| 00:00.48 | hmmhesays | ho vicidial is a b1tch to install |
| 00:01.12 | DoDaT69 | is it? |
| 00:01.16 | DoDaT69 | whats so difficult about it? |
| 00:01.25 | hmmhesays | just a lot of steps and it takes forever |
| 00:01.29 | DoDaT69 | really? |
| 00:01.41 | DoDaT69 | I am checking it out now |
| 00:01.45 | DoDaT69 | looks exactly like what we need |
| 00:01.55 | hmmhesays | http://astguiclient.sourceforge.net/scratch_install.html |
| 00:02.03 | RypPn | ~book |
| 00:02.04 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 00:02.19 | DoDaT69 | wow biog page |
| 00:02.33 | hmmhesays | a couple years ago I got about half way through an auto install script |
| 00:02.51 | hmmhesays | then I left that particular employer and didn't bring it with me |
| 00:02.55 | DoDaT69 | doh! |
| 00:03.13 | hmmhesays | I could probably recreate it pretty easily though |
| 00:03.51 | DoDaT69 | we are thinking that we will go with a freepbx for easy administration and the call reporting, we just need something that will provide the receptionist with a script |
| 00:03.57 | DoDaT69 | its for an executive suite center |
| 00:11.31 | mltlnx | is mohsuggest new to Asterisk 1.4? |
| 00:13.27 | *** join/#asterisk yassine (n=yassine@unaffiliated/yassine) |
| 00:13.34 | yassine | good evening everyone |
| 00:14.35 | yassine | i have a problem with my asterisk even when a peer is registered its not able to call or even to try *43 |
| 00:15.22 | DoDaT69 | nat or firewall? |
| 00:16.06 | yassine | DoDaT69: i'am indeed in my local/home network but im trying from the same network since im at home |
| 00:16.32 | DoDaT69 | so they are on the same network? |
| 00:16.42 | DoDaT69 | do you ahve iptables on your asterisk machine? |
| 00:16.54 | yassine | yes but asterisk is configured with the externhost |
| 00:17.32 | DoDaT69 | you can sip show peers and the extension will show up? |
| 00:18.20 | yassine | DoDaT69: let me pastebin what comes out please |
| 00:18.25 | DoDaT69 | k |
| 00:21.52 | yassine | DoDaT69: http://rafb.net/p/68UwRQ29.html |
| 00:22.25 | DoDaT69 | looks like you are registering from 2 different ip's |
| 00:22.32 | DoDaT69 | nm |
| 00:22.55 | yassine | DoDaT69: i did that on purpose |
| 00:23.04 | DoDaT69 | yea, I see |
| 00:23.15 | yassine | i have registred two time to show you the differnce between both cases |
| 00:23.23 | DoDaT69 | right |
| 00:23.26 | DoDaT69 | it works neither way? |
| 00:23.27 | *** join/#asterisk Maxxed (i=foobar@65.59.245.122) |
| 00:23.42 | yassine | none works |
| 00:23.52 | yassine | i can register but can not dial not even the *43 |
| 00:23.53 | DoDaT69 | do you have iptables running on the local machine? |
| 00:24.06 | DoDaT69 | can you receive a call? |
| 00:24.25 | DoDaT69 | what error do you get when you dial? it could just be your phone's dialplan |
| 00:24.43 | Maxxed | any of you guys know if there is anything out there that will let one tie asterisk into exchange? when a voicemail is recived, you get it in your inbox, but when it is delted, it gets deleted on the asterisk box? |
| 00:24.51 | Maxxed | if that makes any sense |
| 00:24.58 | DoDaT69 | (Maxxed): look at doing openser |
| 00:25.03 | Maxxed | delete it once and not have to login to vm and delete |
| 00:25.09 | Maxxed | openser? |
| 00:25.18 | kand | Maxxed: IMAP storage will do that |
| 00:25.18 | DoDaT69 | (Maxxed): open sip express router |
| 00:25.23 | DoDaT69 | (Maxxed): yea |
| 00:25.25 | yassine | DoDaT69: as soon i dial i see in the cli : [Dec 14 01:25:43] NOTICE[10066] chan_sip.c: Peer '007' is now UNREACHABLE! Last qualify: 10 |
| 00:25.55 | DoDaT69 | (yassine): it looks like you might have some messed up networking |
| 00:25.57 | Maxxed | hum, i dont get it, but il have to check it out |
| 00:26.06 | Maxxed | kand thats what im talkin about |
| 00:26.14 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 00:26.22 | Maxxed | imap storage, mm, i like the sound of that |
| 00:26.50 | kand | Maxxed: that is all I know of. If you go the IMAP route make sure you have the latest greatest, it has been a work in progress... |
| 00:26.57 | Maxxed | hehe when i search for it in google, i get a mess of security vuln notices |
| 00:26.58 | WilliamK | does anyone know if SpanDSP works with asterisk 1.4.15+? |
| 00:27.23 | Maxxed | thanks for the tip, this looks like what i was looking for |
| 00:27.27 | kand | np |
| 00:27.36 | r0d3nt | <SecNews> Title: Zaptel 1.2.22.1 and 1.4.7.1 released |
| 00:27.36 | r0d3nt | <SecNews> Link: http://www.asterisk.org/node/48437 |
| 00:27.38 | r0d3nt | weee..... |
| 00:27.38 | DoDaT69 | if I am not mistaken, excahnge 07 can handle tcp sip |
| 00:27.45 | DoDaT69 | which is where the openser would come in |
| 00:27.58 | DoDaT69 | i know you have to use that for lcs 05 integration |
| 00:28.02 | DoDaT69 | or can rather |
| 00:28.11 | r0d3nt | #asterisk on irc.2600.net / irc.hackint.org <3 asterisk |
| 00:28.14 | Maxxed | hum, il have to home work that too |
| 00:28.30 | Maxxed | 2600! the houston meeting is this week |
| 00:28.41 | Maxxed | i have some damn company xmass thing to deal with... eh |
| 00:29.00 | r0d3nt | =( |
| 00:29.22 | *** join/#asterisk Jam0r (i=Jamie@87.127.190.82) |
| 00:29.23 | r0d3nt | good tx 2600 group on 2600net an di hear their meet is cool |
| 00:29.27 | Maxxed | first friday of evy month :D |
| 00:29.40 | r0d3nt | yup yup |
| 00:29.49 | Maxxed | u gota link? |
| 00:29.53 | *** join/#asterisk jwh (i=jwh@scarlett.lon.rewt.org.uk) |
| 00:29.58 | r0d3nt | for the tx 2600 group ? |
| 00:30.09 | Maxxed | yeah? |
| 00:30.41 | r0d3nt | not sure if they have a site, but about a 12-15 people idle the channel, and i hear from their meeting every so often.... |
| 00:31.15 | Maxxed | what chan? |
| 00:31.17 | r0d3nt | http://www.tx2600.com/ |
| 00:31.21 | r0d3nt | #tx2600 |
| 00:31.29 | Jam0r | Hey, just upgraded to 1.4.5, compiled, installed etc, compiled addons, and there doesnt appear to be any mysql module anymore? is there actually supposed to be one in 1.4.5 addons? |
| 00:31.29 | r0d3nt | irc.2600.net or irc.hackint.org , same thing |
| 00:31.31 | Maxxed | ah, those guys are in san antonio i think |
| 00:31.40 | Maxxed | cool, thx :) |
| 00:31.41 | DoDaT69 | (Maxxed): http://technet.microsoft.com/en-us/library/aa996831.aspx |
| 00:32.01 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 00:32.05 | r0d3nt | Maxxed: np =) |
| 00:32.06 | DoDaT69 | Session Initiation Protocol (SIP) over Transmission Control Protocol (TCP) |
| 00:32.17 | DoDaT69 | <PROTECTED> |
| 00:32.24 | Maxxed | DoDaT69 very nice! |
| 00:32.32 | DoDaT69 | take a look @ office communications server 2007 |
| 00:32.34 | Maxxed | im not running 07, and wont be for a while, but thats cool |
| 00:32.36 | DoDaT69 | thats even better |
| 00:32.40 | DoDaT69 | me either.. |
| 00:33.02 | DoDaT69 | office communications server 07 will allow laptop users to use the messenger client for soft phone as well |
| 00:33.30 | DoDaT69 | complete 100% unified communications without cisco, IF you can get openser workign the right way |
| 00:33.50 | r0d3nt | GL with that |
| 00:34.00 | DoDaT69 | exactly |
| 00:34.16 | Maxxed | nah im cool, i think il stick with my good old trusty asterisk |
| 00:34.21 | Maxxed | i dont need any more windows shit |
| 00:34.29 | r0d3nt | haha |
| 00:34.37 | DoDaT69 | wish I could dodat |
| 00:34.51 | r0d3nt | i wonder why asterisk/digium didn't get in on the " Supported VoIP Gateways " |
| 00:35.01 | DoDaT69 | they want to be special |
| 00:35.02 | Maxxed | iv gotten almost all of my cutomers on linux back ends |
| 00:35.10 | Maxxed | hehe |
| 00:35.56 | DoDaT69 | most of my clients are small business server |
| 00:36.01 | DoDaT69 | so 07 doesnt apply yet |
| 00:36.24 | DoDaT69 | 1/2 will need to upgrade their server to x64 |
| 00:37.54 | yassine | DoDaT69: as soon as i dial asterisk claim the sip client is no more reachable |
| 00:38.28 | DoDaT69 | sounds like you have a jacked up network |
| 00:38.35 | DoDaT69 | check and make sure you have no iptables rules |
| 00:38.41 | DoDaT69 | that your machine is wide open |
| 00:38.56 | DoDaT69 | sip is just whats used to initiate and control the session, otherwise its rtp |
| 00:39.01 | DoDaT69 | if anything blocks that you have nothing |
| 00:39.28 | DoDaT69 | if you are unreachable, check your registration time on your endpoint |
| 00:39.38 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 00:39.42 | kand | yassine: what is your setup? are your phones one the same subnet as asterisk? |
| 00:40.01 | yassine | kand: yes |
| 00:40.19 | [hC] | wow.. how misleading.. GROUP()=groupname is not used the same way GROUP_COUNT(groupname) is. |
| 00:40.27 | kand | yassine: there is no NAT correct? |
| 00:40.38 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 00:41.16 | yassine | well im in the same subnet BUT i have the external ip of my asterisk box wich is router via my dsl router |
| 00:41.32 | DoDaT69 | (yassine): you need to just talk directly to your asterisk server |
| 00:41.42 | DoDaT69 | (yassine): and redirect the ports |
| 00:41.55 | DoDaT69 | you are trying to go out and back in, taht usually dont work well |
| 00:42.43 | kand | yassine: so your asterisk is not in the same subnet? If it is use the private ip to access it like DoDaT69 said. |
| 00:43.15 | *** join/#asterisk yassine (n=yassine@unaffiliated/yassine) |
| 00:43.41 | yassine | DoDaT69: now that im at home it does not |
| 00:46.55 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-2faac2fff7b72465) |
| 00:48.47 | NovceGuru | Hello guys, does anybody know of a service similar to voicepulse (basically a cheap DID provider) but also offers a service to forward to another number if your * box goes down |
| 00:49.10 | *** join/#asterisk neoalex (n=chatzill@user-0ccengj.cable.mindspring.com) |
| 00:49.25 | NovceGuru | I'm not sure many, if any exist |
| 00:49.33 | neoalex | hi guys, does anyone know how I can remove the first digit of the CID on an incoming call |
| 00:49.52 | neoalex | I get the caller ID with a 1 in front from one provider, and without one from another |
| 00:50.05 | neoalex | I would like to see it without the one for the first one too |
| 00:50.25 | NovceGuru | wouldn't that be complex to determine how long the CID is, then do things based on that? |
| 00:51.11 | neoalex | the CID is 10 digits long |
| 00:51.40 | neoalex | standard us phone number, they should all be 10 digits long right? |
| 00:51.44 | kand | You have three options, check the length like NovceGuru said, send the provider with the one to a different context and stip there, or use regex |
| 00:52.11 | kand | Actualy there are probably more |
| 00:53.08 | kand | this is what I use: exten => s,n,Set(CALLERID(num)=$["${CALLERID(NUM)}" =~ "${REGX_NUM_VALIDITY}"]) |
| 00:53.32 | kand | and in extensions.conf [global] REGX_NUM_VALIDITY => ^\+{0\,1}1{0\,1}([2-9][0-9]{2}[2-9][0-9]{6}|011[0-9]*|211|311|411|511|711|911) |
| 00:53.51 | kand | You could clean the regex I use it for other things |
| 00:55.20 | neoalex | looks pretty clean to me, us numbers, international and then emergency and whatnot |
| 00:55.28 | *** join/#asterisk nhuisman_work (n=nhuisman@aeko.IfA.Hawaii.Edu) |
| 00:55.45 | nhuisman_work | hey is it possible to use asterisk with a t1 external gateway instead of a pci/pcie card? |
| 00:55.50 | kand | oh sorry, there should be () around the numbers you want to keep in internaltional.... |
| 00:55.52 | nhuisman_work | then use a dial plan to handle redundancy? |
| 00:56.19 | neoalex | so arround these? |
| 00:56.46 | kand | nm....I group all of them |
| 00:56.59 | kand | it is correct as is, getting late |
| 00:57.39 | nhuisman_work | anyone know about using asterisk like that? |
| 00:59.06 | kand | nhuisman_work: ya, it is possible. Give me a moment there is one that a friend of mine used. |
| 00:59.27 | nhuisman_work | it just seems like a nicer way to do it since then your t1 connection is on a box that will almost never fail |
| 00:59.33 | nhuisman_work | no hard drives, moving parts, etc |
| 00:59.46 | kand | nhuisman_work: agreed, http://www.thevoipconnection.com/store/catalog/Redfone-foneBRIDGE2-Dual-p-16428.html |
| 01:00.19 | kand | nhuisman_work: there are other too, I just know of that one |
| 01:00.21 | nhuisman_work | that works? |
| 01:00.30 | nhuisman_work | how does it communicate to asterisk? |
| 01:00.55 | kand | nhuisman_work: I have been told very well. Zaptell. You map it using its MAC (must be on the same subnet) |
| 01:01.08 | neoalex | ummm... kand it changed the CID to 0 |
| 01:01.23 | nhuisman_work | that's a much nicer way then pci cards |
| 01:01.24 | nhuisman_work | by far |
| 01:01.30 | nhuisman_work | as long as that box is robust |
| 01:01.46 | kand | nhuisman_work: no personal experience but I hear it is... good luck |
| 01:02.06 | nhuisman_work | that's for two t1 connections eh |
| 01:02.10 | nhuisman_work | wonder if they make one with 1 |
| 01:02.35 | nhuisman_work | eh screw it, that's only 1k |
| 01:02.37 | neoalex | quick question: does anyone have a stanaphone account their not using anymore |
| 01:02.45 | kand | neoalex: try show globals on the cli and make sure the regex is there |
| 01:03.09 | neoalex | it's not |
| 01:03.17 | neoalex | dialplan reload is not enough I guess |
| 01:03.28 | nhuisman_work | i guess the next question is, how do you get your asterisk servers to talk to it in a manner that allows for a server to fail and the phones to work with the second server |
| 01:03.31 | kand | neoalex: that is why. ya I think you need to restart to load globals |
| 01:04.26 | kand | nhuisman_work: http://blog.tmcnet.com/blog/tom-keating/asterisk/redfone-fonebridge-quad-span-t1-for-asterisk.asp |
| 01:04.32 | *** join/#asterisk marexz (n=marexz@marexz.mil.lv) |
| 01:04.48 | neoalex | kand: did restart still not showing in globals |
| 01:05.08 | kand | pastebin your extensions.conf |
| 01:06.13 | nhuisman_work | yeah but that says it's ha compatible, doesn't explain how to make asterisk ha |
| 01:06.22 | nhuisman_work | anyone know of resources on how to make asterisk ha with a dialplan? |
| 01:07.00 | kand | nhuisman_work: I use dundi for that |
| 01:07.48 | nhuisman_work | that's already in asterisk eh |
| 01:08.08 | kand | nhuisman_work: ya. But I use pure voip termination on a round robin |
| 01:08.20 | nhuisman_work | what do you mean by pure voip termination? |
| 01:08.31 | neoalex | kand: http://pastebin.com/d7f6b73ae |
| 01:08.55 | kand | nhuisman_work: no pri, or direct PSTN connection (sip trunk to multiple providers) |
| 01:09.01 | nhuisman_work | oh |
| 01:09.42 | kand | neoalex: sorry it should be [globals], like I said it is getting late..... |
| 01:10.24 | nhuisman_work | i wonder if asterisk business edition has dundi in it |
| 01:10.42 | kand | nhuisman_work: I cant image it doesnt |
| 01:10.56 | kand | nhuisman_work: but I dont know |
| 01:10.56 | neoalex | kand: no problem, it works now, thanks a lot |
| 01:11.04 | kand | np |
| 01:11.04 | nhuisman_work | yeah i just wasn't sure if dundi was a new thing or not |
| 01:11.23 | kand | nhuisman_work: been around since at least early, 1.2. BRB smoke break |
| 01:13.20 | nhuisman_work | so when you edit a config or add a phone, how does that work with dundi |
| 01:14.32 | yassine | kand: DoDaT69 here is my network : http://img170.imageshack.us/img170/2962/mynetworkaq9.png |
| 01:15.23 | Jam0r | any known bugs using * with mysql, mysql located on a seperate server - seem to get timeouts, then asterisk hangs, have to restart it to be able to re-reg phone etc, and it repeats after x amount of time |
| 01:16.10 | *** join/#asterisk alexmeyer (n=nothing@c-68-54-121-7.hsd1.in.comcast.net) |
| 01:16.49 | alexmeyer | ok, so what exactly does the "ZT_CHANCONFIG failed on channel 1: Invalid argument (22)" error mean? |
| 01:17.08 | Nugget | Arguments is two doors down on the left, this is abuse. |
| 01:17.21 | alexmeyer | lol |
| 01:17.37 | alexmeyer | we have an fxo card (4 ports being used on it), using fxs_ks signalling |
| 01:17.42 | alexmeyer | but it freaks out |
| 01:17.45 | nhuisman_work | kand let me know when you get back, wanted to ask a few more questions. |
| 01:19.49 | kand | nhuisman_work: shoot |
| 01:20.41 | nhuisman_work | kand: so dundi handles between pbx routing of calls |
| 01:21.05 | nhuisman_work | kand: does that also work at the phone level? Can the phones remain working if their pbx dies? ie be re-routed |
| 01:21.18 | kand | yassine: Alright, for your house us * private IP then foward your private IP via your router |
| 01:22.28 | kand | nhuisman_work: Most phones can simulatously register to multiple severs and us the primary unless unavaliable. Then dundi can locate the server the phone is using |
| 01:22.38 | kand | s/us/use/g |
| 01:22.52 | yassine | kand: is this a normal thing: http://rafb.net/p/WB8WbW63.html |
| 01:22.56 | nhuisman_work | hmm |
| 01:23.09 | nhuisman_work | i guess the question is then how do you keep configs in sync? |
| 01:23.18 | nhuisman_work | if you add a phone on one server |
| 01:23.21 | nhuisman_work | do you add it twice |
| 01:23.23 | nhuisman_work | etc etc |
| 01:23.34 | nhuisman_work | or edit phone information. |
| 01:23.41 | kand | nhuisman_work: I have one primary server and the rest rsync every 10 minutes |
| 01:23.53 | nhuisman_work | is it a simple rsync and restart asterisk? |
| 01:24.07 | kand | nhuisman_work: You still have to reload but you could write a script that if rsync made a change then reboot |
| 01:24.10 | kand | sorry restart |
| 01:24.16 | kand | sorry... reload |
| 01:24.19 | nhuisman_work | i was going to say, ouch reboot :P |
| 01:24.20 | kand | *gesh |
| 01:24.34 | nhuisman_work | so your script does restart asterisk then |
| 01:24.37 | nhuisman_work | if there is a change |
| 01:25.06 | kand | nhuisman_work: it use regex in the rsync log and make the decision |
| 01:25.23 | nhuisman_work | sounds reasonable. |
| 01:25.36 | nhuisman_work | do you know of a way to determine of phone x can register to multiple servers? |
| 01:25.49 | nhuisman_work | i have cisco 7940s, 7960s, and 1735s |
| 01:26.00 | kand | yassine: change your phones to use the internal ip and disable qualify for now, see if that makes your problems go away |
| 01:26.39 | yassine | kand: ok let my try that |
| 01:26.40 | kand | nhuisman_work: ciscos and polcoms are the only two I know for sure |
| 01:27.07 | nhuisman_work | cisco is about to get the big boot in the face for trying to charge me so much to upgrade our voip |
| 01:27.37 | kand | nhuisman_work: I hate ciscos but clients like the look, I higly recommend polycoms |
| 01:28.32 | kand | yassine: pastebin your sip.conf with the changes, specificly the general section |
| 01:28.51 | yassine | okay |
| 01:29.19 | *** join/#asterisk cesar_CR (n=cr@celord.ice.co.cr) |
| 01:29.26 | alexmeyer | anyone else? what's the deal with the "ZT_CHANCONFIG failed on channel 1: Invalid argument (22)" error? |
| 01:29.29 | nhuisman_work | we use polycoms |
| 01:29.32 | nhuisman_work | they work pretty good |
| 01:29.57 | DoDaT69 | (alexmeyer): http://www.digitalson.com/content/view/33/32/ |
| 01:29.59 | nhuisman_work | well I think I'll download asterisk now and start trying this out |
| 01:30.33 | Qwell | nhuisman_work: 1735s? |
| 01:30.35 | kand | nhuisman_work: go for it |
| 01:30.35 | alexmeyer | thanks, I'll try that |
| 01:30.47 | DoDaT69 | (alexmeyer): np |
| 01:30.49 | yassine | kand: sip.conf : http://rafb.net/p/4dI1Vj76.html |
| 01:30.52 | nhuisman_work | we use polycom for our h323 |
| 01:30.59 | nhuisman_work | video conferencing |
| 01:31.33 | nhuisman_work | the only think i'm a little afraid of is making sure that redfone thing will work with our pri |
| 01:32.02 | kand | yassine: I need the includes too. Did your changes help? |
| 01:32.08 | nhuisman_work | it lists some set of pri switchs it's compatible with, just not sure what we are connected to |
| 01:33.01 | alexmeyer | DoDaT69: can do I disable the usb then? do I have to unplug something from the motherboard itself? (btw, I'm no n00b... just dunno in this situation... :P) |
| 01:33.06 | d-k-t | alexmeyer, what sort of modules do you have? |
| 01:33.08 | kand | nhuisman_work: no idea about that, but I would image it works will all standard pris |
| 01:33.16 | nhuisman_work | yeah, probably. |
| 01:33.26 | alexmeyer | wow, I dunno what I typed at the beginning there... O_o |
| 01:33.48 | nhuisman_work | kand, so actually dundi isn't very useful for two servers is it |
| 01:33.52 | nhuisman_work | in one site |
| 01:34.07 | alexmeyer | d-k-t: anything specific you're windering about? |
| 01:34.08 | tzafrir_home | alexmeyer, please pastebin your zapata.conf and the output of cat /proc/zaptel/* |
| 01:34.15 | alexmeyer | ok |
| 01:34.16 | nhuisman_work | since there is really only one site and no routing to be done. |
| 01:34.40 | kand | nhuisman_work: not really....redundance is built into multiple registrations and redfone/round robin trunks |
| 01:35.05 | nhuisman_work | we don't have two trunks but shrug if our trunk dies it's the phone companies problem not mine. |
| 01:35.38 | d-k-t | alexmeyer, well, where ztcfg.c specifies this message, it also gives the message, "Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling?" as an additional message for errorcode 22... |
| 01:35.53 | kand | nhuisman_work: there is always a weak point, just make sure it isnt under your control :) |
| 01:36.03 | nhuisman_work | exactly :) |
| 01:36.11 | nhuisman_work | sorry guys no phones, not my fault |
| 01:36.12 | alexmeyer | http://rafb.net/p/9o0d0E63.html |
| 01:36.48 | alexmeyer | d-k-t: yes, it says that when it gives that error. it's an fxo card, so we configured it as fxs_ks |
| 01:37.58 | kand | nhuisman_work: BTY two of my production servers are formated CM boxes |
| 01:38.10 | nhuisman_work | hahaha |
| 01:38.20 | nhuisman_work | i'm at cm 3.1 |
| 01:38.29 | nhuisman_work | and going to 6.0 looks like a huge pain in the ass |
| 01:39.05 | kand | nhuisman_work: it was an early version when I started here and what little experience I had with it I didnt like |
| 01:39.20 | yassine | kand: now the list is complete : http://rafb.net/p/Dpserr59.html |
| 01:39.29 | tzafrir_home | alexmeyer, the entry for that card appears to be in /etc/asterisk-zapata-auto.conf |
| 01:39.37 | tzafrir_home | But the driver has failed to load |
| 01:39.42 | kand | yassine: ok. and are your phones working? |
| 01:41.00 | alexmeyer | tzafrir: say huh? |
| 01:41.58 | yassine | kand: no |
| 01:42.16 | kand | yassine: what did they do? |
| 01:42.33 | yassine | seems that they dont get registred |
| 01:42.52 | yassine | sip show peers: 007 (Unspecified) D N 0 UNKNOWN |
| 01:43.43 | tzafrir_home | alexmeyer, /proc/zaptel/* show that the card's driver has failed to detect it or simply hasn't loaded |
| 01:43.50 | tzafrir_home | lsmod | grep ^zaptel |
| 01:43.53 | *** join/#asterisk klyrelion (n=Kevin@boyne.demon.co.uk) |
| 01:43.57 | kand | yassine: ok comment out fromdomain, externhost, localnet and qualify (in 007) then pastebin a sip debug during registration |
| 01:44.13 | yassine | okay |
| 01:44.15 | kand | yassine: I want to know where your phone think your box is at |
| 01:44.22 | alexmeyer | it's loaded |
| 01:44.36 | alexmeyer | but it's (I think) only using ztdummy |
| 01:44.56 | tzafrir_home | what is the output of that command? |
| 01:45.03 | alexmeyer | lsmod? |
| 01:45.09 | tzafrir_home | lsmod | grep ^zaptel |
| 01:45.11 | alexmeyer | zaptel 196740 11 ztdummy,wctdm,wcfxo,wcte11xp,wct1xxp,wct4xxp,tor2 |
| 01:45.57 | tzafrir_home | rmmod wcfxo; modprobe wcfxo; dmesg | tail |
| 01:46.13 | alexmeyer | we (I'm actually helping my dad set up this server) had to manually enter in the settings because genzaptelconf didn't add in stuff for this card |
| 01:47.45 | alexmeyer | dmesg just has a bunch of "Registered tone zone 0 ...blahblah" messages at the end |
| 01:48.14 | *** part/#asterisk klyrelion (n=Kevin@boyne.demon.co.uk) |
| 01:48.15 | nhuisman_work | does anyone know if switchvox has a cli? |
| 01:48.21 | alexmeyer | and it gave that error again when modprobing wcfxo |
| 01:50.07 | tzafrir_home | alexmeyer, it didn't add it because the card's driver failed to load |
| 01:50.25 | tzafrir_home | now, what error do you get from modprobe? |
| 01:50.34 | alexmeyer | ok... my dad thought it was because it's an anolog card, or something |
| 01:50.46 | alexmeyer | the same, "ZT_CHANCONFIG failed on channel 1: Invalid argument (22)" error |
| 01:50.56 | tzafrir_home | ah, ignore that |
| 01:51.00 | alexmeyer | oh |
| 01:51.08 | tzafrir_home | what did you see in dmesg | tail |
| 01:51.37 | alexmeyer | http://rafb.net/p/9omby849.html |
| 01:52.05 | yassine | kand: now the echo test works |
| 01:52.50 | tzafrir_home | alexmeyer, no. Something immediately after failing to load wcfxo |
| 01:52.54 | tzafrir_home | rmmod wcfxo; modprobe wcfxo; dmesg | tail |
| 01:52.58 | kand | yassine: that is promissing. So they are registed and communicting with *. How about outbound calls? Then we can work on your office phone |
| 01:53.01 | alexmeyer | yeah, I did |
| 01:53.28 | tzafrir_home | What card do you have? |
| 01:53.49 | d-k-t | alexmeyer, tried adding debug to the driver? e.g. options wctdm opermode=CHINA debug=5 |
| 01:53.50 | alexmeyer | I think it's a TDM0800 or something |
| 01:54.08 | tzafrir_home | ah, that one has a different driver |
| 01:54.15 | alexmeyer | ah |
| 01:54.18 | tzafrir_home | modprobe wctdm24xxp |
| 01:54.40 | tzafrir_home | What version of zaptel do you use? |
| 01:54.40 | alexmeyer | should I rmmod wcfxo first? or does that stay anyway? |
| 01:55.15 | tzafrir_home | leave it be. It is harmless |
| 01:55.28 | alexmeyer | ok |
| 01:55.30 | alexmeyer | 1.2.3? |
| 01:55.57 | tzafrir_home | something you downloaded recently? |
| 01:56.20 | tzafrir_home | anyway, any news from the modprobe? |
| 01:56.47 | alexmeyer | well, for our production boxes (my dad and a friend are doing a voip buisiness) we use AAH2.7, with a modification file we have |
| 01:57.14 | *** join/#asterisk mog (n=mog@c-68-62-172-83.hsd1.al.comcast.net) |
| 01:57.14 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:57.43 | alexmeyer | it just gave the same "invalid argument" error, and "FATAL: Error running install command for wctdm24xxp" |
| 01:59.17 | tzafrir_home | cat /sys/module/zaptel/version |
| 01:59.51 | alexmeyer | no version file |
| 02:00.11 | tzafrir_home | bah. centos4? |
| 02:00.21 | alexmeyer | yep |
| 02:00.24 | tzafrir_home | modinfo zaptel | grep ^version |
| 02:00.25 | *** join/#asterisk RoyK (n=roy@ip-219-6-149-91.dialup.ice.no) |
| 02:00.27 | alexmeyer | AAH 2.7 |
| 02:00.35 | tzafrir_home | that's ancient |
| 02:00.41 | alexmeyer | heh |
| 02:01.12 | alexmeyer | it's the latest one my dad liked |
| 02:01.12 | tzafrir_home | I hope that this is not a new installation |
| 02:01.12 | alexmeyer | it is... |
| 02:01.12 | tzafrir_home | If so: trash it and get something newer |
| 02:01.15 | alexmeyer | heh |
| 02:01.25 | tzafrir_home | AAH is now called TrixBox (CE) |
| 02:01.39 | alexmeyer | that's right... the modinfo version info for this version of zaptel is a huge jumble of gibberish |
| 02:01.43 | alexmeyer | yeah, I know. |
| 02:02.02 | tzafrir_home | really. Don't waste your time on that old junk |
| 02:02.07 | alexmeyer | there was some stuff in later versions he didn't like, so just modified aah 2.7 |
| 02:02.10 | alexmeyer | heh |
| 02:02.11 | alexmeyer | I |
| 02:02.18 | alexmeyer | I'll let him know... |
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| 02:02.43 | tzafrir_home | You don't have to use TrixBox / AAH |
| 02:02.56 | alexmeyer | I know |
| 02:03.29 | tzafrir_home | anyway, you'll need a newer zaptel for that card to be detected |
| 02:03.38 | alexmeyer | heh, if I were doing it, I'd probably slap asterisk onto gentoo or something... ;) |
| 02:03.42 | alexmeyer | ok |
| 02:04.05 | alexmeyer | d'oh, netsplit |
| 02:04.07 | tzafrir_home | florz and co. seem to agree |
| 02:04.12 | alexmeyer | lol |
| 02:04.27 | alexmeyer | k, thanks for your help |
| 02:04.30 | alexmeyer | :) |
| 02:04.55 | yassine | kand: outbound works too i mean from intern sip to extern pstn using the local zap trunk |
| 02:05.26 | kand | yassine: Ok, so currently what doesnt work? |
| 02:05.55 | *** join/#asterisk skymeyer (n=skymeyer@bxlsrvit03.itconnect.be) [NETSPLIT VICTIM] |
| 02:06.08 | yassine | sip from outside |
| 02:06.29 | yassine | and from inside if asterisk is configured in NAT mode |
| 02:06.55 | d-k-t | zaptel 1.2.3 didn't even have a wctdm24xxp module did it? |
| 02:07.42 | kand | yassine: good, what kind of router do you have? |
| 02:07.45 | *** join/#asterisk saftsack (n=oliver@p4FC74AD1.dip.t-dialin.net) |
| 02:09.45 | yassine | kand Siemens sx541 WLAN DSL |
| 02:09.46 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) [NETSPLIT VICTIM] |
| 02:10.21 | kand | yassine: Ok, uncomment externhost, localnet but I can't help you with port fowarding on that if you have not done so already. |
| 02:10.53 | yassine | i already did mapped extern port UDP from 10000-20000 to asterisk box |
| 02:10.53 | kand | yassine: leave fromdomain alown |
| 02:10.59 | yassine | and 5060 UDB too |
| 02:11.15 | kand | yassine: good, (udp) |
| 02:11.31 | yassine | oups yes sorry i mean udp |
| 02:11.34 | yassine | :) |
| 02:11.40 | yassine | its late here |
| 02:12.13 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 02:12.27 | kand | yassine: I know how it is... lol. Let me know if when you uncomment externhost and localnet if your local phones keep working. |
| 02:12.59 | *** join/#asterisk rob0 (i=rob0@sorry.nodns4.us) |
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| 02:14.14 | yassine | kand working fine |
| 02:15.12 | kand | yassine: How about your office phone, did it register or is it not going to (ie off, at home)? |
| 02:16.07 | yassine | i can not test now since im at home :) |
| 02:17.57 | kand | yassine: That is what I thought. But I think you shouldn't have any problems. Two things: leave fromdomain commented and unless you really need it qualify. You may want to make sure your ssh port is fowared too :) |
| 02:19.37 | yassine | kand: ssh is always forwarded thanks for your help! |
| 02:19.50 | kand | yassine: np |
| 02:21.53 | *** join/#asterisk Psykick (n=anon@60.234.198.147) |
| 02:23.59 | *** join/#asterisk [Outcast] (n=outcast@203-114-166-26.eth.sta.inspire.net.nz) |
| 02:25.21 | [Outcast] | does anyone know if there is a way to get a sipura device to send a call to * as soon it is taken off hook |
| 02:26.09 | Psykick | [Outcast]: I would suggest maybe using one of the dialplan features called Originate .... but I may be wrong |
| 02:27.30 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
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| 02:32.53 | [Outcast] | are there any phones that support sending call soon as it goes offhook? |
| 02:33.49 | nhuisman_work | does anyone know where I can find information about registering the same phone to 2 asterisk boxes? |
| 02:33.51 | nhuisman_work | via sip. |
| 02:34.38 | Psykick | nhuisman_work: I'd assume that would be a feature of the phone |
| 02:34.56 | nhuisman_work | i'm trying to find out how to find that information for cisco phones |
| 02:35.52 | Psykick | cisco's website would be the best place to start then I would assume |
| 02:36.55 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 02:37.08 | nhuisman_work | yeah hunting there |
| 02:39.02 | WilliamK | is t.38 enabled by default in * 1.4 nowdays? |
| 02:43.13 | kand | nhuisman_work: I can tell you if it is sip |
| 02:43.40 | kand | WilliamK: no t38pt_udptl=yes |
| 02:43.55 | kand | WilliamK: in sip.conf |
| 02:44.03 | WilliamK | I don't have to patch anything correct? |
| 02:44.06 | WilliamK | I set that |
| 02:44.25 | WilliamK | it looked like * was still having a problem with unknown codec 100 |
| 02:44.26 | nhuisman_work | kand where is the config setting to add the secondary server? |
| 02:44.27 | [Outcast] | found it...hahaha http://forums.linksys.com/linksys/board/message?board.id=VoIP_Adapters&message.id=1711 |
| 02:44.30 | nhuisman_work | on the phone or on the pb? |
| 02:44.30 | nhuisman_work | pbx |
| 02:44.32 | De_Mon | nhuisman_work you're looking for a way to setup multiple lines or sip accounts |
| 02:44.37 | De_Mon | on the phone |
| 02:44.48 | nhuisman_work | i want to use two asterisk servers and the phones don't care if one goes down. |
| 02:44.59 | nhuisman_work | instead of using linux ha |
| 02:45.09 | nhuisman_work | if that's possible |
| 02:45.40 | kand | nhuisman_work: for cisco 7940 running sip SIP000E8494F3FB.cnf looks like: http://pastebin.ca/815482 |
| 02:46.03 | kand | nhuisman_work: all the 7940 and 7960 series actually |
| 02:46.22 | nhuisman_work | the proxy address right? |
| 02:46.22 | kand | nhuisman_work: 7970s can but I dont have an example |
| 02:46.25 | kand | ya |
| 02:46.42 | kand | you need polycom examples? |
| 02:46.59 | nhuisman_work | so this is in the firmware load then |
| 02:47.16 | kand | nhuisman_work: ya but we call it provisiong |
| 02:47.17 | nhuisman_work | 7935 would be my polycom phone, I assume it's not that hard to find. |
| 02:47.22 | kand | provisioning |
| 02:47.58 | nhuisman_work | is that a per phone cnf, or does it replace the extension name and stuff with macros before it send sit to the phone |
| 02:48.12 | kand | Just incase <localcfg> <server voIpProt.server.1.address="vg01.gocentrix.net" voIpProt.server.2.address="vg02.gocentrix.net"/></localcfg> |
| 02:48.30 | kand | per phone but it could be done in default |
| 02:48.37 | kand | On my setup I need to do per phone |
| 02:48.41 | nhuisman_work | ah |
| 02:49.04 | nhuisman_work | so is the failover instantaneous ? |
| 02:49.13 | nhuisman_work | obviously calls will drop |
| 02:49.15 | kand | The call you are on is lost but yes |
| 02:49.17 | kand | lol ya |
| 02:49.26 | nhuisman_work | thats good enough for me |
| 02:49.51 | kand | It is actually registered the whole time just doesnt us it |
| 02:50.59 | nhuisman_work | i wonder if it's possible to download asterisk and then upgrade it to be, or if there are actually alot of changes |
| 02:51.55 | kand | nhuisman_work: asterisk so far is comptatable unless you are using a addtional feature that didnt exist in an older version |
| 02:52.31 | nhuisman_work | sorry i should have expanded that acronym, by "be" i meant asterisk business edition |
| 02:53.55 | kand | nhuisman_work: ah, np. As I understand it they are really on in the same just business edition is built with less features to be more stable but I dont really know much about it |
| 02:54.58 | kand | nhuisman_work: I have found the open source version reliable I wouldn't mind having the support sometimes tho.... |
| 02:55.25 | nhuisman_work | i just kind of wanted them to incorporate the asterisk now gui |
| 02:55.26 | kand | nhuisman_work: but I have been able to work through issues |
| 02:55.31 | nhuisman_work | to make it easier to manage once it's setup. |
| 02:55.44 | kand | nhuisman_work: it is over rated.... :) |
| 02:56.02 | nhuisman_work | really? |
| 02:56.08 | kand | the gui |
| 02:56.10 | nhuisman_work | it just seems easier to add and remove phones, conferences, etc |
| 02:56.54 | kand | possibly, if you write a good dial plan and organize your code well I think it is just as easy |
| 02:57.18 | kand | *if you know what your doing* |
| 02:57.26 | kand | there is the catch |
| 02:57.47 | nhuisman_work | yeah plus i'm not going to be around forever so the next person is going to have to learn asterisk |
| 02:59.07 | kand | nhuisman_work: Doesnt sound like your problem. lol but there is something to be said for being able to delegate tasks. |
| 02:59.42 | nhuisman_work | yeah, like create a new extension for this, oh yea and don't fuck up my whole dial plan with one config error mr student hire. |
| 03:00.35 | kand | nhuisman_work: exactly, and just a side not asterisk dial plans are fault tolerant (ie a missed type line is ignored) |
| 03:00.42 | kand | s/not/note/g |
| 03:01.04 | nhuisman_work | in what way? |
| 03:01.53 | kand | they phrase the files but anything that doesn't fit is ignored and a notice is generated but all good lines are processed anyway |
| 03:02.12 | nhuisman_work | oh |
| 03:02.15 | nhuisman_work | that's useful |
| 03:02.22 | kand | nhuisman_work: I make typos dailly on our production server and nobody notices.....lol |
| 03:02.27 | nhuisman_work | snicker |
| 03:02.57 | kand | nhuisman_work: alright buddy. I am heading out. Good luck! |
| 03:03.09 | nhuisman_work | later, thanks much for the info |
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| 03:13.54 | *** part/#asterisk alexmeyer (n=nothing@c-68-54-121-7.hsd1.in.comcast.net) |
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| 03:19.52 | don_pobre | hi |
| 03:19.59 | don_pobre | anybody here can help me? |
| 03:20.16 | don_pobre | my asterisk 1.4.15 just keep on crashing down |
| 03:20.25 | don_pobre | made a bt full and i got this |
| 03:20.26 | don_pobre | No symbol table info available. |
| 03:20.27 | don_pobre | #12 0x080f765b in dummy_start (data=0x9b0ac60) at utils.c:843 |
| 03:20.27 | don_pobre | <PROTECTED> |
| 03:20.27 | don_pobre | <PROTECTED> |
| 03:20.27 | don_pobre | <PROTECTED> |
| 03:20.29 | don_pobre | <PROTECTED> |
| 03:20.31 | don_pobre | <PROTECTED> |
| 03:20.33 | don_pobre | #13 0x0051c3db in start_thread () from /lib/libpthread.so.0 |
| 03:20.35 | don_pobre | No symbol table info available. |
| 03:20.37 | don_pobre | #14 0x0047606e in clone () from /lib/libc.so.6 |
| 03:20.39 | don_pobre | No symbol table info available. |
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| 03:37.40 | JunK-Y | don_pobre: see my answers in #asterisk-bugs. |
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| 03:59.22 | riddlebox | does anyone use the latest version of firmware for the grandstream GXP2000 phones? |
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| 04:57.35 | watchy2 | they tk you there? |
| 04:58.19 | [TK]D-Fender | yup |
| 04:58.21 | [TK]D-Fender | just got in |
| 04:59.36 | watchy2 | you ever work aastras? |
| 05:00.59 | *** join/#asterisk asdx (n=diego@adsl-156-18.click.com.py) |
| 05:03.04 | watchy2 | I have multiple phones ringing when someone calls in and they are getting "missed calls" on phones that arent the answer phone. How do you fix that? |
| 05:03.43 | [TK]D-Fender | watchy2, You don't |
| 05:03.56 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 05:04.07 | [TK]D-Fender | watchy2, * has not way to signal to ANY phone to not count a call that * cancels as being "not MISSED" |
| 05:04.38 | mosty | watchy2, most phones have an option to disable missed call notification. it's not something the PBX controls |
| 05:06.28 | watchy2 | ah |
| 05:06.31 | watchy2 | so like |
| 05:06.38 | watchy2 | would that mean even if it missed a call |
| 05:06.42 | watchy2 | legitly |
| 05:06.45 | watchy2 | you wont know? |
| 05:06.56 | watchy2 | no way to have both worlds huh |
| 05:07.11 | mosty | you can either be notified about missed calls or not. take your pick |
| 05:07.46 | watchy2 | ah |
| 05:07.49 | watchy2 | thanks man |
| 05:07.57 | watchy2 | wheres it at in polys? |
| 05:08.01 | watchy2 | the sip.cfg? |
| 05:08.15 | [TK]D-Fender | There IS a SIP signalling means to pass on the "count or not" IIMN, but not implemented in *, and not consistent in field |
| 05:13.03 | watchy2 | you could do it per line key |
| 05:13.04 | watchy2 | right |
| 05:13.10 | watchy2 | on a poly |
| 05:15.49 | watchy2 | maybe not |
| 05:16.39 | watchy2 | feature.8.name="calllist-missed" feature.8.enabled="1" |
| 05:16.45 | watchy2 | i guess thats it |
| 05:20.54 | watchy2 | ah on a poly it seems you can tell it per call line |
| 05:21.02 | watchy2 | dunno about a aastra |
| 05:21.36 | watchy2 | thanks guy |
| 05:21.36 | watchy2 | s |
| 05:21.39 | watchy2 | i love u gotta go |
| 05:23.04 | *** join/#asterisk otaku42 (n=otaku42@madwifi/developer/otaku42) |
| 05:23.40 | otaku42 | moin |
| 05:23.57 | *** join/#asterisk kc9jbo (i=faction8@24-240-69-105.dhcp.mdsn.wi.charter.com) |
| 05:25.21 | otaku42 | i learned that since i want to use the meetme extension, i need a zap timer such as ztdummy. server is running kernel 2.6.22, asterisk is 1.4.something. |
| 05:26.15 | otaku42 | wondering: do i still need a patch like this to improve the accuracy for 2.6 kernels in current ztdummy versions? http://bugs.digium.com/view.php?id=4301 |
| 05:27.23 | Maliuta | anyone know if the cisco 7936 conference stations can be flashed to do SIP instead of SCCP? |
| 05:31.42 | otaku42 | hmmm... i think http://bugs.digium.com/view.php?id=10314 (last comment) answers my question. |
| 05:40.19 | [TK]D-Fender | otaku42, thats over 2 YEARS old. |
| 05:46.11 | otaku42 | [TK]D-Fender: that's why i asked whether it's still required. |
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| 06:07.36 | otaku42 | to give a final answer on my previous question (in case the channel is logged): zaptel 1.4.7.1 doesn't need to be patched, it comes with the capabilities for high accuracy (including support for high resolution timers on kernel 2.6.22 and later) |
| 06:12.20 | blitzrage | some bugs might be outstanding pretty long... but not 2 years :) |
| 06:13.07 | mosty | depends how you define bug |
| 06:14.29 | otaku42 | blitzrage: just wanted to be sure :) |
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| 06:17.22 | AJaymn | There a way to clear out the CDR database? more or less to start over clean? |
| 06:18.51 | mosty | delete from cdr; ? |
| 06:22.34 | AJaymn | mostly I want to delete cdr records from over 6 months ago on a working system... |
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| 06:30.40 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 06:31.09 | DocHolliday | zaptel is complaining i dont have kernel sources, how do i get them? |
| 06:34.22 | DocHolliday | got it |
| 06:36.03 | DocHolliday | zaptel is telling me i dont have ncurses when rpm -qa |grep ncurses shows me having ncurses and ncurses devel |
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| 06:40.11 | craigk | if i want to redirect a call from within a dialplan, is something like Dial(Local/1234/n,,r) the best way, or is there something better ? my problem is that if 1234 is an external number then it does not always ring :( |
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| 06:59.47 | blitzrage | DocHolliday: did you re-run ./configure? |
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| 07:32.30 | mosty | craigk, it would be simpler if you don't use chan_local |
| 07:33.15 | craigk | mosty: oh - how do i redirect a call to an external number then ? I want to to appear as if the call came in via a 'normal' path from a trunk |
| 07:33.43 | mosty | what kind of external number? sip? iax? zap? |
| 07:33.59 | hmmhesays | any number of ways |
| 07:34.04 | hmmhesays | i've never seen dune |
| 07:34.06 | hmmhesays | I may download it |
| 07:36.25 | craigk | mosty: lets say i have a dialplan which detects mobile numbers and treats them different to non-mobile. Imagine that a call comes in via some trunk, and tries to dial an extension. After a while, I want to stop dialing the extension and redirect the call to some userdefined number. So, I want to feed the user defined number back into the dialplan so it can see if it is a mobile number |
| 07:36.55 | craigk | the user defined number can be anything: another extension, a mobile or a non-mobile |
| 07:37.23 | craigk | i have been using Dial(Local/number) to do it ... it seems to work but i was wondering if there was a better way |
| 07:37.24 | mosty | craigk, why don't you just Goto some context that does something if it's a mobile number, and something else if it's not? |
| 07:38.36 | craigk | doh |
| 07:38.48 | mosty | i try to avoid chan_local unless i have to, since it messes with CDR |
| 07:38.59 | craigk | so just GoTo(context,usernumber) |
| 07:39.02 | hmmhesays | yah I just got my adsense account |
| 07:40.53 | craigk | mosty: thanks, i will try that (and keep an eye on the cdr) |
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| 08:09.04 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 08:10.43 | Dr-Linux | anybody active! |
| 08:13.31 | mosty | not really |
| 08:14.30 | Dr-Linux | hey mosty, how are you today |
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| 08:14.37 | Dr-Linux | you always active :) |
| 08:14.56 | mosty | i'm looking forward to the holidays |
| 08:14.59 | Dr-Linux | mosty: i needed a suggestion |
| 08:15.06 | Dr-Linux | :) yeah |
| 08:15.25 | Dr-Linux | merry CM to you in advance |
| 08:15.27 | Dr-Linux | :) |
| 08:15.46 | Dr-Linux | and "Happy Eid" to me :P |
| 08:16.39 | Dr-Linux | mosty: ignore priority and look at this: |
| 08:16.40 | Dr-Linux | exten => 4565566,2,GoToIfTime(08:00-17:00|mon-fri|*|*?open1,1) |
| 08:16.40 | Dr-Linux | exten => 4565566,3,GoToIfTime(17:01-07:59|*|*|*?open2,1) |
| 08:17.11 | mosty | yes? |
| 08:17.28 | Dr-Linux | mosty: and now help me and tell where will call go during weekend? |
| 08:17.54 | R1ck | i was wondering, how do I make a menu, that I want a person who is calling to enter 6 digits, and then if those 6 digits are in some record in some database, to forward the call to a mobile... is that possible? |
| 08:17.55 | Dr-Linux | mosty: actually i'm not feeling well, just coming from hospital just to fix this issue :) |
| 08:18.22 | mosty | the first GotoIfTime won't jump on the weekend. the second one will, but only in that time range |
| 08:18.26 | Dr-Linux | R1ck: yes it is |
| 08:18.44 | Dr-Linux | R1ck: use asterisk database |
| 08:18.45 | *** join/#asterisk shinao1 (n=shinao1@41.222.65.68) |
| 08:18.47 | R1ck | cool. any pointers on where to start? |
| 08:19.20 | mosty | R1ck, http://www.voip-info.org/wiki-Asterisk+cmd+DISA ? |
| 08:19.24 | Dr-Linux | R1ck: create a family key and vaule for it |
| 08:20.16 | Dr-Linux | R1ck: DISA will give you "tone" |
| 08:20.34 | *** join/#asterisk arif_endro (n=arif_end@124.81.122.66) |
| 08:20.40 | R1ck | then DISA is not really what I want.. |
| 08:20.57 | Dr-Linux | mosty: that's that happening |
| 08:21.35 | Dr-Linux | mosty: where will call go during 08:00-17:00 sat-sun ? |
| 08:21.55 | mosty | R1ck, use Background or Read, then either use astdb or and AGI script to lookup an external database |
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| 08:22.15 | mosty | Dr-Linux, it will go to priority 4, whatever that is |
| 08:22.53 | Dr-Linux | mosty: and that's "hangup" |
| 08:22.57 | Dr-Linux | i see |
| 08:23.28 | Dr-Linux | mosty: so i want during weekend call should go to open2,1 aswell |
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| 08:23.33 | mosty | Dr-Linux, GotoIfTime only jumps if the time specification matches the current time. otherwise the call goes to the next priority |
| 08:24.25 | mosty | you probably want a catch-all Goto(closed,s,1) at priority 4 |
| 08:32.40 | R1ck | thanks mosty |
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| 08:45.18 | Dr-Linux | mosty: what about this: |
| 08:45.19 | Dr-Linux | exten => 4565566,4,Goto(open2,1) |
| 08:45.51 | mosty | you may as well replace priority 3 with that |
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| 08:48.06 | agx | guys, what the hell Snom is doing... 6.5.15 and 7.x totally broken firmwares.... |
| 08:48.32 | mosty | what's wrong with 6.5.15? |
| 08:48.33 | Dr-Linux | hhm.. |
| 08:48.42 | mosty | 7.x is beta, of course it's broken |
| 08:48.49 | Dr-Linux | mosty: why is that? just asking to understand |
| 08:49.12 | agx | mosty, 6.5.15 has broken BFL, i press a button and randomly another one start to blink |
| 08:49.29 | Dr-Linux | mosty: check your pvt |
| 08:49.36 | mosty | Dr-Linux, because priority 3 and 4 both jump to the same place. if 3 doesn't do it then 4 definitely does |
| 08:50.08 | Dr-Linux | makes sense |
| 08:50.34 | Dr-Linux | hhm.. |
| 08:50.37 | mosty | so the gotoiftime at priority 3 isn't needed. remove it and the call flows the same way |
| 08:51.00 | Dr-Linux | mosty: but what if i do n't change and left the settings as i shown you in pvt? |
| 08:51.41 | mosty | then the call flow is the same, but you have one possibly confusing line that you don't need |
| 08:52.20 | Dr-Linux | okey thanks, i understand |
| 08:52.40 | Dr-Linux | then maybe my 3rd priority is useless |
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| 09:17.51 | FlatFoot | agx: i'm using 7.1.19 on Snom 320's that seems to be quite stable |
| 09:18.09 | *** part/#asterisk yassine (n=yassine@unaffiliated/yassine) |
| 09:19.25 | otaku42 | mosty: 7.x is no longer beta, it has been released a few days ago iirc |
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| 09:20.17 | mosty | otaku42, oh ok, i'm still using 6.whatever is latest on most of my snoms, except for the 370's (which iirc only support v7) |
| 09:20.18 | codec | hi ther |
| 09:20.22 | codec | *there |
| 09:20.41 | codec | short question.. the rtp ports I can define on my voip phone.. those are local ports, right? (source ports) |
| 09:21.32 | mosty | codec, i believe so |
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| 09:39.50 | agx | FlatFoot, i got a few phones with 7.x that does not register against asterisk 1.4 and i decide to downgrade but they're stuck at 6.5.15 and seems there is no way to force them to 6.5.10/12 (that is the one i NEVER had 1 single problem) |
| 09:40.36 | FlatFoot | agx: i must say my Snoms are reg'd to 1.2.x hang on i'll try to reg to my new 1.4.11 |
| 09:40.38 | mosty | agx, what model are you using? |
| 09:42.43 | FlatFoot | agx: yep that version works together |
| 09:43.06 | agx | mosty, any model 300,320,360,370 |
| 09:44.09 | mosty | agx, and which firmware version? |
| 09:44.45 | mosty | i have 7.1.6 working on a snom300 here, i can check my other models also if you like |
| 09:45.30 | agx | mosty, with asterisk 1.2 or 1.4? |
| 09:45.40 | mosty | both |
| 09:45.59 | agx | mosty, is there some new settings in firmware v7 that can 'cause the phone do not register? |
| 09:46.18 | agx | someone on forum was talking about "watchdog" |
| 09:47.04 | mosty | i'm not sure if it's a setting or a bug in the firmware, but i have this particular snom300 registered to an asterisk 1.2.17 and 1.4.15 |
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| 10:13.27 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 10:13.37 | ice_croft | hi all |
| 10:13.55 | ice_croft | [Dec 14 16:15:20] ERROR[25166]: chan_zap.c:10836 process_zap: Unknown signalling method 'pri_cpe' |
| 10:14.03 | ice_croft | what's that? 8-O |
| 10:14.48 | ice_croft | help please |
| 10:16.46 | mosty | did you compile libpri before zaptel? |
| 10:17.33 | *** join/#asterisk myiagy (n=Jose@200.215.59.133) |
| 10:17.54 | ice_croft | i c |
| 10:17.56 | ice_croft | thanx |
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| 10:23.53 | skrusty | does anyone know if you can change the SDP without recompiling? |
| 10:24.07 | skrusty | i need to change it from 96 to 101 |
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| 10:25.55 | yang | What does the following error mean Dec 14 11:25:06 WARNING[27839]: rtp.c:463 ast_rtp_read: RTP Read too short |
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| 10:37.48 | admgecko | morning |
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| 10:38.44 | alinuxlb-22 | Hi can anyone provide me with a recommendation for a TURN/STUN daemon ? |
| 10:40.21 | tzafrir_home | alinuxlb-22, there's a stund called (surprise) stund in Debian |
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| 11:24.18 | festr_ | hi, anyone using snom pickup patch for snom phones? |
| 11:24.35 | festr_ | 1.4 asterisk |
| 11:25.04 | mosty | festr_, no? my snom phones answer in intercom mode without any special patches |
| 11:25.50 | festr_ | mosty: so you can pickup ringing extension by pressing blinking lamp without any patches? |
| 11:26.33 | mosty | i think we're talking about different features. you're trying to do directed pickups? |
| 11:28.07 | festr_ | mosty: yes |
| 11:28.14 | festr_ | mosty: using BLF feature |
| 11:28.41 | mosty | that works for me with asterisk 1.2 with bristuff, i have not tried it with asterisk 1.4 |
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| 11:31.31 | festr_ | it works for me to with 1.4 but only with patch. unfortunatly, this patch is causing deadlocks |
| 11:33.34 | mosty | i am not a big fan of bristuff to tell you the truth. i would recommend against it if at all possible |
| 11:33.56 | festr_ | mee to |
| 11:34.01 | mosty | you're basically on your own if you have troubles with it :/ |
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| 11:49.35 | ice_croft | i have some problem with chan_zap |
| 11:49.43 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
| 11:49.52 | ice_croft | it's loaded, but i can't enter any of zap commands |
| 11:49.59 | ice_croft | whats' wrong? |
| 11:50.07 | ice_croft | mosty hi |
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| 11:50.22 | *** part/#asterisk klyrelion (n=Kevin@icsnet.demon.co.uk) |
| 11:50.27 | kaldemar | ice_croft: what makes you think it's loaded? |
| 11:50.44 | ice_croft | well, localhost*CLI> module reload chan_zap.so |
| 11:50.44 | ice_croft | <PROTECTED> |
| 11:50.44 | ice_croft | <PROTECTED> |
| 11:50.44 | ice_croft | <PROTECTED> |
| 11:50.48 | ice_croft | this |
| 11:51.24 | mosty | ice_croft, pri issues? |
| 11:51.54 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 11:52.12 | jmls | I seem to have a network problem somewhere, but cannot figure it out |
| 11:52.20 | jmls | I call SIP/5740, |
| 11:52.46 | jmls | and 4 times out of 5, the call is "delayed" for up to 4 seconds before it starts ringing |
| 11:52.47 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 11:52.55 | ice_croft | i built zaptel with pri |
| 11:53.00 | jmls | the other time, it rings straight away |
| 11:53.08 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 11:53.37 | jmls | anyone seen this before ? |
| 11:53.52 | kaldemar | ice_croft: what does "show modules like chan_zap.so" say? |
| 11:55.14 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 11:55.22 | ice_croft | damn |
| 11:55.47 | ice_croft | so anyway |
| 11:56.06 | ice_croft | what should i check? |
| 11:56.38 | kaldemar | your zaptel.conf and zapata.conf |
| 11:57.02 | ice_croft | min |
| 11:57.05 | kaldemar | it you set verbose at 10 and try to reload, the cli will give you a hint on what's wrong. |
| 11:57.29 | ice_croft | kaldemar> no, verbose 10 empty |
| 11:57.50 | kaldemar | i.e. "set verbose 10" and "module reload chan_zap.so" |
| 11:58.17 | ice_croft | kaldemar> i understand |
| 12:00.04 | ice_croft | localhost*CLI> module reload chan_zap.so |
| 12:00.04 | ice_croft | <PROTECTED> |
| 12:00.04 | ice_croft | <PROTECTED> |
| 12:00.04 | ice_croft | <PROTECTED> |
| 12:00.04 | ice_croft | that's all |
| 12:01.05 | kaldemar | well, check your configs, especially that all channels and signallings in zaptel.conf and zapata.confs match. and in the future, don't paste to the channel, use pastebin. |
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| 12:07.18 | ice_croft | %| |
| 12:07.31 | ice_croft | retry any posts to me plz |
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| 12:19.05 | *** join/#asterisk poor_man (n=chatzill@213.63.2.234) |
| 12:19.28 | poor_man | hi all! |
| 12:19.35 | poor_man | anyone here with snom 320 phones accessing to a common address book stored in a server? |
| 12:20.54 | poor_man | inspite of the small built-in 100 address capable, addressbook? |
| 12:26.01 | mosty | they support ldap don't they? |
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| 12:39.25 | poor_man | yes |
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| 12:41.09 | poor_man | mosty, but i'm not very familirized with ldap, and dont want create a contact in AD for each telephone number, because i want have a mysql database, for example, and access it to get contact from suppliers, customers, freinds, internal, etc |
| 12:42.45 | poor_man | mosty, with ldap i wil have "one more thing" connection to my AD in windows 2003 server, and i want centralize voip/telephony stuff in one box, in one network |
| 12:43.01 | mosty | poor_man, i figure that the snom built-in address book is ok for small numbers (<100?), beyond that i would investigate the ldap support |
| 12:43.41 | poor_man | ok, i'll give a deeper look in ldap |
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| 13:13.07 | yang | <PROTECTED> |
| 13:13.16 | yang | how could i fix this |
| 13:14.41 | mosty | yang, what actual problem are you having? |
| 13:14.45 | coppice | either the other end isn't sending RTP (e.g. it is sending UDPTL) or it is sending bad RTP |
| 13:15.18 | lirakis | yang: i get that all the time on cli .. but i never have "problems" |
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| 13:33.30 | yang | I dont have any problem in callings, jsut an annoying error that doesnt make my CLI clear |
| 13:34.01 | yang | And I use a VOIP phone -> trunk -> GSM hardware dispatcher |
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| 13:36.24 | mosty | yang, ignore it. my asterisk console is full of crap like this, that doesn't mean that there's something wrong |
| 13:36.30 | yang | And I use a VOIP phone -> asterisk --> trunk -> GSM hardware dispatcher |
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| 13:38.50 | mosty | yang: ask for help when something isn't working, ignore the message for now |
| 13:41.18 | awk | anyone have issues running asterisk on 2.6.22 kernel? |
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| 13:45.46 | yang | awk: hm no...works fine |
| 13:46.18 | yang | 2.6.22-2-686 debian testing |
| 13:46.43 | awk | I'm going to use a custom kernel.. just want to make sure what to use.. |
| 13:46.51 | Toerkeium | any way to get a fax machine working with asterisk? |
| 13:46.51 | awk | wonder what the safest stable version is, without any local vulns, etc.. |
| 13:47.05 | awk | Toerkeium yup fxs/fxo card :) |
| 13:47.07 | awk | or quintum unit |
| 13:47.14 | awk | or some other media gw |
| 13:47.36 | Toerkeium | awk: how will it work? |
| 13:47.51 | Toerkeium | fxo > asterisk > fxs > fax ? |
| 13:47.51 | awk | well how do you want it to work? |
| 13:48.25 | awk | I just use hylafax and use print to fax function for outbound |
| 13:48.31 | awk | and fax to mail for inbound |
| 13:48.56 | awk | saves the use of using fax devices |
| 13:49.34 | Toerkeium | ah, so you just don't use asterisk for fax? |
| 13:50.03 | coppice | fxo -> asterisk -> fxs -> fax requries the right kind of interfaces, to get any decent reliability. The Digium TDM cards are not the right kind of interfaces. |
| 13:50.12 | awk | Toerkeium didnt you listen to what I said |
| 13:50.15 | awk | I use asterisk completly |
| 13:50.25 | awk | hylafax asterisk ... fax to mail asterisk |
| 13:50.55 | awk | coppice I would never use digium for fxs/fxo, quintum units work wonders |
| 13:50.56 | yang | hylafax is some pain to configure |
| 13:51.05 | awk | yang takes me no longer than 10 minutes |
| 13:51.07 | awk | its really easy.. |
| 13:51.13 | awk | ive now developed it into my front end |
| 13:51.15 | awk | www.scopserv.com |
| 13:51.15 | yang | hm is it? |
| 13:51.36 | Toerkeium | and does asterisk need so extra configuration besides hylafax? |
| 13:51.38 | awk | can do all the config through the front end.. |
| 13:51.44 | coppice | hylafax is only a pain when you try to use an arbitrary modem. use a well supported one, and most of the configuration is already done |
| 13:51.45 | awk | Toerkeium iaxmodem |
| 13:52.01 | awk | corect some tweaks to inittab |
| 13:52.07 | Toerkeium | ahh good |
| 13:52.08 | awk | and init q and you on your way |
| 13:52.18 | awk | also create some /dev/IAX device |
| 13:52.23 | yang | hm for the first time i see this asterisk GUI, and I was searching throught the whole web for it |
| 13:52.48 | Toerkeium | we are going to cur down all analog lines, and since I was told (some time ago) "Don't work with fax+asterisk", I just wondered if there was any news about it |
| 13:52.54 | Toerkeium | so, great |
| 13:53.03 | awk | jaaa, nothing comes close to ScopServ |
| 13:53.09 | awk | not by a long shot |
| 13:53.15 | yang | I would be interested in setting a fax+asterisk too |
| 13:53.17 | awk | there is nothing we have not thought of |
| 13:53.33 | awk | it even has support for skype, etc |
| 13:53.35 | yang | awk: but scopserv requires redhat platform |
| 13:53.55 | awk | we developed it on centos. but you can use alian, etc and convert our rpms to deb, etc.. |
| 13:54.03 | awk | we have people using it on debian and other distros |
| 13:54.08 | awk | but we have developed it on centos.. |
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| 13:54.41 | Toerkeium | awk, is scopserv some sort of freepbx? |
| 13:55.11 | awk | lol |
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| 13:55.16 | awk | freepbx is nothing like scopserv :) |
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| 13:55.23 | awk | our system does everything and more :) |
| 13:55.27 | Toerkeium | not comparing which is best.. |
| 13:55.35 | awk | well its not at all the same |
| 13:56.02 | Toerkeium | I mean, it has the "same use" than freepbx, right? |
| 13:56.21 | awk | hmm, well freepbx has a few features we have if that is what you mean |
| 13:56.51 | Toerkeium | gonna check the demo :=) |
| 13:57.15 | Toerkeium | demo is broken |
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| 13:57.18 | yang | awk: how much does it cost? You need to buy a hardware or is it simply a software? |
| 13:57.51 | awk | you can buy a full unit, soon we throwing out all cards and completley using quintum for pri/fxs/fxo, etc |
| 13:57.57 | awk | as zaptel is a load of shit |
| 13:58.03 | awk | and causes all the problems on asterisk |
| 13:58.10 | awk | so take that out the picture you have a stable solution |
| 13:58.41 | awk | but you can buy a license.. its completely locked down untill you license it.. all php code is compiled.. so not like freepbx as open source |
| 13:59.16 | awk | it has fill intergrated billing |
| 13:59.18 | awk | etc |
| 14:02.08 | [TK]D-Fender | awk: Yup, as GUI's go ScopServ's probably the best out there. |
| 14:02.51 | Greek-Boy | scopserv? is it well suited for call termination providers? |
| 14:03.02 | yang | what is the price for only the software? |
| 14:03.25 | awk | Greek-Boy if you want it to be, yes.. for call centres, anything.. |
| 14:03.34 | [TK]D-Fender | Greek-Boy: Yes, they scale to ITSP level |
| 14:03.41 | awk | yang software is free, its based on number of users |
| 14:03.47 | [TK]D-Fender | Greek-Boy: (within some limit of reason) |
| 14:03.50 | yang | but you can buy a license.. its completely locked down untill you license it.. |
| 14:04.01 | yang | so how mcuh does the license cost? |
| 14:04.04 | awk | yes, once you license it all the functions open up |
| 14:04.11 | awk | yang how many users? |
| 14:04.17 | awk | it goes from 25/50/100/1000, etc etc |
| 14:04.19 | yang | about 30-50 |
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| 14:04.58 | yang | Its just the thing that we run all asterisk's on debian |
| 14:05.11 | yang | and if its for centos, maybe it would be incompatible |
| 14:05.39 | apocn | Hello, Im trying to apply this patch http://bugs.digium.com/file_download.php?file_id=16059&type=bug on apps/app_queue.c using patch < patch_file but it says: Hunk #4 FAILED at 1835. |
| 14:05.41 | apocn | any help? |
| 14:06.26 | awk | yes we dont support debian.. not yet anyway.. |
| 14:06.30 | awk | from next year maybe |
| 14:06.34 | yang | my boss is crying for some decent asterisk GUI, and all I could find was op-panel |
| 14:06.38 | awk | we do support fc/centos/rhel |
| 14:08.21 | Toerkeium | awk: enable a demo |
| 14:09.14 | Toerkeium | it will sell itself without words :) |
| 14:09.22 | awk | Toerkeium we do have a knoppix base on cd |
| 14:09.31 | awk | people are welcome to download that |
| 14:10.47 | yang | awk: query me your product price please |
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| 14:11.37 | awk | geting it sec... |
| 14:12.00 | awk | about 2370 once off dollars, and then 927 per year ther after.. |
| 14:12.19 | awk | maybe a bit less on both.. trying to work out conversion |
| 14:12.44 | yang | ok |
| 14:13.10 | Toerkeium | eeww, pretty expensive |
| 14:13.14 | awk | lol hardly :) |
| 14:13.17 | awk | look what it can do.. |
| 14:13.27 | awk | you can sit there all day coding macros if you like :) |
| 14:13.35 | coppice | can you actually get the 927 per year out of people? |
| 14:13.38 | awk | 1 click of a button I have can dynamic or static agents, etc |
| 14:13.50 | awk | coppice system locks down if you dont pay.. |
| 14:13.57 | Greek-Boy | why would u go for scopserv though when they are so much other tools available out there? free oss stuff... |
| 14:14.02 | awk | it works on system id , seriel, etc.. so you cant copy it or mac address cahnge, etc |
| 14:14.39 | awk | Greek-Boy well then you surely have not worked with the system if you can ask that question.. do some research on what the product does.. then try compare it to other products, then last work out stability issues.. |
| 14:14.41 | yang | Greek-Boy: find a good GUI for asterisk that its free? |
| 14:15.07 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 14:15.33 | mosty | apocn, what is that patch created against, and what are you tring to patch into? |
| 14:15.46 | Greek-Boy | ok guys, get the point |
| 14:15.47 | Greek-Boy | lol |
| 14:15.59 | yang | Well what I have in mind now, is I read that this HUBlite client for windows is only compatible with freepbx, and my boss told me that it should also work with asterisk, but I think it doesnt... |
| 14:16.06 | Toerkeium | awk: I don't tend to say "it's expensive" trying to reduce the product value. It's just faaaaaaaaaaaaaaaw away of my budget |
| 14:16.11 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 14:16.15 | yang | HUDlite |
| 14:16.16 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 14:16.51 | yang | awk: you have told me the price for a 50 users license? |
| 14:17.06 | webman | ever since updating my asterisk to 1.4SVN a few weeks ago, it has been very unstable, and I keep updating again hoping for a fix, but it still eludes me. How can I debug what seems to be chan_zap stops responding to in/out calls but there is no core dump |
| 14:17.16 | awk | yes I did |
| 14:17.24 | awk | <awk> about 2370 once off dollars, and then 927 per year ther after.. |
| 14:17.36 | awk | maybe less, im trying to work out conversion |
| 14:17.44 | mosty | webman, does the problem occur with 1.4.15? |
| 14:17.56 | awk | 1.4.14/15 are buggy |
| 14:18.01 | awk | I sugest you stick to 13 till 16 is released |
| 14:18.19 | webman | mosty: I did a SVN update after the release of 1.4.15 .... about 8 hours ago |
| 14:18.31 | awk | ahh unless you using the svn trunk :) |
| 14:18.56 | mosty | webman, so if you run 1.4.15 does the problem occur or not? |
| 14:19.05 | webman | the problem is I don't have a reliable method to cause the 'crash' but it does happen at least once every two days, (today is 4 times already) |
| 14:19.41 | Greek-Boy | I wonder if scopserv can provide ASR and ACD values? |
| 14:19.45 | webman | mosty: I never use a specific release, I always use SVN 1.4 branch.... |
| 14:20.01 | *** join/#asterisk sheldonh (i=[4Ycn4dP@66.219.59.32) |
| 14:20.03 | awk | webman honestly.. we have done serious tests.. revert back to 1.4.13 till 1.4.16 is released |
| 14:20.07 | awk | 14/15 is very buggy! |
| 14:20.18 | webman | mosty: so I last updated shortly *after* 1.4.15 and it has died 3 times since then |
| 14:20.21 | awk | we are not releasing it with scopserv due to that fact.. |
| 14:20.26 | mosty | webman, running svn releases on production boxes is asking for trouble, imo |
| 14:20.37 | sheldonh | any idea what codec a peer is sending me if my asterisk complains "Unknown RTP codec 102 received from ..."? suddenly started today, and i suspect the peer is being weird, not me |
| 14:20.57 | sheldonh | awk: buggy as in? |
| 14:20.58 | webman | awk: well, what are the problems, and if you know about them, why don't the fixes get added to SVN? |
| 14:21.11 | awk | there are many problems with 14/15 please read up.. |
| 14:21.16 | webman | mosty: 1.4SVN is meant to be the best available stable release..... |
| 14:21.16 | mosty | awk, what kinds of bugs are you seeing with 1.4.15? |
| 14:21.18 | sheldonh | awk: just joined :) |
| 14:21.20 | *** join/#asterisk X-Filez (i=x-files@x-files.lv) |
| 14:21.35 | mosty | webman, no, svn checkouts are not releases |
| 14:21.42 | awk | gogle bugs in 1.4.14 asterisk and 15 |
| 14:21.50 | sheldonh | awk: we route upward of 900 concurrent calls on a single asterisk box, which does passthrough between sip and iax2, without any hassles |
| 14:21.50 | awk | you will see many.. |
| 14:21.51 | webman | mosty: no, they are releases with bug fixes .... |
| 14:22.10 | webman | sheldonh: do you have any zap channels ? |
| 14:22.11 | awk | sheldonh great i'm running over 6k calls an hour on some machines :) |
| 14:22.19 | awk | asterisk 1.4.15 handles laod very nicely.. |
| 14:22.22 | [TK]D-Fender | sheldonh: G.722.1 |
| 14:22.28 | sheldonh | webman: no, this is coming back from a sip peer |
| 14:22.34 | sheldonh | [TK]D-Fender: thanks man :) |
| 14:22.39 | awk | yesterday 1 box, 44k calls for the day |
| 14:22.59 | mosty | webman, no, they are not releases at all |
| 14:23.04 | X-Filez | Hello ppls, I have problems, need help, I have 2 snom 320, and installed asterisk 1.4.15 + app_devstate , need realy configure see status Line (LED on or off), please help |
| 14:23.04 | awk | sheldonh and yes not with all functionality, are you using dynamic agents, etc.. |
| 14:23.21 | webman | sheldonh: I am only having problems with zap now, last week I was getting core dumps, but that has been fixed up now |
| 14:23.27 | [TK]D-Fender | sheldonh: I see another reference to it as iLBC, but that must mean a varient unknown to *... so I'd start thinking the former |
| 14:23.34 | awk | webman zap on what hardware? |
| 14:23.38 | coppice | [TK]D-Fender: still trying to sell Polycoms? :-) |
| 14:24.05 | [TK]D-Fender | coppice: Nope, and YOU pimp WB whereas I don't :) |
| 14:24.15 | webman | awk: wct4xxp module (quad PRI card I think the TE405p) |
| 14:24.35 | awk | ahh, I was going to say that wanpipe had issues with CRC errors being sent to exchange.. |
| 14:24.41 | awk | causing the line to drop |
| 14:24.45 | coppice | [TK]D-Fender: G.722.1 might just be worth implementing, even though its almost polycom only |
| 14:25.07 | [TK]D-Fender | coppice: I'm sure SOMEBODY out there cares... that somebody however would not be me. |
| 14:25.16 | webman | awk: it is a very lightly loaded box, lucky to get 100 calls in a day (probably more like 30 really) but yet still very unstable |
| 14:25.39 | awk | webman please do listen to what I said roll back to 1.4.13 till 1.4.16 is released |
| 14:25.51 | awk | see if that has any benefit what I think it will |
| 14:26.14 | [TK]D-Fender | X-Filez: have you got basic presence working to monitor another SIP device for instance? |
| 14:26.16 | coppice | [TK]D-Fender: don't you feel modern systems should try to suck at least a little less than the PSTN? |
| 14:26.30 | webman | is there a way to see what SVN version a source tree is? ie, is the SVN version stored in a file on my machine after I do a checkout ? |
| 14:26.42 | mosty | i don't have issues with 1.4.15, and i handle many thousand calls per day |
| 14:27.15 | mosty | webman, look in .svn/ - it must be in there somewhere |
| 14:27.19 | [TK]D-Fender | coppice: When it is limited by it in such a huge % or implementations (call crosses the PSTN), why ways CPU when you're dragged to the lowest common denominator anyways? |
| 14:27.22 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 14:27.38 | [TK]D-Fender | coppice: Its not that I don't am against the ideal, I'm jsut a realist. |
| 14:28.23 | coppice | [TK]D-Fender: If Bell had taken that attitude, he would have given up the day he got something working as a curio |
| 14:28.27 | awk | anyone know if any of the other front ends have provisioning? |
| 14:28.51 | X-Filez | [TK]D-Fender: i don't have others sip device, only have snom 320 |
| 14:29.14 | [TK]D-Fender | coppice: Yeah, and 99.999% of the "next big things" that come out are just "The next big thing that'll never be". |
| 14:29.26 | [TK]D-Fender | X-Filez: then setup a soft phone and get testing. |
| 14:29.26 | mosty | X-Filez, describe the problem for us |
| 14:29.32 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.pa.comcast.net) |
| 14:29.36 | coppice | [TK]D-Fender: the heaviest use of VoIP is currently within compnay networks, and they have no real problem migrating to better voice quality |
| 14:29.42 | [TK]D-Fender | mosty: Let him limit the scope of his testing first |
| 14:30.31 | [TK]D-Fender | coppice: to my awareness, most VoIP would be between phone & PBX within a building, or bridging to otherwise LCD limited systems together |
| 14:31.25 | coppice | i think calls between two phone within the same organisation outnumber the external calls |
| 14:31.58 | [TK]D-Fender | coppice: Quite possible in many cases, but companies don't tend to care that much. |
| 14:32.15 | ice_croft | ppl |
| 14:32.21 | ice_croft | need some advice |
| 14:32.39 | [TK]D-Fender | coppice: This could be termed a "chicken & egg" debate. If enough peeople adaopt, it becomes mainstream and the amrket drives down price, and increases proliferation. |
| 14:32.57 | ice_croft | when using e1 pri and zaptel, what kind of framing should i use? |
| 14:33.21 | coppice | [TK]D-Fender: if all you are going to be is a PSTN alternative, the only thing you offer is price. you really want to scrape in the dust for pennies? |
| 14:33.28 | ice_croft | i mean, should e1 be framed, or unframed? |
| 14:33.34 | *** join/#asterisk bkw_ (n=brian@adsl-64-149-47-251.dsl.tul2ok.sbcglobal.net) |
| 14:33.36 | [TK]D-Fender | coppice: Change IS difficult and slow, and the what I think is the biggest drawback (others could say advantage), is that tech moves so FAST now. Who wants to settle on a standard when a better one comes out every week? |
| 14:35.03 | mosty | ice_croft, what asks you that question? |
| 14:35.12 | coppice | the codecs might keep being replaced, but the basic idea of wideband voice should be sticky |
| 14:35.16 | Corydon76-dig | ice_croft: ccs |
| 14:35.41 | X-Filez | i configure app_devstate and get result : i can see called phone LED ON, and talkind time, config extension.conf is here -> http://pastebin.com/m3edfbed9 , Problem is : I call from SIP/11 to SIP/12, wait 2 second and say cancel call, but in SNOM LEDs not OFF |
| 14:35.56 | X-Filez | why, don't understad... |
| 14:36.21 | ice_croft | Corydon76-dig> plz, more |
| 14:36.56 | [TK]D-Fender | ice_croft: ask your TELCO what they are using. |
| 14:37.10 | kaldemar | ice_croft: ^ then take a look at zaptel.conf.sample in the source package |
| 14:37.32 | [TK]D-Fender | X-Filez: That is sad.. you are using the DefvState to EMULATE something you can ALREADY track! |
| 14:37.46 | [TK]D-Fender | X-Filez: Exten => 11,hint,SIP/11 <------ |
| 14:37.47 | ice_croft | kaldemar> that's not zaptel, it's lower, at device leve; |
| 14:38.26 | Corydon76-dig | ice_croft: where do you think zaptel exists? |
| 14:38.50 | ice_croft | Corydon76-dig> ok, i'll ask another way |
| 14:39.17 | ice_croft | Corydon76-dig> my phone provider gives me unframed e1 |
| 14:39.33 | Corydon76-dig | Um, what? |
| 14:39.43 | Corydon76-dig | You mean a data E1? |
| 14:40.00 | ice_croft | Corydon76-dig> no |
| 14:40.03 | ice_croft | Corydon76-dig> phone e1 |
| 14:40.05 | Corydon76-dig | It's still framed, unless it's a dead line |
| 14:40.09 | X-Filez | [TK]D-Fender: do you want say me, me need rmove DevState and configure use hint ? |
| 14:40.20 | *** join/#asterisk phillipk (n=pkey@fw.datafax.net) |
| 14:40.28 | ice_croft | Corydon76-dig> but my pri adapter supports only framed mode for e1 |
| 14:40.39 | Corydon76-dig | ice_croft: well, duh |
| 14:40.41 | [TK]D-Fender | X-Filez: You're jsut trying to reinvent the wheel here... DevState is a REMARKABLY cool and useful thing... for tasks that NEED it. |
| 14:40.55 | [TK]D-Fender | X-Filez: When you're trying to track a phone, let * do its normal thing. |
| 14:41.07 | ice_croft | Corydon76-dig> i mean, when i'm setting unframed mode, i cant configure channel on the device |
| 14:41.41 | [TK]D-Fender | X-Filez: DevState is cool for track things like flags you use to control night-time call routing, server-based DND, and other stuff your phone can't know about normally |
| 14:41.51 | Corydon76-dig | ice_croft: ALL E1s ARE FRAMED, UNLESS THE E1 IS OFF |
| 14:42.01 | ice_croft | Corydon76-dig> hm |
| 14:42.05 | ice_croft | Corydon76-dig> u sure? |
| 14:42.09 | Corydon76-dig | Yes |
| 14:42.48 | ice_croft | gotta think about it |
| 14:42.55 | Corydon76-dig | Goodie |
| 14:42.56 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 14:44.00 | mocker | According to Google there are unframed E1s. |
| 14:44.15 | mocker | Clear channel mode, also known as unframed mode. |
| 14:44.21 | ice_croft | mocker> i thought so, too. |
| 14:44.29 | *** join/#asterisk tc3driver-nii (n=huh@rrcs-24-199-16-118.west.biz.rr.com) |
| 14:44.47 | Corydon76-dig | mocker: that's still framed |
| 14:44.56 | X-Filez | [TK]D-Fender: ok, i remove DevState and use extention.conf is here -> http://pastebin.com/m1fc53863 , LED is not worked |
| 14:45.15 | Corydon76-dig | mocker: doesn't matter what you call it, the E1 is still framed |
| 14:45.33 | ice_croft | Corydon76-dig> man, u just made some more mess |
| 14:45.48 | ice_croft | Corydon76-dig> tau32_0.e1_0(Twin Pair) unframed=on loop=none line=hdb3 higain=off monitor=off scrambler=off |
| 14:46.02 | [TK]D-Fender | X-Filez: You'll need to use "type=peer", "call-limit=99" for your phone setups, and you may have to restart * as it stil tracks the old hints. |
| 14:46.10 | Corydon76-dig | fine. Don't come begging for help if you won't listen. |
| 14:46.12 | ice_croft | Corydon76-dig> when i have this- link is OK, but channels r down |
| 14:46.38 | ice_croft | Corydon76-dig> when i have unframed = off -- channels r OK, the link is down |
| 14:47.06 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:47.06 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:47.16 | Corydon76-dig | ice_croft: use ccs |
| 14:48.09 | *** join/#asterisk _baconbuttie_ (n=bob@host-77-101-47-146.static.telewest.net) |
| 14:48.27 | Corydon76-dig | Framing is at a lower level than channels. You have to have framing in order to have channels. CCS is what you want. |
| 14:48.48 | ice_croft | Corydon76-dig> it might be: set, pass, cross, of |
| 14:48.49 | ice_croft | Corydon76-dig> it might be: set, pass, cross, off |
| 14:48.55 | ice_croft | Corydon76-dig> what should i set? |
| 14:49.04 | ice_croft | Corydon76-dig> it's off now |
| 14:50.01 | coppice | Actually, not all E1s are framed, but all E1s used for telephony are |
| 14:50.22 | [TK]D-Fender | ice_croft: Ask your TELCO what they are using. |
| 14:50.32 | ice_croft | coppice> my phone operator says they give only unframed e1s |
| 14:50.49 | ice_croft | 8-О |
| 14:50.50 | coppice | then they are data E1s, and not telephony |
| 14:51.24 | Corydon76-dig | coppice: it's a waste of effort, he won't listen |
| 14:51.24 | coppice | however, you might be confusing framing and multi-framing |
| 14:51.37 | coppice | CCS is framed. CAS is multi-framed |
| 14:51.47 | X-Filez | [TK]D-Fender: thanks :) this work, but have small one problem, my sip conf is here -> http://pastebin.com/m2518a3ae |
| 14:52.02 | X-Filez | [TK]D-Fender wait 1 min, mobile call |
| 14:52.24 | ice_croft | coppice> ok, i just don't know technology well enough. |
| 14:52.32 | ice_croft | thanx everybody |
| 14:52.36 | Corydon76-dig | That part was obvious |
| 14:53.14 | *** part/#asterisk bkw_ (n=brian@adsl-64-149-47-251.dsl.tul2ok.sbcglobal.net) |
| 14:53.14 | ice_croft | Corydon76-dig> well, human can't know everything, u know |
| 14:53.19 | *** join/#asterisk bkw_ (n=brian@adsl-64-149-47-251.dsl.tul2ok.sbcglobal.net) |
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| 14:53.21 | X-Filez | [TK]D-Fender: Problem is : I call from SIP/11 to SIP/12, i See SIP/12 LED is ON, but SIP/12 don't see SIP/11 use Line or not... LED is OFF |
| 14:54.17 | mosty | X-Filez, pastebin your sip.conf and extensions.conf |
| 14:54.36 | [TK]D-Fender | X-Filez: You didn't change them to "peer", and you may have to reboot the phones, and do make sure to restart * |
| 14:55.06 | X-Filez | mosty: sip.conf -> http://pastebin.com/m2518a3ae and extensions.conf -> http://pastebin.com/m1fc53863 |
| 14:55.35 | X-Filez | [TK]D-Fender: i restarted 2 snom and Asterisk :) i know :) |
| 14:58.02 | *** join/#asterisk BCS-Satori (n=somewher@75.148.21.113) |
| 14:58.05 | X-Filez | core show hints say : |
| 14:58.05 | X-Filez | 12@sip_default : SIP/12 State:Busy Watchers 2 and 11@sip_default : SIP/11 State:Idle Watchers 2 |
| 14:58.06 | [TK]D-Fender | X-Filez: type = friend <-- I said "type=peer" |
| 14:58.12 | X-Filez | ok |
| 14:58.16 | X-Filez | wait sec |
| 14:58.28 | [TK]D-Fender | X-Filez: and I suggested a "call-limit=99". |
| 14:59.24 | X-Filez | ok, wait.. restart snoms and asterisk |
| 15:01.03 | X-Filez | [TK]D-Fender: big thanks :) |
| 15:01.30 | [TK]D-Fender | X-Filez: All good now? |
| 15:01.35 | X-Filez | yes |
| 15:01.52 | [TK]D-Fender | X-Filez: Excellent. Now you can go come up with coll useful stuff to use DevState for. |
| 15:01.58 | [TK]D-Fender | cool* |
| 15:02.26 | X-Filez | :) |
| 15:03.19 | *** join/#asterisk admgecko (n=admgecko@adsl-87-102-68-32.karoo.KCOM.COM) |
| 15:03.33 | admgecko | hello boys and girls |
| 15:04.05 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 15:05.34 | R1ck | is FXS for ISDN phones or for Analog phones? |
| 15:05.47 | R1ck | and is a PRI card compatible with ISDN-2 ? |
| 15:06.02 | [TK]D-Fender | R1ck: Analog |
| 15:06.10 | nestAr | NI2? Yea |
| 15:06.22 | nestAr | ISDN 2? not sure sure |
| 15:06.26 | nestAr | not sure* |
| 15:06.27 | nestAr | lol |
| 15:06.39 | R1ck | and FXO is for isdn phones? |
| 15:06.50 | nestAr | no |
| 15:06.58 | nestAr | FXO is for connecting to a POTS line |
| 15:07.05 | nestAr | FXS is for connecting to a POTS phone |
| 15:07.31 | R1ck | ahh, I see |
| 15:07.43 | R1ck | so both are analog? |
| 15:07.49 | nestAr | yes sir |
| 15:07.54 | R1ck | allright, thanks |
| 15:10.42 | R1ck | anyone know how old a Cisco 7750 possible is? |
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| 15:14.22 | admgecko | has anyone had any experince with an avaya 4610sw sip phone config? anyone know how to program the softkeys with asterisk? |
| 15:14.41 | otaku42 | i'm looking for a web-based interface for managing conference rooms (meetme application), for asterisk 1.4.x. any suggestions for that? |
| 15:15.01 | *** join/#asterisk _pepo_ (n=c9eea608@190.10.187.20) |
| 15:15.10 | admgecko | doesnt meetme have a web-interface...it does in trixbox |
| 15:15.37 | admgecko | ? |
| 15:15.48 | otaku42 | admgecko: no idea, haven't used trixbox yet. |
| 15:15.51 | [TK]D-Fender | admgecko: the words "THIRD PARTY" come to mind..... |
| 15:16.05 | [TK]D-Fender | admgecko: Which is the direct translation of "TrixBox" |
| 15:16.24 | admgecko | sorry, just trying to help :*) |
| 15:16.40 | [TK]D-Fender | otaku42: Go look on the WIKI . |
| 15:16.42 | [TK]D-Fender | ~wikis |
| 15:16.42 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 15:19.12 | *** join/#asterisk af_ (n=getsmart@88-149-241-31.dynamic.ngi.it) |
| 15:19.40 | otaku42 | [TK]D-Fender: thx |
| 15:25.53 | admgecko | Fender: do you know anything about using avaya phones on asterisk? |
| 15:27.54 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 15:32.15 | [TK]D-Fender | admgecko: Nope, few people sue them. |
| 15:32.39 | admgecko | yeah, im just wondering how the hell i program the softkeys |
| 15:33.06 | admgecko | its a really nice phone, and ive done the config via tftp, loaded it with the sip firmware, and got it to log on |
| 15:33.24 | admgecko | it can make / recive calls, would just be nice to use the softkeys |
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| 16:00.43 | mikecx | does anyone have a working sla config? |
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| 16:01.40 | *** join/#asterisk russellb (i=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:01.40 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:01.45 | mhiku | where can i buy the converter of the fone line to the asterisk cpu? |
| 16:02.23 | philippel | russellb ping |
| 16:02.28 | russellb | pong |
| 16:02.53 | philippel | russellb concering #10690 |
| 16:03.06 | philippel | the #include 'fix' in 1.4 |
| 16:03.33 | philippel | I understand your comment and logic, but there are about 200,000 installations out there and they could use a little time for us to provide the mechanism to fix |
| 16:04.06 | philippel | this could be categorized much closer to a 'feature/bug' grey area then a pure bug fix and features are usually not introduced in new versions |
| 16:04.19 | philippel | comments? |
| 16:05.44 | *** part/#asterisk MindTheGap (n=MindTheG@201.80.194.113) |
| 16:06.35 | mhiku | what does paid asterisk vs the free one? |
| 16:06.58 | *** join/#asterisk callguy (n=callguy@pool-71-162-97-18.bstnma.east.verizon.net) |
| 16:07.01 | mhiku | features and other things? |
| 16:07.04 | Qwell | mhiku: You get support, and it has been tested (of course, all of that testing ends up being reflected back into the open source code) |
| 16:07.29 | mikecx | anyone with SLA working or a good alternative to SLA? |
| 16:08.18 | mhiku | support only? |
| 16:08.18 | *** join/#asterisk lmadsen (n=blitz[as@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:08.18 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 16:08.32 | Qwell | mhiku: warranty, etc |
| 16:08.51 | mhiku | i mean, can the free one like transfer calls to another fone, etc etc |
| 16:08.58 | Qwell | absolutely |
| 16:09.18 | mhiku | so i need to buy Digium AEX880E |
| 16:09.22 | Qwell | there are very very few things in ABE that are not in open source - and those, only due to licensing issues. |
| 16:09.25 | mhiku | to use the asterisk? |
| 16:09.34 | Qwell | you don't need any hardware at all to use asterisk |
| 16:09.51 | mhiku | how can i plug in the phone line to the cpu? |
| 16:10.03 | Qwell | well, for that you would need hardware :) |
| 16:10.04 | lmadsen | Other than a computer :) |
| 16:10.11 | philippel | russellb oops pasted teh wrong number: 0011543 |
| 16:10.17 | mhiku | so i need to buy Digium AEX880E |
| 16:10.19 | mhiku | ? |
| 16:10.29 | Qwell | mhiku: there are several models to choose from - it depends on what you need. |
| 16:10.33 | mhiku | its a mandatory hardware right? |
| 16:10.44 | lmadsen | Using an ITSP is a more economical way of getting asterisk on the pstn |
| 16:10.57 | Qwell | no, absolutely not. in fact, you don't even have to buy Digium (of course, people buying Digium means I get to have a job, but...) |
| 16:11.06 | lmadsen | If you're just testing |
| 16:11.27 | mhiku | how to test without buying Digium AEX880E |
| 16:11.34 | mhiku | or other models? |
| 16:11.39 | mhiku | or its mandatory? |
| 16:11.55 | mocker | heh. |
| 16:12.00 | Qwell | mhiku: With an ITSP, or just using softphones, if you don't need to make calls to the PSTN |
| 16:12.02 | Qwell | ~itsp |
| 16:12.26 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 16:12.26 | Qwell | ~pstn |
| 16:12.27 | jbot | hmm... pstn is Public Switched Telephone Network, or "please stop the nonsense" |
| 16:12.27 | mocker | ~thebook |
| 16:12.27 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 16:12.27 | Qwell | stupid bot |
| 16:12.27 | lmadsen | Maybe no one pays attention.. Maybe it's me :) |
| 16:13.16 | mhiku | okay where can i buy itsp |
| 16:13.26 | mhiku | or pstn? |
| 16:14.40 | mhiku | T_T |
| 16:14.55 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:16.36 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 16:17.16 | mhiku | maybe the hardware is manatory T_T |
| 16:17.47 | krdian_ | hehe PSTN :) |
| 16:18.08 | krdian_ | Please Stop ... ;) |
| 16:18.16 | mhiku | i dont know where to buy PSTN |
| 16:18.54 | philippel | russellb you must have stepped away, I need to brb for 20 minutes, but look forward to hearing back on the #include possibility of defferring that or providing some sort of backward compatibility option |
| 16:18.54 | mhiku | T_T |
| 16:19.04 | mhiku | can i use a modem? |
| 16:19.31 | mhiku | can i use an oldschool telephone modem? |
| 16:21.26 | mhiku | what is Asterisk Appliance |
| 16:21.31 | *** join/#asterisk BadBru (n=hara@86.121.23.144) |
| 16:21.37 | mhiku | can i use it in the philippines? |
| 16:22.04 | BadBru | some1 know how i can run a call in background ? useful for an ivr menu... |
| 16:22.42 | BadBru | if call is not in background, asterisk doesn't listen my keys.. i want during a call when i press 1 to execute a cmd |
| 16:22.50 | mocker | BadBru: Background() ? |
| 16:23.14 | mort_gib | mhiku: Not sure about the Philippines, but you can buy Digium or other Asterisk compatible hardware loads of places |
| 16:23.32 | mhiku | i mean what is the trunk lines? |
| 16:23.35 | BadBru | yea.. but i don't think it works like Background(ChanSpy(SIP/300|q)) |
| 16:23.50 | mhiku | 4 trunk lines using the phone provider here in philippines? |
| 16:23.58 | mhiku | is that possible? |
| 16:24.23 | BadBru | mocker, what do you think.. an alternative of Background(ChanSpy(SIP/300|q)) that sure won't work |
| 16:24.34 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 16:24.39 | BadBru | maybe Background(beep) works |
| 16:24.46 | BadBru | but Background(ChanSpy(SIP/300|q)) doesn't |
| 16:25.10 | BadBru | who knows ? |
| 16:25.13 | mocker | BadBru: I'd use background to play a message and then have the number pressed launch the application. |
| 16:25.33 | mort_gib | mhiku: I don't see why not, but you would need something like TDM04B http://www.voipon.co.uk/digium-tdm04b-4-fxo-p-77.html |
| 16:25.36 | BadBru | yes.. but acutally is not what i need |
| 16:25.56 | BadBru | i want in background to be a call.. a call wich request ChanSpy(SIP/300|q) |
| 16:26.07 | BadBru | and if i press 1 key.. |
| 16:26.26 | BadBru | call will do hangup(SIP/300-081af122) |
| 16:26.31 | mort_gib | mhiku: ISDN might be another option though. Depending on your local telco! |
| 16:27.29 | mhiku | where to buy isdn?? |
| 16:27.57 | mort_gib | mhiku: your local telco, ISDN is another type of PSTN (phone lines) |
| 16:28.03 | BadBru | mocker what do u think ? |
| 16:28.07 | mhiku | im just new here, in voip , the company uses pabx platforms provided by the local telco |
| 16:28.20 | mort_gib | mhiku: In my area they come out somewhat more affordable... |
| 16:28.36 | mhiku | is that an appliance? |
| 16:29.23 | mort_gib | mhiku: No, the appliance is a small PC with Asterisk loaded. They come in a few flavors, depending on how you connect to PSTN (I think) |
| 16:29.48 | mhiku | this asterisk software have a like msn messenger that can be installed in the internet and accept calls? |
| 16:29.58 | mhiku | like skype? |
| 16:30.48 | [TK]D-Fender | ~softphone |
| 16:30.49 | jbot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam |
| 16:30.51 | [TK]D-Fender | ^^^ |
| 16:31.05 | mort_gib | mhiku: NO, Asterisk is a server based phone system solution. MSN messenger happens to support SIP (as a client) the same protocol but on the server end |
| 16:31.25 | BadBru | [TK]D-Fender can u help me with one thing: how menu asterisk prompts during call |
| 16:31.35 | [TK]D-Fender | BadBru: huh? |
| 16:31.51 | mhiku | so what client can i use in asterisk? |
| 16:32.05 | BadBru | during i make chanspy.. i want if i press 1 key to execute another command |
| 16:32.10 | [TK]D-Fender | mhiku: LOOK UP |
| 16:32.20 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:32.23 | [TK]D-Fender | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam |
| 16:32.32 | mhiku | which do you recommend? |
| 16:32.38 | BadBru | actually i did it, but need to finish call, then asterisk wait for new cmd.. 1, 2 , 3 |
| 16:32.45 | *** join/#asterisk ManxPower (n=manxpowe@50.sub-70-221-238.myvzw.com) |
| 16:32.56 | BadBru | i want to do it during a call |
| 16:32.58 | mort_gib | mhiku: In my "test bed" I use 1. Analogue phone, 1. Snom 300, 1. Analogue phone connected via a Zyxel 2002l and Linphone (softphone) |
| 16:33.09 | BadBru | how u can place a call in background.. |
| 16:33.32 | outtolunc | i recommend the ~book <G> |
| 16:33.42 | [TK]D-Fender | BadBru: During a call? I'm not getting what you want to have happen here... |
| 16:34.02 | mort_gib | [TK]D-Fender: If the guy need a bit pointing in the right direction(!) then fine.... Don't flame him! |
| 16:34.07 | BadBru | yes.. During a call.. if press 1 .. then hangup |
| 16:34.20 | mhiku | so i need Zyxel 2002l |
| 16:34.22 | BadBru | During a normal call.. |
| 16:34.31 | [TK]D-Fender | mort_gib: Nowhere did I flame. I gave direct real info and links to even MORE useful and precise info. |
| 16:34.40 | mhiku | or other devices for the wire in the phone to connect to the server |
| 16:34.40 | mhiku | ? |
| 16:34.52 | mhiku | can i use a generic modem? |
| 16:35.03 | [TK]D-Fender | BadBru: So while 2 people are talking you want the user to hang up on DTMF? |
| 16:35.05 | mort_gib | mhiku: Not really, * is just very flexible, with the right hardware you can get anything to work |
| 16:35.07 | [TK]D-Fender | mhiku: No. |
| 16:35.21 | BadBru | tes right |
| 16:35.32 | BadBru | [TK]D-Fender yes, righ |
| 16:35.35 | mort_gib | [TD]D-Fender: Like you did to me when I started here... |
| 16:35.42 | mhiku | with the right hardware?? why cant i use a generic modem |
| 16:35.45 | mhiku | then? |
| 16:36.00 | mort_gib | mhiku: :-) Not the right hardware... |
| 16:36.05 | mhiku | lol ok |
| 16:36.17 | [TK]D-Fender | BadBru: "show application dial" <- read the instructions on what you can do while in a call. |
| 16:36.52 | [TK]D-Fender | mhiku: BEcause there is no zaptel driver for it. You need to buy compatible hardware. This is not a LINUX issue, it is a Zaptel driver issue |
| 16:37.36 | mhiku | ok which products existing can i choose and compare? |
| 16:37.51 | mhiku | the hardware in * is too pricy |
| 16:38.10 | mhiku | i need to test * first so i need a cheap hardware first |
| 16:38.13 | philippel | russellb ping - you back? |
| 16:38.22 | mort_gib | mhiku: This type of hardware IS pricy |
| 16:39.13 | ManxPower | mhiku: Before Digium existed a T-1 port for most devices was about $4,000-$8,000 |
| 16:39.13 | mort_gib | mhiku: you can get started for some ВЈ200, but difficult for less, unless there is a bundle I'm unaware of |
| 16:39.21 | BadBru | [TK]D-Fender, h - Allow the called party to hang up by sending the '*' DTMF digit. |
| 16:39.27 | BadBru | but is not what i need |
| 16:39.42 | mort_gib | outtolunch: we have all been there! |
| 16:39.50 | mhiku | is the pcie hardware a better start? |
| 16:40.11 | mhiku | or il buy the appliance? |
| 16:40.45 | mort_gib | outtolunch:Sure... But some things are difficult to read up, some stuff you have to feel on your own body |
| 16:40.48 | BadBru | [TK]D-Fender when 2 people talks, and i spy on the channel, when i press 1 i want to hangup their call |
| 16:41.11 | BadBru | so i don't want my call drop, i want call i spy to drop |
| 16:41.24 | mort_gib | mhiku: I have no clue! I would ALWAYS build from source |
| 16:41.48 | mhiku | aww, maybe i need to buy the appliance :( |
| 16:41.52 | mocker | BadBru: I think what you are wanting will need some custom crap done. |
| 16:42.04 | mocker | BadBru: Maybe a hacked up meetme room. |
| 16:42.49 | BadBru | i listen my operators, and if say something wrong, i want to drop their call |
| 16:42.53 | BadBru | by pressing 1 |
| 16:43.02 | mort_gib | outtolunch: Well sort of, I logged on here to ask questions that was not answered in the book, and found a strange condesending tone in here |
| 16:43.12 | BadBru | i can drop call from cli, but will be most confortable to do it by phone |
| 16:43.26 | mhiku | does * uses web based configuration? |
| 16:43.30 | mort_gib | Phillipines are also quite different from my home turf |
| 16:43.54 | mort_gib | mhiku: Your better off using vi and -sigh- the book! |
| 16:43.57 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 16:44.03 | outtolunc | mort_gib, that does happen from time to time here |
| 16:44.30 | mhiku | :D |
| 16:45.31 | mort_gib | outtolunch: Which is why I gladly help out someone the superior beings in here can't even ignore... |
| 16:45.33 | BadBru | some1 knows and can help me ? |
| 16:45.35 | outtolunc | badbru, i must have missed the beginning of this.. why do you need/want chanspy? |
| 16:45.46 | variable_office | what is a good residential ata that you all find to be pretty reliable and quality? |
| 16:45.57 | [TK]D-Fender | BadBru: Sorry that doesn't exist. Chanspy doesn't have those capabilities. You'll have to make an external tool |
| 16:46.25 | outtolunc | mort_gib, in this case the person was given suggestions, and that person didn't even lookup that info before going round and round |
| 16:46.26 | BadBru | outtolunc, i have chanspy, i listen my operatos, but during my listen if i press 1 i want to drop their call |
| 16:46.47 | outtolunc | ah |
| 16:46.52 | [TK]D-Fender | mhiku: Get a Linksys SPA-3102 then. That will give you 1 FXS port, and 1 FXO port for about $70USD. |
| 16:47.05 | outtolunc | that should be fairly easy to do |
| 16:47.50 | BadBru | how ? |
| 16:47.50 | outtolunc | note it would be a custom patch |
| 16:48.38 | variable_office | is the pap2t ok? |
| 16:49.03 | [TK]D-Fender | variable_office: I'd prefer spending a few more bucks on the SPA-2102 over it. Can act as a router, has a bigger CPU, and T.38 support |
| 16:49.40 | variable_office | ah, ok; is that what you like to use [TK]D-Fender for consumer-grade stuff? |
| 16:49.58 | Qwell | the linksys is consumer-grade |
| 16:50.06 | Qwell | oh, misread |
| 16:50.23 | [TK]D-Fender | variable_office: I'm talking $10 here.... don't be a cheap-ass! |
| 16:50.39 | Qwell | I hear the 3102 is quite good |
| 16:50.48 | [TK]D-Fender | Qwell: s/good/acceptable/ |
| 16:50.58 | mocker | Woo, this Aspect powerpoint is actually pretty good. |
| 16:50.59 | variable_office | oh, no i am not saying i wouldnt spend an extra $10, i am just looking for the best in that area, or at least what common experience says is best |
| 16:51.02 | Qwell | [TK]D-Fender: compared to a pap2 or gs? :) |
| 16:51.08 | Qwell | it's all relative |
| 16:51.08 | variable_office | i am having some problems with people using pap2t |
| 16:51.11 | [TK]D-Fender | Qwell: it is a very reasonable product that is remarkably flexible for its cost |
| 16:51.20 | mort_gib | outtolunch: If you don't want to answer, just ignore him -Right?? |
| 16:51.35 | outtolunc | i'm talk to him in priv |
| 16:51.37 | mocker | variable_office: Mediatrix is pretty nice. |
| 16:51.43 | [TK]D-Fender | Qwell: 2102 = PAP2 (for FXS only). Not fair to compare PAP2 & 3102 |
| 16:51.47 | coppice | [TK]D-Fender or remarkably expensive for its BOM :-) |
| 16:51.48 | Qwell | sure |
| 16:51.52 | outtolunc | so no, i'm just ignoring YOU now |
| 16:52.38 | coppice | mediatrix are good for really wacky bugs :-) |
| 16:52.42 | mort_gib | outtolunc: Ok... |
| 16:52.57 | mocker | coppice: And for lots of $$$ |
| 16:53.23 | coppice | well, you gotta pay extra for really creative bugs |
| 16:53.28 | *** join/#asterisk badcfe (i=christia@alltid.dritings.no) |
| 16:53.51 | badcfe | hello. how do i configure a sip peer to accept any invite as long as it comes from a specific ip address? |
| 16:54.10 | badcfe | host and permit doesnt do it, the peer is not recognised by that .. |
| 16:54.26 | [TK]D-Fender | badcfe: You can't in mainline. There is a brach out there dubbed "kill-the-user" which can allow that. |
| 16:54.31 | mhiku | there you go, a cheap one a linksys |
| 16:54.40 | [TK]D-Fender | badcfe: Yes, I'm serious. check the * Daily News site |
| 16:55.02 | badcfe | why "kill-the-user" ? |
| 16:55.12 | Qwell | badcfe: because users are bad |
| 16:55.15 | [TK]D-Fender | badcfe: "type=user|friend|peer" |
| 16:55.28 | [TK]D-Fender | badcfe: * 1.4 started trying to phase out "type=user" |
| 16:55.31 | [TK]D-Fender | (and friend) |
| 16:55.37 | coppice | its all part of the holy war on users |
| 16:55.37 | badcfe | i have type=peer |
| 16:55.45 | Qwell | to this day, peer/user confuses me... :p |
| 16:56.20 | russellb | it makes perfect sense in theory ... chan_sip just mangled it to the point where it doesn't make sense |
| 16:56.25 | badcfe | so the only way is to hood my stuff on the default context? |
| 16:56.27 | russellb | and it still makes perfect sense in chan_iax2 |
| 16:56.38 | mocker | Qwell: Glad it's not just me. :) |
| 16:56.52 | philippel | russellb ping - any comment on my above question, or can I call you offline? |
| 16:57.06 | badcfe | i _want_ things from this specific ip to go to my lovely context, how? |
| 16:57.11 | mhiku | is there a cheaper than linksys? |
| 16:57.16 | mhiku | below 70bucks? |
| 16:57.26 | mocker | ~cheap |
| 16:57.27 | jbot | somebody said cheap was a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 16:57.39 | russellb | philippel: I don't know what to say ... I feel like the upgrade process of any package that uses that type of configuration could easily ensure those files exist |
| 16:58.00 | russellb | at this point, I'm not willing to make any changes to what got added, as I feel it is the right thing to do |
| 16:58.14 | mhiku | lol, i want to test * first before i buy the pcie thing |
| 16:58.17 | Qwell | russellb: WELL, I do need to discuss one part of that change with you |
| 16:58.21 | Qwell | I'll come over |
| 16:58.24 | russellb | ok |
| 16:58.40 | mhiku | is linksys ok? |
| 16:59.02 | badcfe | the INVITE has From: "anonymous" <something, .. could i use that to trigger assosiation with a particular peer in order to go into my lovaly context? |
| 16:59.14 | philippel | russellb there's a pretty large installed base out there, changes like that are usually done on cosideration of that - or at least providing a backward compatibility - it strattles a fine line of feature vs. bug - I'm not saying we won't do anything to address it, but it is going to initially cause a lot of pain to your installed base |
| 16:59.36 | mhiku | can i use edimax voip in * ? |
| 17:00.01 | philippel | russellb it's the type of move that creates a lot negative feedback to a project when a change is done like that without a reasonable transition period, same reason you don't insert new features in a stable release |
| 17:00.06 | [TK]D-Fender | mhiku: No. |
| 17:00.23 | mhiku | why? |
| 17:00.25 | [TK]D-Fender | ~skype |
| 17:00.25 | jbot | [~skype] Skype is a free VoIP software and service using a closed client and propritary protocol. Only commercial channel drivers exist, all of which are poor implementations of convoluted hacks. In general, forget about using Skype with Asterisk... |
| 17:00.38 | mhiku | awww |
| 17:00.58 | [TK]D-Fender | mhiku: * doesn't SPEAK Skype. Stop trying to find a cheap-ass way out. You will fail, and suffer in the process. |
| 17:01.17 | [TK]D-Fender | mhiku: Wake up time. telecom isn't FREE, and $70 is CHEAP. thats $35/port! |
| 17:02.00 | coppice | $35 for port, and $35 for starboard |
| 17:02.24 | [TK]D-Fender | BBIAB, lunch.... |
| 17:02.26 | mhiku | i mean, i want to test it first which i can afford in my monthly salary before using the real thing, if i recommend it in the company and i dont even know how to use it, they will fire me, please understand |
| 17:02.27 | Qwell | philippel: the part of this that people don't seem to understand, is that this fixes a very major problem with broken configs. If there is a problem like that, we simply cannot know whether it was done on purpose, or if it was just a typo |
| 17:02.43 | *** part/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 17:03.26 | philippel | Qwell - I'm not disagreeing with your direction, I requesting you provide some transition time so you don't break thousands of your users because they did not have time to react to the change |
| 17:04.07 | mort_gib | mhiku: I DO understand you, but it's REALLY hard to learn this without spending some $ |
| 17:04.07 | philippel | Qwell - I am making mods as wel speak to address it and will be putting somethign out - but timing wise, it is going to break people |
| 17:04.39 | russellb | Qwell: i suppose for the #include case, we could do 1 release that issues like 10 LOG_ERROR messages, heh |
| 17:05.07 | Qwell | russellb: it was all or none, iirc |
| 17:05.16 | russellb | Qwell: oh .. |
| 17:05.18 | Qwell | the #include case doesn't return differently than the broken context case |
| 17:05.22 | Qwell | (again - iirc) |
| 17:05.25 | mhiku | so is the cheap pcie from * then? |
| 17:05.44 | russellb | well, i will certainly include some notes about the change in the next release announcement |
| 17:06.01 | philippel | russellb what's the timing for that release? |
| 17:06.06 | Qwell | russellb: that was something oej would like to see as well |
| 17:06.17 | mort_gib | mhiku: TDM01B - 1 FXO |
| 17:06.19 | russellb | philippel: most likely this week, actually |
| 17:06.47 | philippel | russellb that's going to break people most likely then - any way to delay it to the next release and do the log warnings? |
| 17:06.47 | mhiku | is there a possibility that the hardware i will buy isnt a pcie???? the server have 1 pcie and been used by the videocard |
| 17:07.22 | Qwell | mhiku: Do you need pcie or regular pci? |
| 17:07.25 | philippel | russellb I'll work on getting an update to our 1.4 supporting versions today if possible to start the ball rolling on our side, but if you do it this week - it's going to cause pain |
| 17:08.02 | mhiku | woah, a 100 dollars hardware, much cheaper than the other hardware ranging from 800 dollars above |
| 17:08.14 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 17:08.24 | flujan | hi all. |
| 17:08.59 | russellb | philippel: alright, let me look to see how hard it would be to issue warnings just for the #include case |
| 17:10.19 | philippel | russellb that woud be great |
| 17:12.14 | *** part/#asterisk alrs (i=non-knav@pozug.com) |
| 17:12.53 | russellb | philippel: so you're a FreePBX developer? |
| 17:13.16 | philippel | russellb I run the project |
| 17:13.21 | russellb | ah, cool |
| 17:13.26 | russellb | sorry, hard to keep up ... |
| 17:14.00 | philippel | russellb np, and btw as an editorial comment, we are completely independent and have nothing to do with trixbox/fonaility - they just happen to 'borrow' our stuff |
| 17:14.13 | [TK]D-Fender | russellb: Take the "blue" pill Neo! ;) |
| 17:14.20 | russellb | philippel: understood |
| 17:14.21 | Qwell | "borrow" is a serious understatement |
| 17:14.25 | russellb | yes, it is |
| 17:14.33 | russellb | it is quite unfortunate what they have done ... |
| 17:14.42 | Qwell | I once asked Rob if he ever saw any of the money that Andrew got for trixbox... not a dime, apparently |
| 17:14.44 | Qwell | That's sad |
| 17:14.46 | russellb | but i should probably stop there :) |
| 17:14.52 | philippel | yes - I had an interesting conversation with Mark on that at Astricon |
| 17:15.35 | russellb | philippel: do they contribute anything? |
| 17:15.43 | russellb | because they certainly have never contributed a single thing to asterisk |
| 17:15.52 | philippel | we should probably all - I try to ignore any engative people do and do everything I can to promote the success of Asterisk in which ever flavor as it is critical to all of our success and the sandbox is big enough for everyone |
| 17:15.55 | philippel | russellb no |
| 17:16.01 | russellb | oh good lord |
| 17:16.24 | russellb | that's terrible |
| 17:16.37 | philippel | russellb bu I need to stop here, anything further shoudl be discussed privately |
| 17:16.45 | russellb | :-X |
| 17:17.32 | [TK]D-Fender | x > x |
| 17:18.31 | russellb | so how about that professional sports team in that sporting event? |
| 17:18.37 | philippel | russellb on a separate note, we (FreePBX) are looking into doing a training in the Spring, probably March/April and likely not too far from your neck of the woods (as in SouthEast coast) - any chance of someone interested in making a guest appearance? |
| 17:19.09 | russellb | philippel: it's possible, but I can't answer that question. You would have to contact our Marketing department |
| 17:19.25 | philippel | I'll talk to Jim |
| 17:19.33 | russellb | sounds good |
| 17:20.41 | mocker | Damn, looks like I missed a good conversation. |
| 17:22.23 | russellb | philippel: looks like this change will work .... I can have it issue a ton of errors for a #include error, but not fail config loading |
| 17:22.32 | russellb | and still preserve the other checks we added |
| 17:22.50 | *** join/#asterisk demiv (n=demiv@134.42.128.66.PPPoECali.dynamic.telesat.net.co) |
| 17:22.51 | *** join/#asterisk vetetix (n=vetetix@eclip3.ec-lille.fr) |
| 17:29.21 | *** join/#asterisk adorah (n=Michael@87.69.57.246.cable.012.net.il) |
| 17:29.53 | *** join/#asterisk vetetix (n=vetetix@eclip3.ec-lille.fr) |
| 17:30.33 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:31.30 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 17:34.53 | russellb | excuse the paste bomb, but i'm lazy... |
| 17:35.12 | [TK]D-Fender | russellb: 11 steps to go! |
| 17:35.27 | russellb | philippel: I have just made a change to 1.4, revision 93000, that issues a huge error message if a #include is done of a file that does not exist |
| 17:35.29 | russellb | [Dec 14 11:31:56] ERROR[19081]: config.c:748 process_text_line: ********************************************************* |
| 17:35.29 | russellb | [Dec 14 11:31:56] ERROR[19081]: config.c:749 process_text_line: *********** YOU SHOULD REALLY READ THIS ERROR *********** |
| 17:35.29 | russellb | [Dec 14 11:31:56] ERROR[19081]: config.c:750 process_text_line: Future versions of Asterisk will treat a #include of a file that does not exist as an error, and will fail to load that configuration file. Please ensure that the file 'foo.conf' exists, even if it is empty. |
| 17:35.30 | russellb | [Dec 14 11:31:56] ERROR[19081]: config.c:754 process_text_line: *********** YOU SHOULD REALLY READ THIS ERROR *********** |
| 17:35.33 | russellb | [Dec 14 11:31:56] ERROR[19081]: config.c:755 process_text_line: ********************************************************* |
| 17:35.40 | russellb | That's what it looks like :) |
| 17:36.36 | mocker | russellb: Should standardize on figlet generated error messages |
| 17:36.58 | russellb | mocker: i don't know what you're talking about :) |
| 17:37.18 | coppice | what a waste of time. you could put "FREE BEER" there, and still nobody will read it :-) |
| 17:38.32 | mocker | russellb: You've never used figlet before?? |
| 17:38.32 | philippel | russellb oops - was in a pm channel, that sounds great - I'm also almost there with my change that I can push out which will create the files that we expect might be there so we are ready for when the switch is pulled all the way |
| 17:38.32 | mocker | russellb: figlet.org, it's a CLI utility for text based banners. :) |
| 17:38.32 | russellb | philippel: sounds good |
| 17:38.34 | philippel | russellb btw - while you are at fixing #include, did you do the same with #exec ? |
| 17:38.44 | russellb | mocker: oh, lol ... yeah, we should do that :) |
| 17:38.59 | philippel | probably even more of a security issue for that one8) |
| 17:39.00 | russellb | mocker: no, i didn't touch exec ... |
| 17:39.07 | Qwell | It's handled by the same code, actually |
| 17:40.03 | Qwell | #exec just creates a file to be #include'd |
| 17:40.17 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 17:40.35 | *** join/#asterisk bkruse_home (n=kruz@76.73.154.120) |
| 17:40.35 | *** mode/#asterisk [+o bkruse_home] by ChanServ |
| 17:40.42 | *** join/#asterisk Schumie (i=SteveWri@cpc1-rdng2-0-0-cust441.winn.cable.ntl.com) |
| 17:41.02 | Zeeek | VOIP Users COnference live IRC #voip-users-conference http://voipUsersConference.org |
| 17:41.59 | Zeeek | come on over and say your bit |
| 17:42.51 | *** join/#asterisk klyrelion (n=Kevin@icsnet.demon.co.uk) |
| 17:46.28 | *** join/#asterisk Greek-Boy (n=email@41.221.58.5) |
| 17:46.41 | *** part/#asterisk klyrelion (n=Kevin@icsnet.demon.co.uk) |
| 17:47.49 | Zeeek | http://news.zdnet.com/2422-13569_22-153763.html |
| 17:48.41 | twisted | haha |
| 17:48.45 | twisted | SMDI is so hacked in 1.4.15 |
| 17:49.20 | russellb | huh? |
| 17:49.25 | twisted | there are 4 message types. we should handle all 4. |
| 17:49.47 | twisted | also, we don't need to discard the message desk info. perhaps we should have a variable set for all of this on the channels |
| 17:50.34 | twisted | i've written a patch to handle all 4, and write them into _SMDI_VM_TYPE var |
| 17:50.52 | twisted | (this is a chan_zap thing) |
| 17:51.22 | twisted | but another thing is we should be able to deal with channel and port offsets, which we don't. might ave to patch that too |
| 17:51.34 | russellb | do you have the SMDI spec? |
| 17:51.36 | twisted | if i get permission, i'll submit the patches... otherwise, i'm just giving a heads up :) |
| 17:51.45 | twisted | yes |
| 17:51.55 | russellb | could you send the spec over? |
| 17:51.56 | twisted | i have the SMDI specs as of version 5.0 |
| 17:51.58 | bkruse_home | twisted: bowling? |
| 17:52.09 | russellb | interestingly enough, i'm making some major mods to SMDI handling right now ... |
| 17:52.17 | twisted | unfortunately, it's a pdf with confidential data written into it |
| 17:52.24 | russellb | argh |
| 17:52.26 | russellb | oh well |
| 17:52.29 | twisted | BUT |
| 17:52.33 | twisted | i can transcribe the spec parts out |
| 17:52.35 | twisted | :) |
| 17:54.21 | russellb | actually, here is some of it, heh |
| 17:54.23 | russellb | http://lists.digium.com/pipermail/asterisk-dev/2003-June/000884.html |
| 17:56.59 | X-Filez | [TK]D-Fender : hey, you there ? |
| 17:57.15 | [TK]D-Fender | X-Filez: yes |
| 17:58.42 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 17:58.51 | *** join/#asterisk ManxPower (n=manxpowe@226.sub-75-201-216.myvzw.com) |
| 17:59.36 | X-Filez | [TK]D-Fender : i have isdn line, and isdn card sweex MO128, you know, what me need use for in asterisk ? |
| 18:00.31 | _x86_ | fine, dont wave to me... |
| 18:00.36 | _x86_ | ;) |
| 18:00.37 | [TK]D-Fender | X-Filez: I don't see any reference of this card being compatible with * |
| 18:00.40 | *** join/#asterisk dennisonicc (n=dennis@cpc1-seve11-0-0-cust650.popl.cable.ntl.com) |
| 18:00.41 | [TK]D-Fender | ManxPower: |
| 18:00.44 | [TK]D-Fender | ManxPower: 'lo |
| 18:01.30 | dennisonicc | hi |
| 18:01.35 | X-Filez | :( |
| 18:07.42 | X-Filez | [TK]D-Fender: lspci say : Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] |
| 18:08.43 | coppice | the great majority of ISDN cards use those Cologne chips |
| 18:10.10 | X-Filez | what use : 1) bristuff or 2) zaphfc ? asterisk 1.4.15 |
| 18:10.13 | coppice | that is BRI ISDN. their PRI chips are far less popular |
| 18:10.35 | dennisonicc | is Asterisk the right aplication if I want to do something like: |
| 18:10.52 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:11.43 | bkruse_home | dennisonicc: phone a friend? yes |
| 18:11.56 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 18:12.01 | X-Filez | coppice: okey, Thanks, try install bri soft |
| 18:12.18 | dennisonicc | the person called 'A' calls to a local number whichone is forwarded to my PC my PC calls B in result A can talk with B |
| 18:12.39 | dennisonicc | A is an local phone B for example Mobile |
| 18:12.48 | dennisonicc | A is a local phone B for example Mobile |
| 18:12.53 | bkruse_home | yes |
| 18:12.55 | *** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 18:13.09 | bkruse_home | at the most basic sense, yes, or you can just have person A call person B |
| 18:13.25 | dennisonicc | yes through my pc |
| 18:14.09 | mikecx | does asterisk always go to the default context? |
| 18:14.29 | dennisonicc | is there any howto on the web for something like this? |
| 18:14.32 | bkruse_home | mikecx: if you do not specify otherwise, default is the fallback |
| 18:14.37 | bkruse_home | and if no extension, extension s |
| 18:14.49 | bkruse_home | dennisonicc: http://asteriskNOW.org |
| 18:14.50 | mikecx | bkruse_home: specify where? |
| 18:14.58 | bkruse_home | or just for the gui, http://asterisknow.org/install-related |
| 18:15.05 | bkruse_home | mikecx: you mean on an incoming call? |
| 18:15.06 | [TK]D-Fender | mikecx: there is no such thing as a "default" context. |
| 18:15.08 | mocker | s,1,Playback(hey-you-arent-supposed-to-be-here) |
| 18:15.44 | [TK]D-Fender | mikecx: Every context name and usage should be explicit. |
| 18:16.06 | mikecx | [TK]D-Fender: i guess my question is where is that first set but I realized it's in zapata.conf |
| 18:16.06 | bkruse_home | [TK]D-Fender: agreed |
| 18:16.22 | bkruse_home | mikecx: you can put context=anything in your zapata.conf for individual channels |
| 18:16.29 | bkruse_home | BUT, if none is specified, it goes to default |
| 18:16.34 | *** join/#asterisk vetetix (n=vetetix@eclip3.ec-lille.fr) |
| 18:16.44 | twisted | bkruse: when? |
| 18:16.50 | [TK]D-Fender | bkruse_home: Thats bad... we should remove that fall-back. |
| 18:16.57 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 18:17.02 | bkruse_home | [TK]D-Fender: agreed |
| 18:17.12 | dennisonicc | oh thanks but I dont want to change my distro I just want to experiment. In future maybe... |
| 18:17.38 | flujan | guys, I have a mono 16 bit 8000Khz wav file... asterisk is not playing it back... |
| 18:17.39 | bkruse_home | twisted: go put a tdm400p in a machine, setup for any number of fxo's or fxs' (any combination rather) |
| 18:17.49 | flujan | I first, put this file to be a musiconhold.... |
| 18:17.55 | twisted | bkruse: i meant bowling... |
| 18:17.59 | bkruse_home | twisted: oh |
| 18:18.03 | bkruse_home | twisted: lol! |
| 18:18.07 | flujan | asterisk is not playing it if I play the file on a windows machine it works... |
| 18:18.10 | flujan | any ideas? |
| 18:18.18 | bkruse_home | twisted: saturday? when are you in town? we will go with russellb and some peeps |
| 18:18.28 | twisted | saturday sounds good |
| 18:18.31 | bkruse_home | [TK]D-Fender: that whole failback thing is being talked about now |
| 18:18.36 | twisted | have russellb or someone call me |
| 18:18.51 | bkruse_home | kk |
| 18:20.15 | [TK]D-Fender | bkruse_home: the "kill-the-user" branch is some more great stuff. We need to undo the "filler" and make * explicit and very functional. |
| 18:23.44 | mikecx | [TK]D-Fender: mind helping me figure out why it's giving me a sent to invalid extension error? |
| 18:23.58 | [TK]D-Fender | mikecx: PASTEBIN is your friend...... |
| 18:23.59 | [TK]D-Fender | ~pb |
| 18:24.00 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:24.04 | mikecx | which files? |
| 18:24.11 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 18:24.33 | [TK]D-Fender | mikecx: Full CLI output of yrou call at verbose 10, and if SIP, then SIP DEBUG enabled, etc. |
| 18:24.43 | [TK]D-Fender | mikecx: And of course your dialplan. |
| 18:25.41 | mikecx | where do I set verbose 10? |
| 18:25.50 | mikecx | nm |
| 18:27.51 | mocker | mikecx: All the cool people attach their sessions with -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvr |
| 18:27.54 | mocker | :P |
| 18:28.12 | mikecx | [TK]D-Fender: http://pastebin.com/d1c5fbd15 |
| 18:28.50 | *** join/#asterisk coolfreecode (n=jimmy@190.43.25.29) |
| 18:29.07 | [TK]D-Fender | mikecx: Starting Zap/5-1 at from-pstn,s,1 failed so falling back to exten 's' <-- I don't see a context named [from-pstn] in your dialplan, do you? |
| 18:29.11 | coolfreecode | hey guys, have a way to replace the dial tone with play some sound file, when user pick up phone for dial? |
| 18:29.24 | mikecx | [TK]D-Fender: i thought I changed all of that in my zapata.conf |
| 18:29.27 | *** join/#asterisk BCS-Satori (n=somewher@75.148.21.113) |
| 18:29.28 | [TK]D-Fender | mikecx: Because thats where your zapata channel is SENDING your calls |
| 18:29.33 | *** join/#asterisk BudgetDedicated (n=BudgetDe@s5593c2e9.adsl.wanadoo.nl) |
| 18:29.46 | [TK]D-Fender | mikecx: try again... |
| 18:29.59 | [TK]D-Fender | coolfreecode: What kind of phone? |
| 18:30.08 | bkruse_home | mocker: mine is -vvvvvvvvvvvvvvvvgcT |
| 18:30.13 | BCS-Satori | Question about followme, is there a way to pass a ringing tone during a followme instead of playing musiconhold? |
| 18:30.14 | bkruse_home | gota get that core! |
| 18:30.15 | coolfreecode | ip |
| 18:30.32 | mocker | bkruse: hah, nice. |
| 18:30.38 | mikecx | [TK]D-Fender: missed a comment line, thanks. |
| 18:31.04 | *** join/#asterisk jhb (n=joerg@81-5-139-2.dsl.eclipse.net.uk) |
| 18:31.30 | [TK]D-Fender | bkruse_home: Oh yeah, on redundant : remove follow-ma as an APP. This should have been left to pure dialplan... |
| 18:31.46 | bkruse_home | [TK]D-Fender: also agreed |
| 18:32.05 | jhb | hi *. In my agi script I send the caller to a meetme conf. I would like to do a xml-rpc call after the caller has hunp up, leaving the conference. Is there a way to do it from within the same agi script? |
| 18:32.24 | jhb | s/hunp/hung/ |
| 18:32.27 | BudgetDedicated | sorry a bit offtopic maybe : I have an asterisk server running but on a windows PC on a homenetwork at a friends house I cound not get a good connection to it with any of the sip softphones I tried. Stun seems to work for me but over there it fails. what is a good method te create somekind of tunnel to my asterisk server to get it to work? Any thoughts ? |
| 18:33.01 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 18:33.02 | *** join/#asterisk hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net) |
| 18:33.15 | [TK]D-Fender | BudgetDedicated: Perfectly OT , read this now : |
| 18:33.17 | [TK]D-Fender | ~sipnat |
| 18:33.17 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:33.18 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 18:33.21 | coolfreecode | ip phone |
| 18:33.40 | [TK]D-Fender | coolfreecode: then you'll have to ready your phone's manual to see if you can send it to an exten immediately upon pickup |
| 18:33.54 | BudgetDedicated | thank you! |
| 18:34.20 | [TK]D-Fender | coolfreecode: if you can then you can have it land on an IVR where you can do "whatever" |
| 18:35.26 | coolfreecode | to analogic phones ?? |
| 18:35.38 | mikecx | well, incoming calls are working right(ish) now for sla |
| 18:36.45 | *** join/#asterisk vetetix (n=vetetix@eclip3.ec-lille.fr) |
| 18:38.22 | BCS-Satori | Is there a way to pass ringing once followme begins isntead of music? i had tried to put ,r after the seconds to ring but i still hear music |
| 18:39.49 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:41.29 | Greek-Boy | I want to replace our security departments old walkie talkies with wifi phones or IP PTT devices that will work with asterisk? Can someone recommend anything? My biggest concern is that I need all security guards to hear or ring when one users needs to speak. sort of like a permanent conference? |
| 18:41.50 | hmmhesays | ugh wifi phones? theres a nightmare for you |
| 18:41.51 | *** join/#asterisk clyrrad (n=darryl@CPE000802212b48-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 18:42.13 | hmmhesays | what is wrong with your t-ways? |
| 18:42.37 | mocker | (or are you trying to fix a problem that doesn't exist?) |
| 18:43.04 | hmmhesays | that kind of sounds like it |
| 18:43.05 | Greek-Boy | lol |
| 18:43.12 | Greek-Boy | the two way are getting old |
| 18:43.17 | Greek-Boy | and I want to monitor conversations... |
| 18:43.29 | hmmhesays | you can't do that with a two way? |
| 18:43.31 | mikecx | i can't get my linksys 942 to view the subscriptions |
| 18:43.52 | Greek-Boy | hmmhesays how would I record that? |
| 18:44.17 | [TK]D-Fender | mikecx: the WIKI has a guide for how to setup presence on Linksys phones (I don't know if the base supports it, or just the expansion module) |
| 18:45.19 | mikecx | [TK]D-Fender: know if any phones support it better? |
| 18:45.35 | hmmhesays | Greek-Boy: most radios have a 2.5 or 3.5mm headphone jack... there are recording devices out there.... |
| 18:45.53 | [TK]D-Fender | mikecx: Polycom & Aastra |
| 18:46.15 | Greek-Boy | anotehr problem two way radios is security and private comms |
| 18:47.36 | hmmhesays | a guarantee you that wifiphones and meetme conferences will be some kind of horrible nightmare for you |
| 18:48.27 | X-Filez | Ppls, i have PCI card Cologne Chip Designs GmbH ISDN network controller [HFC-PCI], but i don't understand, what stable drivers and use configure asterisk 1.4.15 ? please |
| 18:49.12 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 18:49.55 | [TK]D-Fender | X-Filez: www.asteriskguru.net <- they have guides for that chipset |
| 18:50.45 | clyrrad | Wonder if any of you have had this issue before? You call into voicemail, and you can hear Alyson saying the time stamp and promps, but as soon as the actual voicemail message starts to play, the call DIES, on the CLI you get a Maximum retries exceeded on transmission for seqno 102 (Critical Response) message. Anyone know whats happening here? |
| 18:51.42 | [TK]D-Fender | clyrrad: Usually thats packets getting lost. NAT issues are most common. |
| 18:52.27 | clyrrad | [TK]D-Fender: would that affect the acutal calls too? Becase calls are not getting dropped or one way audio.... just when you check voicemail |
| 18:53.45 | X-Filez | [TK]D-Fender: BRI drivers stable or have problems work in asterisk ? |
| 18:54.14 | X-Filez | [TK]D-Fender or zaphfc ? |
| 18:54.35 | [TK]D-Fender | X-Filez: Just TRY it ok? |
| 18:54.56 | *** part/#asterisk jmls (n=jmls@62.49.235.130) |
| 18:55.05 | clyrrad | [TK]D-Fender: would = wouldnt* |
| 18:55.09 | clyrrad | sorry bout that :p |
| 18:55.13 | [TK]D-Fender | clyrrad: That's an protocol error, not voicemail related... |
| 18:55.29 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 18:55.50 | clyrrad | Yea, thats what its confusing me why it only happens when checking voicemail and not making calls, both are going out over SIP |
| 18:56.18 | [TK]D-Fender | clyrrad: Any reinvites happening? |
| 18:57.15 | clyrrad | Just see SIP NOTIFY and getting response 603 Declined (no dialog) when I SIP debug the peer |
| 18:58.23 | clyrrad | The 603 Declined (no dialog) keeps happening over and over and over, its the only thing I see on this peer when doing sip debug |
| 18:59.23 | *** join/#asterisk becks` (n=flux_@218-173.5-85.cust.bluewin.ch) |
| 18:59.34 | [TK]D-Fender | clyrrad: I'm now guessing maybe a codec issue. end-to-end a voice call matches, but * can't translate solo based on preferences |
| 18:59.49 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 18:59.50 | [TK]D-Fender | clyrrad: Doing G.729 in PASSTHROUGH by any chance? |
| 18:59.52 | ZaVoid | burp |
| 19:00.01 | clyrrad | [TK]D-Fender: nope its all ULAW |
| 19:00.19 | becks` | hi, if my phone puts my voice back in my ear very silent (so i know how i sound like), how's that feature called? :) |
| 19:00.25 | [TK]D-Fender | clyrrad: try and debug a call start to finish and pastebin it along with all peer confis. Might be more telling |
| 19:00.28 | clyrrad | [TK]D-Fender: and the Comedian Mail prompts play, it just dies when the actual voicemail message starts to play |
| 19:00.29 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:00.57 | [TK]D-Fender | clyrrad: Verify the codec it was recorded in.... |
| 19:01.19 | clyrrad | [TK]D-Fender: how can I check that from the voicemail directory? |
| 19:01.24 | tzanger | yay aastra webinar |
| 19:01.31 | [TK]D-Fender | clyrrad: jsut ls it |
| 19:02.05 | [TK]D-Fender | becks`: "show application echo" |
| 19:02.15 | becks` | ok, thanks :) |
| 19:02.33 | tzanger | I get enough echo without an application giving it to me |
| 19:02.42 | clyrrad | [TK]D-Fender: its encoded as .WAV, .wav, and .gsm |
| 19:03.04 | [TK]D-Fender | clyrrad: Hrm... well next step, capture a full call for diagnosis |
| 19:03.16 | [TK]D-Fender | tzanger: now THATS a "feature" :p |
| 19:03.47 | *** join/#asterisk angom (n=angom@200.38.31.239.dsl.dyn.telnor.net) |
| 19:03.51 | clyrrad | [TK]D-Fender: by a full call you mean one to Voicemail that has the issue correct? Not an actual phone call as we know those are working... |
| 19:04.15 | [TK]D-Fender | clyrrad: Yes, it'd be nice to see the PROBLEM :p |
| 19:04.24 | clyrrad | ok let me get that |
| 19:05.52 | clyrrad | I will run the debug just on the peer |
| 19:09.32 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 19:10.35 | clyrrad | [TK]D-Fender: Ok i got it, just cleaning it up will pastebin it momentairly |
| 19:11.49 | *** join/#asterisk gardo (n=gardo@121.97.194.130) |
| 19:13.05 | clyrrad | [TK]D-Fender: Here is is: http://rafb.net/p/L8Sbpm30.txt |
| 19:14.24 | flujan | guys, when I originate a call using the AMI does asterisk generates a event on the ami interface? |
| 19:16.55 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 19:17.27 | *** join/#asterisk TUplink (n=Tommy_Hu@c-24-126-38-248.hsd1.wv.comcast.net) |
| 19:17.53 | TUplink | TOP says astersik is using 291MB or ram at an idle is that normals? |
| 19:17.56 | TUplink | normal* |
| 19:23.08 | [TK]D-Fender | clyrrad: HRM |
| 19:23.23 | *** part/#asterisk TUplink (n=Tommy_Hu@c-24-126-38-248.hsd1.wv.comcast.net) |
| 19:23.29 | [TK]D-Fender | clyrrad: PB your peer & general sip.conf |
| 19:23.42 | clyrrad | [TK]D-Fender: ok comming up |
| 19:25.07 | *** join/#asterisk jhb (n=joerg@81-5-139-2.dsl.eclipse.net.uk) |
| 19:26.21 | *** part/#asterisk jmls (n=jmls@62.49.235.130) |
| 19:27.45 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) [NETSPLIT VICTIM] |
| 19:28.00 | clyrrad | [TK]D-Fender: here ya go: http://rafb.net/p/m94oAR92.txt |
| 19:28.39 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 19:32.58 | [TK]D-Fender | clyrrad: fromdomain=XXX.XXX.XXX.XXX <- only wierd looking bit. |
| 19:33.16 | clyrrad | [TK]D-Fender: whats wierd is, when you listen to the voicemails sometimes they start to play, then they just die |
| 19:33.39 | clyrrad | [TK]D-Fender: LOL, that was me doing xxx.xxx.xxx.xxx for privacy :p |
| 19:36.58 | clyrrad | [TK]D-Fender: Would you agree the 603 error message is unrelated to the call droping its audo? |
| 19:37.16 | clyrrad | audio* |
| 19:39.37 | *** part/#asterisk jmls (n=jmls@62.49.235.130) |
| 19:40.32 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
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| 19:42.04 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 19:42.10 | [TK]D-Fender | clyrrad: I'm a bit at a loss on this... but this should help someone pickup the ball |
| 19:42.43 | clyrrad | [TK]D-Fender: I do belive this is the first time ever I asked a question you didnt know the answer to heheh |
| 19:42.57 | clyrrad | [TK]D-Fender: thanks for your help though appreciate you troublshotting this with me |
| 19:43.04 | hmmhesays | you haven't asked him enough questions |
| 19:43.10 | hmmhesays | lol |
| 19:43.27 | clyrrad | mmmmmmmm chocolate chip :P |
| 19:44.51 | clyrrad | Well if anyone else has any ideas, im all EYES :) |
| 19:47.12 | rob0 | Give my regards to Larry and the other Darryl. |
| 19:48.05 | hmmhesays | X.T in the polycom dialplan would be a catch all right? |
| 19:48.17 | [TK]D-Fender | hmmhesays: All numeric, yes |
| 19:48.45 | hmmhesays | does that differ from x.T? |
| 19:49.15 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 19:51.52 | BCS-Satori | Question: When using followme service, are you able to make the musiconhold go away and instead place ringing of the line? If you cant put the real ring is there an audio file that comes with asterisk that has about 20seconds of a ring? |
| 19:52.05 | Qwell | ~itsp-us |
| 19:52.08 | Qwell | ~itsp |
| 19:52.09 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 19:52.14 | Qwell | ~itsplist-us |
| 19:52.15 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com |
| 19:54.13 | jhb | hi *, I try to do one more action after a MeetMe command - that is, after the user has quit the conf. Any ideas? |
| 19:57.08 | hmmhesays | finally got this server set up |
| 19:57.15 | hmmhesays | what a pain in the ass |
| 20:04.47 | *** join/#asterisk Pagautas (n=bigman@83.171.14.250) |
| 20:04.57 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 20:07.50 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 20:13.06 | ice_croft | ку |
| 20:13.08 | ice_croft | re |
| 20:13.39 | ice_croft | still need some answers with e1 framed/unframed mode |
| 20:14.06 | ice_croft | help please |
| 20:14.26 | BCS-Satori | Hmm, is there a way to disable followme's "you have an incoming call" and if the person on the receiving end picks up their phone its answered? |
| 20:17.55 | *** join/#asterisk RoyK (n=roy@ip-10-5-149-91.dialup.ice.no) |
| 20:19.13 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 20:22.42 | BCS-Satori | ^^anyone, know if you can disable the followme's you have an incomnig call message and just allow auto accept if the phone is picked up |
| 20:23.53 | *** join/#asterisk qdk (n=qdk@195.242.194.41) |
| 20:31.11 | *** join/#asterisk kand (n=kanderso@139.148.188.72.cfl.res.rr.com) |
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| 20:34.02 | e` | Can anyone think of why a polycom soundstation 4000 would work fine on incoming calls, but crash and reboot when trying to call out? |
| 20:36.10 | *** join/#asterisk JonR800 (i=jon@p1mp.org) |
| 20:37.15 | BCS-Satori | e`: Do you have multiple vlans? both our cisco station (made by polycom) and (sound station 4000) did that when they were on a mistagged vlan |
| 20:37.35 | BCS-Satori | e`: our best solution was to place it in a native vlan and not use the device to tag |
| 20:41.09 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 20:41.44 | e` | I have mutiple vlans, but all the phones are the same 1 |
| 20:42.35 | BCS-Satori | e`: are you using tagging? make sure that the device is in a single native vlan or is tagged on a tagged vlan |
| 20:42.36 | DarKnesS_WolF | good evening geeks |
| 20:43.43 | e` | I'm not sure if we are using taggin, i'm kinda new to the company and don't know all the ins-and outs yet |
| 20:45.06 | BCS-Satori | e`: you are only allowed 1 native vlan (typically this is your computer network) and then if you want to use another vlan you need to tag it (making sure teh phone supports the tagging too) |
| 20:45.45 | BCS-Satori | e`: if the device doesnt support tagging you need to exclude it from the native vlan (computers lets say 1) and then make the pvid the phone net and make the switch be untagged on phone net on that port |
| 20:45.56 | nhuisman_work | time to order asterisk be! |
| 20:45.57 | nhuisman_work | woot |
| 20:46.00 | BCS-Satori | e`: your best bet is to untag the phone and place it in a native phone network port |
| 20:46.09 | nhuisman_work | hahaha f call manager in the a |
| 20:47.56 | nhuisman_work | just a quick question : I was going to use a redfone external gateway. Anyone had experience with these? |
| 20:48.35 | nhuisman_work | these things : http://www.voipsupply.com/product_info.php?products_id=2025&searchid=490560 |
| 20:56.30 | [TK]D-Fender | nhuisman_work: YUCK |
| 20:56.39 | bkruse | my box says ntldr is missing? :P |
| 20:56.39 | *** join/#asterisk fors1 (n=forsen@pat-tdc.opera.com) |
| 20:56.39 | nhuisman_work | why do you say yuck? |
| 20:57.11 | bkruse | [TK]D-Fender: are those the like $20,000 boxes? |
| 20:57.13 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 20:57.28 | [TK]D-Fender | nhuisman_work: Because those have no EC, only speak TDMoE which nobody gives a rats ass about, and can only talk directly to * and is consequently non-recyclable |
| 20:57.38 | [TK]D-Fender | bkruse: No, they jsut suck |
| 20:57.47 | bkruse | [TK]D-Fender: oh right :] |
| 20:57.49 | nhuisman_work | it's not like i'm going to recycle them |
| 20:57.51 | fors1 | hi. I just upgraded my debian server from sarge to etch, and from 2.6.8 to 2.6.18. Now my sangoma A200 card is not recognized by zaptel anymore. My guess is some issue with udev, but i'm currently clueless |
| 20:57.58 | nhuisman_work | that's what their purpose is |
| 20:58.14 | nhuisman_work | plus at $1200 who cares |
| 20:58.23 | nhuisman_work | and yes they do have EC |
| 20:58.23 | mocker | fors1: Did you recompile the wanpipe/zaptel stuff? |
| 20:58.24 | [TK]D-Fender | nhuisman_work: Then please refer to the "no EC, and SUCK" comments |
| 20:58.37 | [TK]D-Fender | nhuisman_work: Wouldn't trust it. You call. |
| 20:58.38 | *** join/#asterisk arguile (i=user224@KTNRON06-1242488957.sdsl.bell.ca) |
| 20:58.45 | nhuisman_work | i listed the wrong item, sec |
| 20:58.59 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 20:59.02 | fors1 | mocker: yes, also tried with the newest version (zaptel 1.4.7.1 and wanpipe 3.2.1) |
| 20:59.13 | nhuisman_work | http://www.voipsupply.com/product_info.php?products_id=3663 |
| 20:59.14 | nhuisman_work | there |
| 20:59.22 | nhuisman_work | that one has echo cancellation listed |
| 20:59.30 | fors1 | no errors while recompiling, but "wanrouter hwprobe" doesn't find anything |
| 20:59.38 | nhuisman_work | why wouldn't you trust it if it says it has it? |
| 21:00.09 | nhuisman_work | rrier Class Echo Cancellation: |
| 21:00.09 | nhuisman_work | <PROTECTED> |
| 21:00.09 | nhuisman_work | <PROTECTED> |
| 21:00.09 | nhuisman_work | <PROTECTED> |
| 21:00.16 | *** join/#asterisk mvanbaak (n=michiel@vanbaak.xs4all.nl) |
| 21:00.36 | nhuisman_work | maybe that's somehow crap and I just don't recognize it, I do admit i'm a n00b when it comes to telco stuff. |
| 21:01.02 | [TK]D-Fender | nhuisman_work: Well nobody I know would use them. Go right ahead, and remember that specs don't add up to the whole user experience |
| 21:01.23 | nhuisman_work | there are other t1 gateway devices |
| 21:01.29 | nhuisman_work | that one was just really cheap |
| 21:02.07 | nhuisman_work | like audiocodes mediant 2000 type stuff. |
| 21:02.43 | nhuisman_work | do you have any suggestions as to a different external t1 gateway that you think is good? |
| 21:03.10 | nhuisman_work | I only have one t1 pri so my solution was to create to * boxes and mirror them via rsync, then tell all my phones to dual register. |
| 21:04.10 | [TK]D-Fender | nhuisman_work: Indeed anyone looking for a scalable redundant solution I'd aim at a Mediant |
| 21:04.25 | nhuisman_work | it looks like the digium cards also use the g.168 algorithm |
| 21:04.33 | nhuisman_work | here is my situation |
| 21:04.38 | nhuisman_work | I only have 65 phones |
| 21:04.48 | nhuisman_work | We are only going to grow by about 3 phones a year |
| 21:04.53 | bkruse | [TK]D-Fender: what are the super expensive redundant/failover t1 box that everyone always mentions? |
| 21:05.03 | nhuisman_work | it's a university setting and we don't hire faculty or people very often. |
| 21:05.30 | [TK]D-Fender | bkruse: I think I missed the answer to that one every time... that more for like actual LINK failure though.. this is more like SERVER failover |
| 21:06.06 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 21:06.22 | bkruse | [TK]D-Fender: gotcha |
| 21:06.24 | bkruse | ty |
| 21:07.04 | clyrrad | [TK]D-Fender: what do you make of this? That same phone that dies when you dial internally to voicemail, can call the main PBX DID and access voicemail "remotely" and the voicemail message does not die........ make any sense? |
| 21:08.21 | mocker | clyrrad: define 'dies' |
| 21:09.24 | mocker | fors1: Are there any dmesg errors? Do the modules get loaded? |
| 21:09.37 | clyrrad | mocker: the audio disapears, the phone still says connected, but there is no audio and there is a Maximum retries exceeded on transmission 952fa60c-d8c44628@192.168.3.195 for seqno 102 (Critical Response) error message on the CLI |
| 21:12.35 | mocker | clyrrad: can you pastebin your extensions.conf for that section? |
| 21:13.35 | clyrrad | mocker: Here http://rafb.net/p/m94oAR92.txt and here http://rafb.net/p/L8Sbpm30.txt |
| 21:13.52 | fors1 | mocker: not so much, ztcfg -v gives me this though "line 0: Unable to open master device '/dev/zap/ctl'" zaptel module and wanrouter module is loaded |
| 21:14.02 | *** join/#asterisk mvanbaak (n=michiel@vanbaak.xs4all.nl) |
| 21:14.06 | fors1 | lspci reports that there is a Sangoma A200 card installed |
| 21:16.26 | mocker | clyrrad: extensions.conf section? |
| 21:16.46 | clyrrad | mocker: extensions.conf is quite big, what part you wondering on? |
| 21:16.57 | mocker | clyrrad: can you pastebin your extensions.conf for that section? |
| 21:17.02 | mocker | :) |
| 21:17.08 | clyrrad | mocker: for the voicemail section? |
| 21:17.11 | clyrrad | thats where the issue is |
| 21:18.40 | mocker | Yup. |
| 21:18.52 | mocker | I'm guessing that there are different sections for the voicemail from internal and external?\ |
| 21:18.58 | mocker | Just paste the section that's failing. |
| 21:20.06 | clyrrad | ok |
| 21:20.16 | *** join/#asterisk MooingLemur_ (n=troy@unaffiliated/mooinglemur) |
| 21:21.42 | *** join/#asterisk VJFROMGT (n=vjfromgt@user-387g9sf.cable.mindspring.com) |
| 21:21.52 | fors1 | mocker: finally! I solved it :) make install-udev in zaptel did the trick. Thanks for your help anyway :) now i'm not getting killed by my boss |
| 21:22.21 | mocker | fors1: np, next time do them after hours, not on Friday. :P |
| 21:22.36 | clyrrad | mocker: here it is http://rafb.net/p/RHqBQ596.html |
| 21:23.21 | fors1 | mocker: well, here it is after hours.. on a friday.. then I have all weekend to try :) |
| 21:23.42 | clyrrad | mocker: there is not an issue to dial the voicemail, it dials fine, and you can hear the Comedian mail prompts, the problem is when it starts to play back the voicemail to you, it plays it for a few seconds, then the audio dies, the phone still sais connected, but there is no audio, and there is that crital error message on the CLI that mentioned above |
| 21:25.01 | mocker | clyrrad: What's the sip.conf look like for that phone? |
| 21:25.03 | nhuisman_work | does anyone know where I can find out what versions of asterisk and zaptel are included in asterisk business edition? |
| 21:25.25 | mocker | Might make sure that canreinvite=no so it doesn't try to drop out of the media path. |
| 21:25.38 | mocker | Also allow all codecs to make sure that's not the issue. |
| 21:25.39 | clyrrad | mocker: its in the first paste URL i sent you |
| 21:25.46 | clyrrad | mocker: there was a sip debug and the sip.conf |
| 21:25.59 | clyrrad | mocker: Here http://rafb.net/p/m94oAR92.txt and here http://rafb.net/p/L8Sbpm30.txt |
| 21:29.11 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 21:30.26 | VJFROMGT | for some reason only 1 iax2 trunk works on my server at a time, i have 2 trunks but 1 becomes unreachable all the time |
| 21:32.29 | mocker | clyrrad: Everything I'm googling seems to indicate this is a NAT issue. |
| 21:32.33 | mocker | Is your phone behind NAT? |
| 21:32.47 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 21:33.18 | clyrrad | mocker: yes it is, it always has been, but the phone uses STUN and has been connected and working for more than a year now, it just started doing this today |
| 21:33.24 | DarKnesS_WolF | VJFROMGT: may be it is really unreachable? |
| 21:34.25 | kand | clyrrad: I took a look at your sip debug and asterisk is struggling to communicate with your phone. Has your internet been wonky? |
| 21:35.20 | kand | clyrrad: it looks like lost packets or highly delayed packets |
| 21:35.27 | clyrrad | kand: nope, thats the strang thing, if you make a phone call, its perfect, no dropped calls, no sound issues nothing, its only when you check voicemail internally it has the issue. But if you check voicemail by dialing the PBX number (from the same affected phone) it works |
| 21:39.16 | kand | clyrrad: It looks like a NAT issue, lots of invites ACK, and 200 ok are ignored/retransmitted. But they wouldnt be correlated to just vm. Have you listen to any other vm messages on the off chance this message is corrupt and giving asterisk a hard time? |
| 21:39.19 | mocker | clyrrad: Is this just one phone exhibiting the issue? |
| 21:39.49 | mocker | Also, you aren't doing something wonky and storing VM on NFS are you? |
| 21:40.03 | clyrrad | kand: yes we have attempted to clear the mailbox, leave new message and the same thing happens every time |
| 21:40.16 | clyrrad | mocker: yes its just one phone in this office on this PBX |
| 21:40.43 | *** join/#asterisk rtasterisk (n=mozveren@se167-1-82-242-148-65.fbx.proxad.net) |
| 21:40.48 | rtasterisk | hello all |
| 21:41.05 | tzanger | http://194.90.248.2/ <-- can anyone identify that SIP GW? |
| 21:41.11 | kand | clyrrad: There went corrupt message spiking CPU.... hmmm. |
| 21:41.14 | clyrrad | mocker: and this phone only does it when you dial Voicemail Internally. If you call the PBX DID and access voicemail there is no issue |
| 21:41.41 | nhuisman_work | does anyone know of a way to search the asterisk mailing lists? |
| 21:41.45 | clyrrad | kand: nope we completely dumped the mailbox, and left new message many times - it happens consistantly |
| 21:41.53 | nhuisman_work | is there some website that has them all archived and searchable? |
| 21:42.06 | Qwell | nhuisman_work: like lists.digium.com? |
| 21:42.18 | nhuisman_work | uh is that searchable? |
| 21:42.55 | nhuisman_work | because it doesn't seem like it is. |
| 21:43.02 | Qwell | tzanger: Nateks? |
| 21:43.15 | Qwell | "nateks networks voicecom" |
| 21:43.17 | lirakis | later all, have a good weekend |
| 21:43.18 | rob0 | There's a search engine called "Google" which might be able to search it. |
| 21:43.31 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 21:44.02 | muiro | so I have a sip trunk. The calls are coming in fine. My problem is that it calls in with the extension an extension of "+XXXXXXXXXXX". Problem is, asterisk won't match with the "+" sign. I've been trying _+XXXXXXXXXXX but it won't work. _. won't match it either. |
| 21:44.24 | mocker | Qwell: And there's tons of login pages if you search for otgw.cgi |
| 21:44.33 | Qwell | yep |
| 21:44.54 | hmmhesays | oh drupal you are killing me today |
| 21:45.25 | rtasterisk | <PROTECTED> |
| 21:45.25 | rtasterisk | <PROTECTED> |
| 21:45.25 | rtasterisk | <PROTECTED> |
| 21:45.25 | rtasterisk | <PROTECTED> |
| 21:45.25 | rtasterisk | <PROTECTED> |
| 21:45.26 | rtasterisk | <PROTECTED> |
| 21:45.28 | rtasterisk | <PROTECTED> |
| 21:45.30 | rtasterisk | <PROTECTED> |
| 21:45.32 | rtasterisk | <PROTECTED> |
| 21:45.34 | rtasterisk | <PROTECTED> |
| 21:45.37 | rtasterisk | <PROTECTED> |
| 21:45.39 | rtasterisk | <PROTECTED> |
| 21:45.43 | mocker | wtf. |
| 21:45.54 | mocker | rtasterisk: You type very fast. |
| 21:46.28 | rtasterisk | hello |
| 21:46.36 | rtasterisk | what is your opinion ?K |
| 21:46.38 | tzanger | Qwell: looks like it, yeah, thanks :-) |
| 21:47.54 | kand | muiro: I am using _+X. on a production box with np |
| 21:48.11 | Deeewayne | rtasterisk: you are not asking a simple question |
| 21:48.30 | rtasterisk | Its why i ask it :) |
| 21:49.37 | Deeewayne | I don't know the yate design, so I couldn't get into a low level comparison between the 2 implementations |
| 21:49.56 | hmmhesays | yate still being developed by the same people it was 2 years ago |
| 21:49.56 | hmmhesays | ? |
| 21:49.57 | muiro | kand: I just tried exactly that and it still isn't matching. The contexts are named correctly. The first has priority 1. |
| 21:50.18 | hmmhesays | I can't remember the guys name he spoke at the first cluecon though |
| 21:50.43 | kand | muiro: pastebin the sip.conf and related context |
| 21:50.48 | muiro | can do |
| 21:51.08 | *** join/#asterisk funxion (n=x@63.214.236.169) |
| 21:51.44 | Deeewayne | I would have to assume that yate does not use a single queue for all channels, frames, events, etc.. Does it ? |
| 21:52.08 | muiro | kand: http://pastebin.com/m26e0504d |
| 21:52.44 | rtasterisk | Dont know really |
| 21:52.47 | rtasterisk | have to check it |
| 21:53.11 | rtasterisk | But the architecture schema is oriented around a message dispatcher |
| 21:53.18 | rtasterisk | and message routing protocoll |
| 21:53.39 | muiro | kand: oh, let me add the error |
| 21:54.12 | muiro | kand: http://pastebin.com/m1b0d7965 |
| 21:54.21 | kand | muiro: ya because I dont see any error there. Tell me tho, have you ever had one way audio after a call was placed on hold with Bandwidth.com? |
| 21:55.49 | *** join/#asterisk Havokmon (n=rick@64-198-2-66.ip.mcleodusa.net) |
| 21:55.59 | Deeewayne | rtasterisk: I think, like everything with computers, that there are pros and cons of either approach |
| 21:56.17 | muiro | kand: I literally got this trunk an hour ago. One way audio? and do you mean on hold like, on their support line? |
| 21:57.23 | muiro | kand: ah, wait. It's very possible I have the ip's mixed up for the inbound and outbound |
| 21:58.26 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 21:58.32 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-68-93-82-209.dsl.hstntx.swbell.net) |
| 21:58.36 | kand | muiro: no, like when you place a caller on hold then retrieve it. Try from the cli: sip show peer bandwidth.com_inbound (make sure context is fromtrunk); show dialplan fromtrunk (make sure it matchs what is in your dial plan) |
| 22:00.33 | kand | muiro: nm you are right they are backward: Call from 'bandwidth.com_>>outbound<<' to extension '+12345428018' rejected |
| 22:00.41 | muiro | yup |
| 22:00.45 | muiro | that just fixed it |
| 22:01.02 | muiro | thanks for your help |
| 22:01.41 | kand | muiro: np. If you have one way audio after retrieving a call from hold let me know please. I will be on here when I can. |
| 22:02.30 | muiro | kand: ko, I'll write your name down and let you know as soon as I get a chance |
| 22:02.49 | kand | muiro: Thank you! |
| 22:03.49 | muiro | :w |
| 22:03.53 | muiro | whoops |
| 22:06.00 | Havokmon | Anyone tried to get a TE100P from PhonicEQ to work? I'm trying to find a nice asterisk distro to start with before getting into the nitty gritty (compiling drivers,etc)... |
| 22:06.40 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 22:10.45 | kand | Havokmon: did you have a specific question? |
| 22:11.46 | Havokmon | Has anyone successfully used a TE100P, and if so, how did you do it? |
| 22:11.58 | Havokmon | I would prefer the easiest method, as I'm an asterisk newbie. |
| 22:13.31 | Havokmon | Is hardware not an asterisk question? I'm kinda unsure why the OS see's the card, but TrixBox CE and Pro do not. I don't know how it all works yet. |
| 22:13.42 | Corydon76-dig | Easiest method is to buy a card from the originator and not a clone card |
| 22:14.17 | Havokmon | So that would be a no then ;) |
| 22:14.56 | Corydon76-dig | ~trixbox |
| 22:14.56 | jbot | [~trixbox] Trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 22:15.20 | mvanbaak | jbot should be ajusted |
| 22:15.46 | mvanbaak | [~trixbox] Trixbox is a full virus that will render your system useless |
| 22:16.17 | nhuisman_work | hey kand |
| 22:16.19 | Havokmon | Ok, but if the OS detects my ethernet card, Apache doesn't decide if it's going to bind or not ;) |
| 22:16.38 | Havokmon | Does asterisk have internal drivers of some sort? |
| 22:16.46 | mvanbaak | Havokmon: zaptel |
| 22:16.50 | dexpdx | trixbox would be a lot better if it wasn't written in php |
| 22:17.02 | dexpdx | infact lots of things would be better if they wernt written in php |
| 22:17.02 | mvanbaak | dexpdx ;0 |
| 22:17.23 | Havokmon | mvanbaak: so the zaptel is completely separate from the kernel module? |
| 22:17.49 | dexpdx | Havokmon: zaptel is the abstraction layer that allows asterisk to talk to the kernel drivers |
| 22:17.51 | mvanbaak | Havokmon: the zaptel package contains kernel module sources for all digium supported hardware |
| 22:18.16 | Havokmon | gotcha. Thanks. |
| 22:18.45 | Qwell | Havokmon: Digium doesn't support that card - therefore, no driver. You're SOL |
| 22:19.01 | Qwell | Call your CC company, and do a chargeback. |
| 22:19.15 | mvanbaak | and get a card from digium |
| 22:19.24 | Havokmon | Serious? Really that bad? |
| 22:19.29 | mvanbaak | yup |
| 22:19.30 | dexpdx | one with echo cancellation |
| 22:19.43 | Havokmon | I thought you guys were just like the Windows guy when someone says Linux :P |
| 22:20.02 | Corydon76-dig | Do what? |
| 22:20.54 | mvanbaak | nothing ;) |
| 22:21.04 | dexpdx | your mom |
| 22:21.23 | dexpdx | oh now your sister |
| 22:21.28 | mvanbaak | lol |
| 22:21.49 | Corydon76-dig | dexpdx: necrophiliac |
| 22:21.56 | mvanbaak | hahahahahaha |
| 22:22.04 | dexpdx | Corydon76-dig: not if I'm a time traveller |
| 22:22.07 | mvanbaak | 'I see dead people' |
| 22:22.12 | dexpdx | I could be yer daddy |
| 22:22.22 | mvanbaak | who's your daddy ? |
| 22:22.25 | Qwell | hell, he'd probably still let you be his daddy... |
| 22:22.32 | dexpdx | hahah |
| 22:22.57 | dexpdx | it's hard to talk shit on IRC while eating rice with chopsticks |
| 22:23.16 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 22:25.33 | mvanbaak | use the chopsticks to grab the shit and throw it on irc ;) |
| 22:26.31 | X-Filez | have ppls use mISDN ? need help, I'm installed mISDN, load modules , i see in dmesg = have 2 PCI ISDN.. but i dont understand, in asterisk 1.4.15 misdn.conf where select PCI 1 and PCI 2... |
| 22:28.01 | dexpdx | 30wpm using chopsticks |
| 22:29.03 | mvanbaak | X-Filez: what's the problem ? |
| 22:29.27 | X-Filez | mvanbaak: don't understand to configure misdn.conf... :( |
| 22:30.35 | X-Filez | http://pastebin.com/m6f292c69 , port 1 have 2 line, and port 2 have 2 line.. |
| 22:34.32 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 22:35.44 | *** join/#asterisk dexpdx (n=jason@66-162-134-242.static.twtelecom.net) |
| 22:37.53 | *** part/#asterisk dexpdx (n=jason@66-162-134-242.static.twtelecom.net) |
| 22:39.34 | X-Filez | Have Phone Provider :lattelecom:, Lattelecom give me 2 ISDN cabels, were 1 cabel 2 number, i buy ISDN two card where "Cologne Chip", and installing drivers misdn, run misdnportinfo i see 2 my pci cards : http://pastebin.com/m6f292c69 : , i want configure ISDN to asterisk, please help.. |
| 22:43.58 | nhuisman_work | does anyone know of a way to take a single pri and basically make it into two, so if the a primary asterisk box fails then I don't have to plug it into the second box. For instance is it possible to get some manner of t1 pri hardware splitter. |
| 22:45.10 | funxion | anyone know of way to jump priorities after dial command but before the called party picks up? |
| 22:46.56 | putnopvut | funxion: That sounds like you could maybe use the 'M' option for dial. |
| 22:48.30 | funxion | M |
| 22:49.33 | funxion | isnt that for Macros? |
| 22:50.01 | putnopvut | Yeah, but it's the first thing I thought of when you said "after dialing but before the called party picks up" |
| 22:50.28 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 22:50.38 | funxion | that doesnt work |
| 22:50.54 | funxion | I need to be able to press 1 for voicemail while the phone is ringing |
| 22:51.32 | putnopvut | I certainly know how that can be done with a queue, but not sure about Dial() though. |
| 22:51.41 | funxion | yeah |
| 22:51.42 | funxion | well |
| 22:51.46 | funxion | I tried with a queue |
| 22:51.47 | putnopvut | Ah, wait a sec... |
| 22:51.50 | kand | funxion: check into features.conf |
| 22:51.54 | putnopvut | perhaps the 'd' option. |
| 22:52.08 | funxion | Im not so good with features.conf |
| 22:52.28 | funxion | I'm using the t option and doing an Answer before the dial |
| 22:52.40 | funxion | but not able to do blind trasnfer |
| 22:52.45 | funxion | I tried |
| 22:53.16 | funxion | putnopvut I think your on to something there |
| 22:53.32 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 22:53.46 | putnopvut | Yeah, it sounds exactly like what you were talking about. |
| 22:53.56 | funxion | thats it |
| 22:54.02 | funxion | thanks man |
| 22:54.21 | funxion | I totally missed that option when reading |
| 22:54.26 | funxion | kewl |
| 22:54.29 | kand | funxion: you need to make sure that your dial has access to a feature, the quickest way is to [globals] DYNAMIC_FEATURES => atxfer#blindxfer |
| 22:54.40 | funxion | thanks kand |
| 22:54.44 | kand | np |
| 22:55.40 | Havokmon | FYI - I got it (TE100P) working. They have 2 custom trixbox ISO's on their site, the latest didn't work, but the previous did. |
| 22:55.59 | Havokmon | later all |
| 23:14.20 | mvanbaak | hhmm, nice |
| 23:14.36 | mvanbaak | a cloned TE100P with some custom trixbox hack |
| 23:14.53 | mvanbaak | I wonder if that uses the password 'please_take_my_system' |
| 23:16.58 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 23:18.57 | *** join/#asterisk dexpdx (n=dexpdx@66-162-134-242.static.twtelecom.net) |
| 23:22.43 | X-Filez | ISND Layer 4 protocol 0x04000001 is detected, but not allowed for TE lib. , what this is ? |
| 23:25.30 | X-Filez | ISND = ISDN :) |
| 23:26.31 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 23:32.32 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 23:33.26 | X-Filez | hm, in mISDN , where i can find check, ISDN line UP or Down ? |
| 23:34.24 | *** join/#asterisk galeras (n=galeras@190.25.220.211) |
| 23:35.30 | *** join/#asterisk Greek-Boy (n=email@41.221.58.5) |
| 23:35.58 | *** join/#asterisk bmg505 (n=leon@196.209.180.166) |
| 23:40.15 | galeras | Sirs, i'm getting many of this messages: "Primary D-Channel on span 2 up". Any idea? |
| 23:40.49 | *** join/#asterisk |Johny| (n=gomesper@bacus.corp.fccn.pt) |
| 23:41.01 | |Johny| | Hi, where can I find some help to make a simple Asterisk SIP trunk with a Patton GW? |
| 23:41.08 | |Johny| | I dont know if its Asterisk who must register in Patton GW |
| 23:41.15 | |Johny| | or if I should register the Patton GW with Asterisk |
| 23:45.36 | nhuisman_work | when your asterisk ha kicks in and another asterisk box switches to take the ip do all the phone have to re-register? |
| 23:52.13 | jer | hrmm. i just replaced an x100p with a tdm400p with an fxo in port 4. my system sees it ("Module 3: Installed -- AUTO FXO") and i've adjusted my zaptel.conf, but i'm guessing the Zap channel is not still Zap/1 since i keep getting a channel unavailable error when trying to dial out. how can i find out the zap channel that it is on now? |
| 23:53.26 | rob0 | generatezaptelconf (I think) is your friend |
| 23:53.35 | rob0 | ztcfg -vvv maybe too |
| 23:53.59 | rob0 | bbl |
| 23:54.07 | *** part/#asterisk galeras (n=galeras@190.25.220.211) |
| 23:56.16 | jer | ztcfg -vvvv shows channel 04, i've tried to use Zap/4 as well, to no avail. i always get the following: http://pastebin.com/m38b61836 |
| 23:58.09 | tzafrir_home | jer, did you get any error from ztcfg? |
| 23:58.14 | jer | nope |
| 23:58.40 | jer | http://pastebin.com/m26e42b5b <-- ztcfg -vvvv output |
| 23:58.57 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 23:59.06 | tzafrir_home | as for galaras up there: that specific message is a good this. the only question is why the span was down in the first place |
| 23:59.53 | tzafrir_home | jer, next: asterisk -rx 'zap show channels' |
| 23:59.57 | tzafrir_home | what is the output? |
