| 07:17.06 | *** join/#asterisk infobot (ibot@rikers.org) |
| 07:17.06 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.6, 1.6.1.18, 1.6.0.26 (2010/03/12), 1.4.30 (2010/03/12), *-Addons 1.6.2.1, 1.6.1.3, 1.6.0.5, 1.4.11 (2010/04/15), dahdi-linux 2.3.0 + dahdi-tools 2.3.0 (2010/04/13), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 07:17.35 | *** join/#asterisk c0rnoTa (~c0rnoTa@178.176.226.94) |
| 07:24.09 | blaines | When testing my asterisk with a softphone, what's my SIP username? |
| 07:25.10 | blaines | 101@default? |
| 07:25.12 | ChannelZ | How should we know? |
| 07:25.17 | ChannelZ | what did you put in sip.conf? |
| 07:26.32 | blaines | err yea... is the username equal to the mailbox setting? |
| 07:26.51 | ChannelZ | no |
| 07:27.30 | blaines | ChannelZ: here's my sip config http://pastie.org/926710 |
| 07:27.30 | ChannelZ | your sip username is whatever is between the [ ] 's in your sip.conf for a given device |
| 07:27.37 | blaines | ohhhh |
| 07:27.43 | ChannelZ | so it's 101 then |
| 07:27.47 | blaines | thought that was the context |
| 07:28.14 | ChannelZ | no a context is what is between the [ ] 's in extensions.conf |
| 07:28.29 | blaines | haha, too similar |
| 07:29.30 | ChannelZ | well [xxxx] sort of means "a group of different settings that belong together" in various config files. |
| 07:29.50 | ChannelZ | Extension contexts, voicemail contexts, default settings, etc. |
| 07:29.51 | blaines | yea |
| 07:29.58 | blaines | it's just confusing |
| 07:30.24 | blaines | Everything I thought I knew has been turned upside down |
| 07:30.31 | blaines | which is good, but annoying |
| 07:36.06 | *** join/#asterisk Da-Geek (~Da-Geek@80.235.230.186) |
| 07:42.41 | blaines | ChannelZ: Hmm x-lite couldn't register with asterisk |
| 07:44.33 | *** join/#asterisk kruemeltee (~Maddin@port-92-198-62-82.static.qsc.de) |
| 07:45.06 | kruemeltee | hello all together :-) |
| 07:49.43 | *** join/#asterisk Gopal (~Miranda@122.165.63.115) |
| 07:50.04 | Gopal | i am getting cause 3 no route to destination, can anybody out help over here |
| 07:54.57 | *** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net) |
| 07:55.42 | kruemeltee | successfully registered the SIP client? ("sip show peers" within CLI Console) |
| 07:57.21 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 07:57.43 | Jumpie | gopal, there is only about a thousand reasons why |
| 08:06.26 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 08:25.37 | Gopal | Jumpie: I am using a media gateway as a SIP trunk |
| 08:25.52 | *** join/#asterisk E-bola (~bola@smtp.techbiz.dk) |
| 08:25.58 | E-bola | What does this mean: WARNING[4147]: chan_sip.c:17874 handle_response_invite: just did sched_add waitid(2111913) for sip_reinvite_retry for dialog 3d2642b1066a-h2s3j0d9vi0a in handle_response_invite |
| 08:26.04 | E-bola | I have it spamming in my console |
| 08:26.17 | E-bola | 1.6.2.6 |
| 08:26.19 | Gopal | Jumpie: i am getting a warning as No such host <SIP trunk> |
| 08:26.34 | *** join/#asterisk chendy_ (~chatzilla@204.152.211.137) |
| 08:27.16 | *** join/#asterisk aceio (~5d60a88a@gateway/web/freenode/x-ifxotzwawcjzqlsy) |
| 08:27.21 | aceio | hi all |
| 08:27.42 | aceio | seems like digium site is down |
| 08:28.31 | c0rnoTa | aceio: hi |
| 08:28.45 | *** join/#asterisk Z_God (~julius@wlan231150.mobiel.utwente.nl) |
| 08:28.46 | c0rnoTa | aceio: it works |
| 08:29.30 | aceio | c0rnoTa: mm that weired |
| 08:37.12 | Jumpie | gopal..is it as simple as a dns issue? |
| 08:37.20 | Jumpie | did you see if you can resolve the host first? |
| 08:37.23 | Jumpie | from your box |
| 08:48.15 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 08:48.23 | *** join/#asterisk Milad (~milad@unaffiliated/slackark) |
| 08:48.58 | *** join/#asterisk darkskiez (~dz@62-50-207-78.client.stsn.net) |
| 08:50.13 | aceio | iam trying to compile zaptel-1.4,12.1 |
| 08:50.54 | aceio | i am at the stage of make b410p and i am getting the following msg http://pastebin.com/c6r5RqT7 |
| 08:51.01 | Milad | I need a solution in asterisk context, I set a VAR in a context and after that run Dial command, I need to access that VAR after running Dial , but that VAR is empty do you have any clue what should do ? |
| 08:51.28 | aceio | can some check were i am going wrong |
| 08:51.58 | *** join/#asterisk bzing2 (~dr105@dhcp-194-66-208-236.canterbury.ac.uk) |
| 08:52.13 | aceio | scripts/Makefile.build:46: *** CFLAGS was changed in "/home/csadmin/asterisk/zaptel-1.4.12.1/mISDN-1_1_7_2/drivers/isdn/hardware/mISDN/Makefile". Fix it to use EXTRA_CFLAGS. Stop. |
| 08:58.38 | *** join/#asterisk Weazel (~something@mail.business-center.co.il) |
| 08:59.41 | Weazel | Hey guys, is anyone able to help me with a custom i need, searching for an option to manually choose which CID will be written into cdr when outgoing call is being made |
| 09:02.05 | kaldemar | Weazel: core show function CALLERID |
| 09:03.08 | Weazel | i'm a noob <--= |
| 09:07.00 | kaldemar | you can set the caller id for a channel with a function caller CALLERID. the previous is a CLI command that shows documentation for the command. |
| 09:08.12 | kaldemar | use the function inside the Set application in your dialplan, like exten => 123,1,Set(CALLERID(num)=12345). |
| 09:10.35 | tzafrir_laptop | aceio, any chance you moveto dahdi ? |
| 09:11.07 | tzafrir_laptop | This has been fixed in zaptel svn, IIRC, but not sure. |
| 09:11.14 | Weazel | thanks kaldemar |
| 09:11.16 | tzafrir_laptop | The latest version of zaptel is really dahdi |
| 09:12.14 | aceio | so can i use dahdi with asterisk 1,4 ? |
| 09:12.38 | tzafrir_laptop | aceio, yes, if it's 1.4.22 or higher |
| 09:13.23 | aceio | tzafrir_laptop: i have 1.4.30 |
| 09:13.32 | festr_ | does asterisk.org work for you? |
| 09:13.36 | E-bola | Is this valid syntax? exten => 1,n,Dial(IAX2/51201096@318295_peer) .It should dial the DID 51201096 via our IAX trunk named 318295_peer |
| 09:14.14 | tzafrir_laptop | So it should work |
| 09:14.33 | aceio | okay let me check it out |
| 09:15.44 | festr_ | anyone register problems with digium dns? |
| 09:16.00 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:16.47 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 09:17.40 | aceio | festr_: i see this problem |
| 09:17.47 | festr_ | ok |
| 09:18.33 | festr_ | anyone know IP of downloads.digium.com? :) |
| 09:19.10 | carrar | host downloads.digium.com |
| 09:20.18 | festr_ | carrar: try it |
| 09:20.26 | carrar | why |
| 09:20.30 | festr_ | lets do it |
| 09:20.36 | Chainsaw | festr_: You could try asterisk.org instead of digium.com |
| 09:20.42 | Chainsaw | festr_: I've changed all my download URLs. |
| 09:20.54 | festr_ | dns for asterisk.org does not work too |
| 09:21.04 | carrar | You didn't ask me if it would work or not |
| 09:21.17 | festr_ | carrar: does it work for you? |
| 09:21.43 | WIMPy | It doesn't for me |
| 09:21.46 | carrar | everything works fo rme |
| 09:21.52 | Chainsaw | carrar: Your knee-jerk reaction was unhelpful and inappropriate. Don't act so high & mighty about it. |
| 09:21.54 | carrar | Address: 76.164.171.233 |
| 09:22.20 | carrar | jerks a little more |
| 09:22.21 | festr_ | carrar: thank you. it seems there is problem in czech |
| 09:22.41 | festr_ | carrar: what is dns server for asterisk.org pls? |
| 09:23.06 | carrar | host -t NS asterisk.org |
| 09:23.06 | carrar | asterisk.org name server nsx3.digium.com. |
| 09:23.06 | carrar | asterisk.org name server nsx2.digium.com. |
| 09:23.07 | carrar | asterisk.org name server nsx1.digium.com. |
| 09:23.20 | festr_ | it starts to work now |
| 09:24.03 | carrar | Digium needs to get onboard the IPv6 train |
| 09:27.32 | *** join/#asterisk chendy (~chatzilla@219.134.112.137) |
| 09:32.16 | *** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com) |
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| 10:07.33 | *** join/#asterisk eject_ck (~eject_ck@haitek.cosmonova.net.ua) |
| 10:07.43 | aceio | okay i am going too hold off on dahdi drivers |
| 10:08.03 | eject_ck | how I should add my sip provider if it not require registration ? |
| 10:08.29 | aceio | <PROTECTED> |
| 10:19.16 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 10:28.47 | *** join/#asterisk Dovid (~annon@213.8.118.62) |
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| 10:49.24 | *** join/#asterisk jmkgreen (~chatzilla@fentech.gotadsl.co.uk) |
| 10:49.57 | jmkgreen | Having upgraded from 1.4 to 1.6 I've discovered IAX2 audio is now only working in one direction. What should I be looking at? |
| 10:50.36 | jmkgreen | The audio is working fine from the phone, through the firewall back to the asterisk box, no I hear nothing from the asterisk box through the firewall into the PSTN |
| 10:50.56 | *** join/#asterisk fleebailey33 (~fleebaile@unaffiliated/fleebailey33) |
| 10:51.03 | jmkgreen | previously I've fixed this by disabling dahdi (no idea what had happened) but this time the problem persists |
| 11:08.05 | *** part/#asterisk eject_ck (~eject_ck@haitek.cosmonova.net.ua) |
| 11:13.00 | *** join/#asterisk Weazel (~something@mail.business-center.co.il) |
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| 11:49.57 | photographe | i have 1 sip trunk working, with a incoming route, when i had an other sip trunk the first one stop working and when i call it asterisk tell me that i call on the second trunk what i did wrong |
| 12:01.08 | TJNII | Sounds like you messed up the authentication tokens. Pastebin your sip.conf |
| 12:06.32 | Gopal | Jumpie: i solved the issue, the issue is the context been defined |
| 12:06.32 | photographe | http://pastebin.ca/1867696 |
| 12:10.20 | TJNII | photographe: There are no settings for you sip accounts in there. What did you use to generate that? |
| 12:10.38 | *** join/#asterisk Akiraaa (~Akiraaaa@79.112.35.181) |
| 12:10.53 | photographe | its a ippbx runninx asterisknow gui |
| 12:11.02 | TJNII | Then go to #asterisknow |
| 12:11.11 | photographe | k |
| 12:11.18 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:11.23 | TJNII | ~asterisknow |
| 12:11.24 | infobot | asterisknow is, like, based on Asterisk, but is difficult to support in #asterisk for a number of reasons. Please seek support in #asterisknow instead. |
| 12:12.50 | photographe | thanks but you think i need to work under the sip.conf |
| 12:12.50 | TJNII | If you wern't running a GUI, I would say yes. However, since you are using a GUI, I have no idea where it buried the settings. |
| 12:12.51 | photographe | im sorry i was sure it was the same kind of file |
| 12:24.17 | *** join/#asterisk DJF5 (~email@84-105-183-83.cable.quicknet.nl) |
| 12:27.28 | *** join/#asterisk hipitihop (~denis@203.132.229.236) |
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| 12:31.11 | hipitihop | is it possible to drop a sip call from from the cli console and is it possible to temporarily disable a registered extension |
| 12:33.08 | [TK]D-Fender | hipitihop: "soft hangup [channel]" |
| 12:33.23 | [TK]D-Fender | hipitihop: And no, you can't block off a peer from CLI. |
| 12:34.40 | hipitihop | [TK]D-Fender, so I woul dhave to remove it from sip.conf and reload ? |
| 12:34.57 | [TK]D-Fender | hipitihop: yes |
| 12:35.46 | hipitihop | [TK]D-Fender, and a sip reload would drop the channel anyway I suspect |
| 12:36.13 | [TK]D-Fender | hipitihop: No, a call in progress remains alive |
| 12:37.28 | *** join/#asterisk angler_ (~angler@74.51.96.193) |
| 12:39.12 | *** join/#asterisk pentanol (~pentanol@77-35-10-058.pppoe.primorye.net.ru) |
| 12:39.24 | hipitihop | [TK]D-Fender, can I set up a diaplan using time ? iow, I want to restrict calls after a certain time |
| 12:39.42 | [TK]D-Fender | hipitihop: "core show application gotoiftime" |
| 12:39.49 | [TK]D-Fender | hipitihop: "core show function IFTIME" |
| 12:40.33 | *** join/#asterisk Dovid (~annon@213.8.118.62) |
| 12:41.30 | Dovid | morning/afternoon all |
| 12:42.35 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 12:43.55 | ariel_ | Morning folks |
| 12:44.53 | WereCat | ~answes |
| 12:44.58 | WereCat | ~answers |
| 12:44.59 | infobot | rumour has it, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 12:45.04 | WereCat | welcome back infobot! |
| 12:45.22 | WereCat | infobot: did you have a nice vacation? 8-) |
| 12:45.41 | [TK]D-Fender | ~areyouadog ? |
| 12:45.42 | infobot | Bark! Bark! |
| 12:45.48 | [TK]D-Fender | ~botsnack |
| 12:45.48 | infobot | [TK]D-Fender: thanks |
| 12:45.53 | [TK]D-Fender | infobot: Good boy! |
| 12:45.53 | infobot | thanks, [TK]D-Fender |
| 12:47.52 | *** join/#asterisk viq (~viq@unaffiliated/viq) |
| 12:48.15 | *** join/#asterisk Da-Geek (~Da-Geek@80.235.230.186) |
| 12:48.28 | Pan3D | ~book |
| 12:48.29 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 12:49.07 | hipitihop | is it posisble to restrict registration of a peer to a time range in sip.conf or is the only way via daiplan rule |
| 12:49.20 | [TK]D-Fender | hipitihop: Dialplan has nothing to do with registration |
| 12:49.39 | [TK]D-Fender | hipitihop: control of calling = dialplan |
| 12:50.40 | hipitihop | [TK]D-Fender, understand, was just wondering if I could restrict a given peer in sip.conf |
| 12:51.25 | *** join/#asterisk angler_ (~angler@nat/digium/x-vflbopyruypochyq) |
| 12:51.40 | WereCat | hipitihop: no, you cannot |
| 12:53.48 | hipitihop | WereCat, however can I test for a peer in a dialplan and use gotoiftime , e.g. if my curretn outgoing calls is "exten => _X.,1,Dial(SIP/${EXTEN}@gotalk)" |
| 12:54.23 | WereCat | yes |
| 12:56.28 | hipitihop | so if one of my peers in sip.conf is [LinksysPAP], how would I restrict that peer to a given time |
| 12:57.03 | hipitihop | if someone has a handy link .. the book does not seem to cover it, unless I missed it |
| 12:57.11 | WereCat | hipitihop: I think you need to read channelvariables.tex |
| 12:57.45 | WIMPy | The easiest way would be to put it in it's own context, I guess. |
| 12:58.09 | WereCat | WIMPy: that is another option |
| 12:59.07 | Dovid | can anyone help me out with this ? |
| 12:59.08 | Dovid | http://pastebin.ca/1865951 |
| 12:59.42 | WereCat | Dovid: when are you going to file a bug on bugs.digium.com? When are you going to start asking this in #asterisk-dev? |
| 13:00.07 | WereCat | sorry, asterisk doesn't have bugs anymore it has issues. issues.digium.com I think. |
| 13:00.47 | Dovid | WereCat: Cause when ever i put in an issue on Mantis I get shouted at :( |
| 13:00.54 | Dovid | shrugs shoulder. |
| 13:01.11 | hipitihop | ok thanks guys I'll do some more reading |
| 13:01.12 | WereCat | Dovid: that is the way it works, but nobody here is going to be able to help you with a coredump. try #asterisk -dev |
| 13:01.29 | Dovid | wanted to see if anyone would help here first. i only use the bug tracker as a last resort - don't wana piss off the digium god's |
| 13:01.57 | Dovid | ok |
| 13:02.00 | WereCat | Dovid: Asterisk CRASHED. Crashing should never be a config, setup, or design issue. It is a software bug. |
| 13:02.25 | *** join/#asterisk DelphiWorld (~Miranda@196.20.124.153) |
| 13:02.32 | DelphiWorld | hi all |
| 13:02.40 | DelphiWorld | anyone in france? |
| 13:02.57 | WereCat | DelphiWorld: several million people, I think. |
| 13:03.01 | Nugget | about 62 miliion people. |
| 13:03.14 | Dovid | DelphiWorld: I have servers in France. what do you need |
| 13:03.32 | DelphiWorld | Dovid: but are you from france |
| 13:03.47 | WereCat | ~ask |
| 13:03.48 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 13:03.55 | Dovid | nope |
| 13:03.56 | Nugget | even if David is from france that doesn't mean he's in france. |
| 13:04.06 | Dovid | just go there when my servers need some TLC |
| 13:04.17 | DelphiWorld | Dovid: ok see PM. |
| 13:04.20 | Dovid | try #vuc |
| 13:07.58 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 13:08.10 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
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| 13:12.12 | DelphiWorld | TheDavidFactor: sup |
| 13:14.39 | *** join/#asterisk Da-Geek (~Da-Geek@80.235.230.186) |
| 13:16.54 | TheDavidFactor | another monday. :-) |
| 13:18.31 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 13:18.53 | pentanol | hi, anybody here use odbc extensively? |
| 13:19.51 | zomenox | I also have odbc questions |
| 13:20.07 | pentanol | okay, why I got this ? func_odbc.c:466 acf_odbc_read: Unable to execute query [select count(distinct(t_dst)) from av_trunks where room = 'CR_762'] |
| 13:20.15 | *** join/#asterisk oldhack (~jfincher@cpe-24-27-56-221.austin.res.rr.com) |
| 13:20.39 | pentanol | meanwhile this one works well select distinct(t_dst) from av_trunks where room = 'CR_762' |
| 13:22.25 | *** join/#asterisk joako_ (~joako@opensuse/member/joak0) |
| 13:22.27 | pentanol | zomenox use while in extensions.conf? |
| 13:25.14 | TheDavidFactor | off hand I would suspect this is a problem the SQL statement and not with ODBC. Have you tested this manually? |
| 13:27.18 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:27.18 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 13:27.22 | pentanol | TheDavidFactor huh? of course it works just in mysql client |
| 13:28.07 | *** join/#asterisk spenguin[work] (~penguin@122.182.0.38) |
| 13:28.12 | spenguin[work] | hi |
| 13:28.48 | spenguin[work] | Ive got this Digium TE121 card inserted on the box, rebuild dahdi and the rest of them packages |
| 13:29.02 | spenguin[work] | # dahdi_hardware |
| 13:29.03 | spenguin[work] | pci:0000:05:08.0 wcte12xp- d161:8000 Wildcard TE121 |
| 13:29.23 | spenguin[work] | but dahdi_tool doesnt show any devices |
| 13:29.45 | spenguin[work] | whats the proper way to get this card installed |
| 13:30.32 | WereCat | then you have not configured the card |
| 13:31.09 | spenguin[work] | WereCat: does libpri need to be built before asterisk? |
| 13:31.15 | spenguin[work] | or would it matter in this case |
| 13:31.19 | WereCat | spenguin[work]: yes, but that has nothing to do with your problem. |
| 13:31.26 | WereCat | configure the stuff in /etc/dahdi |
| 13:31.33 | WereCat | then run dahdi_cfg -vvv |
| 13:31.42 | WereCat | confirm the output looks like what you expect |
| 13:31.55 | WereCat | spenguin[work]: you have never installed a Digium card before? |
| 13:32.11 | spenguin[work] | WereCat: no, and the digium site seems to be flapping |
| 13:32.26 | spenguin[work] | WereCat: I did install a sangoma card, but that was different |
| 13:32.31 | *** part/#asterisk DelphiWorld (~Miranda@196.20.124.153) |
| 13:32.38 | WereCat | No, it's not really all that different. |
| 13:33.42 | spenguin[work] | WereCat: any docs I can look at? |
| 13:34.06 | WereCat | spenguin[work]: voip-info.org has plenty of docs. none of the docs files included in DAHDI were helpful? |
| 13:34.29 | WereCat | why don't you look at the /etc/dahdi on the server you installed the Sangoma card on? |
| 13:34.41 | E-bola | Im using asterisk 1.4, when setting call-limit for a sip peer does it count both in and outgoing or only ingoing? |
| 13:34.44 | spenguin[work] | ok |
| 13:35.09 | WereCat | E-bola: read sip.conf.sample, pay special attention to the option to track counters on peers. |
| 13:35.23 | WereCat | E-bola: you have been here long enough you should know to check the sample config file. |
| 13:35.52 | E-bola | Well i seem to be finding conflicting info |
| 13:36.42 | WereCat | E-bola: only if you look at all the OUTDATED docs on the internet. If the sip.conf.sample is wrong then you need to file a bug./ |
| 13:37.18 | E-bola | Say i have a peer that is a member of a queue, i set call-limit = 2, to be able to do attended transfers, but wouldnt the queue send a second call to the phone if its currently talking to some1 and then if the phone wants to do an attended transfer it will be out of "slots" because the queue call has taken its second slot? |
| 13:37.35 | WereCat | MAYBE BECAUSE YOU DON'T HAVE LIMITONPEER SET TO YES????? |
| 13:37.45 | E-bola | umm relax? |
| 13:37.54 | WereCat | now stop wasting our time and read the damn sample config file. |
| 13:38.13 | E-bola | and limitonpeers is not relevant at all??? |
| 13:38.22 | E-bola | its for type=friend are you just blurting out random stuff now?' |
| 13:38.28 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.77) |
| 13:38.50 | WereCat | (08:34:41 AM) E-bola: Im using asterisk 1.4, when setting call-limit for a sip peer does it count both in and outgoing or only ingoing? |
| 13:39.19 | E-bola | Yes i know what i said, do you have a point? |
| 13:39.48 | WereCat | yes, almost all issues people have with call-limit are caused by the lack of the option to limit on a peer in addition to a user. |
| 13:40.30 | WereCat | type=friend, as you know means both user and peer |
| 13:41.30 | E-bola | yes but i have type=peer |
| 13:41.56 | WereCat | good. |
| 13:42.50 | WereCat | E-bola: I wish you the BEST of luck. |
| 13:43.26 | E-bola | ... |
| 13:49.01 | eppigy | lol |
| 13:51.13 | *** join/#asterisk hluesea (~hluesea@88.247.127.66) |
| 13:51.17 | Dovid | WereCat: SeanBright said that it seemed like a bug. put in a report and now we play the waiting game |
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| 13:55.33 | pentanol | no one else not use odbc? |
| 13:58.04 | TheDavidFactor | pentanol, I use odbc, but I'm familiar enough with it to know what that error means |
| 13:58.13 | TheDavidFactor | s/I'm/I'm not/ |
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| 14:00.44 | *** join/#asterisk bullium (~wbradshaw@216.68.250.30) |
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| 14:03.39 | hluesea | hello channel, can anyone advice me how can i do the when i get 404 sip error on a channel redirect another trunk ? |
| 14:04.35 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 14:05.00 | Chainsaw | hluesea: Enable verbose & debug on the Asterisk server giving you the 404. That'll force it to admit what's wrong. |
| 14:09.49 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 14:10.51 | E-bola | Is it possible to have more than 1 channel availeble on a phone, but only have asterisk use 1 of them for sending calls via a queue to the phone? |
| 14:11.15 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:11.15 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:11.19 | hluesea | Chainsaw, actually i want to redirect if the sip_trunk_1 give any error to another trunk, is it possible ? |
| 14:12.11 | freezey | odd issue... so from internal i have 1 extension that uses follow me and internal it works fine i can dial the ext and it goes to the external number... from external i can call the direct dial # and it fwds me to the correct follow me number as well... but from another site i have hooked up through dundi it fails and gives a 503 error any ideas? i think its because the phone is not actually plugged in? |
| 14:12.41 | Chainsaw | hluesea: It should set a DIALSTATUS result in most cases. Best check that. |
| 14:14.11 | *** join/#asterisk wdbl (~not@ool-44c0668f.dyn.optonline.net) |
| 14:14.30 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 14:15.43 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:16.22 | WereCat | Chainsaw: be careful! If you keep helping people you'll become as bitter and disillusioned as me and [TK]D-Fender |
| 14:17.27 | *** join/#asterisk evangelion (~manzy_zet@212.183.172.126) |
| 14:18.41 | evangelion | hello |
| 14:19.05 | evangelion | what's considered the current production/most stable release of asterisk? |
| 14:19.42 | pabelanger | Dovid: ping |
| 14:20.22 | Naikrovek | evangelion: check the titls |
| 14:20.23 | Naikrovek | uhh |
| 14:20.24 | *** join/#asterisk utahsaint (~utahsaint@mail.ntegratedsolutions.com) |
| 14:20.25 | Naikrovek | title |
| 14:20.34 | Naikrovek | pabelanger: pong |
| 14:22.09 | evangelion | Naikrovek: do you mean the topic? |
| 14:22.16 | Naikrovek | i do |
| 14:22.59 | evangelion | it does not answer my question! |
| 14:23.18 | Naikrovek | yes it does |
| 14:23.33 | E-bola | no it does not |
| 14:23.51 | Naikrovek | current production versions of asterisk are: 1.4.30, 1.6.0.26, 1.6.1.18, and 1.6.2.6 |
| 14:23.56 | Naikrovek | they're all different branches |
| 14:24.16 | Naikrovek | and they're all considered production-ready |
| 14:25.12 | Naikrovek | if there's a "most stable" version i don't know what it is. i don't think there are any that are considered "most stable" |
| 14:26.33 | evangelion | what are the reason to deploy 4 "stable" branches? |
| 14:26.52 | Naikrovek | because they all have different feature sets and support lifetimes |
| 14:26.53 | E-bola | being very friendly to your users |
| 14:26.54 | *** join/#asterisk s4msung (~s4msung@dice.s4msung.de) |
| 14:27.08 | E-bola | remember pbx systems are less frequently upgraded compared to webservers and what not |
| 14:27.29 | E-bola | i was asking the same questions my self a week ago hehe |
| 14:27.40 | Naikrovek | 1.8 comes out this year (according to last schedule I saw) and it's going to be a long-term support version, like 1.4 is now |
| 14:27.45 | evangelion | is there a feature comparaison table? |
| 14:28.02 | Naikrovek | evangelion: excellent question. I don't know |
| 14:28.33 | *** join/#asterisk moy (~chatzilla@bas1-unionville55-1177733627.dsl.bell.ca) |
| 14:28.36 | Naikrovek | here's half the answer: http://www.asterisk.org/asterisk-versions |
| 14:28.40 | Naikrovek | that covers support |
| 14:28.44 | Naikrovek | not what you asked for, but important |
| 14:28.50 | Naikrovek | other half i'm googling now |
| 14:28.56 | E-bola | A feature comparison would indeed be nice... |
| 14:29.11 | Naikrovek | mostly they're the same |
| 14:29.15 | *** join/#asterisk LemensTS (~LemensTS@adsl-70-238-175-27.dsl.stlsmo.sbcglobal.net) |
| 14:29.39 | E-bola | The devil's in the details.... |
| 14:29.46 | Naikrovek | 1.6.X support SIP over TCP (as opposed to UDP) and G722. 1.6.2.X supports Siren7 & Siren14 |
| 14:29.56 | Naikrovek | little things like that |
| 14:30.03 | Naikrovek | i don't remember all (or even most) of them |
| 14:30.16 | WereCat | Naikrovek: *nod* it's all documented in the standard place |
| 14:30.18 | E-bola | 1.6 also introduced device state's |
| 14:30.27 | *** join/#asterisk CunningPike (~CunningPi@S01060014bf81366b.vc.shawcable.net) |
| 14:30.33 | LemensTS | http://pastebin.ca/1867885 <= I cannot get the variables that are sent to an AGI script to be processed correctly in the AGI array. I put 2 different ways of sending it and the CLI output in this pastebn. I upgraded to 1.4.30 from a version from January 2009, didnt fix it. |
| 14:30.40 | E-bola | He needed a comparison, not documentation |
| 14:30.55 | E-bola | you can always read the changelog, but thats not that userfriendly |
| 14:31.06 | LemensTS | Qwell: ^^^ upgraded it for you |
| 14:31.10 | WereCat | I almost never read the changelog. I read UPGRADE*.txt |
| 14:31.21 | WereCat | since UPGRADE*.txt is where all significant changes are documented. |
| 14:31.25 | Naikrovek | yeah upgrade*.txt will have that |
| 14:31.57 | E-bola | mm well if ur upgrading ya... |
| 14:32.18 | Naikrovek | it shows the feature additions for several versions back, as i recall |
| 14:32.24 | WereCat | no, for everyone. There is so much old docs out there, reading the upgrade files is the only way to know what applies and what doesn.t |
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| 14:33.06 | E-bola | sure but then you need to read all the upgrades, if u dont know anything at all from scratch, hence a normal feature comparison is better |
| 14:35.12 | Naikrovek | the files aren't very big |
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| 14:39.36 | crazeekennee | maybe this isn't the right place for this -- sorry if it isn't -- but I'm trying to learn how to connect a Broadvox SIP trunk to a gentoo Linux router I built. Can soneone point me in the right direction ? Thank you ! |
| 14:40.23 | WIMPy | crazeekennee: You could install Asterisk. |
| 14:40.51 | *** join/#asterisk GameGamer43|Mac (~GameGamer@cpe-74-65-36-91.rochester.res.rr.com) |
| 14:42.55 | crazeekennee | WIMPy: we already have an IP-PBX in-house -- and we just need to mate the SIP truck with our VPN. As I understand Asterisk -- isn't it an IP/PBX by itself ? We don't need to duplicate that funcitonality. |
| 14:43.04 | Slugs_ | Is there a way to find out if the channels are boing selected from the highest first on outbound calls? |
| 14:43.23 | Slugs_ | being* |
| 14:44.43 | fenrus | crazeekennee, asterisk is an ip-pbx. |
| 14:45.14 | *** join/#asterisk captiancrash (~jonathan@12.71.218.232) |
| 14:45.57 | WereCat | ~toolkit |
| 14:45.58 | infobot | i heard toolkit is Remember Asterisk isn't really a PBX. Asterisk is a TOOLKIT that helps you build a PBX from scratch, much like libraries help you build an application from scratch. |
| 14:46.52 | WereCat | Slugs_: watch the cli |
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| 14:53.06 | Slugs_ | WereCat, what if its so fast ;/ |
| 14:53.30 | WereCat | Slugs_: then get a terminal with scrollback or look at the logs |
| 14:53.40 | fenrus | :D |
| 14:53.40 | Slugs_ | ahh yes |
| 14:53.44 | fenrus | shift+pageup = <3 |
| 14:53.48 | WereCat | heck the logs and the CDRs both have the info |
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| 14:54.16 | fenrus | crazeekennee, did you disappear? |
| 14:57.13 | evangelion | dumb question chan_h323 needs http://sourceforge.net/projects/openh323/ |
| 14:57.16 | evangelion | ? |
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| 15:06.37 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:06.48 | InHisName | my asterisk responds to UTC time and not the local time. How to adjust. The linux knows the local time and show that. |
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| 15:13.16 | spenguin[work] | Ive got this PRI card/line and up, but seem to get a lot of these - 12581 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 |
| 15:13.47 | spenguin[work] | I have switchtype=euroisdn |
| 15:13.51 | spenguin[work] | which should be right |
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| 15:16.20 | malcolmd | spenguin[work]: installation support comes with the purchase of the card. www.digium.com/support |
| 15:16.41 | spenguin[work] | malcolmd: the website seemed to be flapping earlier |
| 15:17.11 | malcolmd | ah, right, sorry. forgot about that. we're having provider troubles. |
| 15:17.12 | slashtom | has anyone used the oh323 module to interface to gnugk? |
| 15:17.29 | malcolmd | spenguin[work]: make sure your timing bit is set correctly in /etc/dahdi/system.conf set for 0 if you're not recovering timing, set for 1 if you are |
| 15:17.40 | slashtom | i'm getting no sound when calling from H323 to asterisk clients, but going the other way works fine |
| 15:17.50 | spenguin[work] | malcolmd: ok |
| 15:18.14 | malcolmd | spenguin[work]: also make sure you're on latest 2.3 dahdi drivers |
| 15:18.29 | spenguin[work] | ok |
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| 15:42.38 | ChannelZ | OT: Anyone know if the Intel ICH10 chipset is supported in Linux 2.6.? for SATA? Having trouble finding hard info |
| 15:43.12 | *** join/#asterisk joako_ (~joako@opensuse/member/joak0) |
| 15:43.41 | brad_mssw | pretty sure it just uses ACHI, which is probably why you're not finding info |
| 15:44.31 | fenrus | at least i know ICH8 works :) |
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| 15:46.15 | ChannelZ | yeah this sucks |
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| 15:47.16 | ChannelZ | I've got a dying server, which is running a rare Athlon MP board which has a non-standard power connection. I can't find an adapter cable locally to replace the PS so I'm trying to throw together a new system quickly |
| 15:47.41 | fenrus | hm, replace the motherboard and psu? :) |
| 15:48.01 | brad_mssw | what kernel version are you running? |
| 15:48.07 | fenrus | athlon mp isnt that fresh, an dual core would whip it's ass :) |
| 15:48.33 | ChannelZ | fenrus: that's what I'm doing |
| 15:48.48 | ChannelZ | I just want to make sure the thing will work. |
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| 15:49.05 | ChannelZ | brad_mssw: 2.6.24 at the moment |
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| 15:49.49 | brad_mssw | ChannelZ: what version of 'achi.ko' does that have? |
| 15:49.55 | brad_mssw | ChannelZ: use 'modinfo' on it |
| 15:50.08 | brad_mssw | ChannelZ: unless it is compiled into the kernel, the 'dmesg' might tell you |
| 15:51.46 | brad_mssw | pretty sure 2.6.24 is new enough though |
| 15:52.01 | ChannelZ | 3.0 |
| 15:52.07 | brad_mssw | yeah, 3.0 has it |
| 15:52.10 | brad_mssw | you're good |
| 15:52.25 | brad_mssw | if you were on 2.0, you'd need to add the pci ids to the ahci driver for ich10 |
| 15:52.38 | ChannelZ | Cool thank you very much |
| 15:52.47 | ChannelZ | Off to go shopping. Server just croaked again @*#& |
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| 16:18.04 | WereCat | ChannelZ: I bet the money you saved on the box is not quite so attractive now, is it? |
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| 16:41.31 | Naikrovek | WereCat: you speak like a channel veteran but i've been hanging in here for like 8 months and can't remember you. |
| 16:41.57 | [TK]D-Fender | Naikrovek: ManxPower |
| 16:42.02 | fenrus | is there such a thing as a "channel weteran" ? :) |
| 16:42.04 | Naikrovek | ah |
| 16:42.05 | fenrus | *veteran |
| 16:42.10 | Naikrovek | sure |
| 16:42.17 | [TK]D-Fender | Naikrovek: And He's been around longer than both of us. |
| 16:42.26 | Naikrovek | yeah everyone's been here longer than me |
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| 16:42.49 | [TK]D-Fender | WereCat: module unload chan_anachronism.so! |
| 16:42.53 | Naikrovek | i still got a lot of newb in me |
| 16:43.11 | [TK]D-Fender | Naikrovek: You may be somewhat new, but you are fairly bright |
| 16:43.22 | Naikrovek | thanks |
| 16:43.27 | [TK]D-Fender | Naikrovek: I'll take intelligence over experience in these things most of the time |
| 16:44.04 | Naikrovek | i wish i could just set aside part of my attention and just be in IRC all the time. the knowledge in here is amazing |
| 16:44.11 | Naikrovek | nice to know that i've absorbed some of it |
| 16:44.27 | Naikrovek | hopefully i'll absorb more |
| 16:44.38 | Naikrovek | and be able to help people |
| 16:46.35 | fenrus | anyone heard of any collaborative debugging tool like pastebin, but where you're able to do net-drawings (via some web-application) and publish them with comments on line ? |
| 16:46.47 | fenrus | this would really be awsome. |
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| 16:51.19 | nickfennell | hmmm |
| 16:51.27 | nickfennell | anyone use TelAppliant for their VOIP ? |
| 16:51.32 | nickfennell | or VoIPTalk |
| 16:51.35 | nickfennell | as they're the same people |
| 16:53.57 | *** join/#asterisk devoid (~62db04fa@gateway/web/freenode/x-ybboaaywoijkyjxz) |
| 16:53.59 | devoid | hi |
| 16:54.15 | devoid | http://s44.radikal.ru/i105/1004/a1/34b42e32b3d6.jpg |
| 16:54.23 | Naikrovek | uh |
| 16:54.25 | Naikrovek | not clicking that |
| 16:54.31 | Naikrovek | you need to explain it |
| 16:55.56 | Qwell | devoid: what about it? |
| 16:56.42 | Qwell | oh |
| 16:57.38 | pabelanger | it tastes like burning. |
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| 17:08.26 | hardwire | anybody have a list of test destinations in Manila? |
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| 17:11.23 | *** join/#asterisk p3nguin (gpz5GvdFkf@mtop-mpls.a2infotech.com) |
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| 17:18.33 | *** join/#asterisk SSJGotenks (~41ffe7e3@gateway/web/freenode/x-dtbfdyujsdhinynr) |
| 17:26.10 | *** join/#asterisk alexs77 (~alex@80-219-214-131.dclient.hispeed.ch) |
| 17:34.47 | *** join/#asterisk catpants (debian-tor@gateway/tor-sasl/catpants) |
| 17:43.52 | ariel_ | hardwire: I do a google search for hotels then give them a call |
| 17:44.03 | ariel_ | most have 24 hours operators |
| 17:47.01 | *** part/#asterisk catpants (debian-tor@gateway/tor-sasl/catpants) |
| 17:48.20 | R-Guy | hardwire: call the Cathay Pacific recorded airport record info number |
| 17:49.07 | p3nguin | What are we trying to accomplish? |
| 17:50.16 | alexs77 | Hi. I'm new to Asterisk and have installed trixbox 2.8.0.4 in a VirtualBox VM. I've setup one IAX2 trunk (to personal-voip.de), one Extension & 1 outbound route. I verified that the username, secret, hostname & register are correct (at least I think so *G*). But asterisk just doesn't register... I *CAN* dial out, though. On http://j.mp/9ZzjoG you can find a screenshot showing the configuration. |
| 17:50.28 | Qwell | alexs77: #trixbox |
| 17:50.35 | alexs77 | Qwell: thx |
| 17:53.15 | *** join/#asterisk Z_God (~julius@wlan229184.mobiel.utwente.nl) |
| 17:59.41 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 18:15.19 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 18:16.10 | bmoraca_work | customer wants to dial Mali, Africa... |
| 18:16.42 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:16.42 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 18:16.50 | bmoraca_work | my rate to them is .25 - .30 per minute |
| 18:18.15 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 18:18.18 | p3nguin | Mine is .23 - 31, depending on the terminating carrier. |
| 18:18.21 | Naikrovek | anyone in here have an ASA |
| 18:18.29 | Naikrovek | #cisco is being unhelpful as per usual |
| 18:18.43 | bmoraca_work | what are you trying to do? |
| 18:18.53 | Naikrovek | my ASA is shunning packets from a host in the whitelist |
| 18:19.26 | Naikrovek | from the ASDM syslog: 4Apr 19 201012:33:20401004Shunned packet: 202.153.33.123 ==> ASA_Outside_Interface on interface outside |
| 18:19.44 | Naikrovek | and the config line that is appropriate: threat-detection scanning-threat shun except ip-address 202.153.33.123 255.255.255.255 |
| 18:19.54 | bmoraca_work | p3nguin: the tough thing, though, is that if i allow them international calling (which i don't for anyone else), i have to figure out how to bill them properly |
| 18:20.27 | bmoraca_work | right now, i don't allow international, and everything is flat-rate billing |
| 18:21.10 | p3nguin | Is this a VIP customer? |
| 18:21.17 | bmoraca_work | it is, yes |
| 18:21.30 | bmoraca_work | to my boss anyway |
| 18:21.33 | bmoraca_work | i could care less |
| 18:22.22 | bmoraca_work | i'm fairly sure that i'm going to have to make it work |
| 18:22.27 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-172.dslextreme.com) |
| 18:23.39 | p3nguin | Sounds like a bother. |
| 18:23.50 | bmoraca_work | this whole job is a bother |
| 18:23.58 | Naikrovek | i think i know why i don't really know cisco stuff very well |
| 18:24.01 | Naikrovek | it is infuriating |
| 18:24.21 | Naikrovek | half of cisco configuration is magic. the device has to like you |
| 18:24.26 | bmoraca_work | Naikrovek: unfortunately, i don't have any experience with the IDS and IPS stuff |
| 18:24.34 | Naikrovek | that's fine |
| 18:24.51 | Naikrovek | even when i turn shunning off it continues to be shunned by this damned ASA |
| 18:24.55 | bmoraca_work | i have no desire to go down the security tract |
| 18:25.48 | *** join/#asterisk sulex (~sulex@dynamic-adsl-78-14-172-249.clienti.tiscali.it) |
| 18:25.53 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
| 18:25.55 | cusco | hail |
| 18:26.12 | cusco | Im looking to execute a php script froma AEL |
| 18:26.26 | cusco | I could use System(php -q bla.php); |
| 18:26.46 | cusco | but I would like to store stdout in a ${VAR} |
| 18:26.53 | cusco | how can I acomplish that? |
| 18:27.19 | bmoraca_work | executing system scripts from dialplan was pretty much the reason AGI was created, right? why not use AGI? |
| 18:27.57 | cusco | right... I never used it tho |
| 18:28.09 | [TK]D-Fender | cusco: "core show function SYSTEM" |
| 18:28.42 | cusco | No function by that name registered. |
| 18:28.59 | [TK]D-Fender | cusco: Upgrade |
| 18:29.12 | cusco | im running 1.6.2.6 |
| 18:29.19 | [TK]D-Fender | Hrm. |
| 18:29.22 | [TK]D-Fender | 1 sec |
| 18:29.34 | cusco | so basically I place the php script in /var/lib/asterisk/agi-bin/ |
| 18:29.40 | cusco | how do I call it from ael? |
| 18:29.53 | bmoraca_work | AGI() |
| 18:30.14 | cusco | AGI(script-name.php) = |
| 18:30.15 | cusco | ? |
| 18:30.29 | bmoraca_work | pretty much |
| 18:30.38 | bmoraca_work | some googling and reading about how AGI works might help |
| 18:30.47 | [TK]D-Fender | cusco: "core show function SHELL" |
| 18:30.48 | bmoraca_work | it's not too tough to use |
| 18:31.06 | cusco | Not available |
| 18:31.22 | [TK]D-Fender | cusco: then you've done something wrong... |
| 18:31.23 | cusco | bmoraca_work: how do I store output from script.php onto a var? |
| 18:31.27 | bmoraca_work | cusco: capitalization counts. |
| 18:31.36 | bmoraca_work | cusco: you can set dialplan variables from within AGI. |
| 18:31.47 | cusco | [TK]D-Fender: http://paste.debian.net/69769/ |
| 18:31.59 | cusco | bmoraca_work: ah.. ok |
| 18:32.00 | bmoraca_work | cusco: capitalize it properly. eg. "core show application shell" doesn't work, while "core show application SHELL" does. |
| 18:32.34 | p3nguin | application or function? |
| 18:32.42 | bmoraca_work | function of course |
| 18:32.43 | [TK]D-Fender | FUNCTION |
| 18:32.51 | bmoraca_work | just making sure he was paying attention |
| 18:33.00 | p3nguin | heh |
| 18:33.13 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:33.42 | p3nguin | I don't have a SYSTEM or SHELL function. |
| 18:33.54 | bmoraca_work | you guys are weird then |
| 18:33.56 | [TK]D-Fender | p3nguin: What version? |
| 18:34.02 | p3nguin | 1.4.30 |
| 18:34.19 | cusco | bmoraca_work: I am aware of capitalization, asterisk capitalizes on tab-completition |
| 18:34.26 | p3nguin | I have a System application, though. |
| 18:34.29 | [TK]D-Fender | p3nguin: aNTEDILUVIAN |
| 18:35.28 | *** join/#asterisk GameGamer43|Mac (~GameGamer@cpe-74-65-36-91.rochester.res.rr.com) |
| 18:37.27 | Slugs_ | is there a config file to choose which outbound channels to use? like start 1 and go up or vise versa, or even start outward and work to the middle? |
| 18:38.01 | leifmadsen | Slugs_: I think you need to be more specific in your question -- what channel technology are you talking about? |
| 18:38.12 | Slugs_ | zap, pri |
| 18:38.27 | leifmadsen | Slugs_: if you're talking about DAHDI, then you want to use groups and can use either DAHDI/g0 or DAHDI/G0 to control the direction you use channels |
| 18:38.33 | leifmadsen | (bottom to top, or top to bottom) |
| 18:39.32 | Slugs_ | ahh |
| 18:39.36 | Slugs_ | perfect |
| 18:39.37 | Slugs_ | ty |
| 18:40.25 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 18:40.37 | Naikrovek | gotta love cisco |
| 18:40.41 | Naikrovek | reboot it and it works |
| 18:40.49 | Naikrovek | no more stable than a damned windows 95 machine |
| 18:40.54 | Slugs_ | leifmadsen, can you do outside in? |
| 18:41.00 | *** join/#asterisk Knightfal (~j@mailer.1callres.com) |
| 18:41.57 | cusco | and... |
| 18:41.59 | leifmadsen | Slugs_: I don't understand the question |
| 18:42.06 | cusco | how do I set a global VAR trough the manager |
| 18:42.08 | cusco | not a channel one? |
| 18:42.35 | Slugs_ | outside is as in you have 23 channels, and it will usw 23, then 1, 22, 2, 21,3 |
| 18:42.39 | Slugs_ | etc |
| 18:43.08 | leifmadsen | Slugs_: not to my knowledge |
| 18:43.08 | Slugs_ | ive see it setup like that w/ avaya is the only reason i ask |
| 18:43.18 | Slugs_ | got ya, ty again |
| 18:44.03 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.139.3.dsl.dyn.forthnet.gr) |
| 18:46.13 | R-Guy | Slugs_: normally one end goes top-to-bottom and the other end goes bottom-to-top to avoid collisiosn. no sure what the benefit might be grab from topa nad bottom at the same time. |
| 18:48.22 | cusco | bmoraca_work: setvar only sets variable to specified channel |
| 18:48.40 | cusco | can i either set a global var, or get the current channel name? |
| 18:48.41 | Ad-Hoc | hoi |
| 18:48.58 | cusco | ah, I can pass argument with channel name from ael... |
| 18:49.03 | bmoraca_work | why would you be setting global variables within dialplan? |
| 18:49.47 | *** join/#asterisk dmz (~dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 18:50.33 | cusco | right.. |
| 18:50.37 | *** join/#asterisk Alagar (~Administr@122.164.38.27) |
| 18:50.39 | cusco | I wouldn't |
| 18:52.58 | bmoraca_work | from a design point of view, if you're setting global variables at call-time, you probably have a design flaw |
| 18:56.14 | Naikrovek | bmoraca_work: fixed my ASA problem by rebooting. |
| 18:56.19 | Naikrovek | oh i said that already |
| 18:56.22 | Naikrovek | nevermind then |
| 18:56.28 | bmoraca_work | lol |
| 18:56.51 | Naikrovek | what do you guys use for asset management |
| 18:57.03 | Naikrovek | spreadsheet? gpl app? paid app? |
| 18:57.06 | Naikrovek | homegrown? |
| 18:57.29 | bmoraca_work | my freaking brain? |
| 18:57.33 | Knightfal | Im trying to set the device state to inuse on an outbound call should asterisk do this on its own or do I need to use a custom devstate |
| 18:57.51 | Naikrovek | mine is mostly in my brain now but i need to offload that |
| 18:58.57 | Belgarath | hire pa |
| 18:59.12 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 18:59.17 | Belgarath | external brain capacity :) |
| 18:59.32 | Naikrovek | thinking of tossing something together with php |
| 18:59.35 | Naikrovek | and mysql |
| 19:00.12 | Naikrovek | call it NOFRIM for NO FRills Inventory Management |
| 19:00.56 | Naikrovek | my company's initals are AR. ARIS for A.R. Inventory System |
| 19:13.26 | *** join/#asterisk GameGamer43|Mac (~GameGamer@cpe-74-65-36-91.rochester.res.rr.com) |
| 19:17.32 | cusco | bmoraca_work: thanks |
| 19:17.32 | cusco | I made it work calling a php that connects to manager and takes the ${CHANNEL} argument |
| 19:17.32 | cusco | cool |
| 19:20.28 | *** join/#asterisk GameGamer43|Mac (~GameGamer@cpe-74-65-36-91.rochester.res.rr.com) |
| 19:21.06 | *** join/#asterisk Scorcerer (scor@czlug.icis.pcz.pl) |
| 19:21.17 | Scorcerer | hello |
| 19:21.21 | *** part/#asterisk devoid (~Dev_Null@unaffiliated/devemo) |
| 19:22.35 | Naikrovek | hello, Scorcerer |
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| 20:22.17 | beefpastry | anybody in here had any luck with faxing in 1.6? |
| 20:22.30 | Qwell | ~faxforasterisk |
| 20:22.31 | infobot | [faxforasterisk] Digium's commercial Fax For Asterisk module is available at http://www.digium.com/en/products/software/faxforasterisk.php |
| 20:23.09 | beefpastry | Yeah...my issue is that I'm testing it on an in-house fax machine on a pots line, and I'm 1 for 10. |
| 20:23.24 | beefpastry | I'm 100% faxing to efax, though. |
| 20:23.43 | beefpastry | playing with settings to no avail |
| 20:23.52 | *** join/#asterisk InHisName (~rich@c-76-116-145-200.hsd1.pa.comcast.net) |
| 20:24.53 | *** join/#asterisk citywok (~chatzilla@67-134-194-33.dia.static.qwest.net) |
| 20:27.01 | citywok | I have a very weird case where i think * reloading is causing all of my phones to drop connection... http://pastebin.com/5wGi9NSc |
| 20:27.35 | *** join/#asterisk Alric (~nbowyer@anvault.com) |
| 20:27.44 | citywok | for the last several weeks every once in a while all of my phones will lose their registrations, and i hadn't been paying much attention until it happened today while i had my console open |
| 20:28.08 | lordvadr | I'm having a Playtones problem. The tone only plays for the first few hundred milliseconds and then it goes silent. I've tried it with and without Answer, I'm waiting 10 seconds after the Playtones, also tried it with and without Progress |
| 20:28.29 | lordvadr | v1.6.2.6 |
| 20:31.03 | *** join/#asterisk blaines (~blaines@174-17-165-196.phnx.qwest.net) |
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| 20:35.37 | Gugge | im trying to balance requests / registrations to two asterisk servers with srv records, but my sip phones sometimes sends reinvites (on hold) to server1 when they send the original invite to server2, is that a phone problem, or can i do something to avoid that at the asterisk side? |
| 20:37.20 | bmoraca_work | canreinvite=no, potentially |
| 20:37.23 | bmoraca_work | otherwise, probably not |
| 20:37.41 | bmoraca_work | it's probably a phone-specific issue iwth its implementation of SRV |
| 20:37.42 | Gugge | canreinvite does not stop my phones from sending the reinvite when i press hold :) |
| 20:38.08 | Gugge | and the hold reinvite actually starts a new call on the other asterisk, because it does not know the original call |
| 20:38.13 | bmoraca_work | is your SRV set up with weighted entires, or in strinct round-robin? |
| 20:38.25 | Gugge | same weight |
| 20:38.28 | Gugge | so round-robin |
| 20:38.33 | bmoraca_work | that's probably your issue |
| 20:38.50 | bmoraca_work | SRV isn't really meant to be used for round-robin. it's meant to provide preference. |
| 20:39.04 | Gugge | its ment to do both :) |
| 20:39.10 | Gugge | it has both priority and weight :) |
| 20:39.35 | Gugge | but i guess i should just make two srv hostnames, one with server1 as lowest prio, and one with server2 as lowest |
| 20:40.03 | Gugge | and then set half the phones to srv1 and the other half to srv2, then they would fallback to the other server if one dies |
| 20:40.14 | Gugge | i hate that sollution though :) |
| 20:40.38 | bmoraca_work | it's not a very good one |
| 20:41.03 | Gugge | do you have a better idea, if i dont want one of the servers to just be a standby? |
| 20:41.05 | bmoraca_work | can you affect the DNS cache on your phone? maybe see if it's a known (potentially fixed) issue for that model/version? |
| 20:41.40 | Gugge | well, it is wrong on both snom, zoiper, linphone and ekiga :) |
| 20:42.26 | Gugge | ekiga is really wrong, it cant even register, because it sends the first reg to server1, and then the second with auth to server2 :P |
| 20:42.35 | Gugge | and server2 offcause dont know the nonce from server1 :) |
| 20:43.10 | bmoraca_work | i never use softphones, so i don't know if that's by design or not. it sounds like an improper implementation of SIP, to be honest (not responding to the IP contained in the messages, but rather to the configured parameter) |
| 20:43.43 | Gugge | i would expect the phones to make a srv lookup, and then use _one_ host for an entire dialog too |
| 20:43.50 | Gugge | but that is just not what i see :( |
| 20:43.54 | Naikrovek | ekiga is weird like that if you leave a sip proxy configured |
| 20:44.07 | Naikrovek | i think that's what it was |
| 20:44.18 | Naikrovek | some IP/host that is configured separately to the individual accounts |
| 20:44.24 | Naikrovek | turning that off fixed it for me |
| 20:44.24 | Gugge | hmm, maybe i setup the srv name in the proxy field too |
| 20:44.25 | Naikrovek | when i did that |
| 20:44.40 | Naikrovek | performs yet another drive-by diagnosis |
| 20:44.50 | Gugge | ill check that later :) |
| 20:44.59 | bmoraca_work | thinks softphones are a waste of time |
| 20:45.16 | Gugge | i agree :) |
| 20:45.24 | Naikrovek | oh yeah big time agreeings |
| 20:46.07 | Gugge | but my guess was that if snom failed, and the 3 softphones i tried failed ... then other phones would fail too, and i want my setup to work for peoples strange sip devices :) |
| 20:47.02 | InHisName | I'm having trouble with GoToIfTime() reading UTC instead of my local time. linux reports local time, $asterisk -Tr reports local time. BUT gotoiftime uses UTC time for its decisions (I am in old 1.2.10 version) Is there a way to make this work right ? Some kinda fix ?> |
| 20:47.30 | Gugge | InHisName: my first idea would be to upgrade :) |
| 20:48.31 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
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| 20:48.59 | InHisName | got a second idea ? Can'[t up grade for a couple of months anyway. |
| 20:50.07 | deeperror | InHisName: how about an agi script that makes the conversion? |
| 20:50.50 | InHisName | Hmm, has one been made for this yet? |
| 20:51.12 | Gugge | InHisName: write the gotoiftime in utc :) |
| 20:51.29 | Gugge | that would be my second idea :) |
| 20:52.47 | InHisName | I was saving that one for LAST RESORT ! Edits every half year until the upgrade is fixed good enough. |
| 20:53.16 | *** join/#asterisk cdose1 (~chris@2001:0:53aa:64c:4f1:7d2d:9d50:eefb) |
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| 20:53.42 | deeperror | InHisName: could do something where you +$OFFSET then set the offset variable to whatever you need it to be |
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| 20:54.41 | lordvadr | I want to be able to proxy external sip invites for my users but not for the internet at large. All I seem to be able to do is turn on allowexternalinvite globally and not on a per user basis. Is there a way to get external invites to run through some context before sending the on? |
| 20:58.30 | *** join/#asterisk Jumpie (n3rdz@ip68-98-31-152.ph.ph.cox.net) |
| 20:58.33 | Jumpie | ffs |
| 20:58.35 | InHisName | Interesting, deeperror, Say I have this: exten => 8800,n,GoToIfTime(16:29-16:50|mon-sun|*|*?default,8500,1) How to incorporate the +$OFFSET ? Where offset = -4 hours. |
| 20:58.38 | Jumpie | i find myself disconnected every time i wake up |
| 20:58.44 | Jumpie | and the registering part is annoying :D |
| 20:59.19 | fenrus | Jumpie, add some register script then ;) |
| 20:59.53 | *** part/#asterisk deeperror (~deeperror@adsl-76-226-151-213.dsl.sfldmi.sbcglobal.net) |
| 21:01.45 | Jumpie | yea hehe |
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| 21:26.13 | bent_screwdriver | what's wrong with this: exten => s,n,ExecIf($[ ${IBC} = 1 ] ?Playback(silence/1&this-call-may-be-monitored-or-recorded)) |
| 21:26.38 | bent_screwdriver | it's always true |
| 21:27.12 | [TK]D-Fender | bent_screwdriver: Show us |
| 21:27.56 | bent_screwdriver | [TK]D-Fender: Right before it hits this is the noop for the var IBC NoOp("SIP/10120-0000000f", "IBC=") |
| 21:28.25 | bent_screwdriver | [TK]D-Fender: but it runs the playback anyways. |
| 21:28.45 | p3nguin | If it's true, of course it ran the Playback. |
| 21:29.18 | bent_screwdriver | p3nguin: meant false. IBC is not set but it is playing the recording |
| 21:30.02 | [TK]D-Fender | bent_screwdriver: Show us |
| 21:30.42 | bent_screwdriver | here is the error that it put out: syntax error: syntax error, unexpected '=', expecting $end; Input: |
| 21:30.42 | bent_screwdriver | <PROTECTED> |
| 21:31.19 | [TK]D-Fender | bent_screwdriver: Because you can't have "nothing" on either side of an expression |
| 21:31.27 | p3nguin | This probably works in 1.4: ExecIf($["${IBC}" = "1"]|Playback|silence/1&this-call-may-be-monitored-or-recorded) |
| 21:31.50 | p3nguin | I don't know if the pipes are supposed to be changed to commas in 1.6.x. |
| 21:32.13 | bent_screwdriver | [TK]D-Fender: i thought the [ would let you do that |
| 21:32.44 | [TK]D-Fender | bent_screwdriver: $[] contains the expression, but you still have nothing on the left side |
| 21:32.58 | p3nguin | There's also no ? in ExecIf() in 1.4. |
| 21:33.51 | bent_screwdriver | p3nguin: i'm on 1.6 and the example on the wiki is exact, but like tk said, probably cause it is null. will the "" fix that? |
| 21:33.59 | p3nguin | yes |
| 21:34.16 | Gugge | bent_screwdriver: look at it like $[ = 1 ] vs $[ "" = "1" ] |
| 21:34.25 | p3nguin | Because you're comparing "" to something else . ... what gugge said. |
| 21:34.46 | Gugge | then its pretty clean why it wont work |
| 21:34.55 | bent_screwdriver | ahh didn't realize "" wasn't nothing...makes sense and thanks! trying now. |
| 21:35.00 | p3nguin | If the variable is null, you're comparing null to something, which isn't allowed. |
| 21:36.24 | bent_screwdriver | i'll be dammed that fixed it. i'll have to remember the " in the future. thank you both! |
| 21:47.32 | *** join/#asterisk Zizou (~zizou@190.37.18.195) |
| 21:48.31 | *** join/#asterisk hfb (~hfb@pool-96-247-108-157.lsanca.dsl-w.verizon.net) |
| 21:48.47 | Zizou | Hi, does anybody knows where can i find free spanish prompts? im configuring voicemail in an asterisk box and there are a lot of missing sounds, so its giving me trouble |
| 21:49.32 | Zizou | i get the spanish-prompt-es-co package from the debian repository |
| 21:54.10 | *** join/#asterisk cosmicwombat (~cosmicwom@69.7.44.68) |
| 21:55.29 | KavanS | polycom 301 - thoughts, opinions...rants, good or bad? |
| 21:55.40 | [TK]D-Fender | <PROTECTED> |
| 21:55.53 | [TK]D-Fender | KavanS: How are you getting them, and at what price? |
| 21:56.19 | KavanS | [TK]D-Fender, buying them new from online...with ac adapter, looks like 60-70$ a piece |
| 21:56.32 | [TK]D-Fender | KavanS: You in a PoE environment? |
| 21:56.38 | KavanS | nawp, not POE environment |
| 21:56.42 | KavanS | too small of an office for that... |
| 21:56.50 | [TK]D-Fender | KavanS: how many are you considering? What kind of deployment? |
| 21:57.08 | KavanS | maximum of 10...just something better than the current Linksys SIP adapters we're using |
| 21:57.27 | KavanS | looking to utilize an external headset....according to the 301 pdf, they support headsets |
| 21:57.30 | [TK]D-Fender | KavanS: They are decent phones, and have all the basic functionality.... |
| 21:57.35 | [TK]D-Fender | KavanS: But they are EOL |
| 21:58.01 | [TK]D-Fender | KavanS: If you're happy with "basic SIP phone, with no speakerphone, and RJ9 headset, then sure |
| 21:58.04 | KavanS | ok...it'd be a safe assumption they would be an improvement from our SIP adapters |
| 21:58.16 | KavanS | ok, any suggestions for something a "notch" higher than the 301? |
| 21:58.25 | KavanS | something with speakerphone and cleaner headset ? |
| 22:00.47 | WereCat | like maybe the 330? |
| 22:03.53 | KavanS | ahh ok |
| 22:03.55 | KavanS | that might be a good fit |
| 22:04.09 | KavanS | researching now, thank for the word |
| 22:06.12 | [TK]D-Fender | KavanS: Need a passthrough as well? |
| 22:06.41 | KavanS | probably not, not entirely sure what a passthrough is for SIP phones |
| 22:08.52 | WereCat | it means the phone has 2 ethernet ports |
| 22:09.03 | KavanS | nawp, not necessary |
| 22:09.16 | WereCat | one to link to the network, one for a PC. It actually has an internal 3-port switch. LAN, PC, and internal Phone |
| 22:11.36 | [TK]D-Fender | KavanS: IP 321 = $80 and is "current", uses 2.5mm headset, has a speakerphone, and supports Poe. You'll have to get a switch to power them however, or buy bricks. I suggest the latter |
| 22:16.02 | KavanS | [TK]D-Fender, ok roger that, lemme see if I can locate that model online...sounds like a good fit |
| 22:16.07 | *** join/#asterisk sulex (~sulex@dynamic-adsl-78-14-169-200.clienti.tiscali.it) |
| 22:16.09 | KavanS | thank you for the pro tips, this def. helps narrow it down... |
| 22:21.32 | citywok | every time asterisk reloads my PAP2 goes unreachable |
| 22:21.43 | citywok | any ideas? |
| 22:23.37 | R-Guy | citywok: reloads or is restarted? |
| 22:23.46 | citywok | reloads :P |
| 22:23.48 | citywok | lol |
| 22:24.47 | R-Guy | what are you reloading? sip? |
| 22:24.55 | citywok | asterisk -rx 'reload' |
| 22:25.11 | p3nguin | stop doing that. |
| 22:25.19 | p3nguin | The problem should go away. |
| 22:25.26 | citywok | thanks, very very insightful :P |
| 22:25.39 | \malex\ | does anyone here use an allworx 9112 phone with asterisk? i'm looking for info on its config file, but my google skills are failing me |
| 22:28.04 | *** join/#asterisk QbY (~kelvin@c-24-126-145-123.hsd1.ga.comcast.net) |
| 22:28.24 | QbY | Two crashes today.. http://pastebin.com/YeFF5gFZ -- Anyone see anything? |
| 22:29.24 | leifmadsen | QbY: using app_confbridge? |
| 22:29.27 | Qwell | What version of Asterisk? |
| 22:29.36 | QbY | leifmadsen: yes.. |
| 22:29.50 | leifmadsen | QbY: I see bridge_softmix in the mix, and only app_confbridge uses that afaik |
| 22:29.51 | QbY | Qwell: 1.6.2.6 |
| 22:31.24 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 22:31.41 | leifmadsen | QbY: I'm not sure your backtrace shows enough information to really determine what is going on though |
| 22:32.45 | QbY | <PROTECTED> |
| 22:33.22 | QbY | leifmadsen: yeah, i didn't garner much from it either |
| 22:34.21 | leifmadsen | QbY: see http://svn.asterisk.org/svn/asterisk/trunk/doc/backtrace.txt |
| 22:34.24 | leifmadsen | recently updated |
| 22:36.09 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 22:45.55 | *** join/#asterisk paulc_ (~paulc@unaffiliated/paulc) |
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| 22:54.23 | *** join/#asterisk sulex (~sulex@dynamic-adsl-78-14-170-149.clienti.tiscali.it) |
| 23:02.18 | *** join/#asterisk rare1980_ (~rare1980@115.186.20.131) |
| 23:02.34 | rare1980_ | hi all i have an error on asterisk |
| 23:02.46 | rare1980_ | call get disconnected after 60 sec |
| 23:02.57 | rare1980_ | error msg is lack of RTP activity in 61 seconds |
| 23:03.11 | rare1980_ | any idea wht could be the issue? |
| 23:03.19 | *** join/#asterisk jks (~jks@193.189.93.254) |
| 23:03.36 | fenrus | it says that there is no RTP |
| 23:03.40 | *** join/#asterisk galeras (~galeras@190.146.13.27) |
| 23:03.48 | rare1980_ | wht is rtp |
| 23:03.49 | rare1980_ | ? |
| 23:04.01 | fenrus | the protocol that carries the audio/video |
| 23:04.13 | rare1980_ | ooh ok |
| 23:04.26 | rare1980_ | fenrus: how do i fix it |
| 23:04.28 | fenrus | you get sound through in both directions ? |
| 23:04.34 | rare1980_ | nop |
| 23:04.39 | rare1980_ | i don't get any sound |
| 23:04.50 | fenrus | well, that's the REAL problem then |
| 23:04.57 | rare1980_ | humm |
| 23:05.02 | rare1980_ | soo any idea how to fix it |
| 23:05.25 | fenrus | is your phones nated? |
| 23:05.34 | rare1980_ | nated mean??/ |
| 23:05.40 | fenrus | ~nat |
| 23:05.41 | infobot | rumour has it, nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 23:05.44 | rare1980_ | nop |
| 23:06.10 | *** join/#asterisk TJ^ (~TJ@193.47.83.49) |
| 23:06.25 | rare1980_ | my softfone is not behind any nat |
| 23:07.03 | galeras | is posible to have a sip trunk to group many sip friends?, i mean something like dahdi group. |
| 23:07.22 | galeras | and dial it like SIP/TRUNK/NUM |
| 23:07.41 | leifmadsen | galeras: no, you'd use multiple Dial() lines |
| 23:07.45 | \malex\ | galeras: why would you want to do that? |
| 23:07.58 | leifmadsen | to fail over -- mostly because each peer can typically handle multiple calls |
| 23:08.09 | \malex\ | ah |
| 23:08.09 | galeras | my dialer is so poor :( |
| 23:08.35 | leifmadsen | you could build a subroutine that would apply GROUP() and GROUP_COUNT() to each peer and track when each has a call associated with it, then roll over to the next one in the list |
| 23:08.45 | fenrus | rare1980_, try calling the asterisk and playback some sound so you can verify that the phone is working properly. |
| 23:09.02 | leifmadsen | Echo() is good for that |
| 23:09.10 | *** join/#asterisk boodu (~antoine@175.158.129.128) |
| 23:09.14 | boodu | hello |
| 23:09.28 | rare1980_ | fenrus: i have tried that but i can't hear any sound |
| 23:09.35 | rare1980_ | from my asterisk box |
| 23:09.57 | *** join/#asterisk jstapleton (~jstapleto@c-24-125-171-223.hsd1.va.comcast.net) |
| 23:10.02 | galeras | leifmadsen: Thanks |
| 23:10.12 | fenrus | verify that the RTP ports your softphone use is open in any firewall |
| 23:10.23 | fenrus | and the same for the asterisk |
| 23:10.37 | rare1980_ | let me check .. |
| 23:10.38 | ChannelZ | WereCat: (reply from hours ago) The machine in question (that died) was 8 years old, in use for all that time by me. So I think I got my money's worth out of it! |
| 23:11.14 | fenrus | ChannelZ, must have been a hell of a box back then |
| 23:11.56 | ChannelZ | Yeah it was! A big green Alienware boc built in a Tyan 2462 motherboard (dual Athlon) |
| 23:12.23 | ChannelZ | Great machine, sans completely bizzarre power supply. |
| 23:13.30 | ChannelZ | It's gotten a complete organ transplant now (new motherboard, CPU, RAM, PSU) |
| 23:13.44 | fenrus | =) |
| 23:14.08 | boodu | I am a beginner with asterisk, what is the utility of a sound card in the ippbx ? |
| 23:14.13 | ChannelZ | I might try to get a PS adapter cable and resurrect the board in another case for something.. it's still functional |
| 23:14.47 | ChannelZ | boodu: Live Music On Hold, spying on calls in the console.. |
| 23:14.57 | ChannelZ | Other than that, no so much |
| 23:15.18 | boodu | ChannelZ, all the service with sound ? |
| 23:15.25 | boodu | all right |
| 23:15.28 | \malex\ | i think i've heard of people using a soundcard for an overhead paging system |
| 23:15.38 | fenrus | i'd like a jazzband sitting next to my pbx playing live MoH |
| 23:15.41 | fenrus | that would be awsome |
| 23:15.44 | ChannelZ | Yeh that's another use. |
| 23:17.28 | \malex\ | fenrus: but you'd have to setup a system for the callers on hold to request particular songs |
| 23:17.43 | boodu | and what type of sound card's quality is needed by asterisk ? |
| 23:18.00 | fenrus | :) |
| 23:19.02 | ChannelZ | 0 quality.. you don't "need" a sound card in an asterisk server |
| 23:19.20 | *** join/#asterisk digiv (~mhess@sfcc-208-115-75-235.smartcity.com) |
| 23:20.23 | boodu | no card need for voicemail ? |
| 23:20.28 | boodu | :O |
| 23:20.31 | Naikrovek | nope |
| 23:20.41 | Naikrovek | not unless you wanna listen to it at the server |
| 23:20.51 | Naikrovek | usually people listen to voicemail over the phone |
| 23:21.11 | ChannelZ | The only reason you'd need a sound card is if you were running * on the same computer as a softphone and you needed it for the softphone. |
| 23:21.35 | boodu | ok |
| 23:21.37 | Naikrovek | my asterisk server doesn't have a sound card at all and i have about 80 phones running on it at a time. going to be 85 tomorrow |
| 23:21.38 | boodu | thank you |
| 23:21.40 | jstapleton | what might cause "No application 'Jabbersend' for extension" |
| 23:21.41 | *** join/#asterisk foo (~foo@unaffiliated/foo) |
| 23:22.07 | Naikrovek | jstapleton: somewhere in the dialplan for that extension is a jabbersend() line |
| 23:22.12 | foo | I'm on freepbx 2.4 + asterisk and when I make a call, the person on the other end has a bad echo. any thoughts on why? |
| 23:22.15 | boodu | you run the sound service (voicemail, ...) on the hardphone |
| 23:22.32 | Naikrovek | used for instant messaging |
| 23:22.33 | p3nguin | foo: The volume on the phone is too high. Turn down the volume. |
| 23:22.50 | Naikrovek | foo: if speaker phone, turn down the volume. or get a polycom phone. if not speakerphone i dunno |
| 23:22.54 | Naikrovek | either way #freepbx |
| 23:23.10 | jstapleton | Naikrovek, i know. I am trying to call Jabbersend. I get that error when I do. |
| 23:23.29 | foo | Naikrovek / p3nguin: no speakerphone, hmm, if you're talking about the general volume, that seems strange. *tries on another phone* |
| 23:24.02 | p3nguin | foo: Usually that is the problem, since SIP/RTP doesn't otherwise echo. |
| 23:24.32 | Naikrovek | the echo another person hears is caused by your end, usually. could also be that you have a hardware card without echo cancellation |
| 23:24.40 | p3nguin | Having the handset volume too high does cause echo. |
| 23:24.47 | Naikrovek | jstapleton: make sure the jabbersend application is installed |
| 23:24.58 | ChannelZ | Repating everything someone says to you causes echo |
| 23:25.21 | Naikrovek | echo happens where analog switches to digital and vice versa |
| 23:26.15 | ChannelZ | Anyways if the remote end hears echo, isn't it their problem out of your control? |
| 23:26.41 | Naikrovek | not usually |
| 23:26.53 | Naikrovek | echo is usually caused by the end that doesn't hear it |
| 23:27.06 | Naikrovek | according to a few places i read but cant' remember atm |
| 23:27.20 | Naikrovek | oh you were making a joke |
| 23:27.21 | foo | p3nguin / Naikrovek: weird, I just tried with another phone... no problem, must be a problem with that phone/volume too high. thanks |
| 23:27.23 | Naikrovek | hehe |
| 23:27.32 | Naikrovek | foo: what phone is doing it |
| 23:27.49 | foo | Naikrovek: eh, some phone we have in the office. not a voip phone |
| 23:28.02 | Naikrovek | foo: try from the same phone a couple times, see if it comes & goes |
| 23:28.08 | foo | Naikrovek: ok |
| 23:28.18 | p3nguin | Oh, not a SIP phone? |
| 23:28.36 | Naikrovek | yeah that would be something that could cause echo |
| 23:29.09 | p3nguin | Analog technology does have a tendency to echo without proper echo cancellation hardware. |
| 23:29.41 | ChannelZ | No, I meant that echo cancellation is to get rid of echo coming back from the other end. So if you call someone and the remote end is hearing echo, it might be your fault, but you can't do anything about it |
| 23:30.02 | ChannelZ | At least that was always my understanding |
| 23:30.25 | p3nguin | Yeah, if you hear echo, the other side is usually causing/creating it. |
| 23:38.12 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
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