00:01.11 | *** join/#asterisk shinao1 (n=shinao1@62.173.48.42) |
00:02.36 | *** join/#asterisk timburke (i=timburke@unaffiliated/timburke) |
00:05.50 | *** join/#asterisk squish102 (n=squish10@cpe-098-024-067-184.carolina.res.rr.com) |
00:06.36 | squish102 | could any1 give me advice on what to use for the following simple home setup? |
00:07.04 | squish102 | i have family in bouth south africa and the US. i live in US and would like to make a skype to voipbuster gateway |
00:10.55 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
00:19.34 | *** join/#asterisk thing1 (n=Dwayne@64.42.227.97) |
00:20.42 | thing1 | hi, i have a cisco 186 ata, which connects to a call manager, i would like to plug phone1 and 2 into my asterisk box is this possible, if so is it fxs or fxo signalling? |
00:20.58 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
00:23.42 | *** join/#asterisk fors1 (n=forsen@pat-tdc.opera.com) |
00:23.51 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
00:35.35 | *** join/#asterisk etm124 (n=edmiller@216.37.164.100) |
00:37.00 | etm124 | hello all. im having bit of trouble making outgoing phone calls. ive set up asterisk servers before, but was using zaptel. |
00:37.11 | etm124 | i 'believe' i have set up my TDM400p card correcty. |
00:37.33 | etm124 | i am getting now a -- Executing [s@macro-trunkdial:2] Dial("SIP/208-08e34c18", "Zap/g1/17174211011") in new stack |
00:37.33 | etm124 | [Nov 9 14:13:03] WARNING[7469]: channel.c:3051 ast_request: No channel type registered for 'Zap' |
00:37.34 | etm124 | error |
00:37.43 | etm124 | can anyone be of assistance on where to start checking? |
00:37.49 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
00:48.45 | lmadsen | etm124: are you using DAHDI now? |
00:49.04 | etm124 | i believe so. |
00:49.13 | beek | etm124: If you're using DAHDI then you'll need to use DAHDI/g1.. |
00:49.17 | lmadsen | etm124: core show modules like dahdi |
00:49.35 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
00:49.49 | lmadsen | etm124: or read the Zaptel-to-DAHDI.txt file in the root of your asterisk source directory for how to make it understand 'Zap' |
00:50.10 | etm124 | ill check it out, thanks lmadsen. |
00:51.38 | etm124 | DAHDI DAHDI Telephony Driver w/PRI no yes no |
00:51.42 | etm124 | looks like i am using DAHDI |
00:51.52 | etm124 | however, Devicestate is 'no'. |
00:53.45 | *** join/#asterisk km2 (n=x@mobile-166-217-117-167.mycingular.net) |
00:54.33 | beek | etm124: You'll probably need to do a dahdi_cfg to configure that card. |
00:55.04 | beek | etm124: Basically anything you did with zaptel_* will need to be done with dahdi_* |
00:55.49 | etm124 | interesting. thank you beek. |
00:56.28 | Tuxguy | Is it possible to use the Vonage ATA for another network? |
00:59.46 | thing1 | is this possible: analog phones ----------- channelbank(FXS) ---------- asterisk ----------- channelbank(FXS) ---------------- Cisco ATA ----------- Callmanager |
01:01.12 | thing1 | want to use call manager for outgiong calls |
01:02.26 | ManxPower | thing1: it will work, but calls won't disconnect properly all the time, maybe never at all |
01:03.09 | thing1 | will the FXS signalling work for the ATA? |
01:03.42 | ManxPower | Another option is Asterisk -> SIP -> Call Manager |
01:03.53 | etm124 | beek: thanks again for your help. i think dahdi is correctly configured. i took your suggestion and changed Zap/g1 to DAHDI/g1, but still get the same error. COuld ayou point me somewhere else to look? |
01:03.59 | ManxPower | or maybe Asterisk -> T-1/E-1 crossover -> Call Manager |
01:04.19 | beek | etm124: You're restarted * and dahdi, right? |
01:04.25 | etm124 | Yessir. |
01:04.27 | ManxPower | or heck Telco -> PRI -> Asterisk -> Call manager |
01:04.45 | beek | what does "dahdi show channels" give you? |
01:05.01 | etm124 | asterisk01*CLI> dahdi show channels |
01:05.01 | etm124 | <PROTECTED> |
01:05.20 | thing1 | i have asterisk -> T1 -> Adtran Channel Bank -> ATA -> Call manager, thats the only way my provider will allow me to connect with thier call manager |
01:06.06 | beek | You should have two files: /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf. Are they present and populated? |
01:06.29 | ManxPower | thing1: ATAs don't provide the correct signal to tell Asterisk or Call Manager the call has been disconnected. |
01:06.49 | ManxPower | Who, exactly, is your provider? |
01:07.12 | ManxPower | In any case, you have my recommendation. Take it or leave it. |
01:07.18 | etm124 | yes they are. system.conf: http://pastebin.com/d58879e6f |
01:08.04 | etm124 | chan_dahdi.conf: http://pastebin.com/d2f1e0f7b |
01:08.18 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-fce1e3c18a4465ac) |
01:09.03 | beek | etm124: Are the modules for that card loaded? |
01:09.53 | etm124 | as far as i know. i did a modprobe wctdm |
01:10.02 | etm124 | is there a command to double check that? |
01:11.55 | thing1 | i run a vonager box no problem |
01:12.21 | beek | etm124: Does "core show channeltypes" show DAHDI? |
01:12.39 | etm124 | yes: DAHDI DAHDI Telephony Driver w/PRI no yes no |
01:12.51 | etm124 | the Devicestate says no, however. |
01:13.46 | beek | How about "dahdi show channel 1" |
01:14.00 | etm124 | asterisk01*CLI> dahdi show channel 1 |
01:14.00 | etm124 | Unable to find given channel 1 |
01:14.02 | etm124 | :( |
01:14.34 | beek | Hmmm... then it's definitely not configured properly yet. I hate to sound microsoftesk, but perhaps a quick reboot may get everything loaded and discovered properly. |
01:14.43 | etm124 | gaaaaasp. a reboot? |
01:14.51 | etm124 | ha, let me give it a whirl. thanks beek. |
01:14.52 | beek | I know... I know. |
01:16.24 | *** join/#asterisk `Sean (i=Shaan@CPE0016d3fe20a0-CM0014045acc3c.cpe.net.cable.rogers.com) |
01:18.00 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.4) |
01:21.34 | etm124 | im afraid the channel was still not found beek. |
01:21.42 | etm124 | asterisk01*CLI> dahdi show channel 1 |
01:21.42 | etm124 | Unable to find given channel 1 |
01:22.14 | beek | etm124: Hmmm... well that's not good. I don't understand -- a reboot always fixes a MS machine. ;-) |
01:22.21 | etm124 | haha. |
01:22.29 | etm124 | im running centos 5 |
01:22.33 | orkid | where as a reboot always efs up a lnux one |
01:22.39 | *** join/#asterisk Bad_Robot- (n=BadRobot@cpe-76-173-219-25.socal.res.rr.com) |
01:22.46 | beek | You did install CentOS5 and turned off SELINUX, right? |
01:23.01 | etm124 | asterisk 1.4.22 |
01:23.30 | beek | orkid: That's because we all start/stop things manually on a Linux box and forget to update the startup. |
01:23.39 | beek | etm124: You turned off SELINUX? |
01:23.46 | etm124 | are you referring to iptables, beek? |
01:23.51 | beek | etm124: No. SELINUX. |
01:25.00 | Katty | lmadsen: do you twitter? |
01:25.07 | beek | do a cat /etc/sysconfig/selinux and look for the line that says: SELINUX=[enforcing|permissive|disabled] |
01:25.08 | lmadsen | Katty: I do... sometimes |
01:25.09 | etm124 | just turned it off |
01:25.25 | lmadsen | Katty: leif_madsen |
01:25.34 | beek | etm124: That will kick you in the privates. |
01:25.47 | etm124 | ha, lets try. |
01:26.09 | Katty | lmadsen: following. you know there's an app that will update your twitter to facebook, right? |
01:26.18 | beek | Just change /etc/sysconfig/selinux to "SELINUX=disabled" so that it won't start again on your next boot. |
01:26.22 | lmadsen | I do... I choose not to use it :) |
01:26.26 | Katty | kk |
01:26.29 | lmadsen | but twitter now loads on my server :)_ |
01:26.35 | lmadsen | s/server/website/ |
01:26.55 | etm124 | hmph. still nothing beek. still getting the same 'Unable to find given channel 1'. |
01:27.21 | beek | etm124: Reboot once again. |
01:27.22 | lmadsen | lspci ? |
01:27.38 | lmadsen | computer sees the card? |
01:27.53 | Katty | i'll card YOU in a minute. |
01:28.09 | lmadsen | sounds kinky |
01:28.23 | etm124 | wow. no signs of my card when lspci |
01:28.48 | lmadsen | might want to reseat it then |
01:31.27 | mankash | what is the reason for this error: |
01:31.28 | Katty | hm. |
01:31.28 | mankash | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
01:31.33 | etm124 | taking a break from this. thanks for your help lmadsen and beek. ill be back on a bit later. |
01:31.44 | etm124 | mankash, asterisk isnt started. |
01:31.55 | etm124 | just type asterisk |
01:31.55 | lmadsen | mankash: means asterisk isn't running |
01:31.58 | beek | etm124: you're welcome. Good luck. |
01:32.06 | *** join/#asterisk BeeBuu (n=beebuu@219.130.254.164) |
01:32.09 | mankash | it is started I can see it running on the console |
01:32.24 | mankash | but from remote if I try to connect it gives me this error |
01:32.32 | mankash | my sip phone is regsitered |
01:33.10 | mankash | I think it is a permission issue |
01:33.56 | BeeBuu | is Set(foo=${ODBC_USER_DATABASE(${EXTEN})}) put result to foo ? |
01:34.04 | lmadsen | yes |
01:35.36 | Katty | i'm hungry. |
01:36.33 | BeeBuu | lmadsen: how can i know the query runing? set verbose ? |
01:38.58 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
01:39.30 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
01:40.03 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
01:43.28 | lmadsen | BeeBuu: with ODBC, it's pretty much obscured and you can't unless you enable logging on your sql server |
01:43.51 | lmadsen | do 'odbc show' to make sure your ODBC connection is connected |
01:44.04 | lmadsen | you should see, "connected: yes" |
01:45.34 | BeeBuu | i just get one line--->DSN: asterisk |
01:46.54 | BeeBuu | lmadsen: what's that mean?is it oK? |
01:47.14 | lmadsen | you should see, "connected: yes" |
01:47.23 | lmadsen | if not, then it is not connected |
01:47.42 | lmadsen | BeeBuu: follow the instructions in the book at www.asteriskdocs.org, I think chapter 12 |
01:47.47 | lmadsen | "database integration" |
01:48.05 | BeeBuu | thanks,lmadsen,reading... |
01:48.56 | orkid | is the online pdf the final version? or has it been proofed afterwards? it has a few spelling mistakes, etc, iirc |
01:49.06 | orkid | thanks for the book btw, it's great :09 |
01:49.08 | orkid | :) |
01:51.26 | [TK]D-Fender | lmadsen: Given they app runs a pretty limited number of commands it'd be great if it could verbose them |
01:51.50 | [TK]D-Fender | lmadsen: Not like its something that would flood CLI disproportionately |
01:53.28 | Katty | ello fender. |
01:53.59 | etm124 | Katty: go eat. i just picked up some dinner. well, reheated :( |
01:55.24 | Katty | etm124: had a bagel. |
01:57.04 | *** join/#asterisk joesuffceren2 (n=Dell@75.143.183.26) |
01:57.54 | joesuffceren2 | I have a cisco 7940 setup as a remote extension (or, that's what I'm trying). It will register, I can use it to call extensions connected to my asterisk server, and I can call it from the pstn via it's did, but I cannot use it to place outgoing calls to the pstn. This pastebin shows the CLI output (verbosity 10) when I try to place a call: http://pastebin.com/m7707bea2 |
01:57.54 | joesuffceren2 | incidentally, connecting to the extension via xlite produces the exact same behaviour, so I don't think it's a cisco 7940 thing |
01:58.30 | ManxPower | joesuffceren2: Sorry, I can't help with GUI setups. |
01:58.39 | ManxPower | ~freepbx |
01:58.39 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
02:00.24 | joesuffceren2 | I asked in freepbx, and I understand you're not wanting to dive into their dialplans, etc., but this isn't a freepbx specific issue... |
02:00.26 | joesuffceren2 | *sigh* |
02:00.57 | ManxPower | joesuffceren2: Tell ya what. If you remove all the freepbx crap form your dialplan for whatever number you are dialing then I'll take a look at it. |
02:01.03 | lmadsen | [TK]D-Fender: possibly... but seeing asterisk try and execute the commands isn't very useful if it isn't getting to the server. Basically if ODBC is connected via res_odbc.conf, and 'odbc show' shows connected, everything else can easily be debugged from outside of asterisk (and most likely should) |
02:01.13 | ManxPower | BTW, why are you using something you can't even get support for? |
02:02.27 | ManxPower | You should simply have a Dial, and for troubleshooting a Noop(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is ${DIALSTATUS}) as the priority after the Dial line. |
02:03.16 | ManxPower | That should give less than 5 lines of output. Far better than the EIGHTY NINE lines of output the FreePBX crap has. |
02:04.42 | [TK]D-Fender | lmadsen: True but it'd be more than easy to provider that "what * parses" bit |
02:05.26 | ManxPower | Looks like it will be about 9 lines of output. |
02:05.30 | [TK]D-Fender | joesuffceren2>I asked in freepbx, and I understand you're not wanting to dive into their dialplans, etc., but this isn't a freepbx specific issue... <- Yes, it IS> If you can dial some things but not others then it IS dialplan and FreePBX's problem |
02:05.43 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
02:06.19 | ManxPower | Of course it's a FreePBX problem. I just offered to help him troubleshoot it without FreePBX. Then he stopped talking. |
02:07.09 | joesuffceren2 | even though that happens only with my one extension that's offsite and not with the 9 that are, it's a diaplan issue and not a sip header issue? |
02:07.16 | drmessano | joesuffceren2: You have x number of extensions working, one not.. with 2 different SIP clients on that extension |
02:07.32 | drmessano | Your extension is screwed, and there's no way to troubleshoot it |
02:07.39 | drmessano | Delete, recreate, and move on |
02:08.03 | joesuffceren2 | drmessano, I am in the process of trying that, but the one extension that isn't working is outside the firewall, while all the others are inside |
02:08.13 | ManxPower | funny thing is, the pastebin he pasted indicates the call worked just fine. |
02:08.16 | *** join/#asterisk terracon (n=garry@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
02:08.25 | drmessano | So it's a NAT audio problem |
02:08.29 | drmessano | ~sipnat |
02:08.30 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
02:08.30 | joesuffceren2 | yeah, it seems to, but it just gives me dialtone again |
02:08.41 | joesuffceren2 | the call doesn't actually place (because I called my cell and it never rang) |
02:08.44 | ManxPower | dialtone is generated by the local phone |
02:08.46 | ManxPower | not Asterisk |
02:08.52 | mankash | why when I do ps aux | grep asterisk show me too many lines for asterisk |
02:09.09 | ManxPower | Â Â -- Called 1/8005551212 Â Â -- Zap/1-1 is proceeding passing it to SIP/113-00b62d10 Â Â -- Zap/1-1 answered SIP/113-00b62d10 Â Â -- Hungup 'Zap/1-1' |
02:09.11 | joesuffceren2 | right, but my point is when I call a local extension, I get ringing and the vm of the extension that I call |
02:09.24 | ManxPower | This is a VALID and WORKING call. Now you are saying the call doesn't even get placed. |
02:09.36 | joesuffceren2 | my cell phone never rings |
02:09.37 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
02:09.41 | joesuffceren2 | I replaced my cell number with 5551212 |
02:09.58 | ManxPower | joesuffceren2: Your cell phone number is 8005551212? |
02:10.04 | ManxPower | don't do that. |
02:10.14 | *** join/#asterisk prodyan (n=ian@124.104.71.66) |
02:10.33 | ManxPower | so we really don't know if it dialed a 1 before the area code or it did not dial a one before the area code. |
02:10.41 | drmessano | Your cell never rings because the calls hangs up before it makes it to the cell |
02:11.03 | joesuffceren2 | sorry if that offends you, manxpower. I'm new to IRC and I wasn't sure about the safety of displaying my personal phone number to 260 people I've never met before. forgive me if I'm paraniod |
02:11.20 | joesuffceren2 | paranoid* |
02:11.22 | drmessano | Ok, i've already said twice how to fix it |
02:11.25 | drmessano | Let him keep pasting |
02:11.30 | ManxPower | drmessano: It could be a reinvite problem or a localnet/externip problem, or a "dial before telco is ready problem" |
02:12.12 | ManxPower | My bet is localnet/externip, but you already gave him the link to fix that. |
02:12.25 | drmessano | Yeah, I think thats the solution |
02:12.39 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
02:12.54 | ManxPower | joesuffceren2: in the future, if you are concerned replace JUST the last 4 digits of the number. Generally that info is not important. |
02:13.11 | ManxPower | when pasting config files mask ONLY the passwords. |
02:13.27 | joesuffceren2 | will do. do you prefer fake numbers for those last 4 or xxxx? |
02:13.45 | ManxPower | drmessano: I wonder if he can figure how to adapt the "generic asterisk" instrucitons into "works in freePBX" |
02:14.00 | ManxPower | joesuffceren2: xxxx is best as we know it's fake. |
02:14.12 | *** join/#asterisk nikko (n=nikko@12-218-250-137.client.mchsi.com) |
02:14.12 | etm124 | welp, found out that my card was listed when i did an lspci, just nothing i've heard of. Was recognized as Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
02:14.36 | BeeBuu | i passed follow command: echo "select 1" | isql -v asterisk-connector, but still get odbc connect,why? |
02:14.37 | ManxPower | etm124: that just means you have an older version of the pci vendor id library for your distro |
02:14.43 | etm124 | i tried calling the number in which the phone line is registered, and im not seeing anything in the cli |
02:15.01 | etm124 | thanks ManxPower. i am used to seeing it as TDM Wildcard... |
02:15.25 | ManxPower | etm124: upgrade your libpci and you might see it again |
02:16.17 | etm124 | when u say upgrade, do you mean to a newer version? |
02:16.25 | ManxPower | etm124: correct. |
02:16.32 | joesuffceren2 | drmessano, the contents of my sip_nat.conf are: |
02:16.44 | joesuffceren2 | externip=66.83.116.154 |
02:16.46 | joesuffceren2 | localnet=10.0.0.0/255.255.255.0 |
02:16.52 | ManxPower | joesuffceren2: We don't use sip_nat.conf. |
02:17.55 | etm124 | ManxPower: im running 1.4.7 |
02:17.55 | etm124 | i believe that's the latest. |
02:17.55 | ManxPower | etm124: This is an OPERATING SYSTEM issue. |
02:18.01 | etm124 | my mistake. |
02:18.06 | joesuffceren2 | and deleting and recreating the extension didn't change the situation unfortunately |
02:18.35 | ManxPower | joesuffceren2: I'm sure it didn't. Now are you going to edit extensions.conf so we can do it the old fashioned way? |
02:18.43 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
02:19.21 | ManxPower | joesuffceren2: how many ports did you forward in your router? |
02:19.32 | ManxPower | well, what ports and what protocols? |
02:29.08 | *** join/#asterisk joesuffceren2 (n=Dell@75.143.183.26) |
02:29.18 | *** join/#asterisk rcahilig (n=sysad@202.78.75.246) |
02:29.22 | *** join/#asterisk lolipops (n=lolipops@modemcable238.118-82-70.mc.videotron.ca) |
02:30.05 | joesuffceren2 | drmessano and manxpower, thanks for the help. I'll keep working on it. I won't bother you any more. Have a good night |
02:32.33 | lolipops | im trying to wait for user input with Background();, but it seems that it ignores the response timeout and just exists as soon as the message is done playing. any ideas? |
02:32.42 | lolipops | exits* |
02:34.27 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
02:34.47 | [TK]D-Fender | lolipops: "autofallthrough=no" <- should be under [genera] in extensions.conf |
02:34.53 | [TK]D-Fender | lolipops: "autofallthrough=no" <- should be under [general] in extensions.conf |
02:35.21 | lolipops | ill try that. thanks. |
02:38.15 | squish102 | is there any free skype -> asterisk gateways? i only need one skype extension |
02:39.34 | drmessano | no |
02:39.56 | Mark_Logan | I second that "no". |
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02:41.09 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
02:41.09 | squish102 | thanks |
02:42.35 | *** join/#asterisk sasargen (n=chatzill@72.58.152.132) |
02:43.34 | *** join/#asterisk [gnubie] (n=bintut@cm61.omega118.maxonline.com.sg) |
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02:46.19 | *** mode/#asterisk [+o d3wayne] by ChanServ |
02:48.28 | Mark_Logan | Like "Thanks!" thanks or "Thanks for nothing" thanks? :P |
02:50.40 | squish102 | lol, no, like thanks that someone just gave me an answer |
02:51.02 | Katty | jbot: hi |
02:51.03 | jbot | hello, katty |
02:51.27 | Katty | [TK]D-Fender: have you ever had a tuna melt> |
02:51.49 | [TK]D-Fender | Katty: Yup, oven baked open faced goodness |
02:52.22 | Katty | oh? never tried the oven baked variety. |
02:52.24 | Katty | do you have a recipe? |
02:52.31 | lolipops | [TK]D-Fender, i guess ill fix my little problem with silence/10.wav. easier and cleaner. |
02:55.37 | [TK]D-Fender | Katty: Basiacally just a tuna-salad sandwich, open faced... maybe a slice of cheese. |
02:55.54 | RB2 | Has anyone here used lylix to run a hosted asterisk instance? |
02:56.58 | *** join/#asterisk rcahilig (n=sysad@202.78.75.246) |
02:57.49 | Katty | [TK]D-Fender: what kind of bread to you use? |
02:57.51 | Katty | [TK]D-Fender: broil? |
03:02.05 | lolipops | [TK]D-Fender, why is it TIMEOUT(response) seems to apply to Read(); but not Background();? |
03:02.35 | [TK]D-Fender | Katty: For bread we've got a Quebec made brand whose 12 grain variety is un-fucking-believeably good. Having visited around I've never seen anything like it anywhere in standard bagged form |
03:02.46 | [TK]D-Fender | lolipops: You are mistaken |
03:03.01 | Katty | [TK]D-Fender: pity. hard to get good bread around here. |
03:04.15 | Katty | maybe i should start making my own mini loaves |
03:04.17 | lolipops | [TK]D-Fender, i have an extension here that uses Read(); and it waits the whole default 10 seconds, this other one with Background(); falls out as soon as it's done playing |
03:05.13 | [TK]D-Fender | lolipops: Yes and I told you what to add and you aren';t showing me the problem. |
03:05.21 | [TK]D-Fender | lolipops: I know full well how timeouts work. |
03:06.05 | lolipops | calm down. im just pointing out something that seems illogical to me here. |
03:06.39 | Katty | fender is always calm. |
03:06.41 | [TK]D-Fender | lolipops: PB <- |
03:06.48 | Katty | except when he chops a digit nearly off. |
03:07.13 | [TK]D-Fender | Katty: thats right, and if 1 more person slanders me like that I'LL KILL THEM! |
03:07.22 | [TK]D-Fender | :D |
03:07.30 | *** join/#asterisk BeeBuu (n=beebuu@125.95.28.194) |
03:07.40 | Katty | pats [TK]D-Fender |
03:07.43 | Katty | you do that dear. |
03:08.00 | Mark_Logan | Jeez, we're having a real emotional crash going on here. |
03:08.05 | BeeBuu | where can i set the transfer press key time? |
03:08.27 | Mark_Logan | maybe next time lets go with "asterisk -rvv" and not "asterisk -rvvvvvvvvvvvv" alright? |
03:10.18 | [TK]D-Fender | Katty: On the thumb topic I need to see a doc about having that bit removed entirely. the muscle bit that was left there curled into a bit of a knot and it useless, and a little painful under pressure. Its like it not even "me" |
03:10.49 | Katty | [TK]D-Fender: sounds icky. |
03:11.10 | etm124 | ouch. |
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03:11.48 | [TK]D-Fender | Katty: if they can do it with confidence of success I have NO issue with it. |
03:12.07 | [TK]D-Fender | Katty: its a quality of life issue for me. |
03:12.16 | Katty | you're also not a hypochondriac |
03:12.22 | Katty | iw ould have already been to the doctor half a dozen times ;) |
03:12.26 | Katty | OMG IT"S CANCER |
03:12.27 | Katty | A TUMOR |
03:12.30 | Katty | etc. |
03:13.26 | [TK]D-Fender | Katty: No, its a "healed-over" lump of useless muscle thats F-ing up my thumb! |
03:13.27 | lolipops | http://pastebin.com/d37c04282 |
03:13.51 | Katty | [TK]D-Fender: indeed. |
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03:15.42 | [TK]D-Fender | lolipops: failed call please |
03:15.54 | lolipops | im sorry? |
03:16.21 | Katty | [TK]D-Fender: also, pork chops. |
03:16.39 | Katty | [TK]D-Fender: with chicken broth, honey, soy sauce, ketchup, ginger, and a lil garlic. |
03:16.54 | [TK]D-Fender | lolipops: Show me the failed call |
03:17.10 | `Sean | :o, TASTES yummy |
03:17.26 | Katty | `Sean: oh yes, yes it does. |
03:17.40 | Katty | `Sean: especially when you simmer the pork chops alllll day long. |
03:17.56 | Katty | `Sean: with some rice and veggies, and cornstarch with the leftover sauce. |
03:18.20 | etm124 | is getting hungry now. |
03:19.48 | Katty | etm124: http://bp1.blogger.com/_WqvyAw872Ko/RnGWugLUiOI/AAAAAAAAAXM/A4UL6qTxzkA/s320/PorkChopsYumYum.jpg |
03:20.22 | Katty | etm124: tell me that does not look good, after a long day of work and cold weather. |
03:20.30 | etm124 | mmmmmmmmm. |
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03:21.30 | lolipops | [TK]D-Fender, i feel retarded, but i dont know what you want to see exactly. |
03:22.37 | lolipops | [TK]D-Fender, autofallthrough=no DOES work, but it just strikes me as odd that with autofallthrough=yes, Read(); waits 10 seconds but background returns immediately. |
03:24.02 | [TK]D-Fender | lolipops: because autofallthrough says what happens at the end of an EXTEN <- |
03:24.19 | [TK]D-Fender | lolipops: When you run out of priorities then you fall through |
03:24.37 | [TK]D-Fender | lolipops: Which would be blatantly visible in the CLI of the call. |
03:24.38 | lolipops | the same behavior applies if I put a Playback(vm-goodbye); after Background(); |
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03:29.55 | [TK]D-Fender | lolipops: Do not put playbacks after backgrounds. |
03:30.03 | lolipops | why not? |
03:33.07 | [TK]D-Fender | lolipops: because it devalidates * waiting at the end. |
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03:41.08 | prodyan | guys, how do i get the parameter of an agi script call like this AGI(test.agi|2222) - i want to get 2222 |
03:41.42 | [TK]D-Fender | prodyan: Go read the docs for whatever language you wrote your script in |
03:42.01 | prodyan | hmm oki let me check |
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03:42.30 | prodyan | thanks btw D-Fender |
03:42.33 | lolipops | [TK]D-Fender, i guess i got myself an okay setup for this now. thanks for all. |
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03:49.30 | etm124 | ok. ive been fiddling with my setup for a bit now. it seems like my channels aren't being recognized... |
03:49.53 | etm124 | DAHDI Tools Version - 2.0.0 |
03:49.53 | etm124 | DAHDI Version: 2.0.0 |
03:49.53 | etm124 | Echo Canceller(s): MG2 |
03:49.53 | etm124 | Configuration |
03:49.53 | etm124 | ====================== |
03:49.54 | etm124 | Channel map: |
03:49.56 | etm124 | Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) |
03:49.58 | etm124 | Channel 02: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02) |
03:50.00 | etm124 | Channel 03: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 03) |
03:50.02 | etm124 | Channel 04: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04) |
03:50.04 | etm124 | 4 channels to configure. |
03:50.06 | etm124 | Setting echocan for channel 1 to mg2 |
03:50.08 | etm124 | Setting echocan for channel 2 to mg2 |
03:50.10 | etm124 | Setting echocan for channel 3 to mg2 |
03:50.12 | etm124 | Setting echocan for channel 4 to mg2 |
03:50.16 | drmessano | STOP |
03:50.24 | etm124 | sorry for the big paste. |
03:50.30 | drmessano | Dont do it |
03:50.31 | drmessano | Ever |
03:50.45 | drmessano | ~pb |
03:50.46 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
03:51.17 | etm124 | i am, however, getting an unable to find given channel 1 error. |
03:51.56 | drmessano | .... |
03:52.11 | [TK]D-Fender | etm124: You didn't show us any error |
03:52.54 | etm124 | [TK]D-Fender: Unable to create channel of type 'DAHDI' |
03:53.12 | [TK]D-Fender | etm124: Did you compile * AFTER you installed DAHDI? |
03:53.27 | [TK]D-Fender | etm124: and please pastebin the entire failed call attempt |
03:54.27 | etm124 | [TK]D-Fender: yes, i did. and http://pastebin.com/d6a16e3bd |
03:54.54 | [TK]D-Fender | etm124: And now your configs.... |
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03:57.11 | Katty | hai cunning |
03:57.44 | CunningPike_ | Katty: Hai! |
03:57.44 | etm124 | [TK]D-Fender: chan_dahdi.conf: http://pastebin.com/d2f1e0f7b |
03:57.56 | CunningPike_ | How are you? |
03:58.46 | [TK]D-Fender | etm124: "dahdi show channels" |
03:59.19 | etm124 | [TK]D-Fender: Chan Extension Context Language MOH Interpret |
03:59.26 | etm124 | no other info from that. |
03:59.43 | [TK]D-Fender | etm124: http://pastebin.com/d2f1e0f7b <- line 7. You have NOT set the group you are tryig to dial |
04:00.00 | [TK]D-Fender | etm124: 1st slear error. next, try to reload chan_dahdi after fixing it. |
04:00.03 | [TK]D-Fender | clear* |
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04:01.47 | etm124 | [TK]D-Fender: would that be a reason my channels aren't showing up? |
04:01.59 | [TK]D-Fender | etm124: 2 separate issues |
04:02.41 | [TK]D-Fender | etm124: next if the fact that your spam above lists *4* channels, and here I see you trying to configure *8* |
04:02.58 | [TK]D-Fender | etm124: Someone needs their head screwed on straight |
04:08.21 | etm124 | [TK]D-Fender: your two suggestions fixed it. |
04:08.22 | etm124 | thank you. |
04:08.33 | [TK]D-Fender | etm124: You're welcome |
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04:45.35 | *** join/#asterisk wnspark (i=0cd8f6bd@gateway/web/ajax/mibbit.com/x-6da09c6c410e8907) |
04:46.00 | wnspark | What directory is asterisk installed into when you install it via the ports collection in FreeBSD? |
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04:54.24 | postel | wnspark: which asterisk or use the ports command (man ports) to query the system for where it put the files |
04:54.57 | postel | wnspark: most things go under /usr/local |
04:55.27 | [TK]D-Fender | usuallt /usr/local/etc/asterisk |
04:55.36 | [TK]D-Fender | the config anyways |
04:55.48 | wnspark | i found it |
04:55.49 | [TK]D-Fender | "which asterisk" should answer the other. |
05:12.19 | thing1 | how can i do an autoanswer on a zap channel |
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05:13.58 | thing1 | i have a ata hooked into a channel box, now i would like to be able to dial 9 to get that line and it just keeps ringing |
05:18.00 | [TK]D-Fender | thing1: What is a "channel box"? And what do you mean "autoanswer"? Describe the call from beginning to end |
05:18.52 | thing1 | i want to dial an extension, like 9 and asterisk will bridge me with the line i have the ata plugged into, zap/48 right now i have dial zap/48, and it rings, but never picks up |
05:19.08 | thing1 | i assume i have to have asterisk pickup the call or something |
05:19.31 | thing1 | i have a 24 port adtran channel bank hooked into asterisk via t1 |
05:19.51 | [TK]D-Fender | thing1: Please start from the BEGINNING |
05:20.24 | [TK]D-Fender | thing1: don't just say "I dial". Start from the description of exactly what phone plugged into what device, calling on what channel. |
05:20.36 | thing1 | i ahve a cisco ata which connects to a remote callmanage, i would like to be able to use the two lines for my pbx |
05:21.28 | [TK]D-Fender | thing1: in your zapata.conf or chan_dahdi.conf you specify the context the channels will send incoming calls to. |
05:21.28 | thing1 | i've connected the phone one and two into zap ports and would like to dial an extension like 9 and have asterisk give me a dial tone |
05:21.43 | thing1 | how about outgoing calls? |
05:22.05 | [TK]D-Fender | thing1: lets be clear on this since you have failed again. |
05:22.17 | [TK]D-Fender | thing1: what is the phone you are holding in your hand plugged into? |
05:22.35 | thing1 | its a sip phone plugged into the network |
05:22.46 | [TK]D-Fender | ok, SIP phone direct to *, correct? |
05:22.52 | thing1 | yes |
05:23.51 | [TK]D-Fender | thing1: Good. then make an exten => 9,1,Dial(Zap/1) for example to pull that zaptel channel which should pull you dialtone from the ATA attached to it |
05:24.24 | [TK]D-Fender | thing1: What is the ATA using to talk to the CM? |
05:24.43 | [TK]D-Fender | thing1: because this is a lot of A>D>A for nothing... |
05:24.52 | [TK]D-Fender | thina fugly setup for sure... |
05:25.02 | thing1 | it's used skinny |
05:25.05 | thing1 | using |
05:25.21 | [TK]D-Fender | thing1: You have no SIP license available? |
05:25.36 | [TK]D-Fender | thing1: It'd be far better if you could jsut send the call direct from * to CM |
05:26.06 | thing1 | well the idiot on the other end is making me do it with a ata, i could do it straight with astierks i know but some head-techs are propietery idiots |
05:26.19 | thing1 | i know |
05:26.48 | [TK]D-Fender | thing1: Its doable, but its just a stupid expense and loss of functionailty, etc |
05:27.05 | [TK]D-Fender | thing1: Pass it on that others "in the know" agree that they are morons. |
05:27.07 | thing1 | right now when i dial the zap channel the ata is plugged into it just rings |
05:27.33 | thing1 | i have an fxs signal card |
05:27.54 | thing1 | i also have 4 telco lines coming in on fxo lines and they work perfectly |
05:28.07 | thing1 | but this is diff because it has diff signalling |
05:28.50 | [TK]D-Fender | thing1: you need an FXO signal card, not theother way around |
05:29.19 | thing1 | what is the one for telco, i always get mixed up |
05:29.27 | [TK]D-Fender | thing1: Your ATA acts as FXO. You therefore need an FXO card for your adtran, not an FXS. |
05:29.53 | thing1 | ok |
05:29.58 | thing1 | i guess thats the problem |
05:30.03 | [TK]D-Fender | thing1: Adtran FXO port = act like a phone (uses fxs_ls signalling in Zapata) |
05:30.15 | [TK]D-Fender | thing1: Great so retarded AND backwards |
05:30.18 | [TK]D-Fender | :) |
05:31.02 | thing1 | thanks |
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05:49.35 | orkid | ubuntu or centos? |
05:49.37 | orkid | :) |
05:49.42 | orkid | or debina :O |
05:49.44 | orkid | debian |
05:50.39 | orkid | ? |
05:52.03 | [TK]D-Fender | orkid: Thoughts your coherent very aren't much. |
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05:59.11 | drmessano | [TK]D-Fender: He is amusing, despite being useless |
06:00.52 | [TK]D-Fender | drmessano: Talks doe funny Yoda hmmmmmmMMMMMM!??!?!??! |
06:00.53 | troy- | drmessano, i concur |
06:01.04 | [TK]D-Fender | does* |
06:02.43 | troy- | [TK]D-Fender, it would seem the only way i'm getting inbound SMS is with a USB GSM Radio |
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06:05.30 | orkid | [TK]D-Fender: what distribution of linux would you recommend for asterisk. centos, debian, ubuntu server ? |
06:05.32 | [TK]D-Fender | troy-: Which still seems to have nothing to do with *. |
06:05.36 | orkid | a basic asterisk install |
06:05.40 | orkid | not business |
06:05.41 | orkid | just home |
06:05.49 | [TK]D-Fender | orkid: Whichever you are capable of using to fulfil *'s requirements with |
06:06.08 | orkid | all this talk of 'realtime' and 'transcoding' makes me scared |
06:06.11 | orkid | :L |
06:07.34 | [TK]D-Fender | orkid: Almost everybody does transcoding. Its always a question of what to what, and how. |
06:07.43 | orkid | it seems like centos is preferred in these circles? i mean, thats what pbxin a flash uses right? |
06:08.00 | drmessano | troy-: chan_mobile? |
06:08.00 | [TK]D-Fender | orkid: And realtime... you don't need to care about, and more than half of those that use it probably shouldn't |
06:08.07 | orkid | [TK]D-Fender: really? i would think many people stay ulaw only |
06:08.17 | orkid | [TK]D-Fender: at least for the call portion, not the ivr sounds |
06:08.22 | [TK]D-Fender | orkid: Who cares what PBIAF uses. Look at what YOU can manage |
06:08.23 | troy- | drmessano, yeah |
06:08.42 | drmessano | Why do you need a USB GSM radio? |
06:08.54 | drmessano | Proper Bluetooth phone should do it |
06:09.34 | troy- | drmessano, much less reliable |
06:09.48 | drmessano | How so? |
06:11.46 | [TK]D-Fender | orkid: Any sound not in the codec of your call is transcoded. |
06:11.52 | [TK]D-Fender | orkid: the load is the same |
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06:12.19 | [TK]D-Fender | orkid: Naturally its best if you aren't transcoding |
06:12.49 | [TK]D-Fender | ok, checkout tie. Later all |
06:12.52 | [TK]D-Fender | time* |
06:12.56 | orkid | bye |
06:13.01 | drmessano | Who the hell wants to use ULAW for everything? |
06:13.30 | troy- | bluetooth is largely irrelevant - you are relying on a consumer phone for transmit/receive versus a purpose built radio chipset |
06:14.36 | drmessano | Same device, metal box |
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06:21.26 | orkid | people with packet loss? |
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06:22.47 | CrazyTux | Does anyone here know how AMI works with using Originate and queues? |
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06:35.17 | ManxPower | drmessano: um, we use ulaw for almost everything. |
06:37.27 | denon | as do we |
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06:38.31 | CrazyTux | Hey guys, sorry on a real flaky connection, so any responses I did not see. I was asking if anyone knows a way to view / access the AMI originate CALL queue, when sending numerous originate calls |
06:40.37 | Micc | what is AMI? |
06:41.36 | Micc | If your talking about the api that astman uses, I had to modify the source to send variables with each event. |
06:41.50 | Micc | So I could know what originate went with what newexten |
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06:43.36 | Micc | CrazyTux, I thought that was fixed in 1.4, it should send the unique id with each channel. |
06:44.02 | CrazyTux | Micc unique ID of the channel? |
06:44.22 | Micc | no, its just a unique id you can use for each originate I think. |
06:44.24 | drmessano | AMI is the asterisk management interface |
06:44.36 | Micc | Or set a variable that gets passed back to you in each channel. |
06:44.50 | Micc | With the originate I think you send a setvar too. |
06:45.13 | Micc | Then it passes that var back with the events. |
06:45.53 | Micc | CrazyTux, I haven't done anything with that for more than a year. I'd have to look into that to remember exactly what I did. |
06:46.21 | Micc | There is a way, you might have to modify the asterisk source, but there is a way. |
06:47.12 | CrazyTux | I'm thinking from more of a global view, so I can see it all, by issuing a command possibly? core show channels perhaps, but that is not 100% reliable |
06:47.24 | CrazyTux | without keeping consistent state/information on each channe |
06:47.58 | Micc | oh, then I don't know. |
06:48.11 | Micc | CrazyTux, you'll have to make your own AMI function. |
06:51.33 | drmessano | http://www.voip-info.org/wiki/view/AstManProxy |
06:51.35 | drmessano | Try that |
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08:19.55 | phpboy | hey all |
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09:04.08 | pif | hi, does 1.4.22 fix some 100% CPU peaks + random crashes that 1.4.21.2 has ? |
09:05.20 | mort_gib | pif: I use 1.4.21.X and have no such issues..... |
09:05.51 | aliraja | Hi all, i want to hangup all calls after 5 minutes coversation b/w customer and agent ...any idea of how i call achieve it |
09:07.29 | aliraja | *can achieve it. |
09:08.18 | pif | mort_gib: debian? |
09:08.18 | tzafrir_laptop | pif, reproducable? |
09:08.36 | pif | oh hi tzafrir_laptop I just sent you a query about 1.4.22 debs |
09:09.03 | mort_gib | pif: both Debian and CentOS |
09:09.11 | pif | not really reproducable yet, but very annoying (production server upgraded last week) |
09:09.13 | mort_gib | But compile from source |
09:09.43 | pif | mort: amd64 arch ? |
09:11.27 | mort_gib | pif: No Xeons |
09:13.50 | pif | tzafrir_laptop: our two newly installed debian/sid dell servers with TE410P card have the same symptoms |
09:14.29 | tzafrir_laptop | pif, looking into this |
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09:14.44 | tzafrir_laptop | (well, at least for something that builds) |
09:20.28 | pif | thanks! |
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09:25.12 | CrazyTux | Does anyone know if AMI sends back event/action data to the same socket that created the action, or does it broadcast it to all sockets reading? |
09:28.06 | creativx | CrazyTux: AMI broadcasts it.. doesnt matter where it came from |
09:28.55 | CrazyTux | creativx: are you sure about that? |
09:29.24 | CrazyTux | creativx: I'm listening on another socket, aside from the invoking client request, and not getting anything through |
09:29.37 | creativx | CrazyTux: mine does in 1.2.. sending "originate" events from one client, amiproxy reads ami, and another client reads the amiproxy output.. |
09:30.43 | CrazyTux | creativx: this is 1.4.21.2 so I'd imagine hmmm |
09:31.56 | creativx | but |
09:32.00 | creativx | when i think about it |
09:32.23 | creativx | well.. ive forgotten |
09:32.27 | creativx | and dont have time to check source atm :) |
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10:19.42 | masus | anyone know where i can buy an opensource predicitive dialer script ? |
10:19.56 | masus | or recommend one ? |
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10:33.05 | kippi | hey |
10:33.27 | kippi | has anyone got a SoundStation IP 6000 working with openser/asterisk? |
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11:01.47 | viraptor | is there some way to get sip "From:" domain in AGI? |
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11:12.35 | beniwtv | Hi all... I've got an AGI script in which I execute a DIAL command. This works fine, and I would like to execute a DeadAGI when the channel is hung up, so I put a h extension in my dialplan. However, once my AGI script has finished sending the DIAL command to Asterisk, it finishes executing. That somehow causes Asterisk to execute the h channel inmediaely, when the channel is not really hung up. Do I have to wait in the AGI scr |
11:12.36 | beniwtv | ipt for the call to finish? Is there any signal Asterisk sends? |
11:12.59 | beniwtv | (sorry for the long explanation) |
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11:41.25 | pif | tzafrir_laptop: I'm trying to debuild 1.4.22 (removed all patches) and it fails with "dpkg-source: error: cannot represent change to asterisk-1.4.22/doc/lang/hebrew.ods: binary file contents changed" |
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11:41.42 | scruz | good day |
11:42.18 | scruz | i'm an asterisk newbie...while my copy of the book is downloading, how can i turn off debug mode in asterisk? |
11:42.59 | viraptor | scruz: core set debug 0 |
11:43.07 | viraptor | or remove debug from your logger.conf |
11:44.08 | phpboy | [Nov 10 13:43:34] WARNING[30475]: res_agi.c:2129 deadagi_exec: Running DeadAGI on a live channel will cause problems, please use AGI |
11:44.21 | phpboy | What kind of problems would this cause? |
11:45.27 | scruz | thanks...set debug 0 worked |
11:45.42 | scruz | i think it's an old version :) |
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11:48.39 | scruz | or it didn't |
11:58.19 | phpboy | <PROTECTED> |
11:58.24 | phpboy | that does not look right |
11:58.25 | phpboy | the time :( |
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12:10.11 | ifnotwhynot | hi there any pri span experts active? |
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12:20.03 | plantseeker | How all |
12:20.09 | plantseeker | Hi all |
12:21.09 | plantseeker | can anyone recommend a good site to learn more about call recording. |
12:22.28 | plantseeker | I am particular interested to find out more about both identifying and dealing with recordings that are out of sync |
12:22.32 | Meaw | google can recommend you one |
12:22.52 | phpboy | ifnotwhynot: what do you wanna know? |
12:23.31 | plantseeker | tried asterisk calling recordings out of sync |
12:23.49 | phpboy | plantseeker: what do you mean? |
12:23.54 | phpboy | 'out of sync'? |
12:24.36 | plantseeker | 2 legs joined but out out of sync |
12:24.50 | plantseeker | 2nd out =of |
12:26.41 | plantseeker | I have 39666 reocrdings to check |
12:27.03 | phpboy | I'm still not with you on the out of sync part |
12:27.33 | Meaw | our billing system records the call.. I just simple click to listen |
12:27.34 | plantseeker | ok phpboy I will try to explain what I mean |
12:27.43 | Meaw | but it's too evil to record a 39666 calls |
12:27.59 | Meaw | we record only when a customer complain about bad signal :) |
12:28.10 | plantseeker | Meaw lol |
12:28.25 | plantseeker | I have listen to a recording |
12:28.48 | plantseeker | although everything ok at the beginning |
12:29.23 | plantseeker | as the call progresses the two callers start talking at the same time |
12:29.47 | plantseeker | out of sync |
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12:32.37 | plantseeker | my question is how do I fix an out of sync recording and what is the best way to identify an out of sync recording. I obviously don't want to listen to 39666 recordings |
12:34.36 | plantseeker | I don't expect to be be given the answers just some pointer to the solutions. some good google search terms would be nice :-) |
12:34.50 | plantseeker | pointer=pointers |
12:35.16 | plantseeker | perhaps a touch typing tutorial as well :-) |
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12:46.39 | ultrav1olet | I have two questions: if asterisk supports that, and if there's software which can do that |
12:47.29 | HeMan | it can do that with some subsets of that |
12:47.33 | tzafrir_laptop | pif, nice . so we need to find why that file changes at build time |
12:47.54 | ultrav1olet | that is: imagine such a situation: someone calls into your office, with a call automatically redirected to a IP phone or SIP/IAX2 account, there's a conversation, then I would like to transfer that call to another person. |
12:48.22 | ultrav1olet | I tried playing with Zoiper and I couldn't do that |
12:59.03 | magronez | is away: cliente |
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13:05.52 | ifnotwhynot | hi there waht is the best way to reload zap drivers once one make changes to the zaptel.conf? |
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13:06.52 | ifnotwhynot | phpboy : sorry had to run are u still available? |
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13:08.14 | hi365 | anyone ever get this? |
13:08.14 | hi365 | app_queue.c: The device state of this queue member, Local/230@from-internal/n, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings |
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13:16.58 | clintc | plantseeker: does this sound like your problem: http://bugs.digium.com/view.php?id=12837 |
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13:18.13 | *** mode/#asterisk [+o lmadsen] by ChanServ |
13:18.32 | clintc | plantseeker: here is a bug we have filed with digium: http://bugs.digium.com/view.php?id=13745 |
13:20.36 | plantseeker | Thanks clintc . I will look at both the links provided. Appreciate it. |
13:21.10 | clintc | plantseeker: one the recording has been done, I don't think you can fix it as both sides have been mixed into a mono signal |
13:22.19 | clintc | plantseeker: a workaround we are thinking about until this is fixed is to use the monitor app instead of mixmonitor... monitor can put each side in it's own left right channel... then you could fix after the fact |
13:22.34 | phpboy | I really wish I could figure out why this silly asterisk install is dropping zap calls every now and then :( |
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13:26.58 | hi365 | i think its a 'feature' |
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13:27.09 | astrOdz | hey |
13:27.15 | astrOdz | has anyone tried asterisk with amazon's ec2? |
13:28.26 | hi365 | there have been reports and how-to's. Want more info? Ive got a friend that know how to do it. His name is... (hint: it begins with a big blue G) |
13:28.42 | lmadsen | anyone know how I can check to see why asterisk is blocking and won't return any data from the CLI when I type in a command? |
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13:30.12 | lmadsen | hrmm... interesting, other commands will come back as long as I don't run "sip show peers" first... if I do that, then the console blocks until I exit and jump back in |
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13:32.02 | lmadsen | oh now other commands don't work anymore either |
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13:36.37 | stoffell | is there any way to keep the ringing volume to maximum on a polycom 320? (it resets after every reboot) |
13:37.24 | stoffell | lmadsen, maybe dns issues? |
13:38.35 | lmadsen | stoffell: never seen that happen.. but entirely possible |
13:40.11 | masus | hi all , does anyone know where i can find the (Mysql SQL) db structure for asterisk 1.6 |
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13:47.46 | hi365 | how can i find which revision are included in which rleases? |
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13:54.13 | *** join/#asterisk feeds (n=feeds@85-135-235-105.adsl.slovanet.sk) |
13:54.22 | feeds | hi |
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13:54.54 | feeds | I have a problem with permissions of my created user asterisktest |
13:56.30 | feeds | I put him into the asterisk group, and changed the asterisk.conf so, that he is the astctl owner, and the runuser, without changing the rungrp or astctl own group from asterisk, assuming that he already is in that group |
13:57.03 | feeds | but when I try running asterisk as asterisktest, it says: Unable to install capabilities. |
13:57.08 | feeds | why is this? |
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13:59.15 | Katty | morning |
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13:59.51 | feeds | good morning |
14:00.26 | jaytee | Katty good morning |
14:00.40 | Katty | jaytee: allo |
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14:04.05 | Katty | jaytee: how're the new cubs settling in? |
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14:11.09 | jaytee | ok, time for me to head off to Digium for 1st day of class |
14:11.11 | jaytee | later all |
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14:19.34 | Katty | so quiet this morning. |
14:19.35 | Katty | who died? |
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14:19.56 | anonymouz666 | Katty! |
14:20.10 | anonymouz666 | how was the weekend? |
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14:21.15 | [TK]D-Fender | goes off to hide the bodies |
14:21.39 | anonymouz666 | [TK]D-Fender: Dexter |
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14:21.58 | [TK]D-Fender | kicks skyphyr in the nads |
14:22.06 | Katty | hugs anonymouz666 |
14:22.21 | Katty | anonymouz666: pretty good. took the pup to mom's house and we all raked leaves. |
14:22.29 | Katty | anonymouz666: well riddick didn't. he jumped and chased them around. |
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14:24.18 | tzanger | odd, this te405 doesn't get detected in this old p3 pc |
14:24.23 | lmadsen | neato |
14:24.28 | tzanger | indeed |
14:24.33 | Katty | hugs lmadsen |
14:24.40 | lmadsen | I believe I will make pancakes for breakfast... |
14:24.47 | lmadsen | hugs the Katty |
14:25.07 | [TK]D-Fender | tzanger: OLD PCI spec perhaps? Not 2.1? |
14:25.13 | Katty | oh? pancakes? |
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14:27.26 | Katty | gingerbread muffin mix makes excellent pancakes. |
14:27.35 | Katty | as does the blueberry and chocolate muffin mixes. |
14:27.59 | [TK]D-Fender | Katty: Pie-Rat pup! http://xs433.xs.to/xs433/08450/peglegdog862.png |
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14:28.09 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
14:28.18 | *** kick/#asterisk [skyphyr!n=joe@216.191.106.163] by [TK]D-Fender ([TK]D-Fender) |
14:28.44 | [TK]D-Fender | hopefully that'll wake up his client... |
14:29.16 | *** mode/#asterisk [-o [TK]D-Fender] by [TK]D-Fender |
14:29.34 | Katty | [TK]D-Fender: that did not parse. |
14:29.48 | Katty | maybe it's too early. |
14:29.55 | [TK]D-Fender | Katty: Keep reading and looking at the pic.... |
14:30.03 | [TK]D-Fender | Katty: Go caffeinate |
14:30.18 | Katty | oh |
14:30.20 | Katty | i didn't see the leg. |
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14:30.29 | Katty | heh |
14:30.32 | Katty | goes to get soda. |
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14:33.18 | Katty | heh. on my way back from getting a soda, the receiptionist asked me if the company had a copy of 'microsoft' she could take home and load on her personal computer so she wouldn't have to go buy it. |
14:33.29 | tzanger | [TK]D-Fender: that's kind of what I am thinking |
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14:34.39 | hi365 | Katty: I need a copy too. Y'know, for my ipod. |
14:35.17 | Katty | [TK]D-Fender: this is the same receiptionist who blogged Obama being the antichrist. |
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14:38.21 | [TK]D-Fender | Katty: there's a great word for people like that : sombitch |
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14:44.04 | Katty | how incredibly annoying. |
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14:46.51 | clintc | plantseeker: after doing a little more digging this might be the answer to our problem: http://bugs.digium.com/view.php?id=12296 |
14:47.17 | clintc | can someone tell me if these patches at http://bugs.digium.com/view.php?id=12296 have been merged into the 1.4.22 release? |
14:47.30 | clintc | or even just a pointer for figuring it out myself |
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14:54.51 | *** part/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
14:54.59 | kaldemar | clintc: seem to be. 1.4.22 ChangeLog 2008-04-08 15:03 says the issue is closed. |
14:55.36 | *** join/#asterisk grEvenX (n=even@apb9hb.ip.ssc.net) |
14:56.20 | clintc | kaldemar: yes, was just looking at revs in 1.4.22... but we are still having mixmonitor sync problems with recordings longer than 3 to 5 minutes or so |
14:56.37 | disposable | does anyone have an init.d startup script for asterisk 1.6 on debian? the one from 1.4 only starts astcanary. asterisk doesn't start but prints out Unable to open pid file '/var/run/asterisk.pid': Permission denied |
14:56.39 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
14:57.26 | Katty | hugs anthm |
15:02.46 | *** join/#asterisk dynamite-- (n=patrick@212.112.255.47) |
15:03.12 | dynamite-- | hey all. im running asterisk 1.4.18 and want to upgrade to the last stable version of asterisk. how do i proceed? |
15:03.45 | [TK]D-Fender | disposable: thats not a lack of init script error, thats a file permissions. Check what user you're running as and who owns the PID |
15:03.53 | [TK]D-Fender | dynamite--: Just compile over |
15:04.01 | [TK]D-Fender | dynamite--: (withing 1.4 series) |
15:04.08 | *** join/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
15:04.09 | dynamite-- | okay |
15:04.13 | fcois93 | hello all |
15:04.14 | dynamite-- | simply get new code |
15:04.20 | dynamite-- | recompile and reinstall? |
15:04.32 | [TK]D-Fender | dynamite--: Yes. As long as you don't do "make samples" you'll be fine |
15:04.39 | fcois93 | I need to know how asterisk can forward some headers ? |
15:04.39 | dynamite-- | right, cheers.. |
15:06.14 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
15:06.14 | tzanger | hmm I'm pretty sure the PIIX4 is PCI 2.1 compliant |
15:08.09 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-67454e1c5616a06d) |
15:08.09 | *** mode/#asterisk [+o putnopvut] by ChanServ |
15:08.18 | *** join/#asterisk IsUp (n=nocturne@unaffiliated/isup) |
15:08.23 | hi365 | im having a weird situation where queues is showing a member as avalible, while in truth he's not |
15:08.27 | IsUp | hi ya |
15:08.50 | hi365 | there seems to be some bugs about this, but I think I applyed all the lates patches - and its still not reporting the proper state |
15:09.00 | stintel | \ |
15:09.29 | IsUp | can anyone explain how can i use G option in Dial? i am trying to transferring caller and called party to extension and want to bridge calls. but call is dropping after execution of dialplan. |
15:09.34 | IsUp | any ideas?.. |
15:11.21 | guax | IsUp: show application Dial |
15:11.32 | guax | ooops |
15:11.44 | guax | wrong anderstand of the question sorry |
15:11.50 | *** join/#asterisk [8none1] (n=aherbert@cerberus.franklinamerican.com) |
15:12.08 | *** join/#asterisk CrazyTux (n=brandon@user-vcauh95.dsl.mindspring.com) |
15:12.15 | IsUp | i did |
15:12.32 | plantseeker | Thanks clintz |
15:12.40 | IsUp | but call is not bridging, its dropping after execution. and |
15:12.49 | disposable | [TK]D-Fender: i've added user asterisk, group asterisk, chowned /etc/asterisk to asterisk:asterisk, i'm using the init script from 1.4 (on 1.6). astcanary is started under root, and i can start 'asterisk' under root with no problem. only when i do it with the startup script, i get the permissions error. how do i start it as asterisk? |
15:12.57 | IsUp | its just execute dialplan for 'called' party, i cant set any variables to caller |
15:13.09 | guax | IsUp: why you need to transfeer them? |
15:13.10 | [TK]D-Fender | IsUp: That option does not bridge after |
15:13.23 | IsUp | well, let me explain my problem. |
15:13.56 | IsUp | for example, i am sending a call with Dial at 14:01, but call is answered on 14:03, and caller is hangs up. how can i get actual 'answered' time? |
15:14.03 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
15:14.44 | IsUp | so theres 2 mins lost, on connection state, on ringing etc... i want to know exact time when channel answered. |
15:15.02 | guax | IsUp: theres a variable ANSWEREDTIME |
15:15.04 | IsUp | and i am using AGI, i want to save this information to my database. but i cant read a dead channel. |
15:15.25 | guax | you can read a dead channel on DeadAGI call |
15:15.28 | lmadsen | IsUp: DeadAGI()? |
15:15.39 | IsUp | yeah, i am using DeadAGI |
15:15.54 | jameswf | okay good news and bad news |
15:15.59 | jameswf | Nortel Cuts 1,300 Jobs and Lowers Its Outlook <----good |
15:15.59 | IsUp | when caller hangs up, i need the actual answered time =) |
15:16.18 | guax | IsUp: after dial get the variable ANSWEREDTIME |
15:16.40 | jameswf | Gm SUV production plant working over time to keep up with demand after drop in gas prices <--bad... god america is stupid |
15:17.04 | *** join/#asterisk freakazoid0223 (n=mattc@pool-72-81-0-69.phlapa.east.verizon.net) |
15:17.27 | IsUp | let me tell if its working. |
15:17.28 | coppice | jameswf: there is nothing good about nortel having problems. it means the entire telecoms outlook is gloomy |
15:17.32 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
15:17.51 | *** join/#asterisk tobias (n=tobias@user-0c8hj2l.cable.mindspring.com) |
15:19.08 | jameswf | The proprietary telecom industry going down means people are looking for cheaper alternitives.... in flys asterisk with tights and a cape |
15:19.36 | coppice | at times like thing it generally means people are not looking for anything at all |
15:19.38 | jameswf | Circuit city is bankrupt that kinda sucks |
15:19.48 | phpboy | :( |
15:19.55 | jameswf | I dont shop there but still sucks... |
15:19.57 | phpboy | this is happening with most companies these days :T |
15:20.14 | jameswf | no more state of the art service |
15:22.58 | *** join/#asterisk jmacz (n=jmacz@200.26.159.42) |
15:23.10 | jameswf | thats funny apple and rim battle it out to be second to nokia |
15:23.19 | IsUp | yeah, i can get with ANSWEREDTIME but, how can i get it without using DeadAGI?.. |
15:24.48 | [TK]D-Fender | IsUp: same way you would normally. |
15:25.04 | *** join/#asterisk mark_csi (n=csiadmin@host217-41-18-3.in-addr.btopenworld.com) |
15:25.33 | IsUp | but when i am hangup it does nothing. and i am using AGI script. |
15:26.38 | mark_csi | Hello, anyone know if you can buy hardware echo cancellation after you've already got a digium analogue card? |
15:26.52 | hi365 | belives so |
15:30.15 | mark_csi | Thanks hi365 - just found it. VPMADTO32 |
15:30.22 | [TK]D-Fender | IsUp: Clrealy should be using DeadAGI |
15:30.29 | disposable | ls |
15:30.39 | disposable | sorrywrong window |
15:31.29 | *** join/#asterisk Bad_Robot- (n=BadRobot@cpe-76-173-219-25.socal.res.rr.com) |
15:35.57 | mark_csi | exit |
15:36.01 | *** part/#asterisk mark_csi (n=csiadmin@host217-41-18-3.in-addr.btopenworld.com) |
15:36.09 | *** join/#asterisk mark_csi (n=csiadmin@host217-41-18-3.in-addr.btopenworld.com) |
15:36.21 | mark_csi | hehe wrong window |
15:36.23 | *** part/#asterisk mark_csi (n=csiadmin@host217-41-18-3.in-addr.btopenworld.com) |
15:36.52 | IsUp | lol |
15:36.54 | *** join/#asterisk mbranca (n=matteo@93-62-230-243.ip24.fastwebnet.it) |
15:36.57 | IsUp | LOL |
15:40.55 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
15:44.52 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
15:54.46 | getsql08 | silly question but how to i start asterisk |
15:54.53 | getsql08 | via the CMD prompt |
15:55.02 | *** join/#asterisk mark_csi (n=csiadmin@host217-41-18-3.in-addr.btopenworld.com) |
15:55.31 | IsUp | CMD prompt? |
15:55.49 | getsql08 | command prompt |
15:55.51 | getsql08 | in linux |
15:56.05 | getsql08 | IE /usr/sbin/asterisk start? |
15:56.20 | [TK]D-Fender | getsql08: "asterisk -gvvvvvvvc" |
15:56.34 | [TK]D-Fender | getsql08: if * is not already running as a daemon |
15:56.55 | [TK]D-Fender | getsql08: Which is normally the case. At which point you jsut access CLI via "asterisk -r" |
15:57.39 | getsql08 | [Nov 10 09:57:04] WARNING[6478]: manager.c:3159 init_manager: Unable to bind socket: Address already in use |
15:57.58 | [TK]D-Fender | getsql08: Looks like * is already running. |
15:58.01 | [TK]D-Fender | getsql08: try conencting to it. |
15:58.14 | *** join/#asterisk keith4_ (n=keith@lust.CC.Lehigh.EDU) |
15:58.56 | getsql08 | [Nov 10 09:58:34] WARNING[6487]: pbx.c:4693 add_pri: Unable to register extension 'abn', priority 3 in 'gdincoming', already in use |
15:59.00 | getsql08 | how |
15:59.06 | getsql08 | unable to register add_pri EXT |
15:59.15 | [TK]D-Fender | getsql08: Go look at your dialplan. You've clearly duplicated things |
15:59.33 | [TK]D-Fender | getsql08: its telling you to your face exactly what. |
16:01.04 | getsql08 | how about this |
16:01.05 | getsql08 | <PROTECTED> |
16:01.05 | getsql08 | * make sure you read the INSTALL doc and apply the one of the following patches * |
16:01.05 | getsql08 | * channel.c.hangup_callerid_ast12.diff * |
16:01.05 | getsql08 | * channel.c.hangup_callerid_ast14.diff |
16:01.13 | getsql08 | how do i apply these patches |
16:01.30 | [TK]D-Fender | ~pb |
16:01.31 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
16:01.32 | *** join/#asterisk gazzerh (n=garryh@host-212-158-227-149.dslgb.com) |
16:01.32 | [TK]D-Fender | ^^^^ |
16:01.38 | [TK]D-Fender | getsql08: do not spam in here please. |
16:01.59 | [TK]D-Fender | getsql08: And where did you see that? |
16:03.42 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
16:04.31 | getsql08 | [TK]D-Fender: one sec.. i am back to my dial plan |
16:04.36 | getsql08 | i have clearly duplicated things |
16:04.41 | getsql08 | but I do not see any duplicate entries |
16:04.45 | getsql08 | in manager.conf |
16:04.48 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
16:04.58 | [TK]D-Fender | getsql08: that isn't the dialplan <- |
16:07.45 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
16:07.49 | *** part/#asterisk SteelSide (n=SteelSid@217.76.87.68) |
16:13.14 | [TK]D-Fender | BRB, rebooting. |
16:15.15 | *** join/#asterisk nikko (n=nikko@69.57.49.100) |
16:16.00 | ManxPower | BTW, why do you have an extension called "abn"> |
16:16.02 | ManxPower | ? |
16:16.52 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
16:17.16 | *** join/#asterisk l2trace99 (n=jr@static-71-251-65-16.tampfl.fios.verizon.net) |
16:17.25 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
16:21.36 | IsUp | brb reboot |
16:22.03 | Katty | ponders lunch |
16:22.05 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
16:22.11 | Katty | ZEEEK! |
16:22.15 | Zeeek | hey there ho there ho there |
16:22.15 | Katty | hugs Zeeek |
16:22.28 | Zeeek | air kisses Katty |
16:22.48 | Zeeek | Ladies and gentlement, for the first time ever on this channel |
16:22.57 | Zeeek | I HAVE AN asterisk QUESTION |
16:23.00 | Katty | omg |
16:23.01 | Katty | faitns |
16:23.04 | Katty | oh. |
16:23.06 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
16:23.07 | Katty | let's try that again. |
16:23.09 | Katty | faints. |
16:23.14 | Bad_Robot- | heh |
16:23.15 | Zeeek | where does SipAddHeader() go in the dial plan? Right beofre the dial() ? |
16:23.18 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
16:23.19 | Katty | yes. |
16:23.23 | [TK]D-Fender | Zeeek: Any time before |
16:23.30 | Katty | i can pastebin something, if you need it. |
16:23.44 | Zeeek | you mean like in another file? </joke> |
16:24.03 | Zeeek | no I think that's all I need |
16:24.12 | [TK]D-Fender | Zeeek: Sure... as long as its executed before the dial in the call you want it to apply to :) |
16:24.32 | Zeeek | so to call Talkshoe, apparently they've added a SIP "failsafe" way to defeat DTMF timing problems |
16:24.37 | Katty | oh noes. i'm out of checks. |
16:24.49 | Zeeek | how about balances? Out of those too? |
16:25.10 | [TK]D-Fender | Katty: Executive Branch may be right for you! |
16:25.21 | Zeeek | So if I wanted to send a Subject header: |
16:25.35 | ifnotwhynot | where can i list of hangup causes on PRI for asterisk, got hangup cause 88 can't seem to google it right |
16:25.52 | *** join/#asterisk sdaniels (n=chatzill@216.65.195.52) |
16:26.07 | Zeeek | SipAddHeader(Subject: <passocde>12345</passcode><pin>1</pin>) |
16:26.12 | Zeeek | Correct? |
16:26.13 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
16:26.18 | [TK]D-Fender | Katty: If it makes you feen any better, MO is looking at a recount that you can hold over your receptionist :) |
16:26.25 | [TK]D-Fender | Zeeek: Sure |
16:26.34 | Zeeek | no quotes or anything silly like that |
16:26.40 | Katty | [TK]D-Fender: i'm not going to even go there. |
16:26.58 | Zeeek | of course, there is an error in my call above |
16:27.02 | [TK]D-Fender | ifnotwhynot: http://www.google.ca/search?hl=en&q=isdn+code+88&btnG=Google+Search&meta= |
16:27.18 | [TK]D-Fender | ifnotwhynot: Funny I see a lot of answers on that first page without even DRILLING the links |
16:27.36 | Zeeek | Drill Baby Drill |
16:27.40 | [TK]D-Fender | ISDN cause codes help you diagnose problems with calls. ... 88. Incompatible destination. 89. Non-existent abbreviated address entry ... |
16:27.54 | Zeeek | shudders at the memory evoked by that line |
16:27.58 | *** join/#asterisk astrOdz (n=astrOdz@119.92.213.100) |
16:28.02 | Katty | seanbright: LIAR |
16:28.36 | seanbright | i is |
16:28.56 | seanbright | i'll update to something more appropriate |
16:28.59 | ifnotwhynot | thx |
16:29.06 | seanbright | there. |
16:29.30 | Katty | seanbright: that i can believe. |
16:29.33 | Katty | lights seanbright on fire. |
16:29.42 | [TK]D-Fender | grabs some marshmallows |
16:30.17 | [TK]D-Fender | Seans-nuts roasting on an open fire! |
16:30.22 | [TK]D-Fender | carols... |
16:30.31 | Katty | haha |
16:30.33 | Katty | you're awful |
16:30.35 | Katty | also! |
16:30.35 | Bad_Robot- | feels bad for sean |
16:30.40 | Katty | seanbright: what is 'twitterBar'? |
16:31.07 | ManxPower | There's nothing wrong with twittering, as long as you do it in private and wash your hands after. |
16:31.25 | Katty | seanbright: something for a browser? |
16:31.48 | Katty | oooh. firefox addon. schnazzy. |
16:32.39 | *** join/#asterisk Greek-Boy (n=email@41.222.89.77) |
16:33.46 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
16:33.59 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
16:36.12 | Katty | SwK: GET OUT |
16:36.14 | Katty | hugs CunningPike |
16:36.22 | Zeeek | http://food4wine.ning.com/page/vuc-and-other-news |
16:36.28 | Zeeek | thanks everyone |
16:36.42 | Zeeek | air kisses Katty again |
16:37.02 | Katty | i never really got air kisses. |
16:37.03 | Qwell | oh god, another sip extension? |
16:37.05 | Katty | the point of them, that is. |
16:37.14 | Zeeek | wait, you're Americans. {{{{{{Katty}}}}} |
16:37.27 | Katty | Zeeek: are you french? |
16:37.37 | CunningPike | hugs Katty right back |
16:37.42 | Zeeek | Katty: I am today |
16:37.51 | *** join/#asterisk synchris (n=synchris@athedsl-86032.home.otenet.gr) |
16:37.56 | Zeeek | Qwell: referring to my header? |
16:38.16 | Katty | i'm lost. that's okay tho. i stay lost. |
16:38.31 | Zeeek | Lost is good. Don't ever change |
16:38.32 | Katty | CunningPike: how're you today, deary? |
16:38.49 | CunningPike | Katty: Excellent, thanks - et vous? |
16:39.07 | Zeeek | Qwell: was that comment directed at I ? |
16:39.16 | Qwell | Zeeek: yes |
16:39.18 | Katty | CunningPike: hungry, but plotting lunch. |
16:39.27 | Zeeek | In that case it deserves an answer |
16:39.33 | CunningPike | Katty: Cool - just after breakfast here :) |
16:39.35 | Zeeek | It isn't an "extension" |
16:39.40 | Katty | CunningPike: must get some groceries on the way home from work. I'm making pork chops, homemade mashed potatoes, and some sort of veggie for dinner. |
16:39.57 | CunningPike | Katty: I'll be right over |
16:40.04 | Katty | plans to make extra. |
16:40.07 | CunningPike | ;) |
16:40.17 | Zeeek | Talkshoe's SIP client uses it to send "pseudo" callerID and conference number info |
16:40.23 | Katty | CunningPike: want a tip on the mashed potatoes? |
16:40.30 | Katty | CunningPike: used diced canned potatoes. already cooked. |
16:40.36 | CunningPike | Ah |
16:40.43 | Katty | CunningPike: just mash and mix with the milk and butter--insta homemade goodness! |
16:40.53 | Katty | complete with those mashed potato chunks. |
16:41.09 | Zeeek | so, the faithful participants of VUC who constantly complain about Inband/RFC DTMF won't need to anymore |
16:41.15 | CunningPike | is Irish - not sure I could use canned spuds - it's against religion or something |
16:41.16 | Qwell | Zeeek: it's an extension |
16:41.17 | Zeeek | ya see? |
16:41.35 | ifnotwhynot | hi there i am lost,, i am receiving a call to a PRI span 1 for some reason(http://pastebin.com/m261bb11d) i wan to forward the call to another pri span 2 connected to a legacy pbx.. any help welcome. |
16:41.36 | Katty | CunningPike: well i'm only half irish, so it's okay if i cheat |
16:41.38 | putnopvut | That seems like an odd use of the Subject header |
16:41.44 | CunningPike | Katty: ;) |
16:41.49 | Zeeek | Qwell I prefer to see it as the opposite of "deprecated" |
16:41.57 | Qwell | putnopvut: probably violates some rfc :p |
16:42.00 | Zeeek | maybe it's "reprecated" |
16:42.04 | Qwell | like, "don't put meaningful crap in here" |
16:42.24 | Zeeek | "If it works, use it"⢠|
16:42.35 | Qwell | You just described SIP in 5 words. |
16:42.42 | Zeeek | indeed |
16:42.50 | putnopvut | Qwell: Well, it's not really violating anything. But I would have expected some sort of X- header to be used. |
16:43.07 | *** join/#asterisk italorossi (n=italoros@201.76.154.111.intranet.digi.com.br) |
16:43.09 | Zeeek | which is where logic forbidding masturbation falls short |
16:43.25 | putnopvut | Like Zeeek said, "if it works, use it" :) |
16:43.35 | Zeeek | "If I shouldn't do this, why is it there?" |
16:43.52 | *** join/#asterisk fogo (n=Paul@69.169.132.35.provo.static.broadweave.net) |
16:44.04 | Katty | Zeeek: then how do you explain shock collars :< |
16:44.09 | Zeeek | If RFC is a "request" then who said I said it was ok to use? huh? |
16:44.35 | Zeeek | Katty: tu m'intéresses! You're into that stuff? I wish I wasn't so far away |
16:44.44 | ManxPower | ifnotwhynot: Cause 88 means "incompatible destination" |
16:44.52 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
16:44.59 | Katty | Zeeek: erm. |
16:45.06 | pif | is it still possible to use zapte with 1.4.22 ? |
16:45.10 | pif | zaptel |
16:45.15 | putnopvut | pif: yes |
16:45.24 | Zeeek | but it's desecrated |
16:45.37 | ManxPower | ifnotwhynot: But I have no idea where this message comes from "-- Requested transfer capability: 0x10 - 3K1AUDIO" |
16:45.41 | Arsenick- | Zeeek, t'es en amour ? ;) |
16:45.45 | *** join/#asterisk Ropeguru (n=yeah@adsl-067-035-094-144.sip.mia.bellsouth.net) |
16:45.51 | Zeeek | non, en chaleur |
16:45.59 | Arsenick- | haha |
16:46.04 | Zeeek | à mon âge avancé |
16:46.05 | Katty | oh wow. i just read that obama's chief of staff wants people through the age of 18 to 25 to serve in the military |
16:46.12 | Qwell | yay for being 26 |
16:46.13 | tzafrir_laptop | Zeeek, I guess you read the SIP 4.0 RFC |
16:46.22 | Katty | Qwell: i need another year STAT |
16:46.22 | Zeeek | Great idea. Cuts way down on anyone wanting to approve of wars |
16:46.26 | ManxPower | Katty: change.gov is their new web site |
16:46.31 | putnopvut | tzafrir_laptop: that is the best RFC |
16:46.32 | pif | where is the last version of 1.4.x zaptel? |
16:46.35 | Katty | ManxPower: this makes me want to strangle someone. |
16:46.40 | Qwell | pif: same place as the version before that |
16:46.47 | pif | can't get it from digium |
16:46.49 | ManxPower | Katty: Maybe so, but I bet we won't have as many wars |
16:46.51 | [TK]D-Fender | pif: in the channel topic |
16:46.51 | Qwell | pif: why not? |
16:47.01 | Katty | ManxPower: not if it's manditory. |
16:47.10 | Katty | ManxPower: there will be an uprising in the religious communities tho. |
16:47.20 | Zeeek | yeah right |
16:47.32 | Katty | ManxPower: Jehovah's Witnesses would rather go to jail than serve in the military |
16:47.37 | ManxPower | Katty: Sure it will. It's one thing for people to volunteer -- if they want to be killers that's up to them, but when people are drafted their parents get pretty upset |
16:47.47 | Katty | nods |
16:47.49 | [TK]D-Fender | pif: http://downloads.digium.com/pub/zaptel/ <- sure as heck works for me. |
16:47.56 | Zeeek | and therefore war becomes very unpopular |
16:48.01 | ManxPower | Katty: there are and have always been options for people who object to being drafted. |
16:48.08 | pif | [TK]D-Fender: thx |
16:48.11 | Zeeek | In fact, it has been said the ending the raft was a way to stiffle protest |
16:48.26 | Zeeek | when you think about that, it makes a little more sense |
16:48.26 | [TK]D-Fender | pif: Might want to try a tiny hit harder next time. |
16:48.35 | Katty | ManxPower: never read about them. growing up as Jehovah's witness, and a female, has a way of turning you against all things military. |
16:48.47 | ManxPower | Katty: Personally I don't like the idea of required military service -- but many, many, man countries around the world have it. |
16:48.48 | pif | kisses [TK]D-Fender 's feet |
16:49.29 | Katty | United States of <strike>America</strike> SPARTA! |
16:49.59 | Zeeek | I may be the only person here who has served in a US branch of milserv? |
16:50.13 | Katty | Zeeek: my fiance served 10 years |
16:50.14 | ManxPower | Zeeek: There's a BIG difference between "Mom, Dad, this is what I want to do! <put in reasons here> and I'm 18 and you can't stop me" and "I don't want to go to war Mom and Dad! I'm scared! " |
16:50.22 | [TK]D-Fender | ManxPower: You wouldn't mind it as much if not for who's running it :) |
16:50.25 | Zeeek | BUT HE's NOT HERE. NANANANANA |
16:50.36 | Katty | thank goodness ;) |
16:50.37 | pif | Sparta didn't have mixed breeds as leaders |
16:50.46 | pif | hence their demise |
16:51.34 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
16:51.55 | jblack | The US free market economy approach to waging war has been an interesting experiment. |
16:51.56 | Zeeek | Well, I got an asterisk answer here, possibly the first in the last three years. We now return you to an endless sequence of OT comments. THANK YOU ALL |
16:52.10 | Katty | i live for OT |
16:52.11 | WHYS | OF the people I know, the ones not going to war were the brave ones. There is a fine line between brave and stupid. |
16:52.17 | ManxPower | jblack: It worked much worse than I expected. |
16:52.19 | Zeeek | Katty: me too. Asterisk is boring |
16:53.01 | jblack | Oh? |
16:53.39 | WHYS | Not that they are the only two reasons for going to war. I hear "adventure" is a big one. |
16:53.44 | pif | iraq, vietnam are not real wars, only bullying of a weaker nation that turned into quagmires |
16:53.53 | WHYS | a hollow voice says "plugh" |
16:53.57 | fcois93 | I need to know how asterisk can forward some headers ? |
16:54.00 | ManxPower | jblack: I felt that an all volunteer army was a good thing. The problem with my idea is that there is much less resistance to war when the only people getting killed WANT to be there. |
16:54.35 | *** join/#asterisk Toerkeium (n=Miranda@201.216.206.221) |
16:54.43 | jblack | Ohhh, so it's not the direct results, but the implications. That's a very interesting point. |
16:54.50 | WHYS | ManxPower, on our side anyway. |
16:54.51 | ManxPower | fcois93: Asterisk is a B2BUA SIP device, it does not forward ANY headers. |
16:55.01 | Toerkeium | guys, "show g729" shows "0/2 encoders/decoders of 10 licensed channels are currently in use". How/what can I check who is using those 2 decoders? |
16:55.08 | Zeeek | ManxPower: unfortunately it lowers the quality both of the army and of the experience gained. (not including war, which is awful) |
16:55.39 | Zeeek | Anyway, we should stop attacking shit for oil and solve the problmes more intelligently |
16:55.42 | jblack | Zeeek: Do you really think that? I think quite the opposite. |
16:55.44 | Zeeek | and there I meave you |
16:55.54 | Zeeek | I meave you bvery much |
16:56.04 | ManxPower | Zeeek: IF they started drafting congrespeople's and senator's kids this war would be over by now. |
16:56.07 | *** join/#asterisk Defraz (n=T0tal@63.228.246.229) |
16:56.10 | Zeeek | jblack: what that war is awful? |
16:56.11 | fcois93 | ManxPower: Is it possible to imagin a solution for that? |
16:56.20 | jblack | But they never draft congress' kids. |
16:56.21 | Zeeek | ManxPower: so we seem to agree? |
16:56.30 | Zeeek | bye for now |
16:56.45 | Zeeek | and thanks for all the phish |
16:56.48 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
16:57.02 | ManxPower | jblack: I would have to find a source to cite, but if Elvis was drafted, I doubt politician's kids were exempt. |
16:57.10 | Carlos_PHX | The real solution to unpopular wars is to eliminate the government's ability to steal our money to fund it. If they had to ask us to vote to pay for a war, it wouldn't happen unless it was truly necessary. |
16:57.24 | WHYS | Look up draft board 100 |
16:57.34 | coppice | Elvis and Mohammed Ali were drafted as show pieces |
16:57.37 | WHYS | It's where important peoples kids went. |
16:57.50 | Katty | off i go. cheerio. |
16:58.11 | jblack | Congressional children, unlike 85% of the population, usually go to college. There's a draft exception for that. |
16:58.59 | jblack | I don't think elvis is the example you mean it to be. Didn't the folks in congress dislike elvis, socially speaking, due to his gyrating hips? |
16:59.09 | coppice | people like Elvis ended up entertaining the troups, and were always kept away from danger. morale would have suffered if they were hurt |
17:00.08 | jblack | Look at george bush. Didn't he serve in the national guard, and weren't there questions as to whether he actually bothered to show up? |
17:00.37 | coppice | no. no. nobody dares to even ask such questions :-) |
17:00.53 | jblack | At leat not twice, right? :) |
17:02.39 | *** join/#asterisk devilsoulblack (n=aandaluz@srv.ec-gye.internet.geainternacional.com) |
17:02.52 | *** join/#asterisk chebyte (n=chebyte@host172.190-225-228.telecom.net.ar) |
17:02.57 | chebyte | hi people |
17:03.00 | chebyte | i have a question |
17:03.12 | Carlos_PHX | We probably have an answer, want to see if they match? |
17:03.20 | chebyte | i need buy a board asterisk with support for 4 lines |
17:03.20 | jblack | That's good. Everyone should have a question. |
17:03.20 | pif | it better be about politics |
17:03.31 | chebyte | anyone can recommend me? |
17:03.43 | *** join/#asterisk keith4_ (n=keith@lust.CC.Lehigh.EDU) |
17:03.46 | jblack | POTS ? |
17:04.07 | pif | didn't Dali say "computer are boring, they only have answers" ? |
17:04.14 | chebyte | I ' am viewing a openvox a400p |
17:04.17 | ManxPower | chebyte: 4 of what kind of ports? |
17:04.25 | chebyte | o digium tdm400p |
17:04.33 | ManxPower | chebyte: if you go with OpenVox almost nobody here will be able to help you. |
17:04.36 | chebyte | fxo |
17:05.29 | chebyte | ManxPower: ok what recommend me? |
17:05.49 | pif | coppice: are you still contributing to callweaver? |
17:06.01 | *** join/#asterisk djin (n=djin@84-104-110-116.cable.quicknet.nl) |
17:06.08 | devilsoulblack | hi any one have exprience with openr2 |
17:06.59 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
17:07.38 | coppice | pif: yes |
17:07.41 | chebyte | what kind of board is good ? |
17:07.45 | *** join/#asterisk aiksa[LV] (n=root@mx.fiveplus.lv) |
17:07.53 | aiksa[LV] | beep, tzafrir_laptop there? |
17:08.01 | pif | coppice: so the project is still alive and well ? |
17:08.33 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
17:08.48 | tzafrir_laptop | aiksa[LV], yes |
17:08.52 | coppice | yes |
17:09.08 | *** part/#asterisk WHYS (i=lpfm@137.28.94.209) |
17:09.52 | devilsoulblack | try to make and make install over openr2 and get "make: *** No targets specified and no makefile found. Stop." |
17:10.18 | *** part/#asterisk mark_csi (n=csiadmin@host217-41-18-3.in-addr.btopenworld.com) |
17:12.42 | *** join/#asterisk erth64net (n=erth64ne@mail.opensourcery.com) |
17:13.18 | *** part/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
17:13.30 | *** join/#asterisk spokra (n=spokra@blockhead.sea0.speakeasy.net) |
17:15.02 | tzafrir_laptop | devilsoulblack, looks like you need to run ./configure |
17:15.12 | *** join/#asterisk wiscados (n=mint@81.25.184.155) |
17:17.09 | *** join/#asterisk l2trace99 (n=asd@static-71-251-65-16.tampfl.fios.verizon.net) |
17:18.13 | *** join/#asterisk lejun (n=lejunhu@66.178.134.235) |
17:21.03 | [TK]D-Fender | chebyte: Sangoma A200d |
17:22.13 | *** join/#asterisk Spydre (n=nbaker@75.147.255.17) |
17:23.11 | ManxPower | tzafrir_laptop: So basically he needs to read the INSTALL or README file. |
17:23.43 | Spydre | Anyone around that can offer some advice on videoconferencing? |
17:23.54 | ManxPower | I must admit Digium putting almost all the docs in TeX format almost guarantees nobody will read them. Great going Digium! |
17:24.16 | Qwell | ManxPower: RTFPDF |
17:25.04 | ManxPower | [root@bourbon asterisk-1.6.0.1]# find . -name "channelvariables*" -print |
17:25.05 | ManxPower | ./doc/tex/channelvariables.tex |
17:25.05 | ManxPower | [root@bourbon asterisk-1.6.0.1]# |
17:25.09 | ManxPower | Where was that PDF again? |
17:25.50 | [TK]D-Fender | ManxPower: Yeah add me to the picketer's list... |
17:26.02 | tzafrir_laptop | ManxPower, the README has nothing about it. The INSTALL is the generic autotools one |
17:26.13 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
17:26.14 | *** join/#asterisk freakazoid0223 (n=mattc@pool-72-81-0-69.phlapa.east.verizon.net) |
17:26.33 | hardwire | Keizer: hows things? |
17:26.37 | ManxPower | [TK]D-Fender: They must have been drunk or high when they made that decision. |
17:26.41 | hardwire | rolls Keizer |
17:27.14 | Spydre | Is there any active development of video conferencing solutions for asterisk? I've tried app_conference but that doesn't seem very complete, and hasn't been updated lately. |
17:27.50 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
17:28.05 | *** join/#asterisk CrazyTux (n=brandon@user-vcauh95.dsl.mindspring.com) |
17:28.15 | ManxPower | [TK]D-Fender: The decision will make ME stop telling people to look at the docs included in the tarball. |
17:28.45 | ManxPower | tzafrir_laptop: and INSTALL doesn't tell you to run ./configure? |
17:29.00 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
17:29.24 | *** join/#asterisk boolean12 (n=boolean1@tandem.uplinktel.com) |
17:29.52 | ManxPower | People complain about us being "unix elitists" No better elitism than making all the poor sods figure out how to read TeX formatted files. |
17:30.02 | tzafrir_laptop | it does. Though it could have been a bit clearer |
17:31.01 | ManxPower | ~mailinglist |
17:31.02 | jbot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
17:35.41 | [TK]D-Fender | ManxPower: I still haven't figured out how... |
17:38.26 | ManxPower | [TK]D-Fender: Maybe the marketing people said "The docs are too easy to find! Lets make it harder!" |
17:38.28 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
17:38.45 | [TK]D-Fender | ManxPower: What does TeX really add to our lives? |
17:38.57 | ManxPower | [TK]D-Fender: maybe for 1.8 they will require EMACS in order to edit the config files! |
17:39.13 | *** join/#asterisk Wayhigh (i=wayhigh@www.kevinlynn.com) |
17:39.21 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
17:39.32 | Zeeek | I'm back with question #2! 2 in one day |
17:39.47 | ManxPower | [TK]D-Fender: it's a vendor/platform independent document format. Much like oh say HTML or TEXT. |
17:39.49 | Zeeek | How can I find out who this number belongs to (provider): Call 1-800-949-1284 |
17:40.15 | Zeeek | I just got a domain scam and they want me to call Call 1-800-949-1284 and pay $75 for one name |
17:40.39 | Zeeek | I encourage you to get them tazlking, they must be real scumbags to do that |
17:40.47 | [TK]D-Fender | ManxPower: I mean it isn't any more searchable or anything is it? No in-lin pics or anything fo value? |
17:41.09 | ManxPower | Zeeek: http://www.google.com/search?hl=en&q=800-949-1284&btnG=Google+Search&aq=f&oq= |
17:41.21 | ManxPower | [TK]D-Fender: It's a desktop publishing format. |
17:42.09 | ManxPower | formatting, pictures, generated table of contents, citations, everything you need for desktop publishing. |
17:42.51 | [TK]D-Fender | ManxPower: And how much of that did we USE in converting to it? |
17:44.04 | ManxPower | [TK]D-Fender: No idea, as I'm not going to install 100MB of TeX and required libraries in order to find out. |
17:45.19 | ManxPower | [TK]D-Fender: The thing that really annoys me is that SOME of the docs are still in text format. |
17:45.30 | ManxPower | just not the ones I was looking for. |
17:46.19 | *** join/#asterisk kotique (n=picachu@host-static-89-41-72-85.moldtelecom.md) |
17:46.27 | ManxPower | I predict that UPGRADE.txt will become UPGRADE.tex before the next relase. |
17:46.31 | kotique | guys. do you know any good graphical SIP proto analyzer ? |
17:46.37 | ManxPower | That will raise the barrier even more. |
17:46.39 | kotique | something like this http://bugs.digium.com/file_download.php?file_id=19827&type=bug |
17:47.09 | magronez | is back |
17:47.56 | *** join/#asterisk ElSonico (n=tav@xdsl-179-7.nblnetworks.fi) |
17:52.12 | pif | is that a zaptel.conf problem? "ZT_SPANCONFIG failed on span 1: Invalid argument (22" |
17:52.21 | [TK]D-Fender | pif: Clearly |
17:52.41 | pif | I just upgraded from zaptel versions without changing the conf |
17:53.01 | pif | 1.4.11 to 1.4.12.1 |
17:53.26 | ManxPower | pif: did you also upgrade your kernel? |
17:53.30 | pif | yep |
17:53.51 | ManxPower | pif: Did you recompiled zap after upgrading the kernel? |
17:53.55 | pif | wct4xxp loads |
17:54.02 | ManxPower | That was not my question |
17:54.08 | pif | yes, after reboot to new kernel |
17:54.31 | ManxPower | pif: Odd. |
17:55.00 | pif | didn't do distclean |
17:55.09 | pif | leftovers from older kernel? |
17:55.22 | ManxPower | no idea |
17:55.26 | pif | trying |
17:55.28 | ManxPower | did you at least do a make clean? |
17:55.38 | pif | yep and it failed |
17:55.44 | ManxPower | failed? |
17:55.47 | *** join/#asterisk sp00k3y (n=sp00k3y@wsip-98-190-136-194.ph.ph.cox.net) |
17:55.53 | ManxPower | as in error message or as in "no rule to make clean" |
17:56.32 | l2trace99 | does anyone know how do bridge the channels once in attendend transfer ? |
17:56.41 | [TK]D-Fender | pif: Did you upgrade * as well? |
17:56.49 | pif | yes, 1.4.22 |
17:57.01 | [TK]D-Fender | l2trace99: huh? |
17:57.02 | ManxPower | l2trace99: however your phone docs say to do it. |
17:57.22 | l2trace99 | it is a function of the phone or * ? |
17:57.26 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
17:57.29 | [TK]D-Fender | l2trace99: How are you doing it? |
17:57.47 | l2trace99 | using atxfer in features |
17:58.00 | ManxPower | l2trace99: That depends on the kind of transfer. I recommend using phone based transfers rather than the ugly DTMF transfer hack people seem so fond of. |
17:58.42 | l2trace99 | know any good softphones that allow for a warm tranfer ? |
17:58.47 | pif | is ztdummy supposed to load with wct4xxp ? |
17:58.53 | pif | does it interfere ? |
17:58.53 | ManxPower | pif: no |
17:58.56 | *** join/#asterisk Daejeo (n=chatzill@118.219.208.218) |
17:59.07 | Daejeo | anyone have a iphone? |
18:00.23 | [TK]D-Fender | pif: No, you should not load ztdummy. |
18:00.41 | pif | without it ztcfg completes |
18:01.18 | pif | but I only have red alarms :( |
18:01.28 | *** join/#asterisk farkus (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
18:02.28 | pif | with 1.4.11 they were green |
18:02.45 | Zeeek | how cool would it be if IRC had tags like the web 2.0 stuff? |
18:02.58 | Zeeek | How large would ZTDUMMY be in this channel? |
18:03.01 | Zeeek | HIGE |
18:11.53 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
18:13.48 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
18:14.07 | *** join/#asterisk smk (n=scott@cobra.httpd.org) |
18:14.56 | Katty | ah, screen, how i love thee. |
18:14.57 | *** join/#asterisk Segnale007 (n=Pietro@host36-253-dynamic.36-79-r.retail.telecomitalia.it) |
18:15.16 | pif | do alarms stay RED when asterisk is stopped? |
18:17.02 | ManxPower | pif: try it and see |
18:17.34 | pif | they stay RED with 1.4.22, but go to GREEN with 1.4.21.2 |
18:17.56 | Daejeo | Katty: :) meow meow |
18:19.22 | Katty | Daejeo: murrrrrrrrow |
18:19.22 | *** join/#asterisk Carlos_PHX (n=Carlos@64.129.235.200) |
18:19.22 | Katty | herroes Carlos. |
18:20.03 | Daejeo | Kattyt Meowing and What it Means |
18:20.04 | Carlos_PHX | 'morning |
18:21.00 | Carlos_PHX | notices new political discussion on file formats. |
18:21.46 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-e6d994b987bedbde) |
18:22.11 | *** join/#asterisk mbranca (n=matteo@93-62-230-243.ip24.fastwebnet.it) |
18:22.55 | Katty | Daejeo: mewtacular. http://www.youtube.com/watch?v=6HMIRDEYoEI |
18:24.01 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
18:24.20 | Daejeo | Katty: beautiful :) |
18:24.28 | pif | get a room |
18:24.33 | Katty | quite. |
18:24.53 | Katty | pif: play nice, or get out (= |
18:26.10 | Katty | Carlos_PHX: did you hear about Obama's chief of staff wanting to put all 18-25 year olds into active duty? |
18:26.27 | Carlos_PHX | wonders if I'm being set up for a joke |
18:26.44 | jameswf | is over 25 YAY |
18:26.47 | Carlos_PHX | Surely, no democrat would be such a fascist, no? |
18:26.56 | Katty | Carlos_PHX: REF: http://www.examiner.com/x-536-Civil-Liberties-Examiner~y2008m11d6-Obamas-chief-of-staff-choice-favors-compulsory-universal-service |
18:27.00 | Zeeek | ok gad, still on that? |
18:27.21 | jameswf | examiner == enquirer? |
18:27.30 | Carlos_PHX | Holy shit, they've already started. I'm Cuban, and everything about the Obama campaign reminds me of another socialist whose rule I lived under. |
18:27.31 | pif | universal service is _not_ military service and is an excellent idea |
18:27.33 | De_Mon | I heard he's going to review all of the executive orders made by the current president and recend them all as his first act in office too. |
18:27.46 | pif | Katty: stop casting false aspersions |
18:27.47 | *** join/#asterisk rene- (n=renemend@200.34.66.137) |
18:27.50 | rene- | hey guys |
18:27.50 | [TK]D-Fender | De_Mon: "looking at". We'll see. |
18:27.56 | Katty | Carlos_PHX: it's basically 3 months of service. |
18:28.05 | Katty | Carlos_PHX: from the way i understand the news article. |
18:28.10 | Carlos_PHX | Yes, that's how it started in Cuba |
18:28.12 | pif | so what big whoop |
18:28.14 | De_Mon | you can hear all sorts of stuff, I'll wait till he actually "does" something |
18:28.16 | Carlos_PHX | My dad did 4.5 years. |
18:28.20 | jameswf | I think the US should have manditory service like all other countries |
18:28.24 | Katty | Carlos_PHX: eeesha. |
18:28.39 | Zeeek | bye again :) |
18:28.41 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
18:28.44 | Katty | Carlos_PHX: i can't even begin to imagine the ammount of religious uprising about all this. |
18:28.52 | Carlos_PHX | I think the US should have filtered speech, it works for China. |
18:29.02 | jameswf | America should also require a foriegn language like all other countries (not spanish) |
18:29.06 | De_Mon | but speaking of hearing things, someone told me at lunch the credit card companies are raising interest rates to something like 20% |
18:29.14 | De_Mon | starts googling |
18:29.16 | Katty | jameswf: i agree with the foreign language. |
18:29.30 | De_Mon | jameswf ahh hahaha not spanish? |
18:29.33 | rene- | where should i be looking at when calls from a VOIP provider drop at random times but not all the calls drop at once, it looks totally random i dont think it is an audio issue since quality is very good |
18:29.44 | Carlos_PHX | I agree everyone should know it, but that's very different from abrogating liberty to force people to learn. |
18:29.48 | Carlos_PHX | Which is just retarded. |
18:30.04 | Katty | Same with military service. |
18:30.19 | jameswf | spanish would lean towards having two national langues like canada which would be completely stupid but you should have to pick one and learn it |
18:30.35 | Carlos_PHX | checks to see if he still lives in "the land of the free." |
18:30.36 | pif | "liberty" is a red herring to keep ppl into submission, freedom is often obtained through compulsion |
18:30.40 | jameswf | I went to a private school foe k-6 and we did french |
18:30.43 | Katty | The concept is okay, but quite another when Mommy Dearest has her child taken away. |
18:31.02 | Katty | Carlos_PHX: heh. |
18:31.07 | Katty | Carlos_PHX: not since the patriot act :/ |
18:31.11 | De_Mon | jameswf guys or girls? |
18:31.12 | jameswf | Carlos_PHX: Freedom is no excuse for ignorance |
18:31.13 | Carlos_PHX | I learned French (already knew Spanish), but got stuck at that whole white flag thing. |
18:31.33 | Carlos_PHX | Freedom includes the freedom to choose ignorance. |
18:31.40 | pif | what white flag? |
18:31.48 | Carlos_PHX | If you can't choose what you put in your brain, you are not free. |
18:31.53 | De_Mon | pif surrender? |
18:32.03 | pif | who surrendered? |
18:32.09 | jameswf | Carlos_PHX: On that line of thinking we should eliminate all warning labels so those who choose to be stupid dont last long :) |
18:32.19 | Carlos_PHX | I fully agree. |
18:32.24 | Carlos_PHX | Darwin |
18:32.37 | *** join/#asterisk CrazyTux (n=brandon@user-vcauh95.dsl.mindspring.com) |
18:32.44 | jameswf | I totaly see America turning it to idiocracy (the movie) |
18:33.20 | Carlos_PHX | That movie was a view of the future. |
18:33.30 | Katty | i'm picturing more civil war. |
18:33.31 | De_Mon | white flag.. nevermind |
18:34.13 | Carlos_PHX | I mean, Idiocracy pretty much predicted Youtube. |
18:34.28 | Katty | don't think i've ever seen the movie. |
18:34.37 | De_Mon | me either |
18:34.50 | Carlos_PHX | Recommended, it's funny. |
18:34.50 | Katty | well there we go, somethign to do this week (= |
18:35.01 | De_Mon | sounds like a religulous and expelled and the like though |
18:35.03 | jameswf | Obama won a popularity contest and like most winners of American Idol people will be like "Who?" in 3 months |
18:35.15 | Katty | Terry Fator! |
18:35.18 | Katty | Paul Pots! |
18:35.23 | Katty | or was that Britian's got talent? |
18:36.02 | Katty | those two guys have skills (= |
18:36.51 | De_Mon | I was watching comedy central last night and the guy on stage said the same thing. "america doesn't vote for president, but they do vote for american idol" they should come up with a similiar show for electing presidents to get the public involved he went on to say... funny, yet very sad. |
18:37.03 | jameswf | I was floored Friday i think it was America is going to a bad place in a hand basket, Obama lays out his plan to fix it then opens up for questions where they are like ummmmmokay thats awesome but what america really want to know is what Dog are you getting |
18:37.30 | pif | jameswf: everybody needs lighter moments, you know |
18:37.44 | pif | it's not a sign of stupidity to have them |
18:37.46 | De_Mon | has he picked a dog yet? |
18:37.48 | sdaniels | How can I set * to send all the info that I usually see in an asterisk -r window to syslog? |
18:37.58 | De_Mon | sdaniels edit logging.conf |
18:38.00 | jameswf | he wants a Mut like him |
18:38.00 | Katty | De_Mon: i believe they are adopting a puppy |
18:38.05 | De_Mon | logger.conf? |
18:38.16 | De_Mon | Katty but what Kiiiind of puppy |
18:38.31 | Katty | De_Mon: Mix breed, i'd presume. |
18:38.41 | Katty | De_Mon: Peeing on American History. |
18:38.48 | De_Mon | hahaha |
18:38.56 | Katty | that should be a movie. |
18:38.58 | jameswf | unless he is going to sell the dog to the chinese for sweet and sour and make cash to fix things I DONT CARE ABOUT HIS DOG :) |
18:38.58 | De_Mon | I thouth it was because he was of mixed race |
18:39.21 | sdaniels | De_Mon: Found it, thanks. |
18:39.28 | Katty | jameswf: i'm glad he's getting a dog. |
18:39.30 | De_Mon | the first african american president has the firs dog of a mixed breed! |
18:39.33 | Katty | jameswf: those children are going to have a very hard life. |
18:39.42 | Katty | jameswf: the dog will be a comfort to them. |
18:39.53 | pif | aw, cry me a fucking river |
18:39.56 | jameswf | Obama should perpitrate the sterio type and get a pit bull |
18:39.57 | pif | poor kids |
18:40.05 | Katty | jameswf: haha. |
18:40.19 | jameswf | Katty: he could save his kids and quit |
18:40.34 | De_Mon | that would put biden in charge, right? |
18:40.47 | *** join/#asterisk lucky|aba (n=lucky@ip70-177-7-204.sb.sd.cox.net) |
18:40.51 | Katty | maybe we could put mister paul in office. |
18:41.02 | jameswf | Is bidens daughter a lesbian... oh wait that was cheney |
18:41.05 | Katty | it's nice to think about, anyway. |
18:41.14 | [TK]D-Fender | one of the few dislikes I have with Biden is some of his tech stance, esp on net neutrality. Obama = for, Biden = against |
18:41.51 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
18:43.02 | ManxPower | Time will tell. |
18:44.09 | jameswf | waits for email postage |
18:44.11 | Katty | How depressing. Time for a new OT. |
18:44.22 | [TK]D-Fender | cranks up "The Best of Asia" |
18:45.04 | RypPn | pr0n-time again tk, yum |
18:45.23 | De_Mon | O_o |
18:45.40 | [TK]D-Fender | backs away slowly.... |
18:45.49 | RypPn | now we know where the lumpy thumb came from |
18:46.24 | [TK]D-Fender | RypPn: came from a nasty sword accident |
18:46.35 | jameswf | the government should tax internet porn.... no more $$$ worries |
18:47.38 | De_Mon | why not just tax the internet! |
18:47.59 | jameswf | well taxing porn taxes 98% or the net |
18:49.36 | jameswf | http://www.examiner.com/x-668-TV-Examiner~y2008m11d10-Fox-Orders-Full-Season-Of-The-Cleveland-Show <<Cleveland has his own show... |
18:49.52 | *** join/#asterisk MindTheGap_ (n=MindTheG@189.59.133.91) |
18:50.10 | *** join/#asterisk seaq (n=seaq@190.144.113.26) |
18:52.16 | *** join/#asterisk rgsteele||work (n=rgsteele@75.147.74.137) |
18:52.45 | De_Mon | yada yada yada king of the kill got the axe too |
18:52.50 | rgsteele||work | Hey folks. Is there a way to force a particular sip peer to re-register? I've got a phone acting up, and I'd like to be able to handle that server-side. |
18:52.57 | rgsteele||work | Instead of having the user power cycle the phone. |
18:53.35 | De_Mon | i didn't think you could "request" a registration from the server |
18:53.53 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-1ad46cafb405ae51) |
18:53.57 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
18:54.16 | De_Mon | the phone should have a registration timeout and it'll just reregister when it expires |
18:54.42 | rgsteele||work | De_Mon: No way to pre-empt the timeout? |
18:55.33 | rene- | what could cause randomly dropped sip calls not all calls from the trunk drop at the same time and not all of them drop at all. audio quality is very good |
18:56.49 | De_Mon | none that I've ever heard of |
18:57.59 | *** join/#asterisk killfill (n=killfill@200.63.96.244) |
18:58.03 | killfill | hey... |
18:58.05 | hardwire | poor fill |
18:58.15 | killfill | does zopier support g729 in iax?... |
18:58.26 | killfill | cannot find the codec.. :S |
19:00.53 | killfill | ah..it requires to pay.. :P |
19:03.18 | *** join/#asterisk zchaos (n=none@CPE0080c828f609-CM0019474d28a8.cpe.net.cable.rogers.com) |
19:04.28 | zchaos | if i'm getting bad sound quality with my voip phone is it due to a poor internet connection? |
19:05.19 | rene- | yes |
19:05.30 | zchaos | it was so choppy |
19:05.31 | Katty | possibly. |
19:05.33 | zchaos | and cutting out |
19:05.51 | rene- | probably you dont have enough upstream bandwidth |
19:06.01 | Katty | [TK]D-Fender: what is your opinon of this? http://www.newegg.com/Product/Product.aspx?Item=N82E16830120239 |
19:06.50 | zchaos | what is the min upstream required |
19:07.32 | rene- | it depends on how many calls you need |
19:07.40 | Daejeo | ping [TK]D-Fender: |
19:07.46 | Daejeo | dig [TK]D-Fender: |
19:07.51 | Daejeo | host [TK]D-Fender: |
19:07.53 | zchaos | i can't even make 1 call |
19:07.54 | zchaos | its so choppy |
19:07.56 | rene- | for one call 100 kbps should be enough if not using compression, 50ks will do with g729 |
19:08.05 | rene- | but then again are u sharing the connection? |
19:08.11 | rene- | running servers, sending email |
19:08.15 | rene- | uploading stuff? gaming? |
19:08.27 | *** join/#asterisk Bilano (n=no@66.54.249.50) |
19:08.40 | zchaos | actually |
19:08.42 | zchaos | i think my bittorrent was going |
19:08.43 | zchaos | :P |
19:08.45 | [TK]D-Fender | Daejeo: Yes? |
19:08.51 | [TK]D-Fender | zchaos: SMRT |
19:08.52 | zchaos | not that you mentioned it |
19:08.59 | Bilano | Guys, I got me a small newbie issue. |
19:09.01 | zchaos | now* |
19:09.42 | Bilano | I can get incoming calls, but only if my first extension has the Direct DID filled out. I can't seem to send an incoming call into the IVR... |
19:10.38 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
19:10.46 | Bilano | If I remove the Direct DID from my extension, I get something in the full log that says I've sent the call to an invalid extension. |
19:10.53 | [TK]D-Fender | Katty: OUCH.. pricey |
19:10.54 | StephenF | Bilano pastebin your extensions.conf |
19:10.57 | Bilano | The extension it thinks I've sent the call to is the incoming DID. |
19:11.11 | Bilano | I've told the IVR to send it to extension 200, however. |
19:11.15 | Bilano | Okay, one sec. |
19:12.28 | *** join/#asterisk Keithdizzle (n=temp@static-72-86-26-36.sttlwa.fios.verizon.net) |
19:14.09 | *** join/#asterisk hi365_m (n=hi365@213.151.62.103) |
19:14.21 | Bilano | My extensions.conf is the default, it has not been touched. |
19:14.34 | Keithdizzle | can someone help me? i don't understand how to apply a patch i'm trying to use. |
19:14.56 | Katty | [TK]D-Fender: do you have a nice lil camcorder recommendation? |
19:15.01 | Keithdizzle | this patch in particular: http://bugs.digium.com/view.php?id=8824 |
19:15.29 | [TK]D-Fender | Katty: Ask yourself what it offers considering there is stuff out there thats 1/2 the price |
19:15.37 | [TK]D-Fender | Katty: And how muc you're really going to use it |
19:15.43 | Keithdizzle | i looked up how to apply the patch and i applied it and recompiled, but it doesn't seem like it did anything. |
19:15.56 | Keithdizzle | i can't find the functionalit that it should have added. |
19:16.01 | Keithdizzle | functionality. |
19:17.22 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
19:18.30 | Keithdizzle | if anyone could help me, i'd really appreciate it. |
19:19.18 | Keithdizzle | i mean, first of all, are the files on that bugs.digum.com page what i'm supposed to be applying? am i supposed to be checking it out from an svn repository? |
19:19.46 | Keithdizzle | there doesn't seem to be much of a faq on that site. |
19:20.32 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
19:20.48 | [TK]D-Fender | Keithdizzle: the bug entry says what versiont he patch is for. It may or may not be compatible with a newer release |
19:21.52 | Keithdizzle | well in the attachments, he has a file for 1.4.21.2, which is the version i'm using. |
19:22.19 | Keithdizzle | i applied that file to the 1.4.21.2 source and the output indicated that everything went well. |
19:22.39 | Katty | [TK]D-Fender: it's just for riddick, and family events. |
19:22.46 | Keithdizzle | then i did "make clean" "./configure" "make" and then "make install". |
19:23.10 | Katty | [TK]D-Fender: i've no idea what i'm looking for in a camcorder, the one listed above was the highest rated on newegg. |
19:26.08 | *** join/#asterisk CrazyTux (n=brandon@user-vcauh95.dsl.mindspring.com) |
19:26.17 | Keithdizzle | once i started up asterisk, i don't see the function listed when i do "core show functions" and any attempt to use it in extensions.conf fails. |
19:27.41 | jer | recommendations on moving a production 1.4 pbx to 1.6 this early? |
19:28.19 | *** join/#asterisk iCEBrkr (i=icebrkr@cyberdyne.org) |
19:28.33 | *** join/#asterisk CrazyTux (n=brandon@user-vcauh95.dsl.mindspring.com) |
19:28.33 | Katty | hugs iCEBrkr |
19:28.36 | iCEBrkr | yo |
19:28.42 | Keithdizzle | so uh yeah, if anyone could help me or maybe point me to what i might be doing wrong, i'd sure appreciate it... |
19:28.51 | CrazyTux | [TK]D-Fender: Hey, quick question in 1.4.2.*, using AMI (asterisk manager), can I write a command to one client connected to the server, and read from another client connected to the server at the same time, i.e. does AMI broadcast the events / dialog, or does it send back to the same client socket. |
19:29.15 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
19:29.37 | iCEBrkr | CrazyTux: It broadcasts |
19:30.02 | tzafrir_laptop | Keithdizzle, what version of Asterisk do you use now? asterisk -V |
19:30.04 | iCEBrkr | I mean, it'll spit back the results. |
19:30.21 | CrazyTux | iCEBrkr: to all clients connected to socket server with events ON? |
19:30.34 | CrazyTux | iCEBrkr: or only back to the requesting client |
19:30.46 | iCEBrkr | Yeah |
19:30.57 | CrazyTux | iCEBrkr: which one? |
19:31.07 | iCEBrkr | CrazyTux: If one client sends an Action: All the other clients will see the Events: |
19:31.20 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
19:31.42 | iCEBrkr | Speaking of AMI events. |
19:31.58 | Keithdizzle | i'm using asterisk 1.4.21.2 |
19:31.58 | iCEBrkr | I wrote a quick and dirty v2 of my callerID app with URL launching! weeee! |
19:32.03 | Keithdizzle | and i applied 1.4.21.2.patch from http://bugs.digium.com/view.php?id=8824 |
19:32.57 | Keithdizzle | it seemed to go just fine and i recompiled alright, but it didn't seem to do much. |
19:33.06 | Keithdizzle | it certainly didn't seem to add anything new... |
19:33.11 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
19:33.23 | tzafrir_laptop | Keithdizzle, what do you mean by " any attempt to use it in extensions.conf fails."? What do you do? What happens? |
19:35.50 | [TK]D-Fender | Keithdizzle: Doesn't list something you say... PASTEBIN your "list" and tell us what you think you should be seeing... |
19:37.34 | Keithdizzle | exten => 701,1,Noop(Parked Test ${CALLERID(ALL)}) |
19:37.35 | Keithdizzle | exten => 701,n,Set(CALLEDID(all)="Voice Mail" <500>) |
19:37.35 | Keithdizzle | exten => 701,n,Answer() |
19:37.43 | Keithdizzle | so that's what i've got for extensions as a test. |
19:38.07 | Keithdizzle | [Nov 10 11:36:55] ERROR[9639]: pbx.c:1564 ast_func_write: Function CALLEDID not registered |
19:38.11 | Keithdizzle | and that's what i get. |
19:39.25 | [TK]D-Fender | Keithdizzle: http://bugs.digium.com/view.php?id=8824 <-- where do you see that function listed in this link? |
19:40.07 | Keithdizzle | as far as my list, what i mean is when i do "core show functions" i don't see calledid in there. |
19:40.16 | Keithdizzle | The CALLEDID() function replaces the RemoteParty() application, it can be used |
19:40.16 | Keithdizzle | to name channels that otherwise have no set callerid such as trunks and other |
19:40.16 | Keithdizzle | internal applications (VoiceMailMain, MeetMe etc.) |
19:40.22 | Keithdizzle | gareth (reporter) |
19:40.22 | Keithdizzle | 2007-08-26 05:49 |
19:40.26 | [TK]D-Fender | Keiwhere did you even get the idea that that function name is VALID> |
19:40.40 | Keithdizzle | that post is on the bugs page. |
19:41.37 | Keithdizzle | am i misunderstanding what he's saying? |
19:41.51 | Keithdizzle | i tried RemoteParty() as well, but to no avail. |
19:42.00 | [TK]D-Fender | keithPB <- |
19:42.22 | Keithdizzle | ??? |
19:42.26 | [TK]D-Fender | Keithdizzle: and that function you saw referenced was a SUGGESTION |
19:42.29 | [TK]D-Fender | PASTEBIN |
19:43.58 | Keithdizzle | how...uhhh...do i use pastebin? |
19:44.57 | [TK]D-Fender | kei~pb |
19:44.59 | [TK]D-Fender | ~pb |
19:45.00 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
19:45.19 | Keithdizzle | ok. thanks. |
19:46.35 | Keithdizzle | so what are you looking for me to paste then? just the function list or...? |
19:49.26 | [TK]D-Fender | keithWhat you are trying to do, your lists at CLI, etc |
19:49.44 | Keithdizzle | yeah i was just doing "core show functions". |
19:50.15 | [TK]D-Fender | Keithdizzle: Include that |
19:50.19 | Keithdizzle | i thought that i should see "remoteparty" or "calledid". |
19:50.33 | [TK]D-Fender | Keithdizzle: Keeping in mind no document I see says that function should even exist |
19:50.46 | Keithdizzle | http://pastebin.com/d302d6b94 |
19:51.12 | Keithdizzle | what about the description at the top? it talks all about using remoteparty. |
19:51.17 | [TK]D-Fender | Keithdizzle: Now pastebin the doc that says that function should exist |
19:51.22 | *** join/#asterisk bazilek (n=bazil@mail.generation-p.com) |
19:51.42 | [TK]D-Fender | Keithdizzle: what "description"? Sho me the exact statement that says "this is the thing to call" |
19:52.04 | Keithdizzle | http://pastebin.com/d1f8732 |
19:52.39 | [TK]D-Fender | keithdo YOU see a reference to a FUNCTION in there?\ I sure don't |
19:54.01 | *** join/#asterisk jfarrell (n=jfarrell@unaffiliated/jfarrell) |
19:54.03 | Keithdizzle | what about step 1? i guess that would be an application and not a function? |
19:54.06 | jfarrell | greetings all |
19:54.46 | jfarrell | i have a question, this may or may not be relevant to this channel, if it isnt, please let me know |
19:54.50 | *** join/#asterisk masus (i=masus@88.248.14.186) |
19:54.53 | jfarrell | does anyone know if the possibility of interop is available for .NET with respect to the HUDclient |
19:55.11 | jfarrell | that is, can an external application tie into HUD at all and extended it? |
19:55.14 | jfarrell | *extend |
19:56.03 | [TK]D-Fender | Keithdizzle: Step 1 |
19:56.16 | [TK]D-Fender | Keithdizzle: Step 1 - follow the INSTRUCTIONS |
19:58.56 | Keithdizzle | i...guess i don't understand. i didn't see anything besides his list of current functionality. |
19:59.21 | *** join/#asterisk IPconfig (n=zig@wimax.emtelworld.com) |
20:00.16 | rwaite | can i nest $[] blocks? so can i go GotoIf($[ $[] | $[]]?this:that) |
20:00.22 | Keithdizzle | am i missing something about his description? i don't really see any instructions. |
20:00.53 | *** join/#asterisk Mark_Logan (n=mark_log@S01060014bfc81343.ed.shawcable.net) |
20:00.55 | [TK]D-Fender | Keithdizzle: exten => s,1,RemoteParty("Voicemail" <123>) <-- is this sample not blatant enough for you? |
20:01.15 | [TK]D-Fender | Keithdizzle: Its in your own pastebin for the instructions you claim to be following |
20:01.37 | [TK]D-Fender | Keithdizzle: Instead you seem to be pulling function names out of THIN AIR and ignoring the stated sample. |
20:01.43 | Keithdizzle | right right, i mean i tried that as well. |
20:02.04 | Keithdizzle | <Keithdizzle> i tried RemoteParty() as well, but to no avail. |
20:02.04 | Keithdizzle | <[TK]D-Fender> keithPB <- |
20:02.38 | *** join/#asterisk Akiyuki (n=jimi@rrcs-70-63-90-226.midsouth.biz.rr.com) |
20:02.44 | Akiyuki | Can a vonage ATA be unlocked? |
20:03.01 | Akiyuki | Or, can you put the IP address it is using in /etc/hosts |
20:03.05 | Keithdizzle | so his description says to use RemoteParty but then his post later on says that "The CALLEDID() function replaces the RemoteParty() application, it can be used |
20:03.05 | Keithdizzle | to name channels that otherwise have no set callerid such as trunks and other |
20:03.05 | Keithdizzle | internal applications (VoiceMailMain, MeetMe etc.) |
20:03.06 | Keithdizzle | " |
20:03.11 | [TK]D-Fender | Keithdizzle: Feel free to try again, this time following the instructions and showing that you are able |
20:03.28 | [TK]D-Fender | Keithdizzle: pastebin the entire entry and stop spamming the channel |
20:03.39 | Katty | dies from spam overload. |
20:03.46 | *** join/#asterisk numbshot (n=numbshot@189.205.75.92) |
20:03.50 | [TK]D-Fender | Akiyuki: Some yes, others no. You are asking for pain. |
20:04.00 | Keithdizzle | http://pastebin.com/m2c5c8194 |
20:04.44 | [TK]D-Fender | Keithdizzle: Go look in your modules folder. |
20:04.58 | Akiyuki | [TK]D-Fender: Just trying to save $50.. I guess I coul djust shell it out and save myself some trouble. Or get a wireless SIP phone. |
20:05.24 | [TK]D-Fender | Keithdizzle: http://bugs.digium.com/view.php?id=6643 <-- that paragraph didn't come from here which is where you said you patched rfom. |
20:05.41 | Katty | Dear Lord, please grant me the ability to punch people in the face over standard TCP/IP. |
20:05.57 | [TK]D-Fender | Katty: AT&T hasn't extended that far yet ;) |
20:06.17 | Daejeo | <PROTECTED> |
20:06.19 | Keithdizzle | <[TK]D-Fender> Keithdizzle: http://bugs.digium.com/view.php?id=8824 <-- where do you see that function listed in this link? |
20:06.30 | Keithdizzle | you listed...the wrong bug report. |
20:06.34 | Daejeo | <PROTECTED> |
20:07.18 | CrazyTux | Katty: :) |
20:07.27 | *** join/#asterisk qdk (n=qdk@94.191.219.74.bredband.3.dk) |
20:07.32 | [TK]D-Fender | [14:14]<Keithdizzle>can someone help me? i don't understand how to apply a patch i'm trying to use. |
20:07.34 | [TK]D-Fender | [14:15]<Keithdizzle>this patch in particular: http://bugs.digium.com/view.php?id=8824 |
20:07.38 | [TK]D-Fender | Keithdizzle: YOU said it. |
20:07.50 | [TK]D-Fender | Keithdizzle: You are massively inconsistent. |
20:08.13 | [TK]D-Fender | Keithdizzle: Get your head screwed on straight already |
20:08.18 | sdaniels | I musty be an idiot or something, can someone give me a hand getting all the console messages to syslog? |
20:08.23 | Keithdizzle | <[TK]D-Fender> Keithdizzle: http://bugs.digium.com/view.php?id=6643 <-- that paragraph didn't come from here which is where you said you patched rfom. |
20:08.36 | Keithdizzle | see...you've got the wrong bug report that you listed. |
20:08.58 | Keithdizzle | and what you pasted from me above is 8824 and not 6643. |
20:09.10 | rwaite | SIP/blah-082c6df0 is circuit-busy << would this be the same as 'CHANUNAVAIL'? |
20:09.31 | [TK]D-Fender | Keithdizzle: You'd referred to 2 patches at this point. Just go look in your modules folder. |
20:09.46 | [TK]D-Fender | rwaite: No. |
20:09.53 | Keithdizzle | ok, i'm looking. want the output? |
20:10.02 | Katty | anyone know a better gallery than gallery v2.2? |
20:10.04 | [TK]D-Fender | Keithdizzle: feel free |
20:10.29 | Keithdizzle | http://pastebin.com/d53f49a32 |
20:11.01 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
20:11.07 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
20:11.17 | Keithdizzle | you know, i'm just looking for help. if you would like to see more people use asterisk, maybe you could try being a bit friendlier. i already said i've never done this before and the site says to come here for help, not to talk to assholes. |
20:12.01 | [TK]D-Fender | Keithdizzle: Doesn't look installed... |
20:12.14 | sdaniels | nevermind, I got it. thanks |
20:12.34 | Keithdizzle | yeah i know, i mean that's what i'm asking. his .patch file didn't seem to do much and i am trying to figure out if i just don't understand how to apply this patch or what. |
20:12.57 | [TK]D-Fender | Keithdizzle: Normally the pathc mods your source in place and you need to recompile to get the new apps |
20:12.59 | italorossi | anyone use py-asterisk manager interface? |
20:13.10 | [TK]D-Fender | Keithdizzle: Have you done this? You should see it in menuselect |
20:13.40 | Keithdizzle | yeah i went to recompile, but it's not in menuselect. |
20:14.17 | [TK]D-Fender | Keithdizzle: I'd verify that your source was actually patched. |
20:14.46 | Keithdizzle | well if i try to apply the patch again, it sees it's already been patched. is there something else i can do to verify it? |
20:15.00 | [TK]D-Fender | Keithdizzle: Just trash your whole folder and extract again |
20:15.06 | Keithdizzle | ok. |
20:16.00 | devilsoulblack | hi anyone have this error http://lists.digium.com/pipermail/asterisk-r2/2008-November/000138.html |
20:16.08 | jfarrell | ahh kernel patching, memories |
20:16.15 | jfarrell | no longers users linux |
20:16.20 | seanbright | he's not patching the kernel |
20:16.22 | jfarrell | *uses |
20:16.25 | seanbright | he's patching asterisk |
20:16.29 | jfarrell | ahh |
20:16.38 | Keithdizzle | ok, i've deleted the whole /usr/src/asterisk-1.4.21.2 directory and decompressed the whole tar again. |
20:16.38 | jfarrell | i saw menuselect and patching - i asumed |
20:16.42 | jfarrell | my apologies |
20:16.49 | seanbright | there's a menuselect in asterisk as well |
20:16.52 | seanbright | i forgive you |
20:16.52 | seanbright | ;) |
20:16.57 | jfarrell | sweet |
20:17.15 | [TK]D-Fender | devilsoulblack: Here's an error... your channel ranges don't match at all |
20:17.33 | Keithdizzle | so this is the file i should be using to patch, am i correct? http://bugs.digium.com/file_download.php?file_id=20203&type=bug |
20:17.52 | seanbright | Keithdizzle: what bug is that attached to? |
20:18.00 | [TK]D-Fender | Keithdizzle: what are you running? |
20:18.07 | Keithdizzle | sean: this one: http://bugs.digium.com/view.php?id=8824 |
20:18.18 | Keithdizzle | you mean what version? |
20:18.24 | [TK]D-Fender | os asterisk |
20:18.27 | [TK]D-Fender | of |
20:18.33 | seanbright | Keithdizzle: cd /path/to/asterisk/src ; wget -O - "http://bugs.digium.com/file_download.php?file_id=20203&type=bug" | patch -p1 |
20:18.40 | devilsoulblack | [TK]D-Fender, the telco give that info |
20:19.01 | Keithdizzle | i'm using linux, centos 5 i believe. |
20:19.13 | [TK]D-Fender | devilsoulblack: dahdi_cfg is souwing a completely mismatched range from chan_dahdi.conf |
20:19.21 | [TK]D-Fender | Keithdizzle: ASTERISK <------- |
20:19.30 | seanbright | 1.4.21.2 |
20:20.08 | Keithdizzle | here's the output of me patching: http://pastebin.com/d20506fc |
20:20.12 | [TK]D-Fender | Ok, looks fin so far if thats the ccase |
20:20.16 | *** join/#asterisk feeds_ChZ (n=chatzill@85-135-235-105.adsl.slovanet.sk) |
20:20.23 | seanbright | Keithdizzle: looks perfect |
20:20.26 | Keithdizzle | [root@tobias asterisk-1.4.21.2]# asterisk -V |
20:20.26 | Keithdizzle | Asterisk 1.4.21.2 |
20:21.03 | [TK]D-Fender | Keithdizzle: ok, and then you did "./configure" , "make menuconfig" (looked for the app/function?), "make" make install"? |
20:21.03 | Daejeo | <PROTECTED> |
20:21.10 | [TK]D-Fender | Daejeo: No idea |
20:21.18 | Keithdizzle | that's what i'm working on right now. |
20:21.18 | Qwell | Daejeo: why not just do it locally, and bypass the provider? |
20:21.19 | seanbright | what app/func are you expecting to see? |
20:21.37 | Keithdizzle | i believe it should be app_remoteparty, but i'm not quite sure. |
20:21.45 | devilsoulblack | [TK]D-Fender, i change on chan_dahdi.conf the same number channels and have the same issue |
20:21.52 | seanbright | Keithdizzle: not based on that patch |
20:21.54 | Daejeo | Qwell: it did it, but sometimes trunk fail |
20:22.18 | Daejeo | i want to handle fail over |
20:22.43 | Daejeo | any educated idea? |
20:23.12 | Keithdizzle | seanbright: ok, so am i using the wrong patch? are the files attached to that bug report even the actual patches? |
20:24.27 | Keithdizzle | i have no idea how to use their bug site and their guidelines page doesn't have much information. |
20:24.56 | seanbright | hold on |
20:25.45 | seanbright | it looks like it adds options to the Dial() application |
20:25.54 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
20:25.58 | seanbright | the f, I, and N options |
20:26.07 | seanbright | once you get it all built and installed, do a 'core show application Dial' |
20:27.10 | Keithdizzle | ok...i see those. if that's all it did, what the hell is all this discussion about RemoteParty and CALLEDID? |
20:27.16 | lesouvage | Is it true that with the last version of sox (SoX v14.1.0 ) sox -m came in the place of soxmix or is this bare nonsens. (sorry, kind of ot but surely asterisk related) |
20:27.31 | seanbright | Keithdizzle: it does more... but that was the first thing i noticed. |
20:28.24 | Keithdizzle | ok, i mean i know c and i can see that it doesn't even reference either of those functions, but i know very little about diff and thought maybe i just didn't understand what it was doing. |
20:28.37 | seanbright | Keithdizzle: i think func_connectedline.so is what you are interested in |
20:28.53 | seanbright | Keithdizzle: core show function CONNECTEDLINE |
20:29.15 | Keithdizzle | ok..i get some output from that.' |
20:29.35 | Keithdizzle | http://pastebin.com/d7b249cf6 |
20:29.40 | seanbright | that function was added by the patch |
20:29.47 | seanbright | so i assume it's relevant to what you are looking for |
20:29.56 | Keithdizzle | ok, thanks. i bet that's what it's been renamed to. |
20:29.58 | seanbright | brb |
20:32.27 | beek | [TK]D-Fender: Sangoma's wanfig_dahdi created an interesting chan_dahdi.conf file. Under "Channels" it has "echocancel=yes, echocancelwhenbridged=yes", yet under the individual spans it added "echocancel=no". As a result, Asterisk throws warnings when reloading chan_dahdi. Which should it be? echocancelwhenbridged=no under '[channels]' or echocancel=yes in the span? |
20:33.14 | [TK]D-Fender | beek: Your choice |
20:33.38 | *** join/#asterisk seaq (n=seaq@190.144.113.26) |
20:34.06 | beek | [TK]D-Fender: So bridging these PRIs are unlikely to require echocancelling? |
20:34.36 | [TK]D-Fender | beek: generally not on bridge. Otherwise yes, they almost certainly will |
20:35.05 | beek | [TK]D-Fender: Alrighty -- I'll turn it off for the PRI-PRI bridge and on for the channel bank. Thanks! |
20:35.16 | [TK]D-Fender | beek: np, and glad to hear its going well |
20:35.31 | beek | [TK]D-Fender: I'm having a blast working on this. |
20:39.07 | seanbright | and back |
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20:40.02 | Keithdizzle | hey sean, i'm still having trouble, but i'm going to work on it for a few minutes. thanks for being so helpful. |
20:40.23 | seanbright | Keithdizzle: no sweat. let me know if you need more help. |
20:41.28 | Keithdizzle | seanbright: ok, i've got it to work, but i'm getting errors on the console and i'm not sure if i should be concerned about it. |
20:41.36 | seanbright | pastebin them |
20:41.45 | Keithdizzle | yep, one second. |
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20:43.17 | Keithdizzle | http://pastebin.com/d4d96b49d |
20:43.43 | seanbright | i don't think that is something you need to be concerned with |
20:43.46 | seanbright | (but i might be wrong) |
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20:46.08 | seanbright | Keithdizzle: i think i am wrong, but i don't know how to resolve that particular issue. |
20:46.49 | Keithdizzle | yeah no problems there. i just needed to figure out how to use this. |
20:47.35 | Keithdizzle | thanks so much. |
20:47.40 | seanbright | nods |
20:48.19 | Keithdizzle | and [TK]D-Fender, i guess i'll thank you for your help as well, fraught as it was with berating statements and the general attitude of a complete douche bag. |
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20:52.55 | [TK]D-Fender | Keithdizzle: I refarined from berating, just pointed out the incositant bits as you were working off 2 different sources. |
20:53.19 | [TK]D-Fender | Keithdizzle: We've come prety far so far, just a bit of a bump start |
20:53.43 | Keithdizzle | yeah that was an accident. i'm still learning to use this bug system and those posts are linked. |
20:53.51 | [TK]D-Fender | Keithdizzle: Do you see an app or func being compiled yet? |
20:53.57 | Keithdizzle | yeah, i got it working. |
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20:54.09 | [TK]D-Fender | Keithdizzle: how functional, and using what models? |
20:54.52 | Keithdizzle | following the posts, it went from application RemoteParty() to function CALLEDID to function LINEID and finally it's now called CONNECTEDLINE. |
20:55.15 | [TK]D-Fender | Keithdizzle: Story gets better every time I hear it... can't wait to see what its called next week :) |
20:55.29 | [TK]D-Fender | Keithdizzle: So what phones had you tested this with? |
20:55.49 | Keithdizzle | well it doesn't work with my grandstream, but it worked with a polycom. |
20:55.58 | *** join/#asterisk synchris (n=synchris@athedsl-86032.home.otenet.gr) |
20:56.20 | Keithdizzle | albeit with a shit ton of these errors: |
20:56.20 | Keithdizzle | [Nov 10 12:42:04] WARNING[21099]: chan_sip.c:3707 sip_write: Asked to transmit frame type 64, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4) |
20:56.33 | [TK]D-Fender | Keithdizzle: I'm not surprised at the former.... Polycom, cisco, Linksys, and Aastra have the best bets... Snom is probably a decent bet. |
20:57.11 | [TK]D-Fender | ~gs |
20:57.11 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
20:57.24 | Keithdizzle | yeah i had to get it working on polycom phones else i'm not going to be eating soon. |
20:58.30 | [TK]D-Fender | ~cpid |
20:58.31 | jbot | [~cpid] Called-Party ID is possible with * using patches on Mantis. See : http://bugs.digium.com/view.php?id=8824 |
20:58.34 | Keithdizzle | jbot: a digium rep recommended grandstream when we first looked into this, but it only took two of them to realize they weren't all that nice. |
20:58.57 | [TK]D-Fender | Keithdizzle: Get us a name... we have people on the inside who can deal with him ;) |
20:59.23 | [TK]D-Fender | Keithdizzle: And don't talk with the bot... its too early for that kind of humour :) |
21:00.11 | Keithdizzle | is jbot an actual bot? |
21:00.18 | file | yes. |
21:00.19 | [TK]D-Fender | ~areyouadog ? |
21:00.20 | jbot | Bark! Bark! |
21:00.23 | file | jbot: botsnack |
21:00.23 | jbot | aw, gee, file |
21:00.31 | [TK]D-Fender | file: DAMN YOU! |
21:00.34 | Keithdizzle | wow...that's a lot of work put into something not that exciting. |
21:00.53 | rob0 | It excites me ... shrug |
21:01.07 | [TK]D-Fender | Keithdizzle: jbot is quite useful. |
21:01.17 | [TK]D-Fender | file: How goes? |
21:01.27 | Keithdizzle | i've been working on multivalue database programming for 3 years. that's about as unexciting as it gets. |
21:01.39 | file | [TK]D-Fender: not bad! yourself? |
21:01.40 | Keithdizzle | i guess i would be happy to work on an irc bot instead. |
21:02.10 | [TK]D-Fender | Keithdizzle: You can always distract yourself on the "All snail-racing" channel, or the "Astro-turf growing competition". |
21:02.29 | [TK]D-Fender | file: Getting by... few med situations to clean up I'm hoping don't haunt me down the line. |
21:02.39 | file | [TK]D-Fender: >_< |
21:02.52 | [TK]D-Fender | Keithdizzle: Well jbot is done and well trained with all sorts of useful info. |
21:03.26 | [TK]D-Fender | file: that mal-healed cut I hope to have surgery to correct, fire off all the dental work before year end, etc... |
21:03.39 | *** join/#asterisk klictel (n=klictel@nat/digium/x-f44a2ca44fc16edc) |
21:04.03 | [TK]D-Fender | file: fun, fun, fun... |
21:04.16 | file | [TK]D-Fender: eep |
21:04.25 | file | klictel: Claude! |
21:04.27 | [TK]D-Fender | file: All good news if they're able to... |
21:04.35 | klictel | present |
21:04.39 | [TK]D-Fender | past |
21:04.43 | klictel | heh |
21:04.58 | klictel | i'm sitting jared class in huntsville |
21:05.30 | file | klictel: I thought so, the digium hostname gave it away :D |
21:05.44 | klictel | oh well |
21:06.42 | klictel | file: any snow yet in your corner? |
21:06.52 | file | klictel: thankfully no! but lots of rain |
21:07.06 | klictel | we had some in mtl a couple of weeks back |
21:07.18 | klictel | depressing.... i started crying |
21:07.20 | [TK]D-Fender | klictel: Lol... a tiny bit of slush 1 night... |
21:07.39 | klictel | well yeah maybe but still IT WAS SNOW IN OCTOBER |
21:07.46 | [TK]D-Fender | klictel: Nothing to complain abaout. Now if ew get smashed like last year, thats another matter... |
21:07.48 | klictel | next year it will start in july |
21:08.09 | klictel | i am out to get a snow blower next week |
21:08.30 | klictel | depressing |
21:08.40 | [TK]D-Fender | klictel: I got my winter tires done last friday... jsut about set... took down the shovel for the car this morning... |
21:09.03 | [TK]D-Fender | klictel: I want the green Christmas we had 3 years ago... |
21:09.12 | klictel | i still have to put on my tires... and my wife's car |
21:09.17 | [TK]D-Fender | takes a few more cans of hairspray outside... |
21:09.24 | klictel | heh |
21:11.05 | [TK]D-Fender | 20 mins to checkout time... |
21:11.42 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
21:12.45 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
21:15.55 | Katty | 1hr20 for me :< |
21:17.51 | Keithdizzle | i have another question. |
21:18.10 | Keithdizzle | is there any way when i park a call to have the caller who parked it know which extension it was parked to? |
21:18.22 | *** join/#asterisk seaq (n=seaq@190.144.113.26) |
21:19.20 | *** join/#asterisk RMod_ (n=nicolasj@unaffiliated/rmod) |
21:20.01 | [TK]D-Fender | Keithdizzle: it gets read back to you when you park it. |
21:20.13 | [TK]D-Fender | Keithdizzle: Which is why you need to be doing it in an attended transfer |
21:20.19 | Keithdizzle | thanks. |
21:20.52 | Keithdizzle | well i guess what i mean is if there is any way for the dialplan to know. |
21:21.32 | [TK]D-Fender | Keithdizzle: Well an exten is generated for the parked call... so technically yes its trackable |
21:22.26 | devilsoulblack | openr2 need asterisk-addon ? |
21:22.40 | Katty | pouts at [TK]D-Fender |
21:22.43 | Katty | [TK]D-Fender: can i go home wif you? |
21:22.51 | Keithdizzle | ok, would i just try and list extensions to track it or something? |
21:22.57 | *** join/#asterisk Filippino (n=Filippin@151.61.84.181) |
21:23.15 | [TK]D-Fender | Katty: Sure... c'mon up :) |
21:23.34 | [TK]D-Fender | Keithdizzle: Well what exactly are you trying to accomplish? |
21:23.52 | Filippino | hi. |
21:24.02 | Filippino | just a problem with a grandstream bt200 |
21:24.09 | Filippino | anyone can help me pls ? |
21:24.38 | Keithdizzle | basically my problem is that if a call comes in and gets parked, i need to display the original incoming caller id on the display of the user who actually picks up the parked call. |
21:24.52 | Keithdizzle | and not the extension that parked it? |
21:25.14 | [TK]D-Fender | Keithdizzle: IIRC thats what you get... |
21:25.35 | Keithdizzle | huh? |
21:25.40 | [TK]D-Fender | Keithdizzle: Even through its an attended transfer it keeps the original CID. |
21:26.16 | *** join/#asterisk baliktad (i=baliktad@c-24-17-254-250.hsd1.wa.comcast.net) |
21:26.31 | [TK]D-Fender | Keithdizzle: AH, but you pick up the call and you want it displayed BACK... |
21:26.35 | Filippino | i have my asterisk running with a problem. sometimes I start a phonecall with my BT200 than when I hangup, asterisk doesn't ... after a lot lot lot progblems |
21:26.45 | [TK]D-Fender | Keithdizzle: yeah, that'd be trick, but possible with the CPID patch |
21:26.57 | [TK]D-Fender | tricky* |
21:27.04 | baliktad | I have a provider who wants to send me calls from an entire IP range (a /24) - how can I specify the SIP peer to accept calls from a whole range of IP's? |
21:27.15 | Keithdizzle | ok, let me look up the cpid patch. thanks fender. |
21:27.24 | [TK]D-Fender | Keithdizzle: You'd have to build your own custom exten to do the pickup. |
21:27.36 | [TK]D-Fender | Keithdizzle: thats the patch you just spent all this time installing |
21:27.52 | [TK]D-Fender | Keithdizzle: this is what is required to see it when you pick them up. |
21:28.27 | Keithdizzle | right ok, sorry. i understand what you meant now. |
21:28.38 | [TK]D-Fender | Keithdizzle: You'd ahve to do some real scripting though. Scan for the channel parked on the lot # you are calling for, set the CPID, then do the pickup in a local channel. Rather complex, but possible |
21:29.07 | [TK]D-Fender | Keithdizzle: And it's look nasty from a CDR perspective. |
21:29.11 | [TK]D-Fender | it'd |
21:29.19 | [TK]D-Fender | Ok, well its checkout time here... back later. |
21:29.23 | [TK]D-Fender | good luck to all. |
21:31.25 | diegows | any spa 3102 (firmware 5.1.7) user here? |
21:33.23 | *** join/#asterisk kusznir (n=kusznir@isg-grad-02a.eecs.wsu.edu) |
21:33.57 | kusznir | Hi all: I've got a hardware question: I need to acquire a hardware phone (or something similar) that will function somewhat like an intercom. Basically, I need a speakerphone that will auto-answer. |
21:34.16 | *** join/#asterisk shinao1 (n=shinao1@62.173.48.42) |
21:35.06 | kusznir | It needs decent mic pickup as well. It will be used for announcements followed by comments from whomever is near it. I'm currently doing it with a basic softphone, but the PC I have available is busy doing other stuff and the audio is frequently garbled and such. |
21:35.26 | kusznir | Oh, and it would be nice if it was less than $200, preferably in the $100 range. |
21:37.16 | baliktad | I use the SPA-9xx phones, you can set a SIP header to have it auto answer and playback over the speakerphone |
21:38.08 | baliktad | I've set it up so users can dial the phone directly (extension 2xx) to have the phone ring; to page the phone, they dial 8+extension |
21:41.36 | *** part/#asterisk seaq (n=seaq@190.144.113.26) |
21:42.38 | moy | devilsoulblack: no, it's not needed |
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21:53.47 | *** mode/#asterisk [+o lmadsen] by ChanServ |
21:55.50 | Katty | lmadsen: snow :< |
21:58.39 | kusznir | baliktad: thanks! |
21:59.01 | *** join/#asterisk riddlebox (n=adfad@75-128-170-26.static.stls.mo.charter.com) |
21:59.23 | riddlebox | does anyone use vicidial? does it work well with zap channels? |
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22:08.12 | *** join/#asterisk littlepinkdot (n=thedot@69.7.43.20) |
22:08.24 | littlepinkdot | What would be the recomended echo cancellor? |
22:08.37 | JT | hardware |
22:08.56 | littlepinkdot | I'm currently using OSLEC but finding that it doesn't autostart/load before the zaptel drivers do so it doesn't work 90% of the time. |
22:08.57 | tzafrir_laptop | oslec |
22:09.13 | tzafrir_laptop | littlepinkdot, what do you mean? |
22:09.45 | tzafrir_laptop | this should be fixed by a proper installation, I believe . |
22:09.54 | littlepinkdot | On system boot, zaptel loads, oslec kernel module loads, freepbx/asterisk/etc loads. Since zaptel loaded before oslec did, oslec can't perform the function its designed to since it can't insert itself into the kernel. |
22:10.11 | tzafrir_laptop | what is the output of: modinfo zaptel | grep ^depends |
22:10.22 | littlepinkdot | modinfo zaptel | grep ^depends |
22:10.24 | littlepinkdot | Err |
22:10.29 | littlepinkdot | crc-ccitt |
22:10.44 | tzafrir_laptop | find /lib/modules -name oslec.ko |
22:11.21 | littlepinkdot | My copy resides in /usr/src/oslec/kernel/oslec.ko, should I move it? |
22:11.22 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
22:11.41 | tzafrir_laptop | that's not a proper install |
22:11.59 | tzafrir_laptop | find /lib/modules -name zaptel.ko |
22:12.00 | littlepinkdot | I patched/compiled as the guide suggested |
22:12.18 | littlepinkdot | /lib/modules/2.6.18-8.el5/misc/zaptel.ko |
22:12.18 | littlepinkdot | /lib/modules/2.6.18-53.1.21.el5/misc/zaptel.ko |
22:12.51 | tzafrir_laptop | littlepinkdot, kernel modules should be under /lib/modules/`uname -r` |
22:13.06 | littlepinkdot | Hmm |
22:13.07 | tzafrir_laptop | probably just put it in the same misc/ subdirectory |
22:13.13 | tzafrir_laptop | and then run depmod |
22:13.33 | *** join/#asterisk Akiyuki (n=jimi@rrcs-70-63-90-226.midsouth.biz.rr.com) |
22:13.46 | tzafrir_laptop | once you do that, modinfo would show that zaptel depends on oslec |
22:13.56 | Akiyuki | Is it possible to tell asterisk to "call xxx , then once connected, transfer/confernce to yyyy" ? |
22:14.12 | Akiyuki | I do not want to build this app in perl Net::SIP modules if possible |
22:14.48 | littlepinkdot | tzafrir_laptop, still shows crc-ccitt |
22:15.06 | tzafrir_laptop | only that? |
22:15.11 | littlepinkdot | Correct |
22:15.15 | *** join/#asterisk hohum (n=dcorbe@206.71.169.115) |
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22:17.30 | tzafrir_laptop | did you run depmod? |
22:17.40 | littlepinkdot | Yes |
22:17.54 | tzafrir_laptop | uname -r |
22:19.11 | littlepinkdot | 2.6.18-53.1.21.el5 |
22:19.11 | littlepinkdot | [root@voip1 oslec]# find /lib/modules -name oslec.ko |
22:19.11 | littlepinkdot | /lib/modules/2.6.18-53.1.21.el5/misc/oslec.ko |
22:19.38 | tzafrir_laptop | modinfo /lib/modules/2.6.18-53.1.21.el5/misc/oslec.ko | grep ^vermagic |
22:20.15 | kfife | anyone want to weigh in on whether to use the new g711 codec in 1.6? How much more expensive is it? Docs say 'cleaner' but what practical benefit does it offer? More reliable inband DTMF for example? |
22:20.19 | littlepinkdot | modinfo /lib/modules/2.6.18-53.1.21.el5/misc/oslec.ko | grep ^vermagic |
22:20.33 | littlepinkdot | vermagic: 2.6.18-53.1.21.el5 SMP mod_unload 686 REGPARM 4KSTACKS gcc-4.1 |
22:26.21 | *** part/#asterisk galeras (n=galeras@190.159.51.107) |
22:26.35 | littlepinkdot | Any ideas tzafrir_laptop? =/ |
22:28.11 | tzafrir_laptop | any chance you could test unloading zaptel and loading it (to see if modprobe actually somehow pulls zaptel)? |
22:28.28 | tzafrir_laptop | err... pulls oslec |
22:30.16 | littlepinkdot | FATAL: Module zaptel is in use. when trying to remove |
22:30.29 | *** join/#asterisk marl (n=marl@78.149.78.173) |
22:30.41 | *** part/#asterisk marl (n=marl@78.149.78.173) |
22:31.05 | Akiyuki | Is it possible to tell asterisk to "call xxx , then once connected, transfer/confernce to yyyy" ? |
22:32.51 | seanbright | Akiyuki: yes. call files. |
22:34.10 | Akiyuki | seanbright: Do you have to set cronjobs on them? |
22:34.20 | Akiyuki | seanbright: Trying to build a real time auto-dialer for my CRM. |
22:34.38 | seanbright | Akiyuki: do you want to call people on a scheduled basis? |
22:34.48 | Akiyuki | no |
22:34.48 | seanbright | or in an on-demand fashion |
22:34.51 | Akiyuki | on demand |
22:35.06 | seanbright | then you could just write a script to generate the call file |
22:35.12 | Akiyuki | I want it to Call the customer at xxx then imediately have the phone ringing at yyyy, not whent hey answer. |
22:35.23 | seanbright | or you could connect to asterisk via AMI and send an Originate |
22:35.23 | Akiyuki | Ok, so as soon as i place the .call file in the directory, asterisk will read and dial it? |
22:35.29 | seanbright | Akiyuki: correct |
22:35.31 | Akiyuki | AMI? |
22:35.35 | Akiyuki | Is that a perl module? |
22:35.38 | seanbright | Asterisk Manager Interface |
22:35.40 | seanbright | no |
22:35.46 | seanbright | there is a perl module that helps you connect to AMI |
22:35.53 | *** join/#asterisk zchaos (n=none@CPE0080c828f609-CM0019474d28a8.cpe.net.cable.rogers.com) |
22:35.53 | seanbright | take a look at the wiki for more details |
22:35.58 | seanbright | http://www.voip-info.org/ |
22:36.05 | Akiyuki | Ah ok. trying to do this in php. Might be better off to do it w/ creating the .call files? |
22:36.15 | seanbright | Akiyuki: yeah |
22:36.48 | Akiyuki | Man, that is sweet. I have been trying to do it with Net::SIP and getting no where fast :P |
22:36.49 | zchaos | can anyone tell me of any good canadian VOIP service providers? i was going to use acanac... but i heard they only allow you to call 50 different telephone numbers before they force you to go to the business package |
22:37.02 | zchaos | what is that crap |
22:37.10 | seanbright | ~itsp |
22:37.11 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
22:37.23 | seanbright | zchaos: /msg jbot ~itsplist-ca |
22:37.31 | zchaos | thanks |
22:37.36 | zchaos | is PSTN sip? |
22:37.42 | seanbright | Akiyuki: yeah, call files are trivial in comparison. |
22:37.50 | rob0 | ~pstn |
22:37.50 | jbot | from memory, pstn is Public Switched Telephone Network, or "please stop the nonsense" |
22:38.55 | Akiyuki | seanbright: Can a call file have multiple numbers to attempt? Reading the wiki abuot them now but do not see that |
22:39.13 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
22:39.25 | seanbright | Akiyuki: you can have the call file call a local channel, which in turn can dial multiple people |
22:39.26 | zchaos | do all VOIP service providers allow you to only call 50 different numbers a month? |
22:39.29 | zchaos | before charging you extra? |
22:39.34 | seanbright | Akiyuki: local channels are also detailed on the wiki |
22:39.40 | rene- | riddlebox: i think it was designed with zap channels in mind |
22:41.48 | littlepinkdot | tzafrir_laptop, think I just have to get oslec to load before zaptel does, right now it loads via modprobe.conf but that gets read after zaptel is already loaded. |
22:42.49 | *** join/#asterisk chazz (n=chazz@nat/digium/x-d5445bf82d964ad5) |
22:46.33 | zchaos | do all VOIP service providers allow you to only call 50 different numbers a month? |
22:46.50 | *** join/#asterisk Bilano (n=no@66.54.249.50) |
22:46.54 | Bilano | Howdy gents. |
22:47.01 | littlepinkdot | zchaos, where'd you hear that? |
22:48.37 | zchaos | acanac said that |
22:48.52 | zchaos | thats what i said |
22:48.54 | zchaos | i was like wtf |
22:48.55 | zchaos | tahts gay |
22:49.46 | *** join/#asterisk ManxPower (n=manxpowe@123.sub-75-203-80.myvzw.com) |
22:54.07 | littlepinkdot | Blegh...anyone have experience with OSLEC? |
22:55.02 | Yourname | LOL zchaos that's hilarious! |
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22:57.15 | zchaos | i'm not kidding |
22:57.18 | zchaos | i ordered it then if ound out later |
22:57.22 | zchaos | and the first thing i said |
22:57.29 | zchaos | was... cancel me and refund my shit you never told me that |
22:58.08 | zchaos | wtfff do i do |
22:58.09 | zchaos | http://www.babytel.ca/content_pages/babyPLANS.html |
22:58.15 | zchaos | i need 2 phone lines |
22:58.30 | zchaos | thats $40 a month for 2 lines |
22:58.40 | zchaos | the canadian village package |
23:03.46 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
23:05.33 | ManxPower | littlepinkdot: I have heard good things about OSLEC, but I've never used it. I've use the HPEC before, but these days I just stick to Tellabs hardware EC devices. You plug them in between the telco and Asterisk and never have echo again. |
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23:12.20 | `Sean | Hey, does anyone know where i can get a 1900 number with unlimted channels or lke 10 chanels? |
23:13.51 | ManxPower | `Sean: I doubt you'll fine ANYONE that sells 1-900 VoIP service. |
23:14.01 | ManxPower | Heck, most telcos won't even offer the service. |
23:14.09 | rene- | manx what do u think about digium hardware echo canceller? |
23:14.34 | jer_ | anyone recommend (or not) upgrading a production 1.4 * box to 1.6 ? |
23:14.35 | ManxPower | rene-: "It works good enough for most people" is what I've heard. |
23:14.47 | *** join/#asterisk atlas95 (n=cyril@unaffilated/atlas95) |
23:14.48 | ManxPower | jer_: did you read the upgrade notes? |
23:14.52 | jer_ | ManxPower, yes |
23:15.06 | ManxPower | WOW! Someone finally read them!!!!!! |
23:15.09 | jer_ | =] |
23:15.14 | jer_ | but it doesn't tell me what i want to know |
23:15.24 | ManxPower | jer_: 1.6.x is far too new for me to put it into production. |
23:15.32 | jer_ | it's not about what's changed, i want to know from people actually using it, whether or not they'd recommend upgrading a production 1.4 box to 1.6 |
23:15.39 | jer_ | ManxPower, yeah that's my hangup too |
23:16.02 | ManxPower | jer_: I figure I'll wait until Digium upgrades their corporate PBX to 1.6. That would demonstrate Digium's confidence in 1.6 |
23:16.14 | jer_ | ah, heh, good point =] |
23:16.35 | ManxPower | Right now it's more like "Hey! We have this great new release! Use it! No, we don't use it, but you should!" |
23:17.08 | StephenF | What do you guys use to test your SIP connectivity to your ITSP? Users are complaining about call quality, trying to determine the problem |
23:17.29 | ManxPower | StephenF: many of us don't even use an ITSP. |
23:17.48 | StephenF | ManxPower due to quality? |
23:17.56 | rene- | StephenF: reliability |
23:17.59 | rene- | quality |
23:18.05 | rene- | sometimes cost |
23:18.08 | ManxPower | StephenF: The internet is not reliable to send calls over for my clients. |
23:18.14 | ManxPower | reliable enough, that is. |
23:18.16 | Mark_Logan | True. |
23:18.20 | rene- | it is almost there |
23:18.23 | rene- | but not quite |
23:18.31 | Mark_Logan | Call me when it is okay :P |
23:18.33 | ManxPower | rene-: apparently your users have low expectations. |
23:18.38 | Mark_Logan | lol |
23:18.41 | Daejeo | anyone from austraila? |
23:18.51 | Daejeo | anyone from Australia? |
23:19.01 | rene- | actually they arent |
23:19.12 | ManxPower | VoIPoInternet seems to be about as reliable as a cell phone -- most of the time. |
23:19.13 | rene- | but sometimes they are cheap |
23:19.49 | rene- | if u have good connectivity like > t1 on both ends should be ok, as long as both ends are properly setup for QoS |
23:20.15 | ManxPower | MY problem with VoIPoInternet is if sometime breaks who are you going to call? It could be your ISP, but the problem could be with ANY network between you and your ITSP and since you are not their customer they are not going to fix it for you. |
23:20.15 | rene- | but it does not beats TDM on quality or reliabilty only cost |
23:20.20 | `Sean | damn it this is hard finding a 900 number with like 10 channels |
23:20.28 | littlepinkdot | Guess I'm back to my orignal question...how can I get oslec to load before zaptel so it actually loads? |
23:20.29 | ManxPower | Yo can't do QoS over the public internet. |
23:20.35 | rene- | ManxPower i know |
23:20.46 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
23:20.46 | rene- | but at least you should make sure your upstream bandwidth is managed |
23:20.56 | *** join/#asterisk steliosk (n=Stelios@athedsl-318864.home.otenet.gr) |
23:21.05 | ManxPower | rene-: and their upstream, and their upstream and their upstream. |
23:21.09 | rene- | hehe |
23:21.21 | rene- | yes you can only pray it is going to be ok all the time |
23:21.23 | rene- | and it wont |
23:21.32 | rene- | if u have fiber |
23:21.44 | StephenF | so for a small office with 5 channels, what would your bring in as far as non-voip |
23:21.44 | rene- | at the two points then voip works very well |
23:22.07 | *** join/#asterisk WHYS (i=lpfm@137.28.94.209) |
23:22.14 | ManxPower | One of our carriers (XFone) is having trouble talking to Charter.net, but they go thru some 3rd ISP. Nobody will even admit thereis'a problem. My users don't want to hear "well you can't VPN in because someone we have no influince over has a broken network). |
23:22.16 | StephenF | 5 POTS lines? |
23:22.32 | ManxPower | They would lynch me if I told them that about voice. |
23:23.02 | ManxPower | StephenF: in most markets 8 channels is the min on a PRI to have about the same cost as 8 analog lines. |
23:23.32 | rene- | StephenF: u can try with voip, it works to some extent, it is cheap, it may be right for your application/costumer |
23:23.33 | ManxPower | If you MUST use analog, I'd make sure failover happened between the POTS and the VoIP channels. |
23:24.02 | ManxPower | But really, if you can't afford an 8 channel PRI then you really can't afford my consulting services. |
23:24.09 | StephenF | we are on voip now, its been OK but every once in a while we have quaility issues |
23:24.25 | StephenF | what does an 8 channel PRI for on average? |
23:25.00 | StephenF | cost on average* |
23:25.03 | Mark_Logan | costwise? |
23:25.06 | Mark_Logan | ah |
23:25.17 | ManxPower | StephenF: $300/month to $1,200/month |
23:25.28 | Mark_Logan | So, pocket change. |
23:25.37 | ManxPower | The correct answer is "there is no average, each market is different." |
23:25.41 | *** join/#asterisk Mad|Cow (n=user@static-72-94-249-58.phlapa.fios.verizon.net) |
23:25.42 | StephenF | ok, now what if the customer doesnt need 8 channels? |
23:25.55 | ManxPower | Mark_Logan: Business lines are usually about $50/month. |
23:26.14 | StephenF | ok so thats the next step down just POTS |
23:26.16 | ManxPower | StephenF: Then it's not really my problem because they are not big enough of a client for me to be interested in. |
23:26.24 | Mad|Cow | Does anyone have a example they can share with me from their dialplan on how to enable the "Call Forward on No Answer" feature in asterisk? |
23:26.39 | ManxPower | Really, I don't see much point in doing something other than PRI as your primary PSTN service. |
23:26.56 | StephenF | why is that? |
23:27.01 | ManxPower | Mad|Cow: You mean build a call forward no answer feature on asterisk, don't you? |
23:27.04 | jblack | Mad|Cow: Do two dials one after the other. |
23:27.12 | ManxPower | StephenF: because nothing else is as close to as reliable. |
23:27.50 | Mad|Cow | ManxPower: Yes, do you have an example I could see? |
23:28.03 | ManxPower | As an example, a friend has a cablemodem and service with Vitelity. He says about %5 of the time when he wants to use the phone it does not work. Is that reliable enough for you? |
23:28.06 | Mad|Cow | jblack: What do you mean? |
23:28.17 | ManxPower | Mad|Cow: Press Forward on your polycom phone, follow the menus |
23:28.24 | StephenF | sure, and if you say about $50/mo for a pots line 6 lines = a PRI |
23:29.04 | ManxPower | StephenF: not if your PRI is $1,200/montjh |
23:29.13 | jblack | Madcow: 123, 1, Dial(Somephone) \n 123,n,Dial(Someotherphone) |
23:29.15 | ManxPower | i.e. most of Louisiana and much of the west. |
23:29.16 | StephenF | oh that was a range |
23:29.18 | Mad|Cow | ManxPower: I've tried, asterisk doesnt seem to be playing nicely; I was just on the phone with Digium and they say I have to add something to my dial plan to support it, but couldnt elaborate |
23:29.42 | ManxPower | Mad|Cow: Asterisk does not have a call forward feature. |
23:29.43 | jblack | You have a paid contract with digium, and they wouldn't answer that? |
23:29.48 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
23:30.38 | ManxPower | Mad|Cow: Me, and most people on this channel use the call forwarding features of their phone, they don't build one for Asterisk. |
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23:31.26 | littlepinkdot | ManxPower, *72 isnt call forwarding? |
23:31.50 | *** join/#asterisk MoreAllLess (n=justo@cpe-76-169-252-172.socal.res.rr.com) |
23:32.13 | ManxPower | littlepinkdot: Asterisk is not really a PBX. Asterisk is more of a PBX Toolkit that lets you build a PBX. You could build a service on Asterisk that uses *72 for server based Call forwarding. |
23:32.15 | Mad|Cow | ManxPower: Call forwarding works fine with my Polycom phones; I'm interested in a Call forward on No answer. I have a new Polycom 301 that supports this; but asterisk doesnt seem to acknowledge it. I'm told by support I have to have soemthing in my dialplan. |
23:32.23 | ManxPower | My question is "why bother" |
23:32.38 | littlepinkdot | Ah, forgot thats not a part of FreePBX by default. |
23:32.49 | [TK]D-Fender | Why on Earth would anyone leave that decision up to a PHONE? |
23:32.52 | ManxPower | Mad|Cow: I have very large scripts that have support for ADMINISTRATOR set CF-NA |
23:33.45 | ManxPower | but now I am off to the hottub |
23:33.51 | devilsoulblack | openr2 work fine with freepbx 2.5 ? |
23:34.15 | Mad|Cow | ManxPower: Thanks for the offer; but I would like to figure out how to make it work using the softkeys on the phone |
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23:35.14 | littlepinkdot | Agh I give up |
23:35.31 | *** join/#asterisk italorossi (n=italoros@189.124.186.165) |
23:35.37 | littlepinkdot | Ill just use the built in echo cancellor...for whatever thats worth. |
23:35.50 | kfife | Hi Guys--Question: Compiling Asterisk 1.6: app_fax compile runs into an error. http://pastebin.com/m4b78f187 Is this a know issue or have I missed a dependency |
23:37.09 | [TK]D-Fender | Mad|Cow: Softkeys? Where to go / what to do on on-answer isn't a KEY. You don't push for it. It just happens. So why is this left up to a phone? When is the decision ever going to be different? |
23:37.42 | kfife | ManxPower --> littlepinkdot: well said "Asterisk is more of a PBX Toolkit " |
23:39.08 | voxter | What headsets do you guys use with polycom phones (wired headsets)? I've had some trouble with Plantronics S11/S12 wired headsets + echo |
23:39.17 | [TK]D-Fender | kfife: not even "more of a". it IS a toolkit. Nothing about it is inherently a PBX. You do have to configure everything. |
23:39.35 | [TK]D-Fender | voxter: I use Plantronics M22's + H261 binaural. |
23:40.13 | kfife | [TK]D-Fender: Even more well said. We have one installation here that is NOTHING Like a PBX |
23:40.46 | kfife | [TK]D-Fender: but it's a lot like an amazing business tool |
23:41.45 | voxter | [TK]D-Fender: can you get those in a combo, or do you buy them separate for about $200 in total? |
23:42.11 | [TK]D-Fender | All separate. I get good pricing from my reseller here |
23:45.23 | voxter | [TK]D-Fender: yeah ill check those out. I buy direct too. the S11/S12 are a nice combo package but i think maybe they have their own amp, and so does the phone, and somewhere in the mix echo is introduced... very annoying. Thx for the tip. |
23:45.58 | [TK]D-Fender | voxter: I happily pay for quality |
23:46.06 | voxter | [TK]D-Fender: me too. |
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23:58.42 | ManxPower | [TK]D-Fender: I was trying to be nice. I'm a pretty harsh critic of Digium's marketing department, I was just trying not say bad things about them today |
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