| 00:00.09 | Mavvie | if it is an application error, a sniffer will show you that everything on the network layer is fine. |
| 00:00.20 | voxlinx | think I may just grab the most recent -STABLE and try again. |
| 00:02.26 | voxlinx | or I could just turn off externip in sip.conf. |
| 00:02.43 | voxlinx | that solves the problem nicely. |
| 00:03.24 | voxlinx | I don't recall changing that since the last time everything worked. |
| 00:04.36 | seva | so when you guys setup a brand new asterisk setup |
| 00:04.40 | seva | you start with the sample config |
| 00:04.46 | seva | and then slowly disable everything you don't want? |
| 00:04.56 | seva | (i just got all my test stuff working btw, thanks) |
| 00:05.10 | seva | it just seems like a huge pain to start from the full sample config |
| 00:05.41 | Mavvie | seva: normally I know what is needed, so I start from nothing and add what is needed. |
| 00:06.55 | seva | Mavvie: how do you handle the modules? do you let it autoload everything or do you just not install every module in the distribution? |
| 00:07.01 | seva | or do you list every module you need |
| 00:07.04 | Mavvie | just load everything |
| 00:07.17 | seva | i see |
| 00:08.59 | seva | alright, thanks for the help |
| 00:08.59 | *** part/#asterisk seva (seva@sevatech.com) |
| 00:09.40 | *** join/#asterisk goodnewscd (~goodnewsc@S01060000e8953d28.cg.shawcable.net) |
| 00:10.02 | goodnewscd | Hi!! |
| 00:11.12 | ariel_ | anyone here know what this error is with my new codec g729 installation? 0/0 encoders/decoders of 2 licensed channels are currently in use |
| 00:11.39 | ariel_ | I have no calls in place. |
| 00:12.00 | Mavvie | ariel_: read the message again. |
| 00:12.44 | *** join/#asterisk jetscreamer (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
| 00:13.47 | ManxPwr | ariel_, You are using 0 licenses, out of 2 licenses available |
| 00:13.59 | ariel_ | Mavvie, sorry I did not finish it says I have 2 lisc available. |
| 00:14.09 | Mavvie | ariel_: yes. |
| 00:15.08 | Mavvie | ariel_: you still haven't told an error :-) |
| 00:16.16 | ariel_ | Feb 14 16:14:38 NOTICE[21155]: chan_sip.c:2773 process_sdp: No compatible codecs! |
| 00:16.16 | ariel_ | Mavvie, my connection to the server keep dropping off sorry. |
| 00:17.10 | ariel_ | I am using an sipura and I have put there setup as disallow=all and allow=g729 |
| 00:17.14 | Essobi | MAhahaha |
| 00:17.19 | Essobi | http://funroll-loops.org/ Anyone been there? |
| 00:17.52 | Mavvie | ariel_: and is the sipura configured to use g729? |
| 00:18.55 | wolfson | thats a quite humerous site, read it a while back |
| 00:19.07 | ariel_ | Mavvie, yes it is. |
| 00:22.00 | machinehd | in zaptel.conf I have fxsks=1-4, but I only have 3 pstn lines plugged in. Should it be set to fxsks=1-3 and channel => 1-3 in zapata.conf ? |
| 00:24.40 | Mavvie | ariel_: run "sip show peer" on it and see which codecs are there. |
| 00:25.02 | Mavvie | machinehd: zaptel.conf is the definition of the hardware, not the actuall state of hardware. |
| 00:25.22 | ariel_ | Mavvie, ok thanks go a seg fault on the server trying to get it back up. |
| 00:25.35 | *** part/#asterisk PTG123 (~PTG123@ip68-106-19-249.ph.ph.cox.net) |
| 00:27.50 | machinehd | Mavvie, ok thanks, so stick with 1-4 in zaptel. However, do I need to change to 1-3 in zapata.conf? |
| 00:28.24 | Mavvie | I personally wouldn't do it. |
| 00:29.09 | *** join/#asterisk Nukemizer (~Nuke@65.103.231.133) |
| 00:29.13 | machinehd | Well seeing how there's only 3 lines how do I prevent it from trying to use the fourth? |
| 00:32.07 | Nukemizer | can anybody help me with a module loading problem ? Am trying to use /etc/rc.3/S09zaptel start to load my wcte11xp module but i get the following error |
| 00:32.07 | Nukemizer | Running ztcfg: ZT_SPANCONFIG failed on span 1: No such device or address (6) |
| 00:32.58 | Nukemizer | I think i only need Zaptel and wcte11xp to load for my PRI to work. Would that be wrong ? |
| 00:33.26 | Jlau515 | nukemizer: what happens when you type modprobe wcte11xp |
| 00:33.56 | Nukemizer | it loads |
| 00:34.16 | Nukemizer | but the pri does not work. I get sync errors from the PBX |
| 00:34.40 | Nukemizer | so I wanted to start from the begining and this might be related i thought |
| 00:35.10 | Jlau515 | i get the same error, when using the init script, still trying to figure out why |
| 00:35.27 | Jlau515 | my problem is kudzu keeps thinking the digium card is removed |
| 00:35.33 | *** join/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
| 00:35.42 | Nukemizer | actually let me recant that... |
| 00:35.52 | Nukemizer | I have to modprobe twice to get it to load |
| 00:36.15 | Nukemizer | first modprobe give me this -- ZT_SPANCONFIG failed on span 1: No such device or address (6) |
| 00:36.54 | ctooley | I know there has got to be a way to get Polycom SIP phones to register from behind a NAT firewall |
| 00:37.42 | *** join/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu) |
| 00:40.09 | mountainm2k | Asterisk@Home --> out-of-box setup... Added SIP X.210, and logged into it with X-Lite. I can dial, but when I hit *411 I can't hear anything |
| 00:40.20 | *** join/#asterisk Darwin35 (~Darin@c-24-3-241-22.client.comcast.net) |
| 00:40.23 | mountainm2k | or *98, either, for that matter... |
| 00:40.58 | *** join/#asterisk ZeroXeal (~zeroxeal@ool-44c166d7.dyn.optonline.net) |
| 00:44.15 | Silik0n | w00t i loe finding bidness documents I can snag and rework |
| 00:45.14 | TekLexus | hey, anyone have problems with busy signals after a call ends? When someone calls my box, and they hang up, it keeps a busy signal going, keeping the line off hook and leaving me a 2 hr long busy signal voicemail. |
| 00:46.43 | file[laptop] | analog? X100P or TDM400? |
| 00:50.46 | wolfson | teklexus: you likely need disconnect supervision on the line |
| 00:51.09 | TekLexus | TDM400 |
| 00:51.38 | TekLexus | how do i go about disconnecting supervision? It is a vonage line :( |
| 00:51.48 | mountainm2k | Can anybody help with Asterisk@home ? |
| 00:52.02 | Qwell | TekLexus: Vonage to asterisk? |
| 00:52.03 | Silik0n | is this thing dying? |
| 00:52.07 | Qwell | How's that work out? |
| 00:52.11 | file[laptop] | Silik0n: you are dying. |
| 00:52.37 | TekLexus | Well, its a cisco ATA-186 to a analog line to a TDM400P FXO |
| 00:52.38 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 00:52.48 | Qwell | TekLexus: Other then that, it works out ok? |
| 00:52.49 | file[laptop] | BoRiS!!!!!!!!!!!!!!!!!!!! |
| 00:52.50 | TekLexus | its TEMPORARY... |
| 00:52.55 | *** join/#asterisk implicit (~implicit@lgb-cust-66.18.140.106.mpowercom.net) |
| 00:52.56 | TekLexus | yea |
| 00:52.58 | TekLexus | works fine |
| 00:53.00 | BoRiS | Good evening file!!!!!!!! |
| 00:53.05 | BoRiS | How are you?!?!? |
| 00:53.10 | file[laptop] | not too bad, you? |
| 00:53.10 | FuRR_ | TekLexus: your going ATA -> TDM400P -> * |
| 00:53.18 | TekLexus | FuRR - yes |
| 00:53.32 | Darwin35 | Boris baby |
| 00:53.35 | Darwin35 | I miss you |
| 00:53.41 | FuRR_ | TekLexus: y? |
| 00:54.07 | TekLexus | FuRR - its a temporary line until i get NuFone hooked up |
| 00:54.22 | FuRR_ | ATA's AFAIK are FXS only |
| 00:54.32 | Silik0n | what up file |
| 00:54.55 | terrapen | i can't decide on the best setup for home |
| 00:54.58 | terrapen | IAXy or IP500 |
| 00:54.59 | Silik0n | damn this client is lagged something fierce |
| 00:55.11 | terrapen | im leaning towards IAXy + 5GHz cordless phone |
| 00:55.13 | Cresl1n | Hey!!!! |
| 00:55.13 | TekLexus | ordered a vonage fax line to handle my inbount traffic in the meantime |
| 00:55.13 | Cresl1n | :-) |
| 00:56.03 | TekLexus | so anyone have any idea how to fix the busy signal thing? |
| 00:56.07 | Darwin35 | exit |
| 00:56.12 | Darwin35 | wrong window |
| 00:56.50 | BoRiS | Darwin!!!!!!!!! |
| 00:57.02 | BoRiS | wassup? |
| 00:57.09 | file[laptop] | Silik0n: fighting to stay concious |
| 00:57.12 | BoRiS | file: just about to eat dinner |
| 00:57.24 | BoRiS | Silik0n!!! |
| 00:57.51 | file[laptop] | well that was silly |
| 00:58.18 | hmodes | texlex: can't you use busydetect on the zaptel? |
| 00:58.28 | hmodes | i was doing that for awhile |
| 00:58.44 | TekLexus | ummm. im not too familiar with the commands.. do u have an example? |
| 00:59.30 | hmodes | just set busydetect=yes in zapata.conf for the channel with the problem |
| 00:59.37 | hmodes | and you may want to play around with busycount= |
| 00:59.50 | hmodes | i think i used 10, otherwise calls tend to get cut off occasionally |
| 01:00.59 | TekLexus | so in the [channels] area i just put busydetect=yes? |
| 01:01.00 | roamer323 | hi - on 1.0.5 distro, the handler for a "registered" incoming call looks for a "peer" entry in sip.conf corresponding to the calling proxy, instead of a "user" entry - is this a fixed bug? thx |
| 01:01.12 | hmodes | yeah |
| 01:01.16 | FuRR_ | why cant SBC make it easy to find customer support #s on their website |
| 01:01.30 | TekLexus | what about the busycount=10? where does that go? |
| 01:01.36 | hmodes | same place |
| 01:01.36 | dsmouse | FuRR_: cause they suck. |
| 01:01.37 | FuRR_ | you think the phone company would want to promientaly display PHONE numbers |
| 01:01.42 | TekLexus | oh... ok :) |
| 01:01.45 | mountainm2k | Can anybody help with Asterisk@home ? |
| 01:02.08 | FuRR_ | mountainm2k, whats yer issue |
| 01:02.32 | mountainm2k | Out-of-box setup, added TWO extensions, 210 and 211 using AMP... |
| 01:02.45 | mountainm2k | They can call eachother, but I can't call voicemail, etc... |
| 01:02.49 | Essobi | Jeez. |
| 01:02.52 | mountainm2k | there are no trunks yet, just a test system... |
| 01:03.10 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 01:03.13 | mountainm2k | I can see the * is sending data out the NIC, but I can't hear it... |
| 01:03.13 | TekLexus | isnt it funny when you call vonage, and their on hold music is choppy and keeps crapping out? good service they got there.... |
| 01:03.26 | Essobi | I can't get chan_sip to decode the any ANI.. It's fricking annoying the piss out of me. |
| 01:03.31 | Silik0n | they prolly get MOH from an asterisk server |
| 01:03.38 | shmaltz | TekLexus; there is sum interesting news on Yahoo about vonage |
| 01:03.54 | mountainm2k | schmaltz: URL? |
| 01:04.11 | shmaltz | http://story.news.yahoo.com/news?tmpl=story&ncid=1212&e=2&u=/nm/20050214/wr_nm/telecoms_vonage_dc&sid=95573503 |
| 01:04.16 | Essobi | It's in my CDRs, and it's in my AGIs, but chan_sip refuses to decode cid number.. cidname is fine. |
| 01:04.27 | Essobi | Anyone else running a head build of today? |
| 01:04.36 | shmaltz | very intersting if it's true |
| 01:05.02 | shmaltz | it means that some providers went ahead thru the trouble and blocking vonage? |
| 01:05.08 | TekLexus | wow |
| 01:05.14 | shmaltz | it's like giving you AOL instead of Internet |
| 01:05.14 | file[laptop] | all the leaves are brown, and the sky is grey... |
| 01:05.51 | mountainm2k | interesting... |
| 01:05.58 | mountainm2k | wonder if it's Qworst... |
| 01:06.01 | Essobi | Umm.. If you're 555 then I'm 666? How's it feel to be the heretic? |
| 01:06.15 | shmaltz | Mountanm2k; I dont think its quest |
| 01:06.19 | shmaltz | but who knows |
| 01:06.28 | shmaltz | maybe Cabel |
| 01:06.33 | shmaltz | cable |
| 01:07.19 | *** join/#asterisk IsMe (~some@219.95.222.101) |
| 01:08.14 | Essobi | I'm refetching head, and rebuilding. This is fucking agravating me. |
| 01:08.36 | rvhi | hi i have a voicemail question, hope someone can help me. |
| 01:08.41 | IsMe | Essobi cool down man |
| 01:08.52 | rvhi | to check vm, i have to use voicemailmain. |
| 01:08.58 | shmaltz | interesting: |
| 01:09.00 | shmaltz | http://www.vonage-forum.com/ftopic1082.html |
| 01:09.11 | shmaltz | TekLexus look at above URL |
| 01:09.15 | rvhi | if there a way to let users enter the extension number and then passwd |
| 01:09.27 | shmaltz | rvhi, yep use a macro |
| 01:09.27 | rvhi | *is there? |
| 01:09.42 | rvhi | how to use macro? |
| 01:09.56 | srt | rvhi: thats what voicemailmain prompts for if you call it without arguments |
| 01:10.10 | rvhi | oh, |
| 01:10.37 | rvhi | if i have multiple domains, can i use voicemailmain(@domain)? |
| 01:10.47 | rvhi | i mean context |
| 01:11.04 | shmaltz | here is an example: |
| 01:11.06 | shmaltz | [macro-pwd] |
| 01:11.07 | shmaltz | exten => s,1,Read(PSWD|vm-password|4) |
| 01:11.09 | shmaltz | exten => s,2,Gotoif($[${PSWD} = 0000] ?30) |
| 01:11.11 | shmaltz | exten => s,3,Hangup |
| 01:11.12 | shmaltz | exten => s,30,Goto(phrase-menu,s,3) |
| 01:11.14 | shmaltz | exten => t,1,Hangup |
| 01:11.15 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 01:11.15 | shmaltz | [phrase-menu] |
| 01:11.17 | shmaltz | exten => s,1,Answer ; Answer the line |
| 01:11.19 | shmaltz | exten => s,2,Macro(pwd) |
| 01:11.21 | shmaltz | exten => s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds |
| 01:11.22 | shmaltz | exten => s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds |
| 01:11.24 | shmaltz | exten => s,5,BackGround(custom/phrase-menu) ; Play main menu. |
| 01:11.25 | shmaltz | I use it for phrase recording, here the password is 0000 |
| 01:11.28 | shmaltz | rvhi, RTFM |
| 01:11.28 | file[laptop] | shmaltz: never... paste... like that again |
| 01:11.40 | shmaltz | sorry, file you are right |
| 01:11.55 | shmaltz | didn't realize, shoud have done priv |
| 01:12.08 | dsmouse | shmaltz: http://rafb.net/paste/ |
| 01:12.10 | file[laptop] | but uh rvhi, if you type show application voicemailmain in the asterisk console, it'll tell you all about it - including usage and options! |
| 01:12.14 | dsmouse | shmaltz: go. there. now. |
| 01:12.45 | rvhi | ths folks, appreciate it! |
| 01:13.01 | shmaltz | dsmouse, whats that for? |
| 01:13.58 | dsmouse | you put what you want to paste in the web form. it gives you a url. you paste the url |
| 01:14.48 | TekLexus | schmaltz - glad Optimum online isnt blocking ports like that ISP |
| 01:15.09 | *** join/#asterisk Moc (~Moc@modemcable212.49-80-70.mc.videotron.ca) |
| 01:16.24 | Moc | hi all |
| 01:18.30 | ariel_ | ok after rebooting my asterisk box and starting over I am still not able to use g729 codec. It keep telling me no compatible codec found. |
| 01:18.43 | shmaltz | dsmouse, thanks |
| 01:18.46 | ariel_ | argh |
| 01:19.25 | file[laptop] | ariel_: sip debug and see what the codec negotiation says |
| 01:20.10 | ariel_ | seems like it wants either ulaw But I have on the sipura 2100 set for only g729. |
| 01:20.22 | shmaltz | more on the vonage story: |
| 01:20.23 | shmaltz | http://www.broadbandreports.com/shownews/60323 |
| 01:20.27 | file[laptop] | pastebin the appropriate section |
| 01:20.38 | ariel_ | I have only setup disallow=all allow=g729 |
| 01:20.49 | file[laptop] | I don't care about that, pastebin the sip debug :) |
| 01:21.05 | ariel_ | file[laptop], ok just a sec it's long. |
| 01:21.09 | moonwick | anyone in here have a working T100P? |
| 01:21.14 | file[laptop] | 'datz why we have... pastebin |
| 01:21.17 | Darwin35 | what happen to sqlite and * |
| 01:21.24 | Darwin35 | di sqlite get pulled |
| 01:21.42 | ariel_ | file[laptop], I know that. I will have it in a few minutes. |
| 01:21.44 | Darwin35 | i dont find the res_sqlite |
| 01:21.52 | file[laptop] | go go pastebin! |
| 01:22.14 | *** join/#asterisk HitTop (~Miranda@HSE-Toronto-ppp3491900.sympatico.ca) |
| 01:22.37 | *** join/#asterisk TekLexus (~Mnemonic@206.231.230.230) |
| 01:22.54 | *** join/#asterisk MrEntropy (~entropy@ppp55-252.lns1.adl2.internode.on.net) |
| 01:22.56 | MrEntropy | yo |
| 01:25.20 | rvhi | tried to setup voicemail config using mysql |
| 01:25.22 | rvhi | http://voip-info.org/wiki-Asterisk+voicemail+database |
| 01:25.28 | rvhi | it seems a little bit old |
| 01:25.28 | Essobi | Heh. did it scare the ghost away? >:) |
| 01:25.30 | ariel_ | file[laptop], http://pastebin.ca/5842 |
| 01:25.37 | rvhi | anyway has any suggestion? |
| 01:25.52 | file[laptop] | rvhi: that's for CVS stable, for CVS head search for realtime voicemail |
| 01:26.29 | file[laptop] | ariel_: codec isn't the problem here, 'tis NAT or something |
| 01:26.41 | file[laptop] | or firewall... |
| 01:26.57 | ariel_ | I did not get it all just the last part. argh. |
| 01:27.10 | file[laptop] | Feb 14 17:22:47 WARNING[21155]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call c6f4f360-1ab6ced1@192.168.88.156 for seqno 101 (Critical Response) |
| 01:27.14 | file[laptop] | asterisk couldn't send a packet to her |
| 01:27.34 | *** join/#asterisk NormAst (HydraIRC@Ottawa-HSE-ppp4117547.sympatico.ca) |
| 01:27.41 | ariel_ | yes that was the wrong one. I am trying to get this from the remote server. |
| 01:27.48 | file[laptop] | ah |
| 01:29.54 | Essobi | Decent interface |
| 01:32.21 | Essobi | hah |
| 01:33.28 | rvhi | is the latest stable version 1.0.5? |
| 01:33.38 | file[laptop] | released yes. |
| 01:33.39 | Essobi | Blasted ass stupid fricking -head build. |
| 01:33.48 | file[laptop] | Essobi: did you do a make clean? hmm? |
| 01:33.58 | rvhi | how far is 1.0.6 away? |
| 01:34.04 | file[laptop] | rvhi: eh who knows |
| 01:34.14 | file[laptop] | if you want the latest and greatest, use CVS stable |
| 01:34.16 | Essobi | I wiped it clean, rebuilt all, reinstalled and this bug I've been looking at all for 5 hours just flat disappears and CLID in chan_sip is fine again. |
| 01:34.16 | ariel_ | file[laptop], thank you for the help but it's even more then that. seems that when I do get the g729 going it seg fault. |
| 01:34.21 | file[laptop] | it's the latest... and greatest... stable |
| 01:34.28 | file[laptop] | ariel_: awww |
| 01:34.38 | file[laptop] | Essobi: that's what you get for no cleaning! |
| 01:34.43 | _Vile | rm -rf |
| 01:34.44 | Essobi | PSH. |
| 01:34.48 | *** join/#asterisk zd_eyez (~zd_eyez@Ottawa-HSE-ppp257967.sympatico.ca) |
| 01:34.49 | Essobi | I did. |
| 01:34.50 | ariel_ | yes it's stable.... argh I hate this digium is gone for the night. |
| 01:34.58 | Essobi | I think I had an old .so |
| 01:35.02 | Essobi | in modules. |
| 01:35.04 | rvhi | i guess no reason to wait for 1.0.6 to come out |
| 01:35.08 | file[laptop] | ariel_: run it inside gdb and see where it exactly crashes |
| 01:35.14 | jalsot | what is the correct way to set up rx/txgain with ztmonitor. I mean, practically. Which talk is defined as reference? |
| 01:35.45 | ariel_ | file[laptop], sure thing as soon as I get it to recover it's a remote system |
| 01:36.24 | _Vile | RM -RF |
| 01:36.44 | ariel_ | file[laptop], don't you hate it when the remote side is slow and it starts to crash.... |
| 01:37.31 | file[laptop] | yes |
| 01:42.23 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 01:42.39 | rvhi | one more question about voicemailmain, if i have no option, would it go to the same context? |
| 01:42.56 | rvhi | i.e. is the extension context the same as voicemail context? |
| 01:44.08 | TekLexus | anyone know if the reason i do not get music on hold is because i didint remove the ; infront of ;musiconhold=default in zapata.conf? |
| 01:45.10 | *** join/#asterisk Weezey (WeezeyD@206.210.109.233) |
| 01:45.41 | Weezey | is asterisk reliable enough to sell to customers? |
| 01:47.12 | *** join/#asterisk clinthome (~clinthome@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net) |
| 01:53.29 | implicit | weezey depends on who your customer is |
| 01:54.34 | *** join/#asterisk FirstSword (~FirstSwor@host66146135a8.biz.tor.fcibroadband.com) |
| 01:57.53 | FuRR_ | TekLexus: ";" <- Comment |
| 01:59.32 | TekLexus | i know |
| 01:59.47 | TekLexus | i removed the comment, and i still get no MOH |
| 02:01.05 | *** join/#asterisk znoG (gs@200.115.216.109) |
| 02:01.08 | TekLexus | brb |
| 02:01.09 | *** part/#asterisk TekLexus (~Mnemonic@206.231.230.230) |
| 02:06.09 | dsmouse | hehe |
| 02:06.19 | dsmouse | I just had a conversation with my SO |
| 02:06.45 | dsmouse | "Jamie, I'm going to install something in the kitchen... if you need to use the phone, dial 9 to get out" |
| 02:07.34 | *** join/#asterisk Koshatul (~evangelio@202.9.38.223) |
| 02:10.21 | *** join/#asterisk okieplaya (~jjj@ip68-229-252-53.ok.ok.cox.net) |
| 02:12.50 | yashax | Guys, please help me out to find the latest SIP firmware for Polycom SoundPoint IP500?!! |
| 02:13.36 | yashax | Right now the firmware is for an Altigen PBX |
| 02:15.31 | *** join/#asterisk Nukemizer (~Nuke@66.237.85.58.ptr.us.xo.net) |
| 02:17.22 | yashax | anyone? |
| 02:17.56 | Mavvie | yashax: voip-info.org |
| 02:18.05 | *** join/#asterisk ayano (~erik_leee@adsl-66-51-208-150.dslextreme.com) |
| 02:18.08 | okieplaya | Hi guys im useing a x100p noc off that works ok but the sound is real low if i call them from the softphone but from lan line talking to me its loud thinking it was my mic i put them on hold for hold music and it was stil realy low vol any i deals? |
| 02:18.38 | okieplaya | can i turn it up some where |
| 02:18.40 | okieplaya | ? |
| 02:23.54 | ayano | Did anyone answer? |
| 02:27.15 | *** join/#asterisk znoG (gs@200.115.216.109) |
| 02:28.48 | *** join/#asterisk znoG (gs@200.115.216.109) |
| 02:36.18 | ariel_ | file[laptop], are you still around? |
| 02:36.29 | file[laptop] | yes |
| 02:38.00 | ariel_ | ok I got the codec to work for asterisk calling the device. But now when the device a sipura 2000 tries to call out I am now getting the message Feb 14 18:36:03 NOTICE[21155]: chan_sip.c:2773 process_sdp: No compatible codecs! |
| 02:38.12 | file[laptop] | so do what I said... and we shall see |
| 02:38.35 | ariel_ | easyer said then done |
| 02:38.52 | dsmouse | rxgain in the zapconfig? |
| 02:38.56 | dsmouse | bah |
| 02:40.49 | ManxPwr | ~docs |
| 02:40.50 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 02:41.34 | *** part/#asterisk ayano (~erik_leee@adsl-66-51-208-150.dslextreme.com) |
| 02:42.16 | *** join/#asterisk znoG (gs@200.115.216.109) |
| 02:44.36 | ariel_ | file[laptop], http://pastebin.ca/5845 |
| 02:45.48 | *** join/#asterisk jetx (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
| 02:46.06 | file[laptop] | well that's of no help |
| 02:46.11 | MrEntropy | what exactly is the "next hop uri"? |
| 02:46.15 | bjohnson | ariel_: what are you trying to do? |
| 02:46.49 | ariel_ | trying to get a sipura to use g729 for making calls I can call it via g729 but it can not use it through the asterisk server. |
| 02:46.57 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 02:47.22 | file[laptop] | ariel_: do you have disallow=all allow=g729 in the general section? |
| 02:47.57 | ariel_ | it's in the sip.conf for the user. Not in the general section there are other users on the system. |
| 02:48.07 | bjohnson | ariel_: sorry .. played with SPAs a lot but not with g729 |
| 02:48.28 | ManxPwr | disallow=all and allow=g729 in the [happysipdevice] section. |
| 02:48.35 | ariel_ | bjohnson, thanks I am just trying to get this working right. |
| 02:48.46 | ariel_ | ManxPwr, yes it's there. |
| 02:48.53 | ManxPwr | then do a disallow=all allow=happycodec for each of the sip peers and allow=all in [general] |
| 02:49.06 | file[laptop] | happy happy |
| 02:49.10 | ariel_ | ok let me try that ManxPwr |
| 02:49.45 | ManxPwr | ariel_, In all of my servers that use SIP I do disallow=all and allow= lines for each of the codecs I will use (in [general]) |
| 02:50.00 | ManxPwr | then disallow=all and allow= the one codec I want in each of the peers |
| 02:51.36 | *** part/#asterisk obiyoda (~Jared@70-58-164-219.bois.qwest.net) |
| 02:55.02 | *** join/#asterisk nvadekar (~nvadekar@66.55.113.140.ppp.northrock.bm) |
| 03:01.29 | tzanger | evil rabbi |
| 03:01.30 | tzanger | hahaha |
| 03:02.13 | *** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.we.client2.attbi.com) |
| 03:05.21 | ariel_ | ManxPower, & file[laptop] looks like the sipura 2000 is only sending out ulaw request. It's not sending g729 request at all. It's not the asterisk. I put up sipura 2100 from here and I can use the g729 just fine. Now to see if I can find out why the sipura is now working right. |
| 03:05.47 | ariel_ | thank you for the help. |
| 03:09.21 | rvhi | anyone knows an easy way to do auto attendant? |
| 03:09.34 | rvhi | do i have to write the extension list by hand? |
| 03:10.10 | *** join/#asterisk eipi (~eipi@OL128-44.fibertel.com.ar) |
| 03:10.27 | brc_ | yes, and yes |
| 03:10.59 | *** join/#asterisk muesli (~muesli@mail.muehlhaeuser.de) |
| 03:11.39 | *** join/#asterisk florz (nobody@odnb-d9baa465.pool.mediaWays.net) |
| 03:14.24 | ManxPower | ariel_, do a factory reset on the SIPura |
| 03:14.41 | rvhi | wow, that's a lot of work |
| 03:16.34 | rvhi | with realtime/mysql support of extensions.conf, is it possible to do it on the web interface? |
| 03:17.04 | Moc | anyone use skinny ? |
| 03:18.10 | shmaltz | rvhi, Asterisk is not an out of the box solution, it is howver very flexiable, and the answer is yes |
| 03:19.37 | rvhi | this biggest problem i have with hard coded extensions.conf is that the number must be in sequence |
| 03:19.49 | rvhi | if i add a new line, i have to renumber all the rest |
| 03:20.13 | rvhi | i think mysql backend is the way to go, but not sure how stable it is. |
| 03:20.54 | shmaltz | rvhi, on the wiki there is a scritp for changing this. search google |
| 03:21.00 | ManxPower | rvhi, Uh, CVS-HEAD has the "n" priority. As I'm sure you've seen discussed on the mailing list. |
| 03:21.04 | shmaltz | why don't you use macros |
| 03:21.34 | ManxPower | Granted, CVs-HEAD really should just be used for testing, not production. |
| 03:22.05 | Nugget | it's negligent to suggest CVS-HEAD to someone so afraid of config files that they want a gui to configure asterisk. |
| 03:22.14 | rvhi | this is the second time someone told me to use macros. i'd better do some research on that. :) |
| 03:22.27 | shmaltz | ManxPower, is there any good documentation for n? |
| 03:22.50 | ManxPower | rvhi, Once you learn Asterisk you'll find that renumbering priorities happes much less often than you'd expect. |
| 03:22.51 | tzanger | ManxPower: nonsense, HEAD is great for production :-) |
| 03:22.52 | shmaltz | I understand the n+101, but I don't understand the others |
| 03:23.06 | ManxPower | shmaltz, no idea. I don't use CVS-HEAD. I run production servers. |
| 03:23.30 | shmaltz | I also run production ones, but I use HEAD on those |
| 03:23.36 | Nugget | and yes, let me be the third person to suggest macros. :) |
| 03:23.51 | ManxPower | shmaltz, It's better to forget about n+101 and concentrate on ${DIALSTATUS} |
| 03:23.58 | shmaltz | rvhi, :) |
| 03:24.00 | shmaltz | it was me the first time |
| 03:24.35 | shmaltz | ManxPower whats bad with n+101 and ${DIALSTATUS} togther |
| 03:24.42 | *** join/#asterisk adjacent (~scott@64.203.220.105) |
| 03:24.55 | Nugget | ManxPower: did you see my pastbin alternative to your looping menu approach? I was curious to hear your comments on my alternate approach. I haven't used it much and I'm worried I may have missed an aspect of the technique. |
| 03:24.56 | shmaltz | so only when busy/congested I get ${DIALSTATUS} |
| 03:25.09 | ManxPower | shmaltz, n+101 is SO terribly limiting. |
| 03:25.25 | ManxPower | DIALSTATUS is FAR FAR more flexible. |
| 03:25.39 | ManxPower | I don't think I use n+101 anywhere in my dialplans. |
| 03:25.45 | ManxPower | I use macros, however. |
| 03:25.48 | Nugget | http://pastebin.ca/5539 vs http://pastebin.ca/5540 |
| 03:26.24 | tzanger | ManxPower: regarding your ${DIALSTATUS} post to -users |
| 03:26.33 | tzanger | what are your thoughts on CONGESTION vs CHANUNAVAIL |
| 03:26.34 | tzanger | ? |
| 03:27.11 | ManxPower | Nugget, Other than there being no way your paste could actually work? |
| 03:27.26 | Nugget | it works for me. |
| 03:27.26 | ManxPower | tzanger, My mind is a blank slate when it comes to that. 8-) |
| 03:27.49 | tzanger | ManxPower: :-) I'm trying to convince Mark that CONGESTION should only be returned if the far end TOLD us it was congested |
| 03:27.57 | tzanger | and CHANUNVAIL if the other end could nto be contacted for comment |
| 03:28.22 | ManxPower | Nugget, exten => t,2,Goto(ivrhangup,0,1) and you don't have an exten => 0 in [ivrhangup] |
| 03:28.40 | shmaltz | ManxPower, do you use any multiline phones? |
| 03:28.47 | ManxPower | tzanger, that makes a lot of sense. |
| 03:28.48 | Nugget | ManxPower: it goes to the included 0 from ivr. |
| 03:28.53 | ManxPower | schurig, Yes. |
| 03:29.01 | Nugget | :0 |
| 03:29.02 | tzanger | ManxPower: feel like you're getting ambushed? |
| 03:29.09 | ManxPower | Nugget, far more complicated than you need. |
| 03:29.19 | Nugget | it looks far less complicated than your approach |
| 03:29.27 | ManxPower | tzanger, When I feel that you will no longer see responses from me. |
| 03:29.39 | ManxPower | Nugget, I disagree. |
| 03:29.43 | shmaltz | I'm trying to figure out the best way to ring mutiple line phones at once I right now use Local but I don't like the behavior of Local, |
| 03:29.56 | ManxPower | Nugget, We should get tzanger's opinion! |
| 03:30.01 | Nugget | hah |
| 03:30.02 | tzanger | eh? |
| 03:30.05 | Nugget | he'll just tell us to use HEAD |
| 03:30.14 | tzanger | everyone is entitled to my opinion :-) |
| 03:30.28 | Nugget | http://pastebin.ca/5539 vs http://pastebin.ca/5540 -- tzanger will be the swing vote. |
| 03:30.32 | Nugget | which is more complicated? |
| 03:30.34 | Nugget | :) |
| 03:30.50 | tzanger | what am I judging? |
| 03:30.53 | tzanger | loading both up now |
| 03:31.12 | Nugget | it's an ivr menu that loops once but then hangs up if the user times out the second time. |
| 03:31.19 | tzanger | I have no idea whose is whose so its impartial too |
| 03:31.26 | tzanger | Nugget: ok |
| 03:31.30 | Nugget | manxpower's is easier to iterate more than twice, but there's no reason my technique can't also do it on the third loop. |
| 03:32.16 | tzanger | 5539 looks pretty straightforward, now looking at the other |
| 03:33.01 | tzanger | whoa |
| 03:33.16 | tzanger | 5540 took me like 4 tries to understand what was going on |
| 03:33.22 | Nugget | heh, ok, I lose. :) |
| 03:33.25 | tzanger | it's slick, but I'm not a fan of slick unless there's no oterh way to do it |
| 03:34.42 | tzanger | in general overwriting already-defined extensions (like what ivrhangup does with 't') makes my head hurt, as I get mixed up |
| 03:34.51 | tzanger | but wait |
| 03:34.54 | tzanger | does it work? |
| 03:34.59 | Nugget | yes |
| 03:35.19 | tzanger | since you include [ivr] first, would its definition of 't' not superscede [ivrhangup]'s 't'? |
| 03:35.23 | tzanger | just as |
| 03:35.33 | tzanger | exten => 123,1,Goombah |
| 03:35.43 | tzanger | exten => 123,1,Goozfrahbah |
| 03:35.59 | tzanger | would have the former take place? |
| 03:36.05 | tzanger | oh wait |
| 03:36.05 | dsmouse | is there a way to pick a random extention? |
| 03:36.07 | tzanger | nevermind |
| 03:36.13 | tzanger | I'm thinking of pattern matching |
| 03:36.14 | tzanger | where |
| 03:36.20 | tzanger | 123,1,Goombah |
| 03:36.26 | tzanger | _XXX,1,Goozfrahbah |
| 03:36.40 | tzanger | 123 is the more exact match so it gets precedence |
| 03:36.43 | dsmouse | like Goto(contextT,s,${RAND(0,5)}) |
| 03:36.43 | Nugget | yeah |
| 03:37.02 | tzanger | 5540's is certainly the slicker way of doing things |
| 03:37.22 | tzanger | but it taks more neural activity to understand |
| 03:37.36 | Nugget | I just generally shy away from adding connection variables for tracking state. I find that approach to be more risk-prone over time. |
| 03:37.42 | FirstSword | Hi, i want to ask if there will be nat problem if the setup is : asterisk <-> nat <-> internet <-> ser <-> pat <-> sip client |
| 03:37.56 | Nugget | I accept that my approach works more by accident than by design and is perhaps overly indirect. |
| 03:38.00 | tzanger | Nugget: especially since the asterisk diaplan has no equivalent to 'my' as Perl does |
| 03:38.05 | Nugget | yeah |
| 03:38.07 | tzanger | you end up with spelling errors killing you |
| 03:38.29 | tzanger | or things like ${EXTENSION} and ${EXTEN} and not realizing you want one over the other |
| 03:39.47 | tzanger | hahahaha |
| 03:39.56 | tzanger | One cow is talking to another in the barn and says, "I don't know what to think about this Mad Cow Disease. There are all these complicated scientific issues, economic issues, and political issues! What's a poor cow to think?" |
| 03:40.02 | tzanger | The other cow says: "I don't care; I'm a helicopter!" |
| 03:40.32 | Nugget | je bent niet gek, je bent een vliegtuig! |
| 03:40.46 | tzanger | I can't speak freaky-deaky dutch |
| 03:40.53 | *** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
| 03:40.56 | Nugget | gekke tzanger. |
| 03:40.56 | tzanger | (I'm guessing it's dutch) |
| 03:40.58 | Nugget | tja. |
| 03:41.16 | tzanger | je looks french but nothing else looked french, and it looked slightly german too but I know it's not german :-) |
| 03:41.39 | Nugget | it means "I'm not crazy, I'm an airplane!" |
| 03:41.48 | tzanger | heh |
| 03:41.53 | Nugget | some dutch standup comedian said it in the '80s and now it's an idiom |
| 03:42.12 | tzanger | oh for fuck sakes |
| 03:42.24 | tzanger | the ID4 I downloaded is in German |
| 03:42.59 | Nugget | das ist nicht so gut. |
| 03:43.52 | tzanger | nein, ich kann nicht so gut verstei Deutsch |
| 03:45.10 | tzanger | und ich kann nicht wert eine Schribe auch |
| 03:47.57 | Nugget | heh |
| 03:52.02 | tzanger | bah |
| 03:52.05 | tzanger | I fucked that up |
| 03:52.55 | tzanger | und ich kann nicht wert eine scheisse auch nicht buchstabieren... I had to look that last word up, heh |
| 03:53.20 | tzanger | although it makes sense |
| 03:53.27 | tzanger | booksomething |
| 03:54.15 | *** join/#asterisk TheEmperor (~mattn@203.121.47.100) |
| 03:54.29 | *** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-132.modem.logical.net) |
| 03:54.35 | TheEmperor | anyone know of a good softphone that can use IAX2? |
| 03:54.52 | Nivex | TheEmperor: iaxcomm for cross platform, Firefly if on Windows |
| 03:54.53 | tzanger | firefly? |
| 03:54.59 | Carp1 | I want to plug my cellphone into my box and route the incoming calls on my cell to my asterisk box. Is this possible? |
| 03:55.04 | tzanger | there's also that mozilla based one |
| 03:55.10 | TheEmperor | tried firefly...anymore? |
| 03:55.19 | Nugget | iaxphone is some serious crap in osx. |
| 03:55.26 | Nivex | Carp1: yes. It needs to be an FXO port, and you'll need a Cellsocket (cellsocket.com) |
| 03:55.31 | Nugget | x-lite or x-pro is really the best approach on a mac. |
| 03:55.35 | tzanger | heh |
| 03:55.42 | tzanger | I'm watching Robin Hood, Men in Tights |
| 03:55.44 | TheEmperor | i mean on windows pcs.. |
| 03:55.50 | Carp1 | How much do you think the setup will cost me? |
| 03:55.55 | tzanger | I love the Maitre d'ungeon's name.. .falafel... heh |
| 03:55.58 | Nugget | Nivex said "cross platform" so I thought I'd pitch in./ |
| 03:56.09 | tzanger | and moutard is the head guard |
| 03:56.10 | Nivex | TheEmperor: There is DIAX, but I suppose that depends on your definition of good. |
| 03:56.15 | Nugget | on windows I use x-pro. acoustic echo cancelling is handy. |
| 03:56.24 | TheEmperor | Nivex: what do you mean? |
| 03:56.39 | Nivex | TheEmperor: I was not impressed by DIAX. It was happy-go-crashy. |
| 03:56.40 | TheEmperor | I'd like to use something that has messenging capabilities |
| 03:56.52 | Nugget | asterisk lacks messaging capabilities. |
| 03:56.59 | Carp1 | Is the X100P FXO? |
| 03:57.00 | TheEmperor | i can't seem to get firefly's messenging to work.. |
| 03:57.10 | TheEmperor | even if both extensions are online at the same time? |
| 03:57.21 | implicit | Carp1: yeah |
| 03:57.26 | bjohnson | who has an iaxy? does it have a web interface? |
| 03:57.55 | Nugget | I don't think the iaxy has any interface at all. |
| 03:57.57 | file[laptop] | bjohnson: no it does not. |
| 03:58.09 | Nivex | TheEmperor: firefly's messaging was built for their network which (correct me if I'm wrong somebody) uses proprietary extensions to the IAX2 protocol. |
| 03:58.11 | bjohnson | that makes it a little hard to use |
| 03:58.16 | Nugget | yes it does. |
| 03:58.22 | file[laptop] | asterisk provisions it |
| 03:58.45 | bjohnson | so then it won't work for remote users directly to a voip provider |
| 03:58.50 | TheEmperor | Nivex:oh..no wonder I am having trouble with the messenging bt |
| 03:58.55 | Carp1 | Anyone in here use Cell Socket? |
| 03:59.47 | TheEmperor | also, how do i transfer calls to another extension if i use firefly and iax2? |
| 03:59.56 | TheEmperor | through my * server of course.. |
| 04:00.07 | bjohnson | I'm getting a ringback with my SPA 3ks that definitely seems to be a unit/device issue .. same problem with 3 SPAs on 3 different lines |
| 04:01.07 | okieplaya | TheEmperor download media xphone lot better |
| 04:01.15 | IsMe | i am looking at ebay Access bank II 24 FXS channels for *, any word of advice ? |
| 04:01.17 | TheEmperor | okieplaya: where? |
| 04:02.12 | okieplaya | just sec i find it |
| 04:02.15 | bjohnson | IsMe: buy me one |
| 04:02.34 | IsMe | lol |
| 04:03.00 | okieplaya | TheEmperor http://www.marccharbonneau.com/asterisk/mediaxphone.php |
| 04:03.11 | Sedorox | http://www.voipsupply.com/product_info.php?cPath=99_139&products_id=380&desc=Rhino%20CB24-24FXS%20Channel%20Bank%20Asterisk |
| 04:03.12 | TheEmperor | okieplaya: thanks :) |
| 04:03.14 | Sedorox | for IsMe |
| 04:03.17 | bjohnson | IsMe: make sure it has all the pieces you need. I don't know chan banks but know Adit600, Adtran750, and Carrier Banks are all popular |
| 04:03.21 | okieplaya | np |
| 04:04.21 | bjohnson | anyone have one of those IAX ATAs on ebay yet? Do they have a web interface to setup IAX to directly use a voip provider? |
| 04:05.04 | *** join/#asterisk NTJOCK (~brian@txshirts.com) |
| 04:05.42 | NTJOCK | hello, I'm having trouble with outbound dialing from SIP to Zap. Have read Asteriskdocs and the "VOIP Telephony with *".... |
| 04:05.47 | FirstSword | bjohnson: no it doesn't has web interface |
| 04:05.58 | NTJOCK | I'm not trying to dial 9 first, or do anything fancy... just basic, dial a number send it out. |
| 04:06.06 | bjohnson | FirstSword: how is it configured? |
| 04:06.12 | *** join/#asterisk Mycroft1 (~reece@202.147.104.114) |
| 04:06.18 | Mycroft1 | hi people :) |
| 04:06.28 | NTJOCK | is there a good reason to use 9 for an outside line? |
| 04:06.42 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
| 04:06.52 | Mycroft1 | anyone used spandsp with much sucess, i get the first part of the fax but the rest of the page is cut off |
| 04:06.59 | bjohnson | NTJOCK: make a exten => _NXXNXXX,1,Dial(Zap/1/${EXTEN}) |
| 04:07.10 | bjohnson | going from memory .. might be off a little |
| 04:07.12 | NTJOCK | ok... that's what I needed... I'm using exten=> _NXXNXXXXXX,1,Dial(Zap/1) |
| 04:07.20 | NTJOCK | we have 10 digit dialing... |
| 04:07.24 | NTJOCK | the second / is what was missing |
| 04:07.37 | bjohnson | that's the actual number to dial |
| 04:07.53 | NTJOCK | don't get me wrong, I really appreciate the time people put into these docs... but it's all aimed at people using T1, ISDN, and <>POTS |
| 04:08.05 | NTJOCK | thanks BJ |
| 04:08.17 | bjohnson | I need a Sipura that does IAX |
| 04:08.27 | bjohnson | SIPURA ARE YOU LISTENNING ?? |
| 04:08.28 | NTJOCK | the other thing I'm wondering is if I need to use the 9 for outside line to keep it out of conflict with our extensions |
| 04:08.39 | nestAr | i want everything to use IAX |
| 04:08.45 | nestAr | seems to work much better than SIP |
| 04:08.55 | bjohnson | NTJOCK: 9 is trditional pbx code for getting an outside line .. don't worry about it |
| 04:09.06 | NTJOCK | thanks |
| 04:09.15 | NTJOCK | I'm trying to make this as painless for my employees as possible |
| 04:09.24 | NTJOCK | they all hate the sipura ringtone so far... :) |
| 04:09.25 | NTJOCK | hehe |
| 04:09.25 | bjohnson | yeah .. scrap the 9 |
| 04:09.55 | nestAr | i had to put 8 and 9 in mine.. because part of the company dials 8 to get to outside line.. |
| 04:09.58 | NTJOCK | however, I plan on using a fairly complicated IVR/Call routing. |
| 04:10.03 | bjohnson | sipura 841? I don't have one .. but the SPA 2k and 3k have alternate ring tones .. although I haven't played with them |
| 04:10.05 | okieplaya | why is my lady that talks in voice mail so choppy ? |
| 04:10.11 | NTJOCK | 841 is nice |
| 04:10.12 | nestAr | so you can dial 8 digits (8 or 9) or just 7 digits.. |
| 04:10.13 | NTJOCK | quick to configure |
| 04:10.14 | NTJOCK | :) |
| 04:10.29 | NTJOCK | had them up and running within 5 minutes of plugging it in |
| 04:10.29 | nestAr | i like my spa-3k |
| 04:10.35 | NTJOCK | damn polycom is still giving me fits |
| 04:10.39 | NTJOCK | but I have one of 2 running. |
| 04:10.42 | NTJOCK | :( |
| 04:10.56 | bjohnson | nestAr: I'm getting a ringback after hanging up .. have you found that? |
| 04:10.57 | NTJOCK | brb, off to test the change to the dial string |
| 04:11.39 | FirstSword | Setup: Asterisk <-> NAT <-> Internet <-> SER, will this have NAT problem? |
| 04:12.03 | *** join/#asterisk modulus_ (modulus@rm-f.net) |
| 04:12.04 | modulus_ | wotcher |
| 04:12.09 | dsmouse | FirstSword: it works for me |
| 04:12.22 | Carp1 | none other than extensios dont usually start with 9 |
| 04:12.22 | Carp1 | extensions* |
| 04:12.32 | FirstSword | dsmouse: did u use any ip tunneling? or SER's natHelper? |
| 04:12.57 | dsmouse | oh, wait, SER is a perticular service? |
| 04:13.04 | FirstSword | dsmouse: or simply just port forwarding in NAT layer to asterisk? |
| 04:13.24 | dsmouse | I'm just useing a port forwarding on the NAT box |
| 04:13.33 | FirstSword | dsmouse: well. i just ser as a router to route calls to internal clients |
| 04:13.56 | *** join/#asterisk aday (~aday@aday.net.au) |
| 04:14.07 | NTJOCK | ok dialstring change works |
| 04:14.34 | Nukemizer | I am looking for some guidance on T1 config. I am having a bad time getting the wcte11xp driver to load have tried so many things ( all of which didnt work ) This is the message i get with ztcfg -vv ZT_SPANCONFIG failed on span 1: No such device or address (6) |
| 04:15.23 | Nukemizer | even when i load both zaptel and wcte11xp modules by hand, i get the same error |
| 04:16.17 | okieplaya | why is my comeida mail so choppy ? as she reads mail pass word and what i want to do ? |
| 04:16.30 | *** join/#asterisk juice (~juice@mo-65-41-197-194.dyn.sprint-hsd.net) |
| 04:18.29 | Nugget | comedia mail listen ! many press button work and do what you want . |
| 04:19.14 | GreyFoxx | Is there some way I can configure asterisk to not answer any incoming calls on my landline ? I only want it handling SIP calls and any outgoing calls. Or a document I should look over ? |
| 04:19.18 | modulus_ | unplug the landline from it |
| 04:19.37 | Nugget | configure the zap channel's default context to an empty context. |
| 04:19.42 | okieplaya | Nugget comedia mail choppy on play back |
| 04:19.51 | okieplaya | is it HD speed |
| 04:20.00 | okieplaya | ram maybe? |
| 04:20.09 | bjohnson | okieplaya: run top and watch ram and cpu usage |
| 04:20.10 | GreyFoxx | modulus_: Well I still want to be able to call out, just don' twant it answering incoming landline calls :) |
| 04:20.31 | okieplaya | cpu 3% if that ram used all up but 13mb |
| 04:20.40 | GreyFoxx | Nugget: Cool. I'll try that |
| 04:20.41 | bjohnson | okieplaya: mpg321 on my system also played havoc with playing voice prompts |
| 04:20.47 | okieplaya | i have to reboot to get ram to reset |
| 04:21.13 | *** join/#asterisk pbxman (~tmcarter@ip68-226-15-136.nc.hr.cox.net) |
| 04:21.19 | Nugget | unused ram is wasted ram. as long as it's all in buffers there's no harm it its being used. |
| 04:21.35 | okieplaya | k |
| 04:21.53 | bjohnson | sounds like the system is not maxed out anyway |
| 04:21.54 | okieplaya | bjohnson what do u do to fix it? |
| 04:22.11 | bjohnson | removed mpg321 and installed mpg123 |
| 04:22.17 | okieplaya | yea no where near it |
| 04:22.40 | okieplaya | only me on the PBX |
| 04:23.16 | okieplaya | and 1 softphone |
| 04:23.36 | okieplaya | let me see what i am runin |
| 04:30.36 | okieplaya | ok where do i look to see what MPG for music on hold i am runin sorry new to this |
| 04:30.59 | Nugget | okieplaya: download the asterisk source and do "make mpg123" |
| 04:31.27 | okieplaya | ok what all will that do? |
| 04:31.58 | Nugget | it will build the right version of mpg123 for you |
| 04:32.16 | NTJOCK | when I dial out the receiving line can be heard, but I can't. |
| 04:32.18 | NTJOCK | :( |
| 04:32.20 | NTJOCK | any ideas? |
| 04:32.21 | okieplaya | ok thanks |
| 04:32.24 | NTJOCK | SIP to ZAP call |
| 04:34.34 | *** join/#asterisk tecnico (~tecnico@user-24-236-123-31.knology.net) |
| 04:34.40 | NTJOCK | ok, it appears that my polycom doesn't recognize the call as having begun |
| 04:34.48 | NTJOCK | my sipura lets me talk, but I hear ringing (called party doesn't) |
| 04:34.50 | NTJOCK | grrrrr |
| 04:35.04 | okieplaya | bjohnson & Nugget I am runin mpg123 now still choppy |
| 04:36.02 | okieplaya | zap on hold sounds like crap also |
| 04:36.27 | iMediax | what kernel okieplaya? |
| 04:37.32 | okieplaya | hmm |
| 04:39.32 | iMediax | what kernel are you using? |
| 04:40.15 | okieplaya | CentOS 3.4 Asterisk 1.0.5 that what you are askin? |
| 04:40.33 | j_f00b3r | type 'uname -a' in a shell |
| 04:40.42 | okieplaya | ok |
| 04:41.23 | okieplaya | Linux asterisk1.local 2.4.21-27.0.2.EL #1 Wed Jan 19 02:20:34 GMT 2005 i686 i686 i386 GNU/Linux |
| 04:41.36 | okieplaya | ok? |
| 04:43.43 | okieplaya | so you think its just me? |
| 04:44.15 | okieplaya | PIII 500MHZ 128MB AM 20GB ATA 133 |
| 04:45.12 | j_f00b3r | asterisk@home right? |
| 04:45.28 | okieplaya | yes |
| 04:45.38 | okieplaya | that bad |
| 04:46.02 | okieplaya | ? |
| 04:46.10 | j_f00b3r | eh |
| 04:46.16 | j_f00b3r | build it yourself and you learn more |
| 04:46.21 | j_f00b3r | never messed with it |
| 04:47.07 | okieplaya | well i will i still new just getting started |
| 04:47.46 | okieplaya | how did you know it was asterisk@home? |
| 04:48.05 | j_f00b3r | the default hostname |
| 04:48.14 | j_f00b3r | and it runs on CentOS 3.4 |
| 04:48.15 | okieplaya | i c |
| 04:48.27 | okieplaya | u dont like centos? |
| 04:48.44 | j_f00b3r | I don't have much experience with asterisk@home |
| 04:48.58 | j_f00b3r | so I don't know how everything was built and tested |
| 04:49.07 | okieplaya | what about centos ? what do u run? |
| 04:49.33 | j_f00b3r | centos ;) |
| 04:49.37 | j_f00b3r | just never asterisk@home |
| 04:49.40 | okieplaya | haha |
| 04:49.42 | okieplaya | ok |
| 04:49.47 | *** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au) |
| 04:50.05 | okieplaya | can i ask u few more things ? |
| 04:50.43 | ariel_ | Good night folks it's bed time for me. |
| 04:51.14 | okieplaya | j_f00b3r you there ? can i ask u few more things |
| 04:51.28 | j_f00b3r | sure |
| 04:51.31 | j_f00b3r | might not know the answers |
| 04:53.12 | okieplaya | what if i was goin to try and bulid a pc that had no moveing parts and i used a 4GB sandisk for the hard drive and I see that VIA makes a 1.2 anless pc now |
| 04:53.14 | okieplaya | ? |
| 04:53.26 | okieplaya | fanless |
| 04:53.35 | j_f00b3r | not tried it |
| 04:53.38 | j_f00b3r | don't know ;) |
| 04:55.39 | okieplaya | what IP phones do you use? |
| 04:55.52 | j_f00b3r | an IAXy |
| 04:56.01 | j_f00b3r | mostly sofphones though |
| 04:56.07 | j_f00b3r | for what I'm using it for right now |
| 04:56.40 | okieplaya | office? what sofphone? |
| 04:57.02 | j_f00b3r | well |
| 04:57.16 | j_f00b3r | I use the softphone when I'm on the road for work |
| 04:57.24 | j_f00b3r | and have a DID in my local town |
| 04:57.46 | j_f00b3r | haven't used * for an office PBX yet |
| 04:57.51 | j_f00b3r | I am in the minority |
| 04:57.59 | j_f00b3r | only personal use and outbound calling for work |
| 04:58.39 | okieplaya | i c |
| 04:59.16 | j_f00b3r | I will roll it out for an office PBX very soon |
| 04:59.20 | okieplaya | u see how small the new mobos are now you bot this with sandisk http://www.mini-itx.com/ |
| 04:59.41 | j_f00b3r | right |
| 04:59.47 | j_f00b3r | check www.voip-info.org |
| 04:59.55 | j_f00b3r | they have stuff pertaining to that on there |
| 05:00.01 | okieplaya | yea read every thing on the site i can |
| 05:00.24 | okieplaya | my wife is geting piss off at e for readin on this stuff all the time |
| 05:00.45 | Sedorox | lol |
| 05:00.50 | j_f00b3r | heh |
| 05:00.55 | j_f00b3r | join the club |
| 05:00.57 | okieplaya | i work on crestron www.crestron.com |
| 05:02.01 | okieplaya | http://www.avmx.net/about2.html some stuff i do |
| 05:04.36 | okieplaya | we are lookin to settin * in home 10,000SQ FT we have installed samsung phone for now andhave ben for sometime but lookin to do VOIP |
| 05:04.50 | *** part/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 05:04.53 | okieplaya | cisco is the shit right now from what i read |
| 05:06.12 | j_f00b3r | build it from scratch |
| 05:06.26 | j_f00b3r | if you're going to roll it out in a production environment |
| 05:06.30 | j_f00b3r | you have to know how it works |
| 05:06.35 | j_f00b3r | thats the best way to learn |
| 05:07.32 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 05:07.32 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 05:07.34 | okieplaya | yea i will i have used windows for so long and i have setup windows 2k phone but they suck ass its just goin to take some time |
| 05:09.32 | okieplaya | thank for the help... nice talkin with you |
| 05:09.46 | *** join/#asterisk channan (~channan9@66.180.121.185) |
| 05:13.23 | Mycroft1 | anyone used spandsp with much sucess, i get the first part of the fax but the rest of the page is cut off |
| 05:15.14 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 05:19.48 | Sedorox | anyone use a soekris for asterisk stuff? |
| 05:20.45 | NTJOCK | has anyone successfully tricked a credit card terminal to dial out through an FXO/FXS Zaptel combo? |
| 05:21.02 | NTJOCK | I've got a Hypercom terminal that won't "see" the dialtone |
| 05:21.20 | NTJOCK | I know when I call the idiots at the support line they'll tell me to plug it in to it's own phone line |
| 05:21.35 | NTJOCK | which as far as it's concerned the FXS card should be it's own damn line |
| 05:21.53 | NTJOCK | I can use a hand phone (analog) to dial out via the idiot jack on the back of hte terminal |
| 05:22.05 | NTJOCK | Just wondering if anyone else has been there/done that with a CC terminal |
| 05:22.29 | NTJOCK | I'm getting "starting simple switch on zap/6-1" meaning that * is generating dialtone |
| 05:22.33 | NTJOCK | but the terminal never starts dialing |
| 05:22.56 | Mavvie | what does the modem say? |
| 05:23.15 | NTJOCK | it says "dialing" and that's it |
| 05:23.18 | NTJOCK | and then hangs up |
| 05:23.22 | NTJOCK | I'm guessing it's not seeing dialtone |
| 05:23.25 | Mavvie | no, what does the modem say? |
| 05:23.30 | NTJOCK | maybe the frequency doesn't get it excited right or something |
| 05:24.25 | Mavvie | if you put a normal phone in that FXO, do you get a dialtone? |
| 05:24.29 | NTJOCK | yes |
| 05:24.33 | NTJOCK | and I can make calls |
| 05:24.36 | NTJOCK | and * dials them |
| 05:24.51 | NTJOCK | I can even plug a analog set into the back of the credit card terminal |
| 05:24.52 | NTJOCK | and make calls |
| 05:25.05 | NTJOCK | so only thing I can figure is that the credit card machine doesn't like the * dialtone |
| 05:25.10 | NTJOCK | which is just stupid |
| 05:25.23 | NTJOCK | which of course makes sense... a bank is involved. |
| 05:27.34 | Mavvie | in which country are you? |
| 05:27.40 | NTJOCK | USA |
| 05:27.56 | Mavvie | aha. well at least you don't have to change the default country then :-) |
| 05:28.00 | NTJOCK | hehe |
| 05:28.01 | NTJOCK | yea |
| 05:28.18 | NTJOCK | I just can't figure out why the stupid modem in the credit card box doesn't like the dial tone * gives out |
| 05:28.31 | NTJOCK | and I unfortunately can't tell it to go ignore the dial tone |
| 05:28.47 | Mavvie | the credit card box.... |
| 05:28.54 | Mavvie | does it have an external modem? |
| 05:28.58 | NTJOCK | no |
| 05:29.01 | NTJOCK | all in one unit |
| 05:29.05 | NTJOCK | more like "stupid black box" |
| 05:29.11 | Mavvie | I take it's not an PC based thingie? |
| 05:29.14 | NTJOCK | no |
| 05:29.19 | NTJOCK | it's a credit card machine. |
| 05:29.23 | Mavvie | okies. get the screw driver :-) |
| 05:29.30 | NTJOCK | like a store would have |
| 05:29.50 | NTJOCK | I've seen them in Europe... so I know you guys have them. |
| 05:29.51 | NTJOCK | :) |
| 05:29.56 | Darwin35 | its just a card swipe machine |
| 05:30.01 | NTJOCK | exactly |
| 05:30.04 | NTJOCK | a rather nice one |
| 05:30.18 | NTJOCK | but none the less it doesn't like * dial tone |
| 05:30.26 | NTJOCK | no idea why |
| 05:30.36 | NTJOCK | and it's just script readers at the support place |
| 05:30.39 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 05:30.40 | Darwin35 | to much background noise |
| 05:30.45 | NTJOCK | they generally tell you to burn down your office and plug it in at the pole |
| 05:30.46 | Darwin35 | to much static |
| 05:30.52 | NTJOCK | k |
| 05:30.56 | NTJOCK | direct 50' cable |
| 05:31.00 | NTJOCK | i'll move it closer and try |
| 05:31.03 | NTJOCK | that's a good idea |
| 05:31.24 | Darwin35 | jitter buffer |
| 05:31.25 | NTJOCK | but it works if I plug it into the pots jack instead of * |
| 05:31.34 | NTJOCK | hmmm.... on FXS? |
| 05:31.36 | Darwin35 | lots of reasons it might not work |
| 05:31.57 | NTJOCK | might not like the voltage on the line |
| 05:32.03 | Darwin35 | might be |
| 05:32.04 | NTJOCK | isn't zaptel a low voltage toy? |
| 05:32.10 | NTJOCK | i.e. not the full 48vdc? |
| 05:32.16 | brc_ | ~fxs |
| 05:32.17 | jbot | hmm... fxs is foreign exchange system - or the type of port you need to connect a analog device (phone, fax machine) to a pbx |
| 05:32.18 | brc_ | ~fxo |
| 05:32.19 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 05:32.19 | Darwin35 | you have to set it for low voltage |
| 05:32.25 | heath__ | do you have a long ass wire running from * to the thing? |
| 05:32.35 | NTJOCK | define "long ass" |
| 05:32.39 | brc_ | NTJOCK, are you trying to plug a cc terminal into a fxo port? |
| 05:32.43 | NTJOCK | no |
| 05:32.47 | heath__ | 25 feet+ |
| 05:32.54 | Mavvie | no? |
| 05:32.56 | Darwin35 | into his fxs port |
| 05:32.56 | NTJOCK | CC terminal -> FXS port, dial out on FXO Port |
| 05:33.01 | brc_ | okay |
| 05:33.08 | NTJOCK | CC terminal -> POTS jack works fine |
| 05:33.14 | NTJOCK | stops working when I insert the * box. |
| 05:33.19 | brc_ | uhm |
| 05:33.25 | NTJOCK | 25 ish foot cable from CC terminal to Pots/* |
| 05:33.27 | brc_ | well do you have zaptel and asterisk configured correctly? |
| 05:33.29 | NTJOCK | 3 cable from POTS to * |
| 05:33.31 | NTJOCK | yes |
| 05:33.41 | brc_ | does a normal phone work? |
| 05:33.44 | NTJOCK | I can call out if I put a analog phone on the back of hte CC terminal |
| 05:33.48 | NTJOCK | yes |
| 05:33.54 | NTJOCK | can call from SIP (validate *) |
| 05:33.57 | NTJOCK | can call from other FXS ports |
| 05:34.04 | NTJOCK | and can call from jack on back of CC terminal |
| 05:34.13 | NTJOCK | cc terminal isn't dialing |
| 05:34.21 | NTJOCK | I'm guessing it doesn't like the dial tone |
| 05:34.32 | NTJOCK | but I can't imagine why |
| 05:35.02 | heath__ | you could try a shorter cable, that has fixed probs with my ATA's, but I have no idea on your hardware |
| 05:35.09 | NTJOCK | k |
| 05:35.22 | NTJOCK | it's a digium TDM400 with 3FXS and 1 FXO port |
| 05:35.23 | Sedorox | Question... what would cause a echo with BT100 |
| 05:35.25 | Sedorox | 's on * |
| 05:35.29 | NTJOCK | I also have a second TDM400 with 4 FXO ports |
| 05:35.48 | NTJOCK | running on a PIV 2.2 Ghz with 2GB ram |
| 05:35.51 | NTJOCK | SATA |
| 05:35.53 | NTJOCK | GB ethernet |
| 05:36.00 | NTJOCK | it's a pretty quick box |
| 05:36.06 | NTJOCK | rarely breaks 2% on CPU load |
| 05:36.23 | NTJOCK | let me go play with cables |
| 05:36.24 | NTJOCK | brb |
| 05:36.58 | Mavvie | Sedorox: http://lists.digium.com/pipermail/asterisk-users/2005-February/088794.html |
| 05:37.00 | Mavvie | ~echo |
| 05:37.01 | jbot | somebody said echo was Displays the given arguments on the screen. Syntax: echo (arg1) (arg2) ..(argN). Where arg1 through argN are the arguments to echo. Example: echo "Hello World" displays the string "Hello World". |
| 05:37.03 | heath__ | i saw some old 2 port FXO mods on ebay for dirt cheap, but no such deals for the FXS; do those awesome deals exist elsewhere? |
| 05:37.40 | *** join/#asterisk datareactor (datareacto@203.81.192.33) |
| 05:37.53 | Mavvie | jbot: echo is also "Why echo occurs" at http://lists.digium.com/pipermail/asterisk-users/2005-February/088794.html |
| 05:37.54 | jbot | Mavvie: okay |
| 05:38.37 | Sedorox | thats different.. mine does it on BT100 -> Asterisk - IAX2 - Asterisk -> BT100 |
| 05:38.47 | Mavvie | not it's not. |
| 05:39.08 | Sedorox | ? |
| 05:39.17 | NTJOCK | no dice |
| 05:39.19 | Sedorox | 1. It is not in the Asterisk box because IP to IP calls do not suffer |
| 05:39.19 | Sedorox | this malady |
| 05:39.25 | NTJOCK | changin cables doesn't fix it |
| 05:39.42 | NTJOCK | I even dialed the merchant # from a handset plugged into the back of the terminal |
| 05:39.44 | NTJOCK | worked fine |
| 05:39.50 | NTJOCK | it's just not detecting the dial tone for some reason |
| 05:41.32 | *** join/#asterisk muesli (~muesli@mail.muehlhaeuser.de) |
| 05:42.28 | *** join/#asterisk trig_hm (~jb@home.monkeypr0n.org) |
| 05:43.00 | Essobi | Mmm. |
| 05:43.27 | Essobi | Anyone got an idea why I can't exec Monitor from an AGI with more the one argument? |
| 05:47.45 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 05:52.14 | *** join/#asterisk Ron-Na (~ronald@203.70.36.126) |
| 05:54.38 | Ron-Na | Can anybody lend me his ear? |
| 05:55.07 | Ron-Na | I have SuSE 9.2 and asterisk gives me an error while compiling it |
| 05:55.35 | Ron-Na | It complaints cannot find -lssl but I have installed ssl |
| 05:55.37 | *** join/#asterisk Beirdo (~gjhurlbu@beirdo.user) |
| 05:57.38 | Essobi | Mmm. Perl AGI is acting wack. |
| 06:01.23 | Nugget | Essobi: did you solve the callerid wackness? |
| 06:01.40 | *** join/#asterisk af_ (~af@ip-148-227.sn1.eutelia.it) |
| 06:02.21 | Essobi | Yea. |
| 06:02.30 | Essobi | reuped |
| 06:02.37 | Essobi | make clean all install did the mojo |
| 06:02.39 | Essobi | :\ |
| 06:02.56 | Essobi | After I stared at it for 5 hours cursing chan_sip |
| 06:03.12 | Essobi | now using a monitor under AGI is acting wacky |
| 06:03.14 | Essobi | :\ |
| 06:05.58 | Essobi | exten => s,1,Monitor(wav,/tmp/test,m) |
| 06:06.01 | Essobi | works fine... |
| 06:07.02 | Essobi | $rc = $AGI->exec('Monitor', "wav,/tmp/test2,m"); |
| 06:07.06 | Essobi | won't work. |
| 06:07.07 | Essobi | :\ |
| 06:07.49 | Essobi | returns -1... |
| 06:14.04 | *** join/#asterisk Astrisk-boob (~falker@69-17-136-9.kingkom.com) |
| 06:14.27 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
| 06:15.52 | Essobi | Baaah |
| 06:16.13 | Sedorox | no kick |
| 06:16.15 | Sedorox | just send to me |
| 06:16.16 | Sedorox | :-p |
| 06:16.33 | Essobi | Pssh. |
| 06:16.41 | *** join/#asterisk shaZwaz (~chatzilla@203.81.196.167) |
| 06:16.45 | Essobi | AGI->Exec is pissing me off |
| 06:16.51 | shaZwaz | hi room |
| 06:18.01 | Astrisk-boob | hello everyone! sup shaZwaz |
| 06:18.13 | shaZwaz | Q: Xlite Choppy voice on out-going ? |
| 06:18.21 | shaZwaz | hi |
| 06:18.37 | Astrisk-boob | i want to try astrisk server.. what flavour of linux do you guys reccomend? |
| 06:19.12 | shaZwaz | one you are more used to |
| 06:19.33 | Astrisk-boob | oh ya.. i have been playing with linux for under 1 year. |
| 06:20.07 | Astrisk-boob | mosty fedora and freeBSD... but installed slackware 10 on a pentium 400mhz 256 ram |
| 06:20.56 | shaZwaz | slackware is fine |
| 06:21.14 | Astrisk-boob | cool.. is there any advantages to using gentoo? |
| 06:21.32 | Essobi | Jees. |
| 06:21.37 | Essobi | I don't get this at all. |
| 06:21.46 | shaZwaz | not any that I have heard :) |
| 06:21.46 | Essobi | I'm about to break AGI in two. |
| 06:22.07 | Essobi | GRRRR |
| 06:22.29 | Astrisk-boob | what about security wise <shaZwaz>? |
| 06:23.35 | shaZwaz | I dunno much but there are ways to secure your * installation, check the wiki |
| 06:23.48 | shaZwaz | ~docs |
| 06:23.49 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 06:24.38 | Astrisk-boob | cool thanks for all the help <shaZwaz>... you will prolly see me again :) |
| 06:24.54 | shaZwaz | ~adn |
| 06:24.55 | jbot | rumour has it, adn is the Asterisk Daily News - http://www.sineapps.com/news.php for HTML and http://www.sineapps.com/rssfeed.php for RSS |
| 06:25.23 | *** join/#asterisk justinnnnnn (~dsf@solid.mpa.net.au) |
| 06:26.05 | implicit | sup justinnnnnn |
| 06:27.27 | *** part/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 06:27.44 | shaZwaz | hi implicit |
| 06:27.55 | implicit | hi shaZwaz |
| 06:28.39 | shaZwaz | didn't see much on the channels last days |
| 06:29.04 | implicit | yeah been busy :) |
| 06:29.19 | shaZwaz | halay shumma ? |
| 06:29.54 | implicit | khubam vale khale khaste |
| 06:30.05 | implicit | karam zeyade |
| 06:30.07 | shaZwaz | :) |
| 06:31.24 | implicit | nemidonestam toam farsi balad boodie |
| 06:32.00 | shaZwaz | implicit pm |
| 06:32.17 | implicit | ok |
| 06:32.56 | implicit | i dont have farsi input on here right now so i cant type it into irc |
| 06:33.08 | implicit | and also for some reason my chinese input doesnt work with xchat :-\ |
| 06:33.16 | implicit | stuck to english characters for now heh |
| 06:49.04 | herag | a few months ago, broadvoice required it's users to apply some patch to chan_sip, are those changes now included if I were to upgrade to 1.0.5? (currently using 1.0.2 + bv patch) |
| 06:50.43 | *** join/#asterisk Othello (Othello@nusnet-154-210.dynip.nus.edu.sg) |
| 06:56.20 | *** join/#asterisk elric (~kavit@ppp114-10.static.internode.on.net) |
| 06:56.40 | elric | hey does asterisk support video? what protocol does it use? |
| 06:57.56 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:59.00 | *** join/#asterisk eivindtr (~Eivind@193.91.146.34) |
| 07:09.11 | *** join/#asterisk elric (~kavit@ppp114-10.static.internode.on.net) |
| 07:09.44 | *** join/#asterisk bobr (~bobr@solid.mpa.net.au) |
| 07:10.31 | *** join/#asterisk abbas_ (nidobas@203.81.200.28) |
| 07:11.18 | bobr | nyone know of or can make * send the # key through the system (ie to online banking) but if you press # twice or some other combo then it executes "transfer" |
| 07:11.52 | bobr | atm we can do one or the other but not both so can either set asterisk for our phone banking needs or for transfer needs |
| 07:15.00 | shaZwaz | you can do a number of things |
| 07:15.16 | shaZwaz | like not using t/T in your Dial cmd |
| 07:17.29 | abbas_ | Hi |
| 07:17.48 | abbas_ | where can i get configuration of cisco 5300 with asterisk |
| 07:19.21 | herag | does asterisk not play well with mpg123 .59s? does it have to .59r? |
| 07:19.43 | *** join/#asterisk bobr (~bobr@solid.mpa.net.au) |
| 07:20.58 | djin | as long as it's not 59q |
| 07:21.25 | Mavvie | herag: the rumours are that .59s doesn't work fine. |
| 07:21.37 | djin | and .59s isn't final. |
| 07:21.54 | herag | I see |
| 07:27.03 | *** join/#asterisk gdsm (~gdsm@mk-ns500-1.uk.tiscali.com) |
| 07:27.21 | abbas_ | where can i get configuration of cisco 5300 with asterisk |
| 07:29.31 | *** join/#asterisk gdsm (~gdsm@mk-ns500-1.uk.tiscali.com) |
| 07:43.59 | datareactor | abbas_ this link might help http://lists.digium.com/pipermail/asterisk-users/2004-March/040356.html |
| 07:47.12 | abbas_ | datareactor that seems helpful:) |
| 07:47.42 | abbas_ | but i want to receive calls from as5300 and forward to another termination SIP GW |
| 07:50.48 | Beirdo | hmph |
| 07:51.13 | Beirdo | so it seems my asterisk build doesn't have support for voicemail SQL support :) |
| 07:53.49 | *** join/#asterisk three55ml (~who@cs662589-157.satx.rr.com) |
| 07:56.16 | *** join/#asterisk vmlinux (~dc@wsip-68-15-253-140.dl.dl.cox.net) |
| 07:58.44 | *** part/#asterisk djin (~djin@gridfox.xs4all.nl) |
| 07:59.00 | *** join/#asterisk inspired (mikael@host-81-191-127-166.bluecom.no) |
| 07:59.06 | inspired | yk |
| 08:00.12 | *** join/#asterisk terraHome (~cjs@cs662586-139.satx.rr.com) |
| 08:08.38 | *** join/#asterisk r1 (~erwan@www.thiscow.com) |
| 08:11.25 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 08:22.33 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 08:27.10 | *** join/#asterisk Wireless (~bad@220.233.77.87) |
| 08:27.49 | Wireless | what is the difference between zaptel and zapata packages? |
| 08:28.09 | Wireless | zapata is the driver and zaptel is the user-friendly config tool, correct? |
| 08:30.25 | *** join/#asterisk djin (~marius@62.58.40.196) |
| 08:31.26 | Zeeek | user-friendly config tooL? |
| 08:34.17 | *** join/#asterisk hajekd (~hajekd@21.208.65.212.contactel.net) |
| 08:35.08 | Wireless | sorry, but what's the deal with zaptel/zapata/ztcfg/libpri? |
| 08:36.00 | Zeeek | what is your question? "what's the big deal" doesn't really make clear what you are looking for |
| 08:36.45 | *** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net) |
| 08:37.02 | Wireless | what is the different between zaptel and zapata packages (in debian in particular, but i think other distro also have these two similar packages) |
| 08:37.23 | Wireless | if i have a x100p, are both of these packages needed? |
| 08:37.40 | Zeeek | I have a X100P - zaptel is needed |
| 08:37.45 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 08:37.55 | Zeeek | libpri is for PRI |
| 08:38.11 | Wireless | thanks zeeek, and what does the zapata package do? |
| 08:38.35 | Zeeek | never heard of the zapata package |
| 08:39.03 | Zeeek | where do you see "package" ? in some distro? |
| 08:39.10 | Wireless | yeah, in debian. |
| 08:39.59 | Wireless | the official debian packages only have binaries for zaptel, but source for both zaptel and zaptel. backports has binaries for both. |
| 08:40.33 | Wireless | the zaptel package in debian official is described as "user space util for config" or sth like that. |
| 08:40.45 | Zeeek | never hoid of it |
| 08:41.00 | Zeeek | I downloaded asterisk and zaptel for the make |
| 08:41.06 | Wireless | i see. |
| 08:41.12 | Zeeek | there is a zapata.conf |
| 08:41.31 | Zeeek | and a /etc/zaptel.conf |
| 08:41.33 | {zombie} | Wireless: where did you see this? I see no mention of zapata packages on either packages.debian.org or backports.org |
| 08:41.41 | Wireless | zombie: wait. |
| 08:41.59 | Zeeek | as far as user-friendly... I don't know what that could mean :) |
| 08:42.14 | Wireless | zombie: http://www.backports.org/debian/dists/stable/ - near the end, both zaptel and zapata are listed |
| 08:42.39 | Wireless | so zaptel is the driver, and once installed and configured, asterisk could talk to it? |
| 08:42.57 | Zeeek | Starter tutorial: |
| 08:42.57 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 08:42.57 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 08:42.57 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 08:42.57 | Zeeek | THE reference of the moment: |
| 08:42.57 | Zeeek | http://www.asteriskdocs.org |
| 08:43.06 | {zombie} | zapata contains libzap1 and libzap-dev |
| 08:43.54 | Zeeek | Wireless the articles above will give you a good overview |
| 08:44.05 | *** join/#asterisk schurig (~schurig@p5080A089.dip0.t-ipconnect.de) |
| 08:44.07 | Wireless | zombie: so zapata is just the libraries needed to make the drivers from source? |
| 08:44.11 | {zombie} | right |
| 08:44.29 | Zeeek | and the automated site show the entire process from download to make to use |
| 08:44.46 | Wireless | zeek: reading. thanks. - couldn't get the drivers to compile on debian so moved to asterisk@home - now want to move back but using binary packages |
| 08:45.45 | {zombie} | if you want to create zaptel kernel drives the "debian way"(tm) then you apt-get install zaptel-source, and run "make-kpkg modules_image" from your kernel dir |
| 08:46.54 | Wireless | the debian way gave me all sorts of dependency problems - so i'm sticking to binaries for the moment. :-) |
| 08:47.22 | gdsm | debian unstable is fine |
| 08:47.27 | gdsm | or at least for me |
| 08:47.35 | {zombie} | yeah, but if you want to use the zaptel cards you will need kernel drivers |
| 08:47.59 | gdsm | agreed, but you can compile those from the zaptel source and just put them in |
| 08:48.22 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 08:48.28 | RoyK | ka-pling |
| 08:48.29 | pif | hi, is there a digium reseller in france? |
| 08:48.30 | Wireless | zombie/gdsm - but wouldn't the zaptel binaries do just that? (sorry..., new to linux) |
| 08:48.38 | pif | I am looking for a Wildcard TE410P |
| 08:48.40 | Zeeek | elonex pif |
| 08:48.41 | *** join/#asterisk snewpy (~markl@203-217-69-209.dyn.iinet.net.au) |
| 08:48.47 | {zombie} | Wireless: zaptel binaries are for configuring the card. zaptel drivers are for talking to the card |
| 08:48.49 | pif | thanks |
| 08:48.49 | {zombie} | you need both |
| 08:49.02 | Zeeek | pif oops not elonex |
| 08:49.13 | Zeeek | eikonex but the site isn't up |
| 08:49.22 | pif | ok |
| 08:49.24 | Wireless | zombie: thanks! i see now... so ideally i could get the kernel drivers first THEN install asterisk? |
| 08:49.25 | Zeeek | ah, here we are: https://shop.eikonex.net/catalog/default.php |
| 08:49.39 | Zeeek | I've ordered from them twice - they're ok |
| 08:49.51 | gdsm | I use zaptel debian pkg for configuring card, but get the zaptel driverer from src. |
| 08:49.59 | {zombie} | Wireless: doesn't matter what order you do it in, but you won't be able to talk to your digium cards in asterisk until you install the modules :) |
| 08:50.09 | pif | Zeeek: thanks |
| 08:50.18 | Zeeek | which reminds me....... wasim wasim wasim ? |
| 08:50.29 | {zombie} | gdsm: I compiled them the "debian" way, not that hard really |
| 08:50.30 | Zeeek | ~seen wasim |
| 08:50.33 | jbot | wasim is currently on #asterisk (1d 11h 38m 34s). Has said a total of 63 messages. Is idling for 9h 28m 5s |
| 08:50.40 | gdsm | zombie me too |
| 08:50.48 | {zombie} | and makes a nice .deb you can scp around with your kernels |
| 08:50.51 | Wireless | zombie/gdsm - thanks. |
| 08:51.28 | Wireless | will give it a shot. |
| 08:51.44 | gdsm | zombie, okay, half way there, make-kpkg for the main kernel, but tag the zaptel on extra |
| 08:52.18 | *** join/#asterisk mAsH` (~mAsH@81.208.97.122) |
| 08:52.22 | mAsH` | hi all |
| 08:53.23 | *** join/#asterisk PakiPenguin (~uppal@202.176.254.1) |
| 08:53.40 | Wireless | zombie/gdsm - one more q - will apt-get zaptel-source also get all missing dependencies for me to make sure the make is successful? |
| 08:53.47 | PakiPenguin | hello there , can anyone point me how can i put a failsafe , like if one peer's busy , dial the call thru the other peer? |
| 08:54.09 | mAsH` | i get this errore when i try to make an isdn call |
| 08:54.35 | mAsH` | WARNING[2504]: chan_sip.c:1829 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 8/4) |
| 08:55.15 | Zeeek | PakiP you can just put them in order Dial after DIal |
| 08:56.09 | Zeeek | although come to think that may be if they don't answer |
| 08:58.28 | {zombie} | Wireless: should do, but I can't confirm that, since I compiled it on my development machine which has pretty much everything installed (development wise) already |
| 08:58.50 | Zeeek | mAsH looks like codec mismatch |
| 09:03.36 | PakiPenguin | Zeeek: chanisavail? |
| 09:03.57 | Zeeek | yeah that would probably be better |
| 09:04.22 | Zeeek | but you say "peer is busy" - you are calling a phone or a provider? |
| 09:04.49 | RoyK | eeeek! zeeek! |
| 09:04.50 | Zeeek | if its busy doesn't it go to +101 |
| 09:05.07 | Zeeek | been so long since I set that stuff up I can't remember |
| 09:05.24 | RoyK | Zeeek: ikt tries to jump to +101. if that fails, it continues +1 iirc |
| 09:06.47 | Zeeek | ah RoyK I seem to remember something of that ilk - so - my answer is right - PakiPenguin can just put the fallthrough dials one after another |
| 09:07.08 | Zeeek | btw Voicepulse has 2 servers so IIRC that's exactly what you do for them |
| 09:07.08 | *** join/#asterisk muesli (~muesli@mail.muehlhaeuser.de) |
| 09:09.01 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 09:09.20 | Delvar | morning all |
| 09:09.37 | *** join/#asterisk dnc (~duncan@213.244.225.42) |
| 09:11.02 | Zeeek | drum roll........ |
| 09:11.08 | Zeeek | CYMBAL CRASH! |
| 09:11.13 | Zeeek | good morning |
| 09:19.32 | Zeeek | . |
| 09:19.59 | *** join/#asterisk talkwebhosts (~admin@c-24-130-183-95.we.client2.attbi.com) |
| 09:22.00 | *** join/#asterisk meppl (~mephisto@p548530F5.dip.t-dialin.net) |
| 09:22.07 | meppl | guten morgen |
| 09:29.31 | RoyK | guten morgen, meppl |
| 09:29.37 | PakiPenguin | Zeeek: ever accepted calls from as5300 --> cisco |
| 09:29.40 | PakiPenguin | its urgent , please |
| 09:29.53 | Zeeek | not me! |
| 09:31.01 | *** join/#asterisk hajekd (~hajekd@mail.idoox.com) |
| 09:33.12 | RoyK | PakiPenguin: I have an as5300 in the lab |
| 09:33.15 | RoyK | trying out t.38.... |
| 09:39.21 | BoRiS | Wish T.38 was working with asterisk |
| 09:39.32 | RoyK | BoRiS: add another $1000 to the bounty |
| 09:39.38 | RoyK | perhaps it helps |
| 09:39.55 | meppl | good morning royk |
| 09:40.06 | RoyK | morning |
| 09:40.07 | BoRiS | It was already up to like $8000 and still no one wants to code it :( |
| 09:40.17 | RoyK | BoRiS: coppice is working on it |
| 09:40.27 | RoyK | BoRiS: _when_ was it 8k? |
| 09:40.34 | RoyK | I've only seen the official bounty on 3k |
| 09:40.54 | BoRiS | Read the "comments" page, some guy added an additional 5k |
| 09:41.01 | RoyK | yeah |
| 09:41.04 | RoyK | on the mailing list |
| 09:41.10 | RoyK | but he never repeated it |
| 09:41.18 | RoyK | so that's probably just bladder |
| 09:41.24 | BoRiS | It was on the voip-info.org site |
| 09:41.35 | RoyK | only on the comments page |
| 09:41.47 | RoyK | and it was someone else that added that comment. not the guy posting the email on the list |
| 09:41.54 | RoyK | I tried emailing him, no answer |
| 09:42.02 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 09:42.15 | PakiPenguin | RoyK: sorry was away , the problem is , i have a remote as5300 , from where calls are orignating from pstn side , my * server is in the middle , the call comes into my * server and i forward it from my * to another GSM voip gateway i have , the call passes through fine , but i am not able to hear / both sides cannot hear each other |
| 09:42.20 | BoRiS | Any idea how far coppice has gotten or does he not care? |
| 09:43.15 | RoyK | BoRiS: I think he has some up and going. I sent him an ATA from Yoda.com.tw, but that seemd to not support fax at all |
| 09:43.16 | RoyK | :P |
| 09:43.37 | PakiPenguin | RoyK: i really am stuck with this , can you help me out somehow? |
| 09:43.57 | PakiPenguin | RoyK: can i talk in pvt. with you? |
| 09:44.29 | RoyK | PakiPenguin: tried forcing gsm codec out from asterisk to the gateway? |
| 09:45.26 | vmlinux | hmm.. I blew something up.. getting this error spammed when I start asterisk now: "Ouch ... error while writing audio data: : Broken pipe" |
| 09:46.03 | RoyK | PakiPenguin: sound problems are usually just codec messup |
| 09:46.09 | vmlinux | oh ok |
| 09:46.15 | PakiPenguin | yeah , its g729 all the way around |
| 09:46.23 | RoyK | PakiPenguin: try to ethereal some data and see if it all matches |
| 09:46.30 | vmlinux | I just compiled and installed mpg123 |
| 09:46.31 | RoyK | also, try to sip debug |
| 09:47.47 | PakiPenguin | RoyK: i am trying to |
| 09:48.22 | PakiPenguin | http://pastebin.ca/5856 < -- channels |
| 09:49.28 | PakiPenguin | any idea / pointer? |
| 09:50.16 | RoyK | well |
| 09:50.20 | *** join/#asterisk arseniy_chernov (~ars@84.204.22.118) |
| 09:50.29 | RoyK | it'll be better if you sent the whole debug |
| 09:51.00 | shaZwaz | howdy RoyK |
| 09:51.55 | arseniy_chernov | hello all, nice to be here. can anyone advise an URL of pan-european SIP operator that can terminate calls in major cities, if such exists? |
| 09:51.55 | {-award-} | oh btw ... where do i send patches? ^^ |
| 09:53.08 | *** join/#asterisk Fabe_ (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 09:55.07 | *** join/#asterisk abbas_ (nidobas@203.81.200.28) |
| 10:01.34 | PakiPenguin | RoyK: still around? |
| 10:02.16 | PakiPenguin | http://pastebin.ca/5858 < -- check this guys |
| 10:03.21 | Delvar | iv got an ADSL connection with 16 static ip addresses, when it dials up it seems to get the network address as the router ip address, in this case 217.13.138.32 netmask 255.255.255.240, shouldnt the router ip be .33 |
| 10:04.40 | shaZwaz | PakiPenguin: can u hear it at * ? |
| 10:04.50 | shaZwaz | either end ? |
| 10:05.25 | PakiPenguin | shaZwaz: nopes , no one can hear :( |
| 10:06.19 | PakiPenguin | my setup is like this pstn --->as5300--->my * server ---> my voip gsm gateway |
| 10:07.11 | shaZwaz | any NAT ? |
| 10:07.24 | PakiPenguin | so the debug shows call comes in , it forwards the call to gsm voip gateway ,which rings the number called too , when picked up , it just show gives nothing , no voice |
| 10:07.34 | PakiPenguin | nope , everything's on open ip |
| 10:07.35 | PakiPenguin | s |
| 10:08.03 | shaZwaz | can u ring ur * exts ? |
| 10:08.07 | shaZwaz | and hear anything ? |
| 10:08.20 | *** join/#asterisk zoa (~zoa@pirus.securax.be) |
| 10:08.36 | PakiPenguin | shaZwaz: remote as5300's context , just receives the call and forwards it to the voip gateway |
| 10:09.47 | shaZwaz | isn't * doing the forwarding ? |
| 10:10.15 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 10:10.25 | PakiPenguin | yes it is |
| 10:10.28 | RoyK | PakiPenguin: sorry. I really don't see what's the problem. do you finalize setting up the call here? |
| 10:10.32 | shaZwaz | you sh'd ring one of your ext if nay and check if the as5300-* call is ok ? |
| 10:10.36 | PakiPenguin | lemme post the configs too |
| 10:11.14 | PakiPenguin | * to --> VOIP GSM gateway call is okay , so its just the as5300 and the * now |
| 10:12.51 | shaZwaz | can u hear the voice mail prompts ? |
| 10:13.45 | PakiPenguin | http://pastebin.ca/5860 < -- this is all the config |
| 10:13.48 | PakiPenguin | on my * server |
| 10:17.47 | PakiPenguin | http://pastebin.ca/5861 < -- remote as5300 config for us |
| 10:17.54 | PakiPenguin | anything guys? |
| 10:18.58 | shaZwaz | sip debug |
| 10:19.02 | shaZwaz | check the ports |
| 10:19.52 | PakiPenguin | http://pastebin.ca/5858 < -- sip debug |
| 10:22.48 | *** join/#asterisk vagwin (~vagwin@mk-ns500-1.uk.tiscali.com) |
| 10:36.38 | *** part/#asterisk dnc (~duncan@213.244.225.42) |
| 10:36.44 | *** join/#asterisk expressfone1 (~expressfo@62-15-97-163.inversas.jazztel.es) |
| 10:36.55 | expressfone1 | Cuba hot route 9 hr daily(3:00 p.m -> 00:00 a.m (Eastern time = GMT-5)) . 1 E1. billing 30/6 best quality. only SIP Protocol. ASR 50%, PDD 7sec. Very good quality. |
| 10:37.15 | expressfone1 | or IAX2, no h323 |
| 10:37.22 | expressfone1 | TDM quality |
| 10:37.28 | shaZwaz | PakiPenguin: ignorertpmap? |
| 10:37.29 | elric | anyone know of a *nix IAX2 softphone, apart from iaxComm ? |
| 10:37.46 | shaZwaz | ifurefly |
| 10:38.07 | shaZwaz | expressfone1: FireFly |
| 10:42.41 | expressfone1 | haZwaz> thx |
| 10:44.08 | *** join/#asterisk Fanguin (~Fanguin@p50818411.dip0.t-ipconnect.de) |
| 10:44.29 | vmlinux | woot, got asterisk working. Now to configure the beast :P |
| 10:44.54 | shaZwaz | R.I.P |
| 10:45.21 | wasim | shaZwaz: don't say that, the RIAA may get you |
| 10:45.23 | hajekd | Do you know if LDAPGet can handle LDAP connections via ssl? |
| 10:45.35 | hajekd | looks like it doesn't |
| 10:45.41 | shaZwaz | haha |
| 10:46.00 | shaZwaz | hello wasim |
| 10:46.14 | wasim | heya shaZwaz, any projections for the upcoming series? |
| 10:46.31 | shaZwaz | tell me how it was :) |
| 10:46.39 | shaZwaz | can't take any more |
| 10:47.52 | shaZwaz | can't even get some decent performance |
| 10:48.22 | abbas_ | Salam Wasim |
| 10:48.53 | abbas_ | i am gonnaa call u :) need some help |
| 10:49.01 | *** join/#asterisk netsurfer (~netsurfer@dreambox.myvnc.com) |
| 10:49.20 | RoyK | #asterisk is being invaded by .pk! help! |
| 10:49.54 | netsurfer | they have phonelines in .pk? :D |
| 10:49.56 | shaZwaz | mohaha |
| 10:50.44 | Zeeek | wasim.... tell me.... |
| 10:51.14 | Zeeek | when does the ship leave and when does it arrive? |
| 10:58.03 | abbas_ | Zeeek we are using * betweeen a cisco AS5300 and a GSM Gw call gets completed but no voice |
| 10:58.11 | zoa | and you are using h323 |
| 10:58.13 | zoa | good luck |
| 10:58.24 | RoyK | ~h323? |
| 10:58.25 | jbot | [h323] evil! Don't ask about h323 here. Ask JerJer if you need to bug someone. He says it works, but others don't. |
| 10:58.29 | abbas_ | if we originate the call from a SIP UA it gets completed and has 2way voice |
| 10:59.12 | abbas_ | can u help in it? |
| 11:00.27 | *** join/#asterisk beto75 (~ha@201.128.177.84) |
| 11:00.33 | beto75 | hello guys |
| 11:00.50 | beto75 | excuse me at 5:00 am I'm just a little to get nuts |
| 11:01.04 | beto75 | if I try to call meetme I get this: |
| 11:01.34 | beto75 | Feb 15 05:01:27 WARNING[1570]: file.c:790 ast_streamfile: Unable to open conf-onlyperson (format g729): No such file or directory |
| 11:01.38 | nitram | are there any special config options in zapata.conf for asterisk to talk to a siemens hipath in nt-ptp mode? |
| 11:01.54 | beto75 | please help |
| 11:02.13 | *** join/#asterisk cjk (~cjk@80.92.75.91) |
| 11:02.40 | cjk | hi, what does the src field in the cdr records represent, how is it generated? |
| 11:03.34 | shaZwaz | beto75: dont use any file extenions |
| 11:03.38 | shaZwaz | :) |
| 11:04.49 | beto75 | shazwaz how do I "dont use it" |
| 11:04.50 | shaZwaz | use wav formats if u can |
| 11:05.04 | beto75 | but its the meetme |
| 11:05.15 | beto75 | meetme welcome recording |
| 11:05.20 | beto75 | (only person) |
| 11:05.25 | expressfone1 | any one looking for terminate at +53 ???? .50usd, billing step 30/6, PDD 7sec, ASR 50%, TDM Route |
| 11:05.44 | beto75 | how do I use the wav? |
| 11:06.00 | shaZwaz | oops |
| 11:06.32 | beto75 | Feb 15 05:06:05 NOTICE[1570]: channel.c:1734 ast_set_write_format: Unable to find a path from gsm to g729 |
| 11:07.42 | beto75 | do you think re build asterisk with an older tar (right now is CVS) |
| 11:08.09 | shaZwaz | stable ? |
| 11:08.39 | shaZwaz | do u have a zaptel card ? |
| 11:09.15 | beto75 | I am trying to use ztdummy no zap |
| 11:09.45 | shaZwaz | do u have ztdummy loaded ? |
| 11:09.48 | beto75 | shazwaz how do I CVS stable? |
| 11:09.49 | beto75 | yup |
| 11:09.55 | beto75 | modprobe ztdummy |
| 11:10.17 | shaZwaz | do u have all the sound files ? |
| 11:10.46 | beto75 | yes |
| 11:10.50 | beto75 | I have those |
| 11:12.11 | shaZwaz | what are using at your end a sip phone ? |
| 11:12.15 | *** join/#asterisk ars_ (~ars@84.204.22.118) |
| 11:12.42 | beto75 | xten and a snom 200 |
| 11:12.46 | beto75 | both says the same |
| 11:12.54 | beto75 | xten pro |
| 11:13.18 | shaZwaz | use allow=all in sip .conf |
| 11:14.08 | beto75 | ok let me do that |
| 11:16.57 | ars_ | can anyone advise a voip operator able to terminate calls in European countries? |
| 11:20.39 | shaZwaz | ars_: u can find a bunch at http://www.voip-info.org/tiki-index.php?page=VOIP%20Service%20Providers%20by%20Country |
| 11:20.52 | netsurfer | ars_ - uptime hasnt been great recently, but u could try sipgate.de |
| 11:24.01 | *** join/#asterisk nazgool (~nazgoool@gatekeeper-e0.twc.de) |
| 11:24.05 | nazgool | hi |
| 11:25.30 | *** join/#asterisk makkia (~pietro@nat.xsec.it) |
| 11:27.17 | *** join/#asterisk Savage-S (~savage@c514701e0.cable.wanadoo.nl) |
| 11:27.30 | *** part/#asterisk Savage-S (~savage@c514701e0.cable.wanadoo.nl) |
| 11:29.17 | beto75 | Shazwaz!! you are a genius ,, thank you |
| 11:29.30 | shaZwaz | ;) |
| 11:29.48 | beto75 | but why in other asterisk boxes I have disallow all allow g729 and all this stuff work fine |
| 11:30.17 | beto75 | DAMN for me is not good time to ask silly questions I need to rest a while |
| 11:30.23 | beto75 | bye for now |
| 11:30.23 | shaZwaz | you have a g729 license ? |
| 11:30.30 | beto75 | yes I do |
| 11:30.42 | beto75 | see ya tomorrow |
| 11:30.44 | *** part/#asterisk beto75 (~ha@201.128.177.84) |
| 11:39.06 | jerlique | I'm investiagting starting a voip service for general public access (through subscription). Is asterisk the platform for this, or are there better choices? |
| 11:39.45 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 11:40.22 | ars_ | thanks shazwaz, netsurfer. i'm browsing. |
| 11:56.49 | *** join/#asterisk [ro]nic3try (~iancu@81.181.199.39) |
| 11:56.56 | [ro]nic3try | re all |
| 11:57.24 | [ro]nic3try | how do I manage an ip call ? |
| 11:58.56 | [ro]nic3try | pls :( |
| 11:59.06 | zoa | jerlique: asterisk could do the job |
| 11:59.13 | zoa | as im using it also for that purpose |
| 12:00.03 | [ro]nic3try | yes, but what do i need to config ? .. docs or something |
| 12:01.12 | [ro]nic3try | i don't understand how to use the incoming ip , whici is from another server |
| 12:02.06 | [ro]nic3try | i'm tryng to connect from a SER server to my asterisk serever.. without register.. just using the ip |
| 12:03.41 | *** join/#asterisk ozJames79 (~james@CPE20320889-1842-1.gex.ncable.net.au) |
| 12:03.51 | ozJames79 | anyone here using fwdout? |
| 12:08.28 | Mavvie | [ro]nic3try: try extensions.conf |
| 12:09.33 | [ro]nic3try | yes, but how do i use the ip ?? |
| 12:09.50 | Mavvie | read the samples in the extensions.conf |
| 12:13.04 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 12:13.49 | *** join/#asterisk shaZwaz (~chatzilla@203.81.196.167) |
| 12:17.45 | Zeeek | fwdout is bellster? |
| 12:17.48 | Zeeek | was? |
| 12:19.13 | jerlique | zoa: how do you find it? |
| 12:21.57 | netsurfer | Unable to create channel of type 'Zap' (cause 0) <-- what is cause 0 ? |
| 12:22.29 | RoyK | free((void *)NULL) == cause 0 |
| 12:22.30 | RoyK | :P |
| 12:22.42 | Zeeek | nah! |
| 12:23.14 | RoyK | Zeeek: no? |
| 12:23.48 | [ro]nic3try | i send an invite to 192.168.1.82 |
| 12:23.58 | netsurfer | uhmm... cause 0 is whaaaaaaaaat |
| 12:24.01 | netsurfer | lol |
| 12:24.06 | Zeeek | ~lart RoyK |
| 12:24.20 | Zeeek | oh my. I didn't mean to... geee... |
| 12:24.33 | [ro]nic3try | the i should have someting like exten =>192.168.1.82,1,answer ? |
| 12:24.40 | Zeeek | cause 0 is like "whaaazaaaat?" |
| 12:25.33 | Zeeek | I don need no stinking packages ! |
| 12:26.12 | netsurfer | #define AST_CAUSE_NOTDEFINED 0 <-- baah |
| 12:28.06 | Zeeek | mov a,1 ; mov 1 to a |
| 12:28.30 | netsurfer | asm blows |
| 12:28.48 | Zeeek | asymmetric system management ? |
| 12:28.50 | [ro]nic3try | what should i set.. so my menus to work ?(if i press a button nothing happen) |
| 12:30.22 | [ro]nic3try | asterisk doesnt read my keys after answering |
| 12:30.50 | Zeeek | post your dialplan to pastebin |
| 12:30.59 | Zeeek | just the applicable section |
| 12:31.07 | Delvar | [ro]nic3try: could it be not picking up your DTMF? |
| 12:31.56 | [ro]nic3try | uff .. and how do i fix that ? |
| 12:32.02 | Zeeek | hopefully he had time to wash too, not just sleep |
| 12:32.18 | Delvar | i assume you are using a sip phone into asterisk? |
| 12:32.29 | [ro]nic3try | a HT 286 |
| 12:32.45 | Delvar | you need to log into teh HT and set DETMF mode to 'rfc2833' |
| 12:33.04 | Delvar | then in sip.conf in te sip entity add 'dtmfmode=rfc2833' |
| 12:33.14 | Delvar | do a reaload and wala it works |
| 12:33.21 | Delvar | reload* |
| 12:34.05 | BoRiS | Updated your firmware |
| 12:34.07 | BoRiS | :) |
| 12:34.21 | Delvar | thats the next step |
| 12:34.29 | Delvar | shh im trying to sound smart |
| 12:34.35 | BoRiS | lol |
| 12:35.31 | Zeeek | Delvar hmmmm, sometime INFO is better - depends on codec |
| 12:36.07 | Delvar | realy?.. iv not had any problems with RFC... supose it shouldnt matter too much |
| 12:36.32 | [ro]nic3try | i use info |
| 12:36.56 | Delvar | both should work as long as its the same in teh client and sip.conf |
| 12:37.23 | Zeeek | info should work with most codecs - there are issues with RFC on GS |
| 12:37.35 | Delvar | GSM? |
| 12:37.55 | Zeeek | could nbe - can't remember |
| 12:38.19 | Delvar | ill have to try it with all of them and make sure |
| 12:38.22 | Zeeek | why? for that is reasin there? |
| 12:38.28 | Delvar | i keep telling ppl to use rfc... |
| 12:38.43 | Zeeek | now you've sabotaged half the planet |
| 12:38.48 | [ro]nic3try | i put rfc.. still doesnt read my key |
| 12:38.54 | Delvar | not gona look good if tell them to use rfc and x codec then find it fekers up |
| 12:39.01 | [ro]nic3try | i set the HT also |
| 12:39.16 | Delvar | what firmware is your ht on? |
| 12:39.21 | Delvar | 1.0.5.16 ? |
| 12:39.35 | Zeeek | me use 5.11 |
| 12:39.44 | Delvar | im on 18 |
| 12:39.50 | [ro]nic3try | Program-- 1.0.5.18 Bootloader-- 1.0.0.21 HTML-- 1.0.0.42 VOC-- 1.0.0.7 |
| 12:39.51 | Zeeek | ah you got it working? |
| 12:40.07 | Delvar | 22 and 18 both work prety well |
| 12:40.26 | BoRiS | yeah.. 22 works pretty well |
| 12:42.58 | [ro]nic3try | should i use info on both server and HT ? |
| 12:42.58 | Delvar | yes |
| 12:42.58 | BoRiS | info |
| 12:42.58 | Delvar | Zeeek seems to think its better and who am i to argue with a guy with such a weird name? |
| 12:43.15 | *** join/#asterisk negativecreep (~yama@202.147.174.97) |
| 12:43.28 | Zeeek | hoo hoo heee |
| 12:43.28 | [ro]nic3try | yap... your right.. now seems to work . thx :) |
| 12:43.41 | negativecreep | hi all |
| 12:43.47 | Zeeek | I use whatever works - knowing why it works would be a bonus |
| 12:43.55 | negativecreep | hi Zeeek |
| 12:44.09 | Zeeek | hi positiveSPitirtualLeader |
| 12:44.11 | Delvar | lol |
| 12:44.22 | negativecreep | LOL |
| 12:44.33 | Zeeek | so, what's new in the anti-universe? |
| 12:44.50 | negativecreep | Zeeek: created a new antimatter variant.. |
| 12:45.00 | Zeeek | I stared at one line of odbc code for one hour this morning. Finally I changed the name of one variable and it works |
| 12:45.22 | Zeeek | but why do they insist on beginning array indices at 1? jeeze |
| 12:45.24 | negativecreep | i am trying to configure sip calling on my asterisk server but cant get it done.. |
| 12:45.41 | Zeeek | well positiveEtcJokeIsOver, tells us about it |
| 12:45.58 | Zeeek | where are you runniung into the brick wall? |
| 12:46.02 | negativecreep | Feb 15 17:41:16 WARNING[21244]: chan_sip.c:751 retrans_pkt: Maximum retries exceeded on call 243776234@192.168.16.2 for seqno 3851 (Non-critical Response) |
| 12:46.15 | Zeeek | ah. thtat. Very, very bad. |
| 12:46.15 | negativecreep | Zeeek: thats the error i get.. |
| 12:46.20 | Zeeek | horrible. |
| 12:46.25 | negativecreep | Zeeek: whats the fix? |
| 12:46.45 | Zeeek | first you have to find out why asterisk can't find the sip unit it wants to call |
| 12:46.57 | expressfone1 | any one looking for terminate at +53 ???? .50usd, billing step 30/6, PDD 7sec, ASR 50%, TDM Route |
| 12:47.02 | Zeeek | one reson, when you lose connection to the net |
| 12:47.19 | negativecreep | Zeeek: i am trying to test it on the local lan at the moment |
| 12:47.33 | Zeeek | and the phone is registered? |
| 12:47.38 | negativecreep | Zeeek: it sure is.. |
| 12:47.38 | Zeeek | sip show peers |
| 12:47.44 | *** join/#asterisk robb_ (~robb@matrix.netsoc.tcd.ie) |
| 12:47.46 | negativecreep | sip show peers...says its registered.. |
| 12:47.58 | Zeeek | and the phone can call somthing (echo test?) |
| 12:48.10 | negativecreep | havent tested that..i am just beginning with * |
| 12:49.36 | Zeeek | here is some background reading: |
| 12:49.37 | Zeeek | Starter tutorial: |
| 12:49.37 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 12:49.37 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 12:49.37 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 12:49.37 | Zeeek | THE reference of the moment: |
| 12:49.38 | Zeeek | http://www.asteriskdocs.org |
| 12:50.49 | *** join/#asterisk riksta (~rick@81-178-212-128.dsl.pipex.com) |
| 12:51.24 | negativecreep | Zeeek: i have read all of em.. |
| 12:51.33 | negativecreep | basically i am followin the onlamp tutorial.. |
| 12:51.37 | negativecreep | doesnt seem to work properly for me.. |
| 12:52.29 | Zeeek | where does it go bad? There may be typos |
| 12:53.55 | Zeeek | negatory why don't you pastebin the relevant sections of sip.conf and extensions |
| 12:54.09 | negativecreep | hang on Zeeek |
| 12:54.12 | Zeeek | hundreds of pairs of eyes will then scrutinize, debug, and invoice |
| 12:54.20 | negativecreep | k |
| 12:54.25 | Zeeek | and possibly laugh or poke fun |
| 12:54.35 | Zeeek | but likely help you in the end |
| 12:55.02 | negativecreep | Zeeek: let me try the automated.it link..if it still doesnt work, then i shall ask you guys again..ok |
| 12:55.15 | negativecreep | Zeeek: it looks pretty interesting.. |
| 12:55.21 | Zeeek | that is is damn good guide, that |
| 12:55.21 | negativecreep | Zeeek: thnx for pointing that out.. |
| 12:57.31 | *** join/#asterisk meppl (~mephisto@p548530F5.dip.t-dialin.net) |
| 13:00.38 | modulus_ | word |
| 13:02.10 | Zeeek | byte! |
| 13:03.35 | modulus_ | nibble! |
| 13:05.21 | netsurfer | in debug I see this: Zap/g1/phonenumber|40|t|T <-- thats right, isnt it ? |
| 13:07.32 | Essobi | Anyone got an idea why I can't call monitor in an AGI with more then one paramater? |
| 13:10.48 | modulus_ | essobi, agi doesn't allow it? |
| 13:10.56 | *** join/#asterisk riksta (~rick@81-178-248-194.dsl.pipex.com) |
| 13:11.08 | *** join/#asterisk _Brian (brian@unix01.voicenet.com) |
| 13:11.11 | _Brian | morning all |
| 13:11.51 | Zeeek | bit |
| 13:12.10 | *** join/#asterisk Ubuz (~momo@DSL217-132-49-219.bb.netvision.net.il) |
| 13:12.18 | Zeeek | netsurfer no, not T|T tT |
| 13:12.29 | modulus_ | gawble |
| 13:17.23 | Essobi | modulus_ Umm.. that's retarded. When I say more then 1.. I mean I can't sent... "wav,/tmp/testfilename,mb" to the montior app |
| 13:17.44 | Essobi | which is dumb.. but I can send "wav" |
| 13:17.51 | Essobi | by it's self. |
| 13:18.10 | modulus_ | the only real way to know is look at the code |
| 13:18.32 | modulus_ | or spend endless hours looking online |
| 13:19.08 | *** join/#asterisk k0ga (~roberto@62-15-230-33.inversas.jazztel.es) |
| 13:19.48 | Essobi | The only REAL way to know it to talk to the AGI directly. I did. It doesn't work. |
| 13:19.54 | Essobi | :) |
| 13:20.12 | Essobi | I really don't want to write a patch but I guess I'll have to. :\ |
| 13:20.28 | *** join/#asterisk ManxPwr (~eric@dsl-209-205-172-111.i-55.com) |
| 13:20.34 | ManxPwr | ~docs |
| 13:20.35 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 13:20.35 | netsurfer | Zeeek - didnt make a difference.. still getting same error :o\ |
| 13:21.01 | Essobi | ManxPwr Jbot. |
| 13:21.11 | Essobi | Morning. |
| 13:21.25 | modulus_ | jbot weather klax |
| 13:21.28 | ManxPwr | morning are evil |
| 13:21.55 | RoyK | 'morning' in pluro? |
| 13:22.48 | Nugget | it's too early to think straight. |
| 13:22.51 | Essobi | jbot weather SDF |
| 13:22.59 | Essobi | :P |
| 13:23.12 | *** join/#asterisk DEVILoper (~x@202.5.145.50) |
| 13:23.24 | DEVILoper | Hi All |
| 13:23.42 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 13:23.49 | Essobi | Someone want to test this out for me and have an AGI handy? |
| 13:23.51 | DEVILoper | i wanna integrate my External GSM modem with asterisk any help ? |
| 13:23.52 | *** join/#asterisk jetscreamer (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
| 13:25.59 | DEVILoper | anybody home ? |
| 13:26.04 | Nugget | sure. |
| 13:26.09 | Zeeek | who is it? |
| 13:26.12 | Nugget | what is a external gsm modem? |
| 13:26.29 | Essobi | <PROTECTED> |
| 13:26.35 | Essobi | <PROTECTED> |
| 13:26.56 | Essobi | always works returning 0. Anyone test that for me? I have to head to work, BBIAF. |
| 13:26.59 | bjohnson | I suspect an external GSM modem is designed for data transfers .. not voip |
| 13:26.59 | Zeeek | can't you send your own delimited string and parse it on the other side? |
| 13:27.34 | Essobi | Zeeek IT's AGI killing it. |
| 13:27.37 | Zeeek | oh - I see |
| 13:27.39 | gdsm | anyone got gnugk talking with asterisk on the same box successfully? |
| 13:28.10 | Essobi | even print "EXEC Monitor wav,/tmp/testname "; fails.. |
| 13:28.12 | ManxPwr | Essobi, Check carefully the options for monitor. I think you need a : in there somewhere. |
| 13:35.33 | *** join/#asterisk heison (~heison@gw-yyz.somanetworks.com) |
| 13:35.42 | *** join/#asterisk k0ga (~roberto@62-15-230-33.inversas.jazztel.es) |
| 13:40.20 | *** join/#asterisk sp2 (~rvramos@203.131.113.109) |
| 13:41.05 | *** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net) |
| 13:41.14 | *** join/#asterisk Dabba (dabba@matrix.lgw.ip6net.net) |
| 13:41.21 | DEVILoper | nugget :A GSM modem can be an external modem device, such as the Wavecom FASTRACK Modem. Insert a GSM SIM card into this modem, and connect the modem to an available serial port on your computer. |
| 13:41.29 | negativecreep | Zeeek: done now... |
| 13:41.32 | negativecreep | Zeeek: its working fine.. |
| 13:41.35 | negativecreep | :) |
| 13:42.10 | sp2 | hi everybody |
| 13:42.13 | DEVILoper | NUGGET:A GSM modem could also be a standard GSM mobile phone with the appropriate cable and software driver to connect to a serial port on your computer. Phones such as the Nokia 7110 with a DLR-3 cable, or various Ericsson phones, are often used for this purpose. |
| 13:42.37 | sp2 | wasim: hi this is my new nick and will always use this from now on (breakpoint_bgo) |
| 13:45.50 | netsurfer | baah |
| 13:46.02 | netsurfer | stupid mobo |
| 13:47.05 | DEVILoper | wasim: u there |
| 13:47.11 | Dabba | can anyone tell me how to force the order * looks at the context |
| 13:47.45 | ManxPwr | Dabba, There isn't a lot you can do to force it. But you almost never need to do yhay. |
| 13:47.48 | ManxPwr | that |
| 13:47.56 | Dabba | >ManxPwr http://pastebin.ca/5808 |
| 13:48.34 | Dabba | >ManxPwr it ignores the bit about handle texts |
| 13:49.55 | Dabba | i.e doesnt deal with sms properly, it sends it on as voice even though the clid is defined |
| 13:50.20 | ManxPwr | Dabba, Why are you using _X.? Do you hate your user? |
| 13:50.42 | Zeeek | hey negative - cool! |
| 13:50.47 | tzanger | ManxPwr: hahaha |
| 13:50.54 | ManxPwr | Dabba, Asterisk will use the most specific match. |
| 13:51.16 | ManxPwr | Dabba, using . in patterns can cause unexpected issues. |
| 13:51.24 | ManxPwr | don't use it if you don't have to |
| 13:51.24 | Dabba | so are you saying put 870111/08005875290 blah |
| 13:51.55 | tzanger | I think he's suggesting rethinking your current dialplan to minimize the use of '.' |
| 13:52.12 | ManxPwr | Dabba, Also put a NoOp(${CALLERIDNUM} somewhere in there so you can confirm that the callerid you think is coming in is actually what you are getting |
| 13:52.49 | Dabba | >thanks Manx i tried a capi debug but so much detail flys by and crashes * |
| 13:53.09 | sp2 | manxpwr: what is really being done when you put NoOp($CALLERIDNUM})? |
| 13:53.09 | *** part/#asterisk [ro]nic3try (~iancu@81.181.199.39) |
| 13:53.45 | ManxPwr | sp2, nothing whatsoever except if you look at the CLI you will see the current value of ${CALLERIDNUM |
| 13:54.02 | ManxPwr | NoOp is terribly useful for troubleshooting |
| 13:54.08 | *** join/#asterisk Conductor (~thomas@62.8.240.132) |
| 13:54.38 | ManxPwr | NoOp is great for checking the contents of variables. It's also used as a placeholder for priorities in the dialplan |
| 13:55.05 | sp2 | manxpwr, tnx for that great tip |
| 13:55.15 | ManxPwr | I manage 6 small Asterisk installs ad I never need . in a dialplan. |
| 13:55.45 | ManxPwr | Generally the only people that need . in their dialplan is people that live in countries with variable length dialplans |
| 13:56.00 | ManxPwr | or people that want to call countries with variable length dialplans |
| 13:56.17 | RoyK | which is quite a lot |
| 13:56.45 | ManxPwr | RoyK, Yes, but less so than you might think. |
| 13:56.58 | ManxPwr | Even then you always know the min length of dial strings |
| 13:57.03 | sp2 | how can i get the caller id from a zap channel (FXO connected to a POTS) |
| 13:57.05 | ManxPwr | _X. is just plain lazyness. |
| 13:57.15 | ManxPwr | sp2, it happens automagically. |
| 13:57.33 | Dabba | > ManxPwr its from the wiki re sms |
| 13:58.00 | ManxPwr | One of these days I'm going to go thru the Wiki and fix some of the crappy docs on there. |
| 13:58.33 | ManxPwr | off to work |
| 13:59.05 | sp2 | mnxpwr, i mean in the dialing plan what is the defined variable name is it ${CALLERID}? |
| 14:00.47 | *** join/#asterisk pashah (~pashah@relay.patentica.com) |
| 14:00.54 | *** join/#asterisk sabre (sabre@69.149.209.83) |
| 14:00.55 | pashah | hello! |
| 14:01.13 | pashah | anybody from nufone.net here? |
| 14:04.54 | bjohnson | ManxPwr: I add stuff to the wiki as I discover it |
| 14:06.16 | sp2 | got to sleep now |
| 14:06.22 | *** join/#asterisk amir (~amir@shield.guindehi.ch) |
| 14:06.43 | Essobi | ManxPwr Ahh. I think the options to monitor got changed then. |
| 14:07.50 | Essobi | ManxPwr Monitor([file_format[:urlbase]|[fname_base]|[options]]): |
| 14:10.18 | ariel_ | morning all |
| 14:14.50 | *** join/#asterisk jgaviria (~jgaviria@63.245.86.116) |
| 14:15.00 | *** join/#asterisk _PiGreco_ (~a@adsl-215-48.38-151.net24.it) |
| 14:15.03 | _PiGreco_ | hello |
| 14:15.23 | abbas_ | which the best open source H323 GK? |
| 14:15.26 | jgaviria | hi, somebody using chan_unicall? |
| 14:15.40 | _PiGreco_ | is there a start place to read on ? i found docs pretty confusing |
| 14:16.53 | *** join/#asterisk jayden (~ircatjerr@65.170.43.34) |
| 14:17.21 | Darwin35 | I am learning it |
| 14:17.29 | Darwin35 | I have not fully set it up yet |
| 14:17.41 | Darwin35 | I got all the libs buiilt lastnight for it |
| 14:17.52 | datareactor | PiGreci http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 14:17.56 | jgaviria | Darwin35: are you talking about chan_unicall? |
| 14:18.07 | Darwin35 | yes |
| 14:18.28 | Darwin35 | I am building a new box with all the features installed |
| 14:18.39 | jgaviria | Darwin35: did you get chan_unicall.so in your asterisk libs? |
| 14:18.49 | _PiGreco_ | datareactor: tnx |
| 14:18.53 | jayden | hey, I have not spent much time playing with any of them yet, are any of the GUI's any good? |
| 14:19.17 | Darwin35 | the box is at work when I go in I will let you know |
| 14:19.26 | Darwin35 | waiting on a ride now |
| 14:20.18 | *** join/#asterisk Martohtar (Martohtar@82.196.218.130) |
| 14:20.34 | bjohnson | I have a couple of SPA 3000 issues. Wondering if others have same issues. When fxs and fxo connect .. call usually has a an echo. Also, getting a ring back through the fxs when the phone is hung up (fxs is plugged into LINE in on a Nortel CICS) |
| 14:20.39 | jgaviria | Darwin35: Im using unicall0.2pre8 and asterisk1.03, when i patch the makefile, all seems ok but chan_unicall.so is not created |
| 14:20.54 | modulus_ | jbot perl? |
| 14:20.55 | jbot | it has been said that perl is at http://www.handhelds.org/z/wiki/Perl or at http://www.perl.com, or a knitting stitch, or the Pathologically Eclectic rubbish Lister, or that other "P" language |
| 14:20.55 | bjohnson | ~h323 |
| 14:20.56 | jbot | it has been said that h323 is evil! Don't ask about h323 here. Ask JerJer if you need to bug someone. He says it works, but others don't. |
| 14:21.26 | Darwin35 | hmm |
| 14:21.26 | *** join/#asterisk anok (anok@66-234-37-42.nyc.cable.nyct.net) |
| 14:21.39 | bjohnson | jayden: depends on what you looking for |
| 14:21.49 | Darwin35 | well my ride will be here in 10 min and when I get to the office Iwill login and let you oknow what I fiind |
| 14:21.53 | jgaviria | Darwin35: what version of unicall and asterisk are you using? |
| 14:21.53 | ariel_ | bjohnson, the ring back is message waiting. |
| 14:21.58 | bjohnson | jayden: try some and leave your opinions on the wiki |
| 14:22.20 | bjohnson | ariel_: you think it is a SPA setting or a sip.conf setting? |
| 14:22.21 | Darwin35 | asterisk 1.0.5 |
| 14:22.33 | *** join/#asterisk HiTech69 (~hitech@155-105.202-68.tampabay.rr.com) |
| 14:22.52 | ariel_ | bjohnson, my spa when I first plug them in if there is an issue with a shuttle tone they will ring back. |
| 14:23.24 | jgaviria | Darwin35: and unicall0.2pre8? |
| 14:23.47 | Darwin35 | will have to look when I get to the office |
| 14:24.13 | jayden | ummm, honestly, just looking for a little extra visual wow for a demo |
| 14:24.31 | jgaviria | Darwin: ok thanks a lot... did you get chan_unicall.so right? |
| 14:24.48 | Darwin35 | I will let you know when I get to the office |
| 14:24.54 | Darwin35 | i dont have the box here |
| 14:25.11 | jayden | somthing with basic config stuff and good cdr reporting, ODBC backend |
| 14:25.34 | netsurfer | Mr. Torvalds has such a twisted fxxking sense of humor |
| 14:25.37 | jayden | AMP is cool, but very restructed |
| 14:25.45 | modulus_ | who here knows some perl? |
| 14:25.48 | jayden | restricted that is |
| 14:26.00 | modulus_ | anyone? |
| 14:26.13 | jayden | ~perl |
| 14:26.14 | jbot | perl is, like, at http://www.handhelds.org/z/wiki/Perl or at http://www.perl.com, or a knitting stitch, or the Pathologically Eclectic rubbish Lister, or that other "P" language |
| 14:26.18 | jayden | :) |
| 14:26.21 | bjohnson | modulus_: a very little |
| 14:26.25 | modulus_ | $coderef->(0) |
| 14:26.28 | jayden | knitting stitch |
| 14:26.36 | modulus_ | will that successfully derefence the block and execute? |
| 14:26.41 | ariel_ | jayden, I use asterisk@home to do my basic setup. I then put my own .conf files and edit the php not to have the web setup of amp for my conf files. I use the web editor in asterisk@home. |
| 14:26.54 | modulus_ | or do i need to pass $coderef->(1) ? |
| 14:27.16 | ariel_ | jayden, I use mainly the reports that amp does which makes it work. I also use the Flash Operator panel |
| 14:27.20 | bjohnson | ariel_: had a mailbox defined in sip.conf .. I've removed to try that for a bit. What is a shuttle tone? |
| 14:27.22 | jayden | asterisk@home users AMP, right? |
| 14:27.45 | jayden | uses... need to learn how to type one of these days |
| 14:27.53 | Dabba | quit |
| 14:27.54 | ariel_ | bjohnson, when you pickup the extension or phone connected to the sipura do you hear two quick tones. |
| 14:27.59 | *** join/#asterisk ashus_ (~ashus@p54BCD05E.dip.t-dialin.net) |
| 14:27.59 | *** part/#asterisk Dabba (dabba@matrix.lgw.ip6net.net) |
| 14:28.05 | ashus_ | hi. im trying to get zaphfc from bristuff-0.2.0-RC5 working. kernel is vanilla-2.6.10. it errors with lines like "/mnt/tmpfs/bristuff-0.2.0-RC5/zaphfc/zaphfc.c:359: error: structure has no member named `bytes2transmit'". can anyone help with this? |
| 14:28.09 | ariel_ | jayden, yes |
| 14:28.13 | modulus_ | $coderef->(undef) still dereferences the block |
| 14:28.15 | modulus_ | hmmm |
| 14:28.18 | ariel_ | but that does not mean you need to keep it. |
| 14:28.18 | modulus_ | i wish it didn't |
| 14:28.18 | bjohnson | ariel_: I'll keep an eye on that too |
| 14:29.08 | jayden | thanks |
| 14:29.10 | ariel_ | bjohnson, eye or listen for it? |
| 14:29.12 | ashus_ | or can anyone recommend a bristuff version that works with zaphfc? |
| 14:29.49 | *** join/#asterisk _PiGreco_ (~a@adsl-215-48.38-151.net24.it) |
| 14:31.08 | Darwin35 | ok rides here back in a few |
| 14:31.18 | ariel_ | well off I go to setup a small SOHO for a friend. See you all in about 1 hour or 2. |
| 14:39.20 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 14:40.28 | _PiGreco_ | mh, it appears asterisk with its example configs files has a lot of stuff ready for example..what about security ? i mean, i *think* i see sample accounts, it means ppl can log in into my asterisk ? |
| 14:42.07 | netsurfer | then dont u *think* u should remove them |
| 14:42.48 | _PiGreco_ | oh..i guess it means mv * /somewhere and start just with the prebuilt asterisk.conf ? |
| 14:42.48 | tzafrir | _PiGreco_, is your system available from the internet? |
| 14:42.56 | _PiGreco_ | tzafrir: my adsl line :) |
| 14:43.30 | tzafrir | are users connecting from the internet able to connect to your local extensions? |
| 14:43.59 | *** join/#asterisk switch (~switch@61.206.115.5.user.ad.il24.net) |
| 14:44.01 | tzafrir | And do you actually need such an option? |
| 14:44.10 | netsurfer | samples, are exactly that.. and should NOT be used in a production environment |
| 14:44.34 | _PiGreco_ | not yet able to understand this, im just starting to test..but i find 41 pre-built files, i dont understand which one are necessary to make it work as a minimum.. |
| 14:45.27 | _PiGreco_ | i guess i *have* to leave asterisk.* alone and remove the others.. not sure anyway |
| 14:46.14 | netsurfer | _PiGreco_ - to start with.. make sure only authorised users have entries in sip.conf iax.conf |
| 14:46.49 | shaZwaz | is there a way to input # key during a call without * recognizing it as transfer ? |
| 14:47.15 | netsurfer | shaZwaz - yes.. dont give that extension 't' |
| 14:47.49 | netsurfer | l8rz ppl |
| 14:47.55 | *** join/#asterisk Darwin35 (~Darin@c-24-3-241-22.client.comcast.net) |
| 14:48.00 | shaZwaz | I need it on a outgoing call and I still need to have the transfer capablity |
| 14:48.02 | _PiGreco_ | netsurfer: yes, thats ok |
| 14:48.13 | Darwin35 | <PROTECTED> |
| 14:48.13 | Darwin35 | Feb 15 04:47:36 WARNING[7402]: loader.c:440 load_modules: Loading module app_voicemail.so failed! |
| 14:48.13 | Darwin35 | Ouch ... error while writing audio data: : Broken pipe |
| 14:48.26 | Darwin35 | this is on 1.0.5 |
| 14:48.28 | _PiGreco_ | netsurfer: but i have sample extensions, sample voicemail, sample lot of stuff |
| 14:48.43 | shaZwaz | not using t will permanantly disable transrfers from that line |
| 14:48.47 | *** join/#asterisk switch (~switch@61.206.115.5.user.ad.il24.net) |
| 14:48.58 | *** join/#asterisk eKo1 (~bernd@63.245.57.70) |
| 14:49.38 | *** join/#asterisk clinthome (~clinthome@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net) |
| 14:50.15 | bjohnson | shaZwaz: make a special extension? |
| 14:50.21 | shaZwaz | most of the CC companies and other Customer Support center require the user to punch in # key |
| 14:50.40 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
| 14:50.40 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:50.44 | bjohnson | eg 8 dials out without the t option? |
| 14:50.46 | Darwin35 | grrr |
| 14:50.56 | Darwin35 | app voicemail is not working |
| 14:51.09 | shaZwaz | yeah that a way around but still not user friendly ! |
| 14:51.48 | bjohnson | shaZwaz: what would be user friendly? * reading the user's mind and saying this call I want the t and this call I don't |
| 14:51.54 | clinthome | Is anyone using AgentCallbackLogin with ackcall=no successfully? |
| 14:52.49 | shaZwaz | bjohnson: infact the transfer key should be changable in features .conf like the *8 thing |
| 14:53.13 | clinthome | ...it seems that ackcall=no is simply ignored... |
| 14:53.18 | jalsot | does anybody know what is that annoying beep on heard on phone connected through zaptel (PSTN-PRI-E100P-*-IAX2-iaxComm)? |
| 14:53.47 | shaZwaz | alos there c'd be a key combination to punch take place of # |
| 14:54.27 | jalsot | I guess it relates to echo cancel, but I'm not sure |
| 14:54.46 | eKo1 | Has anybody had problems with Asterisk changing the 'From:' field in the SIP headers after receiving a 407 challenge? |
| 14:55.16 | eKo1 | It seems Asterisk is changing it to the number being dialed an it is screwing up the CID. |
| 14:56.00 | _PiGreco_ | can i show active extensions from commandline ? |
| 14:56.03 | eKo1 | I.e. the CID that people see when being called is their number. |
| 14:56.30 | shaZwaz | _PiGreco_: show channels ? |
| 14:56.35 | _PiGreco_ | oh tnx |
| 14:57.16 | _PiGreco_ | isnt there any FM to read so i wont bother anymore? :) |
| 14:57.35 | *** join/#asterisk jetscreamer (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
| 14:57.35 | shaZwaz | ~docs |
| 14:57.36 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 14:57.54 | shaZwaz | see the Wiki |
| 14:58.13 | _PiGreco_ | mh i found all that doc pretty confusing |
| 14:58.16 | _PiGreco_ | i mean |
| 14:58.34 | _PiGreco_ | it looks like a lot of howtos |
| 14:58.42 | _PiGreco_ | so if i have to do something its ok |
| 14:58.46 | *** join/#asterisk das1234 (~das1234@lizard.disisit.com) |
| 14:58.53 | _PiGreco_ | but to understand how it all works..mmh.. |
| 14:59.01 | _PiGreco_ | :/ |
| 15:00.16 | *** join/#asterisk soulz0 (~soulz0@cm51.epsilon168.maxonline.com.sg) |
| 15:00.19 | soulz0 | hello all |
| 15:00.25 | *** join/#asterisk negativecreep (~yama@202.147.174.97) |
| 15:01.31 | negativecreep | hi all.. |
| 15:01.35 | soulz0 | i saw mark's post, when the problem of ast_channel_make_compatible |
| 15:01.49 | soulz0 | and he suggested in earlier post that |
| 15:01.49 | soulz0 | You have one configured for G.723.1 and one configured for linear or |
| 15:01.49 | soulz0 | mulaw. |
| 15:01.52 | negativecreep | i want to configure for a sip extension to dial a pstn number..how can i do that?? any pointers? |
| 15:01.58 | *** join/#asterisk oej (~oej@54.Red-80-32-211.pooles.rima-tde.net) |
| 15:02.10 | *** join/#asterisk iCEBrkr (icebrkr@chrome.cyberdyne.org) |
| 15:02.54 | negativecreep | hi soulz0 |
| 15:03.18 | stevekstevek | ~docs |
| 15:03.19 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 15:03.32 | stevekstevek | creep: look there |
| 15:03.55 | soulz0 | any pointers? |
| 15:03.58 | negativecreep | stevekstevek: where? |
| 15:04.12 | tzanger | stevekstevek: werd |
| 15:04.41 | stevekstevek | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 15:05.24 | stevekstevek | tzanger: try the latest patches yet? |
| 15:05.29 | tzanger | yup |
| 15:05.37 | tzanger | said some stuff about 'em on -dev |
| 15:05.47 | stevekstevek | oh, yeah? (goes reading). |
| 15:06.14 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 15:06.18 | shaZwaz | _PiGreco_: u sh'd see DCAP classes |
| 15:07.10 | tzanger | #asterisk-dev, not asterisk-dev :-) |
| 15:07.16 | stevekstevek | ahh, i see :) |
| 15:07.28 | _PiGreco_ | shaZwaz: uh ? |
| 15:08.09 | stevekstevek | OK, I guess I'll reply there. |
| 15:15.09 | Essobi | Anyone here a queue wiz? |
| 15:15.50 | Essobi | I need to setup a queue for extensions that are not local to the * box but on a SIP gateway elsewhere.. Can I do that? I read abou agents using agent login and all that jazz.. I don't know if it's possible then. |
| 15:15.51 | *** join/#asterisk mbellion (~michael@e135.stw.stud.uni-saarland.de) |
| 15:19.04 | *** join/#asterisk stevekstevek (~stevekste@slim-eth0.horizonlive.net) |
| 15:19.49 | mbellion | hi! |
| 15:21.10 | mbellion | I am trying to run Asterisk on a gentoo machine with grsecurity and pax kernel. Unforuntately asterisk dies directly after start with: |
| 15:21.24 | mbellion | [res_features.so]Feb 15 16:21:17 WARNING[19198]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/res_features.so: undefined symbol: adsi_available |
| 15:21.38 | mbellion | Feb 15 16:21:17 WARNING[19198]: loader.c:440 load_modules: Loading module res_features.so failed! |
| 15:21.51 | mbellion | Anybody have an idea what could be the reason for that? |
| 15:21.54 | shaZwaz | mbellion: in modules.conf noload =res_features.so |
| 15:22.58 | shaZwaz | compilation erros ! |
| 15:23.02 | mbellion | What is this res_features.so used for? Is it something I can live without? |
| 15:23.22 | shaZwaz | parking and a couple of other things |
| 15:23.59 | sivana | ~seen aginamu |
| 15:24.01 | jbot | aginamu <~AgiNamu@216.230.151.230> was last seen on IRC in channel #asterisk, 3d 7h 20m 4s ago, saying: 'good luck'. |
| 15:24.01 | mbellion | Could it be because of the grsec and pax patches? I mean is there somebody successfully running asterisk in a similar environment? |
| 15:24.47 | zigman | mbellion i run asterisk with grsec no problems |
| 15:25.08 | zigman | 2.6.10-as3 |
| 15:25.14 | zigman | 2.6.10-grsec-as3 |
| 15:25.23 | mbellion | I am still running 2.4 version |
| 15:25.53 | mbellion | I just added noload =res_features.so to the modules.conf and then it is complaining about the next module |
| 15:26.03 | mbellion | It does not seem to be able to load any module |
| 15:26.13 | mbellion | zigman: are you also using pax? |
| 15:27.30 | zigman | yes |
| 15:27.47 | mbellion | strange |
| 15:28.02 | mbellion | zigman: did you have to configure anything special? |
| 15:28.47 | *** join/#asterisk HitTop (~root@host6614613596.biz.tor.fcibroadband.com) |
| 15:29.39 | shaZwaz | anyone successfully got rid of echo on sip phones ? |
| 15:29.48 | shaZwaz | SIP --> PSTN |
| 15:29.57 | hajekd | How is transfer button on grandstream should work? When I press transfer, call is put on hold, then I need to enter phone number and press the Send button...is that correct? |
| 15:30.09 | negativecreep | shaZwaz: can you tell me how can i configure SIP --> PSTN |
| 15:30.28 | *** join/#asterisk TheEmperor (TheEmperor@218.111.51.53) |
| 15:31.09 | shaZwaz | negativecreep: u need a digium card confiure it run * change dialplan and hola |
| 15:31.24 | shaZwaz | in a line that was .. |
| 15:31.38 | negativecreep | shaZwaz: i have pstn 2 sip working...but cant figure out how to configure sip to pstn dialplan |
| 15:32.01 | shaZwaz | what card u have ? |
| 15:32.50 | negativecreep | an X100P |
| 15:32.58 | netsurfer | piece of shit box just dumped core during a kernel compile :( |
| 15:33.06 | negativecreep | i cant figure out the dialplan.. |
| 15:33.07 | TheEmperor | how do i configure |
| 15:33.08 | negativecreep | it works fine.. |
| 15:33.25 | TheEmperor | so that * goes straight to voicemail after a certain time? |
| 15:34.40 | shaZwaz | okay put a exten => _9X. ,1,Dial(zap/1/${EXTEN:1}) , exten => _9X.,2,Hangup in your dialplan |
| 15:34.53 | negativecreep | shaZwaz: thats it? |
| 15:35.03 | shaZwaz | for now :) |
| 15:35.12 | algorithmn | TheEmperor: gotoiftime |
| 15:35.13 | negativecreep | thnx dude |
| 15:35.22 | TheEmperor | algorithmn: ? |
| 15:35.28 | shaZwaz | u can make calls prefixing ur number with 9 |
| 15:35.32 | algorithmn | GotoIfTime within extensions.conf |
| 15:35.42 | TheEmperor | algorithmn: oh... |
| 15:35.50 | algorithmn | in * cli type 'show application gotoiftime |
| 15:35.52 | algorithmn | ' |
| 15:36.08 | mbellion | zigman: I have now disabled pax for the asterisk binary, but still the same problem. Would you mind sending me your grsec config? Probably there is something in my config that is too restrictive. |
| 15:36.24 | TheEmperor | algorithmn: thanks :) |
| 15:36.38 | algorithmn | dont' sweat it... |
| 15:36.49 | shaZwaz | mbellion: do u have any digium cards ? |
| 15:36.50 | algorithmn | just enjoy yourself while you learn linux pbx administration |
| 15:37.03 | TheEmperor | algorithmn: exten |
| 15:37.03 | TheEmperor | algorithmn: exten => 100,5,GoToIfTime 1800 ? is that right? |
| 15:37.22 | algorithmn | for the most part... you need some more parameters |
| 15:37.43 | mbellion | shaZwaz: no, I am just starting with asterisk... I have no digium cards .. perhaps will be added later when the setup grows |
| 15:38.18 | shaZwaz | do u have latest stable ver ? |
| 15:39.13 | mbellion | yes, I have 1.0.5 |
| 15:40.13 | TheEmperor | algorithmn: time format is like 6pm or 1800 ? |
| 15:40.21 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
| 15:41.03 | *** join/#asterisk Martin_Zahn (~Martin@dsl-213-023-223-100.arcor-ip.net) |
| 15:42.42 | *** join/#asterisk Dr-Linux (~sshah@202.125.141.6) |
| 15:43.14 | shaZwaz | TheEmperor: its 24hrs format |
| 15:44.21 | TheEmperor | shaZwaz: thanks |
| 15:45.32 | shaZwaz | SIP --> PSTN echo !!! |
| 15:45.37 | Dr-Linux | my asterisk server was on local ip, i changed it to live ip, then i changed ip in softphones .. but its not working ? |
| 15:45.40 | Dr-Linux | what should i do ? |
| 15:46.20 | shaZwaz | where are the ip phones loacted ? |
| 15:46.32 | shaZwaz | LAN or remote ? |
| 15:46.36 | Dr-Linux | shaZwaz: on lan |
| 15:46.39 | Dr-Linux | as i checked |
| 15:46.49 | shaZwaz | same as * machine ? |
| 15:47.12 | Dr-Linux | yeah, on local LAN |
| 15:47.37 | Dr-Linux | it was working fine, but i changed local ip to live ip, now its not working .. |
| 15:47.48 | shaZwaz | do this on cmd prompt netstat -a |grep 5060 and check which IP its listening on |
| 15:48.05 | Dr-Linux | hhm.. okey wait |
| 15:49.01 | Dr-Linux | shaZwaz: |
| 15:49.02 | Dr-Linux | [root@consultancy-1 local]# netstat -a |grep 5060 |
| 15:49.02 | Dr-Linux | udp 0 0 *:5060 *:* |
| 15:49.19 | jayden | anyone w/ AMP experience around? |
| 15:49.25 | Dr-Linux | what does it eman ? |
| 15:49.47 | shaZwaz | its listening on 0.0.0.0:5060 |
| 15:50.09 | shaZwaz | do u have a privte IP on this machine ? |
| 15:50.09 | Dr-Linux | shaZwaz: so what to do to fix this issue ? |
| 15:50.25 | Dr-Linux | yeah, i have private ip too on this machine. |
| 15:50.52 | Dr-Linux | kya karoon ? |
| 15:51.04 | *** join/#asterisk fishboy1669 (proxyuser@62.69.81.129) |
| 15:51.12 | HitTop | jayden: im uisng amp |
| 15:51.20 | Darwin35 | make[1]: Leaving directory `/usr/src/asterisk-1.0.5/pbx' |
| 15:51.20 | Darwin35 | make[1]: Entering directory `/usr/src/asterisk-1.0.5/apps' |
| 15:51.20 | Darwin35 | make[1]: *** No rule to make target `install'. Stop. |
| 15:51.20 | Darwin35 | make[1]: Leaving directory `/usr/src/asterisk-1.0.5/apps' |
| 15:51.20 | Darwin35 | make: *** [bininstall] Error 1 |
| 15:51.28 | shaZwaz | give it as localnet = your priv ip /subnet mask |
| 15:51.47 | TheEmperor | how can i configure * so that it calls out a number on a certain time everyday? :) |
| 15:52.12 | Delvar | TheEmperor: crongob and a .call file |
| 15:52.13 | shaZwaz | TheEmperor: shcedule a Call file |
| 15:52.14 | jayden | does AMP work with head |
| 15:52.23 | Delvar | cronjob* |
| 15:52.44 | TheEmperor | Delvar: how? |
| 15:53.02 | Delvar | look at /usr/src/asterisk/sample.call |
| 15:53.19 | Delvar | to setup a cronjob look a tthe man pages |
| 15:53.25 | Delvar | its not too hard |
| 15:53.29 | TheEmperor | shaZwaz: ..how.. ? |
| 15:53.31 | shaZwaz | if the call is same most of the time use a .call file to be moved to above path |
| 15:54.31 | TheEmperor | thanks Delvar and shaZwaz |
| 15:54.36 | Delvar | TheEmperor: look at the sample.call file the comments explain how to use it, cronjob is easy to setup using 'crontab -e' the syntax is int eh man pages |
| 15:54.41 | TheEmperor | Delvar: where are the man pages for cronjob? |
| 15:54.48 | Delvar | man crontab |
| 15:54.51 | shaZwaz | #linux |
| 15:54.54 | Delvar | from command line |
| 15:55.01 | shaZwaz | use at |
| 15:55.04 | TheEmperor | ok.. |
| 15:55.58 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 15:56.10 | *** part/#asterisk das1234 (~das1234@lizard.disisit.com) |
| 15:56.31 | shaZwaz | Dr-Linux: u sing * as ur machine name ? |
| 15:56.55 | shaZwaz | weird |
| 15:58.34 | negativecreep | shazwaz set bindport=0.0.0.0 and restart * |
| 15:58.43 | negativecreep | sorry Dr-Linux not shaZwaz |
| 15:59.12 | shaZwaz | I think he is already using it ? |
| 16:00.12 | *** join/#asterisk HuangDi (TheEmperor@218.111.51.53) |
| 16:00.19 | HuangDi | got cut off :( |
| 16:00.22 | negativecreep | Dr-Linux: can the ip phones reach the * machine or not? |
| 16:00.45 | HuangDi | Delvar: what do I need to put in the extensions file? |
| 16:01.42 | Delvar | HuangDi: depends what you want to do? |
| 16:02.04 | HuangDi | Delvar: you were mentioning just now about * calling at a specfic time.. |
| 16:02.10 | Delvar | HuangDi: for testing i usualy dump into Voicemail() app |
| 16:02.22 | HuangDi | Delvar: I got cut off, was TheEmperor just now :) |
| 16:02.47 | HuangDi | Delvar: I need to amend the sample.call first? |
| 16:03.04 | Delvar | HuangDi: yes.. you can make the .call file drop into any context/extension you want, what you do in that context is up to you |
| 16:03.47 | Delvar | HuangDi: usualy i do exten => _X.,1,Voicemail() |
| 16:03.50 | HuangDi | Delvar: ah, so I need to specify the extension in the .call file, and then in the extensions file, specify where to call to? |
| 16:03.51 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 16:04.50 | hajekd | do you have Conference button working on Grandstream? |
| 16:04.57 | Delvar | HuangDi: sort of yes, you should read the comment sin teh file then look on voip-info.org there is a LOT of info there and probably a walk through for this |
| 16:05.18 | HuangDi | Delvar: ok, I'll check it out. Thanks :) |
| 16:05.31 | Delvar | hajekd: doesnt seem to do anything on .16/.18/.22 |
| 16:08.01 | *** join/#asterisk DrNitro|Work (~docnitro@foxxp.wrlc.org) |
| 16:08.11 | *** join/#asterisk jcims (~jcims@cpe-69-135-121-57.columbus.rr.com) |
| 16:09.07 | fishboy1669 | hi guys |
| 16:09.10 | fishboy1669 | hows life |
| 16:09.47 | Martin_Zahn | Hi! Just a quick question: My brother has to build an asterisk server (school/company project). Plan is: All calls schould go over sipgate (IIRC) but the original phone-system they have at the company schould stay to minimize costs. so the server would have to emulate a multiplex-connection (12 lines) to their original phone-system. Is that possible ? (I haven't done anything with asterisk yet but probably would have to set up the se |
| 16:10.50 | hajekd | Delvar: hmm, so there is no way to have a 3way call with grandstream? |
| 16:12.04 | Zeeek | hajekd see meetme |
| 16:12.21 | fishboy1669 | hi zeeek |
| 16:12.24 | Zeeek | lo |
| 16:12.26 | *** join/#asterisk eivindtr (~Eivind@062016241059.customer.alfanett.no) |
| 16:13.40 | hajekd | looks like I have to upgrade BT 102 |
| 16:13.47 | *** join/#asterisk jlewis (~jlewis@solo.atlantic.net) |
| 16:13.49 | Zeeek | won't change anything |
| 16:13.49 | hajekd | to BT102 |
| 16:14.08 | hajekd | it looks like BT102 has Conference button which is assumed to work? |
| 16:14.09 | Zeeek | I have BT102 - it's identical in every way to BT101 except for the extra RJ- |
| 16:14.11 | negativecreep | hi Zeeek |
| 16:14.15 | Zeeek | no it does not |
| 16:14.29 | negativecreep | Zeeek: how can i jump from one context to another.. |
| 16:14.30 | Zeeek | hey I see you were successful nega |
| 16:14.36 | Zeeek | by goto |
| 16:14.42 | Zeeek | show application goto |
| 16:14.51 | negativecreep | ur the man Zeeek |
| 16:14.58 | Zeeek | no, beginner like you |
| 16:14.59 | DrNitro|Work | I was wondering is there a way to allow someone to leave a voicemail message and flag that message as "urgent", then based on the message being urgent it will dial say a pager and page them with a numeric message? |
| 16:15.25 | jlewis | other than "fromuser=" are there sip settings that will override trying to setcidnum in a dialplan? |
| 16:15.54 | fishboy1669 | na zeek is the man |
| 16:15.56 | jlewis | and is there any way to override the sip.conf's fromuser= setting in the dialplan? |
| 16:16.11 | Zeeek | I am *a* man, but no more |
| 16:16.34 | fishboy1669 | modist as well lol |
| 16:16.43 | Zeeek | jlewis have you read any docs? |
| 16:16.49 | fishboy1669 | just got two x100p woring in same box :-) |
| 16:16.59 | Zeeek | fish me too |
| 16:17.02 | fishboy1669 | cool |
| 16:17.06 | *** join/#asterisk E|nyPRI_ (~les@205-200-64-180.static.mts.net) |
| 16:17.10 | tzafrir | anybody managed to install mozphone? |
| 16:17.12 | Zeeek | got all the interrupts turned off |
| 16:17.15 | jlewis | been searching all over www.voip-info.org...are there better docs? |
| 16:17.21 | Zeeek | Starter tutorial: |
| 16:17.21 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 16:17.21 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 16:17.21 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 16:17.22 | Zeeek | THE reference of the moment: |
| 16:17.22 | Zeeek | http://www.asteriskdocs.org |
| 16:17.25 | E|nyPRI_ | anyone know xylome off bugs.digium ? |
| 16:17.36 | fishboy1669 | and finaly got my dell box working as well it was the suse linux unstandard kernel |
| 16:17.40 | Zeeek | the wiki has a lot of info but not easy to find |
| 16:17.44 | fishboy1669 | so using mandrake on it now |
| 16:18.08 | fishboy1669 | lol we should set up one website that has all the urls on it send every one there |
| 16:18.19 | fishboy1669 | rtfm.com |
| 16:18.19 | Zeeek | ManxPower has that |
| 16:18.20 | fishboy1669 | lol |
| 16:18.45 | Zeeek | "RTFM specializes in expert technical consulting for difficult problems, with a particular focus on Network Security and Distributed Systems." |
| 16:18.45 | *** join/#asterisk mtqh (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 16:19.24 | DrNitro|Work | I was wondering is there a way to allow someone to leave a voicemail message and flag that message as "urgent", then based on the message being "urgent" it will dial say a pager to leave a numeric message? |
| 16:19.38 | fishboy1669 | lol i wasnt aware there was already an rtfm.com |
| 16:19.40 | Zeeek | yes, I saw that |
| 16:19.47 | fishboy1669 | should have guessed lol |
| 16:20.16 | Zeeek | Nitro - you could have them press a digit before gouing to vmal and change nboxes |
| 16:20.35 | *** join/#asterisk Flood_of_SYNs (~cory@wbar13.chi1-4.27.95.222.chi1.dsl-verizon.net) |
| 16:20.39 | *** part/#asterisk Flood_of_SYNs (~cory@wbar13.chi1-4.27.95.222.chi1.dsl-verizon.net) |
| 16:20.43 | *** join/#asterisk Flood_of_SYNs (~cory@wbar13.chi1-4.27.95.222.chi1.dsl-verizon.net) |
| 16:20.54 | riksta | i don't suppose anyone here that's in the UK, has an old 1U server they don't use i could buy off them cheap do they? |
| 16:20.58 | DrNitro|Work | there is no way to make it so after they are done recording the message it gives them options to change the "message options" and have them flag it as urgent? |
| 16:21.01 | Zeeek | IOW, line busy/not answering, give everyone the choice (or by cid) to press 1 for urgency, then leave the message at a box that has a pager email |
| 16:21.12 | Zeeek | AFAIK no |
| 16:21.20 | DrNitro|Work | hmm that might work just as well |
| 16:21.27 | Zeeek | if you can use callerid it'd be even better |
| 16:22.22 | *** join/#asterisk Othello (Othello@nusnet-233-130.dynip.nus.edu.sg) |
| 16:22.27 | *** part/#asterisk Flood_of_SYNs (~cory@wbar13.chi1-4.27.95.222.chi1.dsl-verizon.net) |
| 16:22.40 | Zeeek | hey here's a free idea for an extension to asterisk vmail: |
| 16:22.46 | expressfone1 | any one looking for terminate at +53 ???? .50usd, billing step 30/6, PDD 7sec, ASR 50%, TDM Quality Route |
| 16:22.47 | Zeeek | Vmail Filters |
| 16:23.08 | Zeeek | like mozilla-based ones, they can manipulate vmail a,nd decide urgency by cid etc |
| 16:23.17 | Zeeek | that solves your problem |
| 16:23.22 | Zeeek | just need to write it |
| 16:23.39 | DrNitro|Work | I'll get on that ;) |
| 16:23.46 | dsmouse | expressfone1: where's +53 again? |
| 16:23.47 | DrNitro|Work | might take me about a month |
| 16:24.01 | expressfone1 | Cuba +53 |
| 16:24.22 | expressfone1 | proper and cell |
| 16:24.44 | Zeeek | yeah you'd just need to launch an app after taking a message, see who it was, time of day, length, who for and decide what is urgent on that bases |
| 16:24.58 | Zeeek | course my originla comment would work fine too |
| 16:25.10 | DrNitro|Work | I think your original comment is a lot easier |
| 16:25.33 | fishboy1669 | zeeek did u say u had to play with your interupts? |
| 16:25.40 | negativecreep | Zeeek: take a look at this. http://www.pastebin.com/242106 |
| 16:25.50 | negativecreep | i am trying to dial a pstn number from a sip client |
| 16:25.50 | fishboy1669 | i just looked at mine and the cards are sharing with eth and video ! |
| 16:26.45 | Zeeek | negative - hard to believe you read any docs looking at that |
| 16:26.53 | *** join/#asterisk dalabera (~Dalabera@146.82.190.162) |
| 16:27.03 | Zeeek | fish if it works, don't fix it! |
| 16:27.19 | negativecreep | Zeeek: oops |
| 16:27.32 | Zeeek | shame on you! |
| 16:27.36 | Zeeek | shame, shame |
| 16:27.36 | jalsot | does anybody know what is that annoying beep on heard on phone connected through zaptel (PSTN-PRI-E100P-*-IAX2-iaxComm)? I guess it's the echo cancel... any idea? |
| 16:27.48 | jlewis | ARGH!!...finally found the callerid problem |
| 16:28.03 | fishboy1669 | not tested it pluged in phone line yet im sure it will mess up as usual with everything in my life! |
| 16:28.08 | Zeeek | wear a crown of thors for a week! |
| 16:28.15 | fishboy1669 | lol |
| 16:28.18 | Zeeek | or thorns even |
| 16:28.18 | fishboy1669 | good plan |
| 16:28.28 | Zeeek | not you fish - you can eat some fish food |
| 16:28.34 | negativecreep | Zeeek: :( |
| 16:28.39 | jlewis | if you have a sip.conf entry that has a fromuser= setting, and remove it, "sip reload" doesn't remove it...you have to explicitly set fromuser= nothing and do a sip reload to clear the setting |
| 16:28.44 | fishboy1669 | might even win the lottery if i go through suffering like that |
| 16:28.51 | fishboy1669 | zeek are u using udev? |
| 16:28.52 | Zeeek | ok negatory - you have to do some serious reading |
| 16:28.57 | Zeeek | no fish |
| 16:29.01 | fishboy1669 | oh |
| 16:29.02 | fishboy1669 | poo |
| 16:29.04 | fishboy1669 | fish poo |
| 16:29.14 | Zeeek | first look at the dial application and its arguments |
| 16:29.22 | negativecreep | Zeeek: ok |
| 16:29.55 | jlewis | now I can setcidnum whatever I want in the dialplan...and it actually works |
| 16:29.55 | Zeeek | but the worrying part is why do you need to change contexts? So the trouble is |
| 16:29.55 | Zeeek | The dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls. "http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650 |
| 16:29.55 | Zeeek | READ THE ABOVE |
| 16:29.59 | negativecreep | Zeeek: ok |
| 16:30.03 | Zeeek | heh |
| 16:30.09 | Zeeek | I sould like a real authority |
| 16:30.20 | Zeeek | I hope JerJer is nowhere around |
| 16:30.21 | fishboy1669 | u are da man! |
| 16:30.25 | shaZwaz | or a place where u can messup ur whole life :D |
| 16:30.25 | Zeeek | nah |
| 16:30.54 | Zeeek | shaWaz ???? |
| 16:31.17 | Zeeek | wrong window - yiou want the microsoft seminar window |
| 16:31.30 | shaZwaz | heehee |
| 16:31.34 | shaZwaz | nahin Zeeek |
| 16:31.41 | Essobi | Umm. |
| 16:31.46 | Zeeek | I luv Exchange |
| 16:31.51 | Essobi | what's the verbose level to debug extensions.conf? |
| 16:31.53 | Zeeek | it's so cool... NOT |
| 16:31.59 | xkev | essobi 4 |
| 16:32.19 | xkev | e.g. |
| 16:32.20 | xkev | <PROTECTED> |
| 16:32.20 | xkev | <PROTECTED> |
| 16:32.28 | Essobi | preshadit |
| 16:32.43 | Essobi | Mmm. |
| 16:32.43 | shaZwaz | Zeek any idea about how get rid of SIP -> PSTN echo ? |
| 16:32.52 | xkev | shazwaz an echo canceler |
| 16:32.55 | jlewis | was anyone aware of that?...that commenting out a sip.conf entry and doing a sip reload doesn't actually unset the setting? |
| 16:32.55 | xkev | :) |
| 16:32.59 | Essobi | How about sip peers? I got a peer that's not acting right. |
| 16:32.59 | Zeeek | shaWaz I pretend is isn't there |
| 16:33.07 | xkev | essobi sip debug |
| 16:33.10 | eKo1 | shaZwaz: Stop using speaker phone. |
| 16:33.17 | Zeeek | yes, good advice |
| 16:33.33 | shaZwaz | no speaker phone |
| 16:33.34 | xkev | there are good pages on echo in the wiki |
| 16:33.47 | xkev | including an explanation of where it comes from, how to determine if it's you or them, etc. |
| 16:33.51 | Zeeek | someone should assemble all the knowledge on that subject on one page |
| 16:34.00 | xkev | they are linked to each other iirc |
| 16:34.33 | shaZwaz | I have tried every trick in the book and its still there |
| 16:34.43 | Essobi | Mmm. Is there an easy way to prepend digits to a dialed number from certian peers? |
| 16:34.44 | Zeeek | btw, there is a BIG different between asterisk and what we are used to : if a phone is busy when a call comes in and the person hangs up immediately, that phone will still never ring because it's marked as busy |
| 16:34.47 | fishboy1669 | anyone here use udev with zaptel |
| 16:34.56 | shaZwaz | only option left is to use a external echo canceler |
| 16:35.02 | fishboy1669 | need to know where i set it up to do the modprobe automatically |
| 16:35.04 | xkev | shazwaz, are your sip clients on the localnet? |
| 16:35.11 | shaZwaz | yup |
| 16:35.16 | xkev | hrm, then jitter isn't the issue |
| 16:35.27 | xkev | and do you hear it, but not the other end? |
| 16:35.27 | fishboy1669 | shazwas what pstn interface are u using? |
| 16:35.31 | Zeeek | Essobi sure by using contexts |
| 16:35.52 | shaZwaz | fishboy1669: T100p |
| 16:35.59 | Essobi | Mmm. My default context is overriding the context I'm setting for the peer for some reason.. |
| 16:36.08 | xkev | my echo was unmanageable running thru a T100P via the legacy pbx, but now that it's direct to pri, I never hear any |
| 16:36.10 | jlewis | xkev: we've had to use AGGRESSIVE_SUPPRESSOR on our t100p PRIs to get echo mostly gone |
| 16:36.22 | shaZwaz | talking about jitter I have another issue , choppy voice on same path |
| 16:36.25 | fishboy1669 | does that use zapata.conf? |
| 16:36.36 | fishboy1669 | shaz ^^ |
| 16:37.01 | xkev | shaz maybe that's why the canceler can't find a tap? |
| 16:37.24 | xkev | do you have a sound card or other interrupt-heavy traffic going on? |
| 16:37.30 | shaZwaz | I have AGGRESSIVE SUPPERESSOR uncommented |
| 16:37.53 | fishboy1669 | shaz does it use zapata.conf the t100p? |
| 16:38.00 | xkev | <PROTECTED> |
| 16:38.06 | shaZwaz | ofcourse it does fishboy1669 |
| 16:38.13 | fishboy1669 | if so send me a copy of the conf file |
| 16:38.18 | *** join/#asterisk Guigui|taff (~guillaume@217.167.233.150) |
| 16:38.21 | Guigui|taff | hello. |
| 16:38.25 | jlewis | and #define ECHO_CAN_MARK2 |
| 16:38.38 | DEVILoper | Hi i wanna use * with my external modem any way ?? |
| 16:38.50 | shaZwaz | xkev: its a P4 |
| 16:38.57 | jlewis | and we briefly tried ECHO_CAN_MARK3 and didn't like it |
| 16:39.51 | jero_SFLphone | does rxgain influent Caller*ID reception ? |
| 16:40.17 | fishboy1669 | shaz pastebin your zapata.conf |
| 16:40.20 | jero_SFLphone | I only get 2 or 3 of 10 Caller*IDs on digium's fxo |
| 16:40.37 | Guigui|taff | (Could I ask a question please ?) |
| 16:40.58 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 16:42.05 | DEVILoper | Guigui No :) |
| 16:42.12 | Guigui|taff | hmph :( |
| 16:42.23 | jero_SFLphone | dont ask to ask, ask |
| 16:42.36 | Guigui|taff | ok right |
| 16:42.37 | Guigui|taff | :) |
| 16:42.50 | *** join/#asterisk Rick_Hunter (~rhunter@07-034.008.popsite.net) |
| 16:43.20 | Guigui|taff | well I'm testing rxfax feature, and I'm getting the "Unable to find a path from slin to unknown" and "Unable to restore read format on 'Modem[i4l]/ttyI0'". After a google search, I saw that I need to use an other codec. Is there a way in extensions.conf to force the codec to be use when receiving fax ? |
| 16:43.58 | *** join/#asterisk stepcut (~redlion@ip68-107-21-88.sd.sd.cox.net) |
| 16:44.07 | eKo1 | Guigui|taff: On what? |
| 16:44.17 | *** join/#asterisk multrix (~chatzilla@ALyon-252-1-6-169.w82-122.abo.wanadoo.fr) |
| 16:44.23 | Guigui|taff | in extensions.conf file |
| 16:44.45 | Guigui|taff | like an "application" that let changing the codec |
| 16:44.48 | eKo1 | OK, what is the fax connected to. |
| 16:45.05 | expressfone1 | -- Executing ZapScan("IAX2/madcom3-in-alaw01@10.1.6.11:4569/1", "") in new stack |
| 16:45.06 | expressfone1 | Feb 15 17:44:40 NOTICE[11266]: chan_iax2.c:2447 iax2_read: I should never be called! |
| 16:45.08 | expressfone1 | ???? |
| 16:45.25 | Guigui|taff | it's a soft fax in asterisk (rx/txfax), and asterisk is connected on an isdn link |
| 16:45.29 | Guigui|taff | or T0 |
| 16:45.37 | eKo1 | expressfone1: Stop using zapscan. |
| 16:45.51 | eKo1 | Guigui|taff: Get a real fax. |
| 16:45.54 | Guigui|taff | the fax passed, but i see just the header of the page in the tiff |
| 16:45.55 | Guigui|taff | hum |
| 16:46.03 | Guigui|taff | that's not really an answer :) |
| 16:46.04 | stepcut | Since I upgraded from 1.0.3 to CVS head, I can no longer register sip connections. I am on freebsd -- has anyone else seen this ? |
| 16:46.05 | Essobi | Grmm. |
| 16:46.13 | expressfone1 | we using it for monitoring voice quality |
| 16:46.14 | *** join/#asterisk maruz (~maumar@adsl-123-3.38-151.net24.it) |
| 16:46.24 | Guigui|taff | I don't wanna use a real fax |
| 16:46.28 | Essobi | why would I get "Found no matching peer or user for '192.168.0.1:1032'" in sip debug? |
| 16:46.47 | Essobi | then it default to a default context? |
| 16:46.47 | Essobi | the default context? |
| 16:46.47 | eKo1 | Guigui|taff: faxing on * isn't well supported so you're basically on your won. |
| 16:46.48 | Essobi | I've got the peer in there.. |
| 16:46.54 | terrapen | hey, is anyone else getting a 403 on this page: |
| 16:46.55 | terrapen | http://www.google.com/froogle/merchants/ |
| 16:46.56 | Essobi | but it can't find it from some reason. |
| 16:47.00 | eKo1 | s/won/own |
| 16:47.18 | Guigui|taff | hm ok |
| 16:47.46 | eKo1 | Guigui|taff: If and when you do get it working, please make a post about it on the wiki. |
| 16:48.36 | eKo1 | stepcut: Stop using CVS Head. |
| 16:49.11 | xkev | jero_SFLphone, yes rxgain influences caller*id on fxo ports |
| 16:49.24 | xkev | wha? cvs head is love |
| 16:49.55 | eKo1 | xkev: Only if you're a developer. |
| 16:50.08 | xkev | well, ok :) |
| 16:51.29 | Guigui|taff | eKo1: http://www.voip-info.org/wiki-Asterisk+Fax+to+email ? :) |
| 16:52.45 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 16:54.00 | *** join/#asterisk meppl (~mephisto@p548530F5.dip.t-dialin.net) |
| 16:54.12 | xkev | can someone give me an example use of Realtime() app? |
| 16:55.19 | negativecreep | Zeeek: got it working.. |
| 16:55.21 | negativecreep | :) |
| 16:55.36 | xkev | like RealTime(<family>|<colmatch>|<value>[|<prefix>]); wtf is colmatch and value? |
| 16:55.39 | negativecreep | Zeeek: i committed some stupid mistakes..thnx for the help |
| 16:56.59 | *** join/#asterisk abbas_ (nidobas@203.81.200.28) |
| 16:58.11 | *** join/#asterisk dsfr (~dsfr@216.207.244.183) |
| 16:59.07 | *** part/#asterisk maruz (~maumar@adsl-123-3.38-151.net24.it) |
| 16:59.20 | *** part/#asterisk jcims (~jcims@cpe-69-135-121-57.columbus.rr.com) |
| 16:59.33 | jero_SFLphone | xkev, I set up my rxgain/txgain using a milliwatt test application to have a precise setting |
| 16:59.37 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
| 16:59.48 | jero_SFLphone | xkev, but most of my caller*id's are not received |
| 17:00.04 | *** join/#asterisk Skysky (~Jack@host6614613596.biz.tor.fcibroadband.com) |
| 17:02.31 | xkev | jero I never solved my callerid on x100p |
| 17:04.06 | jero_SFLphone | damn |
| 17:04.17 | JerJer | xkev: did you buy a real X100P? |
| 17:04.20 | jero_SFLphone | it's time to move to pri |
| 17:04.35 | shaZwaz | later guys |
| 17:04.36 | shaZwaz | nite |
| 17:04.49 | Essobi | How do I see how many channels have a G729 codec in use? |
| 17:05.00 | jero_SFLphone | is my caller id problem digium or bell related ? |
| 17:05.07 | Essobi | NM |
| 17:05.08 | shaZwaz | show g729 |
| 17:05.09 | Essobi | I found it |
| 17:05.10 | Essobi | ;) |
| 17:05.21 | shaZwaz | show sip channels as well |
| 17:11.56 | *** join/#asterisk jets (~jetsn@guardian.pmt.org) |
| 17:14.47 | *** join/#asterisk buleriando (~chatzilla@f96088.upc-f.chello.nl) |
| 17:16.07 | *** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk) |
| 17:16.35 | *** join/#asterisk jets (~jetsn@guardian.pmt.org) |
| 17:25.00 | redder86 | bjohnson: which ones? |
| 17:25.11 | fishboy1669 | home time |
| 17:25.13 | fishboy1669 | bye |
| 17:27.21 | *** join/#asterisk roamer323 (~sing@HSE-MTL-ppp72652.qc.sympatico.ca) |
| 17:28.29 | roamer323 | hi do I need to install zaptel and have at least one zaptel board to get IAX2 timing working right? thx |
| 17:29.43 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 17:31.16 | Zeeek | bjohnson what providers ? :) |
| 17:31.58 | PatrickDK | roamer, you need to install a zaptel device, or use ztdummy |
| 17:32.24 | PatrickDK | that is only if you want to use iax2 in trunk mode though |
| 17:32.30 | roamer323 | patrickdk - thanks. |
| 17:33.00 | roamer323 | patrickdk - the demo connects okay ... does it use iax2 in truck mode? |
| 17:33.11 | PatrickDK | no, iax2 without trunk |
| 17:33.31 | PatrickDK | trunk is when you want to muliplex more than one call between the same two machines |
| 17:33.41 | PatrickDK | without using two different network packets, but send them in the same packet |
| 17:34.03 | Zeeek | truck mode is slow - you truck the packets between two locations |
| 17:34.04 | PatrickDK | it saves bandwidth when the source and end of each iax2 call is the same, otherwise you can't use trunk |
| 17:34.09 | Zeeek | major lag :) |
| 17:34.30 | PatrickDK | none-trunk just means you don't save bandwidth |
| 17:34.40 | roamer323 | patrickdk - I see, so if I connect IAX2 to iaxtel or FWD... that is not trunk mode, and should work okay with no zaptel |
| 17:34.44 | *** join/#asterisk NTJOCK (~brian@txshirts.com) |
| 17:34.45 | PatrickDK | zeeek, shouldn't be no more than 20ms lag at the most |
| 17:34.51 | NTJOCK | Hello. |
| 17:34.53 | Zeeek | not with truck mode! |
| 17:34.59 | Zeeek | trucks aren't that fast |
| 17:35.03 | PatrickDK | roamer, ya |
| 17:35.08 | riksta | any of you guys been using ADM? i'd appreciate some feedback...bugs etc |
| 17:35.17 | PatrickDK | zeeek, hmm, all trunk mode does is multiplex the voice |
| 17:35.23 | PatrickDK | it has to wait for voice packets to do that |
| 17:35.24 | NTJOCK | I have an issue that I'd like help on where to look for the cause. I've searched Wiki, docs, asteriskdocs. |
| 17:35.24 | *** join/#asterisk emptee (empty@beetle.ispnet.ca) |
| 17:35.28 | PatrickDK | max lag of voice packet is 20ms |
| 17:35.37 | PatrickDK | so max lag of iax trunk vs non-trunk is 20ms |
| 17:35.46 | Zeeek | I said TRUCK actually I thought (tm) trunk mode was for several simultaneous channels |
| 17:35.56 | roamer323 | patrickdk - 1.0.5 , I get garbled sound... I wonder if iaxtel and FWD use newer asterisk? |
| 17:35.56 | Zeeek | as in "duh truck is heah" |
| 17:35.58 | NTJOCK | SIP to ZAP call outbound.... call connects, call proceeds... about 2 minutes in it "vanishes" and asterisk shows it as a hangup. |
| 17:36.27 | PatrickDK | roamer, dunno what that is |
| 17:36.49 | PatrickDK | I know alot of voip people can't do trunk mode, I know voicepulse can't |
| 17:36.59 | Zeeek | roamer323 you fdon't need trunking with FWD |
| 17:36.59 | roamer323 | patrickdk - thanks a gazillion :-) |
| 17:37.08 | *** join/#asterisk Bonbon (~bonbon@83.146.53.34) |
| 17:37.19 | *** join/#asterisk dsfr (~dsfr@216.207.244.183) |
| 17:37.22 | Bonbon | guys, is anyone using the swissvoice phones with sip? |
| 17:37.57 | roamer323 | everybody here seems to use cvs-head... makes me feel like I'm in stone age |
| 17:38.19 | abbas_ | who provides toll free number in US and UK? |
| 17:38.45 | PatrickDK | heh, I need to fix my setup orsomething, cvs head has been crashing my box since a few months ago |
| 17:38.56 | NTJOCK | you may have mixed driver versions |
| 17:39.01 | NTJOCK | I had that problem. |
| 17:39.19 | PatrickDK | hmm, you mean as far as zaptel drivers go? |
| 17:39.20 | NTJOCK | I deleted (purged) /etc/asterisk (make a copy of it) and then the /usr/src directories that apply |
| 17:39.23 | NTJOCK | and grab new copies |
| 17:39.30 | NTJOCK | I think it was a mixup in zaptel and libpri |
| 17:39.39 | NTJOCK | but I zapped all of it and regot from CVS |
| 17:39.45 | NTJOCK | I had mixed luck with deb packages |
| 17:39.46 | PatrickDK | hmm, I always delete all src |
| 17:39.51 | NTJOCK | k |
| 17:39.56 | NTJOCK | i'm running deb/sarge 2.6 |
| 17:39.58 | PatrickDK | and cvs zaptel zapata libpri asterisk each time |
| 17:40.00 | NTJOCK | well behaved for me |
| 17:40.09 | NTJOCK | except for this outbound call limit thing that I can't seem to figure out |
| 17:40.18 | PatrickDK | it was on a p133 box though |
| 17:40.52 | PatrickDK | hmm, I need to get ospf setup on the new network |
| 17:41.13 | roamer323 | anyone knows if the tmcnet show in miami is worth going to? or is it mainly s2s (suit 2 suit)? |
| 17:41.17 | *** join/#asterisk gdb (~cbell@circe.inetdb.com) |
| 17:42.00 | emptee | anyone here developed ip centrex with asterisk? |
| 17:42.44 | buleriando | Hi, can anybody give me a pointer here. No luck with Google, mailing lists, etc. |
| 17:42.52 | buleriando | I've got 2 HFC cards, one TE, one NT, bristuff 0.2.0-RC5 with Florian Zumbiehl's patches |
| 17:43.16 | buleriando | after about 10-20 hours i start getting chan_zap.c:7411 zt_pri_error: PRI: !! Got a UA, but i'm in state 1 |
| 17:43.36 | Bonbon | does anyone know about Sysmaster? |
| 17:43.50 | buleriando | and I need to restart * and reload the zapfhc module |
| 17:44.04 | emptee | Sysmaster gave me a quote once. |
| 17:44.07 | emptee | it was large. |
| 17:44.11 | emptee | like 20k |
| 17:45.03 | Bonbon | it's not based on asterisk in any way is it? |
| 17:46.02 | florz | buleriando: But for the first 10-20 hours, it works without problems? |
| 17:46.16 | buleriando | yes, perfectly |
| 17:46.18 | *** join/#asterisk TekLexus (~Mnemonic@167.206.75.24) |
| 17:47.03 | *** join/#asterisk harryvv (~comming@S010600055d210201.vs.shawcable.net) |
| 17:47.47 | Essobi | Mmm. |
| 17:48.10 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com) |
| 17:48.15 | buleriando | I get D-channel on span 1 up/down msgs every minute or so |
| 17:48.18 | Essobi | I'm tring to setup a queue with members on a remote SIP peer.. anyone know how I would do that? |
| 17:48.39 | buleriando | but that is ok according to various docs, but |
| 17:48.42 | Essobi | I thought member => Sip/peername/number would work.. |
| 17:48.46 | Essobi | but it doesnt.. |
| 17:49.06 | buleriando | the chan_zap errors are synced with this |
| 17:49.54 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 17:50.18 | *** join/#asterisk jsandnes (cryzeck@sms.trangest.no) |
| 17:50.21 | jsandnes | Hey guys, I need some help here :-) - I'm working on a open source Web management interface for asterisk, but i would need some tips from a few people what I should include in that interface, what features people want, and I need a good name :-) |
| 17:50.42 | Essobi | YAAMT :) |
| 17:50.51 | jsandnes | YAAMT? |
| 17:50.53 | Essobi | Yet Another Asterisk Management Tool. |
| 17:50.58 | jsandnes | he he :D |
| 17:50.59 | Essobi | Tehe. |
| 17:51.12 | jsandnes | are there any free ones which works good then? |
| 17:51.13 | Essobi | Wait wait, you can't use that.. that one is mine. ;) |
| 17:51.22 | buleriando | florz: sorry, new to irc, should i be here or talking in the main thread? |
| 17:51.23 | Essobi | AMP is about the closest thing to "good" |
| 17:51.35 | Essobi | even thou it's feature set is pretty limited and it's a PITA to setup. |
| 17:51.37 | jsandnes | Asterisk Management P? |
| 17:51.41 | Essobi | portal |
| 17:51.59 | jsandnes | okay, are you working on one aswell, or? |
| 17:52.00 | florz | buleriando: if you unplug it for a while when that happens, does it work for another 10-20 hours? |
| 17:52.09 | Essobi | and that guys idea of a SQL database.. isn't exactly a robust layout. |
| 17:52.11 | florz | buleriando: main thread? How ya mean? |
| 17:52.24 | Essobi | he needs to go back to database architecture school |
| 17:52.30 | jsandnes | hehe :) |
| 17:52.44 | Essobi | Umm. Yea, a private sourced one, thou likely. |
| 17:52.48 | *** join/#asterisk beto75 (~ha@201.128.177.84) |
| 17:52.52 | jsandnes | heh, ok :) |
| 17:52.55 | Essobi | Depends how much of a raise they want to give me for one. :) |
| 17:53.05 | Essobi | I've wrote one for a client already. |
| 17:53.17 | Essobi | that fit his needs.. |
| 17:53.22 | Essobi | but I need to write a new one. |
| 17:53.35 | Essobi | thats all mojo voodooey |
| 17:53.40 | jsandnes | What are you writing in? |
| 17:53.54 | buleriando | florz: I see your msgs on #asterisk and on the freenode screen: not sure which one to use |
| 17:53.59 | Essobi | PHP, and some perl for the go between probably |
| 17:54.14 | Essobi | I don't like the idea of people editing mysql dialplans directly. |
| 17:54.26 | beto75 | hello guys |
| 17:54.27 | mountainm2k | Asterisk@Home help ? |
| 17:54.28 | buleriando | florz: haven't tried unplugging it, will do so next time it hangs |
| 17:54.30 | florz | buleriando: Uh. No clue either. but #asterisk probably won't be wrong =:-) |
| 17:54.31 | roamer323 | mojo voodooey - perfect match for asterisk :-) |
| 17:54.31 | Essobi | So I'll likely slap them into configs from SQL. |
| 17:54.49 | jsandnes | I only use perl for the web and unix manager, and c++ for the windows manager |
| 17:54.49 | dsmouse | what's the diffrence between asterisk and asterisk@home? |
| 17:55.01 | Essobi | Ahh, mines all web based. |
| 17:55.09 | jsandnes | ahh :) |
| 17:55.16 | mountainm2k | asterisk@home is pre-packaged installs OS and builds everything... |
| 17:55.23 | mountainm2k | Figured I'd use it as a starting point... |
| 17:55.27 | Essobi | That's scarey. |
| 17:55.27 | dsmouse | ah |
| 17:55.42 | mountainm2k | It defines an "internal" extension directory at *411 (or maybe that's standard?) |
| 17:55.47 | Essobi | asterisk@home is a pre-broken * install. ;) |
| 17:56.06 | Essobi | Yea, it's probably SEMI amp based. |
| 17:56.20 | mountainm2k | Anyway, I have two extensions, 210 and 211, both using X-Lite... Both log in, can call each other... |
| 17:56.29 | Essobi | I'm going to do something similar to AMPs idealogy but multi-locals. |
| 17:56.43 | mountainm2k | But anything to do with vmail, I get connected, but no audio |
| 17:56.56 | Essobi | Back to the lab. |
| 17:56.59 | mountainm2k | I can see by the NIC it's sending data out, but I can't hear anything... |
| 17:57.30 | harryvv | mountain, can use use both to make voice calls to other services? |
| 17:57.54 | roamer323 | mountain - have you tried disabling all the codec on the xlites except for GSM? |
| 17:58.12 | harryvv | roamer, that should not make a difference i have the same setup |
| 17:58.40 | harryvv | all codes are loaded in my config. |
| 17:58.44 | mountainm2k | I have tried diff. codecs... GSM doesn't seem to work -- if I un-click everything but GSM, I get "Call failed: 499 Not Acceptable Here" |
| 17:58.49 | mountainm2k | and a fast busy |
| 17:59.20 | mountainm2k | but all the others work, just no audio... :-P |
| 17:59.20 | harryvv | mountainm2k what does cli say when making a call from softphone to softphone |
| 18:00.09 | mountainm2k | actually I just tried iLBC and that doesn't work either... |
| 18:01.24 | mountainm2k | X-Lite seems to like G711u |
| 18:01.37 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 18:01.52 | harryvv | mountainm2k mine is also g711u |
| 18:01.54 | roamer323 | mountain -> is there any NAT between the xlites and the asterisk? This may be either a linksys/dlink box, etc, or the M$'s internet sharing enabled on XP/2000, etc? |
| 18:02.26 | mountainm2k | Nope, corp. network, logically and physically flat... |
| 18:02.30 | mountainm2k | IE one subnet |
| 18:02.35 | jsandnes | Anyone which want to help me to create some ideas on how to make some good features in asterisk? |
| 18:02.37 | riksta | any of you guys been using ADM? i'd appreciate some feedback...bugs etc |
| 18:02.38 | harryvv | roamer323 ms internet sharring would interfear with ths? |
| 18:02.53 | mountainm2k | I even turned off XP SP2 firewall |
| 18:03.35 | mountainm2k | Two X-Lite's can call eachother through *, but neither can hear audio when calling *411 or *98 (vm access) |
| 18:03.41 | dsmouse | jsandnes: non-meetme confrences? i.e. conf two callers togeather off the cuff... (or do I just need to read more web pages...) |
| 18:03.43 | harryvv | when you speak into the microphone do you see the speaker volume bar light up on the interface? |
| 18:04.30 | mountainm2k | When NOT connected? Or when connected to VM? |
| 18:04.39 | *** join/#asterisk Guy- (~korn@chardonnay.math.bme.hu) |
| 18:04.42 | harryvv | lets say when connected |
| 18:04.48 | Guy- | hi |
| 18:05.15 | roamer323 | harryvv -> yes... ms internet sharring is a NAT |
| 18:05.21 | mountainm2k | harryvv: hang on, trying it... |
| 18:05.47 | harryvv | might be a sound releated issue. |
| 18:05.51 | Guy- | all: I'd like to estimate how much it would cost to deploy asterisk at a smallish company - any good primers to read from that perspective? |
| 18:05.52 | mountainm2k | (heh, desktop no microphone, only speaker) |
| 18:05.59 | Guy- | currently, I'm not even sure what we'd need to buy |
| 18:06.05 | harryvv | you have no microphone? |
| 18:06.35 | mountainm2k | Guy: Looking into this myself, as well... we use cBeyond Communications, T1 w/ IP, and then VoIP to a Cisco router... |
| 18:07.03 | mountainm2k | Harryvv: laptop has microphone, desktop only headphones at the moment... |
| 18:07.04 | Guy- | basically, I'd like to use Asterisk as a PBX - I'm not sure what kind of hardware I need to install in the would-be server for it to be able to act as one |
| 18:07.36 | harryvv | guy, get to know asterisk though and though before going to step two and setup a lab to make sure what works there will work in a clients site if you can. |
| 18:07.57 | Guy- | mountainm2k: my situation is completely different - we have three offices in three countries, already connected via IP, and would like to make phone calls between them 'free' |
| 18:08.01 | mountainm2k | harryvv: Noise on microphone causes RED bar to go up, but not green speaker bar... |
| 18:08.21 | NTJOCK | is anyone here very familiar with how a outbound call should progress on a POTS line in *? |
| 18:08.38 | NTJOCK | * doesn't seem to be recognizing the call progress and is disconnecting the call |
| 18:08.58 | Guy- | harryvv: knowing it through and through will, perhaps, come once I begin to use it, but we're too small to to set up a lab or anything - the lab will be the first live server :) I'm reading the handbook draft, but I thought maybe there was something more specific |
| 18:09.08 | roamer323 | jsandes - call by call failover across a mesh of * |
| 18:09.28 | hajekd | any good provider for pstn termination in Europe? |
| 18:09.36 | harryvv | mountain, the green side if for incomming calls |
| 18:09.53 | mountainm2k | figured... "speaker volume"... |
| 18:09.55 | dca[laptop] | hajekd: teliax |
| 18:10.47 | Guy- | OK, I have a specific question: what kind of hardware would I need to terminate a handful (2-3) of ISDN BRIs in a PC and use them with Asterisk? |
| 18:10.56 | roamer323 | mountain - have you tried turning on "sip debug" on the * CLI? |
| 18:10.57 | FuRR_ | Guy-: grab the yellow book from signate |
| 18:11.14 | Guy- | FuRR_: can you give me a URL? |
| 18:11.27 | hajekd | dca: you have some experiences with them? |
| 18:11.43 | *** join/#asterisk Frod (~Frod@201.135.179.199) |
| 18:11.46 | Guy- | FuRR_: ah, found it |
| 18:11.53 | dca[laptop] | hajekd: well, i work with them so i'm definitely biased, but yes :) |
| 18:12.11 | Frod | hello all |
| 18:12.19 | dca[laptop] | hello Frod! |
| 18:12.34 | Frod | does aqny one know it azacall 200 is able to reinvite ?? |
| 18:13.02 | dsmouse | jsandnes: AgentLogout() |
| 18:14.52 | Essobi | One of the most aggravating things about *.. Is I have to wade into source code to find out why something doesn't work. |
| 18:15.21 | Essobi | You can't add a queue member that is on a remote SIP peer, which is shite. |
| 18:15.39 | shido6 | hangovers suck |
| 18:15.47 | shido6 | can I get an amen from the choir? |
| 18:15.59 | shido6 | are hangovers related to dehydration? |
| 18:16.09 | Essobi | Yes. |
| 18:16.26 | Essobi | and amides.. I think that's the word for them. |
| 18:16.32 | shido6 | I shoulda grabbed some pedialite before givin the car to the lady |
| 18:16.39 | dsmouse | shido6: howstuffworks.com has a great artical about hangovers |
| 18:16.45 | shido6 | thanks dsmouse |
| 18:16.54 | roamer323 | essobi - distributed queues... a project for 2008 |
| 18:18.58 | FuRR_ | Gary-: AVM Fritz makes a 4port BRI Card that would work with chan_capi |
| 18:19.01 | hajekd | dca: they offer iax? you have the IP of the gateway? |
| 18:19.22 | Silik0n | distributed queues would be nice |
| 18:20.47 | mountainm2k | harryvv: Any other suggestions for my audio? |
| 18:20.48 | Essobi | Pssh, it's retarded. |
| 18:20.53 | mountainm2k | or anybody else either? |
| 18:20.53 | Essobi | I can dial a SIP peer |
| 18:21.00 | Essobi | why can't I dial a remove sip peer |
| 18:21.06 | Essobi | remote even |
| 18:21.27 | *** join/#asterisk r1 (~erwan@www.thiscow.com) |
| 18:21.49 | NTJOCK | Any ideas on why a call might be scheduled for destruction after it gets started? I'm having my outbound calls destroyed and I can't figure out why |
| 18:21.50 | NTJOCK | :( |
| 18:21.57 | *** join/#asterisk clint_ (~clint@snap.helixsystems.com) |
| 18:21.58 | roamer323 | mountain -> when you turn on sip debug * CLI... in the From, to, and SDP fields of the SIP messages -> check all the IP address |
| 18:23.13 | roamer323 | mountain -> xlites do super voodo-magic in talking through NATs to other xlites and SERs, but * needs all the help it an get |
| 18:23.51 | dsmouse | Essobi: did you 'sip debug' and see what it said? |
| 18:24.51 | harryvv | mountain not at this moment. |
| 18:25.35 | mountainm2k | I'm looking at 'sip debug'... All the IP's look fine... me == 192.168.43.216, * == 192.168.43.215... No other IP's in ther.e.. |
| 18:25.53 | mountainm2k | any X-Lite settings I should check to turn OFF any nat stuff? |
| 18:26.34 | roamer323 | mountain -> did you turn OFF silence supression on the xlite??? |
| 18:26.37 | mountainm2k | Actually, how could it be X-Lite, since I can call from one xlite to the other through * ??? |
| 18:26.50 | mountainm2k | Yes, it's set to transmit silence -> yes |
| 18:27.20 | roamer323 | mountain -> the * would reinvite and the xlite and xlite are talking directly |
| 18:27.46 | mountainm2k | ah, I guess that would make sense... |
| 18:28.01 | roamer323 | mountain -> so prob is between xlite & * , and you can try isolating it there |
| 18:28.32 | mountainm2k | any suggestions? |
| 18:28.51 | NTJOCK | how do you disable sip debug? |
| 18:28.56 | mountainm2k | sip no debug |
| 18:29.03 | roamer323 | mountain -> the sip messages has *all* the information you need to track this down... so, just take your time |
| 18:29.03 | NTJOCK | thanks |
| 18:29.27 | *** part/#asterisk Ogun (~kvirc@h127n2fls34o865.telia.com) |
| 18:29.28 | roamer323 | brb |
| 18:34.01 | mountainm2k | Well, thus far I don't see anything in sip debug messages that would indicate an error |
| 18:34.28 | Connor- | bkw_ wake up, check pvt msg please |
| 18:37.44 | Silik0n | Redneck IVR Prompt recording now available msg Silik0n for details |
| 18:38.18 | roamer323 | mountain -> since you're seeing RTP traffic on the LEDs, there is no error... most likely the asterisk is sending the voice, but the xlite is not listening where it should |
| 18:38.19 | Essobi | dsmouse queue won't even attempt to add a member of "member => SIP/as5400-1/5551212" |
| 18:38.48 | Essobi | or SIP/5551212@5400.s.ip.addy |
| 18:38.53 | roamer323 | mountain -> look into the "body" or SDP part of the SIP messages and check the IP addresss embedded there |
| 18:40.38 | *** join/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu) |
| 18:40.47 | PakiPenguin | uk is 220? |
| 18:41.10 | PakiPenguin | we have 220 -240 here in pk , what should i get , AU ,UK or EU adaptor? |
| 18:42.06 | Essobi | Anyone know how I get all the pretty ast_debug stuff to drop to my console when an app runs? |
| 18:42.44 | Silik0n | set debug like set verbosity |
| 18:42.54 | roamer323 | essobi - can you "tee" stderr to the tty? |
| 18:43.08 | Essobi | I've used it but I don't see anything.. |
| 18:43.47 | roamer323 | essobi - hate to say this again... check the source code... maybe it didn't use stderr :-) |
| 18:44.55 | Silik0n | did you try set debug? |
| 18:45.54 | Guigui|taff | eKo1: with this wiki, I successfully send a fax to asterisk (with rx/txfax module) : http://scottstuff.net/scott/archives/000152.html |
| 18:47.24 | *** join/#asterisk flewid (~flewid@CPE0050ba8c9a95-CM000f9fac6da2.cpe.net.cable.rogers.com) |
| 18:47.26 | flewid | sup |
| 18:49.00 | Essobi | GRRR, Yes. |
| 18:49.17 | Essobi | there's a global debug car for half of the debug statements |
| 18:49.22 | Essobi | var |
| 18:49.27 | Essobi | in asterisk.c |
| 18:49.29 | Essobi | :| |
| 18:49.44 | Essobi | I hate the idea I have to recompile just to do some debugging |
| 18:50.08 | *** join/#asterisk gabb0 (~gabb0@CPE0006258dff02-CM000a73661510.cpe.net.cable.rogers.com) |
| 18:50.59 | Connor- | Anyone using a SPA-2000 behind nat with asterisk on a public? |
| 18:51.36 | Delvar | anyone got an iptell number i can ring for a test? |
| 18:52.01 | *** join/#asterisk JunK-Y (~grepmoo@65.39.228.5) |
| 18:52.13 | gabb0 | hello all |
| 18:53.16 | JunK-Y | ive a core file, when im doing gdb -core (corefile) |
| 18:53.19 | JunK-Y | im getting |
| 18:53.23 | JunK-Y | Core was generated by `asterisk -vvvg -c'. |
| 18:53.23 | JunK-Y | Program terminated with signal 11, Segmentation fault. |
| 18:53.23 | JunK-Y | #0 0x4014825e in ?? () |
| 18:53.23 | JunK-Y | (no debugging symbols found)...Using host libthread_db library "/lib/tls/libthread_db.so.1". |
| 18:53.23 | JunK-Y | (gdb) |
| 18:53.25 | mountainm2k | roamer323: Sorry, bloody phonecalls keep interrupting me... I looked through a complete sip-log, all IP's look right. |
| 18:53.28 | JunK-Y | what that means exactly? |
| 18:53.36 | mountainm2k | Is there another soft-phone I could/should try instead? |
| 18:54.53 | mountainm2k | Hey, now I'm getting little "blips"... |
| 18:54.59 | mountainm2k | hearing them I should say... |
| 18:55.26 | *** join/#asterisk _Raptor_ (RaptorX@p54805194.dip.t-dialin.net) |
| 18:55.30 | _Raptor_ | hi |
| 18:57.13 | _Raptor_ | i hope you can help me finding out what's going wrong here: i want set up a h323 conferencing server and i have installed zaptel and meetme but when i am calling the server i get this in the asterisk console: |
| 18:57.16 | _Raptor_ | Feb 15 19:41:50 WARNING[5951]: app_meetme.c:227 build_conf: Unable to open pseudo channel - trying device |
| 18:57.31 | _Raptor_ | Feb 15 19:41:50 WARNING[5951]: app_meetme.c:230 build_conf: Unable to open pseudo device |
| 18:57.31 | _Raptor_ | <PROTECTED> |
| 18:57.31 | _Raptor_ | <PROTECTED> |
| 18:59.05 | *** part/#asterisk _PiGreco_ (~a@adsl-215-48.38-151.net24.it) |
| 19:01.38 | anthm | load zaptel drivers "modprobe ztdummy" |
| 19:02.27 | mountainm2k | Any other free (or at least shareware) SIP phones I could try? |
| 19:02.50 | *** join/#asterisk zno (~zeno@ip-160-79-174-99.autorev.intellispace.net) |
| 19:03.39 | *** join/#asterisk CSerpent (~me@62.49.253.91) |
| 19:03.52 | CSerpent | evening all <> |
| 19:04.51 | *** join/#asterisk dsfr (~dsfr@216.207.244.183) |
| 19:05.02 | CSerpent | quick question if I may.. in extensions, I want to strip the leading digit from a CID for sending to Zap.. but want to leave it in there for SIP |
| 19:05.35 | CSerpent | so I have exten => 942,2,Dial(SIP/phone1&SIP/206)&SetCidNum(${CALLERIDNUM:1})&Dial(Zap/g3/${EXTEN},30,Ttr) - but it's not recognising the second dial on the same time |
| 19:05.40 | roamer323 | mountain -> xlite works great, no need to switch |
| 19:06.24 | mountainm2k | Well, it's not working for me... :-P |
| 19:06.37 | mountainm2k | Could it be I have something mucked up? |
| 19:07.06 | KalD|Work | does anyone know of a license-free version of libiax that can be used in commercial applications? |
| 19:07.27 | _Raptor_ | anthm: thx, but module not found, is this part of asterisk oder of kernel sources? |
| 19:07.45 | anthm | cvs co zaptel |
| 19:07.56 | anthm | cvs co libpri |
| 19:08.13 | anthm | make install in reverse order that they are listed |
| 19:08.23 | roamer323 | mountain -> can you paste an RTP infomation line from one of your SDP body - here |
| 19:08.57 | mountainm2k | from sip debug ? |
| 19:09.19 | roamer323 | yes - but just a single line with the RTP info |
| 19:09.24 | _Raptor_ | anthm: thx |
| 19:10.04 | Essobi | Jees. |
| 19:10.14 | mountainm2k | Sip read: |
| 19:10.14 | mountainm2k | INVITE sip:*411@192.168.43.215 SIP/2.0 |
| 19:10.14 | mountainm2k | Via: SIP/2.0/UDP 192.168.43.216:5060;rport;branch=z9hG4bK52CD4A15BED24BC0B2EA677037F2ABA7 |
| 19:10.14 | mountainm2k | From: Matt Sturtz <sip:210@192.168.43.215>;tag=2097331258 |
| 19:10.14 | mountainm2k | To: <sip:*411@192.168.43.215> |
| 19:10.14 | mountainm2k | Contact: <sip:210@192.168.43.216:5060> |
| 19:10.16 | mountainm2k | Call-ID: BB378991-96B5-42CA-9F3C-9773FB0FE2FA@192.168.43.216 |
| 19:10.18 | mountainm2k | CSeq: 30843 INVITE |
| 19:10.20 | mountainm2k | Max-Forwards: 70 |
| 19:10.20 | Essobi | Hey anthm .. you got any idea how to add remote SIP peers to a queue? |
| 19:10.24 | mountainm2k | Content-Type: application/sdp |
| 19:10.24 | mountainm2k | User-Agent: X-Lite release 1103m |
| 19:10.26 | mountainm2k | Content-Length: 300 |
| 19:10.29 | mountainm2k | Could keep going from there? |
| 19:10.42 | KalD|Work | mountainm2k, if you do please use pastebin |
| 19:11.10 | Essobi | Or anyone for that matter have an idea how to do that? |
| 19:11.55 | mountainm2k | pastebin |
| 19:12.01 | mountainm2k | nope, that didn't do it... ;-P |
| 19:12.13 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 19:12.24 | KalD|Work | mountainm2k, are you using jitter control on Xten? |
| 19:12.45 | mountainm2k | Checking, where is that in menu |
| 19:13.06 | KalD|Work | i think it is under audio or advanced (I havent used xten in like 6 mo) |
| 19:13.09 | PBXtech | is the PC bus speed the cause of the lag on the digum cards which causes echo? |
| 19:13.20 | KalD|Work | also look in your asterisk conf for jitter in the sip.conf |
| 19:13.47 | mountainm2k | "Hold In Jitter Buffer(ms)" -< 100 |
| 19:14.15 | Essobi | MONEY! |
| 19:14.22 | Essobi | God why wasn't that obvois. |
| 19:14.28 | Essobi | Obvious even. |
| 19:14.46 | KalD|Work | mountainm2k, try playing w/ that... also what sound hardware you using? internal or external? |
| 19:14.52 | Essobi | member => Local/patterninyourdialplantocalloutlocalsipppeer |
| 19:14.54 | Essobi | :P |
| 19:15.16 | Essobi | Docs need updated in queues to reflect that Local is an available channel to. |
| 19:15.44 | *** join/#asterisk darkskiez (~mhb@host-84-9-70-218.bulldogdsl.com) |
| 19:19.40 | mountainm2k | set the jitter buffer in xlite to 0 but no difference... |
| 19:21.08 | *** join/#asterisk hajekd (~hajekd@21.208.65.212.contactel.net) |
| 19:21.21 | PBXtech | Is there a PCI express version of a QuadT1 digium card in development? |
| 19:21.52 | roamer323 | mountain -> are you getting any error/msg on the xlite diagnostic window? |
| 19:22.16 | KalD|Work | PBXtech, are you getting echo on digium hardwware? |
| 19:22.42 | PBXtech | yea, im just thinking i need abetter motherboard. |
| 19:22.53 | PBXtech | im sure someone has tested this |
| 19:23.06 | Silik0n | is there really a need for PCI-Express for a quad T1 card? theres only 6 meg of bandiwdth needed if all channels are in use |
| 19:23.07 | mountainm2k | roamer323: Doesn't appear so... SIP msgs, nothing indicating an error... I could paste here but how to pastebin? |
| 19:23.12 | KalD|Work | do you have echocanel etc set in /etc/asterisk/zapata.conf ? |
| 19:23.19 | KalD|Work | PBXtech, do you have echocanel etc set in /etc/asterisk/zapata.conf ? |
| 19:23.29 | PBXtech | yes |
| 19:23.59 | PBXtech | [Silik0n]: what chipset on a v5 card can handle full 6 meg of bandwidth |
| 19:24.28 | dsmouse | voip-info.org isn't responding :( |
| 19:24.40 | KalD|Work | dsmouse, is for me |
| 19:24.55 | djin | here too. |
| 19:24.55 | *** part/#asterisk Darien (sentry21@mtl.rackplans.net) |
| 19:25.08 | KalD|Work | dsmouse, actually faster now for me than it was last week =-\ |
| 19:25.28 | terrapen | 54-46 Was My Number |
| 19:25.38 | terrapen | Right now, somebody else has that number (one more time!) |
| 19:26.38 | PBXtech | having a slow MB would cause echo because of the latency right? |
| 19:26.43 | bjohnson | found what looks like a deal for Canadians wanting to get about $0.05/minute (prepay style accounts) for cell phone long distance (dial a number, call out from there concept) .. has nothing to do with voip but still is communications related. www.xpresscall.com .. get 40 free minutes (I have info to get another 40 free minutes) .. and account credits never expire |
| 19:27.15 | harryvv | pre pay huu |
| 19:27.24 | KalD|Work | PBXtech, hmmm how do you have it hooked up? I have a single span T1 in my P166 and I get no echo on 8 line conf w/ it |
| 19:27.26 | harryvv | bj i have bell mobility |
| 19:27.31 | bjohnson | so do i |
| 19:27.37 | bjohnson | $.25/minute |
| 19:27.39 | harryvv | have you tried it yet? |
| 19:27.47 | bjohnson | no .. just found out about it today |
| 19:27.50 | PBXtech | i have a quad T1 (1-LD 1-channel bank) and a 4 port FXO |
| 19:27.59 | harryvv | mine is what 30 cents per min prepay which is very expensive. |
| 19:27.59 | Himeko | bjohnson you could do that with your * box |
| 19:28.17 | KalD|Work | PBXtech, do you get echo between those two cards? |
| 19:28.17 | bjohnson | Himeko: not for that cheap .. plus you'd tie up a pstn line |
| 19:28.46 | dsmouse | ok, it's just the page google finds for "asterisk cmd-congestion" on viop-info that does't work. weird |
| 19:28.57 | Himeko | you can't get 5c/m north america LD? |
| 19:29.04 | PBXtech | im not sure, it passes through |
| 19:29.27 | KalD|Work | PBXtech, i think you'd want to move each digium card to a seperate box... the older cards (pre-digium mfg) used all the bus bandwidth - and I imagine a quadspan card comes close to max'n a newer PCI bus |
| 19:29.51 | PBXtech | its a quad card |
| 19:30.25 | *** join/#asterisk roamer323 (~sing@HSE-MTL-ppp72652.qc.sympatico.ca) |
| 19:30.26 | KalD|Work | PBXtech, yeah but didnt you say you also have a 4port FXO board in there too? |
| 19:30.35 | PBXtech | yes |
| 19:30.45 | KalD|Work | PBXtech, try taking that out |
| 19:31.01 | PBXtech | does the quad card still use all the power if its only using 2 of the 4 T1's? |
| 19:31.21 | bjohnson | Himeko: not from the cell phone |
| 19:32.21 | KalD|Work | PBXtech, i dunno - if it is busmaster (is that the correct term for PCI?) it could use all ... and if they both try to busmaster you'd be having issues |
| 19:32.37 | bjohnson | Himeko: cheapest I've seen is $.10 but I chose a package with other options since I don't use cell phone much .. current plan is $.25/minute cell phone LD |
| 19:32.43 | PBXtech | ok |
| 19:33.09 | Himeko | http://www.wintel.ca/ldplans/index.htm |
| 19:33.24 | roamer323 | mountain -> www.pastebin.com (then just put the URL here) |
| 19:34.28 | JunK-Y | when im generating calls, ive some: |
| 19:34.28 | JunK-Y | Feb 15 14:33:22 WARNING[26289]: channel.c:500 ast_channel_walk_locked: Avoided initial deadlock for 'SIP/10.0.0.2-6fd39440', 10 retries! |
| 19:34.33 | JunK-Y | any ideas ? |
| 19:35.00 | *** join/#asterisk mrempire (~user1@h71032.upc-h.chello.nl) |
| 19:35.11 | bjohnson | Himeko: same idea but this has a free 800 number that works in all of Ontario without extra charges, is cheaper per minute, and I can get 80 free minutes to start |
| 19:35.14 | KalD|Work | JunK-Y, do you have dual proc? |
| 19:35.18 | Himeko | hmm, i like that 25c call deal |
| 19:36.48 | bjohnson | Himeko: depends on your calling patterns .. would be cheaper if your average LD call was more than 12.5 minutes |
| 19:36.54 | *** join/#asterisk human39 (~human39@chewie.fyi.net) |
| 19:37.02 | human39 | afternoon all. |
| 19:37.06 | bjohnson | thanks to * at home .. I hope to actually get that type of statistic |
| 19:37.10 | Himeko | i make very few ld calls |
| 19:37.36 | *** join/#asterisk stepcut (~user@207.67.194.2) |
| 19:37.37 | human39 | I was wondering if somebody can recommend a ~$100 IP phone that is good for its price. |
| 19:37.55 | Himeko | the only time i tend to make ld calls from the cell phone is when i am travelling |
| 19:38.01 | Mneumonic | human39 - Sipura 835 is nice |
| 19:38.35 | bjohnson | Himeko: me too .. that's why I like the pay as you go approach, the non-expiring account credit, and the free 80 minutes (probably enough for me for a year) |
| 19:38.44 | human39 | Mneumonic, thanks. |
| 19:38.55 | bjohnson | Sipura 835? |
| 19:38.57 | Mneumonic | err = 841 |
| 19:38.59 | bjohnson | I thought it was the 841 |
| 19:39.00 | Mneumonic | my bad |
| 19:39.17 | Himeko | but wouldn't you break even at 5min not 12.5 min |
| 19:39.30 | JunK-Y | KalD|Work: yes, why? |
| 19:40.41 | bjohnson | Himeko: I don't think the $.25 / call deal is available to cell phone users .. it's in a different section .. so for that I compare against my voipjet or livevoip outgoing at $1.3USD / minute |
| 19:41.43 | Himeko | on the wintel site it doesn't say which ones are not available for wireless use |
| 19:41.51 | Qwell | $1.30 per minute? |
| 19:42.04 | Himeko | just a footnote saying some may not be available |
| 19:42.13 | Himeko | 1.3c |
| 19:42.17 | Qwell | oh |
| 19:42.57 | mountainm2k | damn it, just getting frustrated here, I'm giving it up for a bit... |
| 19:43.56 | bjohnson | Qwell: put the decimal in the wrong spot |
| 19:44.12 | bjohnson | $0.013 |
| 19:44.17 | Qwell | ahh |
| 19:44.35 | netsurfer | hey Qwell |
| 19:44.39 | bjohnson | $0.012 from livevoip right now |
| 19:45.36 | znoG | anyone registered with sipphone or iaxtel? |
| 19:45.37 | human39 | does anybody have the Sipura 841? |
| 19:45.49 | *** join/#asterisk Weezey (Weezey@lan6.LO.iasl.com) |
| 19:45.59 | Weezey | can asterisk be a mgcp client? |
| 19:45.59 | bjohnson | zno: registered with iaxtel but given up on trying to use it |
| 19:46.02 | Qwell | bjohnson: lowest cost routing, heh |
| 19:46.02 | netsurfer | lol bjohnson me too.. iv got so many in my extensions.conf I cant remember the dial prefixes for half of them |
| 19:46.07 | mishehu | has anybody used an IAXy with a fax machine, and if so, how well would you say it worked? |
| 19:46.10 | stepcut | znoG: I am registered with sipphone |
| 19:46.15 | znoG | bjohnson: ah ok.. i got outgoing calls working, just wondering if incoming works |
| 19:46.17 | bjohnson | Weezey: I think so .. there is a mgcp.conf file |
| 19:46.26 | znoG | stepcut: can i ask you to dial my sipphone number? just to check i'm registered and alive with them |
| 19:46.31 | Weezey | bj: last I heard it could only be a server. |
| 19:46.51 | stepcut | znoG: in a minute |
| 19:46.53 | Himeko | if that was a canada wide 1-800 it would be more atractive to me |
| 19:46.55 | znoG | cheers |
| 19:47.05 | stepcut | znoG: you can check your registration status on my.sipphone.com |
| 19:47.06 | Himeko | instead of just ontario |
| 19:47.07 | Qwell | znoG: need your iaxtel tested too? |
| 19:47.13 | znoG | sure, why not :) |
| 19:47.28 | stepcut | znoG: and, they have a number you can call from a regular phone also |
| 19:47.36 | bjohnson | Himeko: yeah .. they have local access numbers but for me in Ont .. the 800 number that works abywhere without paying extra is extra nice |
| 19:47.40 | znoG | stepcut: the virtual numbers? |
| 19:47.44 | znoG | bbs |
| 19:48.07 | Qwell | heh, my outgoing iaxtel doesn't work anymore. fun |
| 19:48.18 | stepcut | http://support.sipphone.com/index.php?_a=knowledgebase&_j=questiondetails&_i=46&nav=+%26gt%3B+%3Ca+href%3D%27index.php%3F_a%3Dknowledgebase%26_j%3Dsubcat%26_i%3D3%27%3ECommon+questions%3C%2Fa%3E |
| 19:48.38 | stepcut | znoG: 1-517-902-0700 and then dial your sipphone number |
| 19:49.06 | Qwell | znoG: Ringing, no answer |
| 19:50.47 | Himeko | ah, the wintel 25c plan is for wireless too |
| 19:51.42 | znoG | stepcut: ahh, great thanks. |
| 19:51.54 | znoG | Qwell: yep just missed your call by a split second :) thanks for the test though |
| 19:52.28 | Weezey | is sipphone.com free? |
| 19:53.41 | stepcut | Weezey: yes, but it cost money to have a virtual number ($5/month) or make outgoing calls (pay as you go) |
| 19:54.40 | Weezey | stepcut: cool, thanks. |
| 19:54.41 | stepcut | I believe incoming (pstn) calls, if you have a virtual number, are free |
| 19:56.10 | Weezey | stepcut: do you know about mgcp support on asterisk? |
| 19:56.26 | stepcut | I do not even know what mgcp stands for :) |
| 19:56.42 | Weezey | heh, MGCP is like SIP as I understand it. |
| 19:56.46 | wasim | only worse |
| 19:56.48 | *** join/#asterisk dsfr (~dsfr@216.207.244.183) |
| 19:56.54 | Weezey | ok |
| 19:57.24 | *** join/#asterisk syslod (~sysglod@65.114.15.70) |
| 19:57.29 | syslod | Sup ppl. |
| 19:57.38 | Weezey | My home phone runs over mgcp, but I want to have asterisk connect to my mgcp account and then my home phone connect to my asterisk |
| 19:58.15 | *** join/#asterisk zoa (zoa@82.103.76.147) |
| 19:58.25 | zoa | yooooooooooooooooow |
| 19:58.32 | Himeko | there is no mgcp client for * |
| 19:58.49 | syslod | Anyone here have BAF/AMI to EMI converter script? |
| 19:59.13 | Weezey | Himeko: that's what I thought, just seeing if that had changed. Thanks. |
| 19:59.32 | Weezey | Himeko: how hard would it be to build one/ |
| 19:59.39 | Himeko | no idea |
| 19:59.45 | Himeko | i am not a programmer |
| 19:59.57 | *** join/#asterisk HitTop (~Jack@host6614613596.biz.tor.fcibroadband.com) |
| 20:00.17 | Himeko | who is your provider? |
| 20:00.32 | *** join/#asterisk three55ml (~who@cs662589-157.satx.rr.com) |
| 20:01.01 | bjohnson | likely Primus |
| 20:01.26 | Himeko | that would be my guess |
| 20:01.50 | Himeko | but is seems primus in the states is using sip |
| 20:02.21 | bjohnson | only primusconnect (or something like that) |
| 20:02.33 | bjohnson | they have 2 different offerings |
| 20:04.09 | *** part/#asterisk Fanguin (~Fanguin@p50818411.dip0.t-ipconnect.de) |
| 20:07.25 | *** join/#asterisk Juggie (agony@CPE00c049d9f271-CM014270110981.cpe.net.cable.rogers.com) |
| 20:08.30 | Juggie | has anyonw written any script such that when a user joins a conference asterisk could play to the entire conference "juggie has joined" etc? |
| 20:09.09 | terrapen | how embarassing |
| 20:09.20 | Juggie | ? |
| 20:09.26 | terrapen | i like to join stealth |
| 20:09.38 | terrapen | and that would be interrupting people, too |
| 20:09.39 | Juggie | well its just an option |
| 20:09.51 | Juggie | i'm writing a conference bridge |
| 20:12.43 | *** join/#asterisk Nix (~Nix@81.213.125.220) |
| 20:12.54 | three55ml | Juggie: I know MeetMe2 plays a sound, I don't know about MeetMe |
| 20:13.10 | three55ml | Wouldn't be too hard to have the user record his name then announce it |
| 20:13.43 | syslod | Juggie: Its already in the current CVS. |
| 20:14.46 | three55ml | Juggie: What features will you be adding not currently in there? |
| 20:15.07 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 20:16.34 | syslod | Anyone worked with BAF files? |
| 20:17.09 | *** join/#asterisk dsfr (~dsfr@216.207.244.183) |
| 20:18.30 | dalabera | what are BAF Files? |
| 20:18.47 | syslod | Bellcore files. Like CDR but more complex. |
| 20:19.29 | znoG | stepcut: 1-517-902-0700 rings out for me (¿?) |
| 20:19.32 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
| 20:19.32 | *** mode/#asterisk [+o anthm] by ChanServ |
| 20:19.46 | PakiPenguin | anyone played with gnugk+asterisk here? |
| 20:20.05 | znoG | stepcut: it does also say: Feb 15 17:19:42 NOTICE[1088506560]: chan_sip.c:6718 handle_response: Failed to authenticate on INVITE to ... |
| 20:20.13 | znoG | <PROTECTED> |
| 20:20.14 | Nix | lol@PakiPenguin |
| 20:20.46 | PakiPenguin | Nix: lol? |
| 20:21.14 | znoG | gotta love it when you're accidentally funny |
| 20:21.26 | Nix | h323.. not very well supported in *.. although it can register to gnugk.. |
| 20:23.37 | *** part/#asterisk DrNitro|Work (~docnitro@foxxp.wrlc.org) |
| 20:23.49 | JerJer | Nix: bullshit |
| 20:24.08 | JerJer | asterisk can make H.323 calls all day long |
| 20:24.23 | JerJer | recieving is a whole different story |
| 20:25.14 | JerJer | so stop spreading more bullshit |
| 20:25.21 | JerJer | state the facts or shut up |
| 20:25.30 | PakiPenguin | JerJer: h323 --> something --> * --> SIP ? How stable is that , and what should i change something with? |
| 20:27.06 | tzanger | PakiPenguin: VOODOO |
| 20:28.13 | riksta | any of you guys been using ADM? i'd appreciate some feedback...bugs etc |
| 20:28.56 | JerJer | change ? |
| 20:29.13 | JerJer | Asterisk can make outbound H.323 calls without issue to any endpoint i have found |
| 20:29.15 | Nix | JerJer: I did state the facts.. |
| 20:29.21 | Juggie | syslod, its in current to play any file i spicify on j oin? |
| 20:29.38 | JerJer | Many endpoints like to play games with RTP for inbound H.323 calls |
| 20:29.54 | JerJer | which leads to issues with ONLY SOME ENDPOINTS (yelling at nix) |
| 20:30.14 | zoa | tsk tsk, lets all behave |
| 20:30.14 | WildPikachu[BED] | hrmmm.... i heard a guy say "Asterisk is not stable enough in high density call routing, use cisco" ... *snicker* |
| 20:30.15 | JerJer | Nix: you did not state any fact |
| 20:30.20 | zoa | and prepare parties for von |
| 20:30.28 | *** join/#asterisk HitTop (~Jack@host6614613596.biz.tor.fcibroadband.com) |
| 20:30.31 | JerJer | you said H.323 is not very wll supported |
| 20:30.34 | Nix | yes. I did.. 2 infact |
| 20:30.38 | three55ml | JerJer: I bet your ego works wonders in the business world |
| 20:30.43 | Nix | 1) h323 is not well supported in * |
| 20:30.52 | Nix | 2) it can register to gnugk though |
| 20:30.53 | Nix | :-) |
| 20:31.05 | JerJer | Nix: tell that to the dozens of people i've setup H.323 based solutions for then |
| 20:31.22 | JerJer | three55ml: i don't accept bullshit from people |
| 20:31.31 | JerJer | if they don't like it, they can go somewhere else |
| 20:31.38 | three55ml | There's a difference between being an asshole and having some decency. |
| 20:31.49 | three55ml | Everytime I see you on here "helping" someone you're condescending and rude about it. |
| 20:31.51 | zoa | lalala, there we go again :) |
| 20:31.56 | JerJer | three55ml: nix has been here for the last few days bitching about asterisk |
| 20:32.12 | Nix | JerJer: I just may not have stated the fact that makes you look good.. |
| 20:32.25 | Nix | ie. 3) JerJer says outbound calls work fine with * and h323 |
| 20:32.26 | Nix | ;-) |
| 20:33.01 | Nix | JerJer: I was not bitching. |
| 20:33.09 | JerJer | then what do you call it? |
| 20:33.12 | Nix | I was actually helping people who asked questions |
| 20:33.16 | JerJer | telling the truth? i think not |
| 20:33.18 | *** join/#asterisk Matthew_I (~matthew@64-89-121-30.arpa.kmcmail.net) |
| 20:33.24 | Nix | please dont be so egotistical just because its your code.. |
| 20:33.33 | JerJer | i could care less |
| 20:33.37 | Nix | there are real life problems with the h323 support in asterisk and you know it.. |
| 20:33.37 | Juggie | when you generate a call via sockets or a .call file can you only use one variable? i cant get more the one to work |
| 20:33.38 | JerJer | don't spread lies |
| 20:33.52 | zoa | juggie, you can use the variable |
| 20:33.59 | JerJer | Nix: then don't use it |
| 20:34.00 | Juggie | i have done that |
| 20:34.00 | zoa | and store more variables inside it |
| 20:34.03 | zoa | with a separated |
| 20:34.06 | Matthew_I | Juggie: this is your luck day |
| 20:34.07 | zoa | separater |
| 20:34.11 | zoa | and then parse it |
| 20:34.12 | Nix | telling people that its all fine and dandy just gives them a bad impression of asterisk as a whole when it fails for them |
| 20:34.15 | Juggie | example? |
| 20:34.20 | Nix | it is much better to be honest up front |
| 20:34.23 | JerJer | Nix: did i say it was fine and dandy? |
| 20:34.24 | Matthew_I | Juggie: a call file you can do SetVar: one=var |
| 20:34.29 | zoa | var1|var2|var3 = 3 vars in one |
| 20:34.29 | Matthew_I | Juggie: multiple times |
| 20:34.33 | *** part/#asterisk mountainm2k (~freenodei@cbit-98.bullseye9.com) |
| 20:34.59 | Juggie | i'll pastebin it |
| 20:35.01 | Matthew_I | Juggie: but with manger you do it like zoa said 'Variable: Var1=stuff|Var2=stuff|Var3=stuff' |
| 20:35.13 | JerJer | Why do you think so many people have problems with H.323 in general? |
| 20:35.19 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 20:35.19 | JerJer | why was SIP even created if H.323 was the answer? |
| 20:35.26 | shmaltz | hi everyone |
| 20:35.27 | Nix | rofl@JerJer |
| 20:35.31 | JerJer | why are major telcos moving away from H.323? |
| 20:35.37 | Nix | rofl again |
| 20:35.38 | JerJer | and deploying SIP |
| 20:35.39 | shmaltz | does anybody have polycom phones? |
| 20:35.41 | Nix | if you say so |
| 20:35.48 | Nix | most telcos run h323 and you know it |
| 20:35.49 | JerJer | ask global crossings why they chose SIP |
| 20:35.58 | JerJer | no they don't |
| 20:36.03 | Matthew_I | sip sucks |
| 20:36.03 | Nix | "moving away" will be a 10 year process if ever... |
| 20:36.15 | Juggie | http://pastebin.ca/5896 |
| 20:36.19 | Matthew_I | non us telco's use h323 much more |
| 20:36.22 | Juggie | that works |
| 20:36.23 | shmaltz | Matthew_l what is better then sip? |
| 20:36.27 | Matthew_I | IAX |
| 20:36.28 | JerJer | Matthew_I: yes it does, but it is far more flexable than H.323 |
| 20:36.37 | Juggie | but only the first variable, any ideas? |
| 20:36.53 | zoa | i use h323 and sip |
| 20:36.55 | zoa | and h323 sux |
| 20:36.56 | Matthew_I | Juggie: dud, did you read it? |
| 20:37.14 | Juggie | Matthew_I what do you mean |
| 20:37.25 | JerJer | nix just because you are an H.323 carrier doesn't give you the right to bitch |
| 20:37.27 | shmaltz | IAX is only better when multiple calls to the same place |
| 20:37.45 | Matthew_I | Juggie: fpusts($socket, "Variable: numdialed=$roomnum|roomnum=$roomnum\r\n"); |
| 20:37.45 | JerJer | have some foresight and move away from the legacy gear |
| 20:37.52 | Matthew_I | Juggie: that will do what you want |
| 20:37.58 | Juggie | perfect |
| 20:38.00 | Juggie | thanks |
| 20:38.04 | Jlau515 | hi, is any body using a digium card in a proliant dl380 g4 |
| 20:38.08 | JerJer | shmaltz: try to get thru multiple layers of NAT using SIP |
| 20:38.18 | Nix | legacy gear.. lol@jer |
| 20:38.18 | shmaltz | doesn work |
| 20:38.27 | Matthew_I | Juggie: from a call file in the spool dir you can do multiple SetVar lines but for the manager interface you have to do it like that |
| 20:38.34 | JerJer | Nix: see you know the truth |
| 20:38.38 | JerJer | all you can do is laugh |
| 20:38.40 | three55ml | Jlau515: Search the Wiki, lots of known issues with that setup |
| 20:38.48 | *** join/#asterisk santiago (~santiago@200.123.226.162) |
| 20:38.59 | Matthew_I | JerJer: put an * box inside both nats and connect them with IAX, all is well :) |
| 20:39.01 | Nix | JerJer: I think you mistake alot of things for bitching.. |
| 20:39.12 | JerJer | nix: ok spreading FUD |
| 20:39.19 | JerJer | telling half truths |
| 20:39.22 | JerJer | what is it then? |
| 20:39.27 | JerJer | if it is not bitching |
| 20:39.32 | Qwell | I vote whining |
| 20:39.41 | JerJer | whining is a good word |
| 20:39.46 | JerJer | which he is doing most def |
| 20:39.49 | Qwell | whining is always a good word |
| 20:39.53 | *** part/#asterisk rvhi (~rv@mail.o-matrix.org) |
| 20:39.56 | Matthew_I | I agree with nix |
| 20:39.57 | Qwell | whining emcompasses alot, heh |
| 20:40.00 | Nix | I still don't understand why you are wasting so much channel space with hot air when PakiPenguin asked a question that clearly your code cannot handle (and you even admitted as much) and I provided him that answer.. |
| 20:40.18 | JerJer | i did no such thing |
| 20:40.35 | JerJer | i stated that some endpoiints play games with RTP |
| 20:40.44 | Nix | yes. thats true |
| 20:40.48 | Nix | both for SIP and h323 |
| 20:41.10 | JerJer | so who is blowing hot air here? |
| 20:41.32 | JerJer | you just counter-dicked yourself |
| 20:41.52 | Nix | how so? |
| 20:42.02 | JerJer | go away |
| 20:42.13 | *** join/#asterisk Ayano (~erik_leee@209.143.187.254) |
| 20:42.32 | Matthew_I | what the heck If i knew we could make up words like counter-dicked, I would have done so a long time ago |
| 20:42.48 | three55ml | JerJer, out of curiousity - do you make your living of NuFone or do you have a day job? |
| 20:42.52 | redder86 | Matthew_I: doing so is a disasterisk |
| 20:42.52 | Qwell | Matthew_I: Its practically a law on IRC |
| 20:43.03 | Weezey | subposably I make up words all the time. |
| 20:43.23 | JerJer | three55ml: i own and run multiple ISPs and VoIP based plays |
| 20:43.23 | Matthew_I | well that's fantabalistic |
| 20:43.41 | bjohnson | fantasterisk |
| 20:43.42 | tzanger | counter-dicked? haha |
| 20:43.51 | tzanger | bjohnson: I own .ca, .com/net/org on that :-) |
| 20:44.08 | JerJer | and support others that some might consider NuFone's competition so you have no clue |
| 20:44.12 | bjohnson | tzanger: only a small development fee need be submitted |
| 20:44.48 | shmaltz | JerJer |
| 20:44.50 | shmaltz | who would use VOIP thru multiple NAT, if you are a provider you get public IP's, if you are private, use good routing practices in NAT on at least one end (DMZ?). |
| 20:44.54 | tzanger | bjohnson: development fee? |
| 20:44.58 | bjohnson | tzanger: needs better stressing to get the right effect |
| 20:45.04 | bjohnson | FANTASterisk |
| 20:45.10 | tzanger | heh |
| 20:45.30 | vaewynAFK | bjohnson: or FANASSterisk |
| 20:45.32 | bjohnson | FANTASTerisk? |
| 20:45.43 | oej | bjohnson: Isn't FANTA a trademark belonging to the Coca Cola Corporation? FANTA(TM)sterisk ? |
| 20:45.58 | tzanger | heh |
| 20:46.02 | bjohnson | oej: likely .. but check FANTAST |
| 20:46.17 | Weezey | mmm cola. |
| 20:47.30 | WildPikachu[BED] | heh |
| 20:47.38 | WildPikachu[BED] | cola + asterisk = perfect |
| 20:47.45 | _Raptor_ | cu |
| 20:47.50 | Himeko | fanta is not a cola though |
| 20:48.00 | WildPikachu[BED] | fanta is nice :() |
| 20:48.01 | bjohnson | somehow colasterisk reminds me of colons |
| 20:48.07 | WildPikachu[BED] | fanta orange/grape |
| 20:48.10 | eKo1 | Fanta was a drink made for the German market by Coca Cola during WWII. |
| 20:48.20 | Weezey | how can I make it so that only people from a list of phone numbers get through to my extension? |
| 20:48.24 | PakiPenguin | eKo1: we have fanta in pakistan too |
| 20:48.38 | bjohnson | "made for the German market by Coca Cola during WWII"? Isn't Coca Cola US? |
| 20:48.39 | WildPikachu[BED] | i went on a tour of a coca cola bottling plant |
| 20:48.41 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 20:48.43 | *** join/#asterisk Uajal (~icechat5@ool-182e86f3.dyn.optonline.net) |
| 20:48.46 | eKo1 | I have Fanta down here in El Salvador so... |
| 20:48.51 | eKo1 | It's everywhere now. |
| 20:48.53 | Matthew_I | Weezey: route by CID |
| 20:48.54 | dsmouse | Weezey: http://www.voip-info.org/wiki-Asterisk+cmd+GotoIf |
| 20:48.54 | bjohnson | Weezey: do callerid checking |
| 20:49.11 | Matthew_I | everyone has fanta |
| 20:49.12 | PakiPenguin | eKo1: we have different flavours too |
| 20:49.18 | Matthew_I | europe and others had it before the us had it |
| 20:49.25 | bjohnson | Weezey: check user auth link from the tips and tricks page on the wiki for an example |
| 20:49.35 | Qwell | Is Fanta even good? |
| 20:49.39 | Matthew_I | Weezey: you don't have to mess with a bunch of gotoifs |
| 20:49.42 | Weezey | can I route by what port they're coming in from too? I want one port to go right to me, the other is filtered. |
| 20:49.42 | Qwell | Looks like it would be really sugary and gross |
| 20:49.43 | WildPikachu[BED] | Qwell, fanta is nice |
| 20:49.49 | eKo1 | I like Fanta. |
| 20:49.51 | PakiPenguin | Qwell: awesome |
| 20:49.53 | Matthew_I | Weezey: like what zap channel they are on? |
| 20:49.59 | eKo1 | Even though it was made for Nazis. |
| 20:50.00 | Qwell | 3 yes's, hmm |
| 20:50.01 | Uajal | I am not so good in cvs. For broadvoice I need to use patch http://edvina.net/broadvoice/broadvoicesip2.txt what exactly should I enter in linux prmpt? |
| 20:50.06 | Qwell | I didn't expect that. heh |
| 20:50.14 | Weezey | Mattew: I only have SIP FXOs. |
| 20:50.16 | Weezey | 2 |
| 20:50.31 | Matthew_I | Weezey: so what SIP port they are coming on? what are you saying? |
| 20:50.55 | Matthew_I | Uajal: check out the code from cvs |
| 20:51.07 | bjohnson | Weezey: use different incoming contexts for each SIP fxo |
| 20:51.25 | Matthew_I | Uajal: run: 'cat broadvoicesip2.txt | patch -p0 --dry-run |
| 20:51.26 | Weezey | Matthew: I have one FXO connected to a Norstar ATA, that one has to come directly to me, the other is coming from my private line, which will be filtered based on the caller. That's possible to do, but I'm just not sure how to make it work. |
| 20:51.32 | bjohnson | Uajal: I think stable 1.0.5 already works with BV |
| 20:51.35 | Matthew_I | Uajal: if it runs clean, take off the --dry-run |
| 20:51.55 | bjohnson | Weezey: use different incoming contexts for each SIP fxo |
| 20:51.59 | Matthew_I | Weezey: just make them go into different contexts, and filter the other based on CID |
| 20:52.11 | Weezey | cool |
| 20:52.15 | Weezey | thanks. |
| 20:52.19 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 20:52.19 | Matthew_I | Weezey: exten => 101/cid,1,Hangup |
| 20:52.25 | *** join/#asterisk TekLexus (~Mnemonic@167.206.75.24) |
| 20:52.43 | Uajal | How to check which version of Asterisk I have |
| 20:52.45 | Matthew_I | Weezey: exten => 101/1877linuxme,1,answer |
| 20:52.47 | *** join/#asterisk t3t (~t3t@galley.pangalacticgargleblaster.com) |
| 20:52.51 | PakiPenguin | show version |
| 20:53.05 | Matthew_I | PakiPenguin: I don't think that is realiable for cvs though |
| 20:53.26 | Uajal | I didn't run it yet. Is it possible to find version before running? |
| 20:53.29 | Weezey | The one thing I haven't grasped yet, how do I make different contexts work? Does it always start at [default] ? |
| 20:53.47 | Matthew_I | Weezey: zapata.conf |
| 20:53.58 | Matthew_I | Uajal: where did you get it? |
| 20:54.17 | Qwell | now, I know what an ata is...but what does it stand for? |
| 20:54.18 | Weezey | Mattew: I don't have any Zap stuff, so sip.conf ? |
| 20:54.37 | Qwell | nevermind, that was a dumb question |
| 20:54.38 | Matthew_I | Weezey: you just said you were using ATA's connected to FXO ports |
| 20:54.39 | Uajal | # cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds |
| 20:54.45 | Qwell | ata == analog telephone adapter? |
| 20:54.48 | Weezey | yeah, they're sip |
| 20:54.52 | Matthew_I | Qwell: correct |
| 20:54.55 | Weezey | SPA-3000 |
| 20:54.59 | Qwell | That was obvious, heh |
| 20:55.08 | *** join/#asterisk dsfr (~dsfr@216.207.244.183) |
| 20:55.14 | Uajal | export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot |
| 20:55.21 | Matthew_I | Weezey: but the ATA is plugged into the pc through the FXO right? |
| 20:55.29 | Weezey | yes |
| 20:55.31 | Matthew_I | Uajal: if you did the recently then you have 1.0.5 |
| 20:55.36 | *** join/#asterisk kietlak (~kietlak@11-mo3-6.acn.waw.pl) |
| 20:55.42 | Matthew_I | Weezey: then you configure it from zapata.conf |
| 20:55.49 | Uajal | yes recently. |
| 20:55.59 | Weezey | Mattew: good to know, you saved me a whole lot of time. |
| 20:56.10 | Weezey | my shit's coming tomorrow. |
| 20:56.16 | Uajal | Matthew: What is --dry-run? |
| 20:57.02 | *** join/#asterisk HitTop (~Jack@host6614613596.biz.tor.fcibroadband.com) |
| 20:57.15 | Matthew_I | Uajal: it makes it pretend to apply the patch |
| 20:57.23 | Matthew_I | Uajal: that way if it fails, you don't get half patched source |
| 20:57.51 | bjohnson | NNNOOOOOOO |
| 20:57.57 | bjohnson | SPA 3ks are SIP devices |
| 20:58.03 | bjohnson | guess where you configure them |
| 20:58.15 | bjohnson | sip.conf |
| 20:58.19 | bjohnson | NOT zapata.conf |
| 20:58.22 | Matthew_I | bjohnson: what are you talking about |
| 20:58.23 | Damin | Weezey: Your "shit"? Have you been constipated lately? |
| 20:58.26 | Uajal | any links explaining dry-run? |
| 20:58.30 | Matthew_I | bjohnson: he is not connecting his ATA's to asterisk |
| 20:58.34 | roamer323 | bjohson - where to get cheap SPA 3k in Canada, do you know? |
| 20:58.43 | Matthew_I | Weezey: right? they ATA are not connecting to asterisk via sip right? |
| 20:59.02 | Matthew_I | Uajal: man patch |
| 20:59.09 | bjohnson | the ATA he refers to is the ATA on his Nortel machine |
| 20:59.13 | Matthew_I | Uajal: all it does is pretend to apply the patch |
| 20:59.16 | bjohnson | (which plugs into a fxo port) |
| 20:59.23 | Matthew_I | bjohnson: what the hizzle? |
| 20:59.25 | bjohnson | in this case the fxo on a SPA 3000 |
| 20:59.47 | Matthew_I | Weezey: are you connecting devices to asterisk via SIP or through an fxo card in the pc? |
| 21:00.16 | bjohnson | * needs the SPA 3000 fxo defined in sip.conf so incoing calls from the NOrtel through it's ATA port come into the SPA 3000 fxo into a default (or other name) context |
| 21:00.28 | *** join/#asterisk pluto- (~pluto@d141-218-238.home.cgocable.net) |
| 21:00.39 | Matthew_I | bjohnson: is the ATA connecting to asterisk via SIP or via fxo? |
| 21:00.42 | Matthew_I | Weezey: ? |
| 21:01.33 | bjohnson | we lost him |
| 21:01.44 | bjohnson | I think he's had a heart attack |
| 21:01.57 | Matthew_I | cram it, facking heart attacks |
| 21:01.59 | bjohnson | call the ambulance |
| 21:02.02 | Matthew_I | the should be abominishilsed |
| 21:02.05 | bjohnson | tell them "no rush" |
| 21:02.12 | mrempire | How can I configure an outgoing call on my isdn card to pstn. In comming calls from pstn are already working |
| 21:02.44 | Matthew_I | mrempire: extensions.conf |
| 21:02.49 | pluto- | is there an 'idiots guide' to setting up asterisk for a simple ipphone/ata to voip provider gateway? my provider supports IAX but my ATA supports SIP only... |
| 21:02.51 | bjohnson | set up one or more extensions that use Dial() to dial out the isdn |
| 21:02.53 | Matthew_I | mrempire: how are you dialing out? |
| 21:03.28 | Matthew_I | pluto-: so you want the ata to connect to ast and ast to connect to provider? |
| 21:03.30 | bjohnson | pluto-: don't know of a simple one .. but they aren't too hard once you get the hang of it |
| 21:03.33 | mrempire | in my extensions.conf i have :exten => 551,1,Dial(Modem/ttyI0/5407616:0650506652) |
| 21:03.40 | *** join/#asterisk sezuan (sezuan@port-212-202-57-119.dynamic.qsc.de) |
| 21:03.45 | pluto- | Matthew_I: yeah, is that doable? |
| 21:03.48 | *** join/#asterisk kiran (~kiran@202.62.88.140) |
| 21:03.50 | Matthew_I | pluto-: sure |
| 21:03.55 | mrempire | Matthew_I: isdn4linux |
| 21:03.58 | Matthew_I | pluto-: if you can provision your ATA |
| 21:04.00 | bjohnson | pluto-: so many examples that it's hard to find one that is good to follow |
| 21:04.02 | bjohnson | ~docs |
| 21:04.03 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 21:04.08 | *** join/#asterisk Nix (~Nix@81.213.125.220) |
| 21:04.09 | Matthew_I | mrempire: like hdlc type styff? |
| 21:04.24 | bjohnson | pluto-: it is DEFINITELY doable |
| 21:04.40 | Matthew_I | mrempire: can't help you with that :) |
| 21:04.50 | pluto- | ok. wanted to make sure it was possible before wasting a lot of time reading up on it |
| 21:04.52 | mrempire | Matthew_I:no just normal way |
| 21:04.57 | bjohnson | pluto-: start with * and the example configs .. what type of ATA you have? |
| 21:05.17 | kiran | hi any one can help me in configuring digium 410 card |
| 21:05.20 | oej | ~seen anthm |
| 21:05.21 | jbot | anthm is currently on #asterisk (45m 49s) |
| 21:05.21 | pluto- | bjohnson: i have a digium IAXy and also a Grandstream HT486 |
| 21:05.27 | Uajal | how to determine whether it is answering machine or real person? |
| 21:05.34 | bjohnson | grandstream is SIP right? |
| 21:05.35 | Matthew_I | what type of isdn card is it? |
| 21:05.37 | mrempire | Matthew_I: logs say:Requested device 'ttyI0/5407616' does not exist |
| 21:05.42 | znoG | pluto-: you just need to configure your asterisk box so that your ATA can auth to it, then setup your dial plans to dial out. If you have a Sipura ATA, you'll most likely need to adjust the dial plan on that too. |
| 21:05.43 | pluto- | but i dont believe you can use the IAXy with a remote iax provider can you? |
| 21:05.52 | Matthew_I | ~seen masteryoda |
| 21:05.53 | jbot | masteryoda <~mnicholso@dhcp-155.digium.com> was last seen on IRC in channel #asterisk, 4d 23h 13m 23s ago, saying: 'some one should be with you shortly'. |
| 21:06.04 | Uajal | how to determine whether it is answering machine or real person and play for them different messages? |
| 21:06.04 | kiran | hi any one from india or of indian origin? |
| 21:06.07 | znoG | pluto-: so, ATA <---SIP---> Asterisk <---IAX---> Your provider |
| 21:06.09 | bjohnson | pluto-: I thinnk you can but you might want to start with something easier |
| 21:06.17 | Matthew_I | Uajal: well, there is no sure way |
| 21:06.24 | pluto- | znoG: yeah, essentially i guess |
| 21:06.24 | bjohnson | like znoG said .. most people go through * |
| 21:06.27 | eKo1 | Uajal: Don't answering machines make beeps before or after answering. You could detect that. |
| 21:06.28 | Matthew_I | Uajal: you can try to use BackgroundDetect |
| 21:06.41 | Matthew_I | Uajal: or WaitForSilence or something |
| 21:06.49 | Matthew_I | Uajal: but it's difficult to be 100% sure |
| 21:06.52 | *** join/#asterisk E|nyPRI_ (~les@205-200-64-180.static.mts.net) |
| 21:06.59 | Matthew_I | eKo1: all the beeps are different |
| 21:07.04 | E|nyPRI_ | anyone have a sipura-841 working with g729 ? |
| 21:07.04 | bjohnson | pluto-: you have the sample configs installed that come with *? |
| 21:07.05 | pluto- | the provider told me they supported SIP and I tried all day yesterday to get the GrandStream HT486 to connect directly through them but it never worked. they weren't much help though. |
| 21:07.13 | eKo1 | Doesn't matter as long as they are beeps. |
| 21:07.13 | Matthew_I | eKo1: the best way is to listen for noise, and then silence |
| 21:07.34 | pluto- | bjohnson: i installed Asterisk@Home so yeah i guess there are some samples |
| 21:07.42 | Matthew_I | eKo1: "You have reached my crappy noisy voice mail, leave a message after the beep, during the slience <beep>" |
| 21:07.44 | kiran | hey any one can help me |
| 21:07.45 | Uajal | Is WaitForSilence the asterisk command (can it be called from perl?) |
| 21:07.58 | Matthew_I | eKo1: so how do you detect the beeps? |
| 21:08.13 | Matthew_I | Uajal: I am not sure, there are several wait commands |
| 21:08.22 | pluto- | don't suppose anyone else here has an account with iax.cc/sixtel ? |
| 21:08.23 | bjohnson | pluto-: SIP devices are defined in /etc/asterisk/sip.conf and iax devices/connections are defined in /etc/asterisk/iax.conf .. there should be some samples already in those files |
| 21:08.24 | Matthew_I | Uajal: it would be an asterisk application |
| 21:08.28 | eKo1 | Matthew_I: No clue, using a wave analyser or something. |
| 21:08.35 | Mavvie | hmm... dumb fxo doesn't hangup all the time, leaving channels open. |
| 21:08.41 | Matthew_I | eKo1: and there you have it..... |
| 21:08.46 | Matthew_I | Mavvie: uk or us? |
| 21:08.46 | bjohnson | pluto-: plus your voip provider likely has an example config for * |
| 21:08.47 | eKo1 | Mavvie: Hah, I have the same problem. |
| 21:08.48 | Uajal | where to find information about these wait commands. I didn't see in documentation? |
| 21:08.52 | *** part/#asterisk sezuan (sezuan@port-212-202-57-119.dynamic.qsc.de) |
| 21:08.53 | Matthew_I | eKo1: uk or us |
| 21:08.59 | pluto- | bjohnson: yeah they do. |
| 21:09.04 | eKo1 | niether. |
| 21:09.10 | eKo1 | *neither. |
| 21:09.10 | kiran | hey any one from india |
| 21:09.12 | pluto- | will give it a shot thanks |
| 21:09.15 | Matthew_I | Uajal: on the ast console type show applications, and show application appname |
| 21:09.18 | kiran | ? |
| 21:09.22 | Mavvie | Matthew_I / eKo1 : US to international lines. |
| 21:09.24 | bjohnson | pluto-: then it's just a metter of editing /etc/asterisk/extensions.conf to connect everything |
| 21:09.33 | Matthew_I | eKo1: where are you? |
| 21:09.35 | mrempire | Kiran: indian in Holland |
| 21:09.42 | eKo1 | Central America. |
| 21:09.46 | bjohnson | don't forget to restart * (or reload the config files) after making changes |
| 21:09.48 | Matthew_I | Mavvie: hmmm.... busydetect not working? |
| 21:09.58 | Matthew_I | eKo1: no busydetect for you? |
| 21:10.20 | pluto- | bjohnson: i tried adding an extension for my IAXy in APM (web gui that came with asterisk@home) but in my log file i keep seeing the IAXy trying to register and failing |
| 21:10.29 | *** join/#asterisk _RaYmAn_ (user@213.237.12.147.adsl.vby.tiscali.dk) |
| 21:10.53 | bjohnson | forget the IAXy for now |
| 21:11.03 | bjohnson | start with the easier one first .. the SIP one |
| 21:11.06 | *** join/#asterisk darby_t (~tom@dni134.neoplus.adsl.tpnet.pl) |
| 21:11.10 | Matthew_I | pluto-: eww apm.... |
| 21:11.34 | pluto- | heh, i thought you guys might say that. purists always hate gui's |
| 21:11.38 | Mavvie | Matthew_I: I have busydetect in the zapata.conf, but I don't know why it doesn't detect right now. |
| 21:11.45 | Mavvie | pluto-: only bad guis :-) |
| 21:12.06 | pluto- | heh. yeah i think it messed up my sample config files pretty good |
| 21:12.16 | pluto- | i'd like to go back to stock but im not even sure asterisk@home provides them |
| 21:12.22 | Uajal | I installed asterisk right now. Should I make samples or it is bad idea? |
| 21:12.29 | Matthew_I | Mavvie: have you ever listened to the tone after a hangup? what happens when the remote end hangs up? |
| 21:12.47 | *** join/#asterisk schwagner (~andrew@68.143.92.248.nw.nuvox.net) |
| 21:12.56 | Matthew_I | Uajal: is this the first install? |
| 21:13.02 | Uajal | yes |
| 21:13.11 | Matthew_I | Uajal: then do make samples |
| 21:13.30 | Matthew_I | Uajal: but only on first installs, it will over write your files otherwise |
| 21:13.32 | *** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl) |
| 21:14.51 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 21:14.57 | schwagner | does anyone know if anyone has considered/tried to integrate asterisk with any sort of CRM solution? |
| 21:15.20 | Matthew_I | schwagner: what do you mean CRM? |
| 21:15.31 | Qwell | call record management |
| 21:15.38 | Matthew_I | like a cdr? |
| 21:15.41 | Matthew_I | or something different? |
| 21:15.53 | schwagner | well, i meant customer relationship manager |
| 21:15.53 | vaewyn | CRM process CDR for billing |
| 21:15.56 | _RaYmAn_ | anyone know of a way to vaguely reliably match SIP/RTP packets with iptables for QoS purposes? As far as I can tell every call's port seems to be setup by an SDP packet so it shouldn't be that hard to do (for someone who can program for iptables.) |
| 21:16.00 | Qwell | right, thats the one |
| 21:16.17 | schwagner | no, like with screen pops with the callers records attached |
| 21:16.45 | schwagner | like, for example, salesforce.com and the like |
| 21:16.47 | Matthew_I | schwagner: could probably be done through the manager interface, if you did not want to write a specialized asterisk module |
| 21:16.49 | Qwell | ie; customer owes us $600 |
| 21:16.57 | Qwell | or; customer is an idiot, hangup on him |
| 21:17.08 | schwagner | yea |
| 21:17.28 | Uajal | any links to run these samples? |
| 21:17.32 | Qwell | I bet Adelphia has one of those for me |
| 21:17.55 | Qwell | "customer is an asshole, transfer to tier 2 immediately" |
| 21:18.01 | kiran | any one can help in configuring digium cards |
| 21:18.05 | kiran | and work wih asteriskr |
| 21:18.11 | Qwell | kiran: Whats the problem? |
| 21:18.12 | schwagner | Quell: yea, pretty much |
| 21:18.17 | *** join/#asterisk farmatel (~farmatel@atlgw01.pharmacentra.com) |
| 21:18.30 | schwagner | so, anybody tried this? |
| 21:18.34 | Qwell | no, thanks |
| 21:18.37 | Matthew_I | kiran: sure we can help, and digium can too :) |
| 21:18.45 | vaewyn | Qwell: Mine at SBC probably reads "know's too much... either don't answer or send a hit man" |
| 21:18.57 | schwagner | ha |
| 21:19.02 | Qwell | vaewyn: I constantly yell at tier 1 techs at Adelphia |
| 21:19.11 | harryvv | can anyone vouch for the S100I — "IAXy" |
| 21:19.11 | Matthew_I | schwagner: I would use the asterisk manager interface |
| 21:19.13 | Qwell | My employer/bank loves me...heh |
| 21:19.16 | *** join/#asterisk gdh (~gdh@80-192-144-33.cable.ubr06.wi.blueyonder.co.uk) |
| 21:19.25 | vaewyn | Qwell: is the fastest way to get to "real" help :P |
| 21:19.25 | Matthew_I | schwagner: do you have a soultion you are looking to tie into asterisk? |
| 21:19.26 | Qwell | I've gotten a 5th level manager once |
| 21:19.34 | *** join/#asterisk folsson (~filip@h87n2fls31o985.telia.com) |
| 21:19.57 | schwagner | not really, i can't find a good crm/ticket tracking package |
| 21:19.59 | Qwell | vaewyn: At Adelphia, everybody claims there is no such thing as tier 3... |
| 21:20.01 | Qwell | I've been there |
| 21:20.21 | kiran | hi qwell |
| 21:20.25 | gdh | Arg. I'm being fantastically lame today. would anyone mind helping me sort my single FXO out? :) |
| 21:20.28 | Qwell | kiran: No dcc, ask a question |
| 21:20.37 | kiran | yeah |
| 21:20.50 | kiran | fine configured 410 digium card to |
| 21:20.55 | kiran | work with e1s |
| 21:21.16 | kiran | i deployed this card in indian scenario |
| 21:21.17 | bjohnson | schwagner: there have been some done but I don't know how many are easy to use (or available). They tend to be pretty task oriented and custom builds |
| 21:21.20 | gdh | "WARNING[8972]: chan_zap.c:769 zt_open: Unable to specify channel 1: No such device or address" .. yet 'ztcfg -vv' says Channel 01: FXO Kewlstart (Default) (Slaves: 01) - and ztmonitor at least shows a display.. |
| 21:21.23 | Matthew_I | gdh: what are you doing over there? |
| 21:21.33 | bjohnson | schwagner: check the usual suspects for info |
| 21:21.33 | schwagner | Mattew_I: i am looking at opencrx now |
| 21:21.44 | Matthew_I | kiran: what problems are you having? |
| 21:21.47 | bjohnson | usual suspects = mailing list archives, wiki, google |
| 21:21.51 | kiran | as e1s in india are bit different |
| 21:22.07 | kiran | there is a green light glowing at the back of the card once i configure |
| 21:22.10 | gdh | Matthew_I: Hm? |
| 21:22.20 | schwagner | bjohnson: thanks, i'll look there |
| 21:22.20 | Matthew_I | gdh: fedora core 3 huh? |
| 21:22.31 | gdh | God no :) |
| 21:22.39 | gdh | Debian sarge, thanks. |
| 21:22.45 | Matthew_I | gdh: oohh good |
| 21:22.51 | Matthew_I | gdh: so ztcfg -vv runs fine |
| 21:22.54 | gdh | I've had this all working on woody , then Stuff happened |
| 21:23.04 | gdh | Matthew_I: Yis. '1 channels configured' (sic) |
| 21:23.07 | Matthew_I | gdh: is this a TDM card? |
| 21:23.21 | gdh | Naw, single FXO 101P |
| 21:23.25 | kiran | can u tell me how to make a sample call and recive a call with small call flow |
| 21:23.38 | Matthew_I | gdh: so in zapata.conf you have the card configured correctly? |
| 21:24.11 | kiran | as when i call the number |
| 21:24.13 | gdh | Matthew_I: I have 5 lines of zapata conf - [channels] signalling=fxo_ks language=en context=incoming channel => 1 |
| 21:24.25 | kiran | it maturing the call but not voice files are playing in |
| 21:24.27 | *** join/#asterisk Rick_Hunter (~rhunter@01-109.008.popsite.net) |
| 21:24.48 | kiran | how to know wheather the pri is working fine or not |
| 21:24.57 | kiran | and how to do dial outs |
| 21:25.01 | *** part/#asterisk farmatel (~farmatel@atlgw01.pharmacentra.com) |
| 21:25.03 | gdh | This is all using the debian package for asterisk and for teh zaptel-source compiled with make-kpkg modules, etc. |
| 21:25.06 | kiran | and where do i see the cdr of this |
| 21:25.11 | Matthew_I | gdh: hmmm |
| 21:25.19 | gdh | That's what I thought! |
| 21:25.27 | Matthew_I | gdh: and you get that error when you try to dial out |
| 21:25.57 | Matthew_I | gdh: I have not had a chance to work with the debian packages (debian is usually good though) |
| 21:26.03 | gdh | Matthew_I: When I try to start * - asterisk -vvvvvvvvc |
| 21:26.07 | gdh | It won't even start |
| 21:26.12 | Matthew_I | gdh: oh |
| 21:26.16 | schwagner | it's official: there are too damn many crm/groupware/cms projects out there |
| 21:26.17 | Matthew_I | gdh: what does it say? |
| 21:26.27 | kiran | qwell: how to procees |
| 21:26.33 | kiran | the calls |
| 21:26.35 | Matthew_I | gdh: what does it say when you do asterisk -c |
| 21:26.39 | gdh | <PROTECTED> |
| 21:26.39 | gdh | <PROTECTED> |
| 21:26.40 | gdh | Feb 15 21:17:56 WARNING[8972]: chan_zap.c:769 zt_open: Unable to specify channel 1: No such device or address |
| 21:26.40 | Qwell | got me |
| 21:26.41 | *** join/#asterisk o-m-a-o-m-a (unknown@43.5-dial.augustakom.net) |
| 21:26.51 | *** join/#asterisk eKo1 (~bernd@207.42.191.66) |
| 21:26.56 | kiran | qwel: do u understand my problem |
| 21:27.01 | znoG | the problem is that anybody who reads a small book on mysql and a bit on php will create some sort of project, probably a crm/groupware/cms project think they are inventing the wheel. |
| 21:27.20 | Matthew_I | gdh: just as I thought |
| 21:27.22 | kiran | qwell: and how to check wheather the pri is working fine or not |
| 21:27.24 | o-m-a-o-m-a | good morning everyone |
| 21:27.25 | Matthew_I | gdh: and the module is loaded? |
| 21:27.26 | Qwell | got me |
| 21:27.38 | Matthew_I | o-m-a-o-m-a: good afternoon |
| 21:27.50 | gdh | Matthew_I: Sure is - ztcfg wouldn't've told me it had one kewlstart interface otherwise |
| 21:27.50 | eKo1 | znoG: Good thing I don't read software books. |
| 21:27.50 | schwagner | znoG: exactly |
| 21:27.54 | kiran | qwell: didnt get u |
| 21:27.58 | gdh | wcfxo 12320 0 |
| 21:27.58 | gdh | zaptel 182788 1 wcfxo |
| 21:28.03 | gdh | dmesg shows it loaded OK |
| 21:28.10 | Matthew_I | kiran: pri show span <NUM> |
| 21:28.19 | kiran | my mmodule is wct4xxxp |
| 21:28.30 | gdh | Matthew_I: and I don't get any unresolved symbols etc. from depmod -ar |
| 21:28.35 | Matthew_I | gdh: and the card is configured as fxsks in zaptel.conf |
| 21:28.49 | Uajal | Does * work with Dialogic T1 boards or only with Digium T1? |
| 21:28.56 | Matthew_I | gdh: and ztcfg works fine? |
| 21:29.08 | Matthew_I | gdh: and your channel is 1 in zapata.conf? |
| 21:29.15 | *** join/#asterisk gdh (~gdh@80-192-144-33.cable.ubr06.wi.blueyonder.co.uk) |
| 21:29.18 | gdh | gahhhhh |
| 21:29.27 | Matthew_I | gdh: I say try stable asterisk from cvs if all of that is in order |
| 21:29.28 | schwagner | gdh: what does your zapata.conf say? |
| 21:29.28 | kiran | join #digium |
| 21:29.55 | gdh | schwagner: See above - it's just 5 lines |
| 21:30.09 | kiran | Qwell can u help me in this |
| 21:30.17 | o-m-a-o-m-a | I need a piece of someones extensions.conf for an incoming CAPI call without VBox |
| 21:30.18 | schwagner | gdh: i think you need fxs_ks signalling |
| 21:30.30 | JerJer | gdh: i say use the latest cvs code |
| 21:30.43 | Matthew_I | gdh: well you do have it configed wrong |
| 21:30.44 | kiran | matthew_i: can u help me |
| 21:30.45 | kiran | ? |
| 21:30.48 | gdh | JerJer: That's exactly what I'd hoped to stay away from this time round =) |
| 21:30.48 | schwagner | gdh: you have an fxo card, right? |
| 21:30.50 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-153-24.dsl.scarlet.be) |
| 21:30.57 | Matthew_I | kiran: I don't know what is wrong with you :) |
| 21:31.04 | gdh | Oh jesus, I'm a wanker. |
| 21:31.15 | kiran | matthew_i:? |
| 21:31.33 | kiran | matthew_i: why? |
| 21:31.50 | gdh | or maybe not :) - still fails to open with fxs_ks in zapata.conf |
| 21:31.56 | schwagner | dang |
| 21:32.30 | gdh | brb, my bladder is about to burst :) |
| 21:32.55 | o-m-a-o-m-a | Using exten => 576264,1,SetLanguage(de) / exten => 576264,2,Dial(Zap/g1/2264,120,tT) I'll get 2 internal Calltones, external 4 and then a busy. It should just ran up to 60 sek |
| 21:33.24 | schwagner | gdh: when you get back, try adding a group line |
| 21:35.34 | kiran | any one from digium? |
| 21:35.43 | kiran | ? |
| 21:36.04 | kiran | any one from digium? |
| 21:36.07 | kiran | any one from digium? |
| 21:36.09 | kiran | any one from digium? |
| 21:36.13 | loud | if they are, you scared them away already. |
| 21:36.16 | schwagner | kiran: apparently not |
| 21:36.18 | bjohnson | ask one more time for a boot |
| 21:36.35 | Silik0n | ask -1 more time for boot... ie: boot him already |
| 21:36.53 | loud | kiran, would you like to leave a message ? |
| 21:36.57 | *** join/#asterisk HitTop (~Jack@host6614613596.biz.tor.fcibroadband.com) |
| 21:38.02 | znoG | kiran: that sort of annoying attitude will get you 0 help |
| 21:38.21 | *** join/#asterisk KalD|Work (~KalD@proxy.corp.telesym.com) |
| 21:38.35 | kiran | sory guys if i annoy any one |
| 21:39.10 | eKo1 | kiran: This is not #digium. |
| 21:39.21 | *** part/#asterisk pluto- (~pluto@d141-218-238.home.cgocable.net) |
| 21:39.34 | gdh | schwagner: OK back. I don't get what you mean by a 'group line' . |
| 21:39.51 | schwagner | gdh: 'group = 1' in your zapata.conf |
| 21:40.16 | schwagner | gdh: although i kinda doubt that will help initilization problems |
| 21:40.20 | gdh | exactly the same . |
| 21:40.30 | *** join/#asterisk hmmhesays (negative3k@66.173.103.108) |
| 21:40.38 | hmmhesays | have no fear, i'm back |
| 21:41.11 | Mavvie | <PROTECTED> |
| 21:41.11 | Mavvie | <PROTECTED> |
| 21:42.03 | bjohnson | Himeko: you still here? |
| 21:42.26 | Matthew_I | i got distracted |
| 21:42.39 | bjohnson | Beirdo: greg_work: no updates to report |
| 21:42.47 | greg_work | hm? |
| 21:43.26 | gdh | schwagner: Am checking out the current CVS just in case =) |
| 21:43.43 | gdh | was really hoping that stable == usable by now :) |
| 21:43.46 | Matthew_I | gdh: this could be because of a version mismatch between zaptel and asterisk |
| 21:43.47 | *** join/#asterisk Chotaire (chotaire@chotaire.net) |
| 21:43.57 | schwagner | gdh: you don't have noload => chan_zap.so in you modules.conf, do you? |
| 21:44.00 | Matthew_I | gdh: what version of asterisk got installed |
| 21:44.04 | eKo1 | stable is usable |
| 21:44.09 | eKo1 | sort of... |
| 21:44.12 | gdh | Matthew_I: You'd hope that the deb packages in sarge would be correctly aligned |
| 21:44.17 | Matthew_I | schwagner: no he dosen't, because that is where the error is comming from |
| 21:44.24 | bjohnson | greg_work: CDN 800 voip search |
| 21:44.28 | gdh | asterisk 1.0.5-1 |
| 21:44.29 | Matthew_I | gdh: not really, packages just kinda of filter into sarge |
| 21:44.33 | gdh | zaptel-source 1.0.2-2 |
| 21:44.37 | gdh | feh |
| 21:44.47 | greg_work | bjohnson: ah. did i tell you what primus told me? |
| 21:44.51 | bjohnson | no |
| 21:44.56 | bjohnson | piss off? |
| 21:45.16 | Matthew_I | gdh: not sure if that will cause a problem or not |
| 21:45.16 | bjohnson | hehe .. they no like * |
| 21:45.33 | gdh | Matthew_I: I wouldn't have thought so, but I'll use the source anyway, For A Laugh. |
| 21:45.34 | Matthew_I | gdh: but having the wrong version can cause a problem |
| 21:45.47 | greg_work | they said since i have local lines with them, they'd do an 800 # to an arbitrary phone number for 4.5c/min, (4c/min if you're a costco executive member, which i think we are) |
| 21:45.54 | Matthew_I | gdh: yeah that's what I would do, just check out the latest stable code |
| 21:46.10 | *** join/#asterisk mountainm2k (~freenodei@cbit-98.bullseye9.com) |
| 21:46.16 | greg_work | so probably the cheapest way to get it is to get a cheap DID anywhere, then get a number pointed at it |
| 21:46.27 | *** join/#asterisk brian23 (~brian@69.20.5.30) |
| 21:46.50 | brian23 | hi, is there a way to detect in the dialplan when a call is coming from a particular source? |
| 21:47.08 | brian23 | for instance, if I want only Zap inbound calls to do certain things, is there a way to discriminate? |
| 21:47.18 | *** join/#asterisk florz (nobody@odnb-d9baa4be.pool.mediaWays.net) |
| 21:47.19 | eKo1 | brian23: Yes. |
| 21:47.28 | brian23 | what would be the format? |
| 21:47.49 | eKo1 | RTFW |
| 21:48.02 | shmaltz | ManxPower; you around? |
| 21:48.06 | Matthew_I | brian23: send stuff into different contexes |
| 21:48.19 | moonwick | greg_work: why not get a nufone 800#? |
| 21:48.26 | shmaltz | ~seen ManxPower |
| 21:48.27 | jbot | manxpower <~eric@dsl-209-205-172-111.i-55.com> was last seen on IRC in channel #asterisk, 18h 18m 31s ago, saying: 'Nugget, We should get tzanger's opinion!'. |
| 21:48.53 | brian23 | thanks - I did. No help. Can you point me in the right direction? If I have a call coming from a switch thru PRI to the Asterisk zap channel, how do I trigger an action without forcing a number as the extension? the dialplan is based on extensions |
| 21:49.12 | harryvv | what directories should asterisk have what persmissions on? im getting some permission problems for asterisk service. |
| 21:49.21 | *** join/#asterisk JMan5Work (~guest@mail.pc-assoc.com) |
| 21:49.31 | bjohnson | greg_work: but then you pay Primus 4c/min plus the DID voip provider likely 2c/min |
| 21:49.52 | *** join/#asterisk wrzf (~chris@cvg-165-103-153.cinci.rr.com) |
| 21:50.06 | brian23 | or, is there no way to trigger an action without actually forcing a number? |
| 21:50.09 | bjohnson | moonwick: he's trying to break the CDN toll free 6c/min barrier |
| 21:50.18 | wrzf | Hello |
| 21:50.29 | bjohnson | moonwick: (for tying into a voip system) |
| 21:50.30 | o-m-a-o-m-a | no one here without a VBox? |
| 21:50.44 | moonwick | ah |
| 21:50.57 | wrzf | I need a little help with something |
| 21:50.57 | greg_work | moonwick: because nufone doesn't even bother to reply to my email |
| 21:51.04 | bjohnson | hehe |
| 21:51.13 | JMan5Work | Hello all. |
| 21:51.14 | moonwick | greg_work: nufone tech support? ha. :) |
| 21:51.18 | wrzf | how do I get a person when they press 2 for asterisk to dial my extension |
| 21:51.24 | bjohnson | greg_work: they replied to mine and I have the info .. they are between 8c and 10c per minute |
| 21:51.26 | wrzf | I have this but it does not seem to work |
| 21:51.28 | wrzf | exten => 2,1.Goto(300) |
| 21:51.32 | greg_work | bjohnson: yes. but 2c/min US + 4c/min Canadian is cheaper than 6.5c/min US for an 800 |
| 21:51.54 | greg_work | cheaper than 8 to 10c/min US too ;) |
| 21:52.10 | greg_work | moonwick: no, nufone sales. |
| 21:52.26 | moonwick | odd |
| 21:52.32 | bjohnson | greg_work: I'm certain that a voip provider somewhere can at least get close to the telco rates for 6c/min or less |
| 21:52.35 | moonwick | ell, I've never dealt with their sales |
| 21:52.37 | JMan5Work | Any recommendations as to which cards to use for standard analog lines both inbound and outbound - 3 lines voice + 1 fax |
| 21:52.57 | greg_work | bjohnson: well. let know if you find one ;) |
| 21:53.06 | JMan5Work | I was looking at the Zaptel TDM04B |
| 21:53.09 | greg_work | bjohnson: you'd think the canadian providers would do it |
| 21:53.18 | bjohnson | JMan5Work: 4 fxo port digium card is one option |
| 21:53.19 | greg_work | JMan5Work: i have that setup here |
| 21:53.46 | bjohnson | JMan5Work: you're probably best off leaving the fax out of the mix |
| 21:54.01 | JMan5Work | bjohnson: why? |
| 21:54.54 | bjohnson | JMan5Work: read the wiki about faxes |
| 21:54.56 | bjohnson | ~docs |
| 21:55.05 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 21:55.20 | JMan5Work | yeah theres a bunch there to wade thru |
| 21:55.29 | bjohnson | also, other practical considerations about availability |
| 21:55.52 | JMan5Work | what about availability |
| 21:56.10 | JMan5Work | one of the lines is already used as a fax |
| 21:56.18 | JMan5Work | fax and fax only |
| 21:56.21 | bjohnson | greg_work: found a couple potential ones but can't get them to confirm via email facts that are unclear on their site |
| 21:56.40 | bjohnson | JMan5Work: and you don't want to keep it as that? |
| 21:56.50 | o-m-a-o-m-a | the first Link @ digium is wrong |
| 21:57.13 | schwagner | greg_work: you look at vonage yet? |
| 21:57.25 | JMan5Work | bjohnson: yeah. I want 3 lines voice + 1 line fax. shouldn't that work with the 4 port FXO card? |
| 21:57.41 | JMan5Work | Or should I look at a combo of Asterisk and hylafax? |
| 21:57.49 | bjohnson | why run the fax line through a voip server if it is just connecting to a fax machine? |
| 21:58.08 | JMan5Work | gonna get rid of the fax machine alltogether |
| 21:58.15 | terracon | my hylafax is connected to multitech modem |
| 21:58.31 | greg_work | bjohnson: yeah, common thing among these shady voip providers. no replying to email, and crappy websites almost devoid of useful info |
| 21:58.34 | *** part/#asterisk djin (~djin@gridfox.xs4all.nl) |
| 21:58.41 | greg_work | and by shady, i mean all |
| 21:58.55 | JMan5Work | Vonage = shady |
| 21:58.56 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com) |
| 21:59.05 | bjohnson | JMan5Work: for simplicity, keep the fax stuff completely separate |
| 21:59.16 | JMan5Work | ok thanks. |
| 21:59.16 | schwagner | i've had really good luck with vonage |
| 21:59.17 | harryvv | fricken asterisk is not loading all the way after a reboot. Some kind of persmisson problem. I can manually start asterisk as su root. |
| 21:59.30 | bjohnson | JMan5Work: for complexity .. try to run incoming and outgoing faxes through * .. and find that it ain't easy |
| 21:59.41 | greg_work | schwagner: no, do they even do DID's? I don't want their almost-the-price-of-a-POTS-line things that have voicemail and all the other stuff I can do myself with * |
| 21:59.41 | dsmouse | schwagner: I'm having prolbems dialing out with vonage at the moment |
| 21:59.44 | JMan5Work | next question - anyone ever migrate someone off Altigen to asterisk? |
| 21:59.47 | gdh | Bollocks to it - will try again when I don't feel like throwing things :) |
| 22:00.11 | bjohnson | schwagner: SHUTTT UPPP !!! vonage ..plu-eeze |
| 22:00.18 | greg_work | i expect to pay $2-5/month for a DID, and 1-2c/min across north america. |
| 22:00.21 | *** part/#asterisk gdh (~gdh@80-192-144-33.cable.ubr06.wi.blueyonder.co.uk) |
| 22:00.31 | schwagner | greg_work: they say the'll give you an 800 number for 4.99 a month and 4.5c / min |
| 22:00.34 | *** join/#asterisk ZX81 (matt@222-153-20-92.jetstream.xtra.co.nz) |
| 22:00.43 | dsmouse | ... |
| 22:00.49 | terracon | I'm on vonage , should I be worried. heh |
| 22:00.52 | greg_work | schwagner: canadian ? |
| 22:00.52 | schwagner | well, then they're high |
| 22:01.03 | schwagner | greg_work: i think so |
| 22:01.16 | schwagner | bjohnson: i take it you're not a fan? |
| 22:01.39 | ZX81 | ~ping you stink |
| 22:01.42 | jbot | pong you stink |
| 22:01.51 | greg_work | schwagner: where is this? all i see on their site is plans from $19.99 to $70 a month |
| 22:01.58 | bjohnson | schwagner: I might be if they didn't lock in their PAP2s and eased incoming and outgoing connections from other SIP hardware/software |
| 22:02.09 | JMan5Work | Anyone have experience with Altigen? |
| 22:02.49 | hmmhesays | wow the new phpagi rocks |
| 22:02.55 | *** join/#asterisk Bentley (~bentley@S01060080c8135e6a.cg.shawcable.net) |
| 22:02.57 | hmmhesays | it's got me all flustered |
| 22:03.00 | dsmouse | schwagner: any chance I can see the part of your config with reguard to vonage? ( even if you x out your name/pw) |
| 22:03.37 | schwagner | dsmouse: it's not there; i don't integrate my vonage line w/ *, it's just at home |
| 22:03.44 | dsmouse | oh |
| 22:03.47 | dsmouse | blah |
| 22:03.48 | bjohnson | schwagner: will they let * recieve that DID? |
| 22:03.53 | FuRR_ | anyone got any docs on intergrating * with VoiceXML |
| 22:03.56 | dsmouse | bjohnson: let? no. |
| 22:04.07 | bjohnson | exactly |
| 22:04.20 | *** join/#asterisk Mneumonic (~Mnemonic@206.231.230.230) |
| 22:04.29 | schwagner | bjohnson: yes, but you have to pay an extra like $5 a month for the sip connectivity |
| 22:04.33 | JMan5Work | how about running asterisk on a mac osx server? |
| 22:04.35 | bjohnson | is it possible other than doing a stupid fxo to their fxs port? I don't think so .. what a waste of hardware |
| 22:05.10 | harryvv | possible to chown root and asterisk for a asterisk directory? |
| 22:05.24 | schwagner | bjohnson: you can if you pay for the "mobile client" software and just don't use it |
| 22:06.35 | schwagner | no, wait, the softphone feature is $9.99 a month |
| 22:07.02 | dsmouse | if I can't get vonage to work, i'm switching to a diffrent service LNP dependant |
| 22:07.55 | three55ml | dsmouse: Vonage such in all regards |
| 22:07.55 | three55ml | sucks |
| 22:08.19 | dsmouse | three55ml: at least with packet8, there's hacks to hijack their hardware |
| 22:08.39 | three55ml | dsmouse: If you're using it with Asterisk, checkout VoicePulse connect |
| 22:08.46 | JMan5Work | * on Mac OSX? anyone? or just not on the x servers? |
| 22:08.56 | three55ml | dsmouse: You can try it for $10 |
| 22:09.05 | schwagner | I personally have a good relationship with my landline provider, and only use vonage at home, so i haven't priced it all up; i was just cuirous if anyone else had |
| 22:09.32 | dsmouse | three55ml: I was looking at them BroadVoice, and a few others |
| 22:09.54 | three55ml | dsmouse: Yeah, there's a lot of them |
| 22:11.57 | greg_work | anyone that uses SPA-841's: just did release 0.2 of my sipura config tools (configures and updates firmware on multiple units, using one config file): http://opensource.mwater.ca/projects/sipuraconfigtools |
| 22:14.26 | dsmouse | voiceplus seems more expencive then broadvoice |
| 22:14.28 | harryvv | chown asterisk /dev/zap/channel seems to resolve my persmission problems but dont think that was the most correct way to resolve it. now i can reboot the server and asterisk now loads. |
| 22:15.26 | bjohnson | schwagner: pay extra to not use their hardware? |
| 22:15.55 | *** join/#asterisk charles___ (~charles@64.35.168.55) |
| 22:15.56 | schwagner | ok, on a slightly different topic, anybody here use any graphical / web-based config tools for *, or just edit the config files directly? |
| 22:16.14 | schwagner | bjohnson: it's software, and yes ;) |
| 22:16.26 | charles___ | schwagner, You need to learn the config files |
| 22:16.49 | JMan5Work | thanks |
| 22:16.50 | schwagner | bjohnson: i just asked, i never said it actually WAS cheaper ;) |
| 22:16.55 | charles___ | schwagner, if your guy screw up the things you need to know how to fix, and also the guy will not give you the same capabilities |
| 22:16.55 | bjohnson | schwagner: looks like you already need an account and then to use the softphone is an additional $12.99 / month (http://vonage.ca/features.php?feature=softphone) |
| 22:16.59 | hajekd | any clue what codecs is using vonage? |
| 22:17.01 | *** part/#asterisk JMan5Work (~guest@mail.pc-assoc.com) |
| 22:17.26 | charles___ | hajekd, probabl G729 |
| 22:17.30 | schwagner | charles___: i did learn them, i wrote my own, i'm just asking |
| 22:17.31 | greg_work | anyone know some free faxing software for windows (that i can put on my laptop to test sending faxes to my normal fax machine connected on a voip extension)? |
| 22:17.45 | *** join/#asterisk jdims (~sleepbsd@24-119-121-6.cpe.cableone.net) |
| 22:17.48 | o-m-a-o-m-a | which windows? |
| 22:17.52 | charles___ | schwagner, so you can answer yourself. |
| 22:17.57 | greg_work | 2k |
| 22:18.06 | schwagner | bjohnson: if you need an account as well, then it's probably not worth it |
| 22:18.15 | charles___ | greg_work, fax ? why don't you send and e-mail ? |
| 22:18.29 | charles___ | greg_work, faxes are waste of bandwitdh |
| 22:18.31 | greg_work | charles___: because people still use faxes |
| 22:18.32 | schwagner | ok, does anyone else use a graphical / web-based config tool? |
| 22:18.50 | terracon | business still love the fax |
| 22:18.53 | charles___ | greg_work, because still send faxes |
| 22:19.01 | greg_work | and though i'd be happy to abolish faxes forever, you can't run a business without a fax machine |
| 22:19.18 | dsmouse | iax > sip, right? |
| 22:19.19 | hajekd | charles: interesting, according to their bandwidth saver they have ~60kbits codec, looks too much for g729 |
| 22:19.26 | charles___ | greg_work, yes you can |
| 22:19.37 | charles___ | greg_work, on the extreme case, I use fax over e-mail |
| 22:19.43 | schwagner | greg_work: i think windows has a built-in fax driver |
| 22:19.56 | charles___ | hajekd, 60Kbits for faxing |
| 22:19.58 | greg_work | ok, let me put it this way. if i removed the fax machine in the office, the rest of the people would string me up and burn me alive |
| 22:20.20 | charles___ | greg_work, use fax over e-mail |
| 22:20.26 | greg_work | anyways, this still doesnt answer my question |
| 22:20.27 | schwagner | charles___: i agree, the paperless office is a myth |
| 22:20.37 | charles___ | greg_work, it will be less painfull than putting fax to work under voip |
| 22:21.06 | greg_work | schwagner: computers just made making paper easier |
| 22:21.35 | schwagner | heh, and making more of it ;) |
| 22:21.47 | greg_work | charles___: while using email is great, its still a lot harder to send certain things, like forms filled out by hand. |
| 22:21.52 | charles___ | bkw_, What calling card do you recommend ? do you know which one is in fully progress, as I can see asterisk-calling-card is stoped since july |
| 22:21.52 | greg_work | charles___: or signed documents |
| 22:22.23 | charles___ | greg_work, who fill forms by hand ? |
| 22:22.25 | greg_work | charles___: scanners to email software generally sucks, and isnt as easy to use as a fax machine, plus not everyone has a scanner |
| 22:23.03 | greg_work | charles___: guess you don't work in an office ;p |
| 22:23.04 | greg_work | anyways |
| 22:23.12 | schwagner | greg_work: did you try the windows fax driver? |
| 22:23.26 | charles___ | greg_work, I work in a cavern. |
| 22:23.34 | schwagner | i know it's in winxp, and i think it is in 2k as well |
| 22:23.51 | greg_work | schwagner: i dont see any windows fax stuff |
| 22:24.11 | schwagner | greg_work: go to printers and faxes |
| 22:24.57 | schwagner | file -> install a local fax printer |
| 22:25.54 | greg_work | apparently my modem driver isn't installed, maybe thats the problem.. |
| 22:26.35 | schwagner | hmmm.. on second thought, maybe you don't get to have faxes on win2k |
| 22:26.56 | schwagner | well, it works on xp anyway (i'm sure this helps you...) |
| 22:27.08 | charles___ | Do you guys know which of the Calling Card app is still being maintained ? |
| 22:27.10 | wankel | hmm. i thought 2k had it, but maybe not. it's been a long time. |
| 22:27.25 | wankel | maybe you can find a copy of winfax on ebay :) |
| 22:27.36 | ariel_ | w2k does support faxing |
| 22:27.39 | charles___ | use hyla fax |
| 22:27.45 | charles___ | hyla fax rulez |
| 22:27.46 | redder86 | Yes, HylaFAX. |
| 22:27.49 | redder86 | :-) |
| 22:27.52 | wankel | ugh. hylafax on windows? |
| 22:28.04 | o-m-a-o-m-a | sendfax/mgetty is my combination :-) |
| 22:28.05 | charles___ | wankel, yes what's the problem ? |
| 22:28.15 | redder86 | Windows? |
| 22:28.28 | charles___ | the thing is, windows cannot do anything by itself |
| 22:28.36 | charles___ | you will need a hylafax server |
| 22:28.44 | redder86 | If you are setting up a system for faxing, don't bother with Windows. |
| 22:29.01 | wankel | i assume this all started with someone wanting to fax something from windows |
| 22:29.03 | greg_work | yeah maybe this would be easier just to setup hylafax... |
| 22:29.24 | wankel | in that case, hylafax is hardly the easiest solution. if you want to set up a big fax server, sure, use hylafax. |
| 22:29.32 | greg_work | wankel: i was trying to find a free windows faxing program that i could install quickly to TEST my real fax connected to a spa-2000 |
| 22:29.51 | wankel | oh. use efax. |
| 22:29.52 | ZX81 | I install a proggy which makes a tif file from windows printer interface and saves it in smb directory |
| 22:29.54 | charles___ | greg_work, man just send a fax to a friend of yours |
| 22:29.57 | wankel | or can you not reach it publicly? |
| 22:29.59 | ZX81 | I check directory with cron |
| 22:30.03 | ZX81 | and send the faxes |
| 22:30.10 | ZX81 | the filename is the destination |
| 22:30.13 | ariel_ | greg_work, Windows 2000 if you have a fax modem can fax out of it with its fax program. |
| 22:30.17 | wankel | or hell, just ask someone to fax you something. |
| 22:30.24 | ZX81 | then for received faxes they go to email |
| 22:30.45 | ZX81 | with spandsp |
| 22:30.50 | ZX81 | for sending etc |
| 22:30.55 | ZX81 | but only on PRI |
| 22:30.56 | ZX81 | :) |
| 22:31.01 | ZX81 | and no voip |
| 22:31.15 | ZX81 | meh |
| 22:31.16 | ariel_ | spandsp works with TDM11b I have |
| 22:31.21 | greg_work | wankel: efax will cost me $33 :p |
| 22:31.38 | wankel | for one fax? jesus. efax used to be cheap. |
| 22:31.54 | greg_work | $15 signup + $18/mo |
| 22:32.00 | wankel | if the number is reachable publicly and isn't some internal pbx extension i'll just fax you something. |
| 22:32.02 | o-m-a-o-m-a | good hint... anyone with an incoming CAPI call without VBox in here? |
| 22:32.16 | znoG | www.faxonline.com.au do fax to email and vice versa too. |
| 22:32.36 | srt | o-m-a-o-m-a: without vbox? what do u mean? |
| 22:32.39 | ariel_ | efax hummmm I pay 12.95 per month for my account they muct of gone up. |
| 22:33.00 | o-m-a-o-m-a | i want to config one of my MSN without any VBox activity |
| 22:33.11 | srt | why do u use vbox at all? |
| 22:33.17 | schwagner | greg_work: just go download an eval, if you're just gonna use it for testing |
| 22:33.19 | greg_work | wankel: i apreciate that, but not quite ready to test yet |
| 22:33.38 | greg_work | thats a good idea schwagner :) |
| 22:33.39 | o-m-a-o-m-a | but I get 4 calling tones for external callers, 2 at my phone then on both busy |
| 22:33.42 | greg_work | didnt even think about it |
| 22:33.56 | *** join/#asterisk luisgrin (~luis@209.99.227.220) |
| 22:34.00 | schwagner | 30-day trials are great :) |
| 22:34.19 | o-m-a-o-m-a | I use the asterisk-internal vbox |
| 22:34.30 | schwagner | so, anybody here use/used AMP (asterisk managment portal)? |
| 22:35.55 | ariel_ | greg_work, if you go to your w2k control pannel you will see an Icon called fax that is what you can use to configure and send faxs via argh the w2k. |
| 22:35.57 | greg_work | schwagner: yeah, i use it |
| 22:36.06 | greg_work | its .. ok. i do lots of editing by hand tho |
| 22:36.14 | znoG | question |
| 22:36.21 | wankel | answer |
| 22:36.31 | znoG | if I dial "17471231233", why would it use iaxtel instead of sipphone? |
| 22:36.31 | znoG | exten => _1[75][41]7XXXXXXX,1,Dial(SIP/${EXTEN}@sipphone,60,tr) |
| 22:36.32 | znoG | exten => _17XXNXXXXXX,1,Dial(IAX2/iaxtel/${EXTEN}@iaxtel) |
| 22:36.33 | schwagner | greg_work: will it clobber over my existing config files? |
| 22:36.45 | greg_work | schwagner: amp's install will clobber half your system |
| 22:36.52 | schwagner | great |
| 22:36.55 | hajekd | Looks like Grandstream does not work with G726 codecs .... with asterisk |
| 22:36.56 | greg_work | :) |
| 22:37.00 | *** join/#asterisk jsolares (~jsolares@200.30.141.85) |
| 22:37.26 | greg_work | its not the nicest install.. hardcoded everything, it just overwrites files without asking. i'd suggest doing it by hand if you're unsure |
| 22:37.26 | ariel_ | Amp is made to be the all controller of you asterisk setup. |
| 22:37.40 | greg_work | it overwrites extensions.conf, and a few others |
| 22:37.52 | schwagner | so, let's say i install an * server / phones system at a client's site; could they use AMP for really basic switch administration? |
| 22:37.56 | greg_work | ariel_: not really, though it is starting to control more and more of it |
| 22:38.01 | greg_work | schwagner: yes |
| 22:38.29 | jsolares | schwagner: yes, but really weird dialplans would be a no go |
| 22:38.33 | ariel_ | I only use amp for the reports |
| 22:38.39 | schwagner | but every time they changed something, it would clobber my extensions.conf file |
| 22:38.44 | jsolares | yeah, that's what i'm doing too ariel_ |
| 22:38.52 | schwagner | hmmm |
| 22:38.53 | greg_work | you can add extensions, configure IVR's, and sort of control inbound calls. you can also setup voip trunks but its a bit limited now (i'm actually just going to start writing some dialplan control stuff for trunks) |
| 22:39.03 | ariel_ | that is what backups work great. |
| 22:39.19 | greg_work | schwagner: theres some hooks in it, where it includes extensions_custom.conf etc, which you can modify. but its still somewhat limited |
| 22:39.25 | schwagner | greg_work: i guess i'll just have to try it out |
| 22:39.30 | ariel_ | I copy all my .conf files to a tmp directory upgrade the amp then move my conf files back. no problems. |
| 22:40.33 | greg_work | ariel_: thats a bit pointless, as some amp upgrades add additional features to extensions.conf |
| 22:40.38 | *** join/#asterisk imagmo (~imagmo@c-24-20-249-117.client.comcast.net) |
| 22:40.44 | znoG | interesting, in the dialplan as soon as i use [] to group numbers, it uses another dial plan. seems to go for the strict ones first |
| 22:41.26 | ariel_ | greg_work, I do my own macro's and dialing rules so I don't need there stuff. I just like there reports. |
| 22:41.45 | ariel_ | In fact I use asterisk@home maintence section and edit my config files there. |
| 22:42.09 | greg_work | ariel_: i've never even seen asterisk@home |
| 22:42.34 | ariel_ | greg_work, it's the quickest way to get an asterisk up a running. |
| 22:43.12 | ariel_ | greg_work, go to http://asteriskathome.sourceforge.net/ |
| 22:43.17 | jsolares | i'm doing text to speech synthesys with my asterisk currently using festival, what other free tts engines are out there that might work |
| 22:44.00 | ariel_ | download there file tar not the iso and using there inst it will install asterisk, spandsp, FOP and lots of good stuff for you. |
| 22:44.31 | harryvv | anyone care to help give me a idea why im getting persmssion errors of asterisk not able to gain access to /dev/zap/channel in debug? I can chown asterisk to that direcoty and debian loads asterisk fine but then my xlite cannot authenticate against asterisk and times out. |
| 22:44.41 | ariel_ | I used it with my Fedora Core 1 and also White Box linux. I am now using it on CentOS which is like Whitebox. |
| 22:45.34 | *** join/#asterisk PatrickDK (patrickdk@dyn-19-218.myactv.net) |
| 22:45.51 | ariel_ | znoG, I use them but off includes |
| 22:46.26 | harryvv | ariel, does your * automaticly load on a reboot? |
| 22:46.28 | greg_work | ariel_: i use xorcom's rapid asterisk debian packages |
| 22:47.16 | ariel_ | I see well that will work. I am used to the RH way of files locations so I stayed away from debian. |
| 22:47.38 | Qwell | harryvv: All my stuff in /dev/zap/ is 644 |
| 22:47.48 | harryvv | okay |
| 22:47.57 | *** join/#asterisk jesse_132 (~chatzilla@207.246.72.150) |
| 22:48.02 | harryvv | qwell and chown? is root? |
| 22:48.03 | ariel_ | harryvv, yes |
| 22:48.07 | Qwell | root:root |
| 22:48.10 | harryvv | k |
| 22:48.21 | Qwell | /dev/zap/channel is 196,254 |
| 22:48.32 | jesse_132 | do I use "Context" to create two different "PBX" environment for users? (1 server, two companies with seperate voicemail/.../music on hold/ ... |
| 22:48.46 | Qwell | jesse_132: I would use different contexts, yeah |
| 22:48.47 | greg_work | ariel_: i switched from redhat to debian a long time ago, and never looked back. after using up2date and rhn, dpkg and apt are amazing |
| 22:49.18 | thieumS | yep debian rox </troll> |
| 22:49.20 | ariel_ | greg_work, I like yum don't use up2date. I have had problems with apt. |
| 22:49.23 | eKo1 | yum is good too. |
| 22:49.34 | harryvv | qwell what do you mean 196 and 254? |
| 22:49.48 | brc_ | jesse_132, yes, more or less... |
| 22:49.51 | Qwell | harryvv: its a char device, those are major/minor |
| 22:50.08 | jesse_132 | brc_, what is the less? |
| 22:50.12 | harryvv | i just checked everythng is 644 for that directory. |
| 22:50.22 | greg_work | ariel_: yum is ok, but its still SO slow. apt on redhat isn't quite the same as apt on debian, with its 18k packages or whatever it is at now |
| 22:50.27 | harryvv | and did a chown root:root /dev/zap and rebooting now. |
| 22:50.38 | Qwell | shouldn't need to reboot... |
| 22:50.46 | Qwell | and you'll need to chown -R |
| 22:50.59 | Qwell | (unless you did /dev/zap/*, of course) |
| 22:51.11 | harryvv | everything in /dev/zap |
| 22:51.17 | harryvv | forgot the -r |
| 22:51.21 | ariel_ | harryvv, If you like it great. That is why there are many flavor of Linux distro's. I am just lasy and don't want to learn another distro type. |
| 22:51.38 | harryvv | im sticking with debian |
| 22:51.52 | harryvv | I have pretty much tried them all over the years. |
| 22:52.37 | ariel_ | I actually like a distro that is based on debian for desktops. It's called Mepis but just for desktop's. |
| 22:53.55 | brc_ | ubuntu |
| 22:54.00 | brc_ | it's very nice |
| 22:54.11 | brc_ | use the new testing release though |
| 22:54.18 | brc_ | hoary iirc |
| 22:54.20 | eKo1 | Debian is an excellent desktop distro.. |
| 22:54.39 | brc_ | debian still uses xfree86... |
| 22:54.43 | brc_ | ubuntu hoary uses x.org |
| 22:54.53 | Qwell | hoary? |
| 22:55.12 | brc_ | it's the next release |
| 22:55.19 | Qwell | oh |
| 22:55.26 | brc_ | warty (the one they have cd's of) is still xfree |
| 22:57.17 | o-m-a-o-m-a | Gute Nacht |
| 22:57.44 | jesse_132 | brc_, I think if you do apt-get update you get xorg on warty |
| 22:58.04 | eKo1 | Mac OS X is the best desktop unix-based OS in my opinion. |
| 22:58.12 | brc_ | jesse_132, okay...I dunno |
| 22:58.25 | ariel_ | Mac's oh boy now that is something else all together. |
| 22:58.29 | jesse_132 | brc_, I've not had sucess with hoary yet :( .. but it was a month ago ... I'll wait for real release ... :) |
| 22:58.50 | brc_ | worksforme(TM) |
| 22:58.56 | *** join/#asterisk afrosheen (afrosheen@txprotoa8.august.net) |
| 22:59.24 | afrosheen | I just did zap destroy channel 7, now channel 7 is gone and won't come back, even if i reset asterisk...what do I do now? |
| 22:59.28 | brc_ | I've also got a dozen or so alternate repositories for *ahem*other stuff |
| 22:59.52 | brc_ | did jah know if you know where to look you can find a repos with .debs for the jdk? |
| 22:59.53 | eKo1 | afrosheen: Power cycle. |
| 23:00.00 | ariel_ | afrosheen, if you do that you need to restart asterisk |
| 23:00.03 | *** join/#asterisk [Latre] (~latre@148.233.19.133) |
| 23:00.10 | brc_ | no insane builddeb dances =) |
| 23:00.24 | afrosheen | ariel_: i did |
| 23:00.24 | brc_ | no no no |
| 23:00.28 | brc_ | just unload the modules |
| 23:00.30 | brc_ | lsmod |
| 23:00.48 | afrosheen | brc_: sounds like a plan, I'll try it |
| 23:00.48 | brc_ | rmmod whatever |
| 23:00.54 | brc_ | modprobe whatever |
| 23:00.54 | ariel_ | afrosheen, stop now, then ztcfg -vvv then safe_asterisk |
| 23:01.12 | brc_ | make sure you rmmod whatever card, AND zaptel, but don't modprobe zaptel, just the module name |
| 23:01.29 | brc_ | yeah, and ztcfg afterwards |
| 23:01.37 | ariel_ | sometimes just service zaptel restart will do it too. |
| 23:01.47 | brc_ | depends on the distro.. |
| 23:01.57 | ariel_ | brc_, like I said sometimes |
| 23:01.58 | [Latre] | hi, where i found howto configure a TDM04B with examples? i check page of digium and voip-info but i dont understand the part of context........ |
| 23:02.15 | afrosheen | service zaptel restart, worked but the channel is still gone |
| 23:02.18 | brc_ | [Latre], you don't understand what a context is? |
| 23:02.27 | [Latre] | yes |
| 23:02.29 | ariel_ | afrosheen, reboot |
| 23:02.41 | afrosheen | ..ugh, the R word |
| 23:02.41 | brc_ | [Latre], best thing I can suggest is to search the wiki with google and find a page on it, or read the asteriskdocs project |
| 23:02.45 | brc_ | ~asterisk docs |
| 23:02.46 | jbot | asterisk documentation project is probably at http://asteriskdocs.org |
| 23:02.51 | afrosheen | how could something like this be so dangerous blargh |
| 23:03.06 | brc_ | afrosheen, personally I'd manually reload the modules |
| 23:03.14 | afrosheen | brc_: they wouldn't unload |
| 23:03.22 | brc_ | what did it say? |
| 23:03.24 | ariel_ | well you did destroy it. |
| 23:03.34 | afrosheen | same thing it always says when it won't unload, busy |
| 23:03.40 | brc_ | ahh |
| 23:03.47 | brc_ | there is a way to force unload a module |
| 23:03.52 | ariel_ | afrosheen, are you getting zombies? |
| 23:03.53 | afrosheen | rmmod -f whatever |
| 23:04.06 | brc_ | and that doesn't work either? |
| 23:04.10 | afrosheen | it's rebooting |
| 23:04.14 | afrosheen | other admin jumped the fence |
| 23:04.14 | brc_ | oh.. |
| 23:04.17 | brc_ | goodluck... |
| 23:04.20 | afrosheen | windows admin |
| 23:04.21 | afrosheen | loves to reboot |
| 23:04.24 | brc_ | hahah |
| 23:04.38 | brc_ | rebooting rarely does anything that can't be done without rebooting |
| 23:04.44 | ariel_ | Windows admin what are they doing in a linux box. |
| 23:04.47 | brc_ | cept kernel replacements |
| 23:04.55 | shmaltz | ~seen ManxPower |
| 23:04.57 | jbot | manxpower <~eric@dsl-209-205-172-111.i-55.com> was last seen on IRC in channel #asterisk, 19h 35m 1s ago, saying: 'Nugget, We should get tzanger's opinion!'. |
| 23:05.02 | darkskiez | brc_: i'm sure theres a patch to change kernel without a reboot somewhere |
| 23:05.10 | brc_ | yup |
| 23:05.14 | afrosheen | ariel_: it's his network, his job...my consultancy :) |
| 23:05.18 | brc_ | that'd be fun to try some time |
| 23:05.28 | znoG | anyone have a 1-747 number I can try and call? (sipphone) |
| 23:05.29 | ariel_ | afrosheen, ah I see.... |
| 23:05.36 | darkskiez | I've needed to reboot to fix a confused dvd writer |
| 23:05.38 | brc_ | I hope yer charging plenty then |
| 23:05.40 | afrosheen | ariel_: and he's real twitchy |
| 23:05.46 | brc_ | I charge extra when stupid people don't let me do my job |
| 23:05.49 | afrosheen | lol |
| 23:05.50 | darkskiez | and to repair my filesystem after a filesystem crash |
| 23:05.56 | brc_ | I'm not kidding |
| 23:06.05 | afrosheen | brc_: idiot tax |
| 23:06.12 | brc_ | yup |
| 23:06.16 | darkskiez | OOooo |
| 23:06.21 | brc_ | you've just gotta use the right phb speak on the invoice |
| 23:06.38 | darkskiez | get a premium rate phone number, register it with e164, and stupid people have to pay. |
| 23:06.40 | darkskiez | Magic. |
| 23:06.50 | ariel_ | znoG, no but 305 or 786.... |
| 23:08.02 | ariel_ | brc_, I had to reboot the yesterday kept getting segfaults.... |
| 23:08.06 | Guigui|taff | anyone know how to load two hisax card ? |
| 23:08.30 | ariel_ | what is a hisax card? |
| 23:08.35 | Guigui|taff | isdn card I mean |
| 23:08.47 | Guigui|taff | and I use hisax linux kernel module |
| 23:08.58 | afrosheen | hizzax |
| 23:09.03 | Guigui|taff | no hisax :) |
| 23:09.32 | ariel_ | afrosheen, did it come back up? |
| 23:09.37 | afrosheen | ariel_: of course |
| 23:09.42 | afrosheen | but I didn't want to reboot |
| 23:09.43 | Guigui|taff | so nobody knows ? |
| 23:09.43 | afrosheen | :) |
| 23:09.49 | afrosheen | Guigui|taff: nobody uses those |
| 23:09.59 | afrosheen | in america, isdn is unnecessary |
| 23:10.06 | Guigui|taff | hm |
| 23:10.31 | harryvv | afro, for anyone outside the boundries of dsl it is. |
| 23:10.36 | Guigui|taff | :( |
| 23:10.39 | afrosheen | we used it about 10 years ago pretty much everywhere |
| 23:10.45 | ariel_ | afrosheen, be nice. I still have a customer with isdn connection for internet access. |
| 23:11.05 | harryvv | dslx is what limites to 24 thousand feet at the most? |
| 23:11.09 | harryvv | limited |
| 23:11.11 | afrosheen | of course, there are exceptions to any rule, but by and large, the US just doesn't use it much anymore |
| 23:11.25 | Guigui|taff | okay |
| 23:11.31 | brc_ | yup |
| 23:11.32 | harryvv | afro, if your in the country might not have a choice. |
| 23:12.09 | moonwick | heh |
| 23:12.25 | *** join/#asterisk nickv111 (~nickv111@69-170-98-48.clspco.adelphia.net) |
| 23:12.58 | jsolares | anyone know what codecs i should use with the avaya 4602 |
| 23:12.59 | afrosheen | harryvv: if you're in the country you don't deserve broadband :p |
| 23:12.59 | nickv111 | Where could I buy an X100P? |
| 23:13.15 | ariel_ | damm kernel-source not loaded... not back to installing.... |
| 23:13.19 | afrosheen | harryvv: at least that's how the phone company looks at it right? |
| 23:13.42 | JunK-Y | how can i solve that shit? |
| 23:13.43 | JunK-Y | Feb 15 18:12:46 WARNING[31484]: rtp.c:874 ast_rtcp_new: Unable to allocate socket: Too many open files |
| 23:13.45 | JunK-Y | ?% |
| 23:13.52 | ariel_ | brc_, says so what...... |
| 23:13.56 | afrosheen | JunK-Y: ooh we had this one last week |
| 23:14.01 | Qwell | ulimit? |
| 23:14.03 | afrosheen | yep |
| 23:14.05 | Qwell | limit.h |
| 23:14.40 | ariel_ | jsolares, ulaw |
| 23:15.02 | jsolares | so disallow all and just ulaw right? |
| 23:15.08 | JunK-Y | ulimit -n is set to 10240 |
| 23:15.34 | ariel_ | jsolares, yes it's called on there phones if I remember there setup g711u |
| 23:15.35 | afrosheen | my brother was so sick of not having broadband in his tiny town he built a wireless link, with an antenna on a grain silo in the next town where they hav dsl |
| 23:16.15 | JunK-Y | Qwell: which value should i put? |
| 23:16.18 | ariel_ | afrosheen, I am going to be starting to make a house in the farmland area down here and the only internet access there will be sat. |
| 23:16.21 | Qwell | JunK-Y: dunno |
| 23:16.26 | Qwell | "unlimited" is valid though |
| 23:16.28 | JunK-Y | afrosheen: and did ya solve that problem? |
| 23:16.55 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 23:17.13 | jsolares | now to find out how to configure those phones to allow dialing 8digit numbers... meh the grandstream's are easier to configure |
| 23:17.28 | afrosheen | JunK-Y: I mean, your question was posed to the channel and solved |
| 23:17.59 | afrosheen | ariel_: satellite is worse than dialup imho |
| 23:18.22 | afrosheen | ariel_: I suggest you figure out where you can bounce a wireless signal in from |
| 23:18.57 | JunK-Y | afrosheen: huh? |
| 23:19.07 | JunK-Y | i cant generate more then 143 calls atm. |
| 23:19.14 | ariel_ | afrosheen, well I have customers on sat and there using voip service as well it's working. |
| 23:19.47 | afrosheen | ariel_: working? or working well? I've seen nothing but 1000ms latency at my brother's house on his old sat connection |
| 23:20.04 | loud | ariel_, sat+g.729 is ok for me. |
| 23:20.08 | loud | 750 ms. |
| 23:22.22 | *** join/#asterisk ReVoK (ReVoK@82.224.60.46) |
| 23:22.26 | ariel_ | afrosheen, I have a few customers in the island out in the caribian and there getting around 650 to 740ms |
| 23:22.34 | ReVoK | hi |
| 23:23.10 | ReVoK | some one to help a noob(me), to get it started ? :) |
| 23:24.04 | Qwell | Get what started? |
| 23:24.31 | Qwell | getting dressed: pants go on one leg at a time |
| 23:24.39 | ariel_ | ReVoK, ask a question. |
| 23:24.51 | ariel_ | Qwell, so funny |
| 23:24.53 | ReVoK | nice :) |
| 23:24.53 | Qwell | starting a car: turn the key |
| 23:26.20 | *** join/#asterisk Firestrm (F1r3570rm@S010600047577bccd.gv.shawcable.net) |
| 23:26.24 | Qwell | still have yet to see a question asked... |
| 23:26.27 | *** join/#asterisk okieplaya (~okieplaya@cdm-208-180-154-4.slsp.cox-internet.com) |
| 23:26.31 | ariel_ | Damm forgot to setup key on Centos GPG key that is... gone to get keys. |
| 23:27.02 | Firestrm | who here knows about setting up PRI's and DID's? I want to start offering local numbers for my area.. |
| 23:27.08 | ariel_ | Qwell, maybe he/she is trying to think. |
| 23:27.25 | ariel_ | Firestrm, ask away. |
| 23:27.29 | Qwell | ariel_: probably. I'm passing the time |
| 23:28.10 | ariel_ | I should go and start dinner. If I don't cook it it's not going to happen. |
| 23:28.14 | *** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com) |
| 23:28.17 | Firestrm | ariel_, what is needed as far as equipment? what do i need to order from the telco besides the PRI line? How are phone numbers assigned? |
| 23:28.48 | ariel_ | Firestrm, oh boy. that is a loaded question. |
| 23:28.54 | Firestrm | :) |
| 23:29.28 | Firestrm | ive allready asked all the easy questions :) |
| 23:29.32 | ariel_ | Lets see frist you need to see what you really want to do. 2nd you start to get quotes on your area for PRI lines and internet data lines. |
| 23:30.00 | Firestrm | ariel_, im working on that.. |
| 23:30.01 | ariel_ | PRI lines here are around 500 per month and did's are about 3.20 for group of 20. |
| 23:30.24 | ariel_ | Next comes the asterisk box and the t100p card. |
| 23:30.27 | Qwell | 3.20 what? |
| 23:30.42 | ariel_ | $ 3.20 for a group of 20 did's. |
| 23:30.44 | Qwell | $3.20 per DID? |
| 23:30.50 | Qwell | oh, wow, $3.20 for all 20? |
| 23:31.08 | ariel_ | well it's a round number I have seen more and have seen less. |
| 23:31.28 | Firestrm | ariel_, is 3.20 per did or for the whole group of 20? |
| 23:31.33 | ariel_ | X/O gave me a quote for 100 did's at 19 dollars per group. |
| 23:31.40 | ariel_ | Group |
| 23:31.44 | ReVoK | well, it's for a project, i just want for the moment, to have a asterix server runing, and two clients conected to the server (using H323) and make them having a VoIP call (using mic and sound card) |
| 23:31.46 | Firestrm | ok, that makes more sense |
| 23:31.48 | Qwell | wow, cheap |
| 23:32.12 | Firestrm | ok, what about full pstn,, ie dial in and out? |
| 23:32.24 | ariel_ | Qwell, yes it's cheap but I like them together so I needed to get them in the group of 20 there picky. |
| 23:32.31 | Qwell | ahh |
| 23:32.32 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 23:32.57 | ariel_ | Firestrm, if you want to setup a pri it comes with both. in and out to your local area in fact here you get 3 different areas |
| 23:33.37 | Firestrm | what does it cost per pstn line out of a pri in your area? |
| 23:33.53 | ariel_ | Firestrm, don't understand? |
| 23:34.19 | ariel_ | If your talking a pots line well here it depends on if it's metered or unmetered. |
| 23:34.23 | Firestrm | i may have my terminology screwed up.. does a pri come WITH 23 phone numbers? |
| 23:34.31 | Firestrm | in and out? |
| 23:35.02 | ariel_ | Firestrm, no it comes with 23 channels which you have 23 of the used at one time. you can have 1000 of did's on them if you want. |
| 23:35.28 | Firestrm | ariel_, but you cant dial out a did can you? |
| 23:35.29 | ariel_ | Firestrm, either for inbound or outbound 23 is all you get per pri. |
| 23:36.04 | ariel_ | Firestrm, hehehe you don't understand don't think in terms of numbers or did but channels for voice. |
| 23:36.21 | Qwell | DIDs are tacked on later, heh |
| 23:36.41 | ariel_ | you can have one did and people call one number and can have 23 people on that the system. |
| 23:36.50 | Firestrm | ariel_, i think i get it.. so the phone number assigned with the DID line is only for incoming, whe you dial out it goes directly out of one of the PRI lines. |
| 23:37.26 | *** join/#asterisk jcims (~jcims@cpe-69-135-121-57.columbus.rr.com) |
| 23:37.33 | file[laptop] | it's not a DID line, it's just a phone number.... it can come in on any of the channels depending on how you set it up |
| 23:37.38 | ariel_ | Firestrm, did are not assigned per channel. |
| 23:37.38 | Firestrm | im just trying to understand it so i dont look like a sucker when asking my telco for service.. |
| 23:37.38 | file[laptop] | you gotta think of channels |
| 23:37.50 | Qwell | Is it normal for a place to have out only DIDs? |
| 23:37.54 | Qwell | my work does that |
| 23:38.41 | ariel_ | Firestrm, you can have 100 numbers each one asigned to a person or extension in your system But you can only get 23 inbound or 23 outbound calls at one time. |
| 23:38.56 | Firestrm | ariel_, ok so if someone assigned did number 555-1234 wishes to use my server to dial out, i take one of my line of the 23 and use it for outdial? |
| 23:39.02 | *** join/#asterisk fdp32 (fdp32@200.63.70.12) |
| 23:39.20 | ariel_ | Firestrm, yes |
| 23:39.27 | *** join/#asterisk syslod (~sysglod@65.114.15.70) |
| 23:39.29 | Firestrm | im starting to understand.. |
| 23:39.38 | harryvv | arial so basicly 23 out of a pool of 1000 can be used at any one time |
| 23:39.38 | Qwell | Usually, each channel isn't given a specific DID either, is it? |
| 23:39.55 | Qwell | ie: in order for somebody to get 2 incoming calls at once |
| 23:39.57 | ariel_ | harryvv, yes |
| 23:40.13 | ariel_ | but it also depends on inbound you share inbound with outbound. |
| 23:40.40 | ariel_ | so if your system made 10 outbound calls there is only 13 inbound calls able to get in. |
| 23:40.43 | Firestrm | so i would set the caller id stuff to 555-1234 before making outboud call, giving the appeariance of coming from that phone number, when in reality its coming from a pool of 1000 or more number all pooled on a 23 line PRI? |
| 23:41.14 | file[laptop] | it doesn't come from any of the numbers, it's just another voice channel in use... if you set the outbound callerid to the phone number, so be it |
| 23:41.18 | ariel_ | Firestrm, yes in most cases yes if the telco allows you to set it. that is something you have to ask them for. |
| 23:41.47 | Qwell | ariel_: goes back to my earlier question. Is it normal to have "outgoing only" DIDs? |
| 23:41.53 | Firestrm | ok .. i understand now.. Do the Telco's use the Terminology DID, or do they call its somthing else? |
| 23:42.18 | Qwell | ie; 555 - 1000 to 1100 are incoming, 555-1101-1110 are outgoing |
| 23:42.34 | ariel_ | Firestrm, yes and also call it something else it depends on who you talk to. Some of there sales people are well programmed incorrectly. |
| 23:43.16 | harryvv | so what happens in say a sales floor setting where there are 30 phones answeing from one phone number? |
| 23:43.26 | Firestrm | ariel_, im dealing with telus.. and with them, the price goes up inversly proportional to your percieved knowlege level. |
| 23:43.26 | Qwell | harryvv: queues |
| 23:43.50 | Firestrm | ariel_, they attempt to rape yuou for all they can |
| 23:43.52 | harryvv | are the lines from the telcos multiplexed into one pri? |
| 23:44.05 | file[laptop] | channels, think of channels! |
| 23:44.26 | Qwell | Not sure if its what he's asking or not... |
| 23:44.29 | file[laptop] | you have 23 channels available, 20 people call the same phone number, telco sends each call down an available channel |
| 23:44.33 | Qwell | but if its more then 23 calls, how does that work? |
| 23:44.41 | file[laptop] | you can have them to another PRI |
| 23:44.44 | harryvv | i see |
| 23:44.53 | Qwell | DIDs aren't connected to a specific PRI then? |
| 23:45.11 | harryvv | makes sence now |
| 23:45.16 | file[laptop] | how do you think high... ugh I forgot the word... capacity places deal with it? |
| 23:45.18 | Qwell | or does the PRI say, "Hey Jon, I'm busy, can you get this?" Then the other PRI says, "Sure Bill" |
| 23:45.40 | Qwell | call centers? |
| 23:45.56 | file[laptop] | we have a few PRIs at work... our telco will send down calls to any available channel |
| 23:45.58 | file[laptop] | regardless of PRI |
| 23:46.01 | Qwell | ahh |
| 23:46.06 | harryvv | what about voip to pstn you are cutting avaible channels in half because one half is data and the other is voice? |
| 23:46.29 | file[laptop] | harryvv: usually you have a separate data connection |
| 23:46.35 | harryvv | okay |
| 23:46.40 | harryvv | and that cost more of course |
| 23:46.42 | file[laptop] | one for TDM interconnect, one for data |
| 23:47.11 | Qwell | are things like DS3 voice also? |
| 23:47.12 | syslod | Anyone got any suggestions for configuring the new T1/E1 card to work with a 12FXO/12FXS CA Bank I ? |
| 23:47.13 | harryvv | im trying to recall my past telco instruction from my instructor. |
| 23:47.23 | harryvv | 24 pair |
| 23:47.29 | *** part/#asterisk mountainm2k (~freenodei@cbit-98.bullseye9.com) |
| 23:47.31 | harryvv | What is a t-1 again |
| 23:47.34 | terrapen | anyone read the article "VoIP for Deployed Soldiers" on /.? |
| 23:47.38 | harryvv | 24 pair of 64 |
| 23:47.42 | file[laptop] | Qwell: can be voice |
| 23:47.46 | greg_work | hm, so i can receive faxes over voip, but not send properly receiving for the most part is error free, sending is either garbled or flat out gives an error |
| 23:47.46 | Qwell | PRI is a T1 |
| 23:47.49 | harryvv | k |
| 23:47.58 | terrapen | i offered the guy a preconfigured * server |
| 23:48.01 | fdp32 | hi, i need information to setup new h323 user with asterisk, can u help me |
| 23:48.08 | file[laptop] | Qwell: PRI is a T1 with a data channel... |
| 23:48.19 | Qwell | file[laptop]: yeah... |
| 23:48.22 | harryvv | how many pair for a t-1 again |
| 23:48.25 | file[laptop] | DS3 can be a T3 with a data channel... |
| 23:48.27 | Sed_bbiab | 2 |
| 23:48.30 | syslod | or 4 |
| 23:48.33 | file[laptop] | or a 44Mbps internet connection if you want |
| 23:48.36 | syslod | usually 4 |
| 23:48.37 | harryvv | 24 pairs total? |
| 23:48.43 | Qwell | thought T3 was 155? |
| 23:48.51 | Sed_bbiab | thats oc3 |
| 23:48.55 | Qwell | ahh |
| 23:48.55 | terrapen | how many phone lines over a DS3? |
| 23:48.59 | syslod | HTU-R <=> HTU-C = 4 pair |
| 23:49.07 | file[laptop] | terracon: 672 _CHANNELS_ |
| 23:49.09 | syslod | 28 * 24 |
| 23:49.15 | terrapen | channel, durr |
| 23:49.16 | terrapen | ok |
| 23:49.20 | terrapen | that's not bad |
| 23:49.24 | Qwell | 28 seems like an odd number |
| 23:49.39 | harryvv | I also know the data is 56kb and the rest up to 64 is overhead ecc and routing information. |
| 23:49.50 | terrapen | i'll bet a voice ds3 is pretty expensive |
| 23:50.02 | syslod | $4600 /monthly |
| 23:50.09 | terrapen | dadgum. |
| 23:50.12 | harryvv | yea but how many channels? |
| 23:50.16 | syslod | ALl |
| 23:50.16 | Qwell | Thats not much, for nearly 700 channels |
| 23:50.26 | harryvv | fill them up! |
| 23:50.26 | harryvv | ;) |
| 23:50.28 | terrapen | i should open up a phone sex line |
| 23:50.30 | Qwell | considering a T1 will cost ~500-1000 |
| 23:50.38 | fdp32 | hi, i need information to setup new h323 user with asterisk, can u help me with any usefull link |
| 23:50.44 | terrapen | with pre-recorded messages and voice recognition |
| 23:50.45 | harryvv | over 1 grand here from what I hear Qwell |
| 23:50.59 | Qwell | harryvv: yeah, it varies alot, from what I've heard |
| 23:51.15 | syslod | depends on tarriffs and how hungry ppl are. |
| 23:51.27 | Qwell | I'm sure a DS3 will vary just as much(if not more) |
| 23:51.44 | syslod | Varies alot. In some places its 10,000+ |
| 23:52.53 | harryvv | I think i will take off and see this |
| 23:52.56 | harryvv | http://www.redbackopenhouseca.com/ |
| 23:53.03 | harryvv | dsl supplier |
| 23:53.12 | harryvv | err mfg |
| 23:53.13 | harryvv | :) |
| 23:53.15 | Sedorox | anyone knoe what a T1 in Telus area runs for.. for half data half voice? |
| 23:53.30 | harryvv | sed, i hear for over 1 grand |
| 23:53.48 | Qwell | ReVoK: no messages please |
| 23:53.49 | *** join/#asterisk iMediax (lklk@00045a809589.click-network.com) |
| 23:54.22 | Sedorox | kk |
| 23:54.32 | Sedorox | I saw a price for $400 |
| 23:54.34 | Sedorox | but dunno |
| 23:54.46 | Qwell | I have a feeling Verizon would rape me for a PRI |
| 23:54.52 | Sedorox | they don't have anything about T1's on their website... |
| 23:54.54 | Qwell | Do I _have_ to go with the local telco? |
| 23:54.55 | Sedorox | lol |
| 23:55.07 | *** part/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 23:55.18 | Sedorox | or host much does a Pri through Telus cost? |
| 23:55.23 | *** join/#asterisk dsmouse (~mouse@rrcs-24-199-146-243.midsouth.biz.rr.com) |
| 23:55.39 | Sedorox | Qwell: wonder if you could do MCI now... since they are the same company.. *crys* |
| 23:55.46 | Qwell | Are they? |
| 23:55.57 | Sedorox | yea... Verizon bought MCI/UUNet |
| 23:55.59 | Qwell | Still, Verizon sucks |
| 23:56.08 | Sedorox | yup |
| 23:56.11 | Sedorox | which sucks |
| 23:56.35 | *** join/#asterisk kimosabe (~kimo@216.60.60.103) |
| 23:57.00 | Qwell | I can just imagine how the call goes. |
| 23:57.04 | Sedorox | http://it.slashdot.org/article.pl?sid=05/02/14/1956216&tid=187&tid=215&tid=218 |
| 23:57.14 | Qwell | "Hello, Verizon." "Yeah, hi...I need a PRI." "A what?" |
| 23:57.20 | Sedorox | lol |
| 23:57.33 | kimosabe | does any one know how i can disable the wan on the 2100 model sipura because if i configure it and once i disconect the router from the wan port i no longer have a dial tone need it to function as a normal 2000 model sipura |
| 23:57.43 | *** join/#asterisk cbachman (~cbachman@129.105.7.250) |
| 23:57.44 | harryvv | Sedorox crap when did verizon by uunet? |
| 23:57.57 | *** join/#asterisk dsfr (~dsfr@216.207.244.183) |
| 23:58.02 | Sedorox | see above link |
| 23:58.07 | harryvv | k |
| 23:58.09 | Sedorox | An anonymous reader submits "Even after a last minute offer from Qwest Communications, MCI board members accepted a less lucrative offer from Verizon to be bought for $6.7 billion in cash, stock and dividends. The acquisition comes after Nextel Communications and Sprint Corp. partnered up in a $35 billion deal and SBC Communications Inc. and AT&T Corp. announced a $16 billion merger plan. So, what's next for the telecom industry?" |
| 23:58.11 | syslod | Qwell:http://www22.verizon.com/regulatory/ |
| 23:58.13 | Qwell | You know its all over when Verizon and the other one merge. |
| 23:58.20 | Qwell | ...why can't I think of the name right now? |
| 23:58.26 | harryvv | man |
| 23:58.29 | harryvv | this is insane |
| 23:58.37 | *** join/#asterisk ACiDV (~joel@122-68-181.dr.cgocable.ca) |
| 23:58.40 | Qwell | syslod: ? |
| 23:58.52 | syslod | A what |
| 23:59.03 | harryvv | btw, my stepdad worked with Craig McCaw when he was just starting the cell biz back in 1978 |
| 23:59.27 | ACiDV | Hi, what is the difference between the S(x) (hangup after x ms) and L(x) (limit the call to x ms) options of Dial cmd ? |
| 23:59.39 | harryvv | he has some interesting stories to say about a service that no one knew what it was back then :) |
| 23:59.41 | Qwell | syslod: Whats that for? Don't feel like reading legaleese right now |
| 23:59.51 | *** join/#asterisk atmel (~vlad@ruxi.dynamic.ucsd.edu) |