00:28.51 | *** join/#asterisk jbot (ibot@c-24-1-99-18.client.comcast.net) |
00:28.51 | *** topic/#asterisk is Asterisk 0.7.1 Released! Please report any bugs at http://bugs.digium.com |
00:29.00 | FuzzyCat | jbot you slag! |
00:29.03 | | FuzzyCat: are you using Windows? |
00:29.03 | *** join/#asterisk davel (~davel@frobnitz.dsp-services.com) |
00:29.24 | FuzzyCat | jbot: shut it slapper |
00:29.28 | | FuzzyCat: I don't know, could you explain it? |
00:29.33 | FuzzyCat | no |
00:33.25 | mishehu | FuzzyCat: whats error 207? |
00:33.29 | mishehu | err issue |
00:33.32 | jimmyz | anyone got an idea how i can make all calls conference then dial the number they called? |
00:34.34 | mishehu | ~theanswer jimmyz |
00:34.37 | | jimmyz: 42 |
00:35.01 | *** join/#asterisk plc5_250 (~chatzilla@pcp03527486pcs.pntiac01.mi.comcast.net) |
00:35.04 | FuzzyCat | mishehu: http://bugs.digium.com/bug_view_page.php?bug_id=0000207 |
00:35.59 | UnixDawg | do you have to put anything in modules.con to get it to record to pgsl |
00:36.00 | mishehu | FuzzyCat: that does not like to work in links it appears |
00:36.00 | *** join/#asterisk brown (~brown@sp1024.rbs-p01.ewol.com) |
00:36.14 | FuzzyCat | ? |
00:36.16 | UnixDawg | do you have to load a module |
00:37.29 | jimmyz | mishehu: what was that |
00:37.34 | *** part/#asterisk ecd (~dslater@ool-44c17c85.dyn.optonline.net) |
00:37.35 | *** join/#asterisk Zebble (~Zebble@Sherbrooke-HSE-ppp3610678.sympatico.ca) |
00:37.50 | FuzzyCat | mishehu: its a SIP issue... if you make a sip call, pull the cable out of your phone (to simulate a client crash) reboot it and put the cable back in the channel never dies in * unless you restart, or do a soft hangup |
00:38.59 | FuzzyCat | that means if you do billing the billing goes on until you kill it |
00:39.23 | *** part/#asterisk _Yog_ (~magnus@hades.27b-6.de) |
00:42.19 | mishehu | FuzzyCat: hmmm... I did finally get to get the page up... so SIP doesn't do any checking to see if a connection is still alive or not... interesting. |
00:42.55 | FuzzyCat | yes.. i think it's the same for h323 and mgcp... but I've not tested those... |
00:44.16 | *** join/#asterisk simprix (~simprix@24.247.181.124.gha.mi.chartermi.net) |
00:44.27 | Tangent | I'm trying to work out where/how I can add in voicemail normalisation to make all the quiet messages louder when they're emailed to me... Would that be app_voicemail.c ? |
00:44.31 | simprix | when i use asterisk i get this with a sip xten client |
00:44.32 | UnixDawg | ok anyone out here get *69 working on the pbx |
00:44.41 | simprix | Jan 14 19:40:44 NOTICE[245776]: chan_sip.c:3635 register_verify: Peer '2000' isn't dynamic |
00:44.41 | simprix | Jan 14 19:40:44 NOTICE[245776]: chan_sip.c:5405 handle_request: Registration from 'simprix <sip:2000@10.0.0.244>' failed for '10.0.0.242' |
00:44.41 | simprix | Jan 14 19:40:44 NOTICE[245776]: chan_sip.c:3635 register_verify: Peer '2000' isn't dynamic |
00:44.41 | simprix | Jan 14 19:40:44 NOTICE[245776]: chan_sip.c:5405 handle_request: Registration from 'simprix <sip:2000@10.0.0.244>' failed for '10.0.0.242' |
00:44.42 | simprix | Jan 14 19:40:44 NOTICE[245776]: chan_sip.c:3635 register_verify: Peer '2000' isn't dynamic |
00:44.44 | simprix | Jan 14 19:40:44 NOTICE[245776]: chan_sip.c:5405 handle_request: Registration from 'simprix <sip:2000@10.0.0.244>' failed for '10.0.0.242' |
00:44.50 | Link | IS Build 0.7.1 done? |
00:44.51 | simprix | sorry |
00:44.53 | *** join/#asterisk PBXtech (Reggie@65.218.37.248) |
00:45.02 | FuzzyCat | Link: topic! |
00:45.13 | km- | I wish I remember where I put my tradewars registration code |
00:45.20 | Link | Is this build available? |
00:46.39 | simprix | can someone help me |
00:47.01 | Tangent | simprix: Depends ;) |
00:47.21 | simprix | i get this error Jan 14 19:44:00 NOTICE[245776]: chan_sip.c:5405 handle_request: Registration from 'simprix <sip:2000@10.0.0.244>' failed for '10.0.0.242' |
00:47.34 | simprix | but everything looks ok im using the xten client on os x |
00:48.30 | Tangent | I get those messages for one of my clients too.. but it all seems to work ok |
00:48.36 | TestMasTer | simprix, i`m having the same problems |
00:48.44 | simprix | mine won't log in |
00:48.51 | simprix | i can't get it working at all |
00:48.52 | FuzzyCat | shitty client? |
00:49.01 | simprix | xten x-lite |
00:49.19 | simprix | i had to do some stupid stuff last time i got it working but i don't remember what |
00:49.33 | TestMasTer | FuzzyCat, agreed |
00:49.37 | TestMasTer | but its needed atm |
00:49.59 | FuzzyCat | sjhpone no good TestMasTer? |
00:50.06 | Tangent | I had to get rid of host=dynamic on my BT101 to make it work |
00:50.07 | FuzzyCat | sjphone |
00:50.18 | simprix | did you just take out the line |
00:50.31 | TestMasTer | FuzzyCat, haven`t tryed it i`m using Xlite for other people and it works fine |
00:50.37 | mishehu | FuzzyCat: I am gathering that IAX does check to see if a link is still alive then? |
00:51.04 | Tangent | I made it host=192.168.254.15 instead... but I've no idea if that's valid... but it 'worked for me'(tm) |
00:51.19 | FuzzyCat | well,thats another problem atm... calls don;t appear to disconnect at either end in that situation |
00:51.40 | UnixDawg | test what error |
00:51.59 | UnixDawg | I keep getting a 484 |
00:52.32 | killerbee | yo yo yo... killerbee is in DA HOUSE |
00:52.45 | *** join/#asterisk Mike1 (~Mike@hoochie.digium.com) |
00:53.00 | Mike1 | Help! |
00:53.02 | Mike1 | =) |
00:54.23 | simprix | tangent when i do that i get this Jan 14 19:51:11 NOTICE[245776]: chan_sip.c:3635 register_verify: Peer '2000' isn't dynamic |
00:54.24 | simprix | Jan 14 19:51:11 NOTICE[245776]: chan_sip.c:5405 handle_request: Registration from 'simprix <sip:2000@10.0.0.244>' failed for '10.0.0.242' |
00:55.32 | Tangent | simprix: I just PM'd you the config for my BT101 |
00:57.12 | joako | Is it possible to detect a fax over IAX or SIP or only when the call comes from a zaptel device? |
00:57.32 | *** join/#asterisk mvand (~mvand@CPE-24-27-138-147.neb.rr.com) |
00:58.15 | mishehu | joako: why would you fax over VoIP instead of scanning in the document and emailing it or something? |
00:58.59 | jimmyz | we voip to a faxserver that e-mails them |
00:59.14 | jimmyz | got to get the call in to a fax server or machine |
00:59.48 | jimmyz | if your voip only how do you get faxes then |
00:59.53 | mishehu | faxing to a faxserver that then emails them? |
01:00.07 | mishehu | I'm still confused... |
01:00.13 | joako | nooo, I mean when you have a fax => extension, can asterisk detect that it is a fax call when its going over IAX? |
01:00.32 | jimmyz | don't understand that though.... |
01:00.42 | jimmyz | why not have an extension for faxes only |
01:01.21 | joako | dont ask me, ask whoever thought of that feature... |
01:02.06 | *** join/#asterisk dolbe (~fujitsu@pcp04566804pcs.jersyc01.nj.comcast.net) |
01:02.39 | jimmyz | mishehu: on my setup some extension has to receive the fax and we are voip only |
01:02.52 | mishehu | jimmyz: I am probably wrong about this, but I think that's a part of the zaptel and not asterisk itself that does the fax detection |
01:03.06 | jimmyz | ok could be right |
01:03.18 | joako | jimmyz: but does fax detection work over IAX? |
01:03.28 | jimmyz | joako: don't know |
01:03.33 | *** join/#asterisk Cripon (~nothanks@pcp02001453pcs.huntsv01.al.comcast.net) |
01:04.44 | mishehu | can't you create virtual zap devices? |
01:04.51 | mishehu | if so, you might be able to get that to work. |
01:04.53 | Cripon | http://www.yottadot.org/download.php?op=viewsdownload&sid=10 |
01:05.08 | jimmyz | joako: are you trying to share one inbound line for voice and data |
01:05.38 | jimmyz | data=fax |
01:13.47 | dalabera | Would anyone tell me where can I download the cisco 7960 image files to upgrade my phone besides http://www.loligo.com |
01:14.47 | dolbe | does asterisk support those? |
01:15.54 | simprix | when i dial extension 2999 which is supposed to go straight to voicemail it gives me a 404 error |
01:16.12 | simprix | and the console says Jan 14 20:10:51 NOTICE[245776]: pbx.c:1211 pbx_extension_helper: Cannot find extension context 'default' |
01:17.55 | bkw_ | Corydon76-work that patch works perfect |
01:18.03 | bkw_ | now we just have to get kram to approve the changes |
01:19.52 | UnixDawg | man asterisk is growing so fast |
01:20.27 | jimmyz | think i got my company to switch to it now |
01:20.41 | plc5_250 | hey folks - I'm having a bit of a problem getting asterisk to work with my (new) TDM-40 |
01:20.55 | bkw_ | ok now lets get sip debug to accept sip debug peername |
01:20.55 | bkw_ | "sip debug peername" |
01:20.58 | jimmyz | want to guess at the cost to record phone calls on two offices about 150 phones...record only no ccm |
01:21.01 | plc5_250 | Jan 14 19:32:36 ERROR[16384]: chan_zap.c:5189 mkintf: Unable to open channel 1: No such device |
01:21.41 | jimmyz | 110k |
01:27.08 | bkw_ | HAHAHAHA |
01:27.10 | bkw_ | chan_sip.c |
01:27.11 | bkw_ | #ifdef THE_SIP_AUTHORS_CAN_SUCK_MY_GONADS |
01:28.38 | jimmyz | how big are gonads |
01:29.32 | rollyson | hmmm. is there a way to kill MOH on a snom 200? |
01:30.12 | bkw_ | GOTTA be a way |
01:30.20 | bkw_ | sip debug [peername] |
01:30.29 | bkw_ | that will be handy if you have a loaded asterisk box |
01:30.31 | bkw_ | DAMN IT |
01:30.51 | *** join/#asterisk VoiceLynx (VoiceLynx@rrcs-central-24-106-64-175.biz.rr.com) |
01:31.03 | rollyson | hmm.. everybody leave the conf? |
01:31.51 | joako | yes, everyone's gone |
01:31.59 | joako | what do you mean by kill MOH on a snom 200? |
01:32.29 | *** join/#asterisk af_ (af@ip381-35-1.adsl.edisontel.com) [NETSPLIT VICTIM] |
01:32.29 | *** join/#asterisk Tangent (authdenied@connerdata-6.dsl.easynet.co.uk) |
01:32.29 | *** mode/#asterisk [+bbb *!*FearOfGOD@*.Level3.net *!*PooPoo@*.ntli.net *!*@*.va.client2.attbi.com] by capek.freenode.net |
01:32.29 | *** mode/#asterisk [+bbb *!*carlossan@*.attbi.com *!*NoHost@*.verizon.net *!*globalthr@*.attbi.com] by capek.freenode.net |
01:32.29 | *** mode/#asterisk [+bbb *!*shane@*.attbi.com *!*tofubar@*.attbi.com *!*@og.latency.net] by capek.freenode.net |
01:32.29 | *** mode/#asterisk [+b *!*dan@194.158.*.*] by capek.freenode.net |
01:32.29 | *** mode/#asterisk [+q sant!*@*] by capek.freenode.net |
01:32.41 | Tangent | Whee... ride the irc rollacoaster |
01:33.12 | rollyson | joako: I'd like to temporarily disable music on hold ;) |
01:33.26 | rollyson | so that when I put someone hold they will get dead air |
01:33.33 | ZX81 | why |
01:33.56 | rollyson | ZX81: in case one of the lines on hold is a conf. |
01:34.03 | ZX81 | aha |
01:34.13 | simprix | what module's do i have to load for the x100p |
01:34.29 | joako | this is when you press the hold button on the phone? |
01:34.37 | rollyson | yeah |
01:34.51 | *** join/#asterisk root (~root@24.214.255.57) |
01:35.06 | root | blah |
01:35.36 | simprix | how do i get the x100p to work |
01:35.41 | *** join/#asterisk angler__ (~angler@24.214.255.57) |
01:36.07 | plc5_250 | this is weird - working with a tdm40 (4 mod). I can dial from extension 1 to extension 4 no problem, but dialing from extension 4 to extension 1 gives a busy. |
01:36.24 | joako | create a new entry in your musiconhold.conf with an empty directory |
01:36.34 | joako | and in your extensions.conf do |
01:36.44 | joako | exten => xxxx,1,SetMusicOnHold(silent) |
01:36.50 | joako | exten => xxxx,2,MeetMe..... |
01:38.25 | *** join/#asterisk juice (~juice@user221.net1199.mo.sprint-hsd.net) |
01:38.28 | simprix | i have this in my zapata.conf how do i get this to transfer to voicemail right away and to accept incoming calls ; |
01:38.28 | simprix | ; Zapata telephony interface sample configuration file |
01:38.28 | simprix | ; |
01:38.29 | simprix | [channels] |
01:38.29 | simprix | ; |
01:38.30 | simprix | ; X100P plugged into PSTN |
01:38.32 | simprix | ; |
01:38.36 | simprix | context=incoming |
01:38.36 | jimmyz | is there anything wrong with the T400P? |
01:38.38 | simprix | signalling=fxs_ks |
01:38.40 | simprix | echocancel=yes |
01:38.42 | simprix | echocancelwhenbridged=yes |
01:38.44 | simprix | relaxdtmf=yes |
01:38.46 | simprix | rxgain=1.5 |
01:38.51 | simprix | txgain=1.5 |
01:38.52 | simprix | immediate=no |
01:38.54 | simprix | busydetect=no |
01:38.56 | simprix | callprogress=no |
01:38.58 | simprix | musiconhold=default |
01:39.00 | simprix | usecallerid=yes |
01:39.02 | simprix | callerid=asreceived |
01:39.06 | simprix | channel => 1 |
01:39.22 | Tangent | simprix: put some entries for extension s in your [incoming] section of extensions.conf |
01:40.13 | simprix | ok i can't get asterisk to open now that i edited zapata.conf |
01:40.28 | simprix | do i need to load the modules first |
01:41.26 | Tangent | simprix: I load the modules from /etc/modules myself.. not sure if asterisk autoloads them |
01:41.58 | simprix | well now that i edited that asterisk won't load |
01:42.51 | plc5_250 | is IAX1 strictly TCP, or does it use UDP as well? |
01:43.37 | Tangent | simprix: you want wcfxo and zaptel modules loaded |
01:43.54 | simprix | Jan 14 20:40:41 WARNING[16384]: Unable to specify channel 1: No such device or address |
01:43.54 | simprix | Jan 14 20:40:41 ERROR[16384]: Unable to open channel 1: No such device or address |
01:43.55 | simprix | here = 0, tmp->channel = 1, channel = 1 |
01:43.55 | simprix | Jan 14 20:40:41 ERROR[16384]: Unable to register channel '1' |
01:43.55 | simprix | Jan 14 20:40:41 WARNING[16384]: chan_zap.so: load_module failed, returning -1 |
01:43.55 | simprix | Jan 14 20:40:41 WARNING[16384]: Loading module chan_zap.so failed! |
01:43.59 | simprix | i have those modules loaded |
01:44.55 | bkw_ | BUG 850 |
01:44.57 | bkw_ | soemoen care to try that |
01:45.51 | *** join/#asterisk Kumbang (~unknown@167.205.22.54) |
01:45.56 | Tangent | simprix: I can't see anything wrong with the config you pasted |
01:47.17 | plc5_250 | can someone try to route a call to me - I want to check my firewall forwarding and make sure I have it right. ip: 68.61.39.105 extension 104 |
01:49.21 | simprix | <PROTECTED> |
01:49.22 | simprix | Jan 14 20:46:28 WARNING[16384]: chan_zap.c:659 zt_open: Unable to specify channel 1: No such device or address |
01:49.22 | simprix | Jan 14 20:46:28 ERROR[16384]: chan_zap.c:5189 mkintf: Unable to open channel 1: No such device or address |
01:49.22 | simprix | here = 0, tmp->channel = 1, channel = 1 |
01:49.22 | simprix | Jan 14 20:46:28 ERROR[16384]: chan_zap.c:7162 setup_zap: Unable to register channel '1' |
01:49.23 | simprix | <PROTECTED> |
01:49.25 | simprix | <PROTECTED> |
01:49.27 | simprix | <PROTECTED> |
01:49.29 | simprix | <PROTECTED> |
01:49.31 | simprix | <PROTECTED> |
01:49.33 | simprix | <PROTECTED> |
01:49.37 | simprix | <PROTECTED> |
01:49.39 | simprix | <PROTECTED> |
01:49.41 | simprix | Jan 14 20:46:28 WARNING[16384]: loader.c:312 ast_load_resource: chan_zap.so: load_module failed, returning -1 |
01:49.41 | puzzled | simprix: stop flooding the chan |
01:49.44 | simprix | <PROTECTED> |
01:49.46 | simprix | <PROTECTED> |
01:49.50 | simprix | Jan 14 20:46:28 WARNING[16384]: loader.c:407 load_modules: Loading module chan_zap.so failed! |
01:49.52 | simprix | thats what it says when i try to load it |
01:49.52 | Tangent | Night all |
01:50.01 | puzzled | nite Tange |
01:50.04 | bkw_ | your config is WRONG |
01:50.59 | bkw_ | simprix whats up..? |
01:51.04 | simprix | bkw_ which config |
01:51.11 | simprix | i can't get asterisk to work |
01:51.13 | bkw_ | zaptel maybe |
01:51.21 | bkw_ | simprix what are you trying to setup?" |
01:51.55 | simprix | a x100p card |
01:52.01 | bkw_ | let me in the box |
01:52.07 | bkw_ | give me ssh and root and 2 min |
01:52.21 | bkw_ | priv msg it to me |
01:52.22 | Exomorph_ | bkw_: About your reply to bug 571... That was my plan to add per sip peer debuging.. But wanted to clean up the sip output first. |
01:52.44 | bkw_ | Exomorph_ ya changing the sipdebug output won't make it into cvs |
01:52.56 | bkw_ | because the output is the exact format as it is on the wire (or very very close) |
01:53.07 | *** join/#asterisk scat (scat@c-24-126-24-177.we.client2.attbi.com) |
01:53.11 | bkw_ | atleast not yet |
01:53.20 | Exomorph_ | bkw_: No its not... and its not at all readable. |
01:53.20 | simprix | ztcfg says channels not configured |
01:53.37 | scat | Hey -- I'm new to Asterisk & VoIP -- is there any Newbie FAQs out there? |
01:53.42 | bkw_ | simprix whats in your zaptel.conf /etc/ |
01:53.48 | bkw_ | Exomorph_ I have no problems reading it |
01:53.53 | Exomorph_ | bkw_: What I put in that patch was to make it a little more readable. |
01:53.59 | bkw_ | ah |
01:54.09 | bkw_ | what I would like to see is > if we are sending a packet |
01:54.09 | Exomorph_ | bkw_: Only because you have been reading it for how long now? |
01:54.18 | simprix | got it thankis |
01:54.19 | bkw_ | and < when we receive a packet |
01:54.28 | Exomorph_ | bkw_: That was in the plans... |
01:54.30 | bkw_ | Exomorph_ you get that? |
01:54.40 | bkw_ | I think tha twould make it more clear if its going in and or out |
01:54.54 | *** join/#asterisk kimo_sabe (foobar@ip68-107-131-120.tc.ph.cox.net) |
01:54.55 | bkw_ | but the peername hack would be SWEET |
01:55.01 | Exomorph_ | bkw_: It was kinda in that patch... Maybe I'll work on it some more agian. |
01:55.14 | bkw_ | Exomorph_ you need to be in the next conf call |
01:55.28 | Exomorph_ | bkw_: When is it? |
01:55.33 | bkw_ | the conf calls are very constructive |
01:55.39 | bkw_ | Exomorph_ we have one daily just about these days |
01:56.22 | Exomorph_ | bkw_: Hmmm Been busy with other stuff lately... But I'll try for the next one... Just send me a pm when you figure it out. |
01:57.03 | jimmyz | bkw: i need your mind fore a sec lol |
01:57.37 | bkw_ | nope fixing chan_zap tweaks |
01:57.44 | jimmyz | ok |
01:58.00 | bkw_ | can you not unregister a manager event? |
01:58.12 | bkw_ | SURE AS FUCK CAN |
01:58.19 | bkw_ | chan_zap doesn't even do any of those |
01:58.20 | bkw_ | man |
01:58.31 | bkw_ | or unregister most of the cli stuf |
01:58.35 | bkw_ | NO WONDER it can't be unloaded |
01:58.36 | bkw_ | hrm |
02:03.01 | jimmyz | anyone up for a dialplan ? |
02:04.01 | *** join/#asterisk igor__ (~igor@66.151.220.60) |
02:04.09 | scat | Can anyone direct me to some information on using asterisk & a linux pc (ie, plugging a headset into the soundcard) to make voip calls? |
02:04.20 | scat | Or anything at all similar to that? |
02:04.26 | bkw_ | WELL HOLY FUCK |
02:04.33 | bkw_ | you can unload chan_zap.so now |
02:04.34 | bkw_ | haha |
02:04.35 | scat | I'm sorry if this isn't the right channel for that |
02:04.46 | scat | But if someone could direct me to something else, I'd reall appreciate it |
02:04.49 | scat | +y |
02:05.17 | jimmyz | scat: you need a softphone to run on the linux pc? |
02:05.25 | bkw_ | asterisk*CLI> unload chan_zap.so |
02:05.25 | bkw_ | <PROTECTED> |
02:05.25 | bkw_ | <PROTECTED> |
02:05.25 | bkw_ | <PROTECTED> |
02:05.25 | bkw_ | <PROTECTED> |
02:05.26 | bkw_ | <PROTECTED> |
02:05.28 | bkw_ | <PROTECTED> |
02:05.34 | bkw_ | AND it doesn't crash |
02:05.34 | bkw_ | haha |
02:05.39 | igor__ | hi all, is it possible to set up limit of SIP incoming calls for _all_ users? so, in total i'll have no more than 10 SIP calls |
02:06.36 | bkw_ | yes |
02:06.40 | bkw_ | incominglimit= |
02:07.42 | bkw_ | now why can't I reload it |
02:08.52 | TestMasTer | bkw_, wanted to let you know something "Redhat and Fedora" You can reload the config files via asterisk -rx reload restart but you can not do reload restart and have it update well you are us asterisk -rcv |
02:11.11 | dolbe | anyone running asterisk on sparc arch? |
02:12.00 | bkw_ | TestMasTer speak english? |
02:12.05 | bkw_ | I don't understand that |
02:12.05 | bkw_ | haha |
02:12.22 | igor__ | bkw_: incominglimit is per user, I need something for the whole system or per peer basis |
02:12.31 | bkw_ | per user |
02:13.12 | bkw_ | ok you can unload chan_zap but you can't load it back now |
02:13.38 | TestMasTer | bkw_, Ok lol let me tell you again Redhat/Fedora Os is what this is on, Anyway if you try to reload ( in the * Console )the extensions config insted of Restarting asterisk It doesn`t work If you do asterisk -rx reload restart it works |
02:13.46 | TestMasTer | bkw_, is that better |
02:14.09 | rollyson | "My caps lock key turns itself on every time I try to type" |
02:14.12 | rollyson | ARGH! |
02:15.07 | bkw_ | asterisk -r |
02:15.08 | bkw_ | reload |
02:15.15 | bkw_ | or asterisk -rx reload |
02:15.17 | bkw_ | that works |
02:15.41 | TestMasTer | rx reload works Fine... From a shell Console but Reload from in the asterisk console Doesn`t |
02:15.58 | bkw_ | set verbose 10 |
02:15.59 | bkw_ | and type reload |
02:16.05 | bkw_ | bet verbose is like 0 |
02:16.45 | bkw_ | I bet this is because the fd isn't getting closed |
02:17.15 | TestMasTer | bkw_, Nopper that doesn`t fix it |
02:17.57 | TestMasTer | bkw_, Ok it worked that time |
02:20.04 | TestMasTer | bkw_, thanx for the info.... I personal didn`t care much if it worked.. for what i need to do but i just thought it was a bug so i thought i would tell you |
02:22.14 | *** join/#asterisk SladeAKT (~mirc@user-0c8h5qn.cable.mindspring.com) |
02:22.25 | *** join/#asterisk Kumbang (~unknown@167.205.22.54) |
02:23.15 | mishehu | bkw_: you guys surprised me with actually having a minor version not zero ;-) |
02:23.33 | bkw_ | haha |
02:23.42 | bkw_ | i'm trying to smash this bug in chan_zap |
02:23.48 | bkw_ | I have it where you can unload chan_zap now |
02:23.57 | mishehu | what is the bug? |
02:24.01 | denon | New! Asterisk 10.0! Even Easier! |
02:24.23 | bkw_ | oh its not a bug yet |
02:24.28 | bkw_ | i'm trying to fix it before i open a bug |
02:26.44 | bkw_ | where is Moc when you need him |
02:27.22 | SladeAKT | is it quiet or am i laggeed? |
02:29.33 | kamileon | quiet |
02:30.10 | *** join/#asterisk cypromis (~michael@80.51.246.186) |
02:30.41 | kamileon | rollyson : try to call my * server |
02:30.44 | rollyson | bkw_: where are we at in the next round of bug squashing? |
02:30.49 | kamileon | i need to test iax calls |
02:30.58 | kamileon | from anywhere out there |
02:31.06 | rollyson | kamileon: got a 700 number? |
02:31.10 | kamileon | no |
02:31.14 | kamileon | how do i get one |
02:31.19 | kamileon | i have an ip address |
02:31.35 | rollyson | www.iaxtel.com |
02:31.45 | kamileon | one sec |
02:32.02 | bkw_ | hrm |
02:32.09 | bkw_ | rollyson let me get JerJer to look at this one |
02:32.14 | bkw_ | and we can start a conf here in a few |
02:32.54 | SladeAKT | is there a test 700 number? |
02:34.35 | bkw_ | AHHHH HA |
02:34.41 | *** join/#asterisk Lee3 (~khgg@211.24.146.12) |
02:35.04 | jimmyz | sladeakt: use an 800 number 8000-555-1212 |
02:35.43 | bkw_ | no you noob |
02:35.46 | bkw_ | an iaxtel number |
02:35.50 | bkw_ | :P |
02:36.03 | *** join/#asterisk james1 (~james@hoochie.digium.com) |
02:36.12 | SladeAKT | actually i did that before i set up with voicepulse. I'm more interested in a decent fwd number to test to. |
02:36.21 | james1 | I am ready to install fedora for my asterisk project |
02:36.24 | Lee3 | hi bkw_ |
02:36.26 | jimmyz | oh |
02:36.31 | james1 | is it better to choose a custome pacakge? |
02:36.44 | james1 | or I shall go for a workstation |
02:36.51 | jimmyz | it was a test number to try |
02:36.54 | james1 | i have a PII 450MHz machine |
02:37.05 | bkw_ | NOW lets see if this works |
02:37.06 | james1 | I want to pout the most juice for * rather than something else |
02:37.25 | james1 | or should I go for custom installation? |
02:37.36 | james1 | please help.....thanks in advance |
02:37.40 | bkw_ | HOLY FUCKING SHIT I DID IT |
02:38.01 | kamileon | can i dial like 'username@server' |
02:38.06 | rob-- | Hi |
02:38.48 | SladeAKT | bkw just out of curiosity, what ya workin on? |
02:39.00 | bkw_ | SladeAKT fixing chan_zap so you can unload it and reload it from the cli |
02:39.04 | bkw_ | and it works |
02:39.04 | bkw_ | haha |
02:39.27 | jimmyz | what was wrong with the 800# to test |
02:39.38 | kamileon | call 6222 |
02:39.54 | SladeAKT | kamileon on iaxtel or fwd? |
02:39.54 | bkw_ | http://asterisk.bkw.org/patch2/chan_zap_unload_reload.diff |
02:40.07 | rob-- | How should I send audio to a channel? Do I have to worry about timing, or do I just send 1 voice frame for every voice frame received? |
02:40.18 | bkw_ | MUHAHAHAHAHAHHHHAHA |
02:40.18 | kamileon | SladeAKT : i really dont know |
02:40.33 | angler__ | kamileon your 6222 number is only available to people registered with our digium server |
02:40.44 | kamileon | ahh |
02:40.46 | rob-- | How do I know how big to make the voice frame (in samples) |
02:40.52 | kamileon | okay well how do i talk to other people |
02:40.57 | james1 | does it matter what type of installation I choose in terms of * performance on 450MHz PII....please let me know... |
02:41.03 | angler__ | sign up on iaxtel |
02:41.06 | kamileon | i talked to someone in germany this morning |
02:41.09 | kamileon | i did |
02:41.23 | angler__ | you can make local calls thru us |
02:41.24 | kamileon | but i have no gnophone, but i have a login/pass |
02:41.28 | angler__ | but not ld |
02:41.38 | kamileon | i want to call people with iax |
02:41.39 | SladeAKT | kamileon, what's your 700 number then? |
02:41.42 | kamileon | and let ppl call me |
02:41.47 | SladeAKT | and do you have it set up in *? |
02:41.53 | angler__ | sign up for iaxtel |
02:41.57 | kamileon | i havent registered my server with iaxtel yet |
02:42.00 | angler__ | people can call your local number |
02:42.06 | *** join/#asterisk tim27 (tim27@229-29.dr.cgocable.ca) |
02:42.13 | angler__ | your local number is 428-6222 |
02:42.16 | kamileon | i dont know how |
02:42.17 | kamileon | yeah |
02:42.18 | kamileon | ok |
02:42.40 | kamileon | so how would i call my friend in romainia ? |
02:42.50 | kamileon | or recieve my calls from iraq? |
02:42.54 | james1 | anyone ...please help me choose an installation type with fedora core on my PII 450MHz so as to get the best perf out of * |
02:43.05 | kamileon | cvs! |
02:43.21 | SladeAKT | kamileon they would have to be on the iaxtel or fwd network themselves |
02:43.23 | angler__ | kamileon only if you sign up at iaxtel or fwd or something |
02:43.36 | kamileon | thats what i mean |
02:43.40 | kamileon | i cant figure out how |
02:43.43 | kamileon | im slow you know |
02:43.47 | angler__ | i doubt mark will let you call international off our lines |
02:43.54 | kamileon | ive been reading for 3 hours! |
02:43.59 | kamileon | noooo |
02:44.06 | kamileon | i just want to call them with iaxtel |
02:44.08 | SladeAKT | sounds like you're just getting started then ;) |
02:44.12 | kamileon | over ip or something |
02:44.21 | angler__ | damn kdebase-audiolibs |
02:44.29 | SladeAKT | that's what it's set up for... |
02:44.29 | jimmyz | kamileon: you need a server there...our use sip computer to computer |
02:44.41 | kamileon | i have a server |
02:44.54 | kamileon | here with a live ip |
02:44.58 | james1 | anyone ...please help me choose an installation type with fedora core on my PII 450MHz so as to get the best perf out of * |
02:45.05 | jimmyz | with asterisk on it |
02:45.11 | kamileon | yeah |
02:45.15 | kamileon | 24.214.198.33 |
02:45.16 | tz-afk | james1: are you having performance issues right now? |
02:45.20 | SladeAKT | james1, do you know what differences the choices make? |
02:45.35 | james1 | my old installation was crap |
02:45.40 | james1 | i am starting from scratch again |
02:45.46 | SladeAKT | were you running X on the box? |
02:45.50 | james1 | got a Adtran 750 |
02:45.59 | rob-- | How should I send audio to a channel? Do I have to worry about timing, or do I just send 1 voice frame for every voice frame received? |
02:46.10 | jimmyz | what about where you are calling |
02:46.24 | rob-- | from an app |
02:46.25 | tz-afk | I am running * on a P233MMX... one channel granted but no performance issues, just stay away from iLBC :-) |
02:46.27 | james1 | SladeAKT, should i choose personal desktop, workstation, server, custom |
02:46.43 | kamileon | i run on a 200mmx |
02:46.50 | james1 | if custom, then can someone list me the minimum set of packages. |
02:47.02 | james1 | kamileon, what installation option did you choose |
02:47.05 | SladeAKT | given those options i would say workstation or server, more likely server |
02:47.19 | tz-afk | kamileon: how many channels? |
02:47.26 | SladeAKT | rob, what kind of app are you writing to the port with? |
02:47.42 | joako | james: custom, and just select development and kernel development (if fendora is anything like prior RH versions) |
02:47.46 | mvand | kamileon: how many GSM channels can you run on that? |
02:47.49 | tz-afk | crontibs: did you get your voicepulse working? |
02:47.53 | kamileon | james, i just installed debian sarge, installed nessecary stuff to install * and use the box (in console mode only) then i downloaded and installed asterisk from cvs.digium.com |
02:48.02 | kamileon | mvand : i dont know |
02:48.09 | tim27 | my 7960 is working now |
02:48.09 | tim27 | :) |
02:48.20 | kamileon | i have a x100p and a tdm400 with 2 modules |
02:48.34 | james1 | okay, serverkamileon, what is the necessary stuff? |
02:48.37 | kamileon | an OLD tdm |
02:48.40 | james1 | is there a list or something? |
02:48.46 | kamileon | ask the digium folk |
02:48.57 | kamileon | im hacking my * box together out of crap parts |
02:49.05 | james1 | hmmm |
02:49.11 | mvand | I have * running with an X100P on a P-233 and it works fine with 1 zap and 1 iax channel |
02:49.17 | angler__ | what kamileon is doing isnt recommended |
02:49.17 | kamileon | theres a hardware list on asterisk.org somewhere |
02:49.18 | angler__ | lol |
02:49.23 | kamileon | under install i think |
02:49.39 | doughecka | size of a box of ceegarrs |
02:49.40 | mvand | same setup didn't work with a p-166 |
02:49.59 | jimmyz | does the T400P work good |
02:50.44 | joako | i think the digium site recommends a 1ghz+ machine for T1/E1 cards |
02:50.56 | james1 | mvand, what is your system conf. (did you install as server?) |
02:51.06 | angler__ | 800mhz for TE410P |
02:51.48 | *** join/#asterisk james1 (~james@hoochie.digium.com) |
02:52.00 | james1 | ooops |
02:52.02 | tz-afk | doughecka: wow... a terabyte for under $1500 |
02:52.03 | angler__ | they need to hurry up and fix this bug in kde package |
02:52.03 | james1 | lost my connection? |
02:52.08 | doughecka | tz-afk: :)) |
02:52.16 | james1 | mvand, what was your install option? |
02:52.23 | james1 | did you install server as well |
02:54.06 | mvand | james1: That box is running RH9.0 ... |
02:54.46 | SladeAKT | imagine if you have to run raid with two those :) |
02:54.49 | james1 | did you install with server option or with custom packages |
02:54.52 | mvand | Did the Custom install with Kernel Development checked, then applied all RH updates |
02:55.07 | jimmyz | mvand: http://216.239.37.104/search?q=cache:zKp5qr9jIYAJ:members.lycos.co.uk/wipe_out/asterisk/asterisk_rh9_install-v1.3.pdf+red+hat+9+asterisk&hl=en&ie=UTF-8 |
02:55.27 | jimmyz | sorry http://members.lycos.co.uk/wipe_out/asterisk/asterisk_rh9_install-v1.3.pdf |
02:55.31 | james1 | so you have all access such as ssh., telnet etc? |
02:56.21 | bkw_ | wooohooooo |
02:56.27 | mvand | I do have ssh, but telnet is evil |
02:56.47 | james1 | thanks mvand and jimmyz for the link |
02:56.47 | angler__ | bkw_ ? |
02:56.55 | james1 | i will follow the same procedure and see how it goes |
02:56.56 | bkw_ | angler__ yes |
02:56.58 | mvand | I'm pretty sure that ssh is part of the base install. |
02:57.03 | james1 | i hope it works out as listed there |
02:57.06 | doughecka | Even if a man chops off your hand with a sword, you still have two nice, sharp bones to stick in his eyes. |
02:57.26 | angler__ | bkw_ was the wooohoo cause you fixed a bug? |
02:57.37 | doughecka | no |
02:57.37 | *** join/#asterisk Link (~svoto@h00095b093ebb.ne.client2.attbi.com) |
02:57.40 | doughecka | he _made_ one |
02:57.58 | angler__ | lol prolly |
02:59.25 | bkw_ | angler__ yes |
02:59.29 | bkw_ | I fixed one |
02:59.31 | bkw_ | bug 851 |
02:59.33 | bkw_ | go read it |
02:59.39 | bkw_ | and test if you feel like it |
02:59.47 | doughecka | oh, the bug with the bounty.. |
02:59.47 | doughecka | :P |
03:00.38 | bkw_ | no twit |
03:01.07 | mvand | james1: I almost forgot> You will need to track down the real mpg123 |
03:01.41 | mvand | Some RH versions install mpg321 and use a symlink. Other versions don't install it at all. |
03:02.01 | mvand | mpg321 plays back too slow |
03:02.31 | bkw_ | muahahhahah |
03:02.41 | doughecka | bwuhahahaha |
03:02.56 | tim27 | bkw_ can i setup a mailbox and use it on 2 phone ??? i suppose yes ??? |
03:03.27 | *** join/#asterisk Kumbang (~unknown@167.205.22.54) |
03:03.45 | doughecka | http://www.wired.com/news/medtech/0,1286,61889,00.html?tw=wn_story_top5 |
03:03.53 | doughecka | who wants to port iax client to it? |
03:04.00 | doughecka | so we can talk by thinking |
03:04.01 | doughecka | :P |
03:08.09 | *** join/#asterisk UnixDawg (~UnixDawg@69-160-1-159.bflony.adelphia.net) |
03:08.23 | mvand | doug: I won't use it if the driver is called "zap"tel |
03:08.55 | *** join/#asterisk A_Guy (~male@211.24.146.12) |
03:08.59 | *** part/#asterisk A_Guy (~male@211.24.146.12) |
03:09.09 | doughecka | lol |
03:09.29 | h3x | zaptel stands for zapata telephony |
03:10.08 | *** join/#asterisk ursenj_ (~ursenj@ns.ursey.com) |
03:10.22 | ursenj_ | what is new in .7.1 |
03:10.35 | mvand | yeah, I know. It was a joke, son. |
03:11.28 | doughecka | hahahahahahhahahahahhhahahaa |
03:12.57 | mvand | 100 electrodes in my brain, I don't want to hear the word "zap" |
03:13.02 | h3x | oh |
03:13.44 | ursenj_ | is there is anything major in 0.7.1 that would prompt me for any upgrade |
03:13.59 | rollyson | timed includes are fixed ;) |
03:14.30 | ursenj_ | g729 access to vmail and confrenece?? |
03:16.39 | *** join/#asterisk mvand (~mvand@CPE-24-27-138-147.neb.rr.com) |
03:16.50 | bkw_ | jrollyson |
03:16.53 | bkw_ | try this |
03:16.54 | bkw_ | http://asterisk.bkw.org/patch2/pri_kill.diff |
03:17.12 | bkw_ | we aren't kill the PRI D channel monitor thread in my previous patch this should kill it .. "should" being the keyword |
03:17.54 | doughecka | is * c or c++? |
03:17.57 | bkw_ | C |
03:18.04 | doughecka | ehm |
03:18.04 | doughecka | why |
03:18.07 | JerJer[ghost] | C |
03:18.07 | bkw_ | because |
03:18.09 | doughecka | isnt c++ better? |
03:18.16 | bkw_ | in who's eyes? |
03:18.22 | doughecka | programmers |
03:18.24 | bkw_ | ok lets write it in GWBASIC |
03:18.27 | bkw_ | NEXT!! |
03:18.30 | JerJer[ghost] | if you like to be that far away from the processor |
03:18.35 | doughecka | oh |
03:18.41 | rollyson | bkw_: this is in place of the other patch? |
03:18.49 | bkw_ | http://asterisk.bkw.org/patch2/pri_kill.diff |
03:18.54 | bkw_ | use that |
03:18.55 | ursenj_ | will * work with a 3COM VOIP 1102 |
03:18.56 | bkw_ | before I test it |
03:19.01 | bkw_ | ursenj_ no |
03:19.02 | doughecka | :P |
03:19.09 | rollyson | bkw_: instead of the other? |
03:19.11 | bkw_ | yes |
03:19.22 | bkw_ | the other didn't kill the pri d channel monitor this should |
03:19.25 | bkw_ | and exit cleanly |
03:19.50 | bkw_ | then again we may have to change where we kill it |
03:20.02 | *** join/#asterisk afg (joemillion@gafachi.rh.rit.edu) |
03:20.03 | bkw_ | but we have to kill it |
03:20.17 | bkw_ | haha |
03:20.23 | bkw_ | JerJer[ghost] sowwy hit the end button |
03:20.23 | bkw_ | haha |
03:20.33 | afg | Hey, I'm trying to get 2 T1 boards to work in a system |
03:20.43 | JerJer[ghost] | quick draw magraw |
03:20.48 | bkw_ | yep |
03:21.10 | bkw_ | jrollyson I sure hope this fixes it |
03:21.26 | afg | is there any reason that I can not put 2 T1 boards in a system? |
03:21.36 | bkw_ | yes |
03:21.39 | bkw_ | if you need two T1's |
03:21.44 | bkw_ | but at that price buy a Quad |
03:21.46 | jimmyz | irq's maybe just depends |
03:22.02 | bkw_ | shit wish I had PRI so I could test this |
03:22.16 | afg | each board gets its own IRQ |
03:22.21 | jimmyz | they are cheap just get one lol |
03:23.13 | ursenj_ | bkw_, get two T100p back to back |
03:23.21 | bkw_ | ursenj_ I wish I had another T100P |
03:23.23 | bkw_ | but I don't |
03:23.24 | afg | bkw_: were you talking to me? if so, do you mean that I can not run 2 PRIs from one system at the same time using 2x t1xxp's? |
03:23.40 | rollyson | bkw_: reloading. |
03:23.41 | bkw_ | afg no |
03:23.46 | rollyson | err, recompiling |
03:23.48 | bkw_ | rollyson call the conf in a few |
03:23.55 | bkw_ | msg me when you do |
03:24.09 | rollyson | I'm in for a sec |
03:24.19 | ursenj_ | bkw_, do you know d3i |
03:24.39 | *** part/#asterisk SladeAKT (~mirc@user-0c8h5qn.cable.mindspring.com) |
03:25.12 | *** part/#asterisk Kumbang (~unknown@167.205.22.54) |
03:25.16 | *** join/#asterisk Kumbang (~unknown@167.205.22.54) |
03:25.45 | afg | If I put two t1xxp in one system, one of the cards led will not light up or blink or anything |
03:26.08 | jimmyz | even if you switch slots? |
03:26.14 | angler__ | afg put it on a different slow |
03:26.15 | angler__ | slot |
03:26.17 | afg | I do not believe it to be the card or the PCI slot, because if I remove the other card, then the one that was not working will work |
03:26.36 | jimmyz | might be bustmastering also |
03:26.39 | *** join/#asterisk brent21 (bdf@paalto-apx-1-144-201.penn.com) |
03:26.45 | afg | cat /proc/interrupts shows that they are on different interrupts |
03:27.02 | angler__ | afg are they taking interrupts? |
03:27.06 | afg | and ztcfg shows that the spans are configured, and asterisk loads the zap channels (i'm talking about when both boards are in) |
03:27.14 | afg | angler__ yes |
03:27.38 | Mike | hows 0.7.1 doing? |
03:27.44 | afg | unfortunately the site where I am working on this does not have internet, so I cant try moving the boards around now |
03:28.07 | afg | jimmyz: what do I need to do as far as the busmastering? |
03:28.21 | jimmyz | it would be in the bios |
03:28.29 | jimmyz | on or off |
03:28.44 | jimmyz | but you probably can't get to it if you are not there |
03:29.00 | kamileon | angler__ |
03:29.00 | afg | well I'll drive back there after I gather some possibilites |
03:29.00 | angler__ | ? |
03:29.18 | kamileon | can you help me with this iaxtel registering thing |
03:29.23 | kamileon | and getting iax calls? |
03:29.28 | afg | jimmyz: so I want to turn busmastering on? |
03:29.28 | kamileon | or can i already? |
03:29.42 | *** join/#asterisk Kumbang (~unknown@167.205.22.54) |
03:30.03 | angler__ | i think theres an example on iaxtel.com |
03:30.13 | bkw_ | hrm |
03:31.08 | ursenj_ | any sucessfuly tied a t100p to a ds1 non-pri card in a Definity |
03:31.14 | ursenj_ | any-anyone |
03:31.46 | *** join/#asterisk _jackhamr (~jackhamr@64.212.11.53) |
03:32.03 | brent21 | Are there any good writeups, howto's, etc on QoS configurations geared towards *. E.g. iax2, sip, etc? |
03:32.48 | kamileon | angler__ : i see... so what would my 700 number be 700-428-6222 ? |
03:33.30 | _jackhamr | helloooo |
03:34.03 | ursenj_ | ok simpler questions,.. will a t100p coneccted to a non-pri T1 |
03:35.18 | Mike | bkw_: should i ask to the ML about my panasonic pbx? |
03:35.27 | bkw_ | Mike they are going to tell you the same thign I told you |
03:35.34 | bkw_ | the PBX isn't and can't send the correct singalling |
03:35.43 | Mike | bkw_: why? |
03:35.56 | Mike | bkw_: what option do i have? if he hangs up the calls when he wants |
03:35.57 | angler__ | kamileon you have to register a number at iaxtel.com |
03:36.18 | angler__ | damn i got 3 monitors now working in linux but got 2 desktops |
03:36.52 | Mike | bkw_: even if i used sipuras with the panasonic phones the transfer and hold etc wont work |
03:37.01 | Mike | bkw_: so i can remove that old pbx |
03:37.25 | Mike | if i could remap the phones keys |
03:37.30 | Mike | that would be great |
03:37.34 | crontibs | mike which panasonic pbx do you got |
03:37.53 | Mike | crontibs: its a 308 |
03:37.59 | Mike | crontibs: its old and big |
03:38.30 | Mike | http://www.telephones-online.co.uk/itm00707.htm |
03:38.33 | Mike | that one |
03:38.46 | crontibs | i was looking at this other day |
03:38.48 | crontibs | http://www.twacomm.com/Catalog/Model_KX-TA624-5.htm |
03:38.49 | ursenj_ | don't plug you 12volt AccessPoint powersupply into your BT-101,.. it will smoke |
03:39.09 | afg | does anyone have 2 T100Ps running in the same system? |
03:39.26 | UnixDawg | http://www.voxilla.com/Article37-nested-order0-threshold0.phtml?POSTNUKESID=6c17b94343252b1114108da28d1ef8db |
03:40.37 | *** join/#asterisk chris007 (~chris007@hoochie.digium.com) |
03:40.42 | chris007 | sup all |
03:40.44 | _jackhamr | hi, i've got a new error on newer cvs chan_sip.c:3635 register_verify: Peer '1111' isn't dynamic |
03:40.44 | _jackhamr | <PROTECTED> |
03:41.10 | chris007 | i sent a few messages on the mailing list but really didnt get very far... |
03:41.54 | chris007 | i was wonderign if someone can point me in the right direction on setting up * for only outbound calling with screen pops when the call goes through for my agents |
03:42.15 | chris007 | with remote agent capibiliaties as well |
03:42.38 | chris007 | i know that my calling list has to be in the spool |
03:43.21 | chris007 | but i need to know if anyone has written a agi for importing a .csv file into the sql backend through a web page or something along thoes lines? |
03:43.28 | ursenj_ | what brands of handsets will work with * |
03:43.48 | *** join/#asterisk PBXtech (Reggie@65.218.37.175) |
03:45.06 | doughecka | "CNET News reports that security flaws have been found in products that use VoIP and text messaging, including those from Microsoft and Cisco Systems. |
03:45.31 | _jackhamr | This is because microsoft and cisco suck ass |
03:45.37 | doughecka | amen |
03:45.45 | doughecka | but tis giving voip a bad name |
03:45.49 | afg | Do I need to run modprobe twice if I have two T100Ps? |
03:46.01 | angler__ | no |
03:46.37 | afg | angler__ any ideas on how I can track down where the problem is? |
03:46.51 | _jackhamr | dough: Cisco gave voip a bad name when they released call manager. merrill lynch just dumped CCM and **10,000** 7960 phones because it sucked so bad. |
03:47.02 | doughecka | haahhahah |
03:47.08 | doughecka | wow |
03:47.23 | doughecka | whos merrill lynch? |
03:47.25 | kimo_sabe | _jackhamr: and I missed that surplus auction? damn :) |
03:47.28 | _jackhamr | stock broker |
03:47.31 | _jackhamr | investments |
03:47.37 | TestMasTer | Anyone know can i have a AGI script auto call me at a sertion time? |
03:47.45 | TestMasTer | Certain even |
03:48.28 | angler__ | afg you have another machine to try? |
03:48.46 | crontibs | TestMaster you mean like cron job that runs script to send call to asterisk |
03:48.59 | afg | angler__ no sorry |
03:49.04 | bkw_ | angler call me |
03:49.10 | TestMasTer | crontibs, Yes but already got the answer thanx though |
03:49.53 | afg | when there is only one T100P in it works fine |
03:50.03 | bkw_ | irq related issue |
03:50.13 | _jackhamr | kimo: I'll hook U up with some cysco phones ;) |
03:50.14 | afg | ie, card1 is in slot1, card2 in slot2. card1 does not work |
03:50.22 | afg | if I remove card2, then card1 works |
03:50.35 | afg | bkw_: cat /proc/interrupts shows that each card has its own IRQ |
03:50.42 | brent21 | doughecka, There are security flaws with analog lines too, e.g. I can hook up a phone to the Demarc on the side of your house. |
03:50.50 | doughecka | indeed |
03:50.56 | doughecka | dont care |
03:51.35 | brent21 | afg you have any extra slots |
03:52.31 | afg | brent21: yes, there is one more slot, I will try that, but I need to drive over to the site... |
03:52.41 | brent21 | afg, I had a problem with a Promise Array controller, everything was on its own IRQ, but my X100P's would crap out and I couldnt figure out why. Pulled the Array controller, rebuit the drives, and all works |
03:53.08 | brent21 | make sure they dont have anything goofy going on with the PCI cards in there, some cards can screew things up even when it looks like its on its own IRQ |
03:53.28 | afg | brent21: thanks, i'll try that, but I think the other slot shares with the ethernet |
03:53.45 | brent21 | in my situation, everything was on its own IRQ, but you gotta watch out for other devices |
03:53.46 | *** join/#asterisk heison (~heison@CPE000a01d49e6f-CM014300011132.cpe.net.cable.rogers.com) |
03:53.51 | brent21 | took me 2 weeks to figure that one out ;) |
03:54.12 | afg | I only have the 2 x T100Ps in that system... no other cards... of course there is onboard ethernet, vga, 2x serials, keyboard, 2x IDEs.... |
03:54.46 | afg | thanks, i'll drive over and try it out |
03:54.56 | afg | is there anything that else that I could try while i'm there? |
03:54.56 | _jackhamr | what causes this punk-ass error: chan_sip.c:3635 register_verify: Peer '1111' isn't dynamic |
03:54.56 | _jackhamr | <PROTECTED> |
03:55.04 | brent21 | if your not using the serials, disable them |
03:55.06 | h3x | make sure you dont put a board in the pci slot next to an agp slot |
03:55.13 | brent21 | I also disabled USB and firewire support on my server |
03:55.21 | TestMasTer | Anyone know where i can get sample.call agi script from? |
03:55.24 | brent21 | free up as much stuff you are not using, and let the PBX be a PBX |
03:55.32 | rollyson | _jackhamr : the host/ip doesn't match and you don't have host=dynamic set. |
03:56.03 | afg | I use one of the serials for console, i think I have the other disabled, i'll check to make sure I have everything else turned off, and try that move |
03:56.12 | doughecka | aha! |
03:56.15 | doughecka | serial console?!? |
03:56.19 | afg | is there any chance my motherboard doesnt support it? |
03:56.21 | doughecka | tis the problem! |
03:56.24 | afg | ? |
03:56.26 | doughecka | I think |
03:56.36 | doughecka | I read someplace that the serial console may cause problems... |
03:56.44 | doughecka | oh, is this a IRQ problem? |
03:56.45 | doughecka | nm |
03:56.53 | _jackhamr | rollyson: thanks :) I don't have dynamic set, I have specific IP's for each host defined with no host=dynamic in the record. |
03:56.54 | afg | well I'm not sure that it is an IRQ problem |
03:56.57 | doughecka | oh |
03:56.57 | *** join/#asterisk simprix (~simprix@24.247.181.124.gha.mi.chartermi.net) |
03:56.57 | atacomm | hows 0.7.x? i havent upgraded yet |
03:56.59 | doughecka | whats the problem? |
03:57.08 | doughecka | atacomm: its feeling fine, and sends it greetings |
03:57.09 | _jackhamr | ata: todays cvs works ;) |
03:57.16 | atacomm | lol |
03:57.22 | simprix | what in sip.conf allows a extension to make outbound calls over a x100p |
03:57.25 | afg | ztcfg -vvvvv works without errors, asterisk starts and loads up all of the zap channels without errors |
03:57.25 | brent21 | afg, yeah just try moving stuff around, shouldnt be the motherboard, but the promise arrays shouldnt caused a problem either (and they did) |
03:57.40 | doughecka | so whats the probme? |
03:57.46 | kimo_sabe | atacomm: aside from "ast_openstream: File demo-echotest does not exist in any format" it seems to be working for me |
03:57.46 | doughecka | problem |
03:57.56 | brent21 | simprix, it basically goes in extensions.conf |
03:58.03 | simprix | what is it |
03:58.17 | simprix | so i don't have to set anything in sip.conf |
03:58.21 | afg | well when there are two cards, the card which isnt working wont light up its LED, and wont run PRI |
03:58.38 | simprix | in sip.conf is it context |
03:58.48 | doughecka | ah |
03:58.49 | afg | if I pull out the second card, then the first card works with no problem (light lights up green and it runs PRI) |
03:58.51 | doughecka | I dont have a pri |
03:58.53 | brent21 | it could be, it depends on how you have Asterisk setup |
03:58.54 | doughecka | so cant say |
03:59.08 | brent21 | simprix, check out: http://www.digium.com/handbook-draft.pdf |
03:59.34 | swirlnets | bkw: why would someone not want a unique id on cdrs? |
03:59.41 | afg | one more thing to mention: when doing a cat /proc/interrupts, the number for the card which isnt working, in the column CPU is 0 |
03:59.42 | brent21 | the book explains everything really well, sorry to refer you to another place, but you really need to read that in order to understand how * works |
04:00.17 | brent21 | afg if you can set an IRQ in bios for PCI slots, try that |
04:00.25 | afg | someone had mentioned bus mastering earlier, does the T100P even utilize bus mastering? |
04:00.58 | afg | brent21: yeah I can do that, it shouldnt matter what IRQ I assign, should it? |
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04:01.36 | brent21 | no but if you force it to take an IRQ, and disable auto's on the other, it may help |
04:01.42 | brent21 | its really an experimenting thing |
04:02.26 | *** join/#asterisk Stuart (~Stuart@65.167.123.238) |
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04:02.42 | afg | thanks brent21 |
04:02.52 | afg | hopefully i wont be back later :)! |
04:03.45 | brent21 | stop back and say hello anyways and let others know what you did to fix it :) |
04:03.59 | brent21 | that way they dont have the same problem hehe |
04:05.01 | afg | will do, hopefully its not this serial console thing that I just starting reading on |
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04:11.19 | *** join/#asterisk Administrator__ (Administra@61.11.96.40) |
04:11.39 | Administrator__ | tim |
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04:12.43 | *** join/#asterisk doughecka (~rooot@adsl-68-133.lou.bluegrass.net) |
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04:19.07 | *** join/#asterisk hmodes (hmodes@66.92.231.2) |
04:19.37 | bkw_ | hmodes wasabi |
04:20.16 | *** join/#asterisk ^tvt^ (KiepLuHanh@65.218.37.175) |
04:20.51 | hmodes | hola bkw |
04:21.05 | hmodes | how is? |
04:21.32 | *** join/#asterisk sobol_ (~druid@80.51.246.186) |
04:21.51 | bkw_ | good |
04:22.31 | Mike | wasim: alive? |
04:22.43 | voip | ~seen stealth-man |
04:22.53 | | voip: i haven't seen 'stealth-man' |
04:22.53 | voip | ~seen stealth_man |
04:22.54 | | stealth_man <Stealth_Ma@80.83.133.249> was last seen on IRC in channel #asterisk, 5d 22h 17m ago, saying: 'how is going ?'. |
04:22.54 | bkw_ | ~seen wasim |
04:22.55 | | wasim is currently on #asterisk |
04:23.04 | bkw_ | wasim wake up boi |
04:23.25 | Mike | someone doing a farfone now? |
04:24.44 | *** join/#asterisk jsmith (~jsmith@smithfam.dsl.xmission.com) |
04:26.08 | Mike | bkw_: you saw the 250$ for bsd drivers? |
04:26.14 | bkw_ | its 950 now |
04:26.19 | bkw_ | read all the notes |
04:26.21 | Mike | who raised it? |
04:26.26 | bkw_ | read the bug notes |
04:26.31 | Mike | okok |
04:26.38 | Mike | are there any developers working on it no? |
04:26.41 | jsmith | Mike: people like bkw_ did... |
04:29.23 | *** join/#asterisk Koplin (~Koplin@hoochie.digium.com) |
04:29.36 | *** join/#asterisk jorgeraidel (~jorgeraid@ip-69-33-73-90.mia.megapath.net) |
04:30.41 | jorgeraidel | wow jajaj and i installed the 0.7.0 one hour ago |
04:30.45 | *** join/#asterisk _gorman (~lehmann@pD9E4ECA0.dip.t-dialin.net) |
04:33.27 | Koplin | how well does asterisk scale? |
04:33.31 | bkw_ | how far can you kick your server? |
04:33.58 | Koplin | very it intergrated into a football :) |
04:34.05 | hmodes | with or without the wheels on? |
04:34.46 | _jackhamr | how about on mars? |
04:34.56 | _jackhamr | did that damn rover get off of the lander yet? |
04:35.20 | _jackhamr | is avt=rfc2833 =true? |
04:35.57 | _jackhamr | are there any scientologists here? :) |
04:37.31 | *** join/#asterisk PlainWhiteTrash (~matt@user-12hcnqu.cable.mindspring.com) |
04:40.33 | *** join/#asterisk miek (~miek@hoochie.digium.com) |
04:41.12 | miek | hello? |
04:41.18 | Koplin | hi |
04:41.29 | hmodes | mushi mushi! |
04:41.34 | bkw_ | yo yo yo |
04:41.58 | miek | can someone answer a technical question for me? |
04:42.21 | _jackhamr | miek: shoot |
04:42.25 | bkw_ | we can try |
04:42.39 | miek | I'm getting a "max retries" error on asterisk when using the X-pro soft phone over nat |
04:42.43 | *** join/#asterisk Administrator__ (Administra@61.11.96.40) |
04:42.46 | *** join/#asterisk l-fy (~diana@home-25022.b.astral.ro) |
04:42.58 | _jackhamr | same here...i don't think it causes a performance issue? |
04:43.14 | _jackhamr | I think the chan_sip.c warning has something to do with nat traversal |
04:43.21 | miek | I can make call but the call is disconnected after a few seconds |
04:43.39 | _jackhamr | do you have the right codecs set up? |
04:43.40 | miek | OK. I should just ignore the error then? |
04:43.55 | miek | Hmmmm. Which codec should I use for the x-pro? |
04:43.57 | _jackhamr | I've been ignoring the error but it doesn't make me feel any more comfortable |
04:44.08 | *** join/#asterisk Administrator_ (Administra@61.11.96.40) |
04:44.10 | _jackhamr | I wish the error would go away :) |
04:44.24 | Administrator_ | hi |
04:44.49 | miek | Which codec should I use for the x-pro? |
04:45.21 | _jackhamr | what are you using now? |
04:45.38 | kimo_sabe | this barbietone is kinda cute, and actually seems to be working in my limited test. |
04:45.49 | kimo_sabe | Are the Snoms as ugly as the look in the pictures? |
04:46.02 | miek | GSM |
04:46.15 | _jackhamr | kimo: do you think a black monolithic slab is ugly? |
04:46.25 | kimo_sabe | _jackhamr: usually, yes. |
04:46.29 | _jackhamr | that's a shame. |
04:46.35 | _jackhamr | heheh |
04:46.49 | kimo_sabe | _jackhamr: I'm a bit of a sucker for shiny translucent bits too |
04:46.59 | _jackhamr | kimo: like the buttons on the barbietone? |
04:47.06 | kimo_sabe | _jackhamr: exactly |
04:47.13 | _jackhamr | jellies. |
04:47.19 | kimo_sabe | the not-depressing black helps too |
04:47.55 | _jackhamr | i remember my dress-in-black days ;) |
04:54.14 | hmodes | pfft |
04:54.29 | kamileon | hello |
04:55.21 | _jackhamr | hiya. |
05:00.01 | *** part/#asterisk ___log (~stats@asterisk.toad.net) |
05:00.02 | *** join/#asterisk ___log (~stats@asterisk.toad.net) |
05:02.00 | joako | bah, IMO the snom's dont look that bad |
05:04.03 | kimo_sabe | I need a sugar-mama to buy me a Cisco 7690 and an Snom 105 to play with |
05:05.26 | carrar | kimo_sabe, their out there |
05:06.23 | crontibs | those barbietones anygood or should i stick with sipura2000 |
05:06.28 | kimo_sabe | carrar: I know it. One of my friends is a "Stay at home dad", his wife is a Nurse Practitioner. Another just imported himself a nurse from Kentucky |
05:06.57 | joako | stick with the sipura |
05:07.11 | crontibs | joako good idea |
05:07.12 | crontibs | he |
05:07.13 | crontibs | h |
05:08.19 | joako | alot more features, and per port actually a few bucks less |
05:08.24 | joako | AND THEY ACTUALLY WORK |
05:08.35 | rollyson | crontibs: stay away from barbietones for now, they have a few more bugs to fix. |
05:09.31 | hmodes | when are they going to fix the ugly as sin bug? |
05:10.18 | bkw_ | hmodes which one? |
05:10.25 | bkw_ | yoru momma? |
05:10.27 | bkw_ | er your |
05:10.28 | bkw_ | haha |
05:10.59 | hmodes | pfffft |
05:11.25 | hmodes | you best watch yourself, my mum has quite a temper |
05:11.34 | hmodes | she's also a raving lunatic |
05:11.37 | hmodes | never a good combination |
05:11.48 | loko-moko | Anyone here own any companies big enough to afford donating some $$$ to kids with cancer charity? |
05:11.50 | UnixDawg | ok what has to be done to get cdr_postgres working |
05:12.11 | UnixDawg | or is it active by default |
05:12.12 | timecop | haha, they found vulnerabilities in h323 |
05:12.17 | timecop | does that mean they will finally kill that shit off? |
05:12.25 | bkw_ | timecop NOT |
05:12.29 | timecop | hehehe |
05:12.36 | bkw_ | we can only wish |
05:12.38 | timecop | bkw_: oh, you like h323? |
05:12.42 | bkw_ | NOPE |
05:12.53 | *** join/#asterisk roger1 (~roger1@hoochie.digium.com) |
05:14.08 | kimo_sabe | hmm, Polycom do SIP yet? |
05:14.17 | timecop | i wonder how sip session timers are doing |
05:14.49 | roger1 | I have exten => 2,3,Dial(ZAP/1/${EXTEN},30) but it appears the X100P card never hangs up, if I turn on "callprogress" then the time out works. Any idea why this is happening ? |
05:15.01 | timecop | "monitor bug" means I'll get mail about it right? |
05:16.44 | crontibs | any advantages of using h323 over sip |
05:16.50 | timecop | no |
05:16.57 | crontibs | as far as call quality |
05:17.12 | timecop | of course not, that has nothing to do with the protocol |
05:17.27 | *** join/#asterisk srinivas (~srinivas@61.11.48.70) |
05:17.31 | timecop | 8khz a-law is going to sound same over SIP or h323 |
05:17.35 | hmodes | h323 is teh devil! |
05:17.46 | hmodes | it sucks even more then sip, and that takes alot of effort |
05:18.00 | crontibs | ahh ok |
05:18.22 | crontibs | g729 codecs still avilable through Digium |
05:18.24 | timecop | haha, thats quotable |
05:20.34 | *** join/#asterisk voip (~voip@c-24-10-202-120.client.comcast.net) |
05:21.29 | hmodes | there needs to be some good motivation for the larger vendors *coughciscocough* to adopt iax2 |
05:22.09 | hmodes | an iax-speaking 7960 and as5300 would be quite sextastic |
05:22.17 | heison | how can i do variable substitution in sed? |
05:22.32 | heison | sed -e 's/${cur_ip}/${new_ip}/' /etc/ipnat.conf > /tmp/ipnat.conf |
05:22.33 | heison | this doesn't seem to work |
05:23.02 | heison | i've tried \${cur_ip} also... |
05:23.21 | hmodes | if the var is in a ' it will be taken literally |
05:23.28 | hmodes | put it outside |
05:25.16 | crontibs | iax2 sipura 3000 would rock |
05:25.20 | crontibs | no longer need sip |
05:25.24 | crontibs | just full 100% |
05:25.25 | crontibs | iax |
05:25.53 | hmodes | feh |
05:26.00 | heison | hmodes: outside where? |
05:26.03 | hmodes | i don't have much use for pots emulation |
05:26.16 | hmodes | nothing i'd want to hook to it works quite right anyway |
05:26.38 | hmodes | a shame there's no workable provision for data over voip-based pots |
05:26.50 | hmodes | not that i really mind the lack of modems in my life |
05:27.44 | hmodes | heison; i think 's/'${cur_ip'/'${new_ip}'/' might be interpreted correctly, don't quote me on that tho |
05:27.59 | hmodes | worth a shot at least |
05:28.35 | heison | not quite... it's okay, i'll look on google |
05:28.41 | UnixDawg | ok I get a errror wen adding cdr_pgsql.so to the addline in cdr |
05:29.00 | *** part/#asterisk kapejod (~kapejod@pD9E83EEF.dip.t-dialin.net) |
05:29.01 | UnixDawg | in the make file |
05:29.17 | UnixDawg | what has to be done to make pgsql work |
05:29.24 | UnixDawg | or is it all active |
05:29.49 | *** join/#asterisk Administrator_ (Administra@61.11.96.40) |
05:30.00 | UnixDawg | it give a -lpg -lz error |
05:30.01 | PlainWhiteTrash | Unixdawg, have you created a database with a table to store it? |
05:30.06 | jorgeraidel | alguien que haya trabajado con el X-lite? |
05:30.11 | UnixDawg | I have the db |
05:30.17 | PlainWhiteTrash | UnixDawg.. then you probably don't have the developer's libraries installed |
05:30.27 | UnixDawg | but when I try to make the cdr module it errors out |
05:30.30 | PlainWhiteTrash | If it won't compile you don't have the developer's libraries installed or they're in the wrong places. |
05:30.50 | PlainWhiteTrash | That's what I'm saying. If it will not compile, you do not have the libraries installed is the most likely problem. |
05:30.51 | UnixDawg | postgres developers lb |
05:30.59 | UnixDawg | ok |
05:31.11 | PlainWhiteTrash | yup. install those and try again |
05:31.52 | jorgeraidel | somebody know works with the X-lite? |
05:32.26 | UnixDawg | I dont find them ion the ports |
05:32.28 | UnixDawg | grrr |
05:32.39 | PlainWhiteTrash | ports for what platform? |
05:33.12 | UnixDawg | found them |
05:33.14 | UnixDawg | fbsd |
05:33.19 | PlainWhiteTrash | ugh. |
05:33.22 | PlainWhiteTrash | that's your problem. |
05:33.27 | PlainWhiteTrash | you didn't say that earlier. |
05:33.39 | UnixDawg | explain |
05:33.41 | PlainWhiteTrash | I don't know if they'll be properly found on *bsd. |
05:33.42 | bkw_ | lkajsdflajsowiejofalksdjfowiejfalksdfj |
05:33.48 | PlainWhiteTrash | You should have said you were on BSD... |
05:34.23 | PlainWhiteTrash | I'm not sure what location the libs will get installed to on *BSD, and i'm also not sure that the build system will find them at compile time on BSD (depends on where bsd sends them) |
05:34.29 | hmodes | falk! |
05:35.07 | UnixDawg | well I am installing we will see |
05:35.14 | PlainWhiteTrash | k |
05:37.11 | voip | How do I change this to make everything dialed with a 9 go out the zaptel? exten => _91800NXXXXXX,1,Dial(ZAP/1/${EXTEN:1}) |
05:37.34 | *** join/#asterisk cman (~cman1@202.51.74.250) |
05:38.50 | UnixDawg | _9X. |
05:39.11 | voip | the . means ininifty? |
05:39.22 | voip | infinity |
05:39.23 | *** join/#asterisk sLeEpLeSs (~sLeEpLeSs@hoochie.digium.com) |
05:40.06 | cman | hi... i just found out that my gs phone is not registered to my *??? |
05:40.11 | cman | how come? |
05:40.57 | UnixDawg | did you login to the phone and set it up |
05:41.11 | cman | yes i can login from brwoser.. |
05:41.21 | cman | but don't know why its not registered to *? |
05:41.28 | UnixDawg | ok and you set it all up ip dns server |
05:41.32 | cman | strange... it was working fine till yesterday evening |
05:41.41 | UnixDawg | do you have a sip exten setup |
05:41.51 | UnixDawg | with the secret= the password |
05:42.00 | cman | yes |
05:42.30 | UnixDawg | make sure the match |
05:42.37 | UnixDawg | then reset the phone |
05:42.48 | UnixDawg | and from cli type sip debug |
05:42.56 | UnixDawg | and watch the errors |
05:46.03 | angler__ | grrr.... |
05:48.17 | UnixDawg | yeah it says it neeeds postgers libs |
05:50.34 | *** join/#asterisk Kumbang (~unknown@167.205.22.54) |
05:51.32 | rollyson | bkw_: your box die? |
05:54.21 | bkw_ | yes |
05:54.25 | bkw_ | but why are you going in and out of the conf? |
05:54.30 | bkw_ | you are bouncing all over the place |
05:54.38 | bkw_ | angler IAX2/guest@asterisk.bkw.org/996 |
05:54.47 | rollyson | trying to reconnect, I'm not getting audio |
05:55.01 | bkw_ | rollyson restart * and see if it works |
05:55.21 | rollyson | no. |
05:55.23 | rollyson | argh |
05:55.30 | bkw_ | wonder whats up |
05:55.35 | rollyson | you guys still in ? |
05:55.45 | bkw_ | yes |
05:55.54 | bkw_ | you keep bouncing in and out |
05:55.55 | rollyson | you hearing me? |
05:55.58 | bkw_ | nope |
05:56.04 | bkw_ | who knows it may be my box |
05:56.05 | rollyson | bkw_: thought it was still dead |
05:56.18 | *** join/#asterisk cman (~cman1@202.51.74.250) |
05:56.27 | bkw_ | nope |
05:56.38 | rollyson | I'm going to shutdown -r now here. |
05:56.45 | bkw_ | damn |
05:57.01 | rollyson | actually |
05:57.06 | cman | why is my GS phone not being registered?? |
05:57.07 | rollyson | lemme try jeremy's conf first |
05:57.12 | cman | its strange... |
05:57.26 | rollyson | my IAX or my SIP is hosed |
05:57.35 | rollyson | or my phone |
05:59.25 | *** join/#asterisk Administrator__ (Administra@61.11.96.40) |
05:59.38 | *** join/#asterisk PilotPTK-Home (~trillian@pcp02587722pcs.shlb1201.mi.comcast.net) |
06:00.08 | *** part/#asterisk PilotPTK-Home (~trillian@pcp02587722pcs.shlb1201.mi.comcast.net) |
06:00.17 | *** join/#asterisk clive- (~pirch@rndf-ip-nas-2-p136.telkom-ipnet.co.za) |
06:00.32 | ReG-Hexer | cool |
06:00.35 | ReG-Hexer | re asterisk |
06:01.01 | *** join/#asterisk Inv_Arp (~junya@adsl-214-36-237.mia.bellsouth.net) |
06:03.14 | h3x | you know, i dont see any DACS documentation in zapata or zaptel config files |
06:03.24 | h3x | i suppose i could RTFS but... |
06:05.08 | *** join/#asterisk T` (~total@RAMY8.RES.cmu.edu) |
06:06.05 | ReG-Hexer | where can i see the changes in 0.7.1 ? |
06:06.17 | ReG-Hexer | ChangeLog? |
06:12.06 | angler__ | 100%[======================================================================>] 37,526,048 2.21M/s |
06:12.08 | angler__ | how nice |
06:12.50 | kimo_sabe | angler__: spiffy |
06:13.32 | voip | wow |
06:13.46 | voip | will you be my sugar daddy? |
06:13.47 | Cripon | angler: where's that from john? |
06:14.14 | hmodes | why can't i have that at home? |
06:14.23 | angler__ | i pulled that off a server from my server |
06:14.31 | angler__ | i wish that was at home |
06:14.39 | hmodes | like seriously, it's been at least 6 years since a significant bump in residential broadband speed :( |
06:14.49 | Cripon | angler: we |
06:14.51 | angler__ | Cripon do i know you? |
06:14.52 | hmodes | hell comcast is just now getting their caps close to what i had during the early @home days |
06:14.53 | hmodes | sigh |
06:15.09 | Cripon | angler: I'd hope so.. |
06:15.14 | angler__ | wish i could get mplayer working with faad right |
06:15.23 | angler__ | ahh |
06:15.26 | angler__ | whats up christian |
06:15.28 | kimo_sabe | hmodes: kazaa not leeching fast enough? There's not much modivation to go any faster |
06:15.34 | Cripon | angler: that's sad man |
06:15.37 | Cripon | :) |
06:15.39 | hmodes | pffft |
06:15.44 | hmodes | screw kazaa |
06:15.45 | angler__ | i didnt know your irc name |
06:15.57 | hmodes | i want more then two channels on my * box without serious degradation ;p |
06:16.20 | kimo_sabe | hmodes: there's room to grow on upstreams, but not really down streams |
06:16.26 | Cripon | angler: did you give mark that kid's resume? |
06:16.34 | hmodes | eh, i'm content with 3mbit |
06:16.37 | angler__ | ummmm |
06:16.39 | hmodes | i'd be much happier with 10 tho |
06:16.41 | angler__ | mail it to me again |
06:16.57 | angler__ | im not in windows and wont be for awhile |
06:17.00 | hmodes | the upstream really needs to be dealt with tho' |
06:17.12 | *** join/#asterisk Administrator__ (Administra@61.11.96.40) |
06:17.25 | angler__ | Cripon you get that routing sorted out? |
06:17.41 | hmodes | although i guess the larger comcast-like megaconglomos are prolly able to get a better price/mbit by gauranteeing async useage |
06:17.55 | Cripon | angler: for 300 to a pro.. he setup vpn between the boxes.. works like a charm. It's actually faster than it was. |
06:18.56 | Cripon | I'm just a Bigelow, and everywhere I go people know the part J's playin. |
06:19.07 | angler__ | haha wtf? |
06:19.37 | Cripon | I'm just a gigello |
06:19.43 | Cripon | gigelo |
06:19.51 | Cripon | i dunno |
06:19.57 | angler__ | i wish i was then i would be rich |
06:19.57 | hmodes | manwhore! |
06:20.33 | Cripon | I whore myself out to myself as often as possible. |
06:20.51 | hmodes | do you tip yourself well? |
06:21.01 | angler__ | lol |
06:21.14 | Cripon | I'm also a cheap bastartd |
06:23.22 | Cripon | Did everyone download this and give it a try from their Windows boxes? http://www.yottadot.org/download.php?op=viewsdownload&sid=10 |
06:26.33 | hmodes | oh my |
06:26.40 | hmodes | that's cute |
06:28.34 | *** join/#asterisk MadNachos (~nachos@rrcs-west-24-199-17-30.biz.rr.com) |
06:29.34 | joako | cripon: what do i seutp on the asterisk side? |
06:29.47 | Cripon | just a user in manager.conf |
06:30.13 | *** join/#asterisk Administrator__ (Administra@61.11.96.40) |
06:30.15 | clive- | joako what are you tryoing to set up |
06:30.33 | Cripon | clive: http://www.yottadot.org/download.php?op=viewsdownload&sid=10 |
06:31.59 | bkw_ | wooohooo |
06:32.04 | bkw_ | bug 851 guys go test it |
06:32.05 | bkw_ | :) |
06:32.43 | rollyson | bkw_: nice work. |
06:33.03 | bkw_ | thanks for testing the PRI goodie |
06:33.04 | bkw_ | s |
06:33.56 | clive- | cripon whats the URL? |
06:35.02 | rollyson | bkw_: wheres the bugs on being unable to unload chan_zap |
06:35.03 | Cripon | clive: it's an * manager written in C# for windows. |
06:35.10 | rollyson | they need to be resolved ;) |
06:36.04 | clive- | cripin sounds interesting, I am a newbie, so I need a windoze and a linux box? |
06:36.39 | Cripon | clive: yes. It's coming along, it's not all that robust yet. But it works and it dials from outlook. |
06:38.28 | clive- | outlook is a mail client?....(which I dont use ..pegasus mail rules:)..so how does it work?, and which protocol? |
06:38.43 | rollyson | outlook is a *groupware* client |
06:38.46 | clive- | it sounds very interesting:) |
06:38.49 | rollyson | not just a mail client |
06:39.02 | clive- | I see |
06:39.08 | Cripon | clive: install it and play around with it. it uses the tcp/ip protocal |
06:39.11 | clive- | does it support SIP ? |
06:39.16 | joako | how do you dial with outlook?? |
06:39.26 | Cripon | Clive: it's not a soft phone.. it's a call manager |
06:39.29 | clive- | not outlook express? |
06:39.56 | hmodes | uurrrgh |
06:40.11 | Cripon | joako: http://www.yottadot.org/download.php?op=viewsdownload&sid=10 has an outlook dialer built in. |
06:40.12 | hmodes | so when does someone donate a 100mbit colo to digium? |
06:40.17 | rollyson | Cripon: I'll try that tomorrow. |
06:40.18 | clive- | ok, so basically a GUI for * |
06:40.23 | Cripon | hmodes: wouldn't that be nice. |
06:40.37 | hmodes | yeeeah.. |
06:40.44 | hmodes | t1s are so 90s ;p |
06:41.00 | *** join/#asterisk Kumbang (~unknown@167.205.22.54) |
06:41.02 | Cripon | clive: it's an interface. It's not as "operator" oriented as gastman yet, but that's coming. |
06:41.11 | TestMasTer | Is there anyway to give Global Access to asterisk -rx? |
06:41.21 | hmodes | that and i bet it would make their lives less annoying if i wasn't mooching off cvs on the same connection they do their thing |
06:41.23 | joako | yes, i installed it, now what? |
06:41.27 | TestMasTer | what i mean is so apache has access to beable to run asterisk -rx |
06:41.41 | joako | what does digium need a 100mbps colo for? |
06:41.51 | kimo_sabe | TestMasTer: sudo? |
06:41.52 | bkw_ | blah timmah |
06:41.54 | Cripon | hmodes: mark told john and I today that they do 9 million email a day over that pipe, plus cvs. |
06:41.56 | clive- | cripon, sounds great....good luck with it...I am not at home where my * box is, but I will keep it in mind.... |
06:41.59 | TestMasTer | kimo_sabe, sodo? |
06:42.01 | TestMasTer | sudo? |
06:42.03 | hmodes | yeeeah |
06:42.13 | angler__ | Cripon lmao |
06:42.15 | hmodes | between maillists and cvs i bet that poor line is constantly lit |
06:42.37 | angler__ | mark did say that though :) |
06:42.54 | joako | what are they running right now? |
06:43.02 | Cripon | fractional t1 |
06:43.22 | Cripon | how many data chans john? 12? |
06:43.29 | rollyson | bkw_: so, how long do you think its going to be before we have to do 0.7.2 ;) |
06:43.40 | bkw_ | rollyson hahah |
06:44.00 | angler__ | Cripon somethin like that |
06:44.12 | joako | hmm i would actually consider giving them a dedicated machine |
06:44.30 | TestMasTer | kimo_sabe, What is sudo |
06:44.39 | bkw_ | super user do |
06:44.47 | Cripon | on how large a trunk? then they run the problem of security. |
06:44.47 | joako | anyways, how do i get this thing to dial from outlook?? |
06:44.55 | bkw_ | its a way to give users su access to some stuff |
06:45.01 | Cripon | joako. you have to have both items installed |
06:45.03 | kimo_sabe | TestMasTer: http://freshmeat.net/projects/sudo/ |
06:45.06 | joako | yes |
06:45.23 | *** join/#asterisk ^sly^ (~jelque@adsl-67-66-120-174.dsl.ltrkar.swbell.net) |
06:45.34 | Cripon | is astring up and running? do you have a little yellow * in your taskbar? is it connected to asterisk? |
06:45.38 | angler__ | Cripon what your wife cook for dinner tonight? |
06:45.40 | joako | yes.... |
06:45.48 | TestMasTer | kimo_sabe, thanx |
06:46.00 | hmodes | sudo is a godsend and teh devil at the same time |
06:46.01 | Cripon | angler: I cooked chilicheese burritos |
06:46.04 | *** join/#asterisk adkr (~adkr@hoochie.digium.com) |
06:46.11 | joako | cripon: was your question directed at me? If I did anything it would be 10mbps and i'd give them 2 or 3 mbps of bandwidth... |
06:46.13 | hmodes | specifically, i really need to ditch %sa ALL=/bin/su |
06:46.22 | hmodes | *sigh* |
06:46.29 | angler__ | Cripon dang i had some cookies and now im eating cereal |
06:46.32 | rollyson | hmm. the only resturant in range is the one across the parking lot |
06:46.50 | Cripon | angler: sorry man.. you need to make more money.. come to work with me. |
06:46.59 | angler__ | Cripon hahaha |
06:47.06 | *** join/#asterisk brc007 (~brc007@ip68-109-131-42.ph.ph.cox.net) |
06:47.12 | angler__ | Cripon ill work with you on the side |
06:47.15 | joako | ok i figued it out, you added a button to my outlook toolbar |
06:47.16 | Cripon | angler: I'm so rich, It's hormel chili everyday. |
06:47.23 | Cripon | joako: bingo |
06:47.25 | angler__ | haah |
06:47.35 | hmodes | someone should hire hmodes! |
06:47.40 | hmodes | he hates his current job |
06:47.43 | angler__ | lol |
06:47.44 | joako | when you right click a contact there an option to call contact |
06:47.46 | hmodes | or is that the definition of 'job'? |
06:47.48 | joako | you should use that... |
06:47.52 | angler__ | does anyone like their job? |
06:48.06 | Cripon | joako: you can't mess with the right click menu's in windows.. so it goes to the taskbar |
06:48.08 | angler__ | JerJer prolly does |
06:48.20 | joako | look @ it |
06:48.23 | Cripon | menu's in outlook I mean.. not windows |
06:48.23 | hmodes | i would enjoy my job if there weren't so many stupid people negating everything i do |
06:48.35 | kimo_sabe | angler__: yes! |
06:48.40 | joako | right click contact -> call |
06:48.42 | Cripon | I know about the outlook dialer.. doesn't work for this. |
06:48.48 | bkw_ | angler I love my job |
06:48.51 | joako | then look at dialing options |
06:48.55 | Cripon | setup h323 and dial that way then. :) |
06:48.58 | joako | any way to add it there? |
06:49.14 | Cripon | no.. you can't add right click menu items in outlook |
06:49.31 | angler__ | i like calling bkw_ and bugging him thru out the day |
06:49.32 | joako | no, add it as an option where the modems / h323 are |
06:50.04 | Cripon | ah.. see.. that only works in outlook.. so it's the same boat |
06:50.33 | hmodes | wait wait, love and job in same sentence? |
06:50.35 | hmodes | that's possible? |
06:50.50 | bkw_ | yes |
06:50.59 | hmodes | neat |
06:50.59 | bkw_ | very possible |
06:51.25 | bkw_ | some days I hate it.. but most days I love it |
06:51.28 | bkw_ | I work from home... |
06:51.39 | hmodes | oooh |
06:51.50 | angler__ | bkw_ is lucky |
06:51.51 | hmodes | that's gotta be nice |
06:51.57 | pros12 | bkw: what kind of work do you do besides this? |
06:51.59 | kimo_sabe | hmodes: what's not to like about working at a liberal arts college that's like 80% female? |
06:52.09 | hmodes | i've tried to explain to my work how unnecessary it is that i drive 100mi each way to get there.. they don't seem to get it. |
06:52.49 | bkw_ | pros12 I am the network bitch for an ISP |
06:52.55 | bkw_ | I do it all...... |
06:53.12 | pros12 | ahhh.. be my bitch too... lol |
06:53.51 | joako | ok if i setup this thing propery, should i get something on the console when I dial? |
06:54.00 | bkw_ | um yes |
06:54.05 | joako | i dont.... |
06:54.19 | *** join/#asterisk dias (~dias@hoochie.digium.com) |
06:55.46 | Cripon | joako: what do you mean, on the console.. it will just initiate a call to the device you specified for it to use, then call the # you told it to dial |
06:56.16 | pros12 | is iaxtel down? |
06:56.22 | bkw_ | down like a french whore |
06:56.27 | pros12 | :( |
06:56.38 | jrollyson | hmm... is mark still around? |
06:56.45 | bkw_ | nope |
06:56.46 | bkw_ | why? |
06:56.49 | bkw_ | got a problem? |
06:56.53 | jrollyson | #738 |
06:57.01 | pros12 | i cant make pstn calls now.. |
06:57.01 | joako | but on the asterisk console it should say SOMETHING, right? |
06:57.17 | *** part/#asterisk benngard (~mabe@81.26.235.3) |
06:57.30 | Cripon | joako: is it doing nothing? |
06:58.15 | hmodes | hrmm |
06:58.30 | hmodes | either i'm being retarted or your zap patch b0rk muh shit |
07:00.07 | Cripon | what's everyone's favorite softphone? |
07:00.31 | pros12 | bkw: can i pm you? |
07:00.34 | jrollyson | iaxcomm |
07:00.48 | jrollyson | hmodes: lol |
07:01.20 | bkw_ | pros12 no |
07:01.24 | pros12 | k |
07:01.30 | bkw_ | keep it in the channel please |
07:01.36 | *** join/#asterisk scat (scat@c-24-126-24-177.we.client2.attbi.com) |
07:01.36 | bkw_ | I have too many things going on right now |
07:01.37 | bkw_ | :P |
07:01.47 | pros12 | lol np |
07:03.25 | jrollyson | hmm. |
07:03.33 | jrollyson | I need to get a sound file |
07:04.01 | jrollyson | "Due to your continued abusive comments, we are now forced to terminate this call. Thank you for calling, have a nice day." |
07:04.30 | jrollyson | and set that up as ext. 666 |
07:05.02 | h3x | haahhaha |
07:05.56 | pros12 | try using this. |
07:05.59 | pros12 | http://www.rhetorical.com/cgi-bin/demo.cgi |
07:07.15 | jrollyson | heh |
07:07.54 | pros12 | like that.. |
07:09.23 | pros12 | download the wav and use it in * |
07:09.39 | carrar | somewhat illegal |
07:09.46 | pros12 | ???? |
07:10.13 | jrollyson | those wavs are only for non-commercial use ;) |
07:10.28 | pros12 | well use it for home |
07:15.31 | hmodes | alrighty bkw, clean checkout, compile, run, stop now, patch, clean, compile, run; |
07:15.41 | hmodes | Jan 15 02:16:41 ERROR[1024]: chan_zap.c:7513 load_module: zapata.conf has changed since last startup. Will not load chan_zap.so |
07:15.52 | hmodes | i haven't touched zapata.conf in months |
07:16.19 | hmodes | + unload chan_zap between run and stop now that is |
07:25.02 | *** join/#asterisk Gazzas (~ghendler@c211-28-134-93.eburwd3.vic.optusnet.com.au) |
07:26.39 | kamileon | hello |
07:27.32 | MSpin | hmodes: I got the same issue...bkw found the bug and is fixing it tho |
07:27.53 | kamileon | bkw_ still having that fax issue? |
07:29.41 | hmodes | ah, well then i feel doubly useless |
07:29.42 | hmodes | pfft |
07:38.22 | kamileon | can someone call 7004286222 |
07:39.50 | hmodes | hrmm, my mwi still doesn't seem to work |
07:40.07 | hmodes | am i some kind of freak or something? |
07:40.53 | Cripon | kamileon: is iaxtel working.. I just go to congestion after a timeout period. |
07:41.10 | *** join/#asterisk deexm (~CraZENiGG@adsl-64-118-253-166.netrox.net) |
07:41.59 | kamileon | wait, its fucked |
07:42.07 | kamileon | wont boot up atm |
07:42.20 | kamileon | sorry |
07:42.46 | kamileon | can i use those voices up there at home since im noncommercial |
07:43.05 | Cripon | kamileon.. I don't see why not |
07:43.13 | *** join/#asterisk RoyK (~roy@19.80-203-29.nextgentel.com) |
07:43.22 | kamileon | good ;) |
07:43.38 | kamileon | i want sexy british female |
07:43.53 | Cripon | kamileon: I'm not the authority, but I say you can use them for whatever you want. Just don't get caught for commercial applications. |
07:43.56 | RoyK | Guten Morgen, #asterisk |
07:44.04 | kamileon | true Cripon |
07:44.30 | kamileon | does anyone know if the tdm400 card and x100p will work in a smp box ? |
07:44.47 | Cripon | yes.. I use them |
07:44.57 | Cripon | dell poweredge 2500 |
07:44.57 | kamileon | smp? |
07:45.05 | kamileon | i have a quad ppr0 box |
07:45.08 | Cripon | dual 933 p3's |
07:45.13 | kamileon | and im running them in a p200mmx box right now |
07:45.19 | kamileon | i notice no faults yet though |
07:45.46 | kamileon | i want to just deck out the intergraph quadppro |
07:45.49 | kamileon | just to do it |
07:45.55 | kamileon | with as much as possible for debian |
07:46.04 | Cripon | kamileon: you in alabama? |
07:46.21 | kamileon | yes |
07:46.22 | kamileon | hsv |
07:46.24 | kamileon | you? |
07:46.36 | Cripon | kamileon: yes |
07:46.44 | kamileon | excellent |
07:46.51 | kamileon | why up so late? |
07:47.33 | kamileon | im kamileon.. dave.. ive posted a few times maybe |
08:06.27 | *** join/#asterisk adam (~arichards@gatekeeper.oremut02.us.wh.verio.net) |
08:10.05 | iewebguy_ | Hello, Can anyone suggest how to connect my ata 186 it is a former vonage device |
08:10.39 | voidptr_ | morning |
08:17.03 | clive- | iewe, it may be locked in |
08:17.26 | iewebguy_ | I have it unlocked... |
08:18.09 | iewebguy_ | It needs sip firmware? or does it already have it? |
08:18.11 | jrollyson | bleh, wrong window |
08:18.19 | clive- | cool, then just reconfigure it to point to the SIP server you want |
08:18.45 | iewebguy_ | ah.... yes there is one big button on the top. |
08:19.15 | iewebguy_ | Any reconfig notes available ? URL? |
08:19.29 | *** join/#asterisk vindex (ldm@zenon.apartia.fr) |
08:19.41 | clive- | http:// (ip-addyof-ata)/dev |
08:20.22 | iewebguy_ | k. |
08:21.02 | clive- | g'luck |
08:21.20 | clive- | bye all |
08:26.08 | *** join/#asterisk dxmdcc (~CraZENiGG@adsl-64-118-253-166.netrox.net) |
08:37.49 | jrollyson | anyone in conf? |
08:40.07 | kamileon | http://www.binarypimpin.com/room/ |
08:41.11 | hmodes | hah! |
08:41.27 | kamileon | i just moved in :( |
08:42.16 | hmodes | http://matrix.gs/random/desk.jpg |
08:42.19 | hmodes | tough call ;p |
08:43.05 | *** join/#asterisk oej (~opr@apollo.webway.se) |
08:43.46 | blll | I will have to take pictures of my home server room some time |
08:44.09 | hmodes | oh, i thought we were comparing desk sizes :) |
08:44.15 | *** join/#asterisk Kumbang (~unknown@167.205.22.54) |
08:44.16 | kamileon | no you should see my old one! |
08:44.54 | blll | hmodes: what speed processor do you have in that ultra 10? |
08:45.01 | Kumbang | ERROR[8192]: File cdr_addon_mysql.c, Line 298 (my_load_module): Failed to connect to mysql database asteriskcdrdb on localhost. |
08:45.04 | hmodes | 440 |
08:45.11 | hmodes | it lacks graphics foo tho :( |
08:45.28 | hmodes | there's 4x400 e450 coming next week :) |
08:45.31 | blll | ah, I need a 440 for mine |
08:45.31 | hmodes | the u10 is getting sold off |
08:45.32 | Kumbang | why did it happened, i do exactly what configurtion need for cdr_mysql |
08:46.44 | hmodes | nice blurry identification tho |
08:46.58 | blll | I have a couple |
08:47.16 | blll | used to work for a sun partner that went out of business, got a lot of sun gear at liquidation |
08:47.20 | hmodes | that should be a hot pic :) |
08:48.21 | hmodes | a martini and a lamp on a 450 between two couches.. that's what i call a still life. |
08:49.27 | hmodes | in the meantime, i shall go play in teh snow |
08:50.47 | *** join/#asterisk dolbe (~dolbe@pcp04566804pcs.jersyc01.nj.comcast.net) |
08:50.52 | blll | oh god |
08:54.55 | *** join/#asterisk RoyK (~asdf@213-187-164-3.dd.nextgentel.com) |
09:01.34 | hmodes | wow, it's nice n' snowy out |
09:01.36 | hmodes | that's fantastic |
09:02.13 | kamileon | where? |
09:03.35 | voip | Does anyone know how nuphone charges? |
09:03.47 | kamileon | out the ass |
09:03.51 | voip | is there a monthly free for a toll-free number plus the 2.9 cents? |
09:04.11 | voip | monthy fee |
09:04.55 | RoyK | hm. how many days wer there between 0.7.0 and 0.7.1? |
09:05.10 | kamileon | .1 ? |
09:06.07 | RoyK | see topic |
09:07.37 | kamileon | when is iax gonna be fixed |
09:07.39 | hmodes | hrmm |
09:07.49 | hmodes | i just spent close to $200 on underarmour |
09:07.55 | hmodes | that seems pretty excessive |
09:07.58 | kamileon | can some tell me how to record my voice greetings and such? |
09:08.12 | kamileon | kevlar? |
09:08.15 | RoyK | where the fsck is the official asterisk download page? |
09:08.25 | kamileon | there isnt one, cvs! |
09:08.33 | hmodes | it's less bullet proof then kevlar, but more fulfilling |
09:08.36 | *** join/#asterisk mmco (~irc@pD9E10B6E.dip.t-dialin.net) |
09:08.46 | kamileon | i sold my kevlar a month ago |
09:08.48 | RoyK | kamileon: then what's the fscking point of releasing versions? |
09:08.53 | kamileon | i didnt plan on getting shot |
09:09.02 | kamileon | cvs update |
09:09.12 | *** join/#asterisk yaboo (~jsirucka@203-213-113-146-vic.tpgi.com.au) |
09:09.32 | RoyK | kamileon: the point of using a release instead of the cvs is that you usually get a more stable version |
09:09.41 | kamileon | oh sorry |
09:09.45 | RoyK | dunno if that's the case with * but |
09:09.45 | RoyK | ... |
09:10.05 | RoyK | but then still - why are there releases if it's all a dynamic heap in the cvs? |
09:10.21 | kamileon | compliance? |
09:13.33 | RoyK | anyone that knows what's the status of zaptel on 2.6? |
09:14.03 | hmodes | i definately noticed special attention getting paid to mantis before .7 was released |
09:14.22 | hmodes | hopefully 1.0 will be truely stable. that would be quitehot |
09:14.40 | hmodes | today's cvs is getting quite close |
09:14.56 | hmodes | the only way i can make it crash is if i do something wouldn't normally |
09:15.08 | hmodes | although my damn mwi doesn't work ;p |
09:17.40 | RoyK | mwi? |
09:17.48 | hmodes | message waiting indicator |
09:18.09 | hmodes | when someone leaves me a message in voicemail2 my 7960's 'you've got mail' led doesn't light up for some reason |
09:19.11 | hmodes | i have a feeling someone swapped fields around in the sip notifies or something, and the 7960 doesn't turn it on for the new format |
09:20.10 | hmodes | i'm too damn lazy to file a formal bug tho, i figure it'll either fix itself or someone else will get irked enough eventually |
09:21.14 | hmodes | or i'll just perform shady hacking like i have for my previous problems |
09:21.29 | hmodes | i should prolly just make myself a diff against cvs |
09:30.33 | *** join/#asterisk deexm (~CraZENiGG@adsl-64-118-253-166.netrox.net) |
09:36.57 | kamileon | snow where? |
09:37.00 | hmodes | as an added bonus the t3 for office use at work is down, and it's certainly not work going in to fight over a couple t1s |
09:37.10 | hmodes | pennsylvania/newjersey |
09:37.43 | kamileon | i have a oc192 in my bedroom!! |
09:37.47 | hmodes | only a couple inches, but around here that's enough to call out of work over |
09:37.50 | kamileon | ph33r me!! |
09:38.23 | kamileon | someone suggested i put all my gear in the closet |
09:38.42 | kamileon | but i couldnt get to the back of the cabinet and still close the closet door. :( |
09:38.55 | kamileon | its my playroom anyways |
09:39.13 | hmodes | my closet has a closet |
09:39.16 | hmodes | it's quite amusing |
09:39.23 | kamileon | very cool. |
09:39.44 | hmodes | eh, not so much that as a good conversation peice |
09:39.59 | hmodes | apparantley at some point it was a former tenant's baby's room |
09:40.03 | kamileon | true enough |
09:40.08 | hmodes | i feel sorry for that kid ;p |
09:40.12 | kamileon | anyone have a qlogic scsi controller card? |
09:40.16 | kamileon | huh huh huh |
09:40.33 | kamileon | no one ever does |
09:41.04 | hmodes | got plenty in suns, none in leenooks tho |
09:41.08 | kamileon | my alpha wont install the digital unix i bought for it unless it has a 'bootable' scsi card installed so the 'firmware bios' can see the disks on it |
09:41.29 | kamileon | i have to boot milo on floppy |
09:41.50 | kamileon | then boot lilo from the ide disk, then mount the scsi disk |
09:41.53 | kamileon | wtf! |
09:42.09 | kamileon | AND |
09:42.15 | kamileon | it doesnt work with my kvm |
09:42.27 | hmodes | ouch? |
09:42.40 | kamileon | yeah, its a good box though |
09:42.48 | kamileon | runs redhat 7.1 i think |
09:42.52 | kamileon | or 7.0 |
09:43.29 | hmodes | not like any 7.x version was any different from any other |
09:43.35 | kamileon | true |
09:43.46 | kamileon | im wanting to go boot it now, damn... |
09:43.48 | hmodes | speaking of which, i need to get up on a whitebox clone next week |
09:43.56 | kamileon | i dont know why, it serves no purpose really |
09:44.16 | *** join/#asterisk digger_ (~digger@penguin.taide.net) |
09:44.20 | hmodes | the two just don't play well together |
09:44.33 | kamileon | i want to put my * hardware in my intergraph 650r |
09:44.57 | kamileon | hmodes : yes, and mine is a discontinued, unsupported platform |
09:45.04 | kamileon | im lucky to log in to it |
09:45.26 | hmodes | yezzz |
09:45.29 | kamileon | what can i possibly do with 4 processors, any suggestions? |
09:48.05 | hmodes | hide one of them and claim smp-ness? |
09:48.13 | kamileon | lol |
09:48.23 | hmodes | und now it be time for sleep |
09:48.25 | kamileon | seriously, what takes advantage of that? |
09:48.26 | hmodes | g'nite fockers |
09:53.04 | *** join/#asterisk dnc (~duncan@213.244.224.118) |
09:54.06 | *** join/#asterisk reseaux (~reseaux@host9-132.pool82105.interbusiness.it) |
09:54.34 | reseaux | Hi To ALL!!!Happy new year!! |
09:55.10 | blll | hmodes: I am hoping the same |
10:03.42 | karimd | does anyone know if phonecore is still in development? |
10:07.13 | *** join/#asterisk clive- (~pirch@rndf-ip-nas-2-p189.telkom-ipnet.co.za) |
10:14.11 | *** join/#asterisk geertn (geertn@193.194.136.225) |
10:15.31 | RoyK | ~seen kape |
10:15.36 | | kape <~kapejod@pD9E8297E.dip.t-dialin.net> was last seen on IRC in channel #asterisk, 21h 3m 12s ago, saying: 'ico: doesnt work yet'. |
10:15.36 | RoyK | ~seen kapejod |
10:15.36 | | kapejod <~kapejod@pD9E82915.dip.t-dialin.net> was last seen on IRC in channel #asterisk, 1d 22h 42m 43s ago, saying: 'sniffe: yes'. |
10:18.34 | *** join/#asterisk john (~john@hoochie.digium.com) |
10:20.48 | john | hello everyone...having a question, what do i need to download such that, i could in my asterisk CLI screen the digits i'm sending and receiving? |
10:21.45 | john | i downloaded already a fresh version, but it seems doesn't to work..if i dial an extension, my CLI screen won't move.. |
10:23.42 | *** join/#asterisk zoa (~john@213.219.141.63) |
10:24.01 | reseaux | john: have you start * with option -vvvvvvvvvc |
10:24.42 | john | yeah.. |
10:25.20 | john | also, i noticed that my t400p card doesn't want to turn green anymore.. |
10:25.35 | john | i do already modprobe it |
10:27.32 | reseaux | have you try a local loop on t400p port? |
10:28.21 | john | just a minute, i'll try to do that.. |
10:29.57 | karimd | is phonecore still being developed? |
10:30.38 | john | reseaux, still red.. |
10:41.05 | voidptr_ | argh, traditional pabx'es are a biatch! |
10:41.36 | voidptr_ | i cannot forward an extension without programming it on a port |
10:42.04 | jrollyson | voidptr: thats lame, what system? |
10:43.33 | stees | how can I pass a variable to a context and 'catch' it again in the context ? |
10:44.21 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
10:48.16 | voidptr_ | jrollyson : philips sopho |
10:48.40 | voidptr_ | it could be that i am too lame to make it work differently ofcourse :D |
10:54.16 | *** join/#asterisk xs (~CAT@a80-126-102-2.adsl.xs4all.nl) |
11:06.08 | stees | hm... |
11:06.27 | stees | why, oh why cant I just have a voicemail box that doesnt play it's own greeting |
11:07.03 | puzzled | stees: you can. don't remember what it was but do show applications and check out the voicemail apps and their settings |
11:12.40 | stees | puzzled: cool, got it |
11:12.50 | puzzled | enjoy |
11:12.51 | stees | Voicemail(s<extension>) |
11:12.57 | stees | preceed it with 's' :) |
11:25.59 | reseaux | john:Sorry...im try to install a server... |
11:26.17 | [Sim] | okay this is sick |
11:26.18 | reseaux | john:so...i still red |
11:26.34 | [Sim] | channel variables can get lost if you do a local to iax bridge |
11:26.51 | voidptr_ | :S |
11:26.52 | [Sim] | but not if you do a local to sip bridge |
11:27.04 | voidptr_ | "can"? |
11:27.07 | voidptr_ | not always? |
11:30.05 | [Sim] | well |
11:30.23 | [Sim] | it seems to be related to a very specific combination of things |
11:31.06 | [Sim] | scenario: I create a callfile in outgoing which connects channel Local/number to [context],s,1 with a variable set |
11:31.34 | [Sim] | if I do this to a local extension that is a SIP phone, everything is fine, and the variable can be used |
11:31.54 | *** join/#asterisk telenieko (~telenieko@80.224.224.228) |
11:31.55 | [Sim] | if I do this to a remote extension (that is routed over IAX) the variable is lost |
11:32.01 | telenieko | Hi everybody. |
11:32.21 | [Sim] | hello |
11:32.40 | [Sim] | voidptr; I noticed this before with local to CAPI. |
11:32.56 | reseaux | hi! |
11:33.16 | telenieko | Problem there: An H323 Client connects to asterisk and asks to be connected with a 2nd H323 client I use CALL application, but then Asterisk hold both calls it codingg-deconding is very slow. Is tehre abyway to 'transfer' the call so asterisk forgets about it conneting both parties directly?? |
11:33.19 | telenieko | Thanks ;) |
11:34.44 | *** join/#asterisk khan1 (~khan@203.82.51.38) |
11:35.00 | khan1 | hello |
11:35.19 | khan1 | i need some help |
11:35.45 | khan1 | my phones have registered with the server and yet i am unable to make a call between them |
11:36.02 | puzzled | hi [Sim] and voidptr_ |
11:36.11 | reseaux | khan1:what kind of problem is? |
11:36.13 | voidptr_ | hey puzzled |
11:36.40 | khan1 | well the last time i was here |
11:36.46 | voidptr_ | [Sim] : hummm ok :( |
11:36.49 | khan1 | some people told me of this site |
11:36.54 | [Sim] | hey puzzled |
11:36.55 | khan1 | which helped alot |
11:37.06 | khan1 | but my companys phones don't seem to work with asterisk |
11:37.14 | khan1 | they keep returning 403 NOT FOUND |
11:37.42 | reseaux | khan1:I think is possible... :-( but you can see some debug trace on *? |
11:37.58 | khan1 | i have |
11:38.19 | khan1 | see my server's ip is 192.168.0.23 |
11:38.23 | khan1 | one phone is 12 |
11:38.29 | khan1 | the other is .14 |
11:38.39 | khan1 | when i dial from .12 to .14 |
11:38.56 | khan1 | everything goes fine from .12 to .23 and .23 to .14 |
11:39.08 | khan1 | but .14 returns 403 NOT FOUND |
11:39.13 | reseaux | khan1:from * CLI you can see some information on incoming call? |
11:39.39 | khan1 | the only thing that seems out of place is that when .23 forwards the call |
11:39.48 | khan1 | it asks .14 to find .14 |
11:39.52 | khan1 | which ofcourse it can't |
11:40.26 | reseaux | khan1:i cant undestand.... |
11:40.34 | khan1 | which part |
11:40.35 | khan1 | ? |
11:41.43 | khan1 | how are u supposed to configure phones? |
11:41.51 | khan1 | you know name their ports? |
11:42.05 | reseaux | if you can call from .23 to .14 what you can see from CLI of * |
11:42.07 | khan1 | they're supposed to have the same names as those in the asterisk server right |
11:42.20 | khan1 | .23 is the server |
11:43.42 | reseaux | your telephone .14 cant call to any other telephone? |
11:44.09 | *** join/#asterisk Filace (~jon@pluto.geekpeople.net) |
11:44.38 | *** join/#asterisk khan1 (~khan@203.82.51.38) |
11:44.51 | Filace | anyone here used chan_capi with multiple BRI lines? |
11:45.59 | khan1 | so anyone |
11:46.05 | khan1 | can you please guide me |
11:46.18 | khan1 | u know like let me know of any web-site that might help me out |
11:46.30 | Filace | www.voip-info.org |
11:46.40 | Filace | http://www.automated.it/guidetoasterisk.htm |
11:46.53 | khan1 | should the sip proxy settings be on on the phone while using asterisk? |
11:47.28 | Filace | not unless you've setup a SIP proxy |
11:47.35 | reseaux | khan1:Check in this site is very usefull... |
11:47.36 | Filace | asterisk doesn't have one built in, afaik |
11:47.38 | khan1 | well i haven't |
11:48.36 | khan1 | so what they're supposed to be in the direct mode? |
11:49.42 | khan1 | yes i've done exactly what it says on these sites |
11:49.42 | khan1 | i've tried it all |
11:49.42 | khan1 | the phones register |
11:49.57 | khan1 | but return 403 NOT FOUND when i dial from one to the other |
11:50.08 | khan1 | the phones work fine on vocal |
11:51.40 | reseaux | khan1:i cant understand... sorry... |
11:52.14 | reseaux | khan1:you are able to register the .14 but you cant call another phone |
11:54.57 | khan1 | the phones show when i type sip show peers |
11:55.12 | telenieko | Problem there: An H323 Client connects to asterisk and asks to be connected with a 2nd H323 client I use CALL application, but then Asterisk hold both calls it codingg-deconding is very slow. Is tehre abyway to 'transfer' the call so asterisk forgets about it conneting both parties directly?? |
11:55.47 | *** join/#asterisk Aviaa_ (~Aviaa@test.incracow.com) |
11:55.48 | khan1 | when i dial from one to the other and sip debug is enabled i can see packets going back and forth between both phones but then a busy tone comes |
11:55.53 | khan1 | and 403 error comes |
11:56.20 | Aviaa_ | halo if on cisco i have set: dtmf-relay rtp-nte , what dtmfmode i must set into * ? |
11:56.48 | *** join/#asterisk illc0mmm (~illc0mm@142-56.34-65.tampabay.rr.com) |
11:57.17 | reseaux | khan1:have check the conf of sip ? |
11:57.24 | khan1 | yes |
11:57.43 | khan1 | i've set up both phones according to the configurations given at both of the above sites |
11:57.50 | *** join/#asterisk ico (~none@pD9531BAA.dip.t-dialin.net) |
11:57.54 | reseaux | the phone is the same? |
11:58.01 | khan1 | but i haven't setup any sip proxy |
12:28.55 | *** join/#asterisk yoyowon (~yoyowon@hoochie.digium.com) |
12:28.59 | yoyowon | hi |
12:29.45 | yoyowon | Does asterisk source include SIP Proxy Server? |
12:30.11 | yoyowon | anybody here? |
12:30.23 | yoyowon | tell me... |
12:31.12 | oej | Yoyowon: Not a full SIP Proxy server, but a SIP server and SIP user agent |
12:33.02 | john | hi reseaux....sorry, got some problems with our servers here... |
12:33.18 | john | yeah, i tried to loop back, led is still red |
12:33.37 | yoyowon | then... |
12:34.39 | yoyowon | Asterisk can connect SIP Proxy Server? |
12:34.39 | yoyowon | real Fully Proxy Server? |
12:39.47 | geertn | yoyowon: Please clarify your question |
12:42.25 | YoYo^ | *BURP* |
12:42.41 | yoyowon | Do i wanna know, asterisk can connect to real SIP Proxy Server? |
12:43.14 | jimmyz | how can i make all out bound calls go to a meetme room and then dail the number |
12:43.27 | jimmyz | where a manager could listen to any call he wanted to live |
12:44.00 | geertn | yoyowon: You mean like ser? |
12:44.33 | yoyowon | Yes.. |
12:45.46 | yoyowon | Sorry.. i can not use Eng.....^^ |
12:46.42 | geertn | yoyowon: You can register with ser from asterisk... you can also route calls from ser to asterisk. |
12:47.12 | yoyowon | correct.. |
12:47.57 | yoyowon | Do I Wanna konw... how route calls from 'sip proxy' to 'asterisk'.. |
12:48.49 | yoyowon | asterisk recieved sip Msg... But Forwarding outbound to external line in PRI(E1) |
12:49.21 | vaewyn | good 12:49UTC all :} |
12:49.28 | yoyowon | Not Forwarding |
12:49.41 | *** join/#asterisk _aggelos_ (~Messenger@cuscon4693.tstt.net.tt) |
12:50.42 | _aggelos_ | Hi everyone, |
12:50.42 | geertn | yoyowon: I'm not sure haven't worked with zap yet only sip. I guess you can setup an extension (with or withouth a prefix) in asterisk which dials out and route the call to the extension. |
12:51.16 | *** join/#asterisk xs (~CAT@a80-126-102-2.adsl.xs4all.nl) |
12:51.16 | _aggelos_ | I get this strange error from asterisk, it has the same installation as it's sister box: |
12:51.30 | _aggelos_ | Wait("Zap/1-1", "5") in new stack |
12:51.30 | _aggelos_ | <PROTECTED> |
12:51.30 | _aggelos_ | <PROTECTED> |
12:51.30 | _aggelos_ | <PROTECTED> |
12:51.38 | _aggelos_ | unable to execture ? |
12:51.50 | bkw_ | chmod 755 DUH |
12:51.55 | bkw_ | ok back to bed for me |
12:52.01 | bkw_ | SOMEONE ELSE TEST BUG 851 |
12:52.15 | yoyowon | hum.... |
12:52.19 | yoyowon | i see |
12:52.26 | _aggelos_ | [daniel@blue sbin]$ ls -l callback.pl |
12:52.27 | _aggelos_ | -rwxr-xr-x 1 root root 2754 Jan 12 00:31 callback.pl |
12:52.28 | yoyowon | Thanks Your comment. |
12:52.29 | vaewyn | _aggelos_: make sure it is chmod 755... and also make sure that if it has a "#!/...." line to tell what to run the script under that it points to a real program |
12:52.43 | _aggelos_ | that script is acctualy running, |
12:52.51 | _aggelos_ | only that it crashes asterisk every time it does that |
12:53.04 | yoyowon | i analyze asterisk.... |
12:53.37 | _aggelos_ | exten => s,1,Wait,5 |
12:53.37 | _aggelos_ | exten => s,2,System(/usr/sbin/callback.pl \"${CALLERID}\") |
12:53.41 | geertn | yoyowon: aha:) |
12:53.52 | _aggelos_ | that's my extensions.conf |
12:53.55 | yoyowon | geertn: ^^ |
12:57.59 | *** join/#asterisk mitya (~mitya@gw.asylumtel.com) |
12:58.02 | _aggelos_ | -- Executing System("Zap/1-1", "/usr/sbin/callback.pl 6961737") in new stackJan 15 08:50:45 WARNING[1226062640]: app_system.c:57 system_exec: Unable to execute '/usr/sbin/callback.pl 6961737' ended with -1 |
12:58.02 | _aggelos_ | <PROTECTED> |
12:58.16 | mitya | Hi everybody |
12:58.26 | _aggelos_ | hello |
12:58.33 | *** join/#asterisk coppice (~Steve@227.168.17.210.dyn.pacific.net.hk) |
12:58.35 | _aggelos_ | anyone any ideas ? |
12:59.13 | JerJer[ghost] | app_system !? |
12:59.15 | JerJer[ghost] | why? |
12:59.33 | YoYo^ | system(/usr/bin/perl /usr/sbin/callback.pl 5551212) |
12:59.52 | JerJer[ghost] | not even close to the right way to dive into a callback system |
13:00.11 | _aggelos_ | it's not the point, |
13:00.24 | _aggelos_ | the callerid is sent to another server via a socket, |
13:00.42 | Filace | does chan_capi support pickupgroups? |
13:00.56 | _aggelos_ | I pickit up via fxo card in one country, send it to another via internet, and get the callback request from there |
13:01.17 | JerJer[ghost] | you don't need app_system for that |
13:03.59 | _aggelos_ | I did now: exten => s,2,System(/usr/bin/perl /usr/sbin/callback.pl \"${CALLERID}\") |
13:04.00 | JerJer[ghost] | on the 'trigger' machine you run a AGI |
13:04.32 | JerJer[ghost] | that sends the call info to the manager port of the other asterisk box |
13:04.47 | _aggelos_ | and I get Jan 15 09:03:16 WARNING[1217669936]: app_system.c:57 system_exec: Unable to execute '/usr/bin/perl /usr/sbin/callback.pl \"6961737\"' ended with -1 |
13:05.02 | JerJer[ghost] | chmod u+x callback.pl |
13:05.34 | vaewyn | JerJer[ghost]: been there... told him that :} |
13:05.53 | _aggelos_ | strangest thing is when I run asterisk -vvvvgc (console) with no -r (remote), it does not crash the asterisk |
13:06.00 | _aggelos_ | if I run simple asterisk |
13:06.07 | _aggelos_ | then asterisk -vvvvr |
13:06.19 | _aggelos_ | when I get the error it crashes and exits asterisk alltogether. |
13:06.35 | JerJer[ghost] | app_system is kludgy |
13:06.43 | _aggelos_ | [daniel@blue sbin]$ ls -l callback.pl |
13:06.44 | _aggelos_ | -rwxr-xr-x 1 root root 2754 Jan 12 00:31 callback.pl |
13:06.53 | _aggelos_ | it used to work on an old version of asterisk |
13:07.04 | JerJer[ghost] | we haven't changed anything in app_system |
13:07.13 | _aggelos_ | since I updated cvs it gives me this ... |
13:07.38 | _aggelos_ | \"${CALLERID}\") <----- is there anything wrong in the notation in your oppinion ? |
13:08.13 | JerJer | why the slashes ? |
13:08.18 | *** join/#asterisk asterisk-bg (~asterisk-@62.73.103.10) |
13:09.33 | _aggelos_ | <PROTECTED> |
13:09.52 | JerJer | duno |
13:10.03 | JerJer | use AGI |
13:11.44 | _aggelos_ | thanks |
13:12.05 | *** join/#asterisk sobol_ (~druid@80.51.246.186) |
13:16.02 | *** join/#asterisk Powerkill (~powerkill@host.161.115.68.195.rev.coltfrance.com) |
13:16.44 | *** join/#asterisk f5-Scr^work (~f5-scr@port-212-202-229-162.reverse.qsc.de) |
13:17.12 | f5-Scr^work | hi got this failure when i try to use my isdn card WARNING[19473]: channel.c:1644 ast_request: No channel type registered for 'capiCD' |
13:17.17 | f5-Scr^work | any idea ? |
13:17.33 | JerJer | no such channel type registered for capiCD |
13:17.54 | f5-Scr^work | when i dial '0' then i want go to my local telefon provider |
13:17.58 | Powerkill | hi all |
13:18.00 | f5-Scr^work | sorry for my english :) |
13:18.10 | RoyK | f5-Scr^work: det er helt greit |
13:18.11 | RoyK | lol |
13:18.12 | JerJer | i've never heard of caipCD channel type |
13:18.23 | JerJer | capiCD |
13:18.37 | [Sim] | jerjer: it comes with chan_capi |
13:19.28 | f5-Scr^work | capi_request: didn't find capi device with outgoing msn = 1. you should check your config! |
13:19.41 | f5-Scr^work | when i try this with normal CAPI channel |
13:19.47 | *** part/#asterisk brown (~brown@sp1024.rbs-p01.ewol.com) |
13:19.48 | f5-Scr^work | this msg comes |
13:20.04 | JerJer | do what the capi god told you to do |
13:20.09 | czmok | hi |
13:20.11 | czmok | w00p |
13:20.15 | f5-Scr^work | in which config ? |
13:20.24 | czmok | i am about to have the 7920 working in asterisk :-) |
13:20.35 | czmok | i think i know where the error is |
13:20.41 | czmok | currently i am recoding some stuff :-) |
13:22.54 | mitya | i would like to compile the asterisk, with openh323-1.12.2 and pwlib-1.5.2 |
13:22.58 | mitya | is it possible??? |
13:23.03 | JerJer | yes |
13:23.06 | JerJer | read the README |
13:23.17 | [Sim] | if I want to have a voip-setup and a secretary that can see who is on the phone and who is not |
13:23.21 | [Sim] | what would I use ? |
13:23.34 | mitya | i have already read it |
13:23.46 | mitya | i says: cd /path/to/openh323 |
13:23.49 | mitya | make clean opt |
13:24.13 | mitya | result:make: *** No rule to make target `opt'. Stop. |
13:24.47 | mitya | what is wrong?? |
13:24.51 | mitya | any idea?? |
13:27.27 | geertn | I wanna make use of SIP url dialing, is this the right way to go?: exten => _9.,1,Dial(SIP/${EXTEN:1}@${SIPDOMAIN})? (it works but I'm not sure if there is a cleaner and better way to do this. |
13:36.51 | discordia | f5-Scr^work: in capi.conf you must add the outgoing MSNs |
13:37.07 | f5-Scr^work | yeah |
13:37.16 | f5-Scr^work | just found my stupid failure |
13:37.20 | f5-Scr^work | :) |
13:37.30 | discordia | hehe |
13:40.26 | *** join/#asterisk simprix (~simprix@24.247.181.124.gha.mi.chartermi.net) |
13:44.49 | *** join/#asterisk crontibs (Crontibs@ool-44c02950.dyn.optonline.net) |
13:45.00 | [Sim] | okay |
13:45.14 | [Sim] | has anyone gotten anywhere with sokol's astmgr ? |
13:47.22 | JerJer | not here... i even tried it on an XP laptop |
13:47.38 | [Sim] | hm |
13:47.53 | [Sim] | I got it running here, but I can't get the hang of how to do what |
13:48.06 | *** join/#asterisk Stuart (~Stuart@smtp.dstoys.com) |
13:49.07 | reseaux | im back... |
13:49.47 | f5-Scr^work | how i configure asterisk to answer a call from my localnetwork "openphone" ? |
13:50.17 | jimmyz | is there something wrong with parking? |
13:50.46 | jimmyz | when i call sip to another sip and try to park the call it doesn't park...it says there is no one parked on that # |
13:50.46 | JerJer | exten => 1234,1,Answer |
13:51.24 | jimmyz | it was work two days ago |
13:51.30 | reseaux | Someone Know if is possible to bring a number to call from signal(code preselection function port in DNID the caller in CLI the called nemuber) and made dial before answer a call and if is not busy join the channels?thz |
13:55.58 | *** join/#asterisk sniffe (~sniffe@ti211110a080-3555.bb.online.no) |
13:56.03 | f5-Scr^work | how i configure asterisk to answer a call from my localnetwork for example "openphone" ? |
13:57.14 | JerJer | exten => 1234,1,Answer |
13:57.19 | JerJer | RTFM |
13:57.31 | f5-Scr^work | cool down |
13:57.45 | f5-Scr^work | i thought this is a channel to help each other |
13:57.46 | JerJer | you have to help yourself before anyone else can help you |
13:58.46 | f5-Scr^work | dont answer my question and ignore me |
13:58.50 | f5-Scr^work | don't flame dude |
13:59.05 | JerJer | you haven't seen a flame yet |
13:59.17 | JerJer | read the fuckin docs <---- that's a flame |
13:59.41 | f5-Scr^work | manual :) |
13:59.43 | *** join/#asterisk jtodd (~jtodd@65.199.209.25) |
14:00.05 | f5-Scr^work | RTFM <-- not RTMC :P |
14:00.10 | f5-Scr^work | ehh D |
14:00.15 | f5-Scr^work | ok im quiete |
14:00.28 | malcolmd | g'mornin' # |
14:00.41 | malcolmd | have we ever hit 250 in this chan? |
14:00.44 | JerJer | howdie |
14:00.55 | malcolmd | darn thing just keeps on growing |
14:03.12 | *** join/#asterisk erik2 (~eanders@host-127-202-220-24.midco.net) |
14:03.23 | voidptr_ | omg |
14:06.00 | *** join/#asterisk coppice (~Steve@34.203.17.210.dyn.pacific.net.hk) |
14:06.36 | mitya | i have already compiled asterisk... OPENH323DIR was wrong... i'm a lamer.. |
14:07.44 | _aggelos_ | JerJer: do you know if anything special needs to be edited in Makefile for asterisk under REDHAT 9 ? tty settings or something like that ? |
14:07.57 | _aggelos_ | apparently the error I get is related to the terminal. |
14:08.10 | _aggelos_ | tty related stuff |
14:08.25 | *** join/#asterisk _discordia (disco@calculates.zero-points.for.zetagrid.de) |
14:08.42 | _aggelos_ | script does not exit with 0 and somehow it bothers asterisk. |
14:08.44 | *** join/#asterisk linuxa (~dave@AMarseille-103-1-3-115.w80-14.abo.wanadoo.fr) |
14:09.08 | *** join/#asterisk martijn (~punisher@213-136-25-66.adsl.bit.nl) |
14:09.11 | martijn | hola |
14:09.13 | mitya | but, now, when i would like to start it, it dies with the following error: libpq.so.3: cannot open shared object file: No such file or directory |
14:09.45 | martijn | is there any thought on when zaptel will be released for linux 2.6 kernels? |
14:10.28 | JerJer | mitya: plus you gota READ the README in asterisk/channels/h323 |
14:10.40 | h3x | i <3 Gentoo |
14:10.53 | malcolmd | h3x: yeah, gentoo's nice |
14:11.09 | martijn | noone has an idea? |
14:11.26 | vaewynAFK | gentoo is nice for single installs... when you gotta admin multiples it sux |
14:11.41 | h3x | no way |
14:11.52 | vaewyn | martijn: A couple people have alreay hacked it over to work... but no official releases yet |
14:12.08 | martijn | ah |
14:12.15 | martijn | are those patches for download somewhere? |
14:12.17 | h3x | a distro shouldnt ever have anything to do with administrating boatloads of machines |
14:12.32 | h3x | theres plenty of applications to aide with that |
14:12.47 | vaewyn | h3x: yeah... and the best ones arn't in gentoo |
14:12.52 | vaewyn | yet |
14:12.55 | vaewyn | :} |
14:12.58 | h3x | such as |
14:13.24 | martijn | vaewyn: are those patches for download somewhere? |
14:13.31 | heison | is iaxtel down? |
14:13.32 | vaewyn | such as the easiest way to clone production machines dpkg --get-selections | ssh remotemachine dpkg --set-selections :} |
14:13.50 | vaewyn | martijn: I am not sure myself... I have just heard mention... not sites sorry :{ |
14:13.54 | h3x | bahg |
14:14.08 | martijn | ok, then I'll have to do it myself also :) |
14:14.47 | *** join/#asterisk mbranca (~matteo@213.140.14.155) |
14:14.53 | h3x | easiest way to clone a machine is pop the new drive in another as a slave |
14:15.00 | h3x | and dump/restore all the shiznit |
14:15.24 | mbranca | hi people |
14:15.41 | h3x | or |
14:15.45 | h3x | use compactflash cards |
14:16.01 | vaewyn | martijn: bkw said to check the README.*26 in the zaptel stuff |
14:16.14 | h3x | if you can live with a small installation |
14:16.41 | martijn | thanks. :) |
14:17.04 | vaewyn | h3x: not when you have 200+ "production" machines... that can't go offline long :P |
14:17.51 | vaewyn | except for kernel upgrades these puppies never go down |
14:17.52 | h3x | netboot the mofos |
14:17.59 | *** join/#asterisk Asterisky (~IAX2_Lost@12-220-205-73.client.insightBB.com) |
14:18.07 | *** join/#asterisk ecd (~dslater@ool-44c17c85.dyn.optonline.net) |
14:18.13 | h3x | welcome to the age of gigE |
14:18.16 | vaewyn | These have to be full local machines... |
14:18.30 | h3x | depends on your application i guess :) |
14:18.42 | vaewyn | gigE hahaha... already saturating 2 channels of those as is :P |
14:19.07 | h3x | i was kind of thinking asterisk farms myself but... |
14:19.13 | voidptr_ | my god |
14:19.14 | vaewyn | academia is such a fun place to play in :} |
14:19.16 | voidptr_ | ripoffs! |
14:19.27 | voidptr_ | first they deliver the wrong materials |
14:19.33 | voidptr_ | i call them, so they send extra things |
14:19.39 | voidptr_ | then they charge me |
14:20.04 | vaewyn | voidptr_: easy to deal with... kill them |
14:20.05 | h3x | haha |
14:20.15 | martijn | yep |
14:20.21 | martijn | take a m4 |
14:20.34 | voidptr_ | i will call them in a bit |
14:22.13 | mbranca | http://slashdot.org/article.pl?sid=04/01/15/022220&mode=nested&tid=126&tid=172&tid=185 |
14:22.15 | vaewyn | pssh... and hear I thought I would get to see your picture on the news :} |
14:22.18 | mbranca | h323 flaws... |
14:22.25 | martijn | heh |
14:22.41 | martijn | some 17 year old turkish guy shot his teacher at school here the other day |
14:22.45 | martijn | whole country at hassle |
14:23.03 | martijn | but we all know his face now |
14:23.05 | martijn | ;) |
14:23.26 | fwd28326 | anyone heard anymore rumours on the tdm400p FXO modules, has anyone actually seen a prototype, or are they mythical :) |
14:24.17 | h3x | *you are getting sleeeeeppy*.... *you know you want a t1....* |
14:24.40 | voidptr_ | fwd28326 : i'm about to send a mail about that :P |
14:25.12 | fwd28326 | voidptr_ can we see it before everyone? |
14:25.15 | fwd28326 | please ! |
14:25.25 | fwd28326 | id love a T1 |
14:25.32 | voidptr_ | no no |
14:25.33 | h3x | i have a PRI at home ! |
14:25.42 | voidptr_ | i'm about to send a mail to digium about it |
14:25.44 | fwd28326 | or even a PRI |
14:25.48 | martijn | hehe.. depends on how good the "service" is :P |
14:26.05 | fwd28326 | but the costs here are off teh scale |
14:26.09 | *** join/#asterisk Cinzas (~cgiirc@p15097463.pureserver.info) |
14:26.13 | h3x | where are you ? |
14:26.17 | fwd28326 | ireland |
14:26.20 | mbranca | fwd28326, the card exists as a prototype, but not released yet due to fxo module problems |
14:26.22 | h3x | haha so E1 |
14:26.24 | Cinzas | ickserv identify icnopleu |
14:26.29 | fwd28326 | yes |
14:26.38 | reseaux | Hi Mbranca ...!!! Im back finaly.... :-) |
14:26.49 | h3x | whats your local carrier out there |
14:26.54 | fwd28326 | well actually partial PRI |
14:27.03 | fwd28326 | is all most people ever see |
14:27.07 | fwd28326 | eircom |
14:27.19 | fwd28326 | incumbant former stae owned |
14:27.26 | mbranca | ciao reseaux, dove cazzo eri finito! :) |
14:28.09 | RoyK | "18.2 The Journaling File System (JFS) code contributed to Linux by IBM was almost certainly copied and adapted for Linux from a version of AIX more recent than the one available for comparison" |
14:28.11 | reseaux | Ciaooo ... non avevo un cazzo e dovevo risolvere un po' di casini!! ex lavoro.. |
14:28.46 | mbranca | reseaux, :) ok. hope your company started well |
14:28.58 | martijn | de mazzel ;) |
14:29.11 | fwd28326 | 3200 for an isdn primary rate!! |
14:29.18 | reseaux | yeah... in this day i bring my two * asterisk server in CUSS in Milano |
14:29.36 | mbranca | CUSS? |
14:30.17 | reseaux | is a MIX where we can join Colt MCI and other telecom company... |
14:30.28 | mbranca | uh, ok |
14:31.15 | RoyK | fwd28326: 3200 what? |
14:31.35 | reseaux | now i have 2 server with 1 TE410P working on every one and work very well... |
14:31.40 | fwd28326 | euro |
14:31.43 | fwd28326 | aplogies |
14:31.59 | fwd28326 | get so used to (nearly) everyone being euro in europe now |
14:32.13 | fwd28326 | thats for installation |
14:32.38 | *** join/#asterisk wreckdiver (~goaway@adsl-068-209-107-007.sip.mia.bellsouth.net) |
14:33.14 | fwd28326 | you get channels in multiples of 16 @ €158 per 16 channels |
14:33.36 | *** join/#asterisk zoa (~john@kn-upc-1.kuleuven.net) |
14:33.42 | zoa | yellow |
14:34.03 | fwd28326 | which isnt to bad |
14:34.12 | fwd28326 | it's the 3200 up front!! |
14:34.39 | *** join/#asterisk dalabera (~Dalabera@206.137.96.110) |
14:35.38 | *** part/#asterisk Cinzas (~cgiirc@p15097463.pureserver.info) |
14:36.21 | heison | how come the version number never changes as I do cvs updates follow by make or even cvs update -d followed by make? |
14:36.24 | *** join/#asterisk Cinzas (~cgiirc@p15097463.pureserver.info) |
14:36.42 | Cinzas | HI |
14:37.18 | Cinzas | I cant make call from H323 to SIP, with asterisk0.7.1 |
14:37.34 | Cinzas | <PROTECTED> |
14:37.51 | *** join/#asterisk Seba2 (~se@OL236-72.fibertel.com.ar) |
14:37.58 | Seba2 | hello all |
14:38.03 | Cinzas | Any idea ? |
14:38.10 | Seba2 | I nees somebody who can help me |
14:38.19 | mbranca | Cinzas, does the user 911 exists in sip.conf AND is authenticated (sip show peers)? |
14:38.23 | Seba2 | I am having troubles using AGI |
14:38.28 | Seba2 | with function STREAM FILE |
14:38.37 | Cinzas | Name/username Host Mask Port Status 911 (Unspecified) 0.0.0.0 5060 Unmonitored910 (Unspecified) 0.0.0.0 5060 Unmonitored |
14:38.44 | Seba2 | When I use this function, nothing happens |
14:38.45 | Cinzas | I can make call from the sip to an h323 endpoint |
14:39.09 | Seba2 | but if I use playback inside extensions.conf all work OK |
14:39.11 | heison | kram: how come the version number never changes as I do cvs updates follow by make or even cvs update -d followed by make? |
14:39.25 | Seba2 | any idea? |
14:39.43 | mbranca | Cinzas, your sip endpoint must register to * to be able to be called |
14:39.58 | *** part/#asterisk LuchoAR (~ll@200.85.102.10) |
14:40.08 | *** join/#asterisk LuchoAR (~ll@200.85.102.10) |
14:40.56 | *** join/#asterisk rocketman (~rocketman@hoochie.digium.com) |
14:40.57 | Asterisky | I am trying to create an iax user while asterisk is running. asterisk -rx reload doesn't seem to do it. Any ideas? |
14:41.30 | rocketman | does anyone know when some of these drivers will be ported to the 2.6 kernel? |
14:41.31 | Cinzas | mbranca: The extensions appear in "sip show peers" |
14:41.41 | Cinzas | Aren't registered ? |
14:41.50 | kram | asterisky: reload should most assuredly do it |
14:41.51 | mbranca | seems not |
14:41.51 | rocketman | or is there a faq I can look at regarding that question? |
14:41.54 | zoa | its already ported to 2.6 |
14:41.56 | kram | what are you seeing when you reload? |
14:42.06 | kram | they are in principle ported to 2.6 already |
14:42.08 | kram | but untested |
14:42.12 | kram | and ztdummy is not yet ported |
14:42.22 | RoyK | kram: seems zaptel's broken |
14:42.30 | rocketman | I see .. |
14:42.33 | kram | royk: more details? |
14:42.38 | mbranca | Cinzas, see the voice (Unspecified)? there must be an ip addr when a sip endpoint registers with * |
14:42.51 | Asterisky | kram, it reloads, but doesn't let the new user registyer. |
14:42.55 | mbranca | kram, zaptel un 2.6.1 & wcusb crash my kernel :) |
14:42.57 | Seba2 | when I use STREAM FILE inside AGI nothing happens, but when I use Playback inside extensions.conf all work OK, any idea? |
14:43.01 | rocketman | thanks guys |
14:43.12 | zoa | if i change something in iax.conf i always reload asterisk completely |
14:43.21 | zoa | although i didnt test it for added users |
14:43.31 | Cinzas | mbranca: Ok. I'll see that |
14:43.33 | Nate187 | can asterisk answer faxes? |
14:43.33 | zoa | *restart asterisk completely |
14:43.38 | zoa | Nate187: yes |
14:43.40 | kram | mbranca: maybe i can get someone to set it up. are you running without preemptive kernel? |
14:43.46 | kram | and without smp? |
14:43.54 | mbranca | kram, yes both |
14:44.01 | kram | with both or without both? |
14:44.24 | mbranca | both without, sorry. I noticed that preempt caused problems |
14:44.26 | Asterisky | ZOP: you can't restart asterisk completely when other users are on calls. That's what reload is for |
14:44.33 | mbranca | but still, no luck |
14:44.41 | zoa | Asterisky: i know |
14:48.10 | *** part/#asterisk ecd (~dslater@ool-44c17c85.dyn.optonline.net) |
14:48.19 | Seba2 | when I use STREAM FILE inside AGI nothing happens, but when I use Playback inside extensions.conf all work OK, any idea? |
14:48.23 | *** join/#asterisk wcd (~dslater@ool-44c17c85.dyn.optonline.net) |
14:48.29 | Seba2 | please somebody can help me? |
14:48.36 | Seba2 | I don't understand why |
14:48.38 | RoyK | kram: seems i was missing some kernel headers |
14:49.02 | kram | royk: i see |
14:49.06 | dalabera | seba what's the command sentence you're using? |
14:49.17 | kram | royk: well, it's fixed now? |
14:49.19 | Seba2 | I will put the line here |
14:49.48 | Seba2 | print "ANSWER\n"; |
14:49.48 | Cinzas | mbranca, i defined the host=ip in sip conf, where ip is the ATA 186 ip ! |
14:49.54 | Cinzas | Now, he creates the channel |
14:49.56 | Seba2 | print STDERR "6a. Testing 'record' playback..."; |
14:49.56 | Seba2 | print "STREAM FILE vm-options \"\"\n"; |
14:50.02 | Seba2 | &checkresult($result); |
14:50.02 | Seba2 | print "HANGUP\n"; |
14:50.06 | Seba2 | that's all |
14:50.29 | Seba2 | is ok I think |
14:50.42 | Seba2 | but I can't hear anything |
14:51.42 | rob-- | kram: can I ask you something about app development? I'm not sure how to handle the timing when sending voice frames to a channel. |
14:51.52 | Seba2 | exten => s,6,BackGround(vm-options) |
14:51.55 | Seba2 | this work ok |
14:52.22 | Seba2 | any idea= |
14:52.23 | Seba2 | ? |
14:52.39 | kram | rob-- what do you mean |
14:53.34 | RoyK | kram: need to build 2.4.24 first |
14:53.41 | Seba2 | dalabera? |
14:53.46 | RoyK | kram: btw. is 2.6 support (zaptel) stable with e100p? |
14:53.47 | zoa | 2.4.24 sux :-p |
14:53.52 | kram | royk: did you read the README.Linux26 already? |
14:53.55 | kram | royk: it's not been tested |
14:53.56 | RoyK | yep |
14:53.59 | RoyK | ok |
14:53.59 | kram | royk: it just compiles |
14:54.05 | zoa | kram i hope to test it very soon |
14:55.16 | *** join/#asterisk Alric (~nbowyer@masq.hyperusa.com) |
14:56.25 | dalabera | zeba your not using the perl Asterisk module. You should look into that. That way it wont that easy... |
14:56.55 | LuchoAR | zoa why does it sux? |
14:57.21 | Seba2 | Lucho sos vos? |
14:57.27 | LuchoAR | si |
14:57.34 | Seba2 | oka |
14:57.38 | Seba2 | dalabera |
14:57.44 | *** join/#asterisk anthm (~anthmct@CPE-24-167-193-161.wi.rr.com) |
14:58.03 | Seba2 | I test Perl module agi-test but it did not work |
14:59.07 | ScaredyCat | Seba2: http://asterisk.gnuinter.net |
14:59.07 | reseaux | OPPORTUNITY: I need someone to develop a mysql frontend for * in php? a pay for it... :-) thanks |
14:59.35 | LuchoAR | reseaux: what do you need it to do? |
14:59.50 | dalabera | you need to intall it in order tu use and it should work |
15:00.18 | Seba2 | can you tell me how? |
15:00.28 | ScaredyCat | follow the instuciton |
15:00.29 | ScaredyCat | s |
15:00.34 | rob-- | kram: I'm trying to write an app for the Cepstral Theta TTS engine. I need to stream the audio which is generated to the channel, and listen for dtmf. When do I send voice frames, and how many samples should be in each frame? I looked at the code in |
15:00.46 | Seba2 | what instructions? |
15:00.55 | ScaredyCat | http://asterisk.gnuinter.net |
15:00.57 | ScaredyCat | http://asterisk.gnuinter.net |
15:01.06 | rob-- | file.c, but I can't get my head arround it. Is there a way of doing it without scheduling a callback? |
15:01.14 | RoyK | ScaredyCat ScaredyCat ScaredyCat ScaredyCat |
15:01.37 | Seba2 | I am reading that page |
15:01.48 | ScaredyCat | ues rob-- do what I go in my app_cepstral .. create the file and stream it back |
15:01.57 | ScaredyCat | ues? yes |
15:02.01 | Seba2 | I think if Asterisk comes wiht and agi-test it should work |
15:02.20 | ScaredyCat | it does work... we are just suggesting you use the perl module |
15:02.36 | ScaredyCat | it will make you life easier |
15:02.38 | ScaredyCat | your |
15:02.47 | Seba2 | yes |
15:02.49 | Seba2 | sure |
15:03.17 | jtodd | rob--: I look forward to the release of app_cepstral. :-) I think that would be pretty cool. |
15:03.19 | Seba2 | but first I want a simple agi script to work |
15:03.24 | jtodd | rob--: Sorry I have no hints for you, though. |
15:03.45 | rob-- | scaredycat: I could do, but it'd be more efficient to stream from memory. Theta can stream the audio as it is being synthesised. |
15:03.46 | Seba2 | I am downloading Perl AGI |
15:03.52 | jtodd | ScaredyCat: do you already have app_cepstral ? Is it released? |
15:04.15 | ScaredyCat | rob--: this is true.. but i wanted it quick |
15:04.16 | zoa | great, one of my servers is under attack |
15:04.29 | ScaredyCat | jtodd: not release but you can have a copy.. |
15:05.24 | *** join/#asterisk Shido6_ (~shido@d57-81-103.home.cgocable.net) |
15:07.25 | *** join/#asterisk Takapa (vegard@svanberg.no) |
15:07.33 | rob-- | It'd be good if the streaming code wasn't file-specific. If I could provide a read function like in file.c and it would just work. |
15:08.01 | rob-- | I suppose I could use a named pipe and use the file streaming code |
15:08.13 | rajo | anyone experience with gnophone and video transmission? |
15:08.18 | rob-- | But it's a bit of a hack |
15:09.15 | *** join/#asterisk montag (~montag@host166-150.pool80105.interbusiness.it) |
15:09.16 | *** join/#asterisk Tekati (~captain@cpe-66-75-211-60.bak.rr.com) |
15:09.18 | montag | <PROTECTED> |
15:09.18 | montag | Failed to load driver chan_modem_aopen.so |
15:09.22 | montag | what can be ??? |
15:09.22 | *** join/#asterisk asterisk-bg (~asterisk-@62.73.103.10) |
15:09.39 | *** join/#asterisk Junbug (~junya@adsl-34-194-140.bct.bellsouth.net) |
15:09.55 | montag | i use a voice modem connected to serial port |
15:09.55 | LuchoAR | is anyone developing a prepaid app with the asterisk API ?? we are about to begin a project of it |
15:10.31 | LuchoAR | and would like to know if anyone is doing something so we can join resources. |
15:10.53 | rob-- | http://www.voiptalk.org have a pre-pay system using asterisk, but I don't know what it is or if it is released. |
15:12.15 | LuchoAR | where?? it don't see it in the webpage. |
15:13.36 | *** join/#asterisk pattieja (~pattieja@63-252-5-1.ip.mcleodusa.net) |
15:14.01 | rob-- | they use it for there siptalk and iaxtalk service. I think it's probably a custom app |
15:14.23 | rob-- | but you could contact them at info@telappliant.com |
15:14.29 | *** join/#asterisk _asr (asr@pimpbox.latency.net) |
15:15.14 | LuchoAR | aa.. ok.. |
15:15.16 | LuchoAR | tnx! |
15:15.42 | crontibs | asr |
15:15.44 | crontibs | whats up |
15:17.20 | sobol_ | any cisco fans here willing to help with 7960 |
15:18.18 | *** join/#asterisk junya (~junya@adsl-81-1-250.mia.bellsouth.net) |
15:20.31 | zoa | rob-- its not a prepay system |
15:20.41 | zoa | not yet |
15:20.44 | zoa | its a rating engine |
15:21.09 | zoa | i'm testing it |
15:21.44 | Powerkill | zoa it's tholo one ? |
15:22.03 | zoa | yes |
15:22.25 | zoa | oh no |
15:22.28 | zoa | its not tholo's |
15:22.40 | zoa | its telappliants |
15:22.42 | Powerkill | it's better ? cause tholo one have lot of segfault |
15:22.44 | zoa | i don't think its opernsource |
15:22.55 | Powerkill | can i test it ? |
15:23.00 | zoa | i don't have it |
15:23.05 | zoa | don't think they will release it |
15:23.08 | zoa | i'll aks |
15:25.35 | Powerkill | ok thanks |
15:25.50 | zoa | powerkill: JerJer has a prepay engine that is supposed to be very good |
15:25.53 | zoa | and its for sale |
15:25.56 | Aviaa | halo again, anyone know how set into extension.conf if I want make this call: want first dial 123 pause 1sec and then 456 ? |
15:27.02 | ManxPower | Dial(Zap/1/123ww456) |
15:27.09 | ManxPower | Only works on analog ports |
15:27.20 | Aviaa | ManxPower i`ll ... |
15:27.34 | Aviaa | ManxPower let me try |
15:27.52 | ManxPower | w is .5 second pause I think, you might try three w's |
15:28.08 | *** join/#asterisk variable1 (~variable1@hoochie.digium.com) |
15:28.16 | *** join/#asterisk brent21 (bdf@paalto-apx-1-144-142.penn.com) |
15:29.58 | Aviaa | ManxPower not working ... |
15:30.06 | ManxPower | What is your dial line? |
15:30.08 | Aviaa | maybe "P" |
15:30.15 | ManxPower | No_Carrier_, "w" is a pause. |
15:30.19 | ManxPower | No "w" is a pause. |
15:30.54 | [Sim] | w00t |
15:31.13 | [Sim] | anyone with amsix lines: peer with us :-) |
15:31.34 | Aviaa | (SIP/123w456${EXTEN}@some.ip.host|60) |
15:31.39 | ManxPower | Aviaa, What Dial line are you using in extensions.conf? What displays on the Asterisk CLI when the Dial is executed? |
15:31.47 | ManxPower | Aviaa, SIP IS NOT ANALOG! |
15:32.02 | Aviaa | aj si .... .<ups> |
15:32.03 | ManxPower | You cannot use "w" with anything except for an analog channel/port. |
15:32.48 | ManxPower | I am not aware of any way to put a pause on a Dial using a digital channel/port |
15:32.48 | Aviaa | ManxPower than we can not set any pause in sip ... am i right |
15:33.10 | ManxPower | Aviaa, make it pause on the remote gateway. |
15:33.14 | *** join/#asterisk Stealth_Man (Stealth_Ma@hyp1-19.dialup.online.ge) |
15:33.25 | Aviaa | isee |
15:33.36 | bkw_ | Corydon you around? |
15:34.05 | zoa | bkw, can i help you ? |
15:34.06 | ManxPower | Aviaa, This is a limitation of VoIP protocols, not Asterisk |
15:34.16 | zoa | its prolly for some testing right ? :) |
15:35.24 | bkw_ | i'm trying to track down the malloc in chan_zap that fails to free when you load it when the signalling is incorrect |
15:35.30 | brent21 | Should I be worried about seeing these messages occasionally? (ast_readaudio_callback): Failed to write frame? |
15:35.36 | bkw_ | I think I see it but I don't know why a "return NULL" is there |
15:35.38 | brent21 | It usually happens when users are in the voicemail system |
15:35.38 | bkw_ | that doesn't seem right |
15:36.32 | zoa | ic, i cant help you then i'm affraid :) |
15:36.44 | bkw_ | line 5192 in mine |
15:36.52 | bkw_ | <PROTECTED> |
15:36.52 | bkw_ | <PROTECTED> |
15:36.52 | bkw_ | <PROTECTED> |
15:36.52 | bkw_ | <PROTECTED> |
15:36.52 | bkw_ | <PROTECTED> |
15:37.04 | bkw_ | i'm going to test something |
15:37.14 | sobol_ | is it possible to load logo to the Cisco7960 from tftp not ftom logo_url "http://"? |
15:37.24 | bkw_ | sobol_ yes |
15:37.38 | bkw_ | it has to be bmp or jpeg(only works in newest firmware) |
15:38.32 | sobol_ | bkw_: and how it looks on config logo_tftp ? |
15:38.44 | ScaredyCat | has anyone got the doxygen docs for * online - wassim gave me a link but i lost it |
15:40.22 | bkw_ | logo_url: http://alsdkfjlasdf.com/blah.bmp |
15:40.47 | doughecka | COOL WEBSITE! |
15:40.51 | doughecka | :P |
15:42.42 | ScaredyCat | /s/in// |
15:42.54 | sobol_ | bkw_: but how t dowload logo from tftp not http |
15:44.10 | bkw_ | you can't |
15:44.15 | scr | I came in this morning and asterisk had segfaulted on me. I'm looking at the backtrace, but I don't really know what I'm looking for |
15:44.21 | *** join/#asterisk huats (~chris@AToulouse-104-2-1-24.w217-128.abo.wanadoo.fr) |
15:44.22 | bkw_ | i'm not aware.. sobol_ why not read cisco's site on this? |
15:44.37 | bkw_ | scr what does the first three lines of the bt say? |
15:44.54 | scr | #0 0x40239f2c in mysql_fetch_row () from /usr/lib/libmysqlclient.so.12 |
15:44.54 | scr | #1 0x4022697e in do_directory (chan=0x82f5170, cfg=0x82eca78, context=0x0, digit=55 '7') at app_directory2.c:227 |
15:44.54 | scr | #2 0x4022669b in directory_exec (chan=0x402275f6, data=0xbd5ff274) at app_directory2.c:300 |
15:45.10 | bkw_ | 1 you are using app_directory |
15:45.12 | bkw_ | with mysql |
15:45.14 | bkw_ | NEXT!!! |
15:45.15 | martijn | ZT_SPANCONFIG failed on span 1: No such device or address (6) |
15:45.21 | martijn | does anyone have a clue? |
15:45.26 | martijn | ztcfg output |
15:45.29 | bkw_ | no such device or address |
15:45.31 | bkw_ | your configs are wrong |
15:45.35 | bkw_ | double check your configs |
15:45.37 | *** join/#asterisk simprix (~simprix@ip-64-32-242-201.dsl.iad.megapath.net) |
15:45.41 | Seba2 | hello |
15:45.42 | bkw_ | and make sure you kernel modlues are in |
15:45.46 | martijn | they are |
15:45.59 | scr | what's wrong with using directory and a db? |
15:45.59 | Seba2 | somebody knows how to use musiconhold inside AGI? |
15:46.01 | martijn | and i copies the configs from a running box |
15:46.13 | *** join/#asterisk nighty (~nighty@hoochie.digium.com) |
15:46.26 | martijn | s/copies/copied/ |
15:46.34 | bkw_ | Seba2 print "SET MUSIC ON default\n"; |
15:46.40 | Seba2 | tx |
15:46.43 | huats | does anybody know a way to connect to a call and just speak to one of the people who are in this call ? |
15:46.58 | MSpin | morning bkw |
15:47.17 | martijn | bkw_- the configs are from a stable and running box |
15:47.24 | martijn | and the mods are loaded |
15:49.03 | bkw_ | lsmod |
15:49.06 | bkw_ | ztcfg -vvvv |
15:49.12 | martijn | did that |
15:49.14 | bkw_ | rmmod's |
15:49.18 | bkw_ | reinstall the mods |
15:49.22 | martijn | did that |
15:49.22 | bkw_ | ztcfg -vvvvv again |
15:49.26 | martijn | did that |
15:49.31 | doughecka | shutdown -r now |
15:49.32 | bkw_ | cat /proc/interrutps |
15:49.39 | doughecka | init 0 |
15:49.42 | bkw_ | doughecka SHUT UP this isn't windows |
15:49.51 | bkw_ | martijn but have you rebooted the box |
15:49.52 | martijn | it says that all channels are configured |
15:49.58 | doughecka | shutdown -r now is the command for redhat |
15:50.02 | doughecka | :P |
15:50.13 | martijn | doughecka- that's not what bkw_ meant |
15:50.13 | bkw_ | doughecka grow up and be helpful please :P |
15:50.17 | martijn | ;) |
15:50.31 | doughecka | :P |
15:50.36 | bkw_ | martijn just for the sake or argument reboot the box |
15:50.41 | martijn | bah |
15:50.44 | doughecka | humbug |
15:51.00 | martijn | box has been rebooted once since install |
15:51.07 | martijn | so that should not be an issue either |
15:51.11 | martijn | mods are in |
15:51.17 | bkw_ | well I hear it has in the past been an issue |
15:51.21 | bkw_ | are you using latest cvs on zaptel? |
15:51.26 | martijn | ztcfg tells me all channels ar configuyred |
15:51.31 | martijn | yea |
15:51.32 | bkw_ | asterisk -vvvvgc |
15:51.34 | bkw_ | what does it say? |
15:51.36 | martijn | just got it out |
15:51.42 | martijn | Channel 123: Individual Clear channel (A-law) (Slaves: 123) |
15:51.42 | martijn | Channel 124: Individual Clear channel (A-law) (Slaves: 124) |
15:51.42 | martijn | 124 channels configured. |
15:51.43 | martijn | ZT_SPANCONFIG failed on span 1: No such device or address (6) |
15:51.57 | bkw_ | you sure your have it rigiht? |
15:51.58 | martijn | zaptel and tor2 are loaded |
15:52.06 | bkw_ | show me your spans in zaptel.conf |
15:52.06 | martijn | i copied the configs |
15:52.18 | martijn | span=1,0,0,ccs,hdb3,crc4 |
15:52.18 | martijn | span=2,0,0,ccs,hdb3,crc4 |
15:52.18 | martijn | span=3,0,0,ccs,hdb3,crc4 |
15:52.18 | martijn | span=4,0,0,ccs,hdb3,crc4 |
15:52.27 | bkw_ | you sure you need that crc4? |
15:52.29 | *** join/#asterisk cypromis (~michael@80.51.246.186) |
15:52.39 | cypromis | hmmm |
15:52.47 | [Sim] | hmm? |
15:52.50 | cypromis | anybody has callerid display with the latst firmware of the barbietones ? |
15:52.51 | bkw_ | hoo? |
15:52.56 | *** join/#asterisk TeleRidd (~TeleRiddl@hoochie.digium.com) |
15:52.56 | martijn | that's what my other config runs |
15:53.04 | bkw_ | martijn very strange |
15:53.08 | martijn | yea |
15:53.13 | bkw_ | reboot the box |
15:53.14 | martijn | i run linux 2.6 though |
15:53.20 | bkw_ | oh and the truth be told |
15:53.21 | martijn | 2.6.1 |
15:53.22 | *** join/#asterisk montag (~montag@host166-150.pool80105.interbusiness.it) |
15:53.22 | bkw_ | NEXT!! |
15:53.28 | martijn | ;) |
15:53.30 | TeleRidd | Is this support channel an "official" digium support forum or is it just users? |
15:53.31 | bkw_ | 2.6 isn't totally supported yet |
15:53.32 | martijn | should not be a problem |
15:53.34 | montag | any tips for a voice modem with asterisk ??? |
15:53.41 | bkw_ | TeleRidd its for the users |
15:53.56 | TeleRidd | Okay, thank you |
15:54.09 | TeleRidd | Here is my problem, as many might have heard from yesterday |
15:54.40 | TeleRidd | Our system was working fine with all configs until we had some power spikes as indicated by our Backup Power Supplies |
15:55.05 | bkw_ | have you treated yoru boxes like windows? |
15:55.07 | TeleRidd | We can receive incoming calls just fine but outgoing calls yield no audio |
15:55.17 | bkw_ | is this a T410P? |
15:55.17 | TeleRidd | Yes, rebooted |
15:55.44 | bkw_ | cards? |
15:55.50 | TeleRidd | When we listen in on the X100P card dialing out we can hear numbers being dialed |
15:56.05 | *** join/#asterisk Jon_UK (Jon_UK@host217-35-96-128.in-addr.btopenworld.com) |
15:56.16 | Jon_UK | Hi everyone! |
15:56.19 | bkw_ | wonder if it was fried |
15:56.22 | TeleRidd | then we receive a message from the phone company that the number could not be completed as dialed |
15:56.29 | Jon_UK | Is anyone here using a SNOM 200 phone with * |
15:56.48 | TeleRidd | But if it still receives incoming calls, how could it not dial out? |
15:56.49 | bkw_ | snom are fugly phones.. CISCO OR DEATH |
15:57.02 | bkw_ | TeleRidd have you modified digits.h in zaptel any? |
15:57.07 | zoa | snoms are not too ugly |
15:57.08 | TeleRidd | using Snom200 phones here. Mess to setup but getting better |
15:57.16 | *** join/#asterisk tim27 (tim27@229-29.dr.cgocable.ca) |
15:57.16 | Jon_UK | bkw_ easy to say if you've got the money for a cisco phone! |
15:57.31 | TeleRidd | No, phone.conf was missing when we rebooted but that is it |
15:57.32 | tim27 | Jon_UK i got the money for 2 7960 |
15:57.35 | tim27 | and 2 7905 |
15:57.38 | bkw_ | Jon_UK I don't have the money.. I just want reliable phones |
15:57.40 | TeleRidd | just regenerated from "make samples" |
15:57.57 | Mike | is queue a good idea insted of call wait? |
15:58.00 | bkw_ | TeleRidd make samples will remove all your stuff |
15:58.06 | bkw_ | and phone.conf is for a quicknet card |
15:58.23 | Jon_UK | I am trying to get my SNOM 200 to light up a different line button depending on which extension is receiving the call, without success! has anyone managed this? |
15:58.26 | tim27 | bkw_ : my 7960, are working now |
15:58.28 | tim27 | :)))) |
15:58.35 | TeleRidd | I know this, I backed up all my conf files before I did this and then moved them back into the directory with the new phone.conf |
15:58.41 | tim27 | register to proxy was set to off |
15:58.47 | RoyK | How many calls would you think I should be able to handle with a Duron 1400? |
15:58.54 | zoa | depends royk |
15:58.58 | zoa | what do you want to do ? |
15:59.01 | zoa | transcoding ? |
15:59.01 | bkw_ | phone.conf don't even need to be there |
15:59.04 | zoa | termination ? |
15:59.07 | zoa | what codecs ? |
15:59.14 | TeleRidd | okay, that is what I thought. |
16:00.13 | Jon_UK | TeleRidd, have you got the function buttons programmed on your SNOM200 ? |
16:00.14 | RoyK | I just setup a ultra-lowcost system with an MSI mb, a duron 1400, 256MB RAM and two ISDN cards for < ¤300 |
16:00.23 | tim27 | bkw_: cisco are reliable, but the LCD screen on the 7960 look cheap as the grandstream phone |
16:00.26 | TeleRidd | yeah I do |
16:00.34 | bkw_ | tim27 UM now |
16:00.36 | bkw_ | er no |
16:00.44 | zoa | royk: what isdn card ? |
16:00.48 | RoyK | so I want to produce an ugly-looking ultra-low-cost pbx |
16:00.50 | RoyK | hfc-pci |
16:00.51 | bkw_ | it may not be highres but damn it its nice |
16:00.53 | TeleRidd | I have just renamed phone.conf -> phone.conf.bak and then restarted Asterisk |
16:00.53 | zoa | thats like the price of 1 isdn card !!! |
16:01.01 | TeleRidd | it says it is unable to load now |
16:01.10 | RoyK | kapijod will come up with zaptel drivers for them soon |
16:01.13 | simprix | how much do channel banks run these day s |
16:01.16 | bkw_ | TeleRidd do you use chan_phone? |
16:01.17 | TeleRidd | so I must need phone.conf then? |
16:01.27 | bkw_ | you must not have an X100p then |
16:01.28 | RoyK | zoa: not el cheapo hfc pci |
16:01.40 | TeleRidd | I have 3 of them in the slots |
16:01.45 | Mike | anyone of you guys know of any cellphone that has wifi in it and that has a sip client that works with asteirsk? |
16:01.47 | TeleRidd | what do I use chan_phone for? |
16:03.25 | TeleRidd | What is phone.conf used for? I know it says linux telephony driver. I must need it if * won't start |
16:04.05 | bkw_ | no you don't need it unless you have quicknet phone cards |
16:04.11 | UnixDawg | MIKE yes the palm phone |
16:04.24 | Jon_UK | Mike X-Lite works well from a pocket PC |
16:04.24 | bkw_ | if you have quicknetphone cards you might as well jump off a off a bridge now |
16:04.26 | pros12 | mike: what phone is this? |
16:05.36 | MSpin | morning kram... |
16:06.19 | TeleRidd | Okay well here is what asterisk is telling me when I start. |
16:06.26 | RoyK | ~x100p |
16:06.35 | zoa | kram: are the E400p's still for sale ? |
16:06.50 | TeleRidd | ..Jan 15 09:59:43 ERROR[1074432736]: chan_iax.c:4826 set_config: Unable to load config iax1.conf |
16:07.01 | TeleRidd | .....Jan 15 09:59:43 WARNING[1074432736]: chan_iax2.c:5510 set_config: Ignoring port for now |
16:07.04 | zoa | thats normal |
16:07.07 | zoa | teleriid |
16:07.09 | bkw_ | TeleRidd that has nothing to do with it |
16:07.10 | TeleRidd | ...Jan 15 09:59:43 ERROR[1074432736]: chan_phone.c:1123 load_module: Unable to load config phone.conf |
16:07.18 | TeleRidd | Jan 15 09:59:43 WARNING[1074432736]: loader.c:312 ast_load_resource: chan_phone.so: load_module failed, returning -1 |
16:07.25 | TeleRidd | Jan 15 09:59:43 WARNING[1074432736]: loader.c:312 ast_load_resource: chan_phone.so: load_module failed, returning -1 |
16:07.26 | bkw_ | then noload => chan_phone.so in modules.conf if you don't need it |
16:07.27 | zoa | teleridd: unload chan_phone, |
16:07.32 | zoa | unload iax1.so |
16:07.41 | zoa | bkw, or is that still iax.so ? |
16:07.46 | bkw_ | noload => chan_iax.so |
16:07.52 | bkw_ | yes its still chan_iax.so |
16:07.53 | zoa | ah k |
16:07.56 | RoyK | When should one use IAX1 and when should I use IAX2? |
16:08.04 | simprix | with a sip client do i have to dial 9 to get a outside line for a pstn call |
16:08.07 | bkw_ | use IAX2 at all times |
16:08.24 | TeleRidd | How do I unload the .so modules |
16:08.32 | bkw_ | modules.conf |
16:08.38 | TeleRidd | sorry for the newbie question, never had to do that before |
16:08.39 | bkw_ | put noload => chan_phone.so |
16:08.49 | TeleRidd | in modules.conf right? |
16:08.54 | bkw_ | yes |
16:09.52 | Powerkill | what the cvs date to upgrade to if i want the 0.7.1 ? |
16:10.27 | bkw_ | you download the tarball from ftp.digium.com |
16:10.34 | tim27 | bkw_ : the half tones on the lcd look crapy... ( i mean the grey tab behin the text of the soft key) ... same on the 2 phone |
16:10.46 | TeleRidd | alright, Asterisk started |
16:11.08 | *** join/#asterisk extremis (extremis@121-17.waldenweb.com) |
16:11.10 | bkw_ | tim27 don't halftone anything then |
16:11.24 | tim27 | what you mean |
16:11.35 | tim27 | i cant no halftone them |
16:11.47 | extremis | so... to change the musiconhold context per extension that gets dialed, do you just add a priority to that extension before it gets dialed for MusicOnHold? |
16:11.59 | tim27 | when you press the softkey HOld, there is a grey tab behin the hold text |
16:12.14 | tim27 | this look very crapy heaven if i ajust contrast |
16:12.24 | bkw_ | oh thats not halftone.. adjust the contrast |
16:12.29 | bkw_ | mine looks perfect |
16:12.34 | bkw_ | maybe your being too picky |
16:12.40 | extremis | bkw_: I've seen it vary between 7960's |
16:12.46 | extremis | the lcd's on the 7960's suck |
16:12.49 | bkw_ | extremis really? |
16:12.52 | bkw_ | mine both look perfect |
16:12.53 | extremis | all mine have different splotches |
16:12.57 | extremis | I have 3 |
16:12.59 | extremis | they all vary |
16:13.00 | tim27 | mines 2 |
16:13.10 | MSpin | the ones we had looked great |
16:13.12 | simprix | with a sip client do i have to dial 9 to get a outside line for a pstn call |
16:13.12 | tim27 | the light grey synch |
16:13.16 | MSpin | (my old company) |
16:13.28 | extremis | tim27: just bitch to cisco and have em send you another one :) |
16:13.35 | extremis | so... about my musiconhold question? |
16:13.35 | *** join/#asterisk Inv_Arp (~junya@68.223.139.104) |
16:13.44 | tim27 | extremis: I didnt buy them new |
16:13.50 | zoa | ah voila |
16:13.56 | zoa | you bought them at ebay |
16:13.57 | zoa | for 100$ |
16:14.02 | zoa | and they ripped you off |
16:14.03 | tim27 | no for 350 $ CAN |
16:14.20 | tim27 | maybe thoses guy selling on ebay sell 2nd quality phone |
16:14.23 | tim27 | i suspect that |
16:14.23 | extremis | tim27: if you buy from ebay you should know your repair options |
16:15.05 | extremis | tim27: honestly I wouldn't worry about it... I have splotchy backgrounds on my 7960 too |
16:15.07 | tim27 | bkw_: buy some phone on ebay too... he like to bid there |
16:15.08 | extremis | the pure black is fine |
16:15.16 | extremis | its only some of the greys that look like ass |
16:15.17 | tim27 | pure black is fine |
16:15.24 | tim27 | greys look ugly |
16:15.27 | tim27 | that same here |
16:15.28 | tim27 | :) |
16:15.40 | tim27 | but phone speak well |
16:15.41 | tim27 | :) |
16:16.16 | extremis | can you add a mmusiconhold context to sip.conf? |
16:16.33 | alex-home | Why would you want to? |
16:16.38 | extremis | exten => 2600,1,MusicOnHold(extremis) |
16:16.38 | extremis | exten => 2600,2,Dial,SIP/2600|20 |
16:16.43 | extremis | that doesn't change the context |
16:16.50 | extremis | it still uses default (defined in zaptel.conf) |
16:17.28 | extremis | and yes, mpg123 is running for that context |
16:17.52 | Filace | could anyone here help me with configuring 2x BRI lines with asterisk.. i've got two of the lines working (except for the fact that BT have broken one of them) |
16:18.33 | *** join/#asterisk s3gal (~leon@cuscon4743.tstt.net.tt) |
16:19.35 | extremis | hrm, maybe I need SetMusicOnHold |
16:20.00 | *** join/#asterisk adke (~adke@hoochie.digium.com) |
16:21.02 | *** join/#asterisk jsmith (~jsmith@209.180.83.10) |
16:21.12 | zoa | wtf: |
16:21.15 | zoa | BANNED FILENAME ALERT |
16:21.15 | zoa | Our content checker found |
16:21.15 | zoa | <PROTECTED> |
16:21.15 | zoa | banned filename in an email to you from: |
16:21.15 | zoa | <PROTECTED> |
16:21.16 | zoa | Not quarantined. |
16:21.24 | zoa | thats not my mailserver doing that |
16:21.30 | zoa | so it must be theirs |
16:21.32 | alex-home | That could be your ISP? |
16:21.35 | zoa | nopez |
16:21.42 | zoa | it the mailserver of that other dude |
16:21.44 | alex-home | hmm, probably be theirs then yea. |
16:21.49 | alex-home | Get him to zip the file and send ;) |
16:22.10 | zoa | i don't even know that guy |
16:22.10 | *** join/#asterisk Sobek (~btatton@209.180.83.11) |
16:22.10 | extremis | nope, its not SetMusicOnHold |
16:22.10 | jsmith | !lart Sobek |
16:22.12 | jsmith | ~lart Sobek |
16:22.19 | ManxPower | That is the usual recommendation for sending EXE files via virus scanning mail servers |
16:22.20 | Sobek | ~whaleslap jsmith |
16:22.24 | | ACTION slaps jsmith upside and over the head with one freakishly huge killer whale named hugh |
16:22.25 | jsmith | ~meepgun Sobek |
16:22.29 | | ACTION shoots Sobek with a hyper-charged flux neutron gun |
16:22.29 | ManxPower | Or just rename the damn file. |
16:22.38 | Sobek | ~lart jsmith |
16:22.40 | zoa | ManxPower it should send the virus message to the sender |
16:22.47 | zoa | not to the recipient of the other server |
16:23.02 | zoa | i might as well get 1000 messages with virus warnings now :( |
16:23.05 | ManxPower | zoa, Our mail server used to do that. Since most virii these days fakes the sender address it's pretty pointless. |
16:23.15 | zoa | true |
16:23.17 | *** join/#asterisk PBXtech (~PBXtechco@67.107.241.3.ptr.us.xo.net) |
16:23.48 | zoa | ooowz |
16:24.39 | ScaredyCat | anyone know what pbx_substitue_variables_temp is? |
16:25.24 | *** join/#asterisk Aharonov (~mcr@desk.marajade.sandelman.ca) |
16:25.29 | extremis | ok, so I want the music on hold context to be changed for all outgoing calls originating from a specific SIP user... is this possible? |
16:25.36 | bkw_ | ScaredyCat oh their is a workaround for your UNIQUEID stuff you wanted |
16:25.50 | bkw_ | it was posted on the bug note before I closed it.. did you get that info? |
16:25.51 | ScaredyCat | oh kewl where? |
16:26.00 | ScaredyCat | no |
16:26.06 | ScaredyCat | again |
16:26.10 | bkw_ | citats I think came up with it |
16:26.21 | bkw_ | its a diffrent API callyou need to use for those |
16:26.35 | extremis | I know how to do it with overloading variables... but I want to do it in sip.conf |
16:26.37 | bkw_ | because they aren't what you would call "channel" variables |
16:26.45 | UnixDawg | cannot find -lpg |
16:26.52 | bkw_ | install pgsql |
16:26.54 | bkw_ | NEXT!! |
16:27.05 | UnixDawg | I have it installed |
16:27.28 | UnixDawg | and pgsql-devel |
16:27.35 | bkw_ | the linker can't find it apparently |
16:27.37 | ScaredyCat | bkw_ is you mean use pbx_substitue_variables* thenit's wrong... or at least the example code is very wrong |
16:27.53 | *** part/#asterisk Aharonov (~mcr@desk.marajade.sandelman.ca) |
16:28.01 | bkw_ | was it james that came up with that? |
16:28.03 | UnixDawg | grr this means we have to fix the paths |
16:28.05 | UnixDawg | grrr |
16:28.12 | UnixDawg | ti fbsd |
16:28.27 | ScaredyCat | bkw918 01-11-04 02:56 use pbx_substitue_variables_* |
16:28.45 | bkw_ | yes I was told that would work |
16:28.53 | bkw_ | check the api specs |
16:28.53 | ScaredyCat | no... |
16:28.56 | bkw_ | hrm |
16:29.02 | bkw_ | then we need to expose those vars also |
16:29.10 | bkw_ | what was that bug number? |
16:29.13 | UnixDawg | frik |
16:29.15 | ScaredyCat | the example uses pbx_substitue_variables_temp - which doesn't exist |
16:29.23 | ScaredyCat | http://bugs.digium.com/bug_view_page.php?bug_id=0000741 |
16:29.49 | drgalaxy | my skinny phone works when I register it with our main * box, but when I connect it to my * box and switch its context to the remote one, the sound from me talking into my phone is not transmitted. any ideas? |
16:30.06 | bkw_ | ScaredyCat ok its reopened |
16:30.12 | *** join/#asterisk VoiceLynx (VoiceLynx@rrcs-central-24-106-64-175.biz.rr.com) |
16:30.15 | ScaredyCat | substitute |
16:30.43 | citats | if you look at app_cut you can get an idea of how to get at those special variables.... ideally pbx_builtin_getvar_helper would call pbx_substitute_variables_temp within it to subst those vars |
16:31.27 | jimmyz | when you do a sip show channels does it tell you what zap channel the phone is using |
16:31.54 | jimmyz | i don't have a zap device yet to check this |
16:32.08 | bkw_ | I need someone with a bit more C clue than myself |
16:32.24 | citats | app_cut doesn't use variables_temp iirc, it uses variables_helper but it addes ${ } around the var so it doesn't need to do some trickery |
16:32.44 | citats | when i get a chance i'll make it work right, but i dont know when I'll have a chance |
16:33.21 | f5-Scr^work | when you dial an empty number, and have early B3 enabled, with: |
16:33.21 | f5-Scr^work | Dial,CAPI/12345678:b |
16:33.21 | f5-Scr^work | the channel will come up at once and give you the dialtone it gets from the local exchange. |
16:33.21 | f5-Scr^work | at this point the channel is like a legacy phone, now you can send dtmf digits to dial. |
16:33.26 | f5-Scr^work | i dont understand this |
16:33.41 | f5-Scr^work | i want to get a dialtone when i dial the number 0 |
16:34.08 | bkw_ | does it? |
16:34.10 | f5-Scr^work | i mean the dialtone of my local provider |
16:34.15 | bkw_ | your TA should do that |
16:34.25 | bkw_ | in this case Asterisk right? |
16:34.41 | *** join/#asterisk coolhp (~Dude@unknown-141-roc.globalcrossing.com) |
16:34.42 | ScaredyCat | citats: thanks for the pointer, I'll take a look |
16:34.46 | f5-Scr^work | hmm dont know |
16:34.55 | f5-Scr^work | TA ? |
16:35.02 | f5-Scr^work | sorry im a german |
16:35.19 | citats | ScaredyCat: hopefully i'll have a chance in the next few days to come up with a patch for that |
16:35.29 | *** join/#asterisk Shido6_ (~shido@d57-81-103.home.cgocable.net) |
16:35.47 | f5-Scr^work | need a dial rule where i only get to my provider and kann normal dial ... |
16:35.54 | f5-Scr^work | has any one a idea ? |
16:36.18 | Filace | if you dial the B channel on a BT provided ISDN2e circuit, you get a regular dialtone |
16:36.33 | bkw_ | HOW? |
16:36.45 | bkw_ | its digital your terminal adapter should provide dialtone |
16:36.52 | Filace | ok, it could be the TA |
16:36.56 | f5-Scr^work | ;exten => _0.,1,StripMSD,1 |
16:36.56 | f5-Scr^work | ;exten => _XX.,2,Dial,capi/${CALLERIDNUM}:bBYEXTENSION |
16:36.56 | bkw_ | or thats how it is in the US |
16:36.58 | Filace | I said you get a dialtone, not where it came from :) |
16:37.00 | f5-Scr^work | when i do this |
16:37.04 | f5-Scr^work | i can phone out |
16:37.08 | bkw_ | dont' use BYEXTENSION |
16:37.15 | f5-Scr^work | but i want to dial 0 and get the dialtone |
16:37.19 | bkw_ | its gonig BYE BYE |
16:37.38 | Filace | bkw_: you know anything about connecting more than one BRI line to a * box? |
16:37.48 | f5-Scr^work | hmm |
16:37.54 | f5-Scr^work | so no one have a idea ? |
16:38.05 | f5-Scr^work | or can help me ? |
16:38.06 | f5-Scr^work | :) |
16:38.15 | puzzled | bkw: chan_capi sports "real" dialtone with e.g. Dial,CAPI/12345678:b |
16:38.29 | Filace | f5-Scr^work: why not just setup sommat like exten => _9.,1,Dial,CAPI/MSN:${EXTEN:1} |
16:38.37 | bug847 | bkw's a nub... don't listen to him |
16:38.51 | Filace | so you can dial 9{number} to call anything external |
16:39.02 | *** join/#asterisk heison (~heison@dyn-65-97.ham.dial.tht.net) |
16:39.25 | f5-Scr^work | hmm dont works |
16:39.30 | *** join/#asterisk Jacky (~Jacky@hoochie.digium.com) |
16:39.39 | puzzled | f5-Scr^work: look in the chan_capi README under "Overlap sending" |
16:39.59 | f5-Scr^work | Overlap sending (a.k.a. real dialtone) |
16:40.00 | f5-Scr^work | when you dial an empty number, and have early B3 enabled, with: |
16:40.01 | f5-Scr^work | Dial,CAPI/12345678:b |
16:40.03 | f5-Scr^work | the channel will come up at once and give you the dialtone it gets from the local exchange. |
16:40.04 | f5-Scr^work | at this point the channel is like a legacy phone, now you can send dtmf digits to dial. |
16:40.14 | f5-Scr^work | but i dont understand this |
16:40.29 | ScaredyCat | bkw_: you can close it again |
16:41.00 | bkw_ | ScaredyCat ok |
16:41.18 | scr | bkw: you objected earlier to my using mysql and directory... was that opposition to mysql (over odbc) or should I not be using the directory w/ a db at all? |
16:41.18 | UnixDawg | frick |
16:41.32 | bkw_ | scr no |
16:41.40 | bkw_ | but you are using something thats not in CVS |
16:41.45 | bkw_ | or is mysql directory in CVS now? |
16:41.58 | bkw_ | nope doesn't appear to be |
16:42.01 | scr | nope, it's my code, I know it's my bug... |
16:42.18 | bkw_ | scr I have an app_directory thats odbc aware but its not pretty |
16:42.32 | bkw_ | I also have voicemail routines that are odbc aware |
16:42.43 | bkw_ | just havne't had time to tweak with it |
16:43.12 | bkw_ | wooo deadlock |
16:44.13 | jsmith | bkw_: Wow... more dreadlocks? I always wanted dreads... |
16:44.19 | Filace | heh |
16:44.23 | UnixDawg | should pq not be pg |
16:44.51 | bkw_ | jsmith No I created this on purpose and it worked |
16:44.51 | bkw_ | haha |
16:44.55 | bkw_ | i'm trying to trap a bug |
16:45.23 | jsmith | bkw_: Oh... |
16:46.01 | Jacky | Does any one use VoiceTronix Openswitch 12 for FXO/FXS card? |
16:46.02 | bkw_ | this is a very strange one |
16:46.58 | vaewyn | Hmm... ok.. so now I need to figure out why my Zap FXS card spent 8 hours yesterday ringing the line after I had hung up |
16:47.23 | vaewyn | wondered why the cordlesses batteries were dead when I had just charged it :P |
16:48.51 | zoa | bkw, how can i retrieve the accountcode from within an AGI ? |
16:49.14 | zoa | or from in a dialplan ? |
16:50.30 | UnixDawg | wich lib pg++ or XX |
16:50.30 | UnixDawg | is needed |
16:50.30 | UnixDawg | i need pgsql suport |
16:50.31 | citats | zoa: accountcode is passed to AGIs as the * vars when the AGI starts |
16:50.31 | zoa | and how can i retrieve it ? |
16:50.31 | zoa | in what variable can i find it ? |
16:50.31 | citats | and from the dialplan as ${ACCOUNTCODE} |
16:50.31 | citats | zoa: agi_accountcode |
16:50.31 | zoa | thnx a lot |
16:50.35 | *** join/#asterisk Havik (~Havik@hoochie.digium.com) |
16:50.35 | Havik | Hey guys, just getting ready to start on my first asterisk system.... was wondering if anyone had any recommendations for phones? |
16:50.49 | crontibs | 7960's |
16:50.54 | crontibs | havik |
16:51.00 | zoa | 7960's or snoms |
16:51.05 | bug847 | Panasonic 1 line corded.. $19.95 @ Office Max |
16:51.11 | bug847 | works wonderful |
16:51.22 | bug847 | we also use a mix of Uniden and Panasonic cordless phones |
16:51.39 | Havik | sorry, good VoIP phones is what i meant :-) |
16:51.42 | UnixDawg | bkw |
16:51.49 | bug847 | crisco or snom |
16:51.50 | Havik | for remote extensions |
16:51.51 | Alric | 7960! |
16:52.04 | bug847 | if you're experimenting, just get a couple grandstreams |
16:52.11 | drgalaxy | boo snom.com resized my browswer window |
16:52.14 | drgalaxy | err snom.de |
16:52.15 | bug847 | but most people will tell you that GS phones suck for production use |
16:52.36 | bug847 | 7940 has 2 line appearance? |
16:52.42 | drgalaxy | are the grandstream ATAs worth anything? |
16:52.53 | bug847 | drgalaxy: dunno... wait for the Eaxy though |
16:53.02 | bug847 | (if you trust Digium hardware that is) |
16:53.03 | drgalaxy | what is Eaxy? |
16:53.03 | bug847 | =D |
16:53.09 | bug847 | iaxy... sorry |
16:53.09 | zoa | IAXy |
16:53.14 | rob-- | drgalaxy I'm using one and it works fine. |
16:53.15 | zoa | digium's ata |
16:53.21 | drgalaxy | ahh excellent |
16:53.46 | drgalaxy | how long until we can get them? |
16:53.48 | rob-- | drgalaxy: If you want to phone me to here the quality my fwd is 212811 and my iaxtel is 17006602034 |
16:54.04 | UnixDawg | grr this pisses me off |
16:54.21 | drgalaxy | rob--: thanks, I trust it will be a good unit. (also, don't have iaxtel setup) |
16:54.34 | *** join/#asterisk jorgeraidel (~jorgeraid@ip-69-33-73-90.mia.megapath.net) |
16:54.37 | drgalaxy | for some reason my skinny phone doesn't transmit the audio I say into it to my office's * box |
16:54.39 | jorgeraidel | hello |
16:55.02 | drgalaxy | Havik: a good place to look for info about phones and such is www.voip-info.org |
16:55.05 | jorgeraidel | hey my asterisk it have one error when I make ectensions reload |
16:55.16 | jorgeraidel | Jan 16 00:54:22 WARNING[1226400960]: pbx.c:4734 ast_context_verify_includes: Context 'asterisk' tries includes non-existant context 'iaxtel' |
16:55.16 | jorgeraidel | voiceip*CLI> |
16:55.39 | zoa | [iaxtel] doesnt exist in extensions.conf |
16:56.36 | *** join/#asterisk Lemmiwinks (~sean@adsl-065-083-169-002.sip.mco.bellsouth.net) |
16:56.38 | rob-- | drgalaxy I was talking about the gs ata, not the iaxy |
16:57.03 | drgalaxy | rob--: ooooh |
16:57.20 | crontibs | yoyo |
16:57.23 | drgalaxy | rob--: either way, I can't speak and be heard currently.. going to work on it later |
16:57.24 | crontibs | whats going with freebsd |
16:57.27 | crontibs | 847 thing |
16:57.57 | jorgeraidel | somebody can helpme |
16:58.09 | *** join/#asterisk seemore (~craig@d205-206-9-104.abhsia.telus.net) |
16:58.20 | *** join/#asterisk Starbuck (~Starbuck@hoochie.digium.com) |
16:58.33 | rob-- | drgalaxy: It's not too bad, although it hasn't got many features. |
16:58.55 | Starbuck | Does anyone know an easy way to make GSM audio files that I can use for the phone menu system? |
16:59.05 | UnixDawg | man this is a pain |
16:59.24 | rob-- | drgalaxy: sometimes when I answer a call I get a loud beep, and have to hangup the phone and pick it up again to answer the call. |
16:59.34 | *** join/#asterisk Stuart (~Stuart@smtp.dstoys.com) |
16:59.59 | swirlnets | does anyone know of a cheap headset for a cisco 7960? |
17:00.13 | jimmyz | we get ours for $50 |
17:00.19 | Starbuck | My company is selling headsets for the 7960. |
17:00.26 | swirlnets | starbuck: URL? |
17:00.32 | Starbuck | About 30 used. |
17:00.42 | swirlnets | starbuck: I don't want used |
17:00.52 | swirlnets | i can get used for 20 bucks |
17:01.08 | rob-- | Kram: what is the best way to stream audio from memory in an app? |
17:01.24 | jimmyz | swirlnets: http://www.acousticalinnovations.com $50 new |
17:01.37 | jimmyz | part number AT-904-6400 |
17:02.20 | swirlnets | jimmyz: do you have a direct link? |
17:02.41 | jimmyz | no i e-mail our order in |
17:03.17 | jimmyz | looks just like this one http://www.acousticalinnovations.com/LX-900.htm |
17:03.33 | *** join/#asterisk Sobek (~btatton@209.180.83.6) |
17:03.36 | swirlnets | jimmyz: thanks |
17:03.39 | jimmyz | np |
17:04.01 | jorgeraidel | alguin me puede ayudar? |
17:04.43 | zoa | what is the price of an LX-900 ? |
17:04.52 | zoa | and how easily can you break them ? |
17:05.01 | zoa | are they the same quality as the GN-Netcoms ? |
17:05.19 | jorgeraidel | but this lx-900 with what kind of voice works? |
17:05.31 | *** join/#asterisk jsmith (~jsmith@209.180.83.18) |
17:05.33 | jimmyz | voice works? it's analog |
17:05.46 | *** join/#asterisk rumba (rumba@cpe-24-160-4-31.sw.rr.com) |
17:05.51 | jorgeraidel | but how I conected? |
17:06.02 | jimmyz | the headset jack on the 7960 |
17:06.10 | jorgeraidel | oh ok |
17:06.16 | *** join/#asterisk benngard (~mabe@81.26.235.3) |
17:06.19 | jorgeraidel | I don't have the 7960 |
17:06.30 | jorgeraidel | only works in this voip? |
17:06.44 | Powerkill | I receice this NOTICE in asterisk |
17:06.44 | Powerkill | Jan 15 18:05:06 NOTICE[1217755952]: rtp.c:418 ast_rtp_read: Unknown RTP codec 19 received |
17:06.58 | Powerkill | kram can you add it i think it's g726 |
17:07.15 | Powerkill | it's from grandstream phone that have it :) |
17:08.05 | jimmyz | Jorgeraidel: don't know...it's what i use on our 7940 and 7960 |
17:12.02 | Lemmiwinks | Hey guys, anyone have any good recommendations for VoIP phones (remote extensions) for the asterisk system? |
17:13.24 | RoyK | Lemmiwinks: snom is nice, but quite expensive |
17:17.58 | swirlnets | lemmiwinks: the best options are Cisco 7960 |
17:18.15 | *** join/#asterisk cypromis (~michael@80.51.246.186) |
17:22.58 | ManxPower | Powerkill, check the archives for the reason and the fix |
17:24.10 | *** join/#asterisk rajo_home (~rainer@p508AE16C.dip.t-dialin.net) |
17:24.13 | *** join/#asterisk clh (~me@216.253.86.210) |
17:24.40 | clh | is the Cisco ATA configurable via browser? |
17:24.45 | clh | anyone.. |
17:24.55 | ManxPower | clh, Yes, see my sample config files. |
17:25.17 | clh | they reachable through links page on asterisk.org? |
17:25.24 | ManxPower | Go to http://www.digium.com/index.php?menu=documentation and look at the "Unofficial Links" section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the "Asterisk Resource Pages". |
17:25.42 | *** join/#asterisk dnc (~duncan@213.244.224.118) |
17:26.11 | clh | k, thanks! I assume you can just plug an el cheapo 900 MHz phone to it and have a "wireless VoIP phone" within the range of the phone base station....is this true? |
17:26.46 | ScaredyCat | bkw_: |
17:27.31 | jorgeraidel | text 'asterisk' tries includes non-existant context 'shortcut' |
17:27.32 | jorgeraidel | Jan 16 01:26:34 WARNING[1226400960]: pbx.c:4734 ast_context_verify_includes: Context 'asterisk' tries includes non-existant context 'iaxtel' |
17:27.45 | jorgeraidel | I have this eror when I make extensions reload |
17:28.06 | *** join/#asterisk outtolunc (~me@h-66-166-19-148.SNVACAID.covad.net) |
17:28.51 | ScaredyCat | nm |
17:29.06 | clh | anyone used cordless phone with Cisco ATA? |
17:29.06 | bkw_ | zoa told you once |
17:29.10 | outtolunc | morn'n |
17:29.13 | bkw_ | iaxtel doesn't exist |
17:29.16 | bkw_ | the [iaxtel] context that is |
17:29.54 | zoa | bkw, he's not listening i think :) |
17:29.58 | jorgeraidel | but is my asterisk? |
17:30.01 | zoa | yes |
17:30.10 | jorgeraidel | I installed the 0.7.0 yesterday |
17:30.13 | zoa | check extensions.conf |
17:30.15 | jorgeraidel | oh |
17:30.17 | zoa | look for include |
17:30.18 | jorgeraidel | ahd them |
17:30.26 | zoa | check if there is an include => iaxtel |
17:30.29 | zoa | if there is, |
17:30.33 | zoa | check if you also have an |
17:30.36 | zoa | [iaxtel] |
17:30.41 | zoa | if you don't have an [iaxtel] |
17:30.41 | jorgeraidel | ok |
17:30.46 | zoa | comment the include line |
17:30.51 | jorgeraidel | ok |
17:30.58 | tclark | clh: sure no problems |
17:31.01 | zoa | NEXT! |
17:31.38 | zoa | kram: that TE410p did work after all |
17:31.47 | UnixDawg | grrr |
17:31.48 | zoa | dunno why it didnt work at first |
17:31.49 | clh | tclark: tx |
17:31.55 | UnixDawg | this pisses me off |
17:32.05 | zoa | kram: i noticed in the changelog that there are now releasecodes for the pri's |
17:32.12 | zoa | where are those being logged ? |
17:32.17 | zoa | CDR ? |
17:32.34 | kram | i don't rembmer |
17:32.45 | zoa | :) |
17:33.21 | kram | hey guess what! |
17:33.59 | zoa | MOOSE PENIS ? |
17:33.59 | kram | iconnecthere wrote me |
17:34.10 | kram | they want someone to help come up with a guide for them for how people can connect with asterisk |
17:34.10 | malcolmd | kram: and? |
17:34.13 | JerJer | ahh sometimsconnecthere |
17:34.13 | bkw_ | kram what did icantconnecthear say? |
17:34.20 | malcolmd | heh |
17:34.24 | zoa | lol |
17:34.40 | kram | also i'm looking for the origination of the term "barbietone" |
17:34.51 | zoa | thats jerjer or bkw |
17:34.51 | zoa | i'm sure |
17:34.56 | zoa | i think i have it logged |
17:34.57 | zoa | :) |
17:36.11 | *** join/#asterisk heison (~heison@dyn-65-97.ham.dial.tht.net) |
17:37.31 | zoa | <Blaisen1> but when i call extension to extension on my grandstream barbietones i get no compatible cdoecs |
17:37.31 | zoa | <Blaisen1> i don't have anything disallowed in sip.conf so i don't understand |
17:37.31 | zoa | <atacomm> HoopyCat: not at all huh? driver issues getting to the Zaptel cards in a virtual machine? |
17:37.31 | zoa | <benjk> Blaisen: make sure that you specify a common codec in your sip.conf entries for those phones and make sure they are codecs that the GS supports |
17:37.31 | zoa | <benjk> for example: |
17:37.32 | zoa | <blitzrage> hmmm.. for some reason I can't seem to get my tdm400p to work in this Celeron 633. I know i |
17:37.42 | zoa | is the first thing i can find for barbietone in my logs |
17:37.51 | JerJer | what's the date of that ? |
17:38.02 | zoa | didnt have timestamps :( |
17:38.10 | zoa | but someone else that is logging might have it |
17:38.13 | *** join/#asterisk heison (~heison@dyn-65-97.ham.dial.tht.net) |
17:38.25 | JerJer | i remember going to shido's place and seeing one for the first time and sayng, "that looks like a toy, man" |
17:38.33 | zoa | :) |
17:38.57 | zoa | i have at least one month worth of logs before that statement |
17:38.57 | UnixDawg | ok this is a pain I need more fbsd people to work on this |
17:39.06 | zoa | but i'm not in the chan 24/24 7/7 |
17:39.10 | UnixDawg | to get postgres usable |
17:39.14 | zoa | but i think that was the first one |
17:39.30 | *** join/#asterisk heison (~heison@dyn-65-97.ham.dial.tht.net) |
17:40.01 | Shido6_ | i think jj started the barbietone name |
17:40.11 | *** join/#asterisk russT (~rusty@65-101-255-24.dnvr.qwest.net) |
17:41.05 | ManxPower | Blaisen's problem is that he does not do a disallow=all and an allow=compatablecodec in his sip.conf |
17:41.23 | ManxPower | JerJer, The skirt does not look good on you. |
17:41.37 | JerJer | damnit that's a kilt |
17:41.42 | JerJer | heheh |
17:41.44 | citats | zoa: i have the same logs in mine and it looks like that Blaisen1 message was about oct 11th or so |
17:42.19 | ManxPower | JerJer, not if you are giggleing like a school girl, it isn't. 8-) |
17:42.30 | JerJer | lol |
17:43.21 | bkw_ | damn it to hell chan_zap will be my bitch today |
17:44.08 | jets | haha |
17:45.58 | LuchoAR | I have installed an E100P (Euro ISDN PRI), which modules should I load before staring asterisk |
17:47.38 | citats | LuchoAR: wct1xxp |
17:48.18 | reseaux | hi luchoAR!! |
17:48.20 | LuchoAR | hi! |
17:48.26 | LuchoAR | Jan 15 14:41:48 WARNING[1074404064]: chan_zap.c:659 zt_open: Unable to specify channel 1: No such device or address |
17:48.34 | LuchoAR | Jan 15 14:41:48 ERROR[1074404064]: chan_zap.c:5189 mkintf: Unable to open channel 1: No such device or address |
17:48.39 | *** join/#asterisk hclai (~hclai@hoochie.digium.com) |
17:48.39 | LuchoAR | here = 0, tmp->channel = 1, channel = 1 |
17:48.44 | LuchoAR | Jan 15 14:41:48 ERROR[1074404064]: chan_zap.c:7162 setup_zap: Unable to register channel '1-3' |
17:48.51 | LuchoAR | that's what I am getting |
17:49.15 | *** join/#asterisk camoa (~camoa@200.71.58.37) |
17:49.16 | reseaux | LuchoAR: have you configure the /etc/zaptel.cfg |
17:49.28 | LuchoAR | channels] |
17:49.28 | outtolunc | .conf |
17:49.28 | LuchoAR | context=default |
17:49.28 | LuchoAR | switchtype=euroisdn |
17:49.28 | LuchoAR | pridialplan=national |
17:49.28 | LuchoAR | signalling=pri_cpe |
17:49.29 | LuchoAR | usecallerid=yes |
17:49.31 | LuchoAR | hidecallerid=no |
17:49.33 | LuchoAR | callwaiting=yes |
17:49.35 | LuchoAR | usecallingpres=yes |
17:49.35 | camoa | hello |
17:49.37 | LuchoAR | callwaitingcallerid=yes |
17:49.39 | LuchoAR | threewaycalling=yes |
17:49.41 | LuchoAR | transfer=yes |
17:49.43 | LuchoAR | cancallforward=yes |
17:49.43 | camoa | one question is the last CVS stable? |
17:49.45 | LuchoAR | callreturn=yes |
17:49.47 | LuchoAR | echocancel=yes |
17:49.49 | LuchoAR | echocancelwhenbridged=yes |
17:49.51 | LuchoAR | rxgain=0.0 |
17:49.53 | LuchoAR | txgain=0.0 |
17:49.55 | LuchoAR | callgroup=1 |
17:49.57 | LuchoAR | pickupgroup=1 |
17:49.59 | LuchoAR | immediate=no |
17:50.01 | LuchoAR | context = default |
17:50.03 | LuchoAR | group = 1 |
17:50.05 | LuchoAR | channel => 1-3 |
17:50.07 | LuchoAR | that's my zapata.conf |
17:50.12 | camoa | kram |
17:50.22 | hclai | hi, LuchoAR |
17:50.26 | UnixDawg | cvs is never stable |
17:50.35 | LuchoAR | hi! |
17:50.38 | hclai | do u have a asterisk with u now? |
17:50.40 | camoa | so, where can i get the last stable version? |
17:50.44 | LuchoAR | yes |
17:50.46 | hclai | can u try to call my IAX? |
17:50.52 | camoa | so, where can i get the last stable version? please |
17:50.54 | UnixDawg | get 7.1 tarball |
17:50.56 | LuchoAR | but I don't have it up |
17:50.59 | hclai | my IP =220.255.46.213 |
17:51.04 | reseaux | have you define D channel |
17:51.05 | camoa | unixdawg, where? |
17:51.12 | UnixDawg | from the website |
17:51.20 | LuchoAR | mmmm.. nope.. |
17:51.20 | reseaux | LuchoAR: span=1,1,0,ccs,hdb3,crc4,yellow |
17:51.30 | LuchoAR | that should go in zapata.conf ? |
17:51.31 | jets | hclai: guest ? |
17:51.32 | reseaux | LuchoAR: bchan=1-15,17-31 |
17:51.32 | reseaux | dchan=16 |
17:51.54 | hclai | who has a asterisk up now???? can u guys try to call my asterisk using IAX? |
17:51.59 | reseaux | LuchoAR: no zaptel.cfg |
17:52.02 | jets | hclai: i do, and what user, guest? |
17:52.17 | hclai | leaptron |
17:52.35 | reseaux | hclai: if i want i try! |
17:52.38 | hclai | leaptron@220.255.46.213 |
17:53.00 | hclai | just want to make sure it works |
17:53.12 | hclai | anyone can let me try theirs? |
17:53.16 | jets | <PROTECTED> |
17:53.16 | jets | <PROTECTED> |
17:53.16 | jets | <PROTECTED> |
17:53.37 | jets | nothing. |
17:53.46 | jets | try the guest@misery.digium.com or whatever |
17:54.04 | hclai | can u try to ping me using that address? |
17:54.51 | camoa | sorry for asking, and the most stable zaptel would be??? |
17:55.02 | *** join/#asterisk AtoB (~shahinkha@62.3.220.66) |
17:55.07 | *** join/#asterisk _jackhamr (~jackhamr@64.212.11.53) |
17:55.08 | camoa | the one from the FTP or the one from CVS |
17:55.08 | jets | 64 bytes from 220.255.46.213: icmp_seq=1 ttl=49 time=536.7 ms |
17:55.17 | hclai | ok |
17:55.18 | jets | did you have any logs from me trying to use your IAX? |
17:55.30 | hclai | noo |
17:55.37 | hclai | no response in the log |
17:55.46 | jets | nothing is in the logs? |
17:55.53 | jets | is your machine behind nat? |
17:56.00 | reseaux | TO ALL: Someone have experience in free PRI messages "The caller is not rechable"?thz |
17:56.01 | hclai | what should i config if i want to call the other IAX? |
17:56.04 | hclai | yes |
17:56.13 | jets | just dial 500 on a phone or soft phone |
17:56.18 | jets | if you still have the demo context |
17:56.22 | hclai | i had forward the port to this pc |
17:56.30 | hclai | yes |
17:56.42 | hclai | it did dial to the digium IAX |
17:58.50 | jets | hclai: i bet you need to forward in the IAX ports |
17:58.58 | jets | otherwise other people cannot use your iax |
17:59.01 | hclai | can u try guest@220.255.46.213? |
17:59.20 | hclai | i already forward the port.but let me confirm again |
18:00.00 | jets | ya it's not getthing through to you |
18:00.14 | hclai | the IAX using port 5036 ya? |
18:00.24 | FuzzyCat | by default |
18:00.27 | reseaux | TO ALL: Someone have experience in free PRI messages "The caller is not rechable"?thz |
18:00.32 | hclai | is it TCP or UDP? |
18:00.53 | Bonbon | anyone here from Ireland (UK)? |
18:01.09 | FuzzyCat | ooo... dangerous words Bonbon |
18:01.22 | Bonbon | Fuzzy? |
18:01.25 | rob-- | hclai: udp |
18:01.27 | jets | http://www.voip-info.org/wiki-Asterisk+firewall+rules |
18:01.36 | jets | the above is a great url for letting in iax, sip, etc |
18:01.40 | hclai | jets, can i call ur IAX? |
18:01.45 | FuzzyCat | Ireland is not part of the UK ... they would prolly tell you in less words... |
18:01.59 | Bonbon | Fuzzy: ok, take your point. |
18:02.01 | Bonbon | ha ha |
18:02.02 | jets | hclai: sure, i'm not sure if i have a guest account |
18:02.06 | jets | try guest@12.174.38.5 |
18:02.09 | FuzzyCat | :) |
18:02.33 | *** join/#asterisk alex4152 (~alex4152@hoochie.digium.com) |
18:03.28 | jets | hclai: if you call it, dial extension 6005 |
18:04.16 | h3x | oops |
18:04.32 | h3x | i was using quickbooks, accidently typed my account number in the $ amount and almost sent nevada power 1.6 million dollars. |
18:04.45 | h3x | that would have sucked. |
18:04.51 | ManxPower | h3x, you have 1.6 million dollars in your account |
18:04.58 | ManxPower | ? |
18:04.59 | citats | heh so now we know the first 2 numbers in your account number are 16 |
18:05.01 | h3x | no |
18:05.02 | citats | :) |
18:05.06 | h3x | i guess then it wouldnt have worked huh |
18:05.08 | hclai | jets, i can get through |
18:05.12 | h3x | it would be cool to bounce a check for a million though |
18:05.14 | hclai | can i speak to u? |
18:05.14 | jets | okay dial extension 6005 |
18:05.45 | rob-- | hclai: you can also try guest@haylott.plus.com if you like. |
18:06.02 | jets | hclai: call my iax and dial 6005 |
18:07.00 | ManxPower | you can also try Dial(IAX2/guest@ext-1.fnords.org/2101) if you want to test IAX2 |
18:07.42 | *** join/#asterisk hclai (~hclai@hoochie.digium.com) |
18:07.48 | hclai | jets |
18:08.09 | hclai | whats the extension i should dial? |
18:08.31 | hclai | i would like to try and listen to the quality of the voice |
18:08.54 | jorgeraidel | zoa? |
18:08.58 | hclai | i already did the port forwarding on the router side |
18:09.02 | rob-- | hclai: he said 6005. |
18:09.07 | hclai | why u still cannot call in? |
18:09.09 | jorgeraidel | I have the same problem and i have include iaxtel |
18:09.12 | reseaux | TO ALL: Someone have experience in free PRI messages signalling like this "The caller is not rechable"?thz |
18:09.16 | hclai | ok |
18:09.17 | rob-- | hclai: you can also try guest@haylott.plus.com if you like. |
18:09.19 | hclai | thanks |
18:09.24 | jorgeraidel | text 'asterisk' tries includes non-existant context 'asterisk-usa' |
18:09.24 | jorgeraidel | Jan 16 02:10:11 WARNING[1226400960]: pbx.c:4734 ast_context_verify_includes: Context 'asterisk' tries includes non-existant context 'sipphone' |
18:09.24 | jorgeraidel | Jan 16 02:10:11 WARNING[1226400960]: pbx.c:4734 ast_context_verify_includes: Context 'asterisk' tries includes non-existant context 'shortcut' |
18:10.12 | JerJer | isn't it obvious? |
18:10.28 | extremis | heh, any digium sales monkeys around? |
18:10.40 | ManxPower | jorgeraidel, ANY time you see that message that means there's an include => thatcontext with not [thatcontext] line in extensions.conf. If you get this with the default extensions.conf then report it as a bug to bugs.digium.com |
18:11.13 | jets | hclai: that is jesse you are talking to |
18:11.16 | jets | my coworker |
18:12.31 | extremis | I love getting voicemail when calling digium :) |
18:12.45 | hclai | i see |
18:12.56 | jets | thank you hclai |
18:13.02 | jorgeraidel | ok |
18:13.12 | hclai | but i still cant let u to call in |
18:13.44 | hclai | i got to make this thing work. i tried that day and it waas working |
18:13.54 | hclai | i am not sure what is the problem now |
18:14.36 | jets | hclai: did you look at the URL i gave you? |
18:14.42 | hclai | yes |
18:14.59 | rob-- | hclai: what is your ip address? |
18:16.02 | *** join/#asterisk erubright (~erubright@216.229.189.150) |
18:16.09 | hclai | guest@220.255.46.213 |
18:16.15 | hclai | can u try to call in |
18:16.37 | hclai | i saw my log file and there is someone call to my asterisk |
18:16.40 | hclai | dial 104 |
18:16.44 | rob-- | I can connect with iax1, but not iax2 |
18:16.45 | hclai | extension |
18:16.52 | hclai | ya...only iax |
18:16.56 | jets | oh sorry |
18:16.58 | jets | i was trying iax2 |
18:17.00 | hclai | can u dial wxtension 104? |
18:17.12 | hclai | i would like to talk to u |
18:17.18 | *** part/#asterisk _jackhamr (~jackhamr@64.212.11.53) |
18:17.19 | hclai | thanks for ur help |
18:18.04 | *** join/#asterisk token (~saa@host2.216.41.24.conversent.net) |
18:18.15 | token | bkw you there bud |
18:18.35 | hclai | thanks guys |
18:18.51 | hclai | whats the differnce of IAX and IAX2? |
18:19.36 | rob-- | no prob. |
18:19.36 | jets | hclai what is your extension? |
18:19.45 | jjanzer | i was pretty impressed with the quality of that call hclai, sounded really clear |
18:20.03 | rob-- | I think iax2 uses udp and is better. |
18:20.08 | hclai | 104 |
18:20.16 | bkw_ | most voip uses udp unless its CRACK |
18:20.23 | bkw_ | well let me say all voip uses udp |
18:20.28 | bkw_ | NEXT!!! |
18:20.38 | jets | hclai: now get an iaxtel.com account, my iaxtel number 1 700 226 4222 |
18:20.46 | ManxPower | IAX and IAX2 both use UDP |
18:20.51 | hclai | i see |
18:20.56 | hclai | cool |
18:20.57 | *** part/#asterisk glyph (glyph@h00095b4e65ab.ne.client2.attbi.com) |
18:21.01 | rob-- | Ok, so what's better about iax2? |
18:21.09 | ManxPower | rob--, trunking |
18:21.20 | hclai | let me check my iaxtel account |
18:22.39 | jorgeraidel | what is asp % and udp ? |
18:22.44 | jsmith | rob--: And, IAX1 is going away :-) |
18:23.06 | jsmith | rob--: Mark wrote IAX2 because he didn't like certain things about IAX1... |
18:23.29 | *** join/#asterisk Junbug (~junya@adsl-158-118-24.mia.bellsouth.net) |
18:23.38 | ManxPower | jorgeraidel, I don't know what "asp" is, but % is Percent and UDP is Unicast Datagram Protocol |
18:23.57 | *** join/#asterisk kapejod (~kapejod@p509241B8.dip0.t-ipconnect.de) |
18:24.03 | kapejod | hi there |
18:24.17 | tx | hi kapejod. |
18:25.06 | hclai | jets, i think i will sign up the account later |
18:25.14 | hclai | i face another problem here |
18:25.22 | hclai | i cant have music on hold |
18:25.35 | jets | hclai: do you have mpg123 installed? |
18:25.48 | hclai | the asterisk is playing the musiconhold"default" |
18:25.56 | camoa | hello, i've just compiled zapter 0.8.0 but now i get tons of unresolved symbols with depmod, i got a pretty short kernel an modules, so i don't know what i'm missing, anyone may help me? please |
18:25.59 | jorgeraidel | I can't enter in downlaod asterisk for 0.7.1 |
18:26.02 | hclai | but i cant hear anything from the phone |
18:26.23 | jets | hclai: make sure you have the program mpg123 installed |
18:26.23 | hclai | how to check that? |
18:26.29 | jets | at a shell |
18:26.31 | jets | type mpg123 |
18:26.33 | jets | or locate mpg123 |
18:27.04 | *** join/#asterisk simprix (~simprix@ip-64-32-242-201.dsl.iad.megapath.net) |
18:27.26 | hclai | after that? |
18:27.35 | jets | does it run |
18:27.42 | hclai | i typed "locate mpg123" |
18:27.47 | hclai | nothing happen |
18:28.26 | *** join/#asterisk DL4GRC (~DL4GRC@pD9ECD87A.dip.t-dialin.net) |
18:28.28 | jets | type just mpg123 |
18:28.33 | hclai | i just downloaded the mpg123-0.59q.tar.gz |
18:28.40 | camoa | and the symbols are those like vprintf and stuff |
18:28.49 | hclai | command not found |
18:30.23 | camoa | the mpg123 executable should be located at /usr/bin or /usr/local/bin not remember exactly, but that makes * fail on MOH |
18:30.31 | hclai | the binary program of mpg123 is in the directory but it said command not found |
18:32.46 | *** join/#asterisk lorena26 (~lorena@78.Red-80-37-200.pooles.rima-tde.net) |
18:32.50 | lorena26 | hi |
18:33.31 | reseaux | hi |
18:34.47 | FuzzyCat | hclai: you need to do an 'updatedb' if you just installed it |
18:35.06 | FuzzyCat | then u can do a locate |
18:35.39 | hclai | how to locate? |
18:35.54 | citats | if its in your path type: which mpg123 |
18:37.09 | hclai | the directory i install the mpg123? |
18:37.26 | FuzzyCat | <FuzzyCat> hclai: you need to do an 'updatedb' if you just installed it |
18:37.26 | FuzzyCat | <FuzzyCat> then u can do a locate |
18:38.38 | hclai | yes |
18:38.41 | lorena26 | work now the Cvs ? |
18:38.41 | *** join/#asterisk slacker (~dhollis@washdc3-ar10-4-63-121-042.washdc3.dsl-verizon.net) |
18:38.41 | hclai | i done it |
18:38.44 | hclai | thank u |
18:38.47 | FuzzyCat | np |
18:39.35 | reseaux | lorena:i dont know |
18:40.04 | reseaux | lorena:download the 0.7.1 |
18:42.24 | lorena26 | yes.. thats i Have |
18:42.25 | *** join/#asterisk dannie (~dannie@s1-09.colo.iglou.com) |
18:42.41 | lorena26 | but .. I have a E100P 1 E1 and take audio in Only One way.. |
18:43.03 | vaewyn | anything bigtime wrong with current cvs? I want to check if it will fix my hanging zap channel |
18:44.25 | *** join/#asterisk mbranca (~matteo@ppp-217-133-173-173.cust-adsl.tiscali.it) |
18:45.46 | mbranca | hi all |
18:48.37 | *** join/#asterisk simprix (~simprix@ip-64-32-242-201.dsl.iad.megapath.net) |
18:48.43 | *** join/#asterisk ok_ (~ok@DSL01.212.114.231.34.NEFkom.net) |
18:49.20 | jtodd | I'm behind the curve here - is there a g.726 codec available for integration into * now? I see it in the translation list, but nothing in the cvs trees that seems to reference it... |
18:50.32 | vaewyn | jtodd: I beleive that is the one you have to purchase from digium for licensing reasons |
18:50.45 | bkw_ | its coming |
18:50.47 | vaewyn | not sure though... that might be 729 |
18:50.55 | bkw_ | its not even written yet I don't think |
18:50.55 | jtodd | I was under the impression that the g.726 stuff was "open source" - yeah, what bkw_ said. |
18:51.03 | jtodd | OK. |
18:51.10 | bkw_ | its coming soon from what I hear |
18:51.16 | jtodd | I'll wait patiently. I don't have any g.726 devices yet anyway, I think. |
18:51.27 | PBXtech | 726 like 729? |
18:51.52 | mbranca | nope. you can buy only g.729 |
18:52.06 | mbranca | g.726 isn't anywhere till now... |
18:52.22 | citats | isnt 726 ADPCM? |
18:52.47 | reseaux | lorena:is not a problem of asterisk i think you use a sip phone? |
18:52.50 | bkw_ | nope |
18:52.52 | reseaux | hi mbranca!!! |
18:52.53 | bkw_ | its like it |
18:52.55 | bkw_ | but diffrent |
18:53.04 | *** join/#asterisk Tili (~Tili@202.133.65.187) |
18:53.25 | puzzled | what's with normal calls getting detected as fax calls in today's cvs? |
18:53.35 | citats | bkw_: actually it is ADPCM, i just looked it up... capable of 40, 32, 24, and 16 Kbps |
18:53.47 | FuzzyCat | USE_OLD_CVS_VERSION :P |
18:53.55 | vaewyn | puzzled: is broken DSP... if you uncomment the "OLD_DSP" ish line it fixes it |
18:54.00 | bkw_ | citats no the header stuff is diffrent from what I was told |
18:54.14 | citats | bkw_: different than what? |
18:54.16 | *** join/#asterisk Om3gAnGeL (~Om3gAnGeL@vwmcches.uncg.edu) |
18:54.25 | vaewyn | puzzled: OLD_DSP_ROUTINES in dsp.c |
18:54.30 | puzzled | vaewyn: thanks but I don't understand why things get broken after a stable brnach release. It used to work fine afaik |
18:54.45 | bkw_ | you can't use ADPCM as g726 |
18:54.47 | bkw_ | it just wont work |
18:54.53 | vaewyn | puzzled: they fixed another bug and created this one |
18:54.59 | bkw_ | we have tried :P |
18:55.04 | *** part/#asterisk Om3gAnGeL (~Om3gAnGeL@vwmcches.uncg.edu) |
18:55.20 | citats | bkw_: thats not what i'm saying... i'm saying that g726 is ADPCM transcoding technique |
18:55.59 | bkw_ | ah ok |
18:56.01 | bkw_ | yes that is true |
18:56.07 | bkw_ | but we can only use g726-32 |
18:56.15 | bkw_ | the rest are still under patent |
18:56.18 | vaewyn | puzzled: see http://bugs.digium.com/bug_view_page.php?bug_id=0000696 and http://bugs.digium.com/bug_view_page.php?bug_id=0000649 for more information |
18:56.20 | bkw_ | g726-32 is accually g721 |
18:56.27 | puzzled | vaewyn: thanks |
18:56.32 | vaewyn | np :} |
18:56.37 | bkw_ | which is really old but compatible with g726-32 (Because they are the same damn thing) |
18:56.46 | Seba2 | hello |
18:57.10 | PBXtech | is there an autoconnect function? so if a handset is picked up it automatically calls security? |
18:57.36 | citats | PBXtech: immediate = yes |
18:57.40 | Seba2 | somebody knows wich variable can tell me information about calltime and duration of a call in AGI? |
18:57.57 | vaewyn | PBXtech... easy... just make your s extension dial them :} drop them into that context and voila! :} |
18:58.35 | vaewyn | Ok... who wants to donate me a quad span T1 card? ;P |
18:58.47 | PBXtech | so it wont want for any digits to be input? pickup and it rings.. right? |
18:59.03 | vaewyn | PBXtech: pretty much :P |
18:59.30 | PBXtech | so how would the dial string look? |
19:00.04 | vaewyn | PBXtech: like exten => s,1,Dial,Zap/1 or such... |
19:00.20 | vaewyn | make the default context for the line include that exten and you are set |
19:00.24 | PBXtech | oh the s thinger.. gotcha thanks |
19:00.26 | puzzled | vaewyn: will I get the OLD_DSP_ROUTINES back also when I enable it in the Makefile or do I need to do it in dsp.c? |
19:00.59 | vaewyn | puzzled: Up to you.. if you define that in the makefile it will do it... if you uncomment the define in dsp.c it works also... up to you :P |
19:01.25 | puzzled | vaewyn: ok, thanks |
19:01.29 | vaewyn | np |
19:03.48 | tclark | vaewyn: current code does not have flag in dsp.c any longer:) |
19:04.00 | vaewyn | tclark: ahh :P |
19:04.20 | vaewyn | am just updating now... stupid fxs card keeps hanging in ring state sol.. |
19:07.05 | *** join/#asterisk zoa (~john@kn-upc-1.kuleuven.net) |
19:07.42 | *** join/#asterisk Moc (~Moc@modemcable213.103-70-69.mc.videotron.ca) |
19:09.05 | bkw_ | Moc |
19:09.09 | bkw_ | we have a patch for you to test |
19:09.33 | bkw_ | who here things asterisk should continue on if it can't load chan_zap or other modules? |
19:09.46 | zoa | depends bkw |
19:09.50 | zoa | on a start ? |
19:09.52 | bkw_ | (right now it segfaults on misconfigured stuff) |
19:09.54 | zoa | on a startup ? |
19:09.55 | vaewyn | should yell no matter what |
19:10.04 | vaewyn | yell loudly |
19:10.07 | zoa | bkw, i think it should work if G729 license starts fucking up |
19:10.13 | bkw_ | haha |
19:10.14 | zoa | as happened on my server this week |
19:10.26 | zoa | suddenly safe_asterisk coredumps all the time |
19:10.34 | zoa | and restart |
19:10.34 | FuzzyCat | ouch! |
19:10.34 | zoa | and restart |
19:10.34 | FuzzyCat | Jan 15 20:09:06 WARNING[180245]: chan_zap.c:1423 zt_call: Unable to ring phone: |
19:10.34 | FuzzyCat | Device or resource busy |
19:10.34 | zoa | and restart |
19:10.40 | bkw_ | ok right now if you have a mismatch in your signalling it will segfault |
19:10.41 | zoa | turns out my g729 license is gone again |
19:10.54 | zoa | bkw, that should not happen ! |
19:10.55 | bkw_ | ok anthm |
19:10.58 | bkw_ | lets do this |
19:10.58 | vaewyn | wb kram |
19:11.03 | kram | thanks but it's only for a sec |
19:11.06 | kram | but i'll be back sooon |
19:11.07 | vaewyn | :P |
19:11.19 | zoa | bkw, a question i have had several times since 0.7.0 |
19:11.34 | zoa | could people use asterisk without compiling zaptel before 0.7.0 ? |
19:11.39 | bkw_ | yes |
19:11.47 | zoa | because now it doesnt work |
19:11.57 | denon | zoa: thats a feature :) |
19:12.07 | bkw_ | here in a few chan_zap will no longer segfault |
19:12.14 | bkw_ | when its totally screwed |
19:12.18 | bkw_ | it will just stop and say BLAH |
19:12.21 | bkw_ | like it should |
19:12.44 | denon | Zaptel: URGENT: I have wet my pants. Please reboot. |
19:13.42 | vaewyn | hehehe I like that |
19:15.22 | bkw_ | that seems to have fixed it |
19:16.16 | vaewyn | even cooler would be config definable actions when things fail... then could have it reboot itself... or page you when things go awry :P |
19:16.23 | vaewyn | but that is a daydream :P |
19:16.43 | vaewyn | living in the past I tell you :P |
19:16.58 | FuzzyCat | well the present isn;t working |
19:17.42 | vaewyn | Heh... FuzzyCat is playing "butterfly effect" with his * install |
19:18.03 | vaewyn | only more serious... and without the moronic banter |
19:18.08 | vaewyn | I hope at lest |
19:18.12 | vaewyn | least even |
19:19.40 | lorena26 | zoa have u problem with zaptel in the last 0.7.1? |
19:19.56 | zoa | nopez i was wrong |
19:20.00 | zoa | i got a flood of messages |
19:20.05 | zoa | but when i retried all went fine |
19:20.19 | lorena26 | u have card for E1 or T1? |
19:20.21 | zoa | probably a little synching problem when i unloaded en reloaded the modules that fast |
19:20.23 | zoa | E1 |
19:20.30 | lorena26 | ok, I the same |
19:20.37 | lorena26 | take u only way AUDIO ? |
19:20.45 | zoa | nopez, i prefer 2 way |
19:20.46 | zoa | :-p |
19:20.50 | lorena26 | jajajaja |
19:20.57 | zoa | hey, where are you from ? |
19:21.05 | zoa | are you dutch ? |
19:21.05 | lorena26 | sorry.. U undertand my question ? |
19:21.05 | zoa | german? |
19:21.07 | lorena26 | Spain |
19:21.14 | zoa | yes, |
19:21.17 | zoa | i don't have that problem |
19:21.25 | vaewyn | Ok... if I am ringing a remote context on another server do I need anything more than: |
19:21.25 | vaewyn | [bell] |
19:21.26 | vaewyn | exten => s,1,Dial,Zap/1 |
19:21.26 | vaewyn | ? |
19:21.37 | vaewyn | (That is the context I am calling |
19:21.39 | vaewyn | ) |
19:21.41 | lorena26 | can u give me a test config ? |
19:21.45 | bkw_ | Moc go test bug 851 now |
19:21.46 | lorena26 | may be is my setup problem |
19:21.48 | bkw_ | tell me if you like that |
19:21.59 | zoa | 851 ? we are already at 851 ? |
19:22.06 | bkw_ | yes |
19:22.06 | zoa | what happened while i was eating ! |
19:22.13 | vaewyn | zoa++ :} |
19:22.29 | zoa | bkw, how could i retrieve the caller id from an asterisk c app ? |
19:22.31 | zoa | not agi ? |
19:22.37 | zoa | pbx_helper blabla ? |
19:23.30 | FuzzyCat | hang zoa |
19:23.43 | vaewyn | that's kindof mean Fuzzy |
19:23.45 | vaewyn | :} |
19:24.01 | angler | my coworker has a turbo in his prelude |
19:24.09 | zoa | but i already know my last wishes !! |
19:24.13 | vaewyn | hahah nice angler |
19:24.16 | puzzled | hehe |
19:24.18 | FuzzyCat | <PROTECTED> |
19:24.22 | angler | he is lying |
19:24.23 | angler | haha |
19:24.31 | zoa | oh shit |
19:24.33 | zoa | i asked caller id |
19:24.36 | zoa | i want accountcode :) |
19:24.37 | angler | nother guy at my pc |
19:24.38 | zoa | sorry dude |
19:24.39 | zoa | :) |
19:24.42 | FuzzyCat | ast_callerid_parse (old_cid, &name, &num); /* this destroys the original string */ |
19:26.24 | zoa | bkw, i'll test that puppy |
19:27.44 | *** join/#asterisk juice (~juice@user221.net1199.mo.sprint-hsd.net) |
19:28.21 | *** join/#asterisk Mike (~mike@dsl-200-67-4-96.prod-infinitum.com.mx) |
19:28.25 | zoa | bkw, look at my remark for chan_zap 851 |
19:29.03 | bkw_ | zoa it doesn't |
19:29.11 | zoa | you sure ? |
19:29.11 | bkw_ | you can accually unload chan_zap during a meetme |
19:29.16 | bkw_ | done it many times lastnight |
19:29.23 | bkw_ | all that stuff doesn't tie to the channel usage |
19:29.27 | zoa | how about during trunking ? |
19:29.34 | bkw_ | same thing |
19:29.48 | zoa | k, so its the kernel modules that do timing ? |
19:30.00 | bkw_ | yes sir |
19:30.05 | zoa | groovy |
19:30.13 | bkw_ | so chan_zap will no longer seg either on load |
19:30.22 | bkw_ | grab the latest patch |
19:30.29 | bkw_ | I changed some of the verbose stuff we added for testing |
19:30.33 | bkw_ | I forgot to remove one line |
19:30.38 | zoa | SYSTEM WARNING: extract() expects first argument to be an array |
19:30.38 | zoa | (file_download.php: line 50) |
19:30.43 | zoa | mantis is fucked |
19:30.48 | bkw_ | also anthm did most of the work tracking down that problem |
19:30.51 | bkw_ | reload the page |
19:31.41 | Seba2 | hello |
19:31.52 | bkw_ | now if chan_mgcp and chan_skinny would just unload |
19:32.27 | zoa | FuzzyCat= do you also know how to grab the accountcode ? |
19:32.32 | zoa | as i made a typo ? |
19:32.36 | zoa | i didnt want the caller id |
19:32.38 | zoa | but the accountcode |
19:32.57 | FuzzyCat | try chan->accountcode :) |
19:32.58 | bkw_ | muhahahahahahahhaha |
19:33.20 | bkw_ | if you have the cdr struct its cdr->accountcode |
19:34.03 | bkw_ | ok who wants to join the conf |
19:34.11 | bkw_ | JerJer you alive? |
19:34.13 | zoa | bkw, i have no headset here :) |
19:34.24 | zoa | do you have speech2text ? :) |
19:34.25 | sxpert | ~seen levon |
19:34.35 | | levon <~levon@mail.feature-it.de> was last seen on IRC in channel #asterisk, 16d 10h 15m 4s ago, saying: ';)'. |
19:34.35 | bkw_ | Moc you testing bug 851 since you were complaining about it so much the other day? |
19:34.35 | FuzzyCat | look at the doxygen docs for the struct ast_chan ZOP |
19:34.35 | FuzzyCat | errm zoa |
19:35.40 | Seba2 | guys |
19:35.52 | Seba2 | how can I back from an AGI file? |
19:36.19 | Seba2 | after processing AGI * continues processiong next in extensions.conf? |
19:37.42 | bkw_ | next on my list |
19:37.42 | *** join/#asterisk ^sly^ (~jelque@65.70.27.77) |
19:37.50 | bkw_ | res_musiconhold should be able to unload now |
19:37.52 | bkw_ | muhahahahhahahahaha |
19:38.12 | Mike | hows 0.7.1 doing? |
19:38.26 | bkw_ | if we can unload res_musiconhold you can in theory unload the kernel modules |
19:38.42 | *** join/#asterisk cypromis (~michael@217.11.142.161) |
19:39.12 | bkw_ | then load them back |
19:39.21 | bkw_ | and bring the modules back in without restarting * |
19:39.31 | bkw_ | har har har |
19:39.54 | FuzzyCat | and I would want to do that ... why? |
19:40.24 | vaewyn | lets you keep SIP or H.323 stuff running even though your zap is messed and such |
19:40.47 | Seba2 | NOTICE[1296014640]: rtp.c:418 ast_rtp_read: Unknown RTP codec 19 received |
19:40.52 | FuzzyCat | is my zap aint working I don;t want sip working |
19:40.52 | Seba2 | What's that? |
19:40.53 | bkw_ | turn off VAD |
19:40.57 | puzzled | anyone seen "stop when convenient" drop a bunch of IAX channels? |
19:40.58 | *** join/#asterisk Inv_Arp (~junya@65.2.9.250) |
19:41.10 | cypromis | maybe they where inconvenient ? |
19:41.11 | Seba2 | ok |
19:41.13 | bkw_ | FuzzyCat if you setup correctly you can direct calls out another provider if zap is down |
19:41.16 | zoa | bkw, we really need a FAQ :( |
19:41.36 | zoa | and people get kicked for asking a question in the FAQ |
19:42.08 | FuzzyCat | bkw_: if my zap if fucked i don;t want to redirect... I want to fix zap |
19:42.46 | zoa | cleopatra*CLI> unload chan_zap |
19:42.46 | zoa | Unable to unload resource chan_zap |
19:42.46 | zoa | <PROTECTED> |
19:42.46 | zoa | Jan 15 20:42:30 NOTICE[18451]: chan_zap.c:3588 zt_read: Fax detected, but no fax extension |
19:42.46 | zoa | cleopatra*CLI> unload chan_zap.so |
19:42.47 | zoa | Unable to unload resource chan_zap.so |
19:42.49 | zoa | Jan 15 20:42:35 WARNING[17426]: loader.c:119 ast_unload_resource: Soft unload failed, 'chan_zap.so' has use count 1 |
19:43.03 | zoa | cool |
19:43.07 | jets | FuzzyCat: But in in the intereium of 45 minutes i don't want my CEO mad he can't make a call |
19:43.08 | zoa | you cant unload if someone is calling :) |
19:43.28 | bkw_ | nope |
19:43.31 | bkw_ | it won't let you |
19:43.34 | bkw_ | usecount is > 0 |
19:44.00 | bkw_ | zoa if you have E1 it will take like 3 seconds for it to respond to the unload |
19:44.11 | bkw_ | because it has to kill the monitor threads for the D channels |
19:44.16 | FuzzyCat | jets: if you zap is fucked then you should hide under your desk to get away from the CEO |
19:44.22 | bkw_ | and if that fails you have larger problems |
19:44.34 | zoa | yeah |
19:44.36 | zoa | it works great |
19:44.43 | zoa | tried to break it |
19:44.45 | zoa | didnt work :) |
19:44.54 | zoa | got an S-frame while it was reloading |
19:44.57 | vaewyn | FuzzyCat: for some installations yes... for some if the zap is down ohh well.. you have lost 10% of your call volume... woopty do :} |
19:45.06 | zoa | btw bkw |
19:45.20 | zoa | it would be great if soft hangup ZAP/1-1 would work |
19:45.22 | zoa | euh |
19:45.27 | bkw_ | it does |
19:45.28 | zoa | soft hangup ZAP/1 |
19:45.32 | bkw_ | press tab |
19:45.33 | zoa | not /1-1 |
19:46.17 | bkw_ | zoa did you restart before you did the unload with the new chan_zap in place the preivous one wouldn't unload |
19:46.32 | zoa | no |
19:46.35 | FuzzyCat | vaewyn: yes, but my point was more to do with the massive effort to allow the unloading and loading of modules when there are still bugs to fix |
19:46.38 | bkw_ | thats why it didn't unload |
19:46.44 | zoa | no no it unloaded fine |
19:46.46 | zoa | it worked great |
19:46.47 | bkw_ | ah ok |
19:46.49 | bkw_ | thank you |
19:46.50 | bkw_ | haha |
19:46.56 | zoa | i did restart asterisk before testing yes |
19:46.58 | bkw_ | was about to say we tested that part ALOT lastnight |
19:47.04 | zoa | i did unload |
19:47.05 | zoa | and reload |
19:47.08 | zoa | and it came back up |
19:47.11 | bkw_ | :) |
19:47.13 | vaewyn | FuzzyCat: Ahh yes.. but for some of these bugs this lets them restart faster to fix it :P |
19:47.14 | bkw_ | handy dandy eh? |
19:47.24 | zoa | had to hangup some conversations though :) |
19:47.25 | zoa | ssst |
19:47.39 | bkw_ | haha |
19:48.09 | bkw_ | now chan_mgcp and chan_skinny need to be fixed so they can unload |
19:48.18 | zoa | that would also be a great app: something to no longer accept new calls |
19:48.25 | bkw_ | correct |
19:48.33 | vaewyn | amen... like an /etc/nologin for * :P |
19:48.36 | bkw_ | yep |
19:48.46 | zoa | without having to reload and also kill the other channel types |
19:49.08 | bkw_ | well you can unload and change signalling also and load it back |
19:49.12 | bkw_ | just amke sure you run ztcfg first |
19:49.13 | bkw_ | haha |
19:49.24 | bkw_ | if you don't it will bitch at you and refuse to load it |
19:50.09 | zoa | bkw, i think a patch to stop chan_zap from flooding when something is wrong with the signalling would be great too |
19:50.13 | zoa | if something goes fubar |
19:50.22 | zoa | i sometimes start flooding your logfiles |
19:50.40 | *** join/#asterisk Bubbag1 (~Bubbag1@12.10.10.226) |
19:50.42 | zoa | and believe me, a 16krpm disk fills very fast :( |
19:50.52 | *** join/#asterisk Ramereth (~lance@ramereth.registered.freenode) |
19:51.04 | jjanzer | 16k, that's an odd number.... usually you see around 15k |
19:51.19 | bkw_ | like a no signalling errors |
19:51.22 | bkw_ | and a signalling errors |
19:51.37 | extremis | so if the telco hands me a t1 with the first 6 channels allocated for voice is there a D channel (he said it wasn't a pri) for signaling? |
19:51.53 | bkw_ | its a CT1 I bet |
19:52.05 | zoa | 15 it is :) |
19:52.09 | extremis | yyeah, I heard him say that |
19:52.15 | extremis | whats a ct1? |
19:52.21 | bkw_ | Channelized T1 |
19:52.36 | extremis | anything special in my zaptel.conf I need? |
19:52.46 | bkw_ | the signalling type is em_w or em |
19:53.05 | bkw_ | chan_zap is my bitch now |
19:53.09 | bkw_ | MUHAHAHA |
19:53.22 | outtolunc | ut oh |
19:53.24 | extremis | bkw_: so is there any drawbacks to a CT1? |
19:53.29 | extremis | are there any drawbacks rather |
19:53.32 | bug847 | WOW! $1050 bounty on the zap drivers now |
19:53.45 | Seba2 | SOMEBODY CAN GIVE ME A URL ABOUT AGI WITH EXAMPLES AND GOOD EXPLANATION? |
19:53.48 | bkw_ | extremis you can't sound outbound callerids |
19:53.55 | extremis | sound? |
19:53.56 | bkw_ | Seba2 SEARCH THE MAILING LIST AND STOP YELLING |
19:54.02 | bkw_ | s/sound/set/ |
19:54.05 | bkw_ | fuck I can't think |
19:54.12 | bkw_ | Seba2 www.voip-info.org |
19:54.25 | Seba2 | tnx |
19:54.43 | extremis | ahh... bkw: they are delivering me a T1 with voice on some channels and data on the other, and it hits an adtran and it breaks it out to ethernet and a T1... should I tell them to configure it a specific way? |
19:54.53 | extremis | because the guy I'm talking to can do it how I want it |
19:54.59 | *** join/#asterisk zoa (~john@kn-upc-1.kuleuven.net) |
19:54.59 | bkw_ | extremis asterisk can do what you want so you don't have to break it out |
19:54.59 | extremis | I just don't know how I want it :) |
19:55.00 | zoa | Mantis - All Projects |
19:55.00 | zoa | APPLICATION ERROR #200: ERROR: A required parameter to this page was not found. |
19:55.02 | zoa | (gpc_api.php: line 37) |
19:55.06 | zoa | matis is not doing so good today |
19:55.14 | bkw_ | mantis is crap |
19:55.17 | extremis | bkw_: whatcha mean? |
19:55.32 | extremis | I don't want my internet traffic terminating on my pbx :) |
19:55.36 | bkw_ | it can |
19:55.45 | bkw_ | otherwise I have no clue how to tell youto set it up |
19:55.51 | bkw_ | :( |
19:56.10 | zoa | what do you mean extremis : |
19:56.15 | zoa | i missed part of your question |
19:56.21 | bkw_ | <extremis> ahh... bkw: they are delivering me a T1 with voice on some channels and data on the other, and it hits an adtran |
19:56.21 | bkw_ | <PROTECTED> |
19:56.36 | extremis | I like having it broken out via ethernet |
19:56.53 | extremis | that way my PBX isn't exposed to the internet |
19:57.04 | bkw_ | why not no voip? |
19:57.07 | extremis | but... I do want to send caller id info on outbound calls |
19:57.13 | bkw_ | you can't with CT1 |
19:57.42 | extremis | so when it breaks out the data dn voice can it be built diff on the voice side? |
19:57.59 | zoa | should chan->account also contain the accountcode bkw ? |
19:58.07 | bkw_ | extremis prob not |
19:58.13 | zoa | i'm telling my programmer what to do |
19:58.16 | *** join/#asterisk oej2 (~opr@apollo.webway.se) |
19:58.19 | zoa | but he doesnt know shit about asterisk |
19:58.24 | zoa | and he has no axx to the server :) |
19:58.32 | extremis | bkw_: so the best option is to terminate teh full t1 at teh T100P card and then route the data to a seperate NIC |
19:58.36 | bkw_ | i'm learning this stuff so FAST |
19:58.53 | bkw_ | extremis still without PRI you can't set the stuff |
19:59.02 | extremis | oh, so I need a PRI |
19:59.05 | bkw_ | yes |
20:00.04 | jets | ya sending digits down a normal t1 with dtmf is kind of a pain |
20:00.14 | bkw_ | if they let you do that |
20:00.21 | bkw_ | MF is what you would have to send |
20:00.25 | *** join/#asterisk brent21 (~bdf@dpvc-207-68-114-121.alt.east.verizon.net) |
20:00.26 | jets | i suffered getting this done correctly with a cisco router |
20:00.28 | jets | ya sorry, mf |
20:00.30 | *** join/#asterisk clh (~me@216.253.86.210) |
20:01.02 | extremis | bkw_: how does the zaptel drivers break out the data? does it create a network interface like the NICs do? |
20:01.14 | kram | greets brent21 |
20:01.48 | *** join/#asterisk glLoadIdentity (~ghjkl@abn143-215.interaktif.net.tr) |
20:02.18 | brent21 | hello, hows it going today? |
20:03.16 | brent21 | Just got my t100P card from digium, great work! reading all about my new Adit 600 right now :) |
20:03.39 | *** join/#asterisk ecs (~ecs@hoochie.digium.com) |
20:03.45 | Cripon | anyone use diax in here? |
20:03.55 | Cripon | I can place calls, but I've only been able to ring it once |
20:04.08 | Cripon | now it says, "Event for non-existant even. dropping." |
20:04.36 | Cripon | substitute session for even there |
20:05.12 | kram | cool :) |
20:05.14 | bkw_ | extremis depends on how they send you the data |
20:06.26 | vaewyn | mmmm channel bank... can't wait for mine to come in :P |
20:07.03 | UnixDawg | ok grr this bites |
20:07.05 | tclark | Cripon: it might be that bug (out of sequence inbound iax2 event) in the iaxlcient ml that is having a hard time getting resolved in IAX2 that the iaxclient.sf.net apps use |
20:07.31 | Cripon | tclark: that sucks. :( |
20:08.15 | kram | tclark: what's your issue with iax2? |
20:08.27 | *** join/#asterisk PilotPTK-Home (~trillian@pcp02587722pcs.shlb1201.mi.comcast.net) |
20:09.05 | brent21 | for the Adit 600, do I leave it on the CB to Loop Start and just use Kewlstart in * config files, or enable Loop Start with Calling Party Disconnect and still use Kewlstart? |
20:09.09 | Cripon | kram: hey mark.. it's christian.. I'm having problems ringing a diax client. |
20:09.10 | UnixDawg | anyone get *69 to work on asterisk |
20:09.14 | tclark | mark steven sokol has track down an issue on the iax2 on win32 where iax2 event come in out of sequence |
20:09.47 | *** join/#asterisk russT (~rusty@65-101-255-24.dnvr.qwest.net) |
20:09.56 | Cripon | tclark: bingo.. switched to iax signalling and it works like a charm |
20:10.04 | reseaux | hi to all!! |
20:10.21 | Moc | salut |
20:10.22 | reseaux | I have a problem with Signaling of libpri... |
20:10.23 | Cripon | reseaux: hello |
20:10.26 | PBXtech | is there a design flow for *? |
20:10.27 | tclark | yea mark even you could work with steve to solve that we be doing a major ahhpy dane |
20:10.31 | reseaux | hi Cripon |
20:10.39 | tclark | err HAPPY dance |
20:11.13 | reseaux | focus in "establishing before answer" |
20:11.30 | reseaux | some help? |
20:11.31 | reseaux | :-) |
20:11.54 | reseaux | this type of function is called "early B3" in CAPI |
20:12.46 | ecs | I am having issues with astman, I get an error, broken pipe |
20:12.53 | ecs | after i log in |
20:13.14 | hclai | jets, are u there? |
20:13.37 | reseaux | some help please :-) |
20:13.53 | hclai | anyone try to use GotoIfTime? |
20:14.50 | hclai | is this syntax correct? 200,1,GotoIfTime(4:00-5:00,mon-fri,*,*,sleep,s,1) ? |
20:14.58 | hclai | anyone can help? |
20:15.05 | *** join/#asterisk ssokol (~ssokol@64-151-42-28-dhcp-kc.everestkc.net) |
20:15.14 | jets | hclai: yes |
20:15.35 | hclai | but it didnt jumo to that context |
20:15.46 | hclai | it just play the next priority |
20:15.57 | jets | is it from 4am to 5am mon-fri? |
20:16.02 | hmodes | grrrr |
20:16.18 | hmodes | it's difficult to justify telecommuting when * keeps thinking conference calls == busy signals |
20:16.20 | PBXtech | bkw, is there a design flow doc? |
20:16.24 | hclai | yes |
20:16.40 | *** join/#asterisk Junbug (~junya@adsl-80-16-148.mia.bellsouth.net) |
20:16.56 | hclai | any idea why it doesn't works? |
20:17.16 | jets | nope |
20:17.18 | extremis | <PROTECTED> |
20:17.25 | extremis | you don't mean 5pm? |
20:17.29 | jets | did you check voip-info.org |
20:17.32 | hclai | did u use this before? |
20:17.35 | hclai | ya |
20:17.37 | hclai | i check |
20:17.45 | jets | what time is it |
20:17.49 | jets | in 24 hour format |
20:17.57 | jets | in singapore i mean |
20:18.02 | vaewyn | on that machine :} |
20:18.09 | hclai | yes |
20:18.16 | extremis | yes |
20:18.35 | hclai | u meant the machine got to be 24 hours format? |
20:18.44 | *** join/#asterisk mike1 (~mike@hoochie.digium.com) |
20:18.45 | jets | on your server type |
20:18.46 | hclai | but here is morning |
20:18.46 | jets | date |
20:18.48 | jets | and paste it here |
20:19.26 | extremis | so does CT1 support e&mwinkstart ? |
20:19.42 | hclai | Thu Jan 15 04:19:09 SGT 2004 |
20:19.57 | Cripon | Jan 15 14:18:04 NOTICE[114696]: chan_iax2.c:4718 socket_read: Registration of 'kraven' rejected: Registration Refused |
20:19.57 | Cripon | toonville*CLI> /usr/sbin/safe_asterisk: line 77: 12198 Killed asterisk ${ASTARGS} 1>&/dev/${TTY} </dev/${TTY} |
20:19.57 | Cripon | Asterisk ended with exit status 137 |
20:19.57 | Cripon | Asterisk exited on signal 9. |
20:19.58 | Cripon | Automatically restarting Asterisk. |
20:20.00 | Cripon | Disconnected from Asterisk server |
20:20.02 | Cripon | anyone? |
20:20.17 | *** join/#asterisk lkc (~lkc@hoochie.digium.com) |
20:20.25 | Seba2 | Sorry again |
20:20.27 | mike1 | Can someone tell me why this does not work ==> exten => 0,1,Dial,ZAP/21 |
20:20.33 | Seba2 | What about EAGI? |
20:20.44 | Seba2 | I don't know nothing about that |
20:20.47 | jets | mike1: zap/2-1 ? |
20:20.48 | Seba2 | What is? |
20:21.21 | mike1 | zap/21 - I am assuming that the 21 is the channel. Is that correct |
20:21.24 | hclai | jets, did u see my date just now? |
20:21.30 | zoa | mike: correct |
20:21.34 | hclai | what should i do? |
20:21.39 | jets | mike1: well you have spans, and channels |
20:21.48 | zoa | that should work |
20:21.51 | mike1 | Asterisk complains that it can find "ZAP" |
20:21.52 | zoa | jets no |
20:21.57 | zoa | its is correct like that |
20:22.06 | *** join/#asterisk PilotPTK-Home (~trillian@pcp02587722pcs.shlb1201.mi.comcast.net) |
20:22.09 | zoa | you don't need to use the span when you dial |
20:22.18 | zoa | you prolly dont have zaptel loaded |
20:22.19 | mike1 | unable to create channel type Zap |
20:22.39 | zoa | do a cat /proc/zaptel/* |
20:22.45 | zoa | tell us what it gives you |
20:23.26 | mike1 | shows all 24 channels as FXOKS |
20:23.45 | mike1 | do I need to specify ZAP somewhere? |
20:23.49 | *** join/#asterisk c4uldr0n (~korri@ns.elitecomp.net) |
20:23.51 | mike1 | is that a context |
20:24.16 | jets | no zap is a device much like sip/ |
20:24.28 | jets | you need a zap interface in zapata.conf |
20:24.29 | jets | etc |
20:24.43 | mike1 | A zap interface? |
20:25.19 | jets | Like a T1 span, or an fxs or fxo card |
20:25.21 | jets | etc |
20:25.46 | c4uldr0n | I'm very new to the Asterisk program, I'm having a problem with the zaptel installation... it fails when I do a "make config"..... error: make: *** [config] Error 127 |
20:26.07 | c4uldr0n | can somebody give me some insight? |
20:26.31 | hmodes | oh, i had tone and silence turned off |
20:26.35 | c4uldr0n | this is with the CVS install |
20:26.36 | hmodes | silly clean tree |
20:28.28 | lorena26 | Jan 15 21:28:12 NOTICE[1258761520]: rtp.c:418 ast_rtp_read: Unknown RTP codec 19 received |
20:28.28 | lorena26 | how I can fix it ? |
20:28.34 | lorena26 | I call with G729 |
20:28.35 | vaewyn | Ok... doing a Dial(IAX/bridge:testing@143.207.x.y/s@bell,15) (bell being context with a single extension to Dial the Zap interface) should it be ringing by itself... or does the dial have to have the ,r for the user to hear the ring? |
20:29.31 | vaewyn | (end phone is rining fine... just caller doesn't seem to be getting ring indication) |
20:29.59 | *** join/#asterisk point (~litw@195.161.106.222) |
20:31.38 | vaewyn | and another question... do I have to call a context? or can I call Zap/1 directly on the remote machine? |
20:33.09 | vaewyn | Hmm.. no.. it seems to want a context/extension :{ |
20:35.42 | *** join/#asterisk krishna (~krishna@hoochie.digium.com) |
20:36.34 | *** join/#asterisk Inv_Arp (~junya@adsl-80-16-148.mia.bellsouth.net) |
20:38.37 | data[out^london | doh? 0.7.1 :) |
20:39.08 | mike1 | How do I correct this ==> No channel type registered for 'ZAP' |
20:39.19 | *** join/#asterisk groz (whocares@h24-87-70-66.vc.shawcable.net) |
20:39.41 | c4uldr0n | I had to manually install my zaptel init scripts because "make config" gives me an error... that sucks huh |
20:40.05 | UnixDawg | anyone have calltrace working |
20:40.22 | UnixDawg | aka last nmbr callback |
20:40.35 | groz | good afternoon folks, is it acceptable to ask a couple of probably dumb newbie q's here ? |
20:40.51 | Cripon | groz: sure |
20:41.06 | Cripon | groz: they may not get answered, but it doesn't hurt to ask |
20:41.08 | groz | ok, the thing i have not been able to quite fathom from rtfm is this |
20:41.18 | vaewyn | 42 |
20:41.19 | vaewyn | :P |
20:42.02 | groz | if i set up asterisk at 2 locations, can i interconnect them in such a way that they have the appearance of being 'one big pbx' with my link between them being all ip on private network ? |
20:42.44 | vaewyn | groz: yes... VERY easily... :} |
20:42.50 | groz | effectively what i'm looking at doing, is trying to get seamless link between 2 locations, one of them is very remote, but it has data via satellite link |
20:43.04 | groz | we have 128K dedicated link over a satellite connection |
20:43.06 | vaewyn | eww... sattelite... your gonna have latency issues |
20:43.15 | groz | ahh, latency we already deal with |
20:43.16 | vaewyn | but yes... it would work |
20:43.24 | groz | using globalstar phones at $1+ per minute |
20:43.31 | vaewyn | ouch hhehe |
20:43.31 | groz | so, latency is the least of the issues, never get rid of it |
20:44.16 | reseaux | I NEED HELP With "early B3 connect" on libPRI Thz |
20:44.24 | groz | ok, that's basically the answer i was looking for, think i'm gonna start setting up a test scenario here, and ordering some hardware to try it all out |
20:44.40 | vaewyn | basic answer is... the remote * install can have a switch statement in the config pointing to your main * box... the dialplan on the main is available to both machines... and it figures everything out for you :P |
20:44.44 | groz | suggestions on the 'right' stuff to buy for setting up a test bench to demo it all ? |
20:45.03 | vaewyn | digium hardware... whatever is appropriate for your needs |
20:45.07 | groz | i was thinking the digium stuff just cuz they seem to be behind it all |
20:45.12 | groz | but never hurts to check |
20:45.16 | vaewyn | *nods* |
20:45.25 | groz | ok, answers that one too, this was 'to easy' today |
20:45.37 | vaewyn | :} |
20:45.53 | Cripon | groz: you want to stick with digium hardware. What kind of test bed are you looking at? |
20:46.06 | vaewyn | If you want H.323 gear or such you would go cisco but.. other than that digium rocks da house :P |
20:46.24 | groz | i'm looking at a couple of pc's with just enuf hardware to support one connection at a time |
20:46.32 | reseaux | I NEED HELP With "early B3 connect" on libPRI Thz |
20:46.35 | groz | basically what i need to do for a true test is this |
20:46.44 | groz | plug an analog phone in on one end, another on the other end |
20:46.44 | *** join/#asterisk Buana (~thomasn@Gcb0a.g.pppool.de) |
20:46.50 | groz | have the ability to dial between them |
20:47.02 | Cripon | groz: why not test with soft phones? |
20:47.05 | groz | then i'll put em on opposite ends of the satellite link, and we'll see how it works |
20:47.14 | Cripon | groz: have you used linux before? |
20:47.16 | groz | oh crip, i have to demonstrate 'real phones' in this case |
20:47.23 | groz | yes, do embedded stuff for a living |
20:47.27 | groz | so not scared of linux |
20:47.35 | groz | <---- rolls own lfs for every project |
20:47.40 | vaewyn | :} |
20:47.57 | Cripon | groz: ok.. for testing you could use s100u's but they're flakey.. You'd need tdm400p's to do a box to box call with no external lines |
20:48.01 | Buana | cool, just found mail-archive from may-2016 on asterisk-side:) |
20:48.03 | groz | first thing i'll be looking at, is load this all up on a dedicated box, busybox boot and 'just run' from flash |
20:48.09 | *** join/#asterisk PlainWhiteTrash (~PlainWhit@ps-ast-router.papersoft.com) |
20:48.20 | Cripon | groz: well good luck |
20:48.23 | groz | ok crip, thanks |
20:48.36 | groz | what i was looking at was the digium devloper kit that comes with one of each |
20:48.40 | groz | just order 2 of those |
20:48.52 | Cripon | do you need the x100p? |
20:48.52 | groz | lets me play with any scenario |
20:49.00 | vaewyn | that would work... would let you take in external calls also to play with |
20:49.03 | groz | it leaves me options to connect it to the existing systems |
20:49.17 | Cripon | yep.. get the tdm dev kit.. don't get light |
20:49.20 | Cripon | it sucks |
20:49.24 | vaewyn | :} |
20:49.30 | groz | yah, already saw that, saw 'usb' and said 'nope' |
20:49.50 | Exomorph | bkw_: You around? |
20:50.36 | groz | my plan is basically a dedicated box running from a flash disk, appropriate hardware plugged in, no monitors or keyboards, just treat it like an appliance, has an off and on switch, that's it |
20:53.27 | fonzai | hi guys! I try to run asterisk on other machine than it was compiled on. But now I get "...Illegal instrucion" when trying to start it. that happens after successfully loading some of the modules. The computers have exactly the same kernel, but different CPUs. any ideas? TIA |
20:54.24 | reseaux | fonzai: different library... dirrent type distribution... i think binary is not good... |
20:54.51 | fonzai | reseaux: the libraries and the distribution are the same, as well |
20:55.09 | reseaux | the kernel is the same? |
20:55.23 | extremis | should I get dtmf on my T1 or DT or MF? |
20:55.25 | fonzai | reseaux: yes |
20:55.44 | reseaux | same hardware? |
20:56.07 | extremis | I need someone with a clue to answer, I have the telco on the phone |
20:56.08 | fonzai | reseaux: not exactly, different cpu etc. |
20:56.11 | extremis | and they are building the circuit |
20:56.23 | extremis | please |
20:56.45 | reseaux | extremis: I cant understand!please explain more.thz |
20:57.14 | fonzai | reseaux: I looked at Makefile, but it seems that it doesn't make any optimizations based on CPU type automaticly. or such.. |
20:57.49 | extremis | whats glare action? |
20:58.55 | groz | another dumb question guys, how much cpu horsepower is needed for a system to handle say 4 lines, and have no problems keeping up with everything, including compression etc ? |
20:59.12 | jets | we need an on,ine guestimator " |
20:59.32 | groz | i looked, but i did ask politely if dumb qs were ok :) |
20:59.41 | groz | couldn't find this stuff online |
20:59.45 | jets | "X t1's, X pri's, X SIP phones" "You will need about a .... xx processor, xx memory" |
21:00.01 | PBXtech | great idea jets |
21:00.14 | groz | yah, i'm just wondering if a epia board with the 533mhz fanless processor will be able to keep up |
21:00.38 | groz | i see a huge value to having a solid state fanless system |
21:00.56 | Exomorph | Ugs.... I'm having lots of problems with Asterisk hanging... |
21:00.57 | extremis | you won't be able to get 4 x100p's in there |
21:01.04 | extremis | irq sharing is a big problem |
21:01.25 | PBXtech | damn the stable release! |
21:01.46 | groz | even on the 4 port board ? |
21:02.04 | extremis | there isn't a 4port x100p |
21:02.13 | groz | no, i meant 4 phones connected |
21:02.16 | PBXtech | 12[4jets12]1: if you have the info I could put up a page to do that |
21:02.16 | groz | no external lines |
21:02.22 | groz | that goes out over data link |
21:02.26 | extremis | oh, helll I have 10 with a 450mhz |
21:02.39 | groz | ok cool |
21:03.27 | groz | cuz of my remote location, solid state is a HUGE benefit, it's in the wheelhouse of a vessel at sea |
21:04.02 | extremis | mmm saltwater |
21:04.11 | groz | yah i know, already got a dozen systems on the boat |
21:04.21 | c4uldr0n | I just started making my Asterisk box... I have my 2 x X100P installed (I have two incoming phone lines)... I'm waiting for my 3 SIP phones to come in a couple days but I want to have my * box ready when they arrive... Can somebody help with a X100P/SIP config |
21:04.28 | groz | saltwater is nothing compared to the problems of the antenna |
21:05.06 | c4uldr0n | if somebody can help, plz private msg me ;) |
21:06.48 | mishehu | hmm... Asterisk 10! hasn't yet been released... I'm disappointed |
21:06.49 | mishehu | heh |
21:07.10 | *** join/#asterisk j35 (~j23@runningawayfrom.avatara.org) |
21:08.55 | reseaux | I NEED HELP With "early B3 connect" on libPRI Thz |
21:09.27 | jets | PBXtech: I don't have that info ;) |
21:09.31 | jets | i have no idea |
21:10.36 | *** join/#asterisk jsharp (~jsharp@www.thegeekworks.com) |
21:10.49 | PBXtech | but it is a great idea :) |
21:11.00 | jsharp | Will setting nat=yes on a SIP peer do bad things if the phone really isn't behind nat? |
21:11.48 | mishehu | it will make your hair turn green |
21:11.50 | jets | jsharp: i don't think so |
21:12.14 | jsharp | That sure beats the grey it is right now. |
21:12.35 | outtolunc | he meant florenscent green |
21:12.51 | outtolunc | er -n |
21:13.09 | oej | jsharp: It means that Asterisk will check for the IP address of the SENDER instead of reading the SIP message. |
21:13.15 | j35 | so i'm trying to install zaptel, and i complie it and then install it and modprobe zaptel and get a BUNCH of unresovled symbol's.. any idea? |
21:13.24 | *** join/#asterisk lkc1 (~lkc@hoochie.digium.com) |
21:13.28 | oej | jsharp: So in most cases, it should not mean anything to turn NAT=on |
21:13.35 | jsharp | As long as its not phosphorescent green & doesn't glow in the dark. |
21:13.51 | mishehu | j35: you run depmod -a recently? |
21:13.58 | mishehu | sometimes taht fixes it. |
21:14.11 | jsharp | oej: Excellent. That's what I thought. I'm building sip.conf out of a database & don't wanna deal with users knowing if they're behind nat or not. |
21:14.12 | mishehu | jsharp: if your uptime is greater than 3 hours, it will |
21:14.15 | j35 | mishehu: it still gives the unresolved symbols in........ i'm using a stock debian kernel |
21:14.18 | j35 | should i build my own? |
21:14.26 | jsharp | I'm safe then, I just rebooted the asterisk box. |
21:14.33 | mishehu | j35: *shrug*, i use slack, and always build my own. |
21:14.35 | *** join/#asterisk w0ss (~w0ss@h00e01455ec48.ne.client2.attbi.com) |
21:14.55 | j35 | lol can't hurt to try |
21:15.02 | PBXtech | 4y5u6p |
21:15.04 | PBXtech | err |
21:15.12 | PBXtech | wrong winder |
21:15.18 | mishehu | wrong colors too |
21:15.24 | PBXtech | 1H2e3H4e5H6e |
21:15.24 | PBXtech | oops |
21:15.25 | PBXtech | hah |
21:15.27 | jets | Whats the best way to do contexts... say all of my phones are in the [corporate] context. |
21:15.43 | jets | and i want then to use long distance dialing say a context [longdistance] |
21:15.56 | jets | in the [longdistance] context do i want to include the corporate context |
21:15.57 | jets | or vice versa |
21:16.21 | mishehu | vice versa i think. |
21:16.37 | PBXtech | why |
21:16.39 | jets | but that means someone at the default s, menu can use the longdistance... |
21:16.48 | jets | because [longdistance is included] |
21:17.08 | PBXtech | what if you had a lobby phone |
21:17.13 | mishehu | jets: but the other way around, you might be putting [longdistance] into other contexts, and you don't want to include corporate in all other contexts |
21:17.40 | mishehu | jets: what is the "default s, menu" ? |
21:19.02 | *** part/#asterisk Buana (~thomasn@Gcb0a.g.pppool.de) |
21:19.09 | PBXtech | i would have the include in the corporate context |
21:19.20 | PBXtech | cause if you had a lobby phone you would just give it a local context |
21:19.29 | mishehu | or a [lobby] context |
21:19.31 | PBXtech | what about international calls? that open for everyone |
21:19.39 | mishehu | or a [whateverthefuckyouwant] context |
21:19.48 | PBXtech | goes against regular PBX methods thou |
21:20.12 | PBXtech | each phone has a restiction a restriction |
21:20.20 | *** join/#asterisk drgalaxy (~bduhan@adsl-66-140-100-19.dsl.lbcktx.swbell.net) |
21:20.51 | mishehu | and what are 'regular pbx methods' then? |
21:21.25 | jets | hrm the doc/SECURITY in the source does it a weird way |
21:21.25 | reseaux | I NEED HELP With "early B3 connect" on libPRI Thz |
21:21.33 | PBXtech | you assign a phone a restiction |
21:22.13 | mishehu | PBXtech: well, that's not exactly the same as going against an rfc. ;-) |
21:22.20 | jtodd | Can someone give me an example of a toll-free prefix in the UK? Here in the US, an example would be 1800xxxxxxx. |
21:23.26 | jtodd | Never mind - found a list. |
21:23.30 | PBXtech | guess what your used to |
21:23.35 | mishehu | hope it's an accurate list ;-) |
21:24.22 | jtodd | probably not. http://mirror.lcs.mit.edu/telecom-archives/archives/country.codes/toll.free.prefixes |
21:24.30 | jtodd | 0345 still valid? |
21:25.07 | zoa | ============= |
21:25.08 | zoa | B e l g i u m |
21:25.08 | zoa | ============= |
21:25.08 | zoa | <PROTECTED> |
21:25.08 | zoa | 11 xxxx |
21:25.08 | zoa | <PROTECTED> |
21:25.10 | zoa | that is so wrong |
21:25.15 | zoa | don't believe that list ! |
21:25.28 | jtodd | zoa: so give me an example of a toll free prefix in Belgium. I just need an example for a demo script. |
21:25.41 | mishehu | belgium is the most horrendous swear word in the galaxy |
21:25.45 | zoa | 08 |
21:25.52 | zoa | mishehu: why ? |
21:25.55 | jtodd | OK, that's fine. What's belgium's country code? |
21:25.58 | zoa | 32 |
21:26.01 | jtodd | thanks. |
21:26.07 | zoa | but that won't work internationally |
21:26.07 | [Sim] | *yawn* |
21:26.46 | *** join/#asterisk heison (~heison@ns1.somanetworks.com) |
21:26.49 | reseaux | There is someone that offer a big Voip Termination for 30000 Euro/month of call |
21:26.55 | Exomorph | Anyone having problems with Asterisk always sending calls to the fax extention? |
21:27.09 | zoa | reseaux what do you mean ? |
21:27.11 | *** part/#asterisk VoiceLynx (VoiceLynx@rrcs-central-24-106-64-175.biz.rr.com) |
21:27.16 | mishehu | zoa: you never read The Guide? |
21:27.24 | zoa | what guide ? |
21:27.32 | mishehu | ~theanswer zoa |
21:27.36 | | zoa: 42 |
21:27.39 | jtodd | zoa: what won't work internationally? the country code or dialing toll-free stuff from outside of Belgium? |
21:27.40 | reseaux | i have a company in italy specialized in Etnic traffic... |
21:27.43 | jets | okay |
21:27.45 | jets | i have a problem |
21:27.48 | jjanzer | heh |
21:27.54 | zoa | from outside of belgium |
21:27.58 | mishehu | zoa: The Hitchhiker's Guide to the Galaxy |
21:28.01 | zoa | aaah no |
21:28.04 | zoa | dont know that |
21:28.11 | jtodd | zoa: Maybe not for much longer. :-) You'll see in my note shortly to the list. |
21:28.15 | daork | does jbot know the question? |
21:28.20 | vaewynAFK | Exomorph: yes... see http://bugs.digium.com/bug_view_page.php?bug_id=0000696 and http://bugs.digium.com/bug_view_page.php?bug_id=0000649 |
21:28.28 | tclark | Exomorph: beaten to death on the ml, new dsp is fsckd, im makefile set it back to odd_dsp_routines |
21:28.29 | zoa | jtodd: i can dial to it from out of europe |
21:28.36 | zoa | but i still have to pay to it though :( |
21:28.55 | Exomorph | tclark: AHHHHHH! Thank-you! |
21:28.59 | jtodd | zoa: OK, so from Germany, as an example, you can dial 3208xxxxxx but you get charged for an international call, right? |
21:29.08 | zoa | no no no |
21:29.11 | zoa | from germany |
21:29.18 | zoa | you need to call 00328 prolly |
21:29.21 | Exomorph | vaewyn: And thanks as well. :) |
21:29.25 | zoa | but you can't call that number |
21:29.31 | zoa | 0800 numbers are for belgium only |
21:29.35 | zoa | they are blocked by the isp's |
21:29.37 | vaewyn | Exomorph: no prob :} |
21:30.21 | jtodd | zoa: So, what would you dial if you were in Belgium, and were dialing a hypothetical toll-free number? |
21:30.51 | zoa | 0800 |
21:31.15 | jtodd | zoa: Let's see a full number, if you have a "real" example. |
21:31.21 | zoa | 080060800 |
21:31.27 | jtodd | ok. |
21:31.46 | jtodd | so is 08 the prefix for toll-free or is 0800 the prefix for toll-free? |
21:31.51 | zoa | 0800 |
21:31.58 | jtodd | OK. |
21:32.07 | zoa | you could have 087 as an area code i think |
21:32.13 | zoa | i could look it up if you want |
21:32.25 | jtodd | zoa: not that important;I just wanted one valid example. |
21:32.44 | zoa | what are you up to ? |
21:32.55 | jtodd | Writing up ENUM examples for freenum.org |
21:33.06 | zoa | cool |
21:33.52 | jets | you shut up jjanzer |
21:34.02 | jets | when you do a [longdistance] -- and then include default |
21:34.03 | jjanzer | eh? |
21:34.12 | jets | oh nevermind |
21:34.15 | jets | nobody will be of any help |
21:34.29 | bkw_ | IDSKLFJLSDFIJSLEIF |
21:34.36 | bkw_ | I ready to shoot res_parking and res_musiconhold |
21:34.51 | bkw_ | why the fuck are they modules? if you can't unload them they should be static |
21:35.12 | mishehu | jets: why would you include [default] in [longdistance] ? |
21:39.57 | ManxPower | bkw_, because you could load a different one that exported the same symbols? |
21:40.03 | token | ANyone know is * can Fork SIP Calls |
21:40.12 | ManxPower | token, Define "fork" |
21:40.13 | denon | fork? |
21:40.20 | denon | you can conference em |
21:40.24 | ManxPower | Usually it's the SIP client that forks up the call, not Asterisk. |
21:40.33 | token | or Multiple users are allowed to register with the Same SIP Reg/Auth different phones |
21:40.48 | ManxPower | token, that's not supported. |
21:41.04 | denon | not entirely wise either |
21:41.06 | ManxPower | Since registration is for Asterisk routing a call to the SIP endpoint it doesn't make a lot of sense to support it. |
21:41.46 | token | Well what if you had A hpone in your office and you had a SIP usb phone in China |
21:42.03 | token | *hpone phone |
21:43.05 | ManxPower | Then deal with it with extensions.conf and Dial(SIP/office&SIP/china) |
21:43.37 | jorgeraidel | que manera de darme problemas el 0.7.0 jajaja |
21:43.38 | jorgeraidel | wow |
21:45.31 | *** join/#asterisk point (~litw@195.161.106.222) |
21:46.25 | [Sim] | ~seen kapejod |
21:46.28 | | kapejod <~kapejod@p509241B8.dip0.t-ipconnect.de> was last seen on IRC in channel #asterisk, 3h 22m 25s ago, saying: 'hi there'. |
21:49.12 | point | ~seen JerJer |
21:49.13 | | jerjer is currently on #asterisk. Has said a total of 25 messages. Is idling for 3h 39m 1s |
21:49.17 | point | :) |
21:49.46 | point | ~help |
21:53.43 | voidptr_ | you see |
21:53.44 | PBXtech | whats a cheep price for a 7960 |
21:53.49 | voidptr_ | eating and drinking is better |
21:53.54 | Corydon76-home | ~seen all-the-new-asterisk-sounds |
21:53.55 | | Corydon76-home: i haven't seen 'all-the-new-asterisk-sounds' |
21:53.58 | voidptr_ | i'm noyt completely drunk now |
21:54.11 | voidptr_ | just a biot |
21:54.27 | *** join/#asterisk clh (~me@216.253.86.210) |
21:56.00 | jtodd | yeah, yeah... .I have a real job that takes up 14 hours a day... the sounds are open on my desktop; alphabet and like 20 phrases done... more work this PM. |
21:56.28 | Corydon76-home | No pressure. ;-) |
21:56.33 | PBXtech | hate those kind of jobs |
21:56.36 | jtodd | not saying that othes don't have real jobs, but this one is a real doozy. |
21:56.58 | PBXtech | telecom jobs are boring |
21:56.59 | jtodd | and Ijust got handed two more full time jobs that I get to do, on top of the two I already had. |
21:57.05 | jrollyson | hello |
21:57.34 | PBXtech | hi |
22:00.04 | bkw_ | anthm I blew it up |
22:00.05 | bkw_ | haha |
22:00.06 | Sobek | Are there no power supplies shipped with an IAXy? |
22:00.19 | zoa | you have an iaxy ? |
22:01.31 | JerJer | iaxy iaxy iaxy! |
22:01.41 | jsharp | telecom jobs sure beat the hell out of "internet" jobs. |
22:01.52 | jtodd | jsharp: there is a difference? |
22:02.01 | PBXtech | packets suck |
22:02.03 | PBXtech | heh |
22:02.19 | groz | jtodd, yes there is, folks in telecom are still working |
22:02.36 | jtodd | groz: I suppose you consider VoIP telecom, eh? |
22:02.39 | *** join/#asterisk Tjardick (Tjardick@15-216.240.81.adsl.skynet.be) |
22:02.39 | PBXtech | and underpaid |
22:02.56 | groz | it's hybrid depending on installation i guess |
22:02.57 | PBXtech | except me |
22:03.08 | *** join/#asterisk omer (~omer@hoochie.digium.com) |
22:03.10 | tholo | On the other hand, some of us that were working in "internet" jobs just plain retired... |
22:03.29 | jsharp | jtodd: Yeah, at least in the telecom industry you don't have a bunch of people hiding in their mommy & daddy's basement claiming to be a "web engineer" cause they know how to use frontpage. |
22:05.36 | jtodd | jsharp: Yeah, you get a bunch of people hiding in their mommy * daddy's basement claiming to be a "telco" because they know how to use asterisk. LOL. |
22:05.54 | jsharp | Heh. True. |
22:06.46 | *** join/#asterisk digger_ (~digger@almestien.com) |
22:08.33 | tclark | jealous |
22:08.35 | bkw_ | har har har |
22:11.07 | JerJer | local config isssue, NEXT |
22:12.06 | bkw_ | tholo |
22:12.17 | bkw_ | this codec problem is a combo of fwd and the 7960 |
22:12.32 | bkw_ | the 7960 offers it so FWD sees that and forces it |
22:14.43 | bkw_ | tholo have you set the perfered code on the 7960 to ulaw or alaw? |
22:14.45 | bkw_ | does it happen? |
22:14.51 | *** join/#asterisk rocketman (~eedgar@c-66-41-182-23.mn.client2.attbi.com) |
22:16.41 | c4uldr0n | if I have 2 X100P's and my phones are all SIP, can I disable chan_modem.so in modules.conf? |
22:19.20 | *** join/#asterisk calvis (~calvis@hoochie.digium.com) |
22:19.42 | Tjardick | c4uldr0n, sure, you only need to modules for things you use ;) |
22:20.31 | Tjardick | but don't forget to disable the other modem modules two |
22:20.32 | Tjardick | too |
22:20.37 | Tjardick | as they depend on the main one |
22:22.06 | ManxPower | c4uldr0n, YES! |
22:22.14 | Corydon76-home | ~seen coil |
22:22.24 | | coil <0@user-69-1-15-56.knology.net> was last seen on IRC in channel #asterisk, 40d 16h 48m 21s ago, saying: 'no'. |
22:22.34 | Corydon76-home | ~seen mr |
22:22.35 | | mr <~fusi@pD953224A.dip.t-dialin.net> was last seen on IRC in channel #debian, 7h 9m 32s ago, saying: 'thanks i try it'. |
22:23.15 | FuzzyCat | fekcing reload fucked again |
22:23.36 | *** join/#asterisk easydone (~easydone@eksel.demon.nl) |
22:24.06 | Sobek | LOL I just read the iax song |
22:24.50 | voidptr_ | oh man |
22:24.54 | Connor | okay.. I really need a SIP--H323 converter or something.. I MUST get Asterisk talking to CCM |
22:25.03 | dnc | hey FuzzyCat, whats up with nl.nixhelp.org |
22:25.20 | c4uldr0n | I just ran "make samples" so I'm getting rid of the unneeded items right now |
22:25.24 | voidptr_ | hey dnc |
22:25.25 | jets | what is the function of ignorepat |
22:25.32 | jets | i don't really understand it i suppose |
22:25.35 | mbranca | Connor use * with chan_h323 |
22:25.45 | jets | oh nevermind |
22:25.45 | voidptr_ | jets: when you press that digit it keeps dialtone |
22:25.46 | jets | yes i do |
22:25.54 | voidptr_ | iirc |
22:26.00 | Connor | mbranca: chan_h323 won't work with CCM |
22:26.01 | JerJer | don't run make samples then |
22:26.13 | easydone | wasim: http://farfon.convergence.com.pk/ is not working..... |
22:26.24 | c4uldr0n | heh, doesn't that just move the conf files into the /etc/asterisk folder? |
22:26.24 | voidptr_ | : idle : 178 hours 46 mins 32 secs (signon: Tue Jan 6 07:25:48 2004) |
22:26.33 | voidptr_ | i dont think wasim will respond easydone ;) |
22:26.42 | easydone | why not? |
22:26.57 | voidptr_ | he hasn't said anything for 178 hours |
22:27.23 | lorena26 | jerjer, for what when i call to cisco in g729, give me error of codec 19 ? |
22:27.30 | voidptr_ | he's quickly approaching the top idlers |
22:27.30 | easydone | voidptr: meditating? |
22:27.34 | lorena26 | Jan 15 21:30:01 NOTICE[1267154224]: rtp.c:418 ast_rtp_read: Unknown RTP codec 19 received |
22:27.40 | voidptr_ | i guess ;) |
22:27.45 | JerJer | lorena26: no cng |
22:28.01 | ManxPower | lorena26, search the mailing list archives |
22:28.38 | lorena26 | I used g729r8 in cisco with No VAD |
22:28.51 | mbranca | Connor, sorry, I'm not a ccm expert... what protos ccm speak, so? |
22:29.31 | Connor | mbranca: h323, sccp and mgcp I think... but sccp and mgcp are for phones only.. not trunking.. which is what I'm needing |
22:30.42 | bkw_ | CUSOTMERS SUCKS |
22:30.50 | PBXtech | yes they do |
22:30.52 | doughecka | lol |
22:31.29 | PBXtech | if they are cute ill let them really suck |
22:31.32 | voidptr_ | i'm not complaining if they pay good |
22:31.33 | voidptr_ | but they dont |
22:31.34 | voidptr_ | :P |
22:31.39 | PBXtech | errr did I say that out loud? |
22:31.59 | mbranca | Connor so why chan_h323 won't work with ccm? |
22:32.43 | PBXtech | if they are complaining they get a higher percentage |
22:32.44 | Connor | mbranca: Ask JerJer. :) It won't hand off the RTP session to the phone. |
22:32.56 | token | BKW is * 0.7.0 stable now |
22:33.01 | token | any notable problems |
22:33.12 | PBXtech | 0.7.1.4.7 is good |
22:33.30 | token | will I get that from default CVS |
22:33.34 | dnc | hey voidptr, hows life? |
22:33.35 | token | :) |
22:33.38 | PBXtech | end of the day silly |
22:33.49 | dnc | i played with a motorola canopy 5ghz antenna today |
22:33.55 | dnc | it was lovely |
22:34.01 | PBXtech | lovely |
22:34.07 | token | Seriously is 0.7.0 Stable!! |
22:34.20 | PBXtech | heard lots of problems dog |
22:34.23 | JerJer | nobody said 0.7.0 was stable |
22:34.31 | rocketman | can anyone tell me a good linux softphone? |
22:34.33 | pros12 | is anyone running asterisk with pstn in montreal? |
22:34.43 | FuzzyCat | dnc |
22:34.47 | token | ok! that is why the question came up I guess |
22:34.47 | FuzzyCat | nothing... |
22:35.00 | dant | CVS still closed? |
22:35.12 | voidptr_ | dnc: ok... it was free beer again this evening |
22:35.45 | Connor | OKay.. someone explain the gk funtion of h323 in chan_h323 |
22:36.30 | mbranca | use it for register aliases to a gk |
22:36.40 | jjanzer | rocketman, kphone is decent |
22:37.07 | rocketman | ok I will look into that |
22:37.12 | Connor | so, is ccm considered a gk ? |
22:37.19 | mbranca | so you can tell to a gk to handle (for example) all number beginning with '2' by passing them to *, in the context specifed in h323.conf |
22:37.45 | *** join/#asterisk Cinzas (~AsHeS@gil.di.uminho.pt) |
22:40.26 | mbranca | nite all |
22:40.54 | jrollyson | "If SCO thought threat-born licensing fees would provide a quick boost to the bottom line, it looks to have miscalculated. As fellow Fool Tom Taulli noted last month, SCO has tried to cast its lawsuits in apocalyptic terms. But with the entire computing world putting its money behind Linux, it appears that, for SCO, the apocalypse is now." |
22:41.51 | c4uldr0n | is there a good tutorial for configuring Asterisk? I'd rather read than bug the hell out of everybody here ;) |
22:42.43 | Cinzas | the best tutorial that i found ..... google |
22:42.48 | Cinzas | :( |
22:43.56 | c4uldr0n | duh ;) |
22:44.04 | Cinzas | really |
22:44.59 | dnc | c4uldr0n: http://www.automated.it/guidetoasterisk.htm |
22:45.32 | c4uldr0n | thanks |
22:46.24 | groz | thanks dnc |
22:46.29 | groz | that's a nice starting point |
22:46.39 | groz | <--- just starting to do first configure after building |
22:47.11 | c4uldr0n | I don't suppose the X100P can dial-out as well as receive phone calls ;) |
22:47.16 | *** join/#asterisk draconius (~j00momma@12-218-60-36.client.mchsi.com) |
22:47.21 | c4uldr0n | sorry for the lame ? |
22:47.27 | PBXtech | it can |
22:47.31 | c4uldr0n | k, whew |
22:47.35 | PBXtech | heh |
22:48.14 | c4uldr0n | I was hoping to get this up and running in one day, it has so many features (Asterisk) that it's hard for me to learn it all in one day |
22:48.33 | PBXtech | try 3 months |
22:49.13 | c4uldr0n | I only have 2 phone lines... I just want to receive calls and have extentions, transfer phone calls between 3 SIP phones in the office |
22:49.22 | c4uldr0n | you would think, 1 day project ;) |
22:49.26 | c4uldr0n | but I'm thinking more |
22:49.34 | *** join/#asterisk FryGuy (~fryguy@c-24-2-50-122.client.comcast.net) |
22:50.57 | Cinzas | :) |
22:51.40 | Cinzas | TOday i've tested * with gatekeeper (h323), ata (h323) anda ata(sip), anda i4l (isdn) |
22:51.53 | Cinzas | Working fine ! Rox |
22:52.06 | Cinzas | anda = and ! |
22:53.06 | PBXtech | i cringe when people say h323 |
22:53.11 | denon | h.323 |
22:53.13 | bkw_ | me too |
22:53.13 | denon | mwahahaha |
22:53.24 | PBXtech | 1L2a3u4g5h6i7n8g 9O11u11t 12L13o14u19d |
22:53.31 | PBXtech | woooo pretty |
22:53.46 | Cinzas | same thing |
22:53.47 | denon | I live if only to annoy bkw |
22:54.08 | denon | so PBXtech, you gonna kick in some $$ on my bounty? |
22:54.32 | PBXtech | I cringe when people say BSD |
22:54.46 | denon | what's wrong with FBSD and OBSD? |
22:54.54 | PBXtech | whats wrong with linux |
22:55.01 | denon | dont get me started |
22:55.06 | Cinzas | whats wrong with u ? |
22:55.21 | PBXtech | umm dont get me started |
22:55.47 | token | anyone know how to get dial by name directory to work? |
22:55.48 | denon | don't worry, I'm not even gonna LET ya get started.. |
22:55.49 | denon | :) |
22:55.53 | token | Is it at all possible? |
22:56.17 | crontibs | denon the bounty starting to look good |
22:56.37 | token | Anyone Know? |
22:56.47 | *** join/#asterisk haighis (~chatzilla@216.58.40.2) |
22:56.53 | token | Anyone? Bueller? Bueller? |
22:57.22 | *** join/#asterisk zwi (~zwi@216.88.131.43) |
22:57.23 | jrollyson | kram: good timing |
22:57.38 | jrollyson | kram: can you look at bug #853 |
22:57.45 | denon | crontibs: yeah? I havnet looked today |
22:57.46 | PBXtech | and #823 |
22:58.05 | denon | crontibs: oh, nothin new .. |
22:58.09 | haighis | bkw_, regarding callerid/canada/stentor switch i tried the same x100p at a different location..in Burlington...still getting now callerid coming through.. i wanted to try on a different switch to see if i get the same errors.. |
22:58.13 | PBXtech | whats the bounty up to now anyway? |
22:58.22 | jrollyson | PBXtech: 823 is already fixed ;) |
22:58.23 | PBXtech | I cringe when people say canada |
22:58.28 | denon | PBXtech: $350 |
22:58.33 | PBXtech | just wanted him to look at it |
22:58.36 | PBXtech | :) |
22:58.42 | PBXtech | though it was like 800 ish |
22:58.45 | haighis | PBXtech, you love canada |
22:59.03 | PBXtech | nuke em |
22:59.07 | c4uldr0n | anybody use MSN to test sip? |
22:59.13 | haighis | we bring lots of good things...like a the Toronto Seal hunt.. |
22:59.28 | PBXtech | cant buy drugs there anymore so nuke me |
22:59.30 | PBXtech | em |
22:59.33 | PBXtech | heh |
22:59.49 | token | Anyone Know if directory sservice dial by name is available on asterisk? |
22:59.54 | haighis | did you hear about the big bust in Barrie..in the molson brewery? |
23:00.13 | haighis | good stuff..... |
23:00.17 | PBXtech | we went to that big old waterfall a year ago |
23:00.20 | token | Anyone? |
23:00.22 | voidptr_ | haggis? |
23:00.23 | voidptr_ | :) |
23:00.23 | haighis | leave some skin for the kids.. |
23:00.43 | token | He puts the lotion on the skin??? |
23:00.53 | haighis | yeah...haighis... |
23:00.58 | haighis | do you know a haighis? |
23:01.13 | PBXtech | 12[4token12]1: I thought you could. never tried it myself |
23:01.22 | zwi | can anybody tell me the exten pattern I need if I want to just direcly dial (10 digits) to an outside line? |
23:01.40 | PBXtech | nxxnxxxxxx |
23:01.57 | token | n= Digits 2-9 x= digits 0-9 |
23:02.06 | *** join/#asterisk johnny_ (~johnny@adsl-63-202-210-220.dsl.snfc21.pacbell.net) |
23:02.16 | zwi | Oh...and here I thought it was much harder than that thanks! |
23:02.17 | haighis | has problem with all calls coming in as fax been fixed or acknowledged as a problem? |
23:02.19 | johnny_ | hi all, i have some dumb questions if ya all wouldnt mind |
23:02.21 | token | PBXTech you know where I can get a working config for Dial directory |
23:02.28 | token | <PROTECTED> |
23:02.36 | token | heading out to sea |
23:03.01 | haighis | token: xten => 411,1,Directory,default |
23:03.12 | haighis | sorry that exten => 411,1,Directory,default |
23:03.19 | PBXtech | http://www.bkw.org/~brian/asterisk-conf/ |
23:03.23 | PBXtech | he has a config |
23:03.44 | haighis | so no thoughts on the callerid? |
23:03.49 | token | can I make # Dial by directory |
23:03.55 | haighis | ?? |
23:03.56 | PBXtech | oh dial by name config.. umm |
23:04.07 | *** join/#asterisk pchitescu (~r00t@home-25022.b.astral.ro) |
23:04.22 | haighis | token, use 411 or some other number |
23:04.38 | c4uldr0n | do I have to do anything special to Asterisk to allow MSN login via SIP for testing purposes? |
23:04.47 | token | I don't mean directory service though |
23:04.58 | haighis | what do you mean then? |
23:05.02 | PBXtech | ut oh a root user hack HIM! |
23:05.15 | token | my 411 no calls out to directory assistance |
23:05.29 | haighis | okay..then make it 511 or *411 |
23:05.31 | johnny_ | I need to have a VoIP solution between remote sites and am I correct in saying that if we get a t1 pri, we would need one Wildcard TE410P w/ asterix? do people ever use asterix w/ cisco routers(sorry for the dumb question) |
23:05.56 | johnny_ | and we just configure the amount of B-channels in the conf? |
23:06.05 | jets | johnny_: correct |
23:06.07 | voidptr_ | t100p should be ok too |
23:06.16 | jets | and yes asterisk can use mgcp |
23:06.34 | voidptr_ | te410p also allows use of channel banks |
23:06.51 | johnny_ | ok, cool..im just really wondering if it will magically work with the PSTN, or how it determines data from voice...? can someone shoot me a relavent FAQ link? im going through as many pages as possible |
23:06.56 | PBXtech | 12[4token12]1: try this exten => *,1,Directory,default |
23:06.59 | ManxPower | Does anyone have any ideas of how to do this: Transfer a call to the parking extension, the parking application reads back the extension the call is parked on, then plays a beep, then connects you to another extension (overhead paging). I can do all of that except automatically connect to the overhead paging. Currently users have to hang up and then dial the paging extension |
23:07.26 | johnny_ | im somewhat confused still about channel banks, all i know is that Direct-ID and caller ID are critical |
23:08.01 | token | PBXtech let me try thanks |
23:08.16 | PBXtech | thats what i had in my config. |
23:09.02 | token | where should I put it under greeting message you think? |
23:09.24 | PBXtech | that was under my autoattendand press * to hear company directory |
23:09.55 | token | PBXtech let me try thanks |
23:10.02 | *** part/#asterisk haighis (~chatzilla@216.58.40.2) |
23:11.11 | JerJer | Openh323 is not vulnerable. This is the old SNMP asn1 stack vulnerability. |
23:11.11 | JerJer | Some vendors might have used this in implementing h323 stacks, but |
23:11.11 | JerJer | openh323 created its only asn1 code, which to my knowledge, does not have |
23:11.11 | JerJer | any such vulnerability. h323 software that is based on openh323 should |
23:11.11 | JerJer | not have any problems related to this CERT. |
23:11.42 | JerJer | s unfortunate that the one has to dig so deep to find out that it was |
23:11.42 | JerJer | a particular ASN1 stack that has the problem, and not h323 in general. |
23:11.45 | denon | JerJer: they claim SIP/etc has the same issues |
23:12.01 | JerJer | Someone should probably contact CERT to explain the difference between |
23:12.01 | JerJer | ASN1 implmenetations and those things that are implemented with ASN1. |
23:12.01 | JerJer | This CERT advisory is like finding a flaw in Linux rpc.statd UDP, and then |
23:12.01 | JerJer | reporting that anyone using IP is vulnerable. |
23:12.19 | voidptr_ | :)) |
23:12.21 | JerJer | -- Dean Anderson, a major Open H.323 contributor |
23:14.06 | Mike | JerJer: alive? |
23:16.05 | JerJer | nope |
23:16.15 | Mike | JerJer: i just want your price list |
23:16.16 | *** join/#asterisk wreckdiver2 (~goaway@216.82.109.5) |
23:18.02 | denon | Mike: give him $5k and he'll do anything |
23:18.09 | denon | couldnt get any simpler than that |
23:18.22 | JerJer | yep |
23:18.29 | Mike | bah |
23:18.42 | JerJer | send an email to sales@nufone.net requesting our rate table, if that's what ur lookin for |
23:18.46 | JerJer | dinner time |
23:20.14 | *** join/#asterisk jsharp (~jsharp@www.thegeekworks.com) |
23:20.27 | jsharp | BORK BORK BORK. |
23:20.38 | Connor | How good is a 7905 phone? |
23:21.44 | *** mode/#asterisk [-o denon] by ChanServ |
23:21.53 | *** join/#asterisk zoa (~john@kn-upc-1.kuleuven.net) |
23:22.24 | jsharp | Has Digium/VoiceAge worked out the stupidness of the G729 codec registry on SCSI devices? |
23:22.26 | rozo | Conner: it works well. Only thing I don't like is the speaker phone (or lack of the speaker phone). |
23:22.34 | zoa | jsharp: it works on scsi !!! |
23:22.38 | jrollyson | kram: bug 853 |
23:22.39 | Connor | whats the diff with it and the 7912? |
23:22.39 | rozo | it does? |
23:22.43 | zoa | i use it on scsi already for a very long time |
23:22.57 | jsharp | I've heard mixed reviews. I couldn't get it to run on SCSI myself. |
23:23.02 | rozo | when we run the register program on a machine with SCSI, it locks up the machine and requires a hard reboot. |
23:23.09 | zoa | funky :) |
23:23.10 | jsharp | That's what I ran across. |
23:23.12 | kram | jsharp: i think there may be a workaround |
23:23.19 | kram | you'll have to talk to martin |
23:23.22 | JerJer | news at 11 |
23:23.27 | kram | but the short answer is that voiceage refuses to fix their bug |
23:23.30 | zoa | what will martin say ? |
23:23.35 | JerJer | hehe |
23:23.35 | jsharp | I'm shocked and surprised. |
23:23.44 | zoa | martin = digium martin ? |
23:23.48 | kram | yah |
23:24.04 | jsharp | Okee. martin it is. |
23:24.53 | dext0r | where is struct ast_channel defined? |
23:24.55 | johnny_ | one more dumb question..if i have 1 T100P and a full t1, I can use that card for both data for our lan and voip to the pstn? |
23:24.57 | rozo | martin was the one that got working for us but he has to register the codec himself. |
23:25.06 | Connor | can someone tell me what the diff is between a cisco 7905g and 6912g ? |
23:25.09 | zoa | how do you mean ? |
23:25.10 | Connor | they look the same... |
23:25.23 | zoa | how did he do that ? |
23:28.23 | *** join/#asterisk s3gal (~leon@cuscon4743.tstt.net.tt) |
23:28.48 | *** join/#asterisk jsharp (~jsharp@www.thegeekworks.com) |
23:29.00 | *** join/#asterisk haighis (~chatzilla@216.58.40.2) |
23:29.26 | dalabera | connor > http://cisco.com/en/US/products/hw/phones/ps379/products_qanda_item09186a00801739d1.shtml |
23:29.42 | *** join/#asterisk guibert (~guibert@hoochie.digium.com) |
23:31.02 | *** part/#asterisk mmco (~irc@pD9E10B6E.dip.t-dialin.net) |
23:31.37 | guibert | hi, I have a TDM400P with 4 FXS ports. On the other side 4 FXS ports given by my provider. Is there a way I can connect one port from the board to the ports of my provider to place calls through there? |
23:31.42 | johnny_ | asterix does not support LDAP, but CCM does, right? |
23:31.46 | *** join/#asterisk scud (~scud@12-219-155-152.client.mchsi.com) |
23:32.06 | Connor | dalabera: Ethernet switch is the diff.. okay.. |
23:32.19 | Tangent | guibert: Isn't that whay FXO cards are for? |
23:33.18 | c4uldr0n | when are they going to make a 4 port FXO? |
23:35.23 | Tangent | c4uldr0n: There's not much point making one when there's BRI and PRI cards already available |
23:36.11 | c4uldr0n | heh, sorry, but what are BRI and PRI? |
23:36.19 | *** join/#asterisk blitzrate_sk00l (blitzrage@CPE0080c6f83fe1-CM.cpe.net.cable.rogers.com) |
23:36.28 | jsharp | Except there's no cheap BRI cards that play well with US National ISDN U interfaces and PRI is damn expensive unless you've got more than 6 or 8 lines. |
23:36.29 | Tangent | Basic Rate ISDN (2 channels) and Primary Rate ISDN (30 channels) |
23:37.04 | *** join/#asterisk simprix (~simprix@24.247.181.124.gha.mi.chartermi.net) |
23:37.04 | c4uldr0n | k |
23:37.04 | Tangent | Hi Simon_ca |
23:37.06 | Tangent | Hi simprix |
23:37.12 | Tangent | Simon_ca: Not you.. sorry |
23:37.20 | simprix | Im kinda new to asterisk I can't call between extensions i get a 404 error but i can make outbound calls over my pstn line |
23:38.14 | c4uldr0n | I just got asterisk working with x-lite, hehe, this is a huge advance for me ;) |
23:38.43 | c4uldr0n | now I guess I have to configure the rest of asterisk so I can hook it up to my phone lines |
23:39.00 | c4uldr0n | w00t |
23:39.06 | guibert | yes, I should have FXO cards but i thought perhaps I can trick this someway. Perhaps using a small pbx as a bridge. Any ideas? |
23:39.14 | Tangent | c4uldr0n: You've leapt the first hurdle.. it gets easier from now |
23:39.25 | c4uldr0n | hehe, cool |
23:39.31 | c4uldr0n | I'm just pissed my phones aren't here yet |
23:39.40 | c4uldr0n | I ordered them Sunday, they shipped them yesterday |
23:39.57 | Tangent | c4uldr0n: That's not overnight delivery |
23:40.13 | c4uldr0n | good thing I found X-lite so I can test and configure until they get here ;) |
23:40.27 | Tangent | c4uldr0n: I installed xlite yesterday :) |
23:40.29 | *** join/#asterisk guibert1 (~guibert@hoochie.digium.com) |
23:40.32 | Tangent | Hadn't heard of it before then |
23:40.41 | c4uldr0n | it's pretty neat |
23:40.50 | Tangent | Yep.. nicer than SJPhone |
23:41.00 | *** join/#asterisk pointer (pointer@aj.catt.com) |
23:41.02 | c4uldr0n | haven't tried anything else, so i wouldn't know |
23:41.10 | c4uldr0n | all I know is that MSN didn't connect properly with SIP |
23:41.14 | c4uldr0n | so I looked further |
23:41.18 | c4uldr0n | and found X-lite |
23:41.22 | derrick | xlite works under wine too fyi |
23:41.32 | Tangent | even better :) |
23:41.34 | *** join/#asterisk {^DaNi^} (~dani@hoochie.digium.com) |
23:41.40 | c4uldr0n | wine rules ;) |
23:41.44 | zoa | derrick: is it good ? |
23:41.51 | zoa | not too much overhead / delay ? |
23:41.51 | c4uldr0n | I should see if Quickbooks works under wine, then I could get rid of Windows ;) |
23:41.53 | derrick | zoa, i didn't use it extensively, just for some testing |
23:42.03 | derrick | zoa, nope, not that i noticed but i keep beefy workstations |
23:42.05 | zoa | i am thinking of using it bugtime |
23:43.01 | *** join/#asterisk {^DaNi^} (~dani@atm0-1.r1-dcp.d1n.net) |
23:45.07 | zoa | everybody go look at the latest bug report |
23:45.13 | zoa | and go add what you'd like to see documented |
23:45.23 | zoa | lets give the documentation project some work ! |
23:45.26 | crontibs | wouldn't this be good error sound for * |
23:45.26 | crontibs | Operator: The fingers you have used to dial are too fat. To obtain a |
23:45.27 | crontibs | <PROTECTED> |
23:45.27 | crontibs | <PROTECTED> |
23:45.37 | crontibs | simpsons ref |
23:45.37 | zoa | yes i'm sure |
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23:47.14 | *** join/#asterisk yaboo (~jsirucka@203-213-113-146-vic.tpgi.com.au) |
23:47.46 | heison | anyone see this NOTICE? |
23:47.51 | dant | anyone fancy helping me test how many people I can fit in a meetme? |
23:47.52 | heison | Jan 15 18:44:04 NOTICE[1225991360]: chan_zap.c:3587 zt_read: Fax detected, but no fax extension |
23:48.21 | Tangent | dant: What do I have to do? |
23:48.41 | zoa | heison: a lot of people |
23:48.47 | zoa | its a little issue with new DSP |
23:48.48 | dant | Dial(IAX2/guest@195.102.146.5/7777) |
23:48.55 | dant | should do it |
23:49.18 | heison | is this supposed to fix the WARNING: http://lists.digium.com/pipermail/asterisk-users/2003-December/030592.html |
23:49.28 | zoa | revert to old dsp in the makefile |
23:49.30 | Tangent | dant: In that case.. no.. I haven't worked out IAX yet or allowed it through my firewall |
23:49.31 | zoa | is all you need to do |
23:49.43 | heison | dant: #define CONF_SIZE 160 |
23:49.51 | Connor | Are snome 200's fully functional in asterisk? I.E. Multiline support, attended and blind transfer, PARKING, and speed dial etc? |
23:50.17 | dant | heison, it's more the size of the box and the quality of the line I want to test :) |
23:50.46 | Connor | I'm putting together a 10-12 phone system and need some cost effective phones.. |
23:51.27 | heison | dant: depends how you gonna test this... 160 ppl via IAX will fill up a T1 pretty quickly. |
23:52.03 | c4uldr0n | context=sip ; Default for incoming calls |
23:52.03 | c4uldr0n | allow=ulaw |
23:52.03 | c4uldr0n | allow=alaw |
23:52.03 | c4uldr0n | allow=gsm |
23:52.03 | c4uldr0n | allow=ilbc |
23:52.10 | c4uldr0n | crap, wrong button |
23:52.12 | c4uldr0n | fux0r |
23:52.18 | c4uldr0n | sorry guys |
23:53.08 | dant | well, anyone else want to try and kill this p3 733? |
23:53.29 | {^DaNi^} | Hi people, I want deploy * as pbx in one of our offices. 150 cisco voip phone. 2 * box. |
23:53.49 | {^DaNi^} | And iax trunk with 2 * box located in our main office with PRI |
23:54.07 | jets | dant: how many are in this meetme |
23:54.10 | {^DaNi^} | any help about the hardware for this purpouse |
23:54.33 | heison | where can i find documentation for meetme? I wanna use the PIN feature, but I can't seem to find up to date docs |
23:54.36 | dant | jet's 3 so far :) |
23:54.45 | {^DaNi^} | in this moment i am thinking in use digium hardware for cards and supermicro as servers |
23:54.51 | {^DaNi^} | with debian as distro |
23:55.32 | rob-- | <PROTECTED> |
23:56.09 | zoa | dani: how much simultaneous calls + what codecs |
23:56.12 | zoa | is you only concern |
23:56.33 | zoa | give us an exact layout |
23:56.37 | zoa | and we will try to help you |
23:56.41 | {^DaNi^} | zoa 30 calls as peak with g729 |
23:56.49 | dant | nice |
23:56.53 | jets | rob--:i'm in the meet me |
23:56.54 | zoa | k, and those 150 cisco's |
23:56.55 | jets | wtf is that |
23:57.00 | zoa | why 2 asterisk servers ? |
23:57.10 | zoa | why two times two ? |
23:57.13 | zoa | redundancy ? |
23:57.15 | {^DaNi^} | zoa i need redundancy in all of our proyects |
23:57.37 | {^DaNi^} | i work for a bank if they dont see two box |
23:57.40 | zoa | k, so i suppose you also have 2 pri's ? |
23:57.59 | {^DaNi^} | in our main office we will have more pri |
23:58.05 | zoa | k |
23:58.09 | {^DaNi^} | this first office is only a test |
23:58.19 | {^DaNi^} | if work we will deploy asterisk in more offices |
23:58.19 | zoa | thought about how handling the redundancy of the incoming pri lines btw ? |
23:58.29 | zoa | i think you need a single xeon |
23:58.33 | zoa | for 30 times g729 |
23:59.04 | zoa | although i have little information on the cpu overhead for trunking |
23:59.07 | zoa | as there seems to be some |
23:59.10 | _aggelos_ | Dani: do you want to run the ip phones instead of phone cables ? evertying via ip ? |
23:59.32 | zoa | i only have 10 shitty g729 licenses |
23:59.40 | {^DaNi^} | _aggelos_ yes, all ip |
23:59.55 | *** join/#asterisk dant (~dan@81-86-69-213.dsl.pipex.com) |