see something like "01:01.0 Network controller: Sangoma Technologies Corp. A101 single-port T1/E1 (rev 01)" |
| 18:42.45 | VoipForces | ReDNeQ-: So you do not see the sangoma card in lspci ? |
| 18:42.45 | ReDNeQ- | ahh fpound it |
| 18:42.46 | ReDNeQ- | hah |
| 18:43.11 | ReDNeQ- | 06:00.0 Network controller: Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card |
| 18:43.46 | VoipForces | ReDNeQ-: Ok, a A200, did you connect the hdd type power connector to the back of the card ? |
| 18:43.57 | VoipForces | ReDNeQ-: And installed your fxs or fxo modules? |
| 18:43.58 | ReDNeQ- | yep |
| 18:44.15 | VoipForces | ReDNeQ-: what does "wanrouter hwprobe" tells you ? |
| 18:44.38 | ReDNeQ- | no wanrouter |
| 18:44.45 | ReDNeQ- | cant find wanrouter |
| 18:44.50 | ReDNeQ- | command that is |
| 18:45.50 | ircnickiuse | guys, do you remember the Michael Jackson Ubituary printed by the NYT? The one saying that all his life Michael Jackson has wished people used Sipsak, pjsip or sipp and shared their experiences on irc? |
| 18:45.55 | VoipForces | and you installed wanrouter? |
| 18:46.03 | VoipForces | ReDNeQ-: You did install wanpipe ? |
| 18:46.05 | jtodd | Back. We have a press request for University/higher-ed users of Asterisk. Contact me via email (jtodd@digium) if you have leads. |
| 18:46.16 | *** join/#asterisk ikks (n=ikks@190.144.69.234) |
| 18:46.19 | ReDNeQ- | VoipForces i think its time i reinstall it |
| 18:46.21 | ReDNeQ- | bbiab |
| 18:46.35 | ReDNeQ- | i htink i got out of order with all these tries |
| 18:47.05 | VoipForces | ReDNeQ-: Follow this to the letter: http://wiki.sangoma.com/wanpipe-linux-asterisk-dahdi |
| 18:47.52 | cheetr | weeeell |
| 18:48.01 | cheetr | i changed the slot the card was in |
| 18:48.09 | cheetr | still getting hdlc bad fcs errors |
| 18:49.00 | VoipForces | cheetr: Before it was HDLC Abort right? |
| 18:49.14 | cheetr | bit of both |
| 18:49.51 | VoipForces | cheetr: Really sounds like a chipset/card incomptability... If I were you my next step would be to contact the card manufacturer. |
| 18:50.20 | cheetr | bah :( |
| 18:50.59 | VoipForces | cheetr: Let me guess they bought it used on ebay? |
| 18:51.08 | cheetr | nah |
| 18:51.13 | cheetr | new from voipon.co.uk |
| 18:54.56 | cheetr | it seems now that i can make and recieve calls |
| 18:54.59 | cheetr | but the quality is shit |
| 18:55.05 | cheetr | it's really echoy |
| 18:55.11 | cheetr | and sounds like i'm talking under water |
| 18:56.41 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 18:56.44 | cheetr | after all, the pri stuff is just warning and notice level alerts |
| 19:00.41 | VoipForces | cheetr: I guess that card does not have hardware echo canceler? |
| 19:00.55 | cheetr | i don't know |
| 19:01.11 | VoipForces | cheetr: hmmm you bought it in the uk, they use BRI there? Are you in the UK or America? |
| 19:01.13 | *** join/#asterisk bryanfe2 (n=chatzill@wsip-72-215-161-251.sb.sd.cox.net) |
| 19:01.20 | cheetr | i'm in the uk |
| 19:01.24 | cheetr | it's definitely PRI tho |
| 19:01.36 | cheetr | i'm in work, and it's 8pm :( |
| 19:01.38 | cheetr | echocanceller=mg2,1-15,17-31 |
| 19:01.45 | cheetr | is in system.conf |
| 19:01.51 | bryanfe2 | is there an asterisk app or sample code for an app, which will show how to send a solid tone (i.e. "A" 440 hz for 1 second)? |
| 19:01.54 | cheetr | but i've tried comenting that out |
| 19:01.56 | cheetr | and that doesn't work |
| 19:02.08 | cheetr | bryanfe2: milliwatt? |
| 19:02.24 | cheetr | or has that been removed |
| 19:02.29 | cheetr | my asterisk book is way old. |
| 19:02.55 | bryanfe2 | cheetr I see it in the sources, worth a shot ;) thank you |
| 19:03.13 | cheetr | it's not A tho, it's 1khz |
| 19:03.17 | cheetr | according to this :P |
| 19:03.40 | bryanfe2 | Looks like it uses another app called Playtones |
| 19:03.57 | *** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7) |
| 19:05.23 | VoipForces | cheetr: I stopped worrying about echo when I started using hardware echo canceler card... what 5 years ago LOL |
| 19:05.51 | jplank | I wish that was the case with one of my clients. I have a client who's had echo ever since we upped him to dahdi |
| 19:05.55 | jplank | w/ hw echo can |
| 19:06.53 | VoipForces | jplank: what card? |
| 19:07.56 | *** join/#asterisk Jacke (i=jacke@valhalla.bofh.org) |
| 19:09.57 | VoipForces | cheetr: I believeyou have to restart asterisk when playing with soft echo stuff. |
| 19:10.03 | cheetr | mmm |
| 19:10.09 | cheetr | i've been restarting it every time |
| 19:10.10 | cheetr | ooo |
| 19:10.49 | cheetr | i just opened 4 lines on a test track using MP3Player |
| 19:10.56 | cheetr | and after 2 mins or so |
| 19:10.58 | cheetr | it cut me off |
| 19:11.08 | cheetr | [2009-06-26 20:10:26] WARNING[6465]: chan_dahdi.c:4664 handle_alarms: Detected alarm on channel 1: Yellow Alarm |
| 19:11.16 | VoipForces | cheetr: PRI or SIP alone? |
| 19:11.23 | VoipForces | cheetr: Ok |
| 19:11.24 | cheetr | PRI |
| 19:11.34 | cheetr | i used the sip phones to dial out and back in again |
| 19:11.48 | cheetr | all the sip stuff is fine, as long as it doesn't go anywhere near the real world. |
| 19:11.55 | VoipForces | cheetr: PRI xover cable? |
| 19:12.09 | cheetr | i don't know what those look like, sorry. |
| 19:12.24 | cheetr | it's just connected to the wall socket with a ethernet patch cable. |
| 19:13.02 | ariel_ | in 20 minutes it's Ice Cream party here at work....hummm I like... |
| 19:13.03 | VoipForces | cheetr: Oh ok. hmmm how long is the cable going from your card to the PRI demarc box? |
| 19:13.19 | cheetr | uh, i guess it's about 30ft |
| 19:13.19 | boch | can i use an outbound proxy when registering a peer ? |
| 19:13.25 | cheetr | i could try and find a short one |
| 19:13.27 | VoipForces | ariel_: IceCream? Friday if for beer ! |
| 19:13.38 | Aiatek | <PROTECTED> |
| 19:13.48 | VoipForces | cheetr: That's long. I always try to keep it like to 6 feet |
| 19:14.05 | e0n` | how can I force or try to from cli get a sip channel to register |
| 19:14.11 | *** join/#asterisk Sajam (n=chatzill@89.108.179.214) |
| 19:14.11 | ariel_ | well it could be for Beer, but I work for a large Telecom co and they don't do beer in the lunch room... only Ice Cream.... |
| 19:15.12 | VoipForces | e0n`: You can not |
| 19:15.12 | ariel_ | e0n`: force |
| 19:15.12 | cheetr | i'll go try find a short one |
| 19:15.12 | ariel_ | sip reload might get it back started |
| 19:15.13 | VoipForces | cheetr: Just a hint, not mean it's the problem |
| 19:15.29 | Sajam | Hello guys, i am newbie in using asterik |
| 19:16.02 | Sajam | i want to install it, what is the best OS for asterisk?? |
| 19:16.02 | ariel_ | cheetr: how long is the cable? |
| 19:16.02 | e0n` | getting all circuits are busy, and noticed the state of my sip provider is Unregistered |
| 19:16.12 | ariel_ | Sajam: which one do you like best? |
| 19:16.25 | Sajam | freebsd |
| 19:16.29 | [TK]D-Fender | e0n`: You don't need to be registered to place calls |
| 19:16.34 | e0n` | got it |
| 19:16.47 | ariel_ | register is for incoming from them |
| 19:16.47 | cheetr | ariel_: the old one was about 10m long, new one about 6' |
| 19:16.55 | Sajam | ariel: Freebsd |
| 19:17.06 | ariel_ | 10m = 33 feet. |
| 19:17.08 | ariel_ | about |
| 19:17.25 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 19:17.33 | [TK]D-Fender | Sajam: * was oriented more towards Linux, but the FreeBSD ports are fairly well maintained |
| 19:17.43 | ariel_ | Sajam: it works on that one, but my choice is CentOS or Debian. |
| 19:17.51 | cheetr | ariel_: yeah |
| 19:17.56 | cheetr | it's no better with a shorter cable |
| 19:18.12 | ariel_ | cheetr: that is ok as long as it's twited cat 5e |
| 19:18.18 | cheetr | yeah |
| 19:18.25 | cheetr | it's just a regular network cable |
| 19:18.34 | cheetr | what was that about a pri x-over? |
| 19:18.42 | *** join/#asterisk ReDNeQ- (n=ReDNeQ@75-148-217-225-Houston.hfc.comcastbusiness.net) |
| 19:19.05 | Sajam | ariel: Cool, i like debian, anyway what version of asterisk do you recommend, and do you have a link for documentation on installing asterisk ?? |
| 19:20.12 | ariel_ | Sajam: depends on what you want to do with it. |
| 19:20.25 | ariel_ | I still use 1.4.25 have not had a reason to go to 1.6.x |
| 19:20.50 | ariel_ | I use Debian Lenny mostly as well |
| 19:21.00 | VoipForces | cheetr: http://www.voip-info.org/wiki/view/T1+Crossover+Cable |
| 19:21.21 | Sajam | ariel: i need to try it inside the company for testing first, i want to replace Cisco products |
| 19:21.39 | [TK]D-Fender | Sajam: 1.6.0.X latest release is a good bet |
| 19:22.26 | Sajam | [TK]D-Fender: Great, what about document for installing it on debian, is there any?? |
| 19:22.29 | cheetr | VoipForces: i'd be surprised if that were the case, (needing a pri x-o cable) as the current working phone server (albeit, 8 year old, and running on a dell beigebox).. jsut connects with a regular patch lead. |
| 19:22.31 | ariel_ | Sajam: wow, I have about over 100 Cisco's here, and also use asterisk |
| 19:22.38 | [TK]D-Fender | Sajam: In the source tarball |
| 19:23.15 | *** join/#asterisk kfife (n=Miranda@home.chicagoventure.com) |
| 19:23.18 | VoipForces | cheetr: You set one span as pri_net and the other one as pri_cpe and you can simulate call passing through the 2 spans |
| 19:23.26 | *** part/#asterisk kfife (n=Miranda@home.chicagoventure.com) |
| 19:23.44 | Sajam | ariel: do you recommend it over cisco? also i want to share in developing it |
| 19:23.50 | cheetr | new error.. |
| 19:23.51 | cheetr | [2009-06-26 20:23:36] ERROR[7358]: chan_dahdi.c:10515 dahdi_pri_error: !! Got a UA, but i'm in state 7 |
| 19:23.59 | cheetr | VoipForces: ahhh |
| 19:24.35 | ariel_ | Sajam: depends on what your doing |
| 19:24.53 | ariel_ | I use Cisco's as pri/E1 to h323 gateways mostly |
| 19:25.48 | Sajam | ariel: Ok, am cisco router 1760, do you think i can use with asterisk |
| 19:25.50 | Sajam | ? |
| 19:25.54 | *** join/#asterisk korihor (n=korihor@190.77.92.57) |
| 19:25.58 | ariel_ | Cisco's are good product, But end points to sip we are replacing them with Asterisk setups, Even the big 5300's but for E1 to tdm or h323 there better. |
| 19:26.37 | ariel_ | 1760 is a small Router what are you running on it? |
| 19:26.52 | cheetr | i think i'm gonna give up for tonight |
| 19:27.30 | Sajam | ariel_: beside it, there's call manager 4.0 and unity voice mail, there's only 50 users here |
| 19:27.43 | DelphiWorld | thank to the infobot developer! is contin mor informations for me! |
| 19:28.14 | [TK]D-Fender | DelphiWorld: It only knows what people in here have trained it for |
| 19:28.29 | ariel_ | Sajam, what phones? |
| 19:28.47 | cheetr | thanks for all your help VoipForces, and co |
| 19:28.49 | ariel_ | I have only 1 minute, it's time to go to the Ice Cream party... |
| 19:28.59 | cheetr | but i'm starting to think this is a losing battle without greater investment. |
| 19:29.01 | Sajam | ariel_: 7940, 7960 |
| 19:29.12 | ariel_ | are you running on them SCCP |
| 19:29.27 | cheetr | one quick question, would it be possible to keep the old asterisk box to do the PRI stuff, then use the new one for the call handling? |
| 19:29.36 | Sajam | ariel_: it's great pleasure to talk to you, why are you going now?? |
| 19:29.39 | DelphiWorld | [TK]D-Fender: how to add other informations to it? |
| 19:29.45 | ariel_ | free food |
| 19:29.49 | ariel_ | ;-) |
| 19:29.53 | ariel_ | bbl |
| 19:30.40 | [TK]D-Fender | DelphiWorld: Like what? |
| 19:30.47 | Sajam | ariel_: lol, eat well, we need to know more info from you later on :P |
| 19:31.30 | *** join/#asterisk timeshell_atwork (n=chatzill@gw.lusi.on.ca) |
| 19:31.44 | DelphiWorld | [TK]D-Fender: open source projects related to asterisk, SIP Routers/PROXY/... |
| 19:32.32 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:32.33 | *** join/#asterisk flrichar (n=toril@tor.ipv6.geexology.org) |
| 19:33.11 | flrichar | any good sites to recommend reading up on proxy authentication? |
| 19:33.12 | *** join/#asterisk oej (n=olle@135.Red-88-2-201.staticIP.rima-tde.net) |
| 19:33.13 | [TK]D-Fender | DelphiWorld: There are pleny of quick-info bits already, but do not mistake infobot as a BOOK that you just pile tomes into |
| 19:33.51 | DelphiWorld | [TK]D-Fender: ok |
| 19:35.18 | VoipForces | cheetr: Yes it's possibl;e. It's a technique I used to slowlly migrate away from a Nortel PBX. |
| 19:36.58 | *** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net) |
| 19:38.32 | *** part/#asterisk flrichar (n=toril@tor.ipv6.geexology.org) |
| 19:39.23 | *** join/#asterisk makafre (n=makafre@64.86.141.133) |
| 19:39.39 | makafre | hi all |
| 19:40.04 | makafre | anybody knows how to use queue_members (realtime)? |
| 19:40.33 | suma | I could not pickup the parked calls, someone please help |
| 19:41.20 | makafre | i was wondering what's the relationship between AddQueueMember and queue_members table? |
| 19:41.46 | suma | queue_members are static members |
| 19:42.01 | suma | AddQueueMember is dynamic |
| 19:42.30 | suma | There is no relationship for what you are talking about |
| 19:42.37 | suma | both are seperate entities |
| 19:42.42 | cheetr | VoipForces: have you had any luck using an rt-kernel? |
| 19:43.09 | makafre | suma: ok, that means queue_members static entries still need to be logged in? using which function? |
| 19:43.52 | suma | AgentLogin |
| 19:44.03 | makafre | suma: sure, ok |
| 19:44.47 | [TK]D-Fender | makafre: No. |
| 19:44.48 | suma | app_directed_pickup.c:159 pickup_exec: No target channel found for 701. <-- why I get this when I have a call parked at the extension 701 |
| 19:45.06 | [TK]D-Fender | makafre: Static memeber are ones you don't have to add Dynamically and are expected to alway be members |
| 19:45.39 | [TK]D-Fender | suma: What are you doing trying to do a directed PICKUP instead of just dialing the parking extension? |
| 19:45.44 | cheetr | right. i'm off. . might have another go on monday! |
| 19:45.55 | *** part/#asterisk cheetr (n=tom@oh.you.can.do.such.a.lot.with.a.wompom.com) |
| 19:46.00 | makafre | suma: sure, and is that table read only from the asterisk point of view, or can it update it for some special reasons? |
| 19:46.07 | [TK]D-Fender | suma: And you should pastebin a failed call attempt for us to look at. Mid you your current methodology is wrong regardless |
| 19:46.26 | [TK]D-Fender | makafre: * does not write to those tables |
| 19:47.37 | suma | [TK]D-Fender: http://pastebin.ca/1475804 |
| 19:48.52 | makafre | tkd-fender: If I implement hot desking, can I leave out the interface field to NULL and just use the membername for the AgentLogin purpose? I would use a SIP extensions as the member name |
| 19:48.56 | [TK]D-Fender | suma: You do not PICKUP a parked call like that. You dial the lot # that it was parked in |
| 19:49.27 | suma | How would the dial parameters would be ? |
| 19:49.28 | [TK]D-Fender | makafre: No, the interface must be specified. |
| 19:49.38 | [TK]D-Fender | suma: No parameters, just dial the # |
| 19:50.42 | makafre | tkd-fender: so is there a way to implement hot desking if the interface is tied to the membername ? |
| 19:51.02 | [TK]D-Fender | makafre: Membername has nothing to do with the device |
| 19:51.15 | [TK]D-Fender | mafaYou can call the memebername "FRED" and * won't care |
| 19:52.33 | suma | [TK]D-Fender: That's awesome, worked great |
| 19:52.44 | makafre | tkd-fender: but isnt AgentLogin validating against this table? sorry, I m totally lost, there is a little something that I am missing |
| 19:53.00 | suma | [TK]D-Fender: What is the Pickup is for ? |
| 19:53.21 | [TK]D-Fender | suma: Go read its instrucitons |
| 19:54.05 | [TK]D-Fender | makafre: No. Agents status and login has nothing to do with what Queues they completely coincedentally happen to be a member of. |
| 19:56.39 | eppigy | D: |
| 19:56.46 | suma | [TK]D-Fender: Ringing channel, how can I make a channel to be ringing until I get someone to pick up ? |
| 19:57.22 | [TK]D-Fender | suma: What are you trying to do here? We already seem to have gotten Parking functional... |
| 19:57.46 | suma | Yes, that is working fine. I'm trying to understand the need of Pickup |
| 19:58.25 | [TK]D-Fender | suma: Your coworker's phone is ringing and you want to grab the call without killing yourself running across a mess office |
| 19:58.40 | makafre | tkd-fender: ok, well, let's restart then: I need to implement realtime hotdeksing, and If I start with the agents, where are they defined? |
| 19:58.50 | [TK]D-Fender | suma: "Imaginations" are in the bin to the left of the door... |
| 19:59.06 | suma | :) that sounds clear. |
| 19:59.53 | [TK]D-Fender | makafre: If you want hotdesking, this is pure dialplan and you should drop the thought of Agents altogether |
| 20:00.16 | makafre | tkd-fender: can you elaborate? |
| 20:01.12 | [TK]D-Fender | makafre: Do your own login purely in the dialplan yourself. *'s use of "Agents" was only for queues, and as fo 1.6 the only way to log in as one is with AgentLogin, which forces them to sit on an active call. |
| 20:03.35 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:06.04 | makafre | tkd-fender: hum, so how you tell the queue your agent is now logged in with a housemade login procedure from the dialplan? |
| 20:06.54 | Greek-B0y | i thought the "m" option in the Dial command is supposed to start playing the song in the class from the beginning when called from a macro? IT doesn't do that on my system... |
| 20:08.18 | [TK]D-Fender | makafre: Hostdesk does not imply QUEUES. |
| 20:08.26 | [TK]D-Fender | makafre: And where would you look for login status? |
| 20:08.54 | makafre | tkd-fender: hotdesk means an agent can login in a queue from any phone, right |
| 20:08.59 | [TK]D-Fender | Greek-B0y: m() has nothing to do with macro. It replaces RINGING with MoH |
| 20:09.29 | makafre | tkd-fender: using his agent id |
| 20:09.47 | [TK]D-Fender | makafre: No, it jsut IDs the user associated with a given device. You can have hot-desking where "John" could be sitting at any desk but not have anything to do with queues |
| 20:10.07 | [TK]D-Fender | makafre: Completely drop the term AGENT unless you're prepared to use AgentLogin. |
| 20:10.37 | makafre | tkd-fender: :-) well, I need agents to answer calls coming in from the queue |
| 20:10.57 | Greek-B0y | [TK]D-Fender: Yeah I know but someone here had said that if you call it from a macro it should start playing the song from the beginning. Do you know of any way to achieve that? |
| 20:11.11 | [TK]D-Fender | makafre: You can use AddQueueMemeber to add them users however you like. |
| 20:11.31 | [TK]D-Fender | Greek-B0y: macro has nothing to do with MoH isntead of ringing. |
| 20:13.01 | Greek-B0y | ok. Anybody know how to make the song play from the beginning with each new call? |
| 20:15.53 | makafre | tkd-fender: all right, what ill probably do then is create a home made table for the agents so I can put in there which queues they each pertain to, and from there whenever they try to login with the correct credentials, I will check against that table, and log them in the queues using AddQueueMember on each of the queue i find in there. Does that make sense at all? |
| 20:19.23 | [TK]D-Fender | makafre: Yes, you seem to be catching on |
| 20:22.55 | makafre | tkd-finally: Which also means that I can still populate the queue_members table, and can use the AgentLogin function to perform the same thing right? But for that option what I am not getting is the interface field? how does that affect the AgentLogin function or how does it use it? An example? |
| 20:26.16 | makafre | When I think interface I see: IAX2/123456 as an example |
| 20:27.06 | *** join/#asterisk WHYS (n=drumm@137.28.94.209) |
| 20:27.41 | WHYS | Is there a backport to allow sip or tcp in 1.4? |
| 20:29.50 | jaytee | WHYS, no there is not at this time and I don't believe there are any plans to "backport" it to 1.4 |
| 20:29.55 | [TK]D-Fender | makafre: AgentLogin only logs that call in as an "agent". It does not make them a member of a queue |
| 20:30.32 | makafre | ahhhhhhhhhhhhhh |
| 20:30.33 | *** join/#asterisk tfrew (n=tfrew@75-145-244-108-Richmond.hfc.comcastbusiness.net) |
| 20:30.36 | WHYS | sighs |
| 20:30.41 | makafre | ;-) |
| 20:31.21 | makafre | tkd-fender: all right, I ll implement first solution, that will be great |
| 20:31.51 | makafre | thanks |
| 20:35.03 | makafre | ! |
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| 20:35.05 | *** mode/#asterisk [+o denon] by ChanServ |
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| 20:41.24 | *** join/#asterisk infobot (i=ibot@rikers.org) |
| 20:41.24 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.1 (2009/06/05), Asterisk 1.6.0.10 (2009/06/05), 1.4.25 (2009/06/05), *-Addons 1.6.1.0 (2009/04/28), 1.6.0.2 (2009/05/21), 1.4.8 (2009/04/28), dahdi-linux 2.1.0.4, dahdi-tools 2.1.0.3 (2009/02/03), Libpri 1.4.10 (2009/04/18) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev #asterisk-commits |
| 20:45.40 | *** join/#asterisk iratik (n=root@74-84-99-12.client.mchsi.com) |
| 20:47.03 | iratik | This may be a dumb question, but if there are 2 network interfaces , which one does asterisk use for outbound connections? . Ie, if i connect to broadvoice's sip servers .. which interface will be chosen? |
| 20:47.34 | Qwell | The one that has the appropriate route |
| 20:47.43 | Greek-B0y | iratik: Which is configured for WAN and has a public gateway assigned to it? |
| 20:48.04 | *** join/#asterisk AndyML (n=AndyML@pool-173-49-143-205.phlapa.fios.verizon.net) |
| 20:48.12 | iratik | eth2 |
| 20:48.32 | iratik | But how can the OS tell which one is private, which is public? |
| 20:48.40 | *** join/#asterisk dmz (n=dmz@64.203.233.195.dyn-cm-pool35.hargray.net) |
| 20:49.43 | Qwell | The one that has the appropriate route |
| 20:50.09 | *** join/#asterisk NirS (n=NirS@95.35.123.5) |
| 20:50.13 | NirS | hi all |
| 20:50.17 | NirS | how is everybody doing ? |
| 20:50.40 | iratik | Appropriate route? |
| 20:51.47 | rob0 | hmm, you'd need to understand basic IP routing |
| 20:52.17 | jaytee | quittin time, bbl |
| 20:52.22 | iratik | I'm guessing i have some holes in my understanding of basic IP routing |
| 20:53.23 | iratik | I asked in #networking and they said it would be application specific |
| 20:53.37 | rob0 | There's no difference between "private" and "public" IP addresses, but generally, a system will only have one default route. Any unknown network will be routed to through the default gateway. |
| 20:53.49 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.36.70) |
| 20:54.04 | iratik | So what determines this default route? |
| 20:54.06 | Qwell | iratik: They were wrong |
| 20:54.17 | Qwell | Your configuration. |
| 20:54.30 | rob0 | "man route" might help you get started. |
| 20:56.21 | iratik | Running route shows that given Destination default, it will use eth2 (which is the public) |
| 20:56.40 | iratik | I wish i knew how it knew to do that automatically |
| 20:57.25 | rob0 | (BTW IRC as root is not a good idea.) |
| 20:57.40 | iratik | my bad |
| 20:57.42 | *** part/#asterisk iratik (n=root@74-84-99-12.client.mchsi.com) |
| 20:58.16 | *** join/#asterisk iratik (n=ctatechs@74-84-99-12.client.mchsi.com) |
| 20:58.18 | rob0 | The kernel looks at the routing table, compares the destination IP address to what is there. |
| 20:58.20 | iratik | Why is it a bad idea? |
| 20:58.40 | iratik | Because of potential vulnerabilities i bet, but how did you know ? |
| 20:58.45 | *** part/#asterisk ikks (n=ikks@190.144.69.234) |
| 20:58.54 | rob0 | 20:45 -!- iratik [n=root@74-84-99-12.client.mchsi.com] has joined #asterisk |
| 20:59.14 | iratik | wow |
| 20:59.29 | rob0 | Actually I would say the main problem is that you get into bad habits. User software should be run as a user, administrative tools as root. |
| 20:59.42 | iratik | I know... sudo -s is evil |
| 21:00.00 | iratik | I used to get pretty frustrated with not understanding basic linux file system permissions |
| 21:00.05 | rob0 | But indeed, a vulnerability or exploit would exploit your whole system. |
| 21:00.10 | iratik | then i got into the habit of doing everything as root |
| 21:00.15 | *** join/#asterisk Aiatek (n=Alfio@190.94.43.138) |
| 21:00.47 | rob0 | Permissions are a good thing. :) When I was learning, and ran into permission problems, it forced me to learn why. |
| 21:01.14 | *** join/#asterisk telecos (n=sergio@104.166.219.87.dynamic.jazztel.es) |
| 21:01.19 | iratik | thanks for the help guys |
| 21:01.38 | rob0 | And now, look at this, I'm a Unix wizard! (He's a Unix wizard, there has to be a twist ...) |
| 21:04.47 | Nugget | <JurassicPark> This is UNIX! I know this! </JurassicPark> |
| 21:06.14 | *** join/#asterisk jplank (n=GBove@cpe-075-181-097-208.carolina.res.rr.com) |
| 21:08.30 | *** join/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 21:10.33 | pa | what does s, mean as extension number? like exten => s,blabla |
| 21:10.59 | Nugget | start of session |
| 21:11.48 | Nugget | Predefined Extension Names on http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf |
| 21:11.55 | pa | thanks |
| 21:12.17 | Nugget | The "s" extension is used when there is no known called number in the context used. |
| 21:12.26 | Nugget | used most commonly in macros |
| 21:13.37 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 21:14.11 | pa | mmh.. |
| 21:15.02 | pa | like if i have an extension 0xxxxxx,dial out, and i dial 0NUMBERIWANTTODIAL, i catch that extension, but if i dial any other number than 0 first , i would catch s, right? |
| 21:16.56 | *** part/#asterisk iratik (n=ctatechs@74-84-99-12.client.mchsi.com) |
| 21:17.01 | pa | but in my default extensions.conf, i have many different sets with exten => s, |
| 21:17.08 | pa | i dont know which one will apply then.. |
| 21:17.21 | [TK]D-Fender | Pa : it will not hit "s" |
| 21:17.25 | pa | mostly under different enviroments |
| 21:17.26 | pa | or.. |
| 21:17.28 | pa | contexts? |
| 21:17.59 | pa | [something] |
| 21:18.15 | pa | why will it not hit s? |
| 21:19.01 | *** join/#asterisk mocker (n=user@shell.mocker.org) |
| 21:19.35 | carrar | Do I smell BBQ? |
| 21:19.39 | carrar | mmmm |
| 21:19.44 | Qwell | ~bbq |
| 21:19.45 | infobot | wtf |
| 21:19.50 | Qwell | ~wtf |
| 21:19.57 | carrar | xWTFx |
| 21:19.57 | seanbright | ~fist Qwell |
| 21:19.58 | infobot | ACTION rules Qwell with an iron fist |
| 21:20.04 | Qwell | ~wtf iirc |
| 21:20.10 | Qwell | nice |
| 21:20.26 | carrar | ~wtc afk |
| 21:20.31 | *** part/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 21:20.31 | mocker | Does setting "insecure=port,invite" override "allowguest=no"? I had to set "allowguest=no" because someone was making calls on my system, but it broke my polycom phone. |
| 21:20.46 | carrar | ~wtf afk |
| 21:20.48 | carrar | heh |
| 21:20.52 | mocker | I set the "insecure=port,invite" on the polycom extension, but want to make sure that doesn't introduce the same hole |
| 21:20.56 | Qwell | ~wtf wtf |
| 21:20.59 | carrar | ~wtf afksfo |
| 21:22.27 | [TK]D-Fender | pa : because "s" is not "if it doesn't match anything land here" |
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| 21:45.35 | luch | anyone out there successfully setup sip trunk to call manager express |
| 21:45.47 | luch | i can recieve calls but not place any outgoing |
| 21:46.20 | *** join/#asterisk megabeef (n=mobile@32.166.92.5) |
| 21:47.07 | luch | i have read quite a few tutorials but i just cannot get it going - error i get is: no such extension context @default while calling local channel |
| 21:52.44 | *** join/#asterisk DarkRift (n=dark@65.92.170.121) |
| 21:54.45 | mocker | luch: Probably want to get rid of the @ ? |
| 21:54.51 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 21:54.56 | joako | I have a setup colocation server ---> public internet -> Paetch router -> Linksys router -> Asterisk server and on the colocation I constantly get a message Peer xxxx is now UNREACHABLE! Time: 80/ Peer xxxx is now REACHABLE! Time: 79 |
| 21:55.45 | joako | I thought the issue was the ancient BEFSR11 router (would not even pass SIP) but after replacing it with somthing running DD-WRT the issue persists. I don't have any problems with any of the other hosts running off that server |
| 21:56.15 | *** join/#asterisk arj_ (n=a@82-169-91-46.ip.telfort.nl) |
| 21:56.36 | [TK]D-Fender | What s the point of double-NAT? You're asking for trouble |
| 21:58.21 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 21:58.35 | joako | [TK]D-Fender: No double NAT. Paetech router terminates the T1 and is managed by the provider. DD-WRT router has a public IP |
| 21:58.59 | joako | It's a pretty standard setup with just about any T1 provider they install their router but you can not configure it yourself |
| 21:59.41 | [TK]D-Fender | joako: Would be nice to see Actual configs & debug |
| 22:00.52 | *** join/#asterisk esaym (n=user@cpe-24-174-186-34.satx.res.rr.com) |
| 22:01.36 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 22:03.31 | joako | [TK]D-Fender: Nothing out of this world: http://pastebin.ca/1475956 |
| 22:03.45 | Sargun | Single Nat? double nat? TRIPLE NAT! |
| 22:04.37 | joako | Sargun: Single nat on the remote end (the one with register => ...) No other NAT anywhere else |
| 22:05.12 | joako | FWIW I always find IAX to be a pain ("No authority found") so I try not to use it |
| 22:05.34 | Nugget | I can't wait for IPv6. :) |
| 22:07.13 | joako | I think I can hold off for a while... I'd probably have to stop smoking pot if I want to memorize IP6 addresses. |
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| 22:10.26 | *** join/#asterisk dpizarro_ (n=dpizarro@190.43.144.169) |
| 22:12.10 | dpizarro_ | good afternoon |
| 22:12.11 | dpizarro_ | <PROTECTED> |
| 22:12.11 | dpizarro_ | <PROTECTED> |
| 22:12.11 | dpizarro_ | <PROTECTED> |
| 22:12.11 | dpizarro_ | <PROTECTED> |
| 22:12.11 | dpizarro_ | <PROTECTED> |
| 22:12.13 | dpizarro_ | <PROTECTED> |
| 22:12.15 | dpizarro_ | <PROTECTED> |
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| 22:29.02 | *** join/#asterisk ACK-NAK (n=Miranda@home.chicagoventure.com) |
| 22:29.59 | ACK-NAK | Is this a clock source error or a signalling type error? "PRI Error on span 0: We think we're the network, but they think they're the network, too." |
| 22:30.27 | eppigy | clock source |
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| 22:40.12 | joako | ACK-NAK: No, it is an issue with setting pri_cpe vs pri_net. If you are connecting to a PRI Provider then you need to set pri_cpe. If you are e.g. connecting Asterisk to a PBX via PRI then set pri_net |
| 22:40.12 | dpizarro2 | does anyone know a variable/way to get the current call duration from AGI script ? |
| 22:40.27 | *** join/#asterisk mykhyggz (n=mykhyggz@evolone.org) |
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| 22:54.37 | ACK-NAK | joako: Thanks! It's confusing becasue when I set signalling to CPE i get the opposite error where both think they're CPE. |
| 22:55.29 | ACK-NAK | joako: PRI Error on span 0? It's not even telling me which span is causing the problem. I should probably disable spans to divide and conquer? |
| 23:02.05 | jaytee | ACK-NAK, what are you connecting the spans to? the PSTN? another PBX in house? |
| 23:04.54 | Aiatek | <PROTECTED> |
| 23:05.11 | joako | ACK-NAK: Is this something like XO or NuVox PRI where they install a Cisco IAD or similar? |
| 23:05.48 | Aiatek | ACK-NAK and dont forget to use a T1/E1 crossover |
| 23:06.19 | joako | Aiatek: Exactly my point. |
| 23:06.31 | Aiatek | yes |
| 23:07.02 | jaytee | I only use a crossover if I'm going straight to another PBX. I use straight cables for my PRIs from Asterisk to my Adtran CSUs. |
| 23:07.52 | joako | jaytee: It seems to be a recent trend these "integrated PRIs" they aren't PRI they install a Cisco IAD or similar and it's VoIP over the network with a PRI interface at your premisis |
| 23:08.21 | jaytee | some of them are adaptive, on demand PRI spans. Funky crap |
| 23:09.03 | Aiatek | jaytee int the scenario i wrote you need a crossover |
| 23:10.52 | jaytee | Aiatek, I've only connected Asterisk using PRI to Time Warner as our PSTN and to a Nortel M1 Option11C. I used a crossover and set my span in Asterisk to be PRI_NET to the Nortel and PRI_CPE for the other span to Time Warner. worked like a charm |
| 23:11.04 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 23:11.09 | jaytee | but the cable to the Adtran is a straight through cable |
| 23:11.18 | Aiatek | jaytee did you see my pastebin |
| 23:11.20 | Aiatek | ? |
| 23:12.26 | jaytee | so you're using E1 in europe |
| 23:12.33 | Aiatek | in that scenario you need a crossover |
| 23:12.54 | Aiatek | because you are conecting to pbx |
| 23:13.03 | Aiatek | by pri cards |
| 23:15.30 | jaytee | Aiatek, right. That's why I used an xover on the span from my Asterisk to my Nortel PBX. |
| 23:15.30 | jaytee | do you use a crossover on the other span to the PSTN? |
| 23:15.31 | Aiatek | nope i did that configuration for one of my clientsin bogota |
| 23:15.35 | Aiatek | man i was talking in the scenario i pasted here |
| 23:15.35 | Aiatek | not for all the conections |
| 23:15.48 | jaytee | yeah, that was understood |
| 23:17.03 | Aiatek | if you going to * to CSUs you can use any 568a or 568b patchcord |
| 23:17.49 | Aiatek | straight of course :) |
| 23:18.34 | jaytee | yeah |
| 23:19.07 | jaytee | but cat5 crossovers won't work on T1 or E1. different pinouts |
| 23:19.23 | Aiatek | i made my owns crossover |
| 23:19.35 | Aiatek | for that scenario |
| 23:19.40 | jaytee | 1 to 4, 2 to 5 |
| 23:20.47 | Aiatek | 12345678 |
| 23:20.51 | Aiatek | 45312678 |
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| 23:23.12 | vegbox | Has anyone here replaced a Reply Voicemail System with asterisk? |
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| 23:51.20 | *** join/#asterisk coolthreads (n=coolthre@203.97.238.71) |
| 23:53.34 | coolthreads | if I use a particular sip phone, make outbounds on this number so that that the person sees this number on there caller ID, if I use this other sip phone use this number instead |
| 23:53.52 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 23:54.22 | coolthreads | Just trying to work out how to achieve that |
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