| 00:02.29 | Deeewayne | farrel_antares, enable sip debug; try again; pastebin output |
| 00:02.50 | *** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com) |
| 00:04.02 | farrel_antares | Deeewayne, can you to give me your email , bacause I have to go now and I'm not on asterisk server to show this. |
| 00:06.20 | Deeewayne | farrel_antares, I'm on IRC about 18 hours a day |
| 00:07.33 | farrel_antares | Deeewayne, okay, I'll be here tomorrow. I'm very tired. I need to go to sleep. thanks for your help. |
| 00:12.47 | Deeewayne | farrel_antares, good luck :-) |
| 00:19.30 | bcrisp | ugh |
| 00:21.02 | Deeewayne | o.o |
| 00:27.38 | *** join/#asterisk hluesea (n=hulusika@88.239.37.69) |
| 00:28.29 | Deeewayne | ~roulette |
| 00:28.29 | infobot | ACTION watches deeewayne pull the trigger: Click! |
| 00:35.38 | *** join/#asterisk Obeliks (i=obeliks@gentoo/contributor/Obeliks) |
| 00:36.07 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 00:38.36 | Obeliks | Heya. I have a FritzCard connected to one S0 running Asterisk (via chan_lcr) and a small switchboard on another S0 which has one analog phone connected. Now if the analog phone port is off the hook, and a call comes in, I'd like to be able to hang up the analog phone and get the other call. Is there any way I can "re-trigger" the phone call on the NT using Asterisk, or is this just a missing feature of my switchboard? |
| 00:39.47 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:40.44 | *** join/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 00:48.01 | *** join/#asterisk sjobeck (n=sjobeck@valdisere.sjobeck.com) |
| 00:48.51 | *** part/#asterisk sjobeck (n=sjobeck@valdisere.sjobeck.com) |
| 00:50.17 | manxpower | ~answers |
| 00:50.17 | infobot | methinks answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 00:51.39 | Obeliks | manxpower, I did search, but I can try again of course... any pointers? |
| 00:58.55 | *** join/#asterisk SaiSoma (n=SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 01:09.09 | drfreeze | <PROTECTED> |
| 01:09.38 | ian6 | drfreeze: ? |
| 01:09.45 | manxpower | drfreeze: should be no "saga". |
| 01:09.51 | manxpower | It just works normally |
| 01:09.51 | drfreeze | I'm looking at the diff of the files and I see some interesting stuff |
| 01:10.35 | drfreeze | I changed Voicemail(699) to Voicemail(699@internal) |
| 01:10.48 | hardwire | pushes over pirulo |
| 01:10.48 | *** join/#asterisk chendy (n=chatzill@58.251.101.141) |
| 01:10.55 | hardwire | pushes over p3nguin |
| 01:10.55 | drfreeze | manxpower: yes, this one office was working for 2 weeks then quit yesterday |
| 01:11.04 | manxpower | you want Voicemail(699@voicemailconfcontext699isin) |
| 01:11.22 | drfreeze | My only changes were to indicate the context |
| 01:11.38 | manxpower | Do you have a [internal] section of voicemail.conf? |
| 01:11.44 | drfreeze | manxpower: what is isin? |
| 01:12.07 | manxpower | "voicemail.conf context 699 is in" |
| 01:12.33 | manxpower | what voicemail.conf context IS 699 in? |
| 01:12.34 | drfreeze | manxpower: so, yes, I have the the vm's defined in [internal] of voicemail |
| 01:13.01 | manxpower | drfreeze: good. Then you should make sure your mailbox= line in sip.conf for device 699 has mailbox=699@internal |
| 01:13.31 | manxpower | If it still doesn't work, run a sip debug and watch for NOTIFY messages being sent from Asterisk to the SIP device |
| 01:13.52 | manxpower | remember to do a reload when you change things |
| 01:13.59 | drfreeze | The manxpower let me check that |
| 01:14.40 | drfreeze | manxpower: in sip.conf I have context=internal and mailbox=699, but I don't have mailbox=699@internal |
| 01:14.51 | drfreeze | That explains it!!!! |
| 01:16.07 | drfreeze | manxpower: do you think a sip reload will do the trick or will I need to reboot the phones? |
| 01:16.15 | manxpower | drfreeze: context= is the extensions.conf |
| 01:16.22 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:16.22 | manxpower | drfreeze: sip reload and maybe voicemail reload |
| 01:16.41 | manxpower | drfreeze: mailbox=699 defaults to adding @default if a context is not specified |
| 01:16.54 | manxpower | i.e. mailbox=699 is the same as mailbox=699@default |
| 01:18.08 | drfreeze | right. I was assuming that the context= line in sip.conf was setting the context for the sip phones |
| 01:18.28 | drfreeze | But I see that you can define a mailbox in any context to a phone |
| 01:18.29 | manxpower | drfreeze: yes, the extensions.conf context. |
| 01:18.50 | drfreeze | I don't think there is a voicemail reload |
| 01:19.10 | *** join/#asterisk Kumbang (n=kumbang@rusnas.paume.itb.ac.id) |
| 01:19.15 | manxpower | context= specifies which context in extensions.conf calls FROM that device land in. |
| 01:19.22 | p3nguin | module reload app_voicemail |
| 01:19.34 | p3nguin | no, that's not right. |
| 01:19.48 | p3nguin | What modules is for vm? |
| 01:19.50 | manxpower | app_voicemail.so |
| 01:20.04 | p3nguin | Oh, I guess I was right the first time. |
| 01:20.42 | *** part/#asterisk korihor (n=korihor@190.205.251.97) |
| 01:20.55 | *** join/#asterisk coppice (n=chatzill@50.131.92.116.dyn.pacific.net.hk) |
| 01:21.09 | *** join/#asterisk dkirker (n=dkirker@udp519393uds.csc.calpoly.edu) |
| 01:23.11 | *** join/#asterisk ali1234 (n=al@robotfuzz.co.uk) |
| 01:24.45 | *** part/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 01:25.42 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 01:32.05 | *** join/#asterisk ectospasm (n=weechat@user-24-236-95-118.knology.net) |
| 01:38.16 | drfreeze | manxpower: Thanks |
| 01:38.25 | drfreeze | Thanks much |
| 01:45.27 | dongs | hey guess wat i fixed my session timer problem... by downgrading back to 1.4 |
| 01:46.14 | ManxPower-work | dongs: what version were you using? |
| 01:48.07 | *** join/#asterisk voipmonk (n=voipmonk@69.172.100.53) |
| 01:59.37 | Katty | hum |
| 01:59.53 | Katty | putty doesn't look too shabby on a 46" |
| 02:10.35 | *** join/#asterisk Deeewayne (n=dwayne@c-71-228-179-90.hsd1.al.comcast.net) |
| 02:10.35 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 02:10.59 | dongs | ManxPower-work: 1.6.1.latest, it was broken, i tried to patch it, it was wasting my time, went back to 1.4 + working patch |
| 02:11.18 | *** join/#asterisk GameGamer43 (n=GameGame@CPE-65-27-76-78.new.res.rr.com) |
| 02:19.36 | x86 | gah |
| 02:20.26 | x86 | can't get any phones to register... asterisk CLI keeps giving me "registration from '<sip:7601@asteriskf.q.d.n>' failed for 'ip.of.phone.here' - not a local domain" |
| 02:21.05 | x86 | I've tried commenting out all domains, commenting out autodomain, and commenting out realm, so as all of those would just get defaulted |
| 02:21.18 | x86 | phones are attempting to register via the server DNS name, and not the IP |
| 02:21.29 | x86 | is that the problem? |
| 02:22.11 | dongs | shouldnt be. |
| 02:22.15 | x86 | running 1.6.1.11 |
| 02:22.33 | x86 | any ideas? |
| 02:23.10 | dongs | is that the exact error message? |
| 02:23.21 | dongs | oic, 'Not' is capitalized. |
| 02:24.10 | x86 | 'Registration' is also capitalized, if that makes a difference ;) |
| 02:24.18 | x86 | other than case, that's the message |
| 02:24.44 | x86 | (FQDN of asterisk server and IP of Polycom phone changed to protect the innocent hehe) |
| 02:25.46 | x86 | ManxPower-work, Qwell, [TK]D-Fender: any ideas here? |
| 02:26.01 | dongs | what does sip show settings show for sip domains |
| 02:27.02 | x86 | "our auth realm" and "from: domain:" are both the DNS domain |
| 02:27.20 | x86 | other than that, it doesn't have a domain list at all (I commented them all out of sip.conf thinking that was the issue) |
| 02:27.52 | x86 | ah... "sip show domains" has a different story |
| 02:28.16 | x86 | it appears it did not unload the domains when I did a "module unload chan_sip" ; "module load chan_sip" |
| 02:29.23 | x86 | even a complete reload of asterisk and the sip domains are still there?! |
| 02:29.26 | x86 | wtf is going on here |
| 02:31.17 | x86 | ok odd, the changes I was making to the file were not getting read |
| 02:31.43 | x86 | perhaps the file was locked somehow... anyway, sip show domains is now saying domain support is disabled (which is what I want, methinks) |
| 02:32.01 | x86 | check it out, polycom 1 just registered :) :) :) |
| 02:32.07 | x86 | sorry to bother you lol thanks |
| 02:33.05 | dongs | excellent. |
| 02:34.48 | x86 | hmm, now to get paging working to a Polycom IP601 |
| 02:36.04 | dongs | i'm glad the only thing my asterisk does is turns voip into analog/isdn so i can plug it into a real pbx. |
| 02:36.11 | dongs | voip phones are a fucking disaster. |
| 02:36.30 | x86 | heh |
| 02:36.31 | x86 | can be |
| 02:36.40 | x86 | Polycoms are mostly sane though |
| 02:43.07 | *** join/#asterisk OrNix (n=ornix@host89-251-107-21.hnet.ru) |
| 02:44.00 | *** join/#asterisk joshaidan (n=brianj@24.109.210.41) |
| 02:48.20 | kam187 | hmm anyone using asterisk trunk? |
| 02:49.00 | bcrisp | hmm why does the QueueStatus AMI action return Events rather than a response? |
| 02:49.48 | bcrisp | and Events to what? there is no unique identifier to match request/ response |
| 02:52.37 | kam187 | hmm netstat -an doesnt show sip or ooh323 up |
| 03:21.58 | *** join/#asterisk gladier (n=gladier@pedobear.molested.me) |
| 03:24.20 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 03:37.10 | x86 | is it possible to do post-Dial processing on a call? |
| 03:37.41 | x86 | for example, I want to MixMonitor a bridged call, and after one (or both) parties hang up, I want to turn the wav into an mp3 |
| 03:37.57 | x86 | so can the dialplan continue to execute further instructions after the call is hung up? |
| 03:38.22 | *** part/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 03:39.43 | ChannelZ | x86: http://www.voip-info.org/wiki/view/Asterisk+h+extension but I dunno if it will always do what you want, I've never used it |
| 03:47.12 | *** join/#asterisk cpoulson (n=chris@pdpc/supporter/active/cpoulson) |
| 03:54.29 | *** join/#asterisk xmitter (n=xmitter@c-24-21-212-187.hsd1.or.comcast.net) |
| 03:57.31 | bcrisp | man its cruel to force agents to listen to the default on hold music |
| 04:00.54 | ChannelZ | Replace it with John Mayer, torture them to the max |
| 04:05.01 | bcrisp | ive been logged in as an agent while i test out my AMI software.. and that music is making me dumber by the second |
| 04:05.34 | bcrisp | ~roulette |
| 04:05.35 | infobot | ACTION watches bcrisp pull the trigger: Click! |
| 04:08.39 | *** join/#asterisk razu (n=razu@razu.data.ee) |
| 04:22.30 | bcrisp | maybe some kenny g ? hah |
| 04:29.55 | *** join/#asterisk retentiveboy (n=pdugas@74.95.28.37) |
| 04:31.50 | *** join/#asterisk Tech_Travis (n=Administ@cpe-76-87-9-130.socal.res.rr.com) |
| 04:33.14 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 04:34.25 | *** join/#asterisk soman (n=somnath@118.102.130.6) |
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| 04:46.54 | *** join/#asterisk blkry (n=chatzill@96.37.24.97) |
| 04:52.47 | *** join/#asterisk voipmonk (n=voipmonk@69.172.100.53) |
| 04:54.38 | *** join/#asterisk hakr (i=bryan@element.techlive.tv) |
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| 04:57.14 | ChannelZ | beats bcrisp over the head with a clarinet |
| 04:59.12 | bcrisp | ChannelZ: Celine! |
| 04:59.26 | bcrisp | ChannelZ: make them listen to the titanic theme over.. and over |
| 05:00.09 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 05:00.37 | *** join/#asterisk ticoit (n=ticoit@201.191.187.203) |
| 05:00.41 | ChannelZ | heh |
| 05:00.56 | ChannelZ | I haven't played with queues extensively, why are the agents made to listen to MOH? |
| 05:02.56 | bcrisp | because i da man? |
| 05:03.17 | bcrisp | nah its just an option.. they can dial in or be dialled |
| 05:03.29 | bcrisp | when they dial in they can hear silence or hear music |
| 05:04.07 | ChannelZ | so then they get connected to someone, do their business, and then wait for the other side to hang up before they are put back 'on hold' waiting for another call? |
| 05:04.18 | bcrisp | right |
| 05:05.31 | bcrisp | ~roulette |
| 05:05.31 | infobot | ACTION watches bcrisp pull the trigger: Click! |
| 05:05.42 | bcrisp | does infobot's roulette ever fire? |
| 05:06.07 | ChannelZ | yeah but it's not as fun when the bot doesn't have ops |
| 05:06.49 | bcrisp | annoying thing that im finding is that calling an action against the manager interface just generates events not tied to a particular request action ID |
| 05:07.27 | bcrisp | so instead of having a client fire QueueStatus and wait for a response, i need to have some service periodically fire QueueStatus, then the clients just listen for events and filter by queue name .. ugh |
| 05:08.09 | voipmonk | why have you created this monster to begin with? |
| 05:08.16 | bcrisp | monster? |
| 05:08.20 | ChannelZ | I was just gonna ask, what exactly are you writing? |
| 05:08.31 | bcrisp | something to visualize queues |
| 05:09.09 | bcrisp | realtime stats etc |
| 05:09.19 | voipmonk | like orderlyq? |
| 05:09.46 | voipmonk | but u thought your time was cheaper than buying www.orderlyq.com :) |
| 05:10.02 | bcrisp | didnt know about it |
| 05:10.10 | bcrisp | asked these questions before and was told repeatedly "learn AMI" |
| 05:10.23 | voipmonk | shakes his head |
| 05:10.46 | voipmonk | doing what you are told will get you into a lot of trouble |
| 05:10.47 | voipmonk | LOL |
| 05:11.31 | bcrisp | hmm orderlyQ doesnt look like what we need |
| 05:12.07 | ChannelZ | http://www.voip-info.org/wiki/view/QueueMetrics ? |
| 05:13.45 | bcrisp | thats closer |
| 05:14.02 | bcrisp | i dont mind doing this.. its not hard.. just some of the AMI is a little .. whats the word... bad |
| 05:14.22 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 05:14.40 | bcrisp | anyways, wrote the program today that gives realtime queue stats etc so ill go with that for now |
| 05:15.07 | ChannelZ | yeah it'd be nice if it were a little more structured |
| 05:15.12 | bcrisp | or consistent |
| 05:15.34 | ChannelZ | it might be as fast to write your own Manager that spits out the information the way you need it :) |
| 05:16.12 | bcrisp | ya.. tho it is helpful in the fact that it teaches me a little bit more about how * "thinks" |
| 05:18.14 | bcrisp | i suppose i could write my own userevents into the dialplan |
| 05:18.51 | bcrisp | part of the reason im doing this is that i need to integrate a crm and helpdesk into it also |
| 05:22.13 | dan__t | So a few weeks ago I saw these errors. I don't remember what I did to resolve them, but I remember it being something really simple and retarded that I didn't do: [Dec 10 22:21:23] NOTICE[3520]: chan_sip.c:9740 sip_reg_timeout: -- Registration for 'ABCDXYZ123@sip.flowroute.com' timed out, trying again (Attempt #24) |
| 05:23.25 | dan__t | It was one of those "DOH!" moments. |
| 05:25.39 | ChannelZ | is that a question? |
| 05:26.22 | dan__t | I blame sleep. |
| 05:26.30 | dan__t | "Anyone seen that before and might be able to clue me in to what's going on?" |
| 05:28.11 | ChannelZ | DNS, firewall, host down.. |
| 05:28.29 | dan__t | I remember what it was. Shit. |
| 05:29.00 | dan__t | damnit. |
| 05:29.08 | dan__t | LumenVox media server listens on port 5060 as well. |
| 05:29.51 | ChannelZ | well thats rude of them |
| 05:30.08 | dan__t | Fact. |
| 05:36.22 | *** join/#asterisk mintos (n=mvaliyav@117.206.19.139) |
| 05:59.21 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) |
| 06:13.26 | razu | does anyone know what is the variable which defines that recordings will go to /var/spool/asterisk/monitor ? |
| 06:14.11 | razu | in 1.6 |
| 06:15.09 | razu | giving folder to Monitor() command doesn't make any difference ... in case you put Monitor(wav,/another/folder/file,mb) ... MONITOR_EXEC still gets /var/spool/asterisk/monitor//antoher/folder/file.wav |
| 06:15.27 | razu | in 1.2 it worked ... 1.4 also AFAIK |
| 06:16.33 | *** join/#asterisk bcrisp (n=irchon@ip72-222-167-229.ph.ph.cox.net) |
| 06:19.05 | *** join/#asterisk bcrisp (n=irchon@ip72-222-167-229.ph.ph.cox.net) |
| 06:19.32 | bcrisp | ~roulette |
| 06:19.33 | infobot | ACTION watches bcrisp pull the trigger: Click! |
| 06:20.42 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 06:22.38 | hardwire | ~roulette |
| 06:22.39 | infobot | ACTION watches hardwire pull the trigger: Click! |
| 06:22.47 | *** join/#asterisk bcrisp (n=irchon@ip72-222-167-229.ph.ph.cox.net) |
| 06:22.52 | hardwire | ~help |
| 06:23.06 | voipmonk | whats up, hardwire ? |
| 06:23.17 | hardwire | voipmonk: looking at infobots pheaturez |
| 06:23.33 | hardwire | it sent me an unholy list of them |
| 06:23.34 | voipmonk | §§§§§§§§§§§§ |
| 06:23.44 | hardwire | type ~help voipmonk |
| 06:24.02 | hardwire | ~piglatin and you'll see what I mean. |
| 06:24.04 | voipmonk | my keys are fux0r3d |
| 06:24.12 | voipmonk | cant choose the sqiuggly |
| 06:24.17 | voipmonk | it comes up § |
| 06:24.19 | hardwire | copy/paste |
| 06:24.30 | voipmonk | ~help |
| 06:24.33 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 06:24.41 | voipmonk | unholy |
| 06:24.42 | hardwire | see |
| 06:24.52 | hardwire | ~spell asthmar |
| 06:24.56 | voipmonk | ~dollar variables |
| 06:24.57 | infobot | If I had a dollar for every brain variables didn't have... I'd have ONE dollar *CREEPERZ* |
| 06:25.06 | voipmonk | ..... |
| 06:25.10 | hardwire | hehe |
| 06:25.17 | voipmonk | ~hostdel |
| 06:25.31 | voipmonk | ~ircstats |
| 06:25.31 | infobot | Currently I'm hooked up to irc.us.freenode.net:6667 but only for 2d 3h 55m 29s. I had to reconnect 4 times. Connectivity: 100.00 % |
| 06:25.50 | hardwire | !upsidedown I'm a monkey. |
| 06:25.50 | hardwire | ~upsidedown I'm a monkey. |
| 06:25.52 | hardwire | aww |
| 06:32.28 | voipmonk | ~unlobotomy |
| 06:32.30 | *** join/#asterisk Tim_Toady (n=moi@188.4.76.130.dsl.dyn.forthnet.gr) |
| 06:38.50 | dan__t | So before MeetMe, what was used to bridge or otherwise conference more than one call? |
| 06:38.50 | voipmonk | your phone's three way function |
| 06:38.50 | dan__t | ...internal to Asterisk. |
| 06:38.50 | voipmonk | or your phones conferencing features |
| 06:38.50 | voipmonk | check the changelogs back before asterisk 1.2 |
| 06:38.50 | voipmonk | what are you looking to do? |
| 06:38.50 | dan__t | Trying to run ChanSpy on a channel that's in MeetMe. |
| 06:38.50 | dan__t | So that kind of works, except the spied on channel cannot hear the spying channel, even with ChanSpy's w flag. |
| 06:38.50 | *** part/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 06:45.13 | voipmonk | what happens when u spy on the channel before it goes into meetme , then what happens while you're spying on it while it goes into meetme? |
| 06:45.13 | dan__t | Haven't tried spying before meetme, I'll give that a shot. |
| 06:45.13 | dan__t | Thanks for the idea. |
| 06:45.13 | voipmonk | Im here all day |
| 06:45.13 | voipmonk | till sat |
| 06:45.13 | dan__t | haha thanks. |
| 06:45.54 | dan__t | Ok, I want to test that but I'm not so sure how I can do that before sending someone to MeetMe. Can I have a channel open to/with Asterisk, that just kind of sits there, without going to an extension? Can I put it in Wait() or something, and if so, can I issue commands on it? |
| 06:46.13 | *** join/#asterisk _bugz_ (n=bugz@adsl-99-154-128-250.dsl.lsan03.sbcglobal.net) |
| 06:48.39 | voipmonk | you can play a message or have it wait a few seconds while u locate the channel |
| 06:48.43 | voipmonk | then listen in on it |
| 06:48.48 | voipmonk | and on the other channel join the meetme |
| 06:49.04 | voipmonk | what are you trying to do? |
| 06:58.32 | ChannelZ | well I just did a crazy test |
| 06:58.40 | ChannelZ | I dialed in IAX, went into a conference |
| 06:58.50 | ChannelZ | dialed in Skype and entered the same conference (using SFA) |
| 06:59.12 | ChannelZ | Called in my cell phone and did a ChannelSpy(IAX) and I was able to hear everything in the conference |
| 06:59.34 | kaldemar | dan__t: i've been using asterisk since 0.7.4 i think, and don't remember any other conference solution than MeetMe. |
| 07:00.48 | dan__t | Hey, sorry, got lost in terminal land. |
| 07:01.13 | dan__t | So maybe SIP and IAX and friends are handled differently.... only slightly, but just so much. |
| 07:01.22 | ChannelZ | differently than what? |
| 07:01.33 | ChannelZ | What are you trying to listen to? |
| 07:02.19 | dan__t | I want the spying channel to be able to talk to the spied on channel, while the spied on channel is in MeetMe. |
| 07:02.31 | dan__t | Even with the w option of ChanSpy, I cannot do this. |
| 07:03.43 | ChannelZ | Hmm I'd need more bodies to test that here |
| 07:05.04 | dan__t | yea i've got a friend as a guinea pig, plus my desk phone, plus my home phone haha |
| 07:05.27 | ChannelZ | well I think it works |
| 07:05.40 | ChannelZ | I IAX'd into a conf, and cellphoned in to spy with 'w' and I am able to talk to myself |
| 07:06.10 | ChannelZ | IE if I yell into my cell phone I hear it in my IAX softphone |
| 07:06.28 | ChannelZ | let me see if my landline is charged enough to work |
| 07:06.30 | dan__t | What did you spy? How diid you spy? |
| 07:06.49 | dan__t | Yeah that's exactly what I'm trying to do. |
| 07:06.56 | ChannelZ | ChanSpy(IAX,w) |
| 07:07.23 | dan__t | WHen you say IAX like that, should I just assume IAX/channelnameyadayada or a literal IAX |
| 07:07.36 | dan__t | And if so, how does that work? Does that apply to ALL AIX channels? |
| 07:07.41 | ChannelZ | I didn't specify a channel, it's like a wildcard |
| 07:07.47 | ChannelZ | it picks up the first, you can cycle through with * |
| 07:08.07 | dan__t | oh, right. |
| 07:08.12 | dan__t | Sorry, its coming together now, kind of. |
| 07:08.15 | ChannelZ | let me see if it works being specific but I don't know why it would be any different |
| 07:08.19 | dan__t | I use ChanSpy(SIP/1000-09d3ee60,w) etc etc |
| 07:08.28 | dan__t | That's what fails. |
| 07:08.58 | dan__t | So I have a VoIP phone stuck in Wait(3600), I originated to an extension to my cell phone |
| 07:09.23 | dan__t | Even with w, I still cannot hear the spying channel from the spied on channel. |
| 07:09.45 | dan__t | So its got nothing to do with MeetMe, it looks. |
| 07:10.19 | ChannelZ | hmm |
| 07:10.55 | dan__t | I'll break it down even simpler, hold on a sec. |
| 07:11.18 | ChannelZ | this is trange |
| 07:11.39 | ChannelZ | when I did ChanSpy(IAX/Bob-Home,w) I got "-- <DAHDI/1-1> Playing 'beep' (language 'en')" |
| 07:11.47 | ChannelZ | and then nothing. I'm not sure what it's doing. |
| 07:12.14 | dan__t | How did you "do" ChanSpy? Did you make an extension and then run it? |
| 07:12.31 | ChannelZ | yeah |
| 07:12.36 | dan__t | Still a newbie at on-the-fly commands, sorry - trying to see how much of it I can do with straightup console and what has to be done through extensions.conf |
| 07:13.09 | ChannelZ | oh duh |
| 07:13.12 | ChannelZ | it's IAX2/* |
| 07:13.15 | ChannelZ | trying again |
| 07:13.16 | dan__t | hehe |
| 07:14.16 | ChannelZ | huh no it still worked |
| 07:14.23 | ChannelZ | Try not putting the entire call ID on your ChanSpy |
| 07:14.27 | ChannelZ | just the channel |
| 07:14.44 | dan__t | Ok. |
| 07:15.02 | dan__t | So like SIP/1000? |
| 07:15.06 | ChannelZ | yeah |
| 07:15.35 | dan__t | trying |
| 07:15.51 | dan__t | Nope |
| 07:16.00 | ChannelZ | That makes no sense |
| 07:16.08 | ChannelZ | So you can hear what is going on, but can't talk to the person? |
| 07:16.30 | dan__t | Correct - spying channel cannot talk to the spied on channel. |
| 07:16.33 | dan__t | [Dec 11 00:15:41] NOTICE[4098]: app_chanspy.c:240 start_spying: Attaching SIP/flowroute-09e3b9c8 to SIP/1000-09d3ee60 |
| 07:16.55 | dan__t | That happens even when I specify ChanSpy(SIP/100,w), so it resolves the actual active channel the same as it would if I simply put in the full channel. |
| 07:17.23 | *** join/#asterisk |Cybex| (n=John@212.178.82.21) |
| 07:17.36 | ChannelZ | do you have canreinvite=no or are perhaps the two endpoints talking directly which is why you're not really able to do anything about it? |
| 07:18.29 | ChannelZ | because unless the stream the spied on channel is getting is coming from * you of course couldn't mix your own audio into that stream being sent to them |
| 07:19.22 | ChannelZ | although thinking about it, in a MeetMe everyone has to be getting their stream from * in order for it to work at all in the first place... hmmm |
| 07:19.47 | ChannelZ | I can try it with SIP phones when I get to work tomorrow |
| 07:20.00 | dan__t | Appears that canreinvite has no effect |
| 07:20.12 | dan__t | I'm not even using MeetMe for this test |
| 07:20.25 | ChannelZ | so you can't spy on a SIP channel and talk to them at all |
| 07:20.29 | dan__t | It appears that ChanSpy is not working for me as I expect it to. I can't move on to using MeetMe until I get this resolved. |
| 07:20.36 | dan__t | That generally sums it up, sure. |
| 07:21.18 | ChannelZ | hmm |
| 07:21.45 | ChannelZ | but without MeetMe my point I think would still be valid |
| 07:22.05 | dan__t | It may be, sure. |
| 07:22.06 | ChannelZ | though you said you just tried it with canreinvite off |
| 07:22.10 | dan__t | Unfortunately I can't test that. |
| 07:22.26 | ChannelZ | eh? |
| 07:22.57 | dan__t | er, yes, sorry, read that wrong. |
| 07:22.57 | *** join/#asterisk newsmafia (n=newsmafi@wsip-98-173-56-18.sd.sd.cox.net) |
| 07:23.24 | ChannelZ | so the SIP channel you're trying to spy on is or isn't local to the * box it's self? |
| 07:24.17 | dan__t | https://issues.asterisk.org/view.php?id=16262 |
| 07:24.25 | dan__t | It is. |
| 07:24.36 | dan__t | Its from my Polycom 601 to the Asterisk box sitting right here via SIP. |
| 07:25.48 | ChannelZ | and the channel you're spying from is also |
| 07:25.50 | *** join/#asterisk Ad-Hoc (n=nimbus@62.169.216.185) |
| 07:26.03 | *** join/#asterisk xrmx__ (n=rm@host143-252-dynamic.8-87-r.retail.telecomitalia.it) |
| 07:26.30 | ChannelZ | I have no idea what VAD and CNG are :! |
| 07:27.00 | ChannelZ | silence suppresion and... |
| 07:28.08 | dan__t | Just fishing for ideas. |
| 07:28.42 | ChannelZ | I'm not sure that's related in your case |
| 07:29.49 | ChannelZ | did you turn canreinvite off globally |
| 07:31.43 | kaldemar | VAD is voice activity detection and CNG is comfort noise generation. VAD is that a phone stops to send voice frames if it thinks that it's not receiving any audio to the speaker. this usually causes a silence to the other end, which often makes people think that the call has dropped. that's why CNG was introduced, that there wouldn't be absolute silence in the other end. |
| 07:32.25 | ChannelZ | so CNG just generates soft white noise on the other end or something? |
| 07:33.00 | kaldemar | yes |
| 07:33.16 | ChannelZ | huh. fun.\ |
| 07:33.28 | ChannelZ | thanks |
| 07:34.14 | kaldemar | see rfc 3389 for more. |
| 07:39.35 | dan__t | Thanks for explaining, kaldemar. |
| 07:39.51 | dan__t | According to the Asterisk book, w sounds like it implies being used only on a bridge. |
| 07:40.08 | dan__t | Were you able to use w and have the spied on channel hear the spying channel, ChannelZ? |
| 07:40.24 | ChannelZ | Yes |
| 07:40.47 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 07:40.50 | dan__t | Even with the simplest of tests, huh. |
| 07:40.57 | dan__t | Wonder why I couldn't do that. |
| 07:40.58 | ChannelZ | Softphone->IAX->Asterisk->MeetMe ; Call in cell phone->DAHDI->Asterisk->ChanSpy(IAX) |
| 07:41.08 | dan__t | And no, did not try canreinvite off globally unless that's a default, kaldemar. |
| 07:41.25 | ChannelZ | that was me. |
| 07:41.59 | ChannelZ | you said you tried canreinvite=no and it didn't help, but did you turn it off for both SIP peers? |
| 07:42.00 | dan__t | Default is yes |
| 07:42.11 | dan__t | I did, yes. |
| 07:42.40 | ChannelZ | hmm. Well as I said I'll have to try it tomorrow when I get to work on SIP phones and see if I get the same results, I don't have a good way to test from here |
| 07:42.59 | dan__t | Undestood. |
| 07:43.01 | dan__t | Thank you for doing that. |
| 07:43.20 | ChannelZ | sure. I'll let you know what happens |
| 07:44.19 | dan__t | Very cool, thanks./ |
| 07:47.26 | dan__t | Which version? |
| 07:52.50 | *** part/#asterisk newsmafia (n=newsmafi@wsip-98-173-56-18.sd.sd.cox.net) |
| 07:54.35 | ChannelZ | 1.4 at work |
| 07:54.48 | ChannelZ | 1.4.27.1 to be precise |
| 07:55.14 | dan__t | Got it. |
| 07:58.30 | creativx | so how painful will it be to replace our iax2 trunks with sip "trunks" ? |
| 07:58.52 | *** join/#asterisk Tech_Travis (n=Administ@cpe-76-87-9-130.socal.res.rr.com) |
| 08:03.40 | voipmonk | not very |
| 08:03.48 | voipmonk | should take you a few minutes |
| 08:03.58 | creativx | i was expecting a few days of pain |
| 08:03.59 | creativx | hehe |
| 08:05.24 | *** join/#asterisk blinkiz (n=blinkiz@unaffiliated/blinkiz) |
| 08:06.10 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 08:06.32 | blinkiz | Hi. In the asterisk CLI, I can enable core debug. I now want to dump all input coming in under the day to a file. How can I do this? |
| 08:10.06 | *** join/#asterisk casix (n=casix@xenpbxedifici.adamvozip.es) |
| 08:10.09 | casix | hello |
| 08:15.52 | ChannelZ | hola |
| 08:16.00 | *** join/#asterisk icyValk77 (n=icyValk7@213.129.64.4) |
| 08:17.08 | ChannelZ | blinkiz: core set debug <level> [filename] |
| 08:17.21 | ChannelZ | see 'help core set debug' |
| 08:17.37 | *** join/#asterisk Chris-NB (n=chris@85-126-61-10.work.xdsl-line.inode.at) |
| 08:18.04 | blinkiz | ChannelZ, Well, I found out. Found the file /etc/asterisk/logger.conf |
| 08:18.14 | *** part/#asterisk blinkiz (n=blinkiz@unaffiliated/blinkiz) |
| 08:18.59 | Chris-NB | hi |
| 08:19.08 | Chris-NB | anyone using SRTP with Asterisk? |
| 08:19.14 | Chris-NB | or tried to? |
| 08:23.33 | ChannelZ | nope |
| 08:25.19 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 08:26.23 | *** join/#asterisk mpe (n=mpe@gate.ipvision.dk) |
| 08:28.00 | dan__t | ChannelZ, I'm going to crash. Thanks again for your help. |
| 08:28.36 | casix | anyone know how to make a mutex access to a variable? |
| 08:30.30 | kaldemar | casix: what variables are you talking about? |
| 08:32.25 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 08:33.18 | ChannelZ | dan__t: me too, no problem.. try to catch up with you tomorrow |
| 08:44.00 | casix | a counter |
| 08:45.01 | casix | kaldemar: I have to make a counter for the calls. I will assign a number to each call but I need to get the number and increment with mutex access on it |
| 08:47.12 | *** join/#asterisk mintos (n=mvaliyav@117.206.19.139) |
| 08:48.20 | kaldemar | casix: aside from normal channel variables you have functions SHARED and GLOBAL to handle variables. but, this sounds like a perfect use case for GROUP functions. see functions GROUP and GROUP_COUNT. |
| 08:49.20 | casix | kaldemar: thx :) |
| 08:50.22 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 08:52.52 | *** join/#asterisk war9407 (i=war@71.6.165.232) |
| 08:54.46 | Chris-NB | no one ever tried SRTP for Asterisk? |
| 08:56.11 | ChannelZ | nope |
| 08:57.01 | kaldemar | Chris-NB: there's no support for it in release branches, afaik. |
| 08:58.52 | ChannelZ | yeah it's a totally separate crazy branch aint it |
| 08:59.29 | kaldemar | http://issues.asterisk.org/view.php?id=5413 |
| 09:01.04 | ChannelZ | 2005 sheesh |
| 09:01.51 | kaldemar | it's had activity though. see the last notes. |
| 09:02.02 | ChannelZ | yah |
| 09:05.46 | *** join/#asterisk Faustov (i=user@gentoo/user/faustov) |
| 09:07.36 | ChannelZ | wanders off to bed |
| 09:07.48 | *** join/#asterisk erbse (n=Andre@voxgate.winet.ch) |
| 09:18.00 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 09:18.25 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 09:22.21 | casix | kaldemar: but I need to assign a unique number to each call, with group when one call finish the group decrement. I was using the database because if asterisk is restarted i can continue with the number |
| 09:22.32 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 09:26.06 | *** join/#asterisk Mark22 (n=mark@unaffiliated/mark21) |
| 09:32.54 | Mark22 | are messages like "[Dec 11 10:27:39] > doing dnsmgr_lookup for 'sip.voipbuster.com'" normall to get them every 1 or 2 minutes? I did remove debug and verbose from the config files. asterisk version: 1.6.1.11 |
| 09:34.17 | TommyBotten | Mark22: Yes. As long as you use DNS SRV, it is normal |
| 09:37.21 | Mark22 | are there ways to keep it out of the cli/logs? |
| 09:38.30 | TommyBotten | Are you using DNS SRV ?... if not, disable it |
| 09:46.12 | Mark22 | as far as I know I am not using it, so I could disable it (at least to test if someone on the system is using it) |
| 10:01.02 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 10:01.11 | *** join/#asterisk yoshx (n=yoshx@88-138-188-188.adslgp.cegetel.net) |
| 10:15.35 | *** join/#asterisk RypPn2 (n=Gloop@rossgroup2.demon.co.uk) |
| 10:23.42 | eppigy | hello |
| 10:26.43 | *** join/#asterisk Gugge (n=gugge@vlan2.dlxhosting.dk) |
| 10:30.31 | Mark22 | hello eppigy |
| 10:31.54 | eppigy | hi |
| 10:34.23 | *** join/#asterisk bbt (n=sam@samuels.id.au) |
| 10:41.15 | *** join/#asterisk |Cybex| (n=John@atwork-21.r-212.178.82.atwork.nl) |
| 10:54.57 | TSM2 | has anyone had issues with their polys messing up any custom ring cadances setup though the xml file? i have a IP550/450 that after a few weeks seems to mess up and just rings, ignoring the silence i setup in the cadance to match UK ring cadance |
| 10:58.02 | *** join/#asterisk xmitter (n=xmitter@c-24-21-212-187.hsd1.or.comcast.net) |
| 10:59.24 | *** join/#asterisk waa (n=waa@balrog.credipar.com.br) |
| 11:14.13 | *** join/#asterisk verywiseman (n=khaled@unaffiliated/verywiseman) |
| 11:15.04 | *** join/#asterisk lost_soul (n=noymfb@cpe-74-71-234-100.twcny.res.rr.com) |
| 11:39.30 | m0e | What's a good link to help with dahdi issues? (setting up the fxo card, etc. etc.) |
| 11:40.04 | m0e | its currently dialing out/in but I get no audio on the sip phone (audio works on the other side though) |
| 11:40.46 | Tim_Toady | then its propably a sip issue not dahdi |
| 11:41.27 | *** join/#asterisk louben (n=lou@212-70-216-131.ath.static.tee.gr) |
| 11:41.48 | m0e | sip to sip works though |
| 11:42.25 | m0e | so i'm guessing it must be something related to dahdi (codec issue?) |
| 11:42.27 | Tim_Toady | one way audio is usually a sip problem |
| 11:42.50 | Tim_Toady | do u have nat between ur phone and ur * box? |
| 11:43.18 | m0e | well.. yes and no.. yes the * box is behind nat.. but the phone i'm testing with is on the same lan so no |
| 11:45.00 | Tim_Toady | what codec are u using? tried using ulaw/alaw? |
| 11:45.21 | m0e | yeah.. i tried using both of them |
| 11:45.27 | drmessano | Incorrect NAT settings could do that too |
| 11:45.34 | m0e | currently in my sip config, i have it so that it tries g729, then alaw, then ulaw |
| 11:45.59 | drmessano | what is your localnet setting? |
| 11:46.48 | m0e | localnet=192.168.10.0/24 |
| 11:47.13 | drmessano | writes that down for haxoring later |
| 11:47.18 | m0e | lol |
| 11:47.25 | drmessano | Formatted correctly |
| 11:47.29 | drmessano | So you're ok there |
| 11:47.41 | m0e | should i try maybe setting the subnetmask to 255.255.255.0? |
| 11:47.47 | drmessano | Naw |
| 11:48.02 | drmessano | Using /24 works fine |
| 11:48.25 | m0e | i had a problem in sip.conf before.. externhost doesnt seem to work for me, so i had to write a small script to insert externip every 10 minutes and do a reload |
| 11:48.48 | drmessano | externhost wont work unless the DNS is configured |
| 11:48.52 | m0e | sip reload that is |
| 11:48.59 | m0e | oh.. local DNS? |
| 11:49.01 | drmessano | yes |
| 11:49.09 | m0e | grrr.. thanks for telling me |
| 11:49.10 | TSM2 | just wandering if anyone knows how US state taxes work, im looking at getting bandwidth.com as a SIP trunk provider and trying to see if we will be stung for any taxes as we currently do with out PSTN lines. |
| 11:49.14 | drmessano | The box needs to perform a standard DNS lookup |
| 11:49.27 | m0e | i was bashing my head against it with no results |
| 11:49.56 | drmessano | externhost just seeds a DNS lookup that asterisk then uses as the IP, since we don't use hostnames in SIP messages |
| 11:50.27 | m0e | hrm.. ok.. back to my initial problem.. anywhere else I can check? any cool document that I can read? anything? :P |
| 11:50.56 | drmessano | This may be your initial problem |
| 11:51.18 | drmessano | Are you using 1.4? |
| 11:51.25 | drmessano | or *gasp* 1.2 |
| 11:51.38 | m0e | I'm using the latest asterisknow, so i believe its 1.4, let me check |
| 11:51.56 | m0e | Asterisk 1.4.26.3 |
| 11:53.33 | drmessano | Fix up your resolv.conf and try it.. I may be on the wrong path, but broken DNS does some weird things on 1.4 and prior |
| 11:54.10 | TommyBotten | Not just 1.4. I've seen the strange DNS behaviour on 1.6.1.10 aswell |
| 11:54.20 | m0e | fixed |
| 11:54.27 | m0e | let me try the dahdi problem now |
| 11:54.52 | drmessano | TommyBotten: What sort? |
| 11:55.19 | kaldemar | m0e: do you have G.729 licenses for asterisk? |
| 11:55.35 | TommyBotten | drmessano: When we lost DNS, the phones would no longer register, and asterisk seemed to refuse any SIP traffic |
| 11:55.43 | m0e | kaldemar, there was some link for downloading the free version (so i followed that) |
| 11:55.52 | drmessano | TommyBotten: Oh, that old bastard is still around |
| 11:55.58 | drmessano | TommyBotten: Weird |
| 11:56.21 | m0e | something about needing higher processing power.. and since this is just going to be used for a couple of calls.. i though tit would be ok |
| 11:56.22 | drmessano | TommyBotten: I always install a local BIND and set as secondary, just for those times |
| 11:56.31 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 11:56.36 | m0e | do you think this is the problem? though if it was.. wouldnt it be the problem with sip => sip calls? |
| 11:56.38 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 11:56.38 | TommyBotten | drmessano: It comes and goes. But considering it's unusual ops, I've never reported an issue |
| 11:56.40 | drfreeze | Anyone know if an extension can have more than one voicmail attached to it? |
| 11:56.42 | TommyBotten | drmessano: Same here |
| 11:56.43 | drmessano | m0e: Illegal and unsupported |
| 11:56.47 | kaldemar | m0e: i'd make sure it's not a codec issue on the SIP side first. |
| 11:56.49 | m0e | illegal? |
| 11:57.00 | drmessano | G729 is not free |
| 11:57.06 | TommyBotten | It's not illegal |
| 11:57.07 | TommyBotten | Or |
| 11:57.19 | TommyBotten | It's only illegal in the US where you people have patents in IP |
| 11:57.24 | m0e | i thought there were a couple of versions of g729 |
| 11:57.26 | TommyBotten | For the rest of the free world, ideas are free |
| 11:57.37 | drmessano | TommyBotten: Its not legal to use in any country that enforces the patent on G729 |
| 11:57.38 | kaldemar | m0e: when you do SIP to SIP, it might just use G.729 passthrough, which is no problem. but from DAHDI to SIP, you'd have to transcode, and that's a different animal. |
| 11:57.42 | TommyBotten | There are free implementations of G.729. |
| 11:57.50 | TommyBotten | drmessano: Exactly... which is US. And maybe canada |
| 11:58.01 | drmessano | TommyBotten: It's more than that |
| 11:58.12 | TommyBotten | drmessano: No country in EU or Asia does it. Except for maybe Japan |
| 11:58.19 | m0e | kaldemar, is there a way to know if its a transcoding problem? |
| 11:58.34 | m0e | though.. i've had a sip call one end on alaw and the other on g729 (and was working) |
| 11:58.40 | *** join/#asterisk lftsy (n=mlr@install.deckpoint.ch) |
| 11:58.45 | TommyBotten | really hates the idea of patents on IP |
| 11:58.58 | kaldemar | m0e: sure, enable sip debug, make a call, and look at CLI output. or even better, pastebin it and someone will surely take a look. |
| 11:59.18 | m0e | ok.. will do that thanks |
| 12:02.55 | gladier | hey guys - im having to change from a binary install of asterisk to a compiled version so i can run chan-sccp-b. anyways i have a list of about 15 modules i need to recompile - http://pastebin.com/m13d89a88 and i'm struggling to find them on the net |
| 12:04.07 | gladier | can anyone point me in the right direction? |
| 12:04.08 | *** join/#asterisk jackal (n=jackal@pool-96-247-204-126.clppva.fios.verizon.net) |
| 12:04.35 | kaldemar | gladier: http://downloads.asterisk.org/pub/telephony/asterisk/ <-- choose a version and download the whole package, some of the modules in your list are in asterisk-addons |
| 12:04.42 | m0e | http://pastebin.com/d69f5dbcc |
| 12:05.25 | *** join/#asterisk |Cybex| (n=John@atwork-21.r-212.178.82.atwork.nl) |
| 12:08.35 | kaldemar | m0e: looks like asterisk finds a common codec, so no obvious errors there. try disable g729 and try again. |
| 12:08.41 | m0e | just that I'm clear, I want to make sure that my dahdi config is correct.. I have the TDM800P (2 FXO modules with echo cancellation).. I have this in system.conf (this is just for 1 channel) |
| 12:09.00 | m0e | fxsks=1 and echocanceller=mg2,1 |
| 12:09.47 | m0e | according to the dahdi docs, i should use fxsks for FXO cards (something about reversing them) |
| 12:10.43 | kaldemar | DAHDI/2-1 was the one used in your paste, so that is irrelevant. |
| 12:10.46 | *** join/#asterisk jmworx (n=jeval@70.54.254.106) |
| 12:10.54 | m0e | ok |
| 12:11.16 | m0e | sorry, this is all very new to me, so I might be saying some stuff that makes no sense :) |
| 12:11.18 | kaldemar | but yes, the signaling should be fxsks. for an FXO device. |
| 12:13.01 | m0e | I tried with alaw (for the sip extention) but now I hear a hissing noise on the sip phone (still i can hear voice on the other end) |
| 12:14.03 | gladier | kaldemar: ok i've gotten asterisk-addons installed - any pointers on the rest? |
| 12:14.50 | kaldemar | m0e: something might be wrong with your nat settings, btw. if you look at where the options messages are actually going. |
| 12:15.27 | kaldemar | gladier: they're either in asterisk, obsolete or renamed. |
| 12:16.07 | kaldemar | gladier: see UPGRADE*.txt for the applications you're using. |
| 12:16.23 | m0e | hmm. |
| 12:16.29 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 12:16.40 | m0e | OPTIONS sip:oZFDJ_lHtwhYUmjCYTx9@192.168.10.66;transport=UDP SIP/2.0 (what should the .66 ip be for?) |
| 12:16.47 | kaldemar | gladier: hope you're not compiling an older version than you had from the binary package. |
| 12:16.51 | m0e | cause my phone has the .65 IP which is weird |
| 12:17.38 | kaldemar | m0e: with the nat settings you better ask around in #freepbx since it seems to be generating your configs. |
| 12:17.53 | kaldemar | in case you've dropped freepbx: |
| 12:17.55 | drmessano | FreePBX has nothing to do with NAT settings |
| 12:17.56 | kaldemar | ~sipnat |
| 12:17.57 | infobot | [~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 |
| 12:18.15 | kaldemar | drmessano: does it not support nat in any way? |
| 12:18.28 | drmessano | FreePBX doesnt care about your NAT.. that' |
| 12:18.32 | drmessano | FreePBX doesnt care about your NAT.. that's done outside of it |
| 12:18.42 | m0e | that is the howto I got for sip->sip to work... but I cant get it from sip->dahdi or dahdi->sip |
| 12:19.02 | kaldemar | but it does generate the device configs. |
| 12:19.06 | drmessano | So ~sipnat applies here |
| 12:19.37 | kaldemar | so it might overrun manual modifications. i don't know since i don't use it. :P |
| 12:19.48 | drmessano | Right, you dont.. I do |
| 12:20.12 | *** join/#asterisk pawpro (n=Miranda@213.166.12.34) |
| 12:20.13 | m0e | hmm.. i'll go digging around a bit more try to find anything and I'll come back and bother you guys a bit more :P |
| 12:20.17 | kaldemar | your show then. :) |
| 12:20.27 | gladier | kaldemar: oops :P never realsied there was a make menuconfig - i've gotten too used to binary systems. :P |
| 12:20.57 | pawpro | Hello everybody. I can't find anywhere to change the default CLI (if no CLI info available) from "asterisk" to something else. Where can I do that? |
| 12:21.05 | drmessano | m0e pastebin your NAT config and tell me which file it's in, pls |
| 12:22.04 | m0e | http://pastebin.com/d69af1365 |
| 12:22.07 | m0e | sip_general_custom.conf |
| 12:22.39 | m0e | the externip clause is put in externip.conf (on its own) ad I check that every 10 minutes and replace it |
| 12:24.12 | kaldemar | pawpro: it uses your hostname, if the CLI prompt is what you mean |
| 12:24.17 | drmessano | Invalid pastebin |
| 12:24.21 | TommyBotten | Doing a dial do a dahdi channel, where an ATA is connected - shouldn't Dial(dahdi/1) be enough, as long as the device is connected to the first channel? |
| 12:25.02 | drmessano | m0e: Ummm? |
| 12:25.06 | kaldemar | TommyBotten: yes, unless you want to pass a number on the channel |
| 12:25.16 | m0e | sorry http://pastebin.com/d206ba4e7 |
| 12:25.25 | TommyBotten | kaldemar: I really don't see the point. It's a "dumb" ata. Analog stuff. |
| 12:25.46 | kaldemar | TommyBotten: but you gotta be quite kinky to connect an ATA to an analog channel. |
| 12:26.07 | TommyBotten | kaldemar: How come? |
| 12:26.13 | pawpro | kaldemar: I meant caller id |
| 12:26.17 | m0e | drmessano, i just removed my include and put the externip clause in there (where it should be) |
| 12:26.28 | drmessano | ok |
| 12:26.28 | kaldemar | pawpro: what protocol? |
| 12:26.45 | pawpro | sip |
| 12:27.04 | TommyBotten | kaldemar: ATA wasn't quite correct. It's a FAX unit. |
| 12:27.09 | TommyBotten | kaldemar: An analog telephony adapter, or analog telephone adapter, (ATA) is a device used to connect one or more standard analog telephones to a digital and/or non-standard telephone system such as a Voice over IP based network. |
| 12:27.13 | TommyBotten | [edit] eh |
| 12:27.14 | kaldemar | TommyBotten: what exactly are you connecting to what? do you have an analog card in your asterisk box? |
| 12:27.15 | TommyBotten | sorry |
| 12:27.18 | drmessano | m0e: Do you have G729 installed? |
| 12:27.23 | pawpro | kaldemar: sip |
| 12:27.44 | kaldemar | TommyBotten: if the ATA is speaking SIP to asterisk, you dial it with SIP/..., not DAHDI/... |
| 12:27.46 | m0e | I believe so.. I've made sip->sip calls with it (one device using alaw and the other using g729) |
| 12:28.10 | m0e | and I've tested the same one way audio from sip->dahdi with alaw (same problem) |
| 12:28.23 | kaldemar | pawpro: there's a parameter for that in sip.conf |
| 12:28.35 | drmessano | g729 <> g729 doesnt transcode |
| 12:28.43 | drmessano | dahdi to g729 does |
| 12:28.54 | m0e | how about alaw -> g729? |
| 12:28.58 | drmessano | yes |
| 12:29.08 | m0e | then it shouldnt be that |
| 12:29.14 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 12:29.16 | m0e | just to make sure, i'll double check |
| 12:29.42 | pawpro | kaldemar: callerid = "" ohhh how did i miss that |
| 12:29.46 | drmessano | Is the phone using the internal IP or external hostname? |
| 12:29.51 | pawpro | kaldemar: thanks |
| 12:29.55 | m0e | internal |
| 12:30.02 | TommyBotten | kaldemar: I wasn't very precise. It's a FAX unit connected to it. Analog stuff. |
| 12:30.09 | TommyBotten | kaldemar: And the CLI says app_dial.c:1547 dial_exec_full: Unable to create channel of type 'dahdi' (cause 0 - Unknown) |
| 12:30.19 | drmessano | remove your fromdomain, reload and try it |
| 12:30.25 | m0e | ok |
| 12:30.53 | kaldemar | TommyBotten: sure, you need to speak SIP to the ATA. it's not some transparent magical DAHDI interface to asterisk. :) |
| 12:31.25 | m0e | same thing |
| 12:31.54 | kaldemar | TommyBotten: assuming the scenario is asterisk <- [SIP] -> ATA <- [analog] -> [FAX device] |
| 12:31.55 | m0e | btw.. i keep getting this (Got SIP response 415 "Unsupported Media Type" back from 192.168.10.1).. it happens with sip->sip too.. but it works with sip |
| 12:33.05 | TommyBotten | kaldemar: It's not a ATA. I mixed up terms. The scenario is [FAX device] <-> [wctdm24xxp+ Wildcard AEX800] <-> [Asterisk via dahdi] |
| 12:33.28 | m0e | i'll be back in a bit (conf call) |
| 12:36.44 | kaldemar | TommyBotten: oh ok. then DAHDI/1 should be fine. check with "module show like dahdi" that you have the module loaded. |
| 12:38.08 | TommyBotten | It's there. And dahdi show status seems fine as well. Is it case sensitive? |
| 12:40.36 | kaldemar | TommyBotten: not 100% sure, worth trying. other channels types are not. |
| 12:40.42 | TommyBotten | kaldemar: And do I need spans setup for analog lines aswell? |
| 12:40.44 | kaldemar | TommyBotten: does dahdi show channels print you the channel? |
| 12:40.55 | kaldemar | TommyBotten: no, spans are not needed. |
| 12:42.42 | TommyBotten | All the channels show up |
| 12:43.09 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 12:43.20 | TommyBotten | But ... in my dahdi-channels.conf it says ;;; line="1 WCTDM/0/0" ... TDM isn't Time divisional multiplexing? |
| 12:43.24 | TommyBotten | In that case, wrong card |
| 12:44.04 | *** join/#asterisk voipmonk (n=voipmonk@69.172.100.53) |
| 12:45.20 | kaldemar | TommyBotten: that's just a comment |
| 12:46.27 | TommyBotten | Hmm.. how about the setting in /etc/dahdi/system.conf fxsks=1 |
| 12:46.41 | TommyBotten | Isn't it supposed to be set to FXO? |
| 12:46.51 | kaldemar | yes |
| 12:47.42 | TommyBotten | Great. Thanks :) |
| 12:47.45 | kaldemar | and signaling as fxo_ks in chan_dahdi.conf |
| 12:48.13 | *** join/#asterisk Failrar (n=Failrar@2001:470:1f15:316:2a0:d1ff:fe4e:e802) [NETSPLIT VICTIM] |
| 12:49.04 | kaldemar | butbut, is the channel on the card really an FXS? with older cards and drivers zaptel/dahdi/asterisk complained if the fxo/fxs setting was wrong. |
| 12:49.27 | *** join/#asterisk mintos (n=mvaliyav@117.206.17.247) |
| 12:50.28 | TommyBotten | Hmm |
| 12:50.48 | TommyBotten | dahdi_cfg says "Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling?" |
| 12:52.03 | TSM2 | Does SDSL exist in US? |
| 12:52.10 | kaldemar | how is the layout of your card? which channels are FXO and which FXS? |
| 12:52.51 | TommyBotten | I really want FXS only. Or what I means is, that I have three faxes that needs to be connected to it |
| 12:52.56 | kaldemar | TommyBotten: you should connect the FAX device in an FXS channel and configure it as fxoks in system.conf and fxo_ks in chan_dahdi.conf |
| 12:52.57 | *** join/#asterisk jmworx (n=jeval@mail.octasic.com) |
| 12:53.28 | TommyBotten | changing to fxoks in system.conf just gived me the error message |
| 12:54.15 | kaldemar | then the port you're configuring is FXO |
| 12:54.47 | *** join/#asterisk lau (n=lau@unaffiliated/lau) |
| 12:55.16 | TommyBotten | What do you mean by that? |
| 12:55.39 | lau | hello I am noob at VoIP, but know a little bit about GNU/Linux systems |
| 12:56.08 | lau | I have a SPA2102 FXS |
| 12:56.35 | lau | and I have a VoIP provider (different from my IAP) |
| 12:56.59 | lau | I use my IAP modem in order to reach the VoIP outside network |
| 12:57.10 | lau | is that clear ? |
| 12:57.16 | kaldemar | TommyBotten: analog ports are either FXO or FXS. phones, fax machines and alike are FXO devices that need to be connected to an FXS port on your card. FXO ports are used to connect a phone line to your box. |
| 12:57.22 | lau | I mean does this description make sense for the moment ? |
| 12:57.25 | voipmonk | lau: what do you want to do? |
| 12:57.37 | *** join/#asterisk Tim_Toady (n=moi@194.219.192.79.dsl.dyn.forthnet.gr) |
| 12:57.41 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 12:57.55 | kaldemar | TommyBotten: to be a bit confusing, FXS ports need to be configured to use FXO signaling and vice versa. |
| 12:58.44 | lau | voipmonk: I want to setup a VoIP at home (3 IP phones, 1 Analog phone) |
| 12:58.59 | *** join/#asterisk scalex000 (n=chatzill@133puntacana02.codetel.net.do) |
| 12:59.11 | scalex000 | hi guy I have a problem |
| 12:59.15 | lau | voipmonk: I want to be able to be redirect call from one home phone to another home phone |
| 12:59.31 | lau | voipmonk: I want to be able to put people on hold with music |
| 12:59.33 | scalex000 | my IAX unregister and not register again |
| 12:59.45 | lau | voipmonk: I want to call outside |
| 12:59.49 | TommyBotten | kaldemar: Aha... So you're saying my system.conf ahd dahdi_channels.conf should actually be fxo ? |
| 12:59.52 | kaldemar | lau: set asterisk to communicate with your VoIP provider, and the VoIP phones and SPA2102 with asterisk. |
| 13:00.51 | kaldemar | TommyBotten: yes, but which ones, depends on the layout of your card. the card comes with different hardware module combinations that define what channels you have. |
| 13:02.48 | TommyBotten | kaldemar: Aha... So it is possible that I only have FXS hardware? |
| 13:03.27 | kaldemar | TommyBotten: sure, but in this case FXS hardware is what you need. |
| 13:03.32 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:03.46 | TommyBotten | Aha... ok |
| 13:04.04 | scalex000 | hello, why when I use iax2 show peers, appear host unspecified |
| 13:04.25 | *** join/#asterisk zoap (n=zoap@53501AB4.static.ziggozakelijk.nl) |
| 13:04.46 | [TK]D-Fender | scalex000: Did you specify it? |
| 13:05.01 | [TK]D-Fender | scalex000: Is it dynamic AND registered? |
| 13:05.06 | kaldemar | 14:59 < scalex000> my IAX unregister and not register again |
| 13:05.32 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.53.211) |
| 13:05.42 | scalex000 | well I check iax2 show registry and see it was a registration in both side |
| 13:06.01 | scalex000 | but when I use this command iax2 show peers |
| 13:06.13 | scalex000 | only the host appear unspecified |
| 13:06.37 | TommyBotten | kaldemar: So if my dahdi_scan says port=1,FXO |
| 13:06.45 | TommyBotten | kaldemar: That means I have the wrong hardware |
| 13:06.45 | TommyBotten | ? |
| 13:07.37 | Gido-E | TommyBotten go read about fxo and fxs |
| 13:08.14 | [TK]D-Fender | [08:05]<kaldemar>14:59 < scalex000> my IAX unregister and not register again |
| 13:08.32 | kaldemar | TommyBotten: i think so, i'm not familiar with the command and what output it provides. |
| 13:08.52 | TommyBotten | Gido-E: Believe you me, I have tried |
| 13:09.14 | kaldemar | TommyBotten: if your hardware is FXO only, you have wrong hw modules for the card. |
| 13:09.50 | scalex000 | TK, theres something wrong when I use iax |
| 13:09.57 | kaldemar | TommyBotten: you need the green ones, unless they have decided to change the colors at some point. :P |
| 13:11.44 | *** join/#asterisk sasargen (i=d04a8d70@gateway/web/freenode/x-wftznwponsznyvof) |
| 13:11.48 | TommyBotten | kaldemar: Much appreciated :) |
| 13:11.48 | lau | I think this discussion may SPA2102 is only a gateway Analo phone/IP network right ? |
| 13:12.05 | lau | s/I think this discussion may // |
| 13:12.07 | scalex000 | TK, I don't know how to explain |
| 13:13.00 | scalex000 | TK, I have 2 asterisk using IAX, one behind Nat, the matter is something I lost connection on both side |
| 13:13.11 | *** join/#asterisk _bradk (n=brad@unaffiliated/-bradk/x-9249860) |
| 13:13.12 | scalex000 | tk, when I try to reload not work |
| 13:14.42 | kaldemar | lau: yes |
| 13:20.23 | lau | again i am still noobing with VoIP but why should I need Asteriks |
| 13:20.29 | lau | what is the value added ? |
| 13:20.55 | lau | I can use my VoIP provider web manager configuration account in order to tune everything cannot I ? |
| 13:21.03 | voipmonk | sure you can |
| 13:21.06 | voipmonk | go for it |
| 13:22.57 | voipmonk | lau - http://www.asterisk.org/support/features are a few features, there are more features you can bring to your installation after you read through the included documentation |
| 13:24.38 | *** join/#asterisk Ast001 (n=uros@89.216.186.50) |
| 13:25.36 | Ast001 | hello I am unable to compile dahdi driver on my pclinuxos box. It complains for /lib/cpp like this : C preprocessor "/lib/cpp" fails sanity check |
| 13:26.27 | Ast001 | I think I have all necessary for compiling including bison, libtermcap ,zlib,openssl,ncurses,kernel-sources |
| 13:27.04 | [TK]D-Fender | Ast001: Apparently you don't |
| 13:27.34 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 13:27.37 | Ast001 | what do you think I am missing ? |
| 13:27.52 | [TK]D-Fender | Ast001: cpp just like it says. |
| 13:27.59 | [TK]D-Fender | Ast001: Which is probably the G++ stuff |
| 13:29.00 | *** join/#asterisk TheDavidFactor (n=chatzill@fw1.safedataisp.net) |
| 13:29.05 | Ast001 | gcc3.3-c++ ? |
| 13:29.13 | *** part/#asterisk anonymouz666 (n=anonymou@189.24.53.211) |
| 13:29.38 | [TK]D-Fender | Ast001: G++ |
| 13:29.56 | Ast001 | when I type g++ at synaptic it shows me only that |
| 13:30.22 | *** join/#asterisk e4 (n=e4@76.79.59.194) |
| 13:31.57 | *** join/#asterisk retentiveboy (n=pdugas@atl.pra-corp.com) |
| 13:32.08 | [TK]D-Fender | Ast001: Go follow the debian install instructions for it then |
| 13:32.19 | *** part/#asterisk TommyBotten (n=noosjent@217-14-12-26-dhcp-osl.bbse.no) |
| 13:33.44 | Ast001 | ok |
| 13:34.09 | Ast001 | one more quick question for you. Can X100P card be forced to work with recent dahdi driver ? |
| 13:34.25 | Ast001 | for version 1.6 of Asterisk |
| 13:35.24 | [TK]D-Fender | Ast001: DAHDI dropped support for it IIRC |
| 13:35.59 | Ast001 | so should I use older zaptel driver ? |
| 13:36.13 | [TK]D-Fender | Ast001: clearly |
| 13:36.52 | Ast001 | hm ok thanks for help . |
| 13:37.16 | *** part/#asterisk icyValk77 (n=icyValk7@213.129.64.4) |
| 13:37.37 | *** part/#asterisk Ast001 (n=uros@89.216.186.50) |
| 13:39.48 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:44.11 | drmessano | I can't use 4 X100P cards on my servers anymore? |
| 13:44.16 | drmessano | CRAPOLA |
| 13:48.11 | *** join/#asterisk ccesario (n=ccesario@189-19-6-236.dsl.telesp.net.br) |
| 13:51.38 | *** join/#asterisk kannan (n=kann@121.246.242.95) |
| 13:54.23 | *** join/#asterisk coppice (n=chatzill@50.131.92.116.dyn.pacific.net.hk) |
| 13:55.28 | kannan | hello, I have 2 E1 Digium cards, asterisk 1.4.27.1. 1st card is to PSTN telco, and the other is a Siemens HiPath 4000 PBX with which i want to integrate. The Siemens E1 span shows yellow alrm. My configs are : http://pastebin.ca/1710953 . Please advise how to troubleshoot |
| 13:56.26 | [TK]D-Fender | kannan: You should not be TAKING timing from your Seimens, you should be providing it |
| 13:56.59 | [TK]D-Fender | kannan: span=2,1,0,ccs,hdb3,crc4 <- SB 2,0,0 |
| 13:59.40 | kannan | [TK]D-Fender, thanks, i will try |
| 14:00.26 | creativx | so [TK]D-Fender, do you think we should upgrade from 1.2 soon. he he |
| 14:00.51 | [TK]D-Fender | creativx: Procrastination : the art of keeping up with yesterday |
| 14:01.13 | creativx | bwaha indeed |
| 14:01.20 | creativx | it must be some 12 months since last time i asked you that now i think |
| 14:01.21 | creativx | hehe |
| 14:01.28 | creativx | thats some hardcore procrastination right there |
| 14:02.37 | *** join/#asterisk TommyBotten (n=noosjent@217-14-12-26-dhcp-osl.bbse.no) |
| 14:02.44 | *** join/#asterisk yang (n=yang@freenode/sponsor/cacert.assurer.yang) |
| 14:02.52 | *** part/#asterisk DMeloUK (n=DominicM@127.159.119.70.cfl.res.rr.com) |
| 14:02.52 | creativx | if it hadnt been that our itsp is terminating their entire iax2 product line, i dont think i would have had to touch that box in another 12 months =) |
| 14:03.15 | yang | I am trying to memorize all asterisk ports required to place calls - SIP 5060, RTP 10000-20000 are there any extra, which I forgot ? |
| 14:04.31 | *** join/#asterisk puzzled (n=foobar@83.163.53.136) |
| 14:04.50 | ehsjoar | LLL |
| 14:04.55 | *** part/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 14:05.07 | kannan | [TK]D-Fender, now the alarm on span 2 shows yell/rec |
| 14:05.35 | kannan | and now back to yellow |
| 14:07.52 | [TK]D-Fender | kannan: Was this previously functioning? |
| 14:08.15 | kannan | no, first time, the siemns of course was working fine, the asterisk works fine with the telco |
| 14:08.16 | [TK]D-Fender | yang: that is the inbound SIP port, and RTP |
| 14:08.29 | kannan | i want to set * as a pass thru between the telco and the siemns |
| 14:08.38 | yang | yes, trying to apply simple QOS rules |
| 14:08.40 | kannan | should we specify termtype as something ? |
| 14:12.04 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:13.12 | x86 | heya _ShrikE |
| 14:13.14 | x86 | ltns |
| 14:14.34 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-yxblflidujsbjyzi) |
| 14:15.11 | *** join/#asterisk cusco (n=trilili@2001:0:53aa:64c:2c82:168a:a077:acde) |
| 14:15.12 | cusco | woa |
| 14:15.17 | cusco | menuselect is different |
| 14:16.04 | cusco | how nice... should still be able to use the spacebar to select/deselect an app |
| 14:16.07 | cusco | item |
| 14:20.06 | ManxPower-work | looks around |
| 14:20.54 | *** join/#asterisk dudee (n=pocketir@124.155.195.6) |
| 14:21.12 | ManxPower-work | yang the RTP ports are defined in /etc/asterisk/rtp.conf |
| 14:21.18 | dudee | Hi there, i have some problem and hoping for some guru to help me. I have a Ring group that rings all the analog interface (FXS) modules. the thing is if an agent is on the phone and a new call comes him, he cannot answer the phone even though |
| 14:21.24 | dudee | callwaiting = yes |
| 14:21.27 | dudee | please help |
| 14:21.53 | yang | ManxPower-work: I know, thank you...I just thought if there are any other ports left, I can see 5038 and 4569 (IAX) |
| 14:22.03 | ManxPower-work | dudee: Are you using FreePBX/AsteriskNOW? |
| 14:22.14 | *** join/#asterisk brad_mssw (n=brad@216.155.101.90) |
| 14:22.14 | dudee | yes |
| 14:22.20 | ManxPower-work | ~freePBX |
| 14:22.21 | infobot | [~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 |
| 14:22.48 | dudee | that bad? |
| 14:23.12 | ManxPower-work | dudee: Not bad. Different. |
| 14:23.16 | dudee | using freepbx with asterisk |
| 14:23.32 | Katty | BRR |
| 14:23.36 | Katty | wtf |
| 14:23.40 | Katty | who authorized this weather |
| 14:23.58 | ManxPower-work | Katty: needs a squirrel coat. |
| 14:24.12 | Katty | i need a visit to florida |
| 14:24.59 | dudee | ManxPower : are the dial plans really that different? |
| 14:26.01 | dudee | freepbx is just not as active as asterisk channel |
| 14:26.32 | Katty | sighs |
| 14:26.36 | ManxPower-work | dudee: You are basically asking a Windows question on a #DOS channel. Is Windows really that different from DOS. I mean Windows runs on top of DOS, doesn't it? |
| 14:26.58 | TommyBotten | ManxPower-work: Not anymore .... .. :> |
| 14:27.03 | dudee | the kernel has been rebuilt |
| 14:27.08 | ManxPower-work | TommyBotten: Windows95 then. |
| 14:27.11 | dudee | so not anymore |
| 14:27.13 | TommyBotten | :) |
| 14:27.13 | TommyBotten | hehehe |
| 14:27.17 | TommyBotten | sorry for nitpicking |
| 14:27.17 | dudee | haha |
| 14:27.21 | Katty | dudee: man, just take a hint already |
| 14:27.22 | dudee | ok get your point |
| 14:27.22 | creativx | nitting |
| 14:27.25 | ManxPower-work | ~guis |
| 14:27.25 | infobot | guis is, like, "FreePBX/Trixbox is to Asterisk as Windows 95 is to DOS" |
| 14:27.31 | Katty | dudee: do not ask here, you are not going to get help |
| 14:27.42 | creativx | i enjoyed win95 |
| 14:27.44 | dudee | katty can see that |
| 14:27.52 | dudee | at least can chat |
| 14:27.58 | Katty | then chat |
| 14:28.00 | ManxPower-work | Katty: especially because FreePBX uses a huge complicated AGI script for call routing and deciding on recording, call waiting, etc. |
| 14:28.06 | Katty | and stop pestering poor manx about freepbx |
| 14:28.06 | dudee | while waiting for help in freepbx |
| 14:28.24 | dudee | ok noted |
| 14:28.31 | Katty | cheers. |
| 14:28.41 | dudee | ) |
| 14:28.41 | Katty | in other mews, i got critter server officially in place this morning |
| 14:28.50 | Katty | annnnnd my sata drive in my gaming machine died :/ |
| 14:29.08 | Katty | Major Bummer. |
| 14:29.20 | TommyBotten | critter server? |
| 14:29.28 | kaldemar | some newer analogy for guis would be in place. :) |
| 14:29.47 | TommyBotten | It's like running FreePBX on top of asterisk |
| 14:29.48 | TommyBotten | oh, wait |
| 14:29.54 | kaldemar | some people were born yesterday. :P |
| 14:29.59 | Katty | infobot: critter cam |
| 14:30.00 | infobot | i guess critter cam is http://www.ustream.tv/channel/squirrel-critter-cam |
| 14:30.00 | dudee | on a seperate note, any idea to reduce irq misses? managed to recude from 79 to 22/23..but would like it to be 0..sign..advice/recommendations? |
| 14:30.04 | Katty | ^- critter server |
| 14:30.06 | ManxPower-work | dudee: Calling a phone in regular Asterisk can be as little as 1 line in the dialplan. In FreePBX it's something between 100 - 200 lines in the dialplan for the exact same thing. |
| 14:30.15 | dudee | done most of the web advicea |
| 14:30.29 | TommyBotten | Katty: Yours ? |
| 14:30.34 | Katty | yes |
| 14:30.38 | Katty | infobot: forget critter cam |
| 14:30.38 | infobot | Katty: i forgot critter cam |
| 14:30.38 | TommyBotten | Neat |
| 14:31.03 | Katty | infobot: critter cam is Katty's broadcast of The Nut House at http://ustre.am/8H5d |
| 14:31.04 | infobot | okay, Katty |
| 14:31.06 | dudee | ManxPower: i noticed that too |
| 14:31.21 | dudee | full of plans for simple entries |
| 14:32.27 | Katty | TommyBotten: it doubles as live video survlance for the front yard until i get that camera in housing |
| 14:32.45 | eppigy | nein |
| 14:32.48 | eppigy | DA |
| 14:32.50 | Katty | hello dear |
| 14:32.54 | eppigy | hiya |
| 14:32.55 | eppigy | :> |
| 14:32.59 | Katty | shall we hug |
| 14:33.02 | Katty | this fine morning |
| 14:33.04 | eppigy | huggle Katty |
| 14:33.08 | eppigy | HUGGLES |
| 14:33.10 | eppigy | HEARTLES |
| 14:33.11 | dudee | how many tvlines for the camera? |
| 14:33.18 | Katty | hugs eppigy |
| 14:33.45 | dudee | 480? |
| 14:33.45 | Katty | dudee: most average cameras are 16 channel i believe |
| 14:33.51 | dudee | 520? |
| 14:33.52 | TommyBotten | Katty: Ah... The webpage was not very nice to my browser, though |
| 14:33.54 | dudee | tvlines |
| 14:33.57 | Katty | meaning the software can handle 16 cameras at once |
| 14:34.02 | dudee | not channels |
| 14:34.15 | dudee | clarity of camera |
| 14:34.17 | Katty | idk, it's 30fps |
| 14:34.21 | Katty | and HD |
| 14:34.27 | Katty | which i forget the resolution exactly |
| 14:34.53 | Katty | uhhh not 60i |
| 14:34.57 | dudee | thats also not the tvlines |
| 14:35.08 | dudee | thats frames per seconds |
| 14:35.08 | Katty | well then make a numbe rup |
| 14:35.14 | Katty | i don't really care |
| 14:35.29 | creativx | birds |
| 14:35.35 | Katty | hmm yes. birds. |
| 14:35.40 | creativx | car |
| 14:35.41 | creativx | ! |
| 14:35.41 | dudee | is it dome or not? |
| 14:35.47 | eppigy | 2^128 |
| 14:36.01 | Katty | dudee: you're viewing through a logitcal c600 720p webcam |
| 14:36.11 | Katty | dudee: there currently is NOT a network camera. |
| 14:36.13 | yang | eppigy: you are Dave ! |
| 14:36.13 | creativx | more birds |
| 14:36.25 | eppigy | yang: I AM |
| 14:36.25 | creativx | oh hai zoom |
| 14:36.36 | Katty | yeah i left the face zoom thing on. i'll turn it off |
| 14:36.46 | creativx | ahah |
| 14:36.49 | creativx | i remember that |
| 14:36.58 | creativx | some half assed code controlling that motor |
| 14:37.12 | Katty | it's irritating |
| 14:37.19 | creativx | it is poorly tested =) |
| 14:37.35 | Katty | i wouldn't mind a set rotation |
| 14:37.56 | eppigy | GIRL WORK ME INTO YOUR ROTATION |
| 14:38.09 | eppigy | impulse problems |
| 14:38.29 | Katty | ^_- |
| 14:41.52 | Katty | the birdies sure are hungry this morning |
| 14:42.04 | creativx | i stopped watching |
| 14:42.06 | creativx | can you give me an update |
| 14:42.08 | creativx | what is happening |
| 14:42.21 | Katty | aliens are lighting the law on fire |
| 14:42.30 | Katty | s/law/lawn/ |
| 14:42.33 | creativx | HOT SHIT! |
| 14:42.33 | *** join/#asterisk gaetronik (n=gaetan@LPuteaux-156-15-5-233.w80-14.abo.wanadoo.fr) |
| 14:42.43 | creativx | better call the lawman |
| 14:42.44 | eppigy | that is just me |
| 14:42.50 | eppigy | please remain calm |
| 14:42.52 | *** join/#asterisk RoPBX (n=IceChat7@200.93.61.20) |
| 14:42.54 | Katty | k |
| 14:43.14 | Katty | while you're in my front law, would you mind waking the squirrels up |
| 14:43.21 | Katty | CAR |
| 14:43.32 | Katty | oh hey! good job eppigy. one just woke up. |
| 14:44.05 | RoPBX | hi all |
| 14:44.05 | eppigy | yes |
| 14:44.14 | eppigy | one does what one can |
| 14:44.27 | Katty | haha. did you see the squirrel |
| 14:44.32 | Katty | climb up the pole, and slide back down |
| 14:44.53 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 14:45.11 | RoPBX | please anyone knows about t1 cards? |
| 14:45.45 | Katty | yes. |
| 14:47.50 | eppigy | i am a t1 card |
| 14:47.54 | [TK]D-Fender | RoPBX: I hear they let you connect T1's |
| 14:48.33 | Katty | eppigy: you would more appropriately be the t1 |
| 14:49.14 | creativx | its all tubes anyways |
| 14:51.11 | RoPBX | i have an extrange problem, when i send followme calls to t1 line, i get dial tone |
| 14:52.11 | *** join/#asterisk guyvdb_ (n=guy@dsl-240-182-94.telkomadsl.co.za) |
| 14:52.11 | RoPBX | if i call directly from an extension i can call fine, but if i send then from followme then i get a few seconds of dialtone and then the call is hung |
| 14:52.34 | Katty | sounds like a real pickle. let's see the call. |
| 14:53.12 | eppigy | I AM THE T1 |
| 14:54.33 | Katty | yes, dear, of course. |
| 14:54.39 | guyvdb_ | Hi, I have sip to sip working, DAHDI (pstn) to sip working but cannot dial out sip to DAHDI. When I attempt to dial out I get error "No translator path exists for channel typ DAHDI (native 0x4c) to 0x1 followed by unable to create channel... |
| 14:55.00 | guyvdb_ | If I do transcoder show on the CLI I get no DAHDI transcodes found |
| 14:55.11 | guyvdb_ | Any ideas? |
| 14:55.51 | Katty | personally |
| 14:55.52 | Katty | i'd have some pie |
| 14:56.00 | eppigy | t1 agrees |
| 14:56.08 | Katty | and possibly pastebin a call so others can see it |
| 14:56.14 | eppigy | time division multiplexinf of pie |
| 14:56.16 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 14:57.07 | Katty | how does one multiplex pie |
| 14:57.27 | eppigy | with a pie cutter |
| 14:57.27 | kaldemar | guyvdb_: what codec are you using on the SIP side? |
| 14:57.44 | Katty | eppigy: that would be the time division bit |
| 14:57.45 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 14:57.53 | Katty | eppigy: that's communication |
| 14:57.58 | eppigy | oh god |
| 14:57.59 | Katty | eppigy: how does one communicate with pie |
| 14:58.04 | RoPBX | my problem is like this: http://www.voipuser.org/forum_topic_16237.html |
| 14:58.09 | *** join/#asterisk ajohnson (n=aaron@65-122-4-130.dia.static.qwest.net) |
| 14:58.10 | eppigy | I have gotten my stuff all confussed |
| 14:58.23 | Zeeek | Katty: it's a very confussing subject |
| 14:58.23 | Katty | maybe the tin can the pumpkin came in |
| 14:58.24 | guyvdb_ | ulaw alaw g723 g729 |
| 14:58.25 | kaldemar | guyvdb_: G.723? |
| 14:58.26 | eppigy | proof that a system cannot know its own nature |
| 14:58.27 | Katty | with a string. and another tin. |
| 14:58.29 | guyvdb_ | yip |
| 14:58.34 | Zeeek | sits and contemplates his Allison Smith Asterisk orange USB bracelet |
| 14:58.44 | guyvdb_ | should i disable g723 |
| 14:58.49 | Katty | Zeeek: eppigy told me he is time division multiplexing pie |
| 14:59.02 | kaldemar | guyvdb_: disable G.723 and try again |
| 14:59.03 | Zeeek | pie is made to be muliplexed |
| 14:59.05 | Katty | Zeeek: verb. |
| 14:59.26 | Zeeek | verbs and bitters |
| 14:59.30 | eppigy | Katty: we can eat the same pie together |
| 14:59.30 | Katty | how does one communicate with pie on several letters |
| 14:59.33 | Zeeek | delicious in a drink |
| 14:59.35 | Katty | err levels |
| 14:59.35 | eppigy | that is multiplexing kinda |
| 14:59.54 | kaldemar | guyvdb_: G.723 is supported as passthrough only by default, it's a proprietary codec. |
| 14:59.55 | Katty | eppigy: that would be multiplexing BY pie |
| 15:00.00 | *** part/#asterisk lau (n=lau@unaffiliated/lau) |
| 15:00.24 | Zeeek | so we're going to learn about how to get more g729 throughput today |
| 15:00.25 | Katty | (time division) + multiplexing x pie |
| 15:00.32 | Zeeek | using pies |
| 15:00.40 | guyvdb_ | kaldemar: so what should i be using ulaw and alaw only |
| 15:00.41 | eppigy | i am at a loss here |
| 15:00.43 | Zeeek | and three of the pies are free |
| 15:00.58 | Katty | Zeeek: you're missing squirrels. |
| 15:01.10 | Katty | Zeeek: ask infobot about critter cam in /query |
| 15:01.11 | kaldemar | guyvdb_: use whatever codecs you have. |
| 15:01.12 | Zeeek | no, don't need squirels, this is hardware |
| 15:01.34 | eppigy | only pie is applicable to hardware |
| 15:01.48 | Katty | and only if you're multiplexing it |
| 15:01.50 | kaldemar | guyvdb_: looks like you don't have a G.723 codec, and for G.729 you need licenses so i'd disable it too. use gsm for lower bandwidth. |
| 15:01.52 | guyvdb_ | kaldemar: where can i list the codecs I have? |
| 15:01.52 | Katty | i mean by it |
| 15:02.00 | guyvdb_ | ok |
| 15:02.15 | Zeeek | So, some free g729 channels, see #vuc in 2 hours. And a Polycom ip335, the Les Paul of SIP phones! |
| 15:02.58 | Katty | blue jay! |
| 15:03.05 | *** join/#asterisk etfonhomey (n=etfonhom@74-143-192-74.static.insightbb.com) |
| 15:03.14 | Zeeek | Katty nice! |
| 15:03.16 | Katty | i bet a bluejay would multiplex a pie |
| 15:03.34 | eppigy | multiplex your face |
| 15:03.38 | Zeeek | who is doing this, you? |
| 15:03.38 | Katty | your mom's face |
| 15:03.47 | Katty | Zeeek: yes'r |
| 15:03.55 | eppigy | your entertainment center |
| 15:04.01 | Zeeek | Let me break it with more viewers |
| 15:04.09 | kaldemar | guyvdb_: module show like codec in CLI will list you codec modules. |
| 15:04.36 | guyvdb_ | kaldemar : seems to be connection on GSM |
| 15:04.47 | guyvdb_ | thx will do some more testing now |
| 15:05.36 | Katty | Zeeek: i don't htink that's possible. |
| 15:06.01 | Zeeek | well I added one so far and it didn't break |
| 15:06.01 | Katty | Zeeek: for ustream to be able to recieve and broadcast as many live feeds as they have... |
| 15:06.09 | Katty | Zeeek: something tells me you shan't be breaking it anytime soon |
| 15:06.16 | Katty | holy crap. the front yard is going crazy |
| 15:06.32 | Zeeek | Actually, we had over 100 viewers for the Mark Spencer "simulcast" a couple of years ago |
| 15:06.41 | Zeeek | on #vuc |
| 15:07.12 | drfreeze | Morning |
| 15:07.27 | drfreeze | What is a common way of handling transfers to voicemail |
| 15:07.39 | Zeeek | ~seen markster |
| 15:07.41 | infobot | i haven't seen 'markster', Zeeek |
| 15:07.44 | RoPBX | [TK]D-Fender, please any ideas with the t1 problem? did you see the link http://www.voipuser.org/forum_topic_16237.html |
| 15:07.53 | Zeeek | ~seen puzzled |
| 15:07.54 | infobot | puzzled is currently on #asterisk-dev (1h 3m 23s) #asterisk (1h 3m 23s), last said: 'Corydon76-dig: yes I have seen a few of those. doesn't make it any easier'. |
| 15:07.57 | drfreeze | Do you just make an extension for that? For example 555 dials the extension and 5551 goes directly to voicemail |
| 15:08.11 | Katty | if you want to |
| 15:08.12 | *** join/#asterisk catphish (n=catphish@gateway.atechmedia.net) |
| 15:08.19 | Katty | most people dial the number, then go to voicemail |
| 15:08.27 | Katty | or possibly use followme |
| 15:08.31 | Katty | followme is irritating |
| 15:08.38 | puzzled | Zeeek: you rang? |
| 15:08.49 | Zeeek | just testing the bot :) |
| 15:09.04 | guyvdb_ | kaldemar: i have now placed a call... :) thx for the help |
| 15:09.07 | RoPBX | well... i'm not using followme from asterisk, i made my own followme, without using the command ifself |
| 15:09.23 | kaldemar | guyvdb_: you're welcome |
| 15:09.30 | Katty | RoPBX: aye. that's what i do as well. |
| 15:09.54 | catphish | i am having a problem with call pickup, i have 2 phones in a pickup group, when one rings, and i dial *8 from another, the first phone is silenced, but the call isn't picked up, it simply continues to voicemail |
| 15:10.59 | Katty | i love it when people use these terms like Call Pickup |
| 15:11.02 | Katty | and Pickup Group |
| 15:11.36 | kaldemar | RoPBX: have you been to #freepbx already? |
| 15:11.52 | catphish | apologises for being a noob, but i assume everyone knows what i'm talking about :) |
| 15:12.47 | *** join/#asterisk Ickmund (n=magnus@cli-5b7ee15c.bcn.adamo.es) |
| 15:13.05 | Katty | catphish: what Phone System are you using? |
| 15:13.25 | catphish | asterisk+sip |
| 15:13.27 | drfreeze | catphish: Its called * 8 pickup, or star eight pickup |
| 15:13.38 | Katty | catphish: where did you buy it |
| 15:13.47 | drfreeze | And it means, litterly, you hit: *, 8 and then pickup the phone |
| 15:13.51 | catphish | i thoght it was called "Group call pickup" |
| 15:13.54 | drfreeze | nothing happens until you pickup |
| 15:14.17 | catphish | can't tell if / why Katty is mocking him |
| 15:14.51 | DocAwesome | ~seen kram |
| 15:14.52 | infobot | kram <n=kram@cm56-214-79.liwest.at> was last seen on IRC in channel #debian, 830d 19h 2m 56s ago, saying: 'hey guys, my debian decided to show me parts of gnome and apt outout in half geran since a few weeks, can anyone tell me how to make it speak english eagain?'. |
| 15:14.52 | Zeeek | Katty: I increased your audience by at least 50% |
| 15:14.52 | RoPBX | no kaldemar, i'm not using freepbx |
| 15:14.59 | Katty | Zeeek: horay! |
| 15:15.11 | Zeeek | only 830 days? |
| 15:15.30 | DocAwesome | well, guess Mark hasn't been on IRC in quite some time :) |
| 15:15.30 | Zeeek | that's right, I said the wrong pseudo |
| 15:15.46 | Katty | someone knocked the tray sideways :< |
| 15:15.54 | Kram | hey |
| 15:16.10 | Kram | who invokes my name in vain? |
| 15:16.15 | DocAwesome | I do! |
| 15:16.37 | Zeeek | is IRC case-sensitive? I guess so? |
| 15:16.56 | Katty | yes |
| 15:17.02 | catphish | Katty: I bought this asterisk from a man in the pub, it was only $100,000 |
| 15:17.11 | Katty | catphish: such a steal! |
| 15:17.24 | Zeeek | catphish: including shipping? |
| 15:17.46 | Katty | i wonder how much it would cost to ship 200lbs to alaska |
| 15:18.08 | coppice | does palin weight that much? |
| 15:18.18 | Katty | giggles insanely |
| 15:18.22 | Zeeek | coppice: :) |
| 15:18.31 | Katty | claps |
| 15:18.47 | catphish | my config doesn't want to play nice :( |
| 15:18.53 | coppice | I guess if you include the $150k of clothes she must |
| 15:19.19 | Zeeek | coppice what's this about a g722 library you wrote? |
| 15:19.29 | Katty | maybe we could ship her books back to her |
| 15:19.41 | Katty | 200lbs should cover the whole lot published, right? |
| 15:19.42 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 15:19.50 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 15:19.50 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:20.46 | Zeeek | those peanuts are making me hungry |
| 15:21.35 | coppice | Zeeek: what about it? |
| 15:21.46 | Katty | Zeeek: they're raw. |
| 15:21.52 | Katty | Zeeek: you wouldn't like them |
| 15:22.04 | Zeeek | I hadn't heard anything about it, everyone I know of uses that pjsip |
| 15:22.22 | coppice | many people use my G.722 |
| 15:22.33 | Zeeek | what SIP clients for OS X use it? |
| 15:23.09 | Katty | i believe zopier supports g722 |
| 15:23.18 | Katty | not the free version tho |
| 15:23.20 | Zeeek | It will soon but he's using the other lib |
| 15:23.28 | coppice | dunno. I just write it. I don't bother following who uses it |
| 15:23.42 | Katty | ^- palin's book |
| 15:23.43 | Zeeek | yes, I recall that from spandsp |
| 15:24.06 | Zeeek | Katty true I'd want em roasted |
| 15:24.18 | coppice | it seems the G.722 pjsip uses is the ancient one. its slow and it only handles one channel |
| 15:24.47 | Zeeek | coppice I wish you'd join us someday to talk about your work in this realm. A lot of people think you are Kilgore Trout |
| 15:25.05 | Zeeek | (does anyone need an explanation of Kilgore Trout?) |
| 15:25.13 | coppice | I do |
| 15:25.34 | Zeeek | then this is the obvious first link: http://en.wikipedia.org/wiki/Kilgore_Trout |
| 15:25.57 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 15:25.57 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:26.22 | Katty | ohoh, what's that guy's name which ones eleventy billion newspapers, magazines, myspace, |
| 15:26.25 | Katty | he's old. |
| 15:26.28 | Zeeek | coppice: let me finish by repeating that we'd love to hear from you anyime |
| 15:26.33 | Katty | murdock! |
| 15:26.40 | Zeeek | murDick |
| 15:26.53 | Zeeek | murdouchebag |
| 15:27.04 | Zeeek | murderofcrows |
| 15:27.22 | gaetronik | hi |
| 15:27.27 | Zeeek | hello |
| 15:28.09 | Katty | rupert |
| 15:28.26 | Zeeek | Simon Rupe |
| 15:28.55 | Zeeek | for the next 43 hours, I am a bachelor |
| 15:29.58 | Katty | oh? |
| 15:30.04 | Zeeek | coppice not this part: "Trout, who has supposedly written over 117 novels and over 2000 short stories, is usually described as an unappreciated science fiction writer whose works are used only as filler material in pornographic magazines." |
| 15:30.30 | Zeeek | Trout is a fictional character |
| 15:30.31 | Zeeek | $ |
| 15:30.36 | *** join/#asterisk Akiraa (n=Akira@79.112.12.171) |
| 15:30.37 | Zeeek | $EOL |
| 15:30.44 | Zeeek | \r\n |
| 15:30.58 | Katty | science fiction? |
| 15:31.03 | Katty | what /kind/ of porno magazines? |
| 15:31.13 | Zeeek | go read Wikipedia |
| 15:31.16 | ManxPower-work | Katty: Science Fiction Porn, of course! |
| 15:31.29 | coppice | Zeeek: well, if I can help the porn industry I'm content |
| 15:31.36 | ManxPower-work | I always liked Robby the Robot does The Andromeda Galaxy. |
| 15:31.49 | *** join/#asterisk jmworx__ (n=jeval@mail.octasic.com) |
| 15:32.00 | Zeeek | coppice in that case, you will especially appreciate hour 2 of the VUC, usually unrecorded, uncensored and rater R |
| 15:32.02 | Katty | Robby the Robot? |
| 15:32.07 | Katty | wasn't he on that forbidden planet |
| 15:32.13 | Zeeek | no, Robby of the Doors |
| 15:32.23 | Katty | ^_- |
| 15:32.25 | Katty | googles |
| 15:32.33 | Zeeek | goggles |
| 15:33.00 | Katty | ah ha! |
| 15:33.03 | Katty | it was forbidden planet |
| 15:33.08 | Katty | http://www.fanboy.com/archive-images/robby-the-robot-lifesized.jpg |
| 15:33.26 | Katty | http://theinvisibleagent.files.wordpress.com/2008/11/forbiddenplanet.jpg |
| 15:34.28 | gaetronik | mISDN_close: fid(18) isize(131072) inbuf(0xb7767008) irp(0xb7767008) iend(0xb7767008) |
| 15:34.28 | gaetronik | P[ 0] Got: 1,2,3,4 from get_ports |
| 15:34.28 | gaetronik | P[ 1] stack_init: Cannot add layer 4 to this port. |
| 15:34.34 | gaetronik | what does this mean? |
| 15:42.03 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:42.52 | *** join/#asterisk ticoit (n=ticoit@201.205.153.166) |
| 15:42.54 | Zeeek | I didn't know this after all these years: Polycoms don't support STUN at all |
| 15:43.00 | Zeeek | or do they? |
| 15:46.58 | *** join/#asterisk muiro (n=muiro@rrcs-24-56-88-130.ma.biz.rr.com) |
| 15:51.08 | ManxPower-work | Zeeek: standby |
| 15:51.37 | SuPrSLuG | Zeeek:no they don't |
| 15:51.42 | ManxPower-work | Zeeek: I have the Polycom SIP Admin Guide open. No string "STUN" in that document. |
| 15:51.56 | ManxPower-work | you seldom *need* STUN. |
| 15:52.31 | coppice | you seldom *need* to make a phone call |
| 15:53.10 | ManxPower-work | I make phone calls all the time without STUN. |
| 15:53.46 | coppice | and how many are really making the world a better place? |
| 15:54.06 | Zeeek | heh |
| 15:54.11 | ManxPower-work | hands coppice a lava lamp |
| 15:55.12 | *** join/#asterisk bcrisp (n=bcrisp@70.102.242.138) |
| 15:55.17 | bcrisp | mornin |
| 15:55.27 | Zeeek | agreed i hv 8 servers that work fine |
| 15:56.11 | bcrisp | * is trying to play a queue-minute sound that doesn't exist.. |
| 15:56.48 | bcrisp | /var/lib/sounds/en contains queue-minutes but not queue-minute |
| 15:57.31 | eppigy | brrr |
| 15:59.09 | Zeeek | ManxPower-work: the service that said I was sending my local IP is ukddi.com |
| 15:59.20 | Zeeek | it's the first time I've heard this from anyone |
| 15:59.40 | Zeeek | I searched all over and found several posts saying Polycom doesn't even support STUN at all |
| 16:00.20 | *** join/#asterisk [T]ank (n=[T]ank@206.71.78.158) |
| 16:00.30 | ManxPower-work | Zeeek: if you are sending your localIP then your localnet= is bad |
| 16:00.55 | [T]ank | does anyone know of a product/company that would allow me to compare a batch of telephone numbers and report back which of them are cell numbers? |
| 16:01.07 | ManxPower-work | localnet= specifies which network(s) to send the internal IP to and externip= specifies what IP should be used for non-local IPs |
| 16:01.23 | [T]ank | I know that you can go online and do a reverse phone look up kind of deal, but I am trying to find one where I can do a batch of numbers. |
| 16:01.29 | ManxPower-work | [T]ank: how many numbers? |
| 16:01.32 | Zeeek | I'm registering the phone directly no atserisk |
| 16:01.44 | [T]ank | Possibly thousands a day. |
| 16:02.13 | ManxPower-work | Zeeek: sucks to be you. Never did Polycom NAT stuff |
| 16:02.51 | [TK]D-Fender | Zeeek: Go listen to your first VUC brodcast again :) |
| 16:02.55 | coppice | Zeeek: pjsip seems to collect the world's worst implementations of various things |
| 16:03.04 | ManxPower-work | Polycoms have NAT support under Network/NAT |
| 16:03.15 | [T]ank | i have a huge list of numbers, but dont want to call cell phones... dont want to be a prick dialer company. |
| 16:04.14 | Zeeek | ManxPower-work: GASP: in the menu? |
| 16:05.14 | ManxPower-work | Try http://tnid.org/ they may have a bulk pay option |
| 16:05.34 | Zeeek | where do you see STUN in Poly setting? |
| 16:05.56 | ManxPower-work | Zeeek: STUN is not required for NAT |
| 16:06.06 | Zeeek | forgetting STUN, you mean you can fill in your ext ip? |
| 16:06.11 | Zeeek | forget STUN, pretend I didn't mention it |
| 16:06.24 | Zeeek | how do I force it to send the ext ip on that line? |
| 16:06.37 | Zeeek | I've never seen such a setting |
| 16:07.11 | ManxPower-work | try setting the "IP Address" to your external IP |
| 16:07.45 | Zeeek | where is ip address setting? I see nothing about that |
| 16:07.49 | ManxPower-work | under the Polycom NAT web interface |
| 16:08.16 | Zeeek | my web interface doesnt work, never has. There must be something in the cfg file that does it then? |
| 16:08.36 | Zeeek | I can see the web but changing things there doesn't work, not even reboot |
| 16:08.46 | Zeeek | IP650 btw |
| 16:09.26 | ManxPower-work | Zeeek: Page A-150 in the Polycom SIP Admin Guide 3.2.2 |
| 16:09.48 | ManxPower-work | Zeeek: the web page uses javascript. has to be enabled to use the web interface |
| 16:09.57 | Zeeek | ok I'll look at that later thx. Like you say, I have never needed anything like this before |
| 16:10.25 | Zeeek | ManxPower-work: actually it might work better in a different browser, some of my phones do. |
| 16:13.26 | Zeeek | ManxPower-work: actually, that part of the interface appears to have worked and the phone rebooted so we'll see if it lakes a diff |
| 16:14.20 | SuPrSLuG | Zeeek: you can add something like -> http://pastebin.com/m7815aa99 to phone.cfg |
| 16:14.44 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 16:15.42 | Zeeek | SuPrSLuG: thx ! |
| 16:16.13 | *** join/#asterisk ticoit (n=ticoit@201.205.153.166) |
| 16:16.48 | SuPrSLuG | Zeeek: NP |
| 16:16.51 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:17.00 | Zeeek | Speaking of javascript, the Siemens Gigasets work a lot better with Opera. SOmething in the ajax hates FF |
| 16:17.21 | bcrisp | opera and javascript?? |
| 16:17.34 | bcrisp | thats like oil and water |
| 16:17.44 | Zeeek | so before I go, and thanks for all the fish, we're giving away some free g729 licenses today in 40 minutes on #vuc http://vuc.me |
| 16:18.03 | Zeeek | bcrisp: I'm not an Opera fan, but I tested the phone web interface and it's much faster on Opera |
| 16:18.15 | Zeeek | in fact that's all I use Opera for |
| 16:18.16 | bcrisp | Zeeek: try it in chrome |
| 16:18.24 | Zeeek | I might! |
| 16:18.36 | bcrisp | opera's javascript is broken in a million ways |
| 16:18.37 | Zeeek | Wave works a lot better in Chrome than FF |
| 16:18.46 | Zeeek | FF had gotten some serious bloat lately |
| 16:19.04 | bcrisp | i hate to blaspheme but IE8 is quite fast |
| 16:19.38 | ppc | Yeah now you can compromise your system 2x as faster than you could with IE7! |
| 16:19.41 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 16:19.55 | coppice | I think IE7 was just a ruse to make IE8 look good |
| 16:19.57 | Zeeek | bcrisp: whatever works! |
| 16:20.08 | *** part/#asterisk [T]ank (n=[T]ank@206.71.78.158) |
| 16:20.14 | *** join/#asterisk lost_soul (n=shawn@cpe-74-71-234-100.twcny.res.rr.com) |
| 16:20.15 | bcrisp | cppice similar to vista vs win 7? |
| 16:20.35 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 16:20.53 | coppice | bcrisp: not really. XP still makes Win 7 look bad |
| 16:21.03 | bcrisp | true |
| 16:22.00 | *** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be) |
| 16:23.03 | bcrisp | hmm my queue is no longer ringing members |
| 16:24.16 | DocAwesome | XP is still pretty awesome |
| 16:24.17 | Tim_Toady | Is there a way to get the digium card serial on a remote machine from cli? |
| 16:24.29 | bcrisp | StarCraft is still awesome |
| 16:24.35 | DocAwesome | Tim_Toady: not that I'm aware of, but you could call Digium and ask :) |
| 16:24.41 | DocAwesome | I prefer Red Alert |
| 16:25.14 | *** join/#asterisk ticoit (n=ticoit@201.205.153.166) |
| 16:25.41 | bcrisp | i hate this network im on... |
| 16:25.55 | DocAwesome | ya, I hate freenode too :) |
| 16:25.55 | bcrisp | i think its screwing with my asterisk testing |
| 16:26.33 | bcrisp | all the sudden callers in the queue are not being connected with queue members |
| 16:26.36 | bcrisp | no warnings |
| 16:26.59 | *** join/#asterisk |Cybex| (n=John@212.178.82.21) |
| 16:29.02 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 16:30.53 | ManxPower-work | sounds like the network the phones are on is down |
| 16:31.14 | *** join/#asterisk mokmeister (n=mokmeist@109.78.107.178) |
| 16:38.27 | dan__t | 'morning. |
| 16:39.54 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 16:42.00 | *** join/#asterisk high-freq (n=hfreq@99.188.100.195) |
| 16:43.10 | high-freq | quick question, do i have to have a SIP provider for asterisk to work and dialout to regular numbers? and if so is there a free SIP provider out there to test this on |
| 16:43.15 | *** join/#asterisk dlynes (n=daniel@bas5-hamilton14-1242444445.dsl.bell.ca) |
| 16:43.20 | *** join/#asterisk ticoit (n=ticoit@201.205.153.166) |
| 16:43.43 | bcrisp | high-freq: yes, and i think no |
| 16:44.20 | *** join/#asterisk vitaminx (n=vitaminx@89.130.31.1) |
| 16:44.21 | bcrisp | high-freq: do you have any telephony hardware that you're using or just looking for SIP trunking? |
| 16:44.58 | bcrisp | ~roulette |
| 16:44.59 | infobot | ACTION watches bcrisp pull the trigger: Click! |
| 16:45.57 | high-freq | hmm i have a dlink hard phone here |
| 16:46.11 | high-freq | have it connected to a layer3 switch |
| 16:46.58 | bcrisp | high-freq: have you read "the book" ? |
| 16:47.04 | bcrisp | ~book |
| 16:47.05 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 16:47.19 | high-freq | ofcourse not |
| 16:47.22 | high-freq | ;) |
| 16:47.36 | bcrisp | high-freq: follow the free ebook pdf link and read it twice |
| 16:47.49 | bcrisp | high-freq: its extremely useful |
| 16:47.52 | high-freq | teacher gave us 2 phones and said to intsall asterisk so intalled in ubuntu and that was it lol |
| 16:47.56 | high-freq | kk |
| 16:48.55 | high-freq | ok will read and thx for reply |
| 16:49.02 | bcrisp | high-freq: no problem |
| 16:49.32 | *** join/#asterisk Tim_Toady (n=moi@77.49.116.218.dsl.dyn.forthnet.gr) |
| 16:50.07 | bcrisp | another thing that i'm seeing is that when the queue does actually attempt to dial an agent, if the agent doesn't answer the first time it won't try again |
| 16:50.30 | bcrisp | user is left with moh |
| 16:53.19 | *** join/#asterisk OmbreNoir (n=OmbreNoi@modemcable070.50-177-173.mc.videotron.ca) |
| 16:54.45 | ManxPower-work | bcrisp: without a pastebin of a failed call it's all just talk |
| 16:54.58 | bcrisp | im not seeing any |
| 16:55.26 | bcrisp | queue member is successfully registered, queue caller is on hold |
| 16:55.45 | bcrisp | rechecks queues.conf for typos |
| 16:57.01 | ManxPower-work | bcrisp: if there is no CLI output then you have major serious problems. |
| 16:57.19 | ManxPower-work | If there's no CLI output you think is important -- well let us decide if it's important or useful. |
| 16:57.32 | bcrisp | ManxPower-work: im trying again now |
| 16:58.46 | bcrisp | ManxPower-work: im thinking i dont have all the debugging messages i need |
| 16:58.51 | *** join/#asterisk xmitter (n=xmitter@c-24-21-212-187.hsd1.or.comcast.net) |
| 16:58.52 | *** join/#asterisk LemensTS (n=customgt@adsl-70-238-155-79.dsl.stlsmo.sbcglobal.net) |
| 16:59.03 | bcrisp | i sip set debug on |
| 16:59.55 | *** join/#asterisk outtolunc (n=me@c-98-248-96-110.hsd1.ca.comcast.net) |
| 17:00.01 | ManxPower-work | bcrisp: see /etc/asterisk/logger.conf |
| 17:00.06 | bcrisp | ty |
| 17:00.13 | LemensTS | what is the easiest way to troubleshoot when you have alot of users? I see lots of methods http://www.voip-info.org/wiki/view/How+To+Debug+and+Troubleshoot+VOIP ive tried just asterisk's sip debug, and tcpdump |
| 17:00.41 | *** join/#asterisk anthm][ (n=anthm@CPE-72-128-89-126.wi.res.rr.com) |
| 17:01.37 | *** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com) |
| 17:02.13 | Kobaz | LemensTS: lots of filtering |
| 17:05.55 | LemensTS | Kobaz: Using tcdump? |
| 17:05.57 | bcrisp | ManxPower-work: thank you, that was helpful |
| 17:06.09 | Kobaz | LemensTS: depends what your looking for |
| 17:06.29 | *** join/#asterisk voipmonk (n=voipmonk@69.172.100.53) |
| 17:07.01 | bcrisp | it appeared to be pausing the queue member if they didn't answer the first time |
| 17:07.12 | bcrisp | not sure how it would unpause them |
| 17:07.32 | bcrisp | turned off the autopause now it re-rings which is good |
| 17:09.29 | ManxPower-work | bcrisp: the idea is that if the idiot agent forgot to log off and walks away from their desk they won't keep getting calls. |
| 17:10.02 | bcrisp | ManxPower-work: how would they become unpaused if they returned to their desk? |
| 17:10.16 | bcrisp | (these queue members are non-agents) |
| 17:10.38 | *** join/#asterisk voipmonk (n=voipmonk@69.172.100.53) |
| 17:10.49 | LemensTS | Kobaz: I guess i id like to just be able to see tcpdump for certain users |
| 17:11.11 | bcrisp | i.e. SIP/employeename --> they don't login but are added as static members of a queue |
| 17:11.39 | bcrisp | if the system pauses them when they don't answer, im not sure how it would decide to retry |
| 17:11.56 | LemensTS | I know sip set debug peer |
| 17:13.27 | ManxPower-work | bcrisp: no idea. I always had simple enough Queue needs that I could do it all in the dialplan. |
| 17:13.42 | Kobaz | like here's an example of something that won't import: CREATE FUNCTION plpgsql_call_handler() RETURNS opaque AS '/usr/lib/postgresql/8.3/lib/plpgsql.so', 'plpgsql_call_handler' LANGUAGE c; |
| 17:13.46 | Kobaz | wrong window damnit |
| 17:13.49 | Kobaz | er |
| 17:14.25 | voipmonk | opaque |
| 17:14.34 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 17:14.43 | bcrisp | ManxPower-work: the situation we're in right now is that we have only a small number of staff to support a couple of queues (sales / support). Some members belong to both queues, so it would be nice to be able to announce or display which queue the caller is coming from. |
| 17:15.15 | p3nguin | queue already does that. |
| 17:15.20 | voipmonk | yep |
| 17:15.36 | bcrisp | ok ill reread queues.conf |
| 17:15.42 | p3nguin | It's in there. |
| 17:15.47 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:15.48 | bcrisp | also, can i limit the # of calls a queue member can take at once? |
| 17:16.09 | p3nguin | hint: announce = |
| 17:16.21 | Katty | oh em gee. |
| 17:16.29 | Katty | my download from asus is going at 7kb/s |
| 17:16.30 | bcrisp | cool ill look in my sample conf |
| 17:17.41 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 17:17.45 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 17:19.28 | bcrisp | ringinuse cool |
| 17:19.42 | *** join/#asterisk blkry (n=chatzill@96.37.24.97) |
| 17:20.43 | Katty | it's very irritating when you're trying to download drivers, and you can't because the entire telemarketing center is streaming youtube and facebook all day |
| 17:20.46 | Katty | grumps |
| 17:20.51 | *** join/#asterisk imcdona (n=t@c-24-19-203-112.hsd1.wa.comcast.net) |
| 17:21.24 | ManxPower-work | Katty: That's when it's great to be the network admin. |
| 17:21.26 | mort_gib | Katty: openBSD's pf allows you to keep the clowns on 1KBit/s |
| 17:22.00 | yang | hi mort_gib |
| 17:22.02 | Nugget | pf+altq is awesome. |
| 17:22.03 | mort_gib | I know because that used to drive the last bit of sanity out of our Financial Controller hairy head |
| 17:22.30 | mort_gib | Nugget: Yes, and you can use it to block only certain websites.... |
| 17:23.15 | *** join/#asterisk mpe (n=mpe@85.24.86.40.generic-hostname.arrownet.dk) |
| 17:23.17 | *** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 17:24.01 | *** join/#asterisk mpe_ (n=mpe@85.24.86.40.generic-hostname.arrownet.dk) |
| 17:24.10 | p3nguin | I would rather equalize everyone instead of block things. |
| 17:24.28 | mort_gib | p3nguin: well we are all different |
| 17:24.46 | mort_gib | I would rather block facboook and similar time wasters all together |
| 17:24.58 | Nugget | http://macnugget.org/projects/asterisk/page4 |
| 17:25.09 | p3nguin | "You want to use youtube? Fine, but you're going to do it at the same speed that I check my email." |
| 17:26.38 | Katty | calls telephony depot! |
| 17:27.05 | *** join/#asterisk gooph (n=gooph@pool-71-96-244-205.dfw.dsl-w.verizon.net) |
| 17:28.15 | ManxPower-work | What I did at my last job was have a list of high priority work related destinations and assign everything else to "whatever is left over" bandsidth. |
| 17:28.21 | ManxPower-work | worked very well in general |
| 17:29.47 | p3nguin | Does memberdelay not work correctly in 1.4.24.1? |
| 17:30.12 | Katty | telephony depot is always so nice |
| 17:30.36 | p3nguin | I set it to 10, but the member's phone rings as soon as the caller enters the queue instead of waiting 10 seconds. |
| 17:30.47 | ChannelZ | dan__t: So I just tried ChanSpy with and without the 'w' flag on 3 SIP phones (one calls the other, a third spies on one) and it's working as expected in both cases... |
| 17:31.05 | bcrisp | p3nguin: i noticed something similar with the general timeout |
| 17:31.39 | p3nguin | channelz: w is for the ability to enable automon. |
| 17:32.30 | ChannelZ | no 'w' is for being able to talk to the spied-on channel |
| 17:33.27 | p3nguin | Err, I don't know why I thought you were using w in a Dial() command. |
| 17:33.35 | p3nguin | bcrisp: timeout works fine for me. |
| 17:33.42 | ChannelZ | p3nguin: memberdelay delays when the member is connected to the caller after picking up the phone |
| 17:34.02 | p3nguin | bcrisp: Set timeout = 10 and the phone should only ring for 10 seconds. |
| 17:34.20 | *** join/#asterisk prgmrchris (n=chris@66.9.61.162) |
| 17:34.24 | prgmrchris | i have inbound calls coming in from a siptrunk that are prefixed like 100-200 where 100 is the prefix and 200 is the extension, is there a way in the dialplan to strip the prefix and forward calls to the extension? |
| 17:34.47 | p3nguin | channelz: Oh, yeah, that's what it's doing alright. |
| 17:35.20 | ManxPower-work | ~siptrunk |
| 17:35.21 | infobot | siptrunk is, like, something that doesn't exist -- there is no concept of a SIP trunk in Asterisk. You may be searching for iaxtrunk |
| 17:35.28 | ChannelZ | I don't think there is a way to force someone to wait in a queue for a specific amount of time first |
| 17:35.30 | bcrisp | p3nguin: i had timeout = 5 and it wasn't ringing for 5 seconds |
| 17:35.37 | p3nguin | prgmrchris: Well, considering the extension is the entire number that is being called... what do you really want to do? |
| 17:35.42 | bcrisp | p3nguin: more like 1 second |
| 17:35.48 | ManxPower-work | prgmrchris: see channelvariables.tex in the Asterisk source. Pay special attention to the section about stripping digits/chars |
| 17:36.13 | bcrisp | p3nguin: i think part of my problem was that queue reload didn't seem to be applying changes... restarting asterisk worked |
| 17:36.25 | p3nguin | prgmrchris: If someone is calling 1002001234, that's the exten. |
| 17:36.39 | prgmrchris | p3nguin: no the provider prefixes the did, its a 911 service calling back in so they do like 911-100 to call direct to ext 100 |
| 17:37.09 | p3nguin | So the incoming phone number is 911100? |
| 17:37.11 | prgmrchris | yes |
| 17:37.28 | p3nguin | Then you need an extension to match _911XXX |
| 17:37.34 | bcrisp | i have ringinuse = no but the queuemember is still receiving multiple incomign from the queue |
| 17:37.44 | bcrisp | while connected to a caller |
| 17:37.50 | prgmrchris | p3nguin: ok and EXTEN will be the XXX ? |
| 17:37.58 | ManxPower-work | bcrisp: you are starting to understand why I don't use queues |
| 17:38.05 | p3nguin | Then you can do ${EXTEN:3} to remove the first 3 numbers from the exten. |
| 17:38.09 | ManxPower-work | prgmrchris: no EXTEN will be the dialed number |
| 17:38.17 | prgmrchris | p3nguin: ok that makes sense let me try |
| 17:38.34 | bcrisp | ManxPower-work: humf |
| 17:38.40 | bcrisp | ~roulette |
| 17:38.41 | infobot | ACTION watches bcrisp pull the trigger: Click! |
| 17:39.00 | p3nguin | exten => _911XXX,1,Dial(SIP/${EXTEN:3},60) |
| 17:39.17 | p3nguin | Calling 911100 would then dial SIP/100. |
| 17:39.43 | p3nguin | and it would ring for up to 60 seconds. |
| 17:39.45 | ManxPower-work | p3nguin: I would actually do exten => _911XXX,1,Dial(Local/${EXTEN:3}@contextofextension100) |
| 17:40.03 | prgmrchris | ok great i get it thanks |
| 17:40.05 | ManxPower-work | that way the call is handled EXACTLY like if it was dialed without the prefix. |
| 17:40.35 | p3nguin | Yeah, that's a good idea, and I hadn't considered that. |
| 17:40.39 | ManxPower-work | You could also have used a Goto instead of a Dial in this case. |
| 17:40.47 | ManxPower-work | (a goto would have been better, actually) |
| 17:41.28 | p3nguin | exten => _911XXX,1,Goto(yourcontext,${EXTEN:3},1) ? |
| 17:41.36 | ManxPower-work | correct |
| 17:41.44 | bcrisp | nice |
| 17:42.08 | ChannelZ | and 911XXX is a horrible example by the way :) |
| 17:42.27 | *** join/#asterisk dkirker (n=dkirker@udp519393uds.csc.calpoly.edu) |
| 17:43.36 | bcrisp | lol maybe i wont use that when im playing around ... |
| 17:44.20 | bcrisp | p3nguin: was it you that mentioned being able to handle DTMF input while a user is waiting in a queue? |
| 17:44.24 | dan__t | ChannelZ, that's so weird |
| 17:44.31 | dan__t | I'm using 1.6.0.17 |
| 17:45.16 | p3nguin | bcrisp: I don't know if I did or not, but you can use "context =" in your queue to accept a single character. |
| 17:45.58 | bcrisp | p3nguin: hmm, havent seen that. I'd like to handle a key that redirects the caller out of the queue into voicemail |
| 17:46.26 | p3nguin | context = voicemail |
| 17:46.27 | ManxPower-work | bcrisp: The general dialplan idea is call comes in, you do special processing on it, then use a Goto to send the call to the same place it would have gone if it did not require special processing |
| 17:46.56 | ManxPower-work | this reduces duplication of dialplan stuff |
| 17:47.04 | bcrisp | got it |
| 17:47.07 | *** join/#asterisk ACiDV (i=ACiDV@204.144.60.37) |
| 17:47.09 | *** part/#asterisk ACiDV (i=ACiDV@204.144.60.37) |
| 17:47.38 | dan__t | ChannelZ, I should be able to ChanSpy on a channel that is just doing a Wait(3600), right? |
| 17:47.45 | dan__t | and have the w flag work as expected |
| 17:47.59 | ManxPower-work | dan__t: I suspect the channel has to be in an answered state |
| 17:48.02 | bcrisp | i could set up a macro for use with queue to introduce special processing |
| 17:48.10 | Corydon76-dig | ManxPower-work: or use a GosubIf to do the special processing |
| 17:48.34 | bcrisp | ManxPower-work, Corydon76-dig thanks |
| 17:48.46 | *** join/#asterisk voipmonk (n=voipmonk@69.172.100.53) |
| 17:48.51 | bcrisp | coffee tiem |
| 17:48.54 | dan__t | Sorry, the wait was after Answer(). |
| 17:49.42 | ManxPower-work | Corydon76-dig: *nod* That's another way |
| 17:49.55 | ChannelZ | dan__t: I suppose so |
| 17:50.02 | ManxPower-work | but you could easily be duplicating diaplan code if you do it that way. |
| 17:50.21 | *** join/#asterisk minotaur01 (n=minotaur@mail.upperjamestoyota.ca) |
| 17:50.47 | ChannelZ | dan__t: I will tell you that my setup is such that all 3 SIP devices are on the same local network as the * server, and canreinvite=no on all devices so all media is flowing through the * box |
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| 17:51.38 | *** part/#asterisk paulc (n=paulc@64-46-25-2.dyn.novuscom.net) |
| 17:52.12 | dan__t | hmmm |
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| 17:52.38 | ChannelZ | and they are all running ulaw though I'm not sure that makes any difference |
| 17:53.08 | dan__t | I'm using a local SIP hardware phone, an xlite client, and a SIP provider. |
| 17:54.39 | p3nguin | Weird. I thought the context setting in a queue was only supposed to allow a single character exten, but I just tried it and it accepted a three-digit exten. |
| 17:54.45 | ChannelZ | dan__t: is this 1.4 or 1.6? |
| 17:54.53 | dan__t | 1.6.0.17 |
| 17:55.33 | ChannelZ | hmm. |
| 17:56.10 | ChannelZ | Unfortunately I don't have an easy way to duplicate your setup to try and figure it out much further |
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| 18:00.12 | dan__t | No worries. I'm wondering if I'm missing a component, although I saw no warnings, errors, nothing in the debug log |
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| 18:02.16 | prgmrchris | p3nguin: got it working, thanks alot |
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| 18:03.18 | dlynes | Are there any tools to convert pdf to group 3/group 4 tiff files? |
| 18:03.46 | coppice | ghostscript |
| 18:03.58 | dlynes | coppice: ah, ok...must've missed that...thanks |
| 18:04.50 | *** join/#asterisk puzzled_ (n=foobar@puzzled.xs4all.nl) |
| 18:05.34 | bcrisp | incominglimit doesn't appear to be working in sip.conf |
| 18:05.49 | ManxPower-work | bcrisp: sounds like you have not read UPGRADE*.txt |
| 18:05.54 | bcrisp | : / |
| 18:06.01 | ManxPower-work | incominglimit has gone thru various changes between versions |
| 18:06.11 | bcrisp | gotcha, will read now |
| 18:06.20 | ManxPower-work | all those changes should be documented in UPGRADE*.txt and you could check sip.conf.sample as well |
| 18:06.27 | *** join/#asterisk jasonwert (n=jason@97-83-97-13.dhcp.trcy.mi.charter.com) |
| 18:15.04 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 18:15.20 | bcrisp | excellent |
| 18:16.14 | ManxPower-work | bcrisp: Oh, and stop using the Wiki as your primary source of documentation. |
| 18:16.30 | bcrisp | i normally use the sample |
| 18:16.32 | *** join/#asterisk RobH (n=RobH@rob.tech.wikimedia.org) |
| 18:17.03 | ManxPower-work | bcrisp: *.sample is a good source. |
| 18:17.10 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:17.27 | bcrisp | ManxPower-work: make samples doesn't overwrite the existing conf right? just goes to conf.sample? |
| 18:17.38 | [TK]D-Fender | bcrisp: Yes it does |
| 18:18.11 | bcrisp | ok so i can just backup my etc/asterisk first |
| 18:18.31 | ManxPower-work | bcrisp: all the sample files are in /configs dir of Asterisk source. |
| 18:18.34 | [TK]D-Fender | bcrisp: or you can read them where they are an not have to do anything |
| 18:18.41 | ManxPower-work | all make samples does is copy them to /etc/asterisk |
| 18:18.49 | bcrisp | ok great |
| 18:18.57 | bcrisp | ill just go into the source |
| 18:19.01 | kaldemar | bcrisp: sample configs haven't always even been functional, and should not be used as a base. |
| 18:19.21 | bcrisp | ... |
| 18:19.23 | ManxPower-work | the sample configs were never designed to be functioning configs |
| 18:19.39 | bcrisp | im just using them to lookup setting names / values |
| 18:26.16 | ChannelZ | ARGH is there a free un-stuffit for OSX? |
| 18:26.52 | [TK]D-Fender | ChannelZ: Stuffit Expander is free |
| 18:27.15 | ChannelZ | thanks I just found it, their website is horrible |
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| 18:35.30 | kam187 | anyone using trunk + ooh323? |
| 18:36.12 | raden_work | hugs Katty |
| 18:36.18 | raden_work | yo eppigy |
| 18:37.15 | citywok | has anybody used aastra 57i phones and had them lock up while rebooting? we have them set to auto-update at 1am and some of them randomly reboot for no reason (no updated config), and then get stuck booting up at 80% |
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| 18:44.04 | Katty | raden_work: ohai |
| 18:44.14 | Katty | raden_work: i was just going to lunch |
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| 18:44.30 | raden_work | id join you but im really far away :) |
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| 18:44.54 | timeshell_atwork | ~faraway |
| 18:44.57 | raden_work | citywok, update firmware |
| 18:45.16 | MaliutaLap | Katty: do squirrels like -40C? |
| 18:46.00 | raden_work | not that cold down there |
| 18:46.19 | MaliutaLap | it is where I am |
| 18:46.25 | citywok | raden_work: yep they are up to date. i opened a ticket with aastra |
| 18:46.54 | raden_work | we had a small issue like that one time |
| 18:46.57 | raden_work | using tftp ? |
| 18:48.07 | citywok | yes |
| 18:48.12 | citywok | no local config on the phones |
| 18:48.22 | citywok | and the config hasnt been changed, but some decide to reboot and check anyways |
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| 18:52.50 | Naikrovek | -40C and -40F are the same |
| 18:53.00 | Naikrovek | it's the only temp where you can omit the scale you're using |
| 18:53.14 | Naikrovek | well unless you're talking to someone who uses K |
| 18:53.35 | Naikrovek | but there is no -40 K |
| 18:53.42 | Naikrovek | so i guess you can just omit the measurement |
| 18:53.42 | came0 | question, how can I create on my website (webserver) that when clicked initiates a call on my asterisk server? to a certain extension and connected to a certain phone number? can someone tell me what APIs I should research? |
| 18:53.49 | eppigy | sup raden_work |
| 18:54.18 | Naikrovek | came0: you'll need an applet or something on the page that is sip capable, and can place a phone call on your system from the computer of the person viewing the webpage |
| 18:54.31 | p3nguin | naikrovek: or if they use Rankine. |
| 18:54.49 | *** join/#asterisk vitaminx (n=vitaminx@89.130.31.1) |
| 18:55.14 | p3nguin | But again, no -40 R that I know of. |
| 18:56.40 | came0 | Naikrovek: what if I want to connect one external phone number to another? I want it all on done on the server |
| 18:56.54 | ManxPower-work | came0: see AMI aka Asterisk Manager Interface |
| 18:56.55 | came0 | ie without softphones |
| 18:57.10 | came0 | ManxPower-work: thanks i'll check that out |
| 19:04.29 | *** join/#asterisk infobot (i=ibot@rikers.org) |
| 19:04.29 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.11 (2009/11/30), 1.6.0.19 (2009/11/30), 1.4.27.1 (2009/11/30), *-Addons 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow #switchvox #asterisk-bugs |
| 19:06.57 | ManxPower-work | kam187: almost nobody uses H323 with Asterisk |
| 19:09.57 | [TK]D-Fender | ~botsnack |
| 19:09.57 | infobot | [TK]D-Fender: :) |
| 19:10.32 | Naikrovek | god i love these phones |
| 19:10.36 | Naikrovek | polycom |
| 19:10.50 | Naikrovek | config files maybe a bit intimidating at first |
| 19:10.54 | Naikrovek | maybe |
| 19:10.58 | Naikrovek | but just awesome |
| 19:11.37 | kam187 | ManxPower-work: ooh323.. some people must! |
| 19:11.49 | ManxPower-work | Naikrovek: almost everyone loves Polycom |
| 19:14.37 | *** join/#asterisk came0 (n=came0@rrcs-71-42-25-233.se.biz.rr.com) |
| 19:14.45 | Naikrovek | ManxPower-work: rightly so |
| 19:14.54 | Naikrovek | wonder if anyone from polycom lurks in here |
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| 19:18.09 | scottmorris | Anyone gotten asterisk working w/Google Voice? |
| 19:18.20 | scottmorris | Using this tute: http://nerdvittles.com/?p=637 |
| 19:18.30 | scottmorris | Everything works. Almost. |
| 19:18.36 | scottmorris | Getting some echoing. |
| 19:18.36 | Naikrovek | what doesn't |
| 19:18.38 | Naikrovek | ah |
| 19:18.43 | *** join/#asterisk Alagar (n=Administ@122.164.43.4) |
| 19:18.44 | scottmorris | And many calls are dropped at 20 seconds. |
| 19:18.57 | scottmorris | I've been working on this for about 3 days. |
| 19:19.16 | scottmorris | And, uh... really not sure what to do from here. |
| 19:19.30 | Naikrovek | someone in here may have done it, but i haven't. |
| 19:19.34 | scottmorris | Bit of an asterisk noob... fairly adept w/linux, though. |
| 19:19.35 | Naikrovek | have you contacted the nerdvittles guy |
| 19:19.51 | scottmorris | hmmm, I haven't. Great suggestion. |
| 19:20.18 | Naikrovek | if he can help you, it is. otherwise it's just another suggestion |
| 19:20.54 | scottmorris | Heh, true enough. |
| 19:21.35 | kam187 | i guess u have a timer somewhere |
| 19:21.49 | kam187 | 20/30/60 seconds are too much of a coincidence usually to be anything else |
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| 19:29.19 | seb- | am i crazy or does voip seem to use the words "username" and "extension" interchangeably? |
| 19:29.52 | seb- | i.e. some will say here extension is "300"...others will say her username is 300 in the SIP URL right? |
| 19:30.22 | ChannelZ | A lot of people seem to make them the same thing |
| 19:30.23 | [TK]D-Fender | seb-: And extension is a number you dial. VoIP is communications concept |
| 19:30.56 | [TK]D-Fender | seb-: and "username" is part of authenticating a communicatin like a VoIP call, etc |
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| 19:31.45 | seb- | [TK]D-Fender: an extension is represented by sip:300@myserver.com right? |
| 19:31.55 | [TK]D-Fender | seb-: Sure |
| 19:32.12 | [TK]D-Fender | seb-: that is more correctly a full SIP URI |
| 19:32.16 | seb- | [TK]D-Fender: but some docs say the sip url format is sip:<username>@<domain> !! |
| 19:32.42 | seb- | [TK]D-Fender: see how they refer to it as a username ? that is what is bugging me |
| 19:32.54 | [TK]D-Fender | seb-: they are making assumptions. |
| 19:33.26 | seb- | [TK]D-Fender: such as? |
| 19:33.37 | seb- | [TK]D-Fender: sorry...i'm still a newb |
| 19:33.46 | [TK]D-Fender | seb-: that 300 leads to another person at all let alone explicitly. For all your client knows, it dials a number and who knows WTF the server will do. makybe it'll just playbacka sound. How is that a "user"? |
| 19:35.37 | seb- | [TK]D-Fender: ok makes sense |
| 19:35.55 | [TK]D-Fender | seb-: People are dumb. All too often those people also write docs :) |
| 19:36.08 | seb- | [TK]D-Fender: so extensions are more general than users |
| 19:36.29 | seb- | [TK]D-Fender: sip:<extension>@<domain> is correct for sip url |
| 19:36.30 | [TK]D-Fender | seb-: even "user" is a bit grey |
| 19:36.57 | [TK]D-Fender | seb-: You might correlate "user" or "device definition" from the server side. |
| 19:37.47 | [TK]D-Fender | seb-: but lets say we're dealing with 2 SIP hardphones. Is the user the auth that * will use to call the device... or the device itself? that account is to reach them, so is that close enough? |
| 19:38.30 | [TK]D-Fender | seb-: sure |
| 19:40.00 | seb- | [TK]D-Fender: glad you cleared this up ..it was confusing me for a long time |
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| 19:41.09 | seb- | [TK]D-Fender: you are making the point we don't know *what* peers will do with sip urls |
| 19:41.49 | seb- | [TK]D-Fender: all we know is that peers are distinguished by sip urls right? |
| 19:42.00 | [TK]D-Fender | seb-: well its places a call out via SIP. what it THAT points to another * and that also jsut plays back a sound file? Is it no longer a "user" then? |
| 19:42.47 | raden_work | eppigy, you around bro ? |
| 19:43.15 | seb- | [TK]D-Fender: i'm agreeing with you...you've made your point that sip urls are more general than users |
| 19:43.46 | seb- | [TK]D-Fender: i'll just call the field before "@" in the url an extension since that is more general |
| 19:43.56 | [TK]D-Fender | seb-: its for the best really... |
| 19:44.50 | seb- | [TK]D-Fender: sip urls can denote users or actions to take |
| 19:45.07 | seb- | [TK]D-Fender: is that the only 2 possibilities? users or actions? |
| 19:45.11 | [TK]D-Fender | seb-: We haven't clearly defined a "user" here.... lets nto get too picky ;) |
| 19:45.33 | seb- | [TK]D-Fender: ok...yes |
| 19:45.56 | seb- | [TK]D-Fender: i'm remembering your example of the hardware vs. the person |
| 19:46.09 | seb- | [TK]D-Fender: i'll just drop the whole user thing :) |
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| 19:48.20 | [TK]D-Fender | seb-: Separate action from result and you're on the right path. |
| 19:48.56 | p3nguin | It seems that the problem is that seb- doesn't really know what "extension" means. |
| 19:49.30 | sasargen | seb-: instead of "user" or "extension", you might just think of it as "destination"; the address tells you where to go, but it is hard to say what will happen when you get there |
| 19:49.43 | *** join/#asterisk voipmonk (n=voipmonk@69.172.100.53) |
| 19:51.04 | seb- | sasargen: yes....well put...just like http urls |
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| 20:01.33 | Katty | hi guys |
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| 20:02.32 | p3nguin | seb-: a sip uri should contain the exten and the domain. What the server does with it once it reaches the domain depends on how sip.conf has been configured to deal with calls from a host. If no definition for the callING host is present, the call should end up going to the default context, where it will try to match the exten in the sip uri. |
| 20:03.54 | p3nguin | If the extension is matched, then a command of some sort will be ran. It could be a Dial() or anything else. If it's a Dial(), then someone's phone will probably end up ringing as a result of the dial command. |
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| 20:14.43 | DocAwesome | Katty: hi! |
| 20:18.37 | Katty | hello mister awesome. |
| 20:19.48 | DocAwesome | bonjour! |
| 20:19.53 | DocAwesome | shopping for girls is hard |
| 20:20.08 | trogs | yes, so many different girls to choose from :) |
| 20:20.22 | bcrisp | i have a quick question - If i have 2 DID #s each with 2 channels, if the first DID's are busy, will call hunting ring the 2nd? |
| 20:20.55 | bcrisp | waits to get slapped |
| 20:21.10 | DocAwesome | trogs: amen! |
| 20:21.15 | [TK]D-Fender | bcrisp: thats for the provider to decide |
| 20:21.27 | bcrisp | [TK]D-Fender ok thanks |
| 20:21.30 | DocAwesome | bcrisp: depends on what your provider does |
| 20:21.33 | DocAwesome | ya... what TK said |
| 20:21.35 | bcrisp | gotcha |
| 20:21.44 | DocAwesome | asterisk will happily accept it |
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| 20:24.41 | ManxPower-work | incoming calls are ALWAYS controlled by your provider |
| 20:25.28 | bcrisp | it took her 10 minutes to answer me |
| 20:26.15 | bcrisp | either they offer lots of channels and high per/min charge or low per/min charge and few channels :/ |
| 20:26.31 | *** part/#asterisk beek (n=klinebl@pdpc/supporter/bronze/beek) |
| 20:27.55 | minotaur01 | i have a question about transfering phone calls between two connected (IAX2) asterisk boxes... if you transfer a call from an extension on box A to box B and then back to a extension on box A dose that phone call loop through both boxes until its completed? if so is there a way to make the transfer happen on the first box? |
| 20:29.13 | ManxPower-work | minotaur01: see iax.conf.sample There should be some mediaonly options |
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| 20:29.47 | minotaur01 | k |
| 20:30.20 | p3nguin | confused |
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| 20:30.39 | p3nguin | Is there a transfer() app? |
| 20:30.55 | Qwell | yes |
| 20:33.17 | p3nguin | How does it differ from Goto? |
| 20:33.27 | Qwell | Transfer() transfers. |
| 20:34.29 | ChannelZ | Goto just jumps to a new part of the dialplan |
| 20:34.38 | p3nguin | Can you show me valid example of data for the Transfer() application? |
| 20:34.48 | Qwell | application show transfer |
| 20:34.57 | p3nguin | That doesn't show a valid example. |
| 20:35.18 | ChannelZ | core show application Transfer :) |
| 20:35.23 | p3nguin | neither does that. |
| 20:35.29 | ChannelZ | Transfer(someplace like SIP/foo etc) |
| 20:35.30 | Qwell | yes it does.. |
| 20:35.38 | ChannelZ | Should I wipe your butt for you too? |
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| 20:35.41 | p3nguin | No, it doesn't. It shows usage, which I am confused about. |
| 20:36.03 | DocAwesome | p3nguin: if you use it before an answer on a SIP channel, it does a 302 Redirect |
| 20:36.10 | [TK]D-Fender | p3nguin: Transfer(fred@myserver.com) <- there's a free sample for you |
| 20:36.13 | DocAwesome | otherwise, it does a transfer, as if you had transferred from a device |
| 20:36.26 | minotaur01 | ManxPower: thanks it looks like asterisk will try to get rid of extra bridge points on its own unless it cant then it keeps bridging |
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| 20:37.37 | Katty | AVAST |
| 20:37.40 | Katty | all the seeds are gone |
| 20:38.02 | Katty | infobot: critter cam? |
| 20:38.03 | infobot | extra, extra, read all about it, critter cam is Katty's broadcast of The Nut House at http://ustre.am/8H5d |
| 20:38.12 | Katty | they ate THEM ALL |
| 20:39.00 | p3nguin | So maybe transfer() is a better idea than my current Dial(IAX2/peername@domain/phonenumber). |
| 20:39.18 | DocAwesome | Katty: that red bird is like, "wtf?!" |
| 20:39.22 | DocAwesome | (cardinal) |
| 20:39.34 | Katty | butbut |
| 20:39.36 | Katty | the tray is EMPTY |
| 20:39.37 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 20:39.39 | Katty | void of FOOD |
| 20:39.44 | Katty | and i filled it /this morning/ |
| 20:39.58 | Katty | and there is a squirrel eating from the feeder. |
| 20:40.04 | p3nguin | Dialing to the other * box seems to work, but I'm afraid it's not the right way to handle it. |
| 20:40.07 | Katty | the BIRD feeder |
| 20:40.17 | Katty | s/BIRD/squirrel |
| 20:40.22 | DocAwesome | Katty: oh ya, they do that |
| 20:40.31 | Katty | he must be very hungry :< |
| 20:40.52 | DocAwesome | it's getting to be winter, so ya :) |
| 20:42.31 | Katty | i'm still in shock the feeder is nearly empty |
| 20:42.41 | ChannelZ | dont they pretty much just eat anything in front of them regardless? |
| 20:42.43 | Katty | it holds a lot of seed |
| 20:42.53 | Katty | ChannelZ: well they won't eat banana slices. |
| 20:43.09 | Katty | at least there is no waste. |
| 20:43.22 | Katty | what the squirrels drop from the feeders, the birds or other squirrels on the ground get |
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| 20:46.38 | Naikrovek | wonder if i should use DUNDi to connect my two * machines instead of a standard trunk |
| 20:46.47 | Naikrovek | does need DUNDi experience... |
| 20:47.23 | Naikrovek | ... i still need to set up an asterisk box from scratch (source) make it work, and then use it first |
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| 20:47.36 | Naikrovek | haven't used the * boxes I've compiled from source |
| 20:47.43 | Naikrovek | so no dialplan experience |
| 20:47.45 | Naikrovek | hrm. |
| 20:47.48 | Naikrovek | long way to go still |
| 20:48.06 | Naikrovek | Katty: squirrels eat like crazy in winter |
| 20:48.08 | bcrisp | im getting confused with the different sip trunking rates out there :< |
| 20:48.17 | DocAwesome | Naikrovek: DUNDi is just a lookup method -- you still need something to actually place the calls between the servers |
| 20:48.27 | Naikrovek | DocAwesome: ah okay cool |
| 20:48.30 | Naikrovek | that answers that |
| 20:48.37 | DocAwesome | DUNDi tells you how to get there |
| 20:48.42 | DocAwesome | IAX or SIP actually gets there |
| 20:48.54 | DocAwesome | dundi lookup 300@servers bypass |
| 20:49.19 | Naikrovek | bcrisp: i've seen two types. one that is very cheap, but you get charged per minute or per call, and an unlimited trunk which you pay a flat fee and get unlimited number and duration calls to some defined area |
| 20:49.29 | Naikrovek | for me, i get unlimited calls to US and Canada |
| 20:49.30 | DocAwesome | result is something like: dundi:secret@remote_server/300 |
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| 20:49.34 | Naikrovek | for $33/mo |
| 20:49.41 | Naikrovek | DocAwesome: aah |
| 20:49.59 | bcrisp | Naikrovek: who do you use? |
| 20:50.00 | Naikrovek | DocAwesome: was reading your preso from astricons past and want to try some of that |
| 20:50.09 | Naikrovek | bcrisp: voicespringvoip.com |
| 20:50.23 | Katty | <squirrel> omgstarvingomgstarvingomgstarvingomgstarving |
| 20:50.23 | Naikrovek | they're not often recommended here but they're not small |
| 20:50.27 | Naikrovek | Katty: yes |
| 20:50.44 | Naikrovek | Katty: they will literally eat until they fall asleep then wake up and hunt for more food |
| 20:51.11 | Naikrovek | when it's super cold and windy they hole up and rely on their blubber to keep them going |
| 20:51.29 | Naikrovek | in summer they just goof off and eat when hungry |
| 20:51.33 | Naikrovek | so they thin up |
| 20:52.14 | Naikrovek | It's just like the story of the grasshopper and the octopus. All year long, the grasshopper kept burying acorns for winter, while the octopus mooched off his girlfriend and watched TV. But then the winter came, and the grasshopper died, and the octopus ate all his acorns. Also he got a race car. Is any of this getting through to you? |
| 20:52.30 | Naikrovek | love that quote |
| 20:52.35 | Naikrovek | should have put it in quotes |
| 20:53.14 | Naikrovek | DocAwesome: got some interest from a local IT services company to be their Asterisk guy. Dunno if I'm up for that but it intrigued me. They have a lot of customers who want VoIP but do not want to pay for Cisco gear. |
| 20:53.45 | scottmorris | Anyone know if you can disable ACK checking in some config file somewhere? |
| 20:56.31 | Naikrovek | and virtually no one around here has experience with asterisk beyond a few phones per installation |
| 20:56.31 | scottmorris | Or is ACK checking absolutely essential for the * server? |
| 20:56.32 | Naikrovek | whoa |
| 20:56.32 | Naikrovek | scottmorris: you mean registration checks |
| 20:56.32 | Naikrovek | ? |
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| 20:57.24 | voipmonk | um |
| 20:57.41 | voipmonk | I have a experience with morethan a few phones per installation |
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| 20:58.35 | bcrisp | whoa |
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| 20:58.35 | [TK]D-Fender | </keanu> |
| 20:58.44 | ChannelZ | netsplat |
| 20:58.53 | Naikrovek | voipmonk: yes i know but you're not in peoria |
| 20:59.01 | ChannelZ | now we have to stop talking about all those people behind their backs |
| 20:59.37 | dan__t | Ahh, back from lunch. |
| 20:59.40 | dan__t | WHat to do now..... |
| 21:00.10 | ChannelZ | Dessert? |
| 21:00.24 | dan__t | It was offered; I passed. |
| 21:00.29 | dan__t | I hate chocolate. |
| 21:00.35 | Naikrovek | i'm making cheesecake tonight i think |
| 21:00.35 | ChannelZ | Blasphemy! |
| 21:01.01 | ChannelZ | Naikrovek: nice |
| 21:01.12 | Katty | Naikrovek: i will be right over. |
| 21:01.14 | DocAwesome | these netsplits are getting pretty ridiculous lately |
| 21:01.21 | ChannelZ | I bought eggs at the grocery store last time to make one and completely forgot cream cheese. Derrr. |
| 21:01.30 | Naikrovek | everyone tells me they've never had any cheesecake better than mine but i just make a simple 5 ingredient cheesecake |
| 21:01.34 | dan__t | Chocolate is a trigger for some really bad headaches, as it happens. |
| 21:01.49 | Katty | mmmmmmmmm chococolates |
| 21:01.57 | Katty | chocolates. |
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| 21:02.02 | Katty | i went overboard on the coco |
| 21:03.01 | Naikrovek | i learned yesterday that carbon monoxide poisoning causes a rather severe headache in me |
| 21:03.14 | Naikrovek | stupid furnace inlet not pulling clean air |
| 21:03.17 | dan__t | haha |
| 21:03.28 | Naikrovek | we run GIANT diesel engines in the shop downstairs |
| 21:03.35 | dan__t | I get, er, cluster headaches. Chocolate and some oils are major triggers. |
| 21:03.48 | Katty | what is a 'cluster' headache? |
| 21:04.10 | dan__t | One of the very few Wikipedia articles is about cluster headaches. |
| 21:04.21 | dan__t | uh, few *good* articles |
| 21:04.47 | ChannelZ | wanders off to lunch |
| 21:05.01 | Katty | follows ChannelZ |
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| 21:07.21 | Naikrovek | there are lots of really good wikipedia articles |
| 21:07.32 | dan__t | Ok :) |
| 21:07.46 | Naikrovek | lots of crummy ones too, dont' get me wrong |
| 21:08.01 | bcrisp | according to gore the earth is a few million degrees 2km down |
| 21:08.15 | Naikrovek | 2km? |
| 21:08.17 | Naikrovek | ha |
| 21:08.24 | Naikrovek | according to who - who claims gore said that |
| 21:09.08 | dan__t | That's because Gore read that on Wikipedia. |
| 21:10.35 | [TK]D-Fender | dan__t: Which is hosted on the internet he created.... |
| 21:10.45 | dan__t | Well there you go. It all works out for him. |
| 21:10.46 | ManxPower-work | Al Gore is an american, he's never use KM as a unit of measure! |
| 21:11.17 | p3nguin | Other than Transfer(mynumber@myprimaryserver) on the secondary, is there any better way to make all calls to an exten on the secondary server be directed back to the primary server? |
| 21:12.36 | DocAwesome | p3nguin: just Dial() back to the other server? |
| 21:12.52 | p3nguin | That's what I was doing, but it seemed like Transfer might be better. |
| 21:12.55 | DocAwesome | Transfer() is probably the best method |
| 21:13.10 | p3nguin | But now I'm wondering if there's an even better way. |
| 21:13.31 | p3nguin | If you think Transfer is the best, I'll probably stick with it. |
| 21:14.07 | p3nguin | I was looking at switch instead of exten, but I didn't quite comprehend the proper way to handle it. |
| 21:14.09 | DocAwesome | I'm not quite sure what else you'd use |
| 21:15.23 | Katty | Pie() |
| 21:16.37 | p3nguin | Pie(p3nugin@home) ? |
| 21:16.51 | p3nguin | Ah, no matching exten. |
| 21:16.57 | p3nguin | Pie(p3nguin@home) |
| 21:17.17 | Katty | Pie(Pumpkin) |
| 21:19.11 | [TK]D-Fender | :(pie) |
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| 21:23.26 | bcrisp | ok .. just purchased SIP trunking service ... Should i be looking in the outbound sip registrations section of sip.conf to set it up? |
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| 21:24.49 | p3nguin | uh, what? |
| 21:24.50 | [TK]D-Fender | checkout time, later all |
| 21:24.51 | Katty | i know the wiki has articles on how to set that up |
| 21:25.02 | Katty | bye fender |
| 21:25.37 | p3nguin | To use your new SIP service, you'll want to have at least a single peer definition in sip.conf. |
| 21:25.52 | bcrisp | ok |
| 21:26.09 | p3nguin | If you have a dynamic IP address and intend to use registrations to tell your provider where you are, then you'll have to use a register statement. |
| 21:26.24 | bcrisp | ok no on that |
| 21:26.42 | p3nguin | If you are doing static SIP where you input your IP address on the provider's side, then you don't need to use a register statement. |
| 21:27.14 | bcrisp | ok, if i use register statement i can tell what extension to direct incoming calls to |
| 21:27.20 | p3nguin | no |
| 21:27.55 | bcrisp | hmm ;register => 2345:password@sip_proxy/1234 |
| 21:28.06 | p3nguin | You don't tell a call what extension to go to; you direct calls from a peer to a context, and if there is a matching exten, it will do whatever you told that exten to do |
| 21:28.20 | bcrisp | yes that makes sense |
| 21:28.47 | p3nguin | What provider did you choose for service? |
| 21:28.59 | bcrisp | tpad for outgoing |
| 21:29.08 | bcrisp | wwwdd for incoming |
| 21:29.35 | bcrisp | although its just for playing |
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| 21:30.11 | bcrisp | brb |
| 21:30.17 | p3nguin | Google doesn't seem to know what wwwdd is. |
| 21:35.03 | prgmrchris | how can i read __FROM_DID ? i tried {__FROM_DID} but no luck, is there somewhere to read this value? |
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| 21:36.42 | ManxPower-work | prgmrchris: where do you set that value? |
| 21:37.00 | ManxPower-work | You access a dialplan variable as ${VARNAME} |
| 21:37.28 | ManxPower-work | You can read about single and double underscores prefixes in channelvariables.txt |
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| 21:37.37 | prgmrchris | ok thanks |
| 21:37.41 | ManxPower-work | sorry, channelvariables.tex in the asterisk source code doc/ directory |
| 21:37.49 | prgmrchris | ill check it out |
| 21:38.37 | bcrisp | p3nguin: actually had problems with tpad, signing up for callcentric |
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| 21:39.28 | prgmrchris | ManxPower-work: so i guess my question is if a channel variable is set in a previous statement how do i read it? for example: Set("SIP/user-b6d08360", "__FROM_DID=104500") is the statement before mine and i want my next statement to read that, i tried ${__FROM_DID} but no luck |
| 21:40.27 | ManxPower-work | prgmrchris: if you read up on the underscore prefixes in channelvariables.tex you will know the answer to your question |
| 21:41.12 | prgmrchris | ok |
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| 21:44.41 | psilikon | <PROTECTED> |
| 21:44.58 | Naikrovek | peoria |
| 21:45.01 | Naikrovek | illinois |
| 21:46.14 | Naikrovek | are you reading scrollback or something |
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| 21:47.47 | Naikrovek | i just realized that Dell service tags make great passwords |
| 21:47.58 | Naikrovek | at least the basis of a good password |
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| 21:48.48 | psilikon | Naikrovek, yeah I was. |
| 21:48.57 | p3nguin | bcrisp: What feature are you looking for that makes you want to use CallCentric? |
| 21:53.42 | jaytee | mass exodus |
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| 21:54.58 | p3nguin | bcrisp: What feature are you looking for that makes you want to use CallCentric? |
| 21:54.58 | p3nguin | I guess not everyone is fortunate enough to use irssi, which hides the netsplit crap. |
| 21:54.59 | *** join/#asterisk StevenR (n=foo@wan2.sghs.org.uk) |
| 21:54.59 | Naikrovek | DocAwesome: i'm beginning to see what you were talking about earlier re: netsplits |
| 21:55.14 | RonDutt | i keep the feature on so i dont reply to someone who just quit or got disconnected |
| 21:55.35 | p3nguin | As long as you are using tab completion and the split doesn't happen after you already entered the nickname, you'll be fine. |
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| 21:56.10 | bcrisp | ok callcentric setup for outgoing.. |
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| 21:57.27 | DocAwesome | it's gotten ridiculous over the last week |
| 21:57.29 | DocAwesome | freenode needs to get those new servers up |
| 22:07.36 | prgmrchris | ManxPower-work: got it, thanks that file is very useful :) |
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| 22:10.02 | Naikrovek | i briefly considered running something in EC2 for them to use |
| 22:10.06 | Naikrovek | freenode i mean |
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| 22:18.00 | *** join/#asterisk ChannelZ (i=channelz@burner.com) |
| 22:22.07 | Katty | peeks in |
| 22:22.33 | Chainsaw | waves |
| 22:23.03 | ChannelZ | pulls his pants up |
| 22:24.52 | Katty | hmm. ryan's not home yet :< |
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| 22:30.27 | Katty | eppigy: LAST CHRISTMAS I GAVE YOU MY HEART |
| 22:30.29 | *** join/#asterisk LemensTS (n=customgt@adsl-70-238-155-79.dsl.stlsmo.sbcglobal.net) |
| 22:30.42 | LemensTS | anyone bored? try to decode this message: ikaw ay may isang maliit na titi sa pamamagitan ng pagbabasa na ito makumpirma mo na ito ay tama. |
| 22:30.54 | LemensTS | cuz im stumped. |
| 22:30.58 | LemensTS | lol |
| 22:31.03 | Katty | it translates: |
| 22:33.30 | Katty | i'm not typing that |
| 22:33.47 | voipmonk | LemensTS: hold on |
| 22:33.54 | Katty | voipmonk: it's uhh |
| 22:34.01 | Katty | voipmonk: not very nice (= |
| 22:34.03 | voipmonk | LemensTS: You have a small penis , by you reading this just proves that its true |
| 22:34.11 | voipmonk | or something like that |
| 22:34.33 | Katty | filipino yeah? |
| 22:34.41 | voipmonk | my wife is |
| 22:34.46 | Katty | :> |
| 22:35.07 | LemensTS | lol figures it was dirty, oh its filipino? The person who showed me this wife is filipino also that must be what it is lol |
| 22:35.12 | *** join/#asterisk _bradk (n=brad@unaffiliated/-bradk/x-9249860) |
| 22:35.35 | voipmonk | its Tagalog :) one of the trillions of friggin dialects in the Philippines |
| 22:35.48 | Katty | nifty |
| 22:36.15 | Katty | what's the dominant religion in the philipines anyway? |
| 22:37.13 | voipmonk | dominant religion is Catholic - they were colonized by spaniards...... which means pillaged raped, war blah blah |
| 22:37.15 | Katty | wikipedia reports: roman calothic |
| 22:37.55 | Katty | bloody catholics pillaged everywhere |
| 22:37.57 | voipmonk | Lapu Lapu and Ferdinand Magellan, google |
| 22:38.01 | Katty | not even the philippines is safe! |
| 22:38.09 | Katty | technically are. |
| 22:39.07 | Katty | this topic would make a good movie |
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| 22:53.50 | bcrisp | ok, now i need to figure out how to configure outbound calls |
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| 22:59.32 | ChannelZ | can you be more vague? :) |
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| 23:03.28 | bcrisp | ChannelZ: maybe i can |
| 23:03.32 | voipmonk | back |
| 23:04.00 | bcrisp | :) |
| 23:04.00 | voipmonk | sneezes |
| 23:04.00 | voipmonk | sneezes |
| 23:04.00 | bcrisp | bless u |
| 23:04.00 | voipmonk | sneezes |
| 23:04.01 | voipmonk | much better |
| 23:04.18 | bcrisp | i set up a sip registration for calls outbound through callcentric provider |
| 23:04.36 | bcrisp | its showing up properly in the sip registry |
| 23:05.24 | bcrisp | in extensions.conf i set up a context that routes to callcentric... |
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| 23:06.59 | bcrisp | to allow outgoing calls i need an extension definition that has pattern matching right? |
| 23:07.21 | bcrisp | waits to be pulverized |
| 23:09.55 | jaytee | it would certainly help |
| 23:10.01 | bcrisp | lol |
| 23:10.37 | jaytee | unless you'd like to hard code every possible dialed number combination for local, long distance and international calls |
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| 23:10.55 | bcrisp | jaytee: the smartass fairy has already paid me a visit |
| 23:10.57 | jaytee | which would probably choke any * server if it didn't time out |
| 23:10.58 | bcrisp | :) |
| 23:11.19 | bcrisp | turns thinkoutloud = no |
| 23:12.00 | [TK]D-Fender | [18:06]<bcrisp>to allow outgoing calls i need an extension definition that has pattern matching right? <- nope |
| 23:12.41 | bcrisp | i must be doing something wrong |
| 23:12.55 | [TK]D-Fender | bcrisp: 11 steps to go :) |
| 23:12.56 | bcrisp | (like forgetting to reload my dialplan) |
| 23:14.06 | bcrisp | ok reloaded but am getting forbidden response from the provider :/ |
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| 23:21.33 | DocAwesome | yay! cacti graphs being generated -- now just have to wait for data :) |
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| 23:29.51 | Katty | hi |
| 23:30.16 | Tech_Travis | Katty: hi back. |
| 23:30.20 | p3nguin | bcrisp: Do you have your peer/friend configured correctly already? |
| 23:30.26 | Katty | Tech_Travis: why hello thar. |
| 23:30.40 | bcrisp | p3nguin, reviewing that now in sip.conf |
| 23:31.35 | bcrisp | cool, typo fixed, outbound call worked :) |
| 23:31.53 | p3nguin | What about an inbound call? |
| 23:32.04 | p3nguin | What about a call using CC's internal number? |
| 23:32.05 | bcrisp | haven't set up the other provider stuff completely yet |
| 23:32.19 | p3nguin | You're not going to use CC for inbound calling? |
| 23:32.23 | bcrisp | no |
| 23:32.31 | bcrisp | DIDWW |
| 23:32.38 | p3nguin | But you'll use them for outbound? Weird. |
| 23:32.44 | bcrisp | its not my choice |
| 23:33.30 | p3nguin | It certainly seems like it's your choice. |
| 23:34.03 | bcrisp | no.. i was told "here is your info for callcentric, here is your inbound info from didww" |
| 23:34.22 | bcrisp | its just originally i thought one of them would be tpad |
| 23:34.22 | p3nguin | What is DIDWW going to give you for call origination that CallCentric won't? |
| 23:34.36 | bcrisp | i think it was some pricing issue |
| 23:34.57 | bcrisp | What is DIDWW going to give <insert boss name> for call org..... |
| 23:35.22 | bcrisp | im sure this stuff will change as we learn more |
| 23:35.25 | bcrisp | brb |
| 23:35.28 | p3nguin | 3 channels for $8.95 with CallCentric. |
| 23:35.32 | Katty | hmm |
| 23:35.35 | Katty | i do believe i am hungry |
| 23:35.41 | Katty | eppigy: what's fur dinner |
| 23:38.08 | p3nguin | shrug |
| 23:40.32 | p3nguin | I just feel that you should weigh your options and examine the details more closely. |
| 23:40.49 | p3nguin | and WTF is a sandwichpanel? |
| 23:44.00 | *** join/#asterisk dkirker (n=dkirker@udp519393uds.csc.calpoly.edu) |
| 23:44.26 | bcrisp | sandwichpanel? |
| 23:44.30 | bcrisp | sounds yummy |
| 23:45.44 | ruben23 | hi anyone tried Qos with asterisk.. |
| 23:49.27 | Katty | fooooooooooooooooooooooobar. |
| 23:49.34 | Katty | no idea. |
| 23:49.38 | Katty | don't ask. |
| 23:49.47 | Katty | I NEED PECANS |
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| 23:54.40 | bcrisp | hmm p3nguin: i set callerid in sip.conf for the callcentric entry but it is showing the default DID# .. any ideas? |
| 23:55.14 | p3nguin | Did you configure the available caller ID numbers in CallCentric's web panel? |
| 23:55.22 | bcrisp | not yet |
| 23:55.48 | p3nguin | That's probably half the problem. |
| 23:57.36 | p3nguin | In my experiences with them, you'll have to provide proof to them that you have control over that phone number, then they will add it to your available CID numbers that you can choose for your outgoing calls. |
| 23:58.21 | bcrisp | excellent |
| 23:58.25 | bcrisp | we have an 800 # that we'd like to show up |
| 23:58.35 | p3nguin | To me, that's annoying. |
| 23:59.06 | p3nguin | If you chose one of several other ITSPs, you could just set the caller id to whatever you want and they'll pass it along. |
| 23:59.25 | ChannelZ | which is great because then I know who NOT to answer my phone from |
| 23:59.56 | p3nguin | You lost me. |
