| 00:01.04 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 00:05.41 | *** join/#asterisk pguima_pg (~pguima@ool-435384c3.dyn.optonline.net) |
| 00:06.40 | pguima_pg | Does anyone know how i can re-register my Free Fax License? I had to change servers. thank you |
| 00:12.43 | pguima_pg | Anyone? |
| 00:16.06 | jpsharp | Just get a new license. |
| 00:16.51 | pguima_pg | i cant get a new one, only if i pay for it. |
| 00:17.12 | pguima_pg | i already have one registered under my account |
| 00:20.20 | jpsharp | Weird. I have like 6. |
| 00:20.44 | *** join/#asterisk fireman_biff (~biff@65.48.176.14) |
| 00:20.48 | pguima_pg | 6 free fax licenses? how did you sign up for the other ones? |
| 00:20.52 | *** join/#asterisk HyperNerdV2_ (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 00:21.11 | *** join/#asterisk resist0r (~resist0r@69.31.131.51) |
| 00:28.10 | fireman_biff | what options are there for allowing people to use skype to call an asterisk PBX? |
| 00:34.40 | resist0r | I think the answer to the question you meant to ask is chan_skype |
| 00:36.13 | WIMPy | was |
| 00:42.17 | fireman_biff | what about skype connect? anybody ever tried that? |
| 00:51.10 | pguima_pg | exit |
| 01:29.19 | *** join/#asterisk l2trace99 (~chatzilla@rrcs-71-43-104-238.se.biz.rr.com) |
| 01:30.31 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 01:39.25 | *** join/#asterisk bintut (~bintut@119.234.1.15) |
| 01:50.29 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 01:50.29 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 02:05.36 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 02:05.36 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 02:09.29 | *** join/#asterisk bintut (~bintut@119.234.1.15) |
| 02:09.30 | jpsharp | chan_skype has been discontinued. |
| 02:13.57 | *** join/#asterisk bjweeks (~brandon@wikipedia/Brandon) |
| 02:18.23 | *** join/#asterisk ThinkGNU- (~ThinkGNU-@174-31-120-200.chyn.qwest.net) |
| 02:40.45 | *** join/#asterisk Rholk (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 02:51.30 | billytwowilly | is there a web gui for asterisk? if so, how do I turn it on? |
| 02:52.09 | ChannelZ | run!!! |
| 02:53.13 | *** part/#asterisk fireman_biff (~biff@65.48.176.14) |
| 02:55.46 | WIMPy | turns around |
| 02:57.29 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 02:58.34 | MJCS | i just installed asterisk using the x64 iso and my /etc/asterisk folder is empty |
| 02:58.55 | MJCS | billytwowilly: openpbx has a gui |
| 02:59.32 | MJCS | billytwowilly: also asterisk-gui.noarch : Graphical User Interface (GUI) for Asterisk, The Open Source PBX |
| 03:01.28 | MJCS | nevermind had to install the config package |
| 03:01.32 | *** join/#asterisk Kraln (~kraln@69.169.90.239) |
| 03:04.07 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-132-58.ks.ks.cox.net) |
| 03:06.48 | *** join/#asterisk PhoenixMage (~Phoenix@CPE-120-146-192-94.static.vic.bigpond.net.au) |
| 03:20.25 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-lutnwrrnyniszkoa) |
| 03:21.02 | ThinkGNU- | If a system has multiple outside lines, what's the best way to make it pick up one line, check if it's in use and move to the next if it is in use? |
| 03:21.11 | ThinkGNU- | I'm thinking If statements |
| 03:21.28 | WIMPy | What knid of lines? |
| 03:21.35 | ThinkGNU- | SIP |
| 03:22.11 | WIMPy | "line" ins an interesting term for that. |
| 03:22.21 | WIMPy | Why do you care if a SIP account is in use? |
| 03:24.06 | ThinkGNU- | How many calls can be made on one external sip account simultaneously? |
| 03:24.35 | WIMPy | Ask whoever gave you the account. |
| 03:24.57 | ThinkGNU- | okay so the limit is in the provider |
| 03:25.37 | WIMPy | yes |
| 03:26.15 | ThinkGNU- | I have six available, trunks I guess you'd call them? So they should allow six simultaneous calls then? |
| 03:27.08 | *** part/#asterisk mcf3782 (~mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 03:27.16 | WIMPy | Apart from the fact that sip and trunk dont really fit togeter, trunk should at least mean multiple calls. |
| 03:27.31 | WIMPy | So 6 trunks should be 12 calls or more. |
| 03:29.10 | ThinkGNU- | I think this is my confusion is that right now we're using an ATA with six analog ports. Our old comdial pbx of course has to see each as a "line". |
| 03:29.30 | ThinkGNU- | I'm trying to wrap my head around the way this will work when I get an * PBX in place. |
| 03:31.17 | ThinkGNU- | Sorry if my questions don't seem well thought out. |
| 03:43.00 | ThinkGNU- | So if I have my pattern matching right for outgoing calls and they all direct at my SIP provider I should be able to send calls there until the limit they set on me is reached right? |
| 03:43.24 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 03:44.06 | WIMPy | yes |
| 03:44.50 | ThinkGNU- | okay |
| 03:44.58 | ThinkGNU- | Sorry for acting like a total noob |
| 03:46.22 | ThinkGNU- | Thanks for your help |
| 03:49.15 | *** join/#asterisk slav3_kitten (~kitten@unaffiliated/slav3-kitten/x-0866809) |
| 04:13.06 | fubada | hi guu |
| 04:13.08 | fubada | hi guys |
| 04:13.13 | slav3_kitten | yo |
| 04:13.16 | fubada | i have a very strange problem thats keeping me at office super late |
| 04:13.22 | fubada | i have an asterisk box with two interfaces |
| 04:13.31 | fubada | and RTP needs to be forcd out over eth2 |
| 04:13.33 | fubada | and not eth0 |
| 04:13.43 | fubada | sometimes, it takes the path of eth0 |
| 04:13.46 | fubada | HOW can i change this |
| 04:13.59 | slav3_kitten | do you need to get traffic in on eth0? |
| 04:14.11 | fubada | just local traffic |
| 04:14.26 | slav3_kitten | local traffic is the phone traffic i assume? |
| 04:14.42 | fubada | yes and also other data like OS updates |
| 04:14.59 | slav3_kitten | hmmm |
| 04:15.08 | fubada | eth0: phones and other OS update shit |
| 04:15.16 | fubada | eth2: direct connection to t1 line for sip |
| 04:15.17 | slav3_kitten | eth2 is to a dmz i take it, and eth0 hits a router before wan? |
| 04:15.21 | fubada | yes |
| 04:15.53 | fubada | i fixed the probem where packets coming in on eth2, leeve out eth2 |
| 04:15.54 | slav3_kitten | make an access list on the router preventing the eth0 ip address from touching the wan |
| 04:15.55 | fubada | that works |
| 04:16.06 | fubada | but I cant force asterisk to do all rtp communication on eth2 |
| 04:16.51 | slav3_kitten | if you force the ip of eth0 not to have a route to wan, it will send it the only place it has a valid route |
| 04:17.12 | fubada | okay hang on let me explain better |
| 04:17.43 | fubada | eth0 is 192.168.60.12, its router is 192.168.60.1, its attached to a cable modem with 100mbit to the inet. I use this for bulk traffic, this connection is NOT very reliable |
| 04:18.00 | fubada | eth2 is verizon t1 public IP, i use this for sip and voip |
| 04:18.05 | fubada | it is a reliable line |
| 04:18.14 | slav3_kitten | right |
| 04:18.15 | fubada | phones are on 192.168.60.x, eth0 |
| 04:18.35 | fubada | so i need asterisk to always do its rtp on eth2 |
| 04:18.42 | slav3_kitten | if on your router you add an access list rule preventing 192.168.60.12 from connecting to wan via the router |
| 04:18.57 | fubada | ok but that will break my need to push all bulk traffic out eth0 |
| 04:18.59 | fubada | such as updates |
| 04:19.04 | slav3_kitten | the only valid route to wan will be eth2 |
| 04:19.14 | fubada | i cant be pulling updates and do voip out the t1, its too slow |
| 04:19.21 | slav3_kitten | right, but you can in an access list state the protocol or port number you're disallowing |
| 04:19.25 | fubada | right |
| 04:19.30 | fubada | thanks will try |
| 04:19.38 | slav3_kitten | so you disallow eth0 rtp to wan |
| 04:19.43 | slav3_kitten | but everything else will work |
| 04:20.01 | fubada | i was hoping to solve this entirely on the asterisk box |
| 04:20.13 | fubada | i guess i cant bind asterisk to only eth2 cos my phones are on eth0 |
| 04:20.51 | slav3_kitten | iirc the book states you either bind it to one interface, or all interfaces |
| 04:21.08 | slav3_kitten | so the router is about the only way to direct traffic |
| 04:22.19 | fubada | or, because i have a local updates server on 60.x anyways, i can just make eth2 my default gw on * box |
| 04:23.26 | slav3_kitten | maybe? i'm not sure of your network config |
| 04:23.38 | fubada | thanks man |
| 04:23.45 | slav3_kitten | but i think throwing an access list rule on your router would be the simple solution |
| 04:24.07 | fubada | so i disallow outgoing rtp traffic from *'s eth0 on the router |
| 04:24.17 | fubada | wont * just keep trying it and failing |
| 04:24.31 | fubada | what makes you think it will retransmit on eth2 |
| 04:24.48 | slav3_kitten | no, there will be no valid route for rtp traffic. linux will send it out the only interface with a valid route |
| 04:26.17 | fubada | so if my rtpstart is 10000 and end is 20000, i need to block that entire range? |
| 04:27.42 | slav3_kitten | yep |
| 04:50.57 | *** join/#asterisk MJCS (~script@ip68-5-48-206.oc.oc.cox.net) |
| 04:51.16 | MJCS | why is it that I can only answer after the 2nd ring? |
| 05:00.19 | ChannelZ | Bad reflexes? |
| 05:01.21 | slav3_kitten | poor hearing? |
| 05:01.54 | ChannelZ | Do more meth and you'll be able to keep answering it even when it's not ringing |
| 05:03.07 | slav3_kitten | why not bath salts, then you can answer it while chewing on a face? |
| 05:04.20 | Kraln | can you guys help me debug a problem with the jabber module? |
| 05:04.23 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 05:04.40 | Kraln | seems to fail to authenticate with gtalk (I'm 100% sure the password is correct, and it has no funny characters in it) |
| 05:04.47 | slav3_kitten | i'm new. and not awake |
| 05:04.58 | Kraln | I just get back a <not authorized/> from el goog |
| 05:05.37 | ChannelZ | are there funny characters in it? |
| 05:05.49 | Kraln | no. |
| 05:07.11 | ChannelZ | well make sure your username in jabber.conf is right I guess |
| 05:07.13 | ChannelZ | works here |
| 05:07.22 | Kraln | how is your username? |
| 05:07.31 | Kraln | email@domain.com/Talk or whatever? |
| 05:07.32 | slav3_kitten | https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google help maybe? |
| 05:07.36 | ChannelZ | me@gmail.com/Talk |
| 05:08.11 | *** part/#asterisk ThinkGNU- (~ThinkGNU-@174-31-120-200.chyn.qwest.net) |
| 05:09.01 | slav3_kitten | night all |
| 05:09.08 | ChannelZ | nighty |
| 05:15.07 | Kraln | slav3_kitten, ChannelZ: http://pastebin.com/B3fupFrq |
| 05:15.19 | fubada | slav3_kitten: i changed my setup to where eth2 is my default gw now, and I have a static route to eth1(phonelan), howver, im having one way audio issues |
| 05:15.22 | fubada | can you help please |
| 05:15.40 | fubada | slav3_kitten: the other way didnt work for other reasons |
| 05:17.59 | *** join/#asterisk nix8n82 (~none@65.161.180.230) |
| 05:18.56 | ChannelZ | Kraln: what version of asterisk is this? |
| 05:19.22 | Kraln | > core show version |
| 05:19.22 | Kraln | Asterisk 1.8.13.0 built by root @ voice.kraln.com on a x86_64 running Linux on 2012-06-04 23:49:51 UTC |
| 05:24.26 | ChannelZ | Is this only when you're trying to make or receive a voice call, or is it not logging in to the account at all? The debug seems to imply something incoming. |
| 05:24.36 | Kraln | not logging at all |
| 05:24.41 | Kraln | this is the initial trying to connect |
| 05:25.02 | Kraln | hmm, seems its blocking logins on meebo too |
| 05:26.08 | Kraln | maybe something else is wrong..? or maybe I fail2ban'd it |
| 05:27.47 | ChannelZ | well if that were the case here I wouldn't expect to be seeing both incoming and outgoing messages, so I doubt it |
| 05:28.02 | Kraln | no as in too many auth failures |
| 05:28.05 | Kraln | and they blocked it from everywhere |
| 05:28.23 | ChannelZ | maybe. Are you logged into that account elsewhere? |
| 05:28.27 | Kraln | no |
| 05:30.09 | Kraln | oh! |
| 05:30.10 | Kraln | hmm |
| 05:30.13 | Kraln | moment! |
| 05:31.41 | Kraln | aha! |
| 05:32.00 | Kraln | VICTORY |
| 05:32.06 | fubada | can someone please help |
| 05:32.16 | fubada | one ay audio |
| 05:32.22 | fubada | one way audio |
| 05:32.24 | ChannelZ | which way? |
| 05:32.29 | fubada | asterisk is not behind nat, phones are |
| 05:32.42 | fubada | ChannelZ: hard to say, most times i cant hear others |
| 05:32.49 | fubada | right now its both ways |
| 05:32.53 | Kraln | fubada: set your extensions to nat=yes |
| 05:33.49 | ChannelZ | is externip or externaddr (depends on your version) and localnet set in sip.conf? Peers with nat=yes ? |
| 05:33.52 | *** part/#asterisk Kalidarn (~unknown@unaffiliated/kalidarn) |
| 05:34.57 | fubada | one sec uys |
| 05:34.59 | fubada | looking |
| 05:35.17 | fubada | i use users.conf...how do i set extensions to yes |
| 05:35.22 | ChannelZ | Kraln: so what did oyu screw up? |
| 05:35.32 | Kraln | ChannelZ: me? nothing. |
| 05:35.46 | Kraln | had to change the jabber module to fix the login |
| 05:36.01 | Kraln | if you try a non-gmail google account, it barfed because it wasn't sending the right xml attributes |
| 05:36.02 | Kraln | ^.^ |
| 05:36.06 | fubada | Kraln: where do I set nat=yes please |
| 05:36.13 | Kraln | fubada: in the extension configuration |
| 05:36.25 | ChannelZ | sip.conf |
| 05:36.27 | fubada | ChannelZ: both externalip and localnat are set |
| 05:37.03 | fubada | http://pastie.org/4036002 |
| 05:37.05 | fubada | there guys |
| 05:37.07 | fubada | can you help |
| 05:37.09 | fubada | thats sip.coinf |
| 05:37.39 | fubada | randmly I will have audio, but 3/5 times i dont |
| 05:37.46 | fubada | or vice versa |
| 05:38.06 | ChannelZ | Then it's probably your phone and an issue on that side, if indeed your Asterisk is not behind NAT or running a firewall |
| 05:38.13 | fubada | im running 10.5.0 |
| 05:38.21 | fubada | i dropped iptab les |
| 05:38.31 | fubada | phone can call inter office no problem |
| 05:38.38 | fubada | i can call confb ridge etc |
| 05:39.08 | ChannelZ | Are both devices you're testing with outside the asterisk box's network? |
| 05:39.34 | *** join/#asterisk sawgood (~sawgood@173-13-158-29-sfba.hfc.comcastbusiness.net) |
| 05:39.37 | fubada | asterisk box has two interfaces; em1 is phonelan 192.168.60.x |
| 05:39.43 | fubada | and em2, is t1 direct to verizon |
| 05:39.49 | fubada | no firewall on em2 |
| 05:40.00 | fubada | * box has em2 as default gw |
| 05:40.31 | fubada | http://pastie.org/4036015 |
| 05:40.39 | fubada | route -n ^ |
| 05:42.32 | ChannelZ | Not enough info to tell what you're doing and how you're testing. You say you can "call inter office no problem" but don't say how or from what side of the network the device(s) are on as it relates to Asterisk |
| 05:42.46 | fubada | okay |
| 05:42.50 | fubada | one second please |
| 05:43.09 | ChannelZ | you have a semi-complicated network setup and I don't know what is what. You need to narrow down what part of the network is working and which part isn't |
| 05:43.10 | fubada | http://pastie.org/4036029 |
| 05:43.21 | Kraln | okay, last google talk question |
| 05:43.27 | fubada | phones are attached to em1 network 192.168.60.x |
| 05:43.36 | Kraln | if gtalk doesn't supply a DID, how can I properly handle incoming routes? |
| 05:43.45 | fubada | they talk to asterisk on 192.168.60.12 |
| 05:43.58 | Kraln | fubada: draw a picture and put it on the internet |
| 05:44.01 | Kraln | a network diagram |
| 05:44.04 | fubada | ok |
| 05:44.19 | Kraln | then use show channel to see what ip addresses each side thinks they are talking to |
| 05:44.39 | Kraln | I had this problem with a misbehaving firewall, incoming audio was being sent to some IP i'd never heard of |
| 05:44.55 | fubada | coming up man |
| 05:45.37 | fubada | iptables i dropped |
| 05:46.19 | ChannelZ | Kraln: see gtalk.conf and you can configure them to go to whatever context you want based on the user they came in on |
| 05:46.19 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 05:47.51 | ChannelZ | from there you can make extensions that match that username which are kind of like a DID |
| 05:48.05 | l2trace99 | Kraln: it is supposed to come in on <gtalk account>@context set in gtalk.conf |
| 05:48.18 | l2trace99 | but I have 1.8.11 and it has issues |
| 05:48.20 | Kraln | ah, okay, lets see. |
| 05:48.47 | l2trace99 | it fell through to s@googlein ( my context is google in ) |
| 05:48.49 | Kraln | yep, got it, perfect. |
| 05:48.51 | ChannelZ | for instance all my crap goes into a context called "incoming-gtalk" and then I have a bob exten, a joe exten, etc. |
| 05:49.06 | l2trace99 | does it work for you ? |
| 05:49.11 | Kraln | got everything working |
| 05:49.23 | l2trace99 | crap must be something with 1.8.11 |
| 05:49.35 | ChannelZ | probably not, you probably just have something wrong |
| 05:49.44 | Kraln | now I have free termination to the continental US, $1/mo unlimited incoming DID, and $0.04/m calling to my european cell phone |
| 05:49.45 | l2trace99 | I can find it |
| 05:49.49 | ChannelZ | I run 1.8.10 on one server and it's fine, so unless they broke something in .11 specifically... |
| 05:49.58 | Kraln | awesome! :) thanks guys |
| 05:50.02 | l2trace99 | there are some bug reports |
| 05:50.40 | l2trace99 | it misses the name match on the dialplan then falls through to s@googlein |
| 05:50.53 | l2trace99 | misses that then I had to set s@default |
| 05:50.56 | l2trace99 | to get it |
| 05:52.28 | fubada | Kraln: http://oi45.tinypic.com/2h5pfls.jpg |
| 05:52.33 | fubada | there it is |
| 05:52.46 | fubada | my sip.conf http://pastie.org/4036002 |
| 05:53.03 | fubada | no audio guys, so frustrating |
| 05:53.10 | fubada | iptables is disabled |
| 05:53.49 | ChannelZ | so by your diagram you say that any phones on the LAN side (em1) all communicate between each other fine |
| 05:53.55 | fubada | yes |
| 05:54.01 | fubada | and can reach * confbridge etc |
| 05:54.06 | fubada | no issues at all |
| 05:54.14 | ChannelZ | It's only when your "sip peer" gets involved that this is a problem, and presumably you're testing a call to/from the internet to one of the localnet phones |
| 05:54.24 | fubada | yes |
| 05:54.26 | fubada | exactly |
| 05:54.28 | l2trace99 | is caninvite=no ? |
| 05:54.33 | l2trace99 | for the phones ? |
| 05:54.36 | fubada | chcking |
| 05:54.38 | ChannelZ | it's directmedia now |
| 05:54.43 | Kraln | fubada: does receiving calls work, but outgoing ones not? or visa-versa? |
| 05:54.54 | fubada | receiving calls work |
| 05:55.00 | fubada | they seem to work with audio |
| 05:55.03 | fubada | only outgoing it seems |
| 05:55.11 | fubada | l2trace99: not defined |
| 05:55.16 | fubada | dont have caninvite set at all |
| 05:55.27 | l2trace99 | sry it is canreinvite = no |
| 05:55.30 | ChannelZ | it's "directmedia" now, not canreinvite |
| 05:55.31 | Kraln | need |
| 05:55.35 | Kraln | canreinvite=no |
| 05:55.36 | fubada | wait, INCOMIGN failed too |
| 05:55.40 | Kraln | directmedia=no |
| 05:55.50 | Kraln | qualify=yes |
| 05:55.53 | fubada | directmedia is set to no |
| 05:55.55 | Kraln | qualifyfreq=60 |
| 05:56.12 | Kraln | host=dynamic |
| 05:56.21 | fubada | this is all in sip.conf? |
| 05:56.25 | fubada | what about nat? |
| 05:56.32 | Kraln | nat=yes |
| 05:56.59 | Kraln | I don't see a section where you've configured individual extensions, though (I am a bit of a noob), so maybe under general? |
| 05:57.03 | ChannelZ | The remote peer you're testing is a phone/softphone, or an ITSP? |
| 05:57.12 | fubada | phone |
| 05:57.13 | fubada | polycom |
| 05:57.27 | fubada | Kraln: im usin users.conf |
| 05:57.30 | fubada | let me show you that |
| 05:58.20 | ChannelZ | turn on rtp debug and see if you're getting any RTP, and where Asterisk thinks it's sending RTP |
| 05:58.23 | fubada | http://pastie.org/4036074 |
| 05:58.32 | fubada | theres my users.conf |
| 05:58.35 | fubada | can you look at that |
| 05:58.44 | fubada | just did a test with 5 calls... 2/5 bad |
| 05:59.06 | fubada | im getting rtp but its such a flood of messages in console i cant figure whats normal and wahts not |
| 05:59.27 | fubada | it sends rtp to my phone (the phone im making a call one), and to my providers media servers |
| 05:59.49 | l2trace99 | inside to inside works ? |
| 05:59.52 | fubada | yes |
| 05:59.53 | ChannelZ | wait.. there IS an ITSP involved? |
| 06:00.04 | fubada | theres a sip provider |
| 06:00.13 | fubada | all my calls get sent to them |
| 06:00.27 | ChannelZ | you just said you were testing just with a remote polycom phone |
| 06:00.54 | fubada | im sorry i was confused, i meant im testing by placing calls from my desk polycom to my cll |
| 06:00.57 | fubada | cell |
| 06:01.11 | ChannelZ | sighs |
| 06:01.17 | ChannelZ | we need to see a sip debug of a call |
| 06:01.24 | fubada | one second |
| 06:04.03 | fubada | cant get one to fail now |
| 06:04.03 | fubada | one sec |
| 06:04.33 | ChannelZ | this sounds more like an ITSP problem |
| 06:04.42 | l2trace99 | or bandwidth |
| 06:05.29 | fubada | bandwidth is plenty |
| 06:05.38 | fubada | one second guys :P |
| 06:08.12 | tapout | what foip provider do you guys use? |
| 06:09.13 | ChannelZ | Vitelity |
| 06:09.35 | Kraln | I'm using localphone |
| 06:10.39 | l2trace99 | gafachi & voxbone |
| 06:11.04 | ChannelZ | wanders off |
| 06:12.18 | fubada | http://pastebin.ca/2158667 |
| 06:12.21 | fubada | this is the busted one |
| 06:12.22 | fubada | sip debug |
| 06:12.26 | fubada | please help:) |
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| 06:17.24 | fubada | anyone can help me figure this one out please? |
| 06:17.31 | fubada | <PROTECTED> |
| 06:20.30 | tapout | is there a way to see which wholesale provider a voip provider is using? |
| 06:20.55 | tapout | i signed up to voxbone and it tells me it won't show the pricing unless i'm going to setup a voip company that spends 500 a month lol |
| 06:21.00 | sawgood | tapout: lots of variables and sub-variables with that question (do you have a goal)? |
| 06:21.43 | tapout | well i use voip.ms, i love them. They don't support FOIP. I wanted to go up one level in that chain to see if they support FOIP (seeing how voip.ms is so cheap, i figured the wholesaler would be cheap as well) |
| 06:21.58 | sawgood | do you need T.38? |
| 06:22.05 | sawgood | or want T.38 SIP circuit |
| 06:22.26 | tapout | to be honest, i don't know what i need to get near 100% faxing to email |
| 06:22.39 | fubada | ChannelZ: yt man :p |
| 06:22.43 | sawgood | right now what process do you have in place to test/confirm? |
| 06:23.03 | Kraln | what's a good, cheap dect handset that's asterisk friendly? |
| 06:23.13 | tapout | sawgood, right now i'm using voip.ms with a 800 number to accept faxes.. it works maybe 75% of the times. it seems only the largest clients have trouble faxing :) |
| 06:23.32 | sawgood | so, your only goal is to send/receive e-faxing (via a computer with a client or a web browser GUI)? |
| 06:23.47 | tapout | kraln, my buddy got a siemens handheld, i can't remmeber what device it is exactly but he loves it |
| 06:23.56 | Kraln | gigaset? |
| 06:24.06 | Kraln | those are like five years old now :/ |
| 06:24.12 | tapout | never send, receive faxes near 100% to email from a 800$ |
| 06:24.13 | tapout | # |
| 06:24.26 | tapout | i don't think it's a gigaset tbh |
| 06:24.37 | sawgood | Are you going to put the Fax for Asterisk module on your box? |
| 06:24.55 | Kraln | something like this, but not $$$: http://www.snom.com/en/products/voip-dect-phones/snom-m9/ |
| 06:25.02 | tapout | i've got that fax setup with p3nguin's help... |
| 06:25.22 | sawgood | You might not be pleased with the Snom M3/M9 units |
| 06:25.41 | Kraln | oh? |
| 06:25.48 | sawgood | tapout: excellent ... I can give you a demo number which you can text for fax receiving via email |
| 06:26.08 | sawgood | tapout: you can use the DEMO number for practice for a week if you want (for receiving incoming e-faxes) |
| 06:26.19 | tapout | sawgood, wait .. are you a provider? |
| 06:26.24 | tapout | a 'foip' provider |
| 06:26.28 | sawgood | I am a provider |
| 06:26.30 | tapout | voip.ms said i need a 'foip' provider |
| 06:27.10 | tapout | don't set all that stuff up, show me your website |
| 06:27.28 | sawgood | if you pvt msg me, I'll help |
| 06:28.15 | tapout | t.38 = foip ? |
| 06:28.25 | tapout | when someone says you need foip, does that imply the t.38 ? |
| 06:42.00 | tapout | i think i'm going to order that snom m9r |
| 06:44.47 | Kraln | sawgood said he doesn't like them |
| 06:45.34 | fubada | ugh |
| 06:45.41 | fubada | so much frustration with these no audio issues |
| 06:46.15 | tapout | he doesn't like the m9's? .. hrmm.. what does he like? |
| 06:46.19 | tapout | sawgood, what do you like for a dect phone ? |
| 06:51.32 | Kraln | silence apparently |
| 06:52.19 | tapout | KX-TGP500 and KX-TGP550 .. have you seen those? someone messaged me those in here (thank you to who did it :) |
| 06:53.37 | awk | Snom M3/M9 is crap. |
| 06:57.35 | sawgood | The Snom M9/M3 get returned about 50% or more of the time |
| 06:57.38 | sawgood | various complaints |
| 06:58.41 | tapout | are the KX-UT670's crap? :) |
| 06:59.20 | sawgood | I don't know about that unit |
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| 07:19.07 | bulkorok | hi |
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| 07:51.47 | ChannelZ | fubada: I'm not really seeing anthing odd in terms of IP addresses.. so unless your network is randomly misrouting traffic, I'm not sure what to say other than it's an issue between you and your ITSP, or them and your cell |
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| 08:27.21 | Chainsaw | wonders why Asterisk 10.5.0 is still a secret |
| 08:27.25 | Chainsaw | Is it surprise? For me? You shouldn't have! |
| 08:27.29 | Chainsaw | +a |
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| 08:32.32 | acrg | WIMPy: you there? |
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| 08:48.23 | schmidts | hi |
| 08:59.49 | bulkorok | question: how can I reload changes in extconfig.conf seperatly without doing "reload" |
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| 09:00.57 | bulkorok | mmpf... answer: reload extconfig ... life can be so easy... |
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| 09:20.00 | aurs | my arg1 to a macro is "+${ARG2}<sip:+${ARG2}@${LOCDOM}>;privacy=off;screen=yes" but it gets cut at the first ; char. If i add \ to escape it, the \ char is added (I'm setting a sip header). Does this look familiar to anyone? |
| 09:23.32 | kaldemar | aurs: escaping and string handling may vary between versions. you could use func FILTER to remove the backslash or encode/decode the argument with BASE64_* functions. |
| 09:26.16 | bulkorok | hey guys... I want to use realtime extensions in mysql. what is the delimiter for the appdata? | is out and , seems not to work too... |
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| 09:38.57 | aurs | kaldemar, yes so it seems. Moving from 1.4 to 1.8 :) |
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| 10:38.04 | aurs | I think I will try to upgrade to the latest 1.8 before doing more on this... very weird how the backslash char is handled in some cases |
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| 11:31.20 | aurs | kaldemar, upgrading to 1.8.13.0 solved my problem... |
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| 11:56.15 | slav3_kitten | yawns an stretches out |
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| 12:22.42 | mirelab | hello, is there any chance i can set a bigger delay on Echo() app ? :) |
| 12:24.03 | [TK]D-Fender | There isn't supposed to be any additional delay |
| 12:24.18 | [TK]D-Fender | But feel free to recode it if you like |
| 12:24.34 | mirelab | hehe thx :) |
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| 12:52.19 | kozakman | hi all |
| 12:52.21 | kozakman | There is no voice on channel E1 on the incoming and outgoing calls. Alarm works. The internal SIP phones work fine. Asterisk 1.8.8, the board E1 Openvox D130P. What might be lamenting the lack of voice? |
| 12:54.05 | kozakman | there are no errors in logs |
| 12:55.43 | [TK]D-Fender | kozakman, So you've answered a call, played audio, and put them on Echo() and they heard nothing? |
| 12:57.46 | kozakman | I answer the call, and silence, nothing |
| 12:58.14 | [TK]D-Fender | Show us a test like the one I just described. |
| 12:58.38 | kozakman | Put on hold, the music still does not play |
| 12:58.50 | [TK]D-Fender | Answer, Playbe, Echo. Do these |
| 12:58.59 | [TK]D-Fender | Playback* |
| 12:59.03 | [TK]D-Fender | that order |
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| 13:01.46 | kozakman | http://pastebin.com/cZi1zCTb |
| 13:02.14 | [TK]D-Fender | koI don't trust your MoH and that is not what I told you to do. |
| 13:02.19 | [TK]D-Fender | kozakman, don't trust your MoH and that is not what I told you to do. |
| 13:06.21 | kozakman | http://pastebin.com/hMU6SJ4u |
| 13:07.09 | [TK]D-Fender | <[TK]D-Fender> Answer, Playback, Echo. Do these. That order |
| 13:07.59 | kozakman | http://pastebin.com/vigSVubp |
| 13:09.01 | [TK]D-Fender | kozakman, So you did not hear the playback at all, and nothing from the Echo() after? |
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| 13:09.15 | kozakman | nothing |
| 13:09.56 | [TK]D-Fender | kozakman, "dahdi show status", "dahdi show channels", and inclide your system.conf and chan_dahdi.conf |
| 13:10.46 | kozakman | http://pastebin.com/hm4aWegh |
| 13:11.48 | kozakman | http://pastebin.com/DhDRMGLB |
| 13:13.17 | [TK]D-Fender | Ok, that all looks pretty good... ok, do one more test call with PRI DEBUG enabled.... |
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| 13:18.40 | kozakman | http://pastebin.com/BtK2ghY1 |
| 13:21.21 | [TK]D-Fender | kozakman, add "alaw=1-31 "to your system.conf, reinitialze DAHDI and restart * |
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| 13:24.03 | kozakman | [TK]D-Fender, nothing |
| 13:25.10 | [TK]D-Fender | kozakman, Well your configs look fine, debug shows nothing out of the ordinary, you're processing the call in a way that should show results. I'd check with OpenVox at this point. |
| 13:29.44 | mirelab | Is there a posibility to play sound/recording while in Dial() app on a callee rx channel side (so that callee can hear it) |
| 13:30.41 | WIMPy | See option m. |
| 13:31.08 | mirelab | music on hold? |
| 13:31.48 | kozakman | [TK]D-Fender, thanks |
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| 13:32.12 | WIMPy | Err. You said callee? That's A, but obviousely after Dial. |
| 13:32.58 | mirelab | yeah |
| 13:33.00 | [TK]D-Fender | No, but that's obviously after being answered.... |
| 13:33.27 | WIMPy | Yes, in Dial(), but after dial. ;-) |
| 13:33.53 | [TK]D-Fender | So we take a left ... right? |
| 13:34.38 | mirelab | i can set dtmf sequence in features to do Playback but that is not heared on peer side cause it probably ends up on wrong rx/tx side |
| 13:34.50 | WIMPy | Two wrongs don't make a right, but three lefts do. |
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| 13:36.29 | NotHere | on asterisk-1.8.13.0-1_centos5 dahdi fails to load with the message WARNING[9049] loader.c: Error loading module 'chan_dahdi.s |
| 13:36.30 | NotHere | o': /usr/lib/asterisk/modules/chan_dahdi.so: undefined symbol: pri_connect_ack |
| 13:36.45 | NotHere | last working version is asterisk-1.8.12.1-1_centos5 |
| 13:36.50 | [TK]D-Fender | WIMPy, Correct |
| 13:37.35 | WIMPy | NotHere: Slap your package manager or whoever built them. Something doesn't seem to fit there. |
| 13:39.45 | NotHere | WIMPy, I will happily slap them, but at the moment I would prefer to just fix the problem |
| 13:40.16 | NotHere | WIMPy, is there a package missing which wasn't listed as a dependency? |
| 13:40.42 | WIMPy | That looks like libpri to me. |
| 13:40.51 | NotHere | (my packages come from http://packages.asterisk.org/centos/$releasever/asterisk-1.8/$basearch/ ) |
| 13:42.05 | NotHere | it worked. Thankls |
| 13:43.00 | NotHere | for the record, asterisk 1.8.13 needs at least libpri 1.4.12 (and not 1.4.11 which was previously installed) |
| 13:44.32 | WIMPy | I'm pretty sure, it doesn;t, but the version in the package seems to do so. |
| 13:45.03 | WIMPy | OTOH, I wouldn't want to use anything less than 1.4.12. |
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| 13:47.34 | slav3_kitten | WIMPy, my cisco phones power up. got a power brick to check today |
| 13:48.27 | slav3_kitten | so i'm going to try getting the skinny chan driver to work with the 7911 and 7960 i packed with me on the trip |
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| 16:04.51 | mboylan | hi guys, hoping someone can point me in the right direction here... we recently upgraded our two routing layer boxes that connect to the PRIs to 1.8 from 1.2 Business Edition. Since then, for inbound calls from the PRIs that don't have both a caller id name and a caller ID number, asterisk sets the sip header information to Anonymous@anonymous.invalid |
| 16:05.52 | mboylan | sendrpid and trustrpid on the sip trunk entries along with sendrpid enabled on my phone entry on our test server seems to fix the problem, but is there any risk to enabling sendrpid globally? |
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| 16:28.53 | pabelanger | ~book |
| 16:28.53 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
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| 16:29.36 | Qwell | mboylan: If something doesn't support rpid, it's very broken |
| 16:30.23 | mboylan | Qwell: I mean security wise |
| 16:31.40 | ThinkGNU- | #mediagoblin |
| 16:32.35 | ThinkGNU- | Sorry, that was supposed to be a join command. It wasn't meant for the channel |
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| 16:35.26 | pabelanger | Anybody know what codec's Velocity supports? |
| 16:43.47 | Nugget | does not support the "extraneous apostrophe" codec |
| 16:44.29 | Qwell | ponders |
| 16:44.34 | Qwell | telnet |
| 16:44.40 | Nugget | pbbbt |
| 16:44.42 | blizzow | One of my two PRIs went yellow this morning and dropped a bunch of calls and then people couldn't make outbound calls. I unplugged the failing PRI and everything started working again. Is there a way to disable a PRI remotely without having to physically unplug the thing? |
| 16:44.44 | Qwell | boo, can't even get to the next test |
| 16:44.54 | Qwell | Nugget: How do you feel about netcat + GAPING_SECURITY_HOLE? |
| 16:45.33 | blizzow | I see I can disable a PRI channel, but I wanted to disable the failing PRI span. |
| 16:45.49 | Nugget | Qwell: just use inetd to bash :) |
| 16:46.01 | Qwell | Nugget: now you're thinking |
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| 16:50.33 | [TK]D-Fender | blizzow, Remove the channel configs for it and reinit DAHDI |
| 16:50.45 | blizzow | Thanks. |
| 16:51.49 | blizzow | [TK]D-Fender: I called my phone company and they say their circuit tests clean to the demarc and claim it's my equipment. The makers of my tdmoe box say their stuff is fine. Is there any way * could cause the yellow alarm? |
| 16:51.58 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 16:53.13 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-86-140.user.veloxzone.com.br) |
| 16:53.13 | coppice | blizzow: yes there is - get the CRC setting wrong, or have faulty equipment |
| 16:55.41 | blizzow | The thing has been up and running for nearly a year so I don't think it's a CRC setting issue. I did upgrade dahdi a couple of weeks ago, but it ran for 11 days with no issues and now I've got two service drops within 5 days. |
| 16:55.43 | anonymouz666 | coppice: what about ilbc from google into asterisk? |
| 16:55.48 | *** join/#asterisk GeoGeek (~steve-o_@12.71.122.227) |
| 16:56.10 | coppice | what about it? |
| 16:56.17 | *** join/#asterisk philipp64|laptop (~chatzilla@66-232-81-197.ctcweb.net) |
| 16:56.39 | *** part/#asterisk GeoGeek (~steve-o_@12.71.122.227) |
| 16:56.54 | anonymouz666 | coppice: don't you think that google can deny somehow the code? |
| 16:58.15 | coppice | I can't imagine they want to. the fact they put fixed point iLBC and both fixed and floating iSAC in webrtc may imply they think they've already put floating iLBC out there in an acceptable form. |
| 16:58.15 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 16:59.03 | *** join/#asterisk resist0r (~resist0r@69.31.131.51) |
| 16:59.28 | coppice | google are *extremely* sloppy about this kind of thing |
| 17:01.34 | coppice | they put some of my stuff in webrtc. in some cases they took my header off and put their own copyright notice on my work. its nothing too important, but it shows how they behave |
| 17:02.11 | Qwell | coppice: if you can point to a specific file, I can pass it along to a friend |
| 17:02.31 | Qwell | It's his job to deal with issues like that. |
| 17:03.40 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 17:04.55 | coppice | Qwell: does he deal with the missing stuff in webrtc as well? some of the codec test data files are missing |
| 17:05.09 | Qwell | coppice: no, his title is something like "Open Source Advocate" |
| 17:05.22 | Qwell | or, works in that dept, or whatever |
| 17:05.51 | Qwell | anyways, he deals with that kind of licensing stuff all the time. |
| 17:07.11 | *** join/#asterisk Bullmoose (~Bullmoose@65-129-14-184.bois.qwest.net) |
| 17:07.50 | anonymouz666 | coppice: your work is LGPL? |
| 17:08.16 | coppice | sometimes :-) |
| 17:08.22 | Qwell | coppice: If you care enough about the headers, feel free to shoot me an email, and I can forward it along. He'd certainly want to be aware of it, at least. |
| 17:25.25 | *** join/#asterisk blitzrage (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:25.25 | *** mode/#asterisk [+o blitzrage] by ChanServ |
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| 18:10.36 | *** join/#asterisk LoboX (~mapineyro@190.167.210.137) |
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| 18:12.30 | LoboX | hi all i have a quick question, can i make in asterisk a call to all my ext, and everybody listen to what im saying from one phone? |
| 18:13.20 | LoboX | if i want to make a announcement to everybody in the office |
| 18:13.40 | LoboX | is that possible? |
| 18:14.23 | malcolmd | app_page |
| 18:15.23 | LoboX | thanks malcolmd let me check thaty |
| 18:15.29 | LoboX | that* |
| 18:23.16 | *** join/#asterisk rossand (~aross@CPE009400809a9c-CM78cd8ed45eb5.cpe.net.cable.rogers.com) |
| 18:25.31 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 18:31.45 | Kraln | hmm, does asterisk support video too? |
| 18:31.54 | *** join/#asterisk last1 (~dood@modemcable153.206-57-74.mc.videotron.ca) |
| 18:33.16 | [TK]D-Fender | somewhat |
| 18:33.43 | Kraln | <PROTECTED> |
| 18:33.52 | [TK]D-Fender | "follow the talker" in app_confbridge. single acller to caller video, and video voicemail. |
| 18:34.05 | Kraln | thanks |
| 18:34.08 | [TK]D-Fender | mieux? |
| 18:34.18 | last1 | whenever I stop an audio file from playing by pressing an option, the audio file stops with a cracking noise or something |
| 18:34.48 | last1 | for example, the audio might say: "Press 1 if you wish to continue" and if I press 1 mid-audio, I might hear: Press 1 <crackling> |
| 18:34.54 | last1 | how do I remove that noise ? |
| 18:34.57 | Kraln | [TK]D-Fender: oui, merci |
| 18:47.17 | *** join/#asterisk rgagnon (~rgagnon@rrcs-71-42-183-54.sw.biz.rr.com) |
| 18:47.19 | LoboX | malcolmd got it working >:) |
| 18:47.22 | LoboX | thank you |
| 18:47.52 | rgagnon | FYI for web people at digium... http://www.asterisk.org/node/51985 is a dead/invalid link on your asterisk homepage |
| 18:48.02 | rgagnon | its the news page for 10.5.0 |
| 18:49.04 | Qwell | mjordan: ^ |
| 18:53.10 | mjordan | rgagnon: fixed, thanks for bringing it to our attention :-) |
| 18:53.22 | rgagnon | np. thanks for the quick fix |
| 18:53.35 | *** part/#asterisk rgagnon (~rgagnon@rrcs-71-42-183-54.sw.biz.rr.com) |
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| 19:12.00 | *** join/#asterisk timahvo1 (~rogue@41.81.94.89) |
| 19:44.58 | malcolmd | LoboX: yay :D |
| 19:49.08 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 19:50.02 | fubada | is it possible to disable the voicemail "chirp" noise with asterisk? |
| 19:50.07 | fubada | and also the stutter vm dialtone |
| 19:50.12 | fubada | or is this a phone side setting |
| 19:51.27 | *** join/#asterisk gusto (~gusto@88.128.70.77) |
| 19:52.04 | [TK]D-Fender | Chirp = phone. Stutter... well.. what kind of phone? |
| 19:52.09 | fubada | polycoms |
| 19:52.56 | [TK]D-Fender | ALL phone. |
| 19:52.59 | fubada | thanks |
| 19:59.40 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 20:04.13 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:10.30 | *** join/#asterisk timahvo1 (~rogue@197.176.161.198) |
| 20:13.22 | philipp64|laptop | how do I override the user@ portion of the URI in an outgoing SIP call from within the the dialplan? |
| 20:14.04 | [TK]D-Fender | Should be based off the callerid |
| 20:14.36 | fubada | [TK]D-Fender: what does it suggest when 1/3 calls ha one-way audio issues |
| 20:14.37 | fubada | ? |
| 20:14.46 | fubada | s/ha/have |
| 20:15.02 | fubada | and can I run my sip.conf by you sir |
| 20:15.12 | [TK]D-Fender | Suggestes you misconfigured something and a few of your calls arrived at a time while ports were temporarily mapped right to survive |
| 20:17.09 | fubada | http://i45.tinypic.com/2h5pfls.jpg <- my asterisk box network diag, http://pastebin.ca/2158835 <- my sip.conf |
| 20:17.41 | fubada | could you help me figure out if ym sip.conf matches my network setup. im confused about nat, and direct media |
| 20:18.08 | fubada | SIP peer is really my ITSP |
| 20:24.19 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 20:31.48 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 20:33.11 | *** join/#asterisk smooth_penguin (~jr@115.69.254.194) |
| 20:37.11 | *** join/#asterisk MJCS (~script@ip68-5-48-206.oc.oc.cox.net) |
| 20:39.12 | MJCS | what is the proper way to send an incoming call to an extension. I did dial(sip/1001) for example but when I answer it keeps ringing on the person dialing in. If I place answer() before it, it works but I dont really want asterisk doing that. |
| 20:40.59 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 20:43.03 | Kraln | time to get some college kid in trouble... http://kraln.com/abuse_msu.txt |
| 20:43.05 | jaytee | MJCS, in 99% of cases you want to have Answer() before you use the Dial app for incoming calls. |
| 20:43.35 | fubada | lol |
| 20:43.38 | MJCS | yeah but that looks like you are answering the call on the other phones |
| 20:44.03 | MJCS | instead of it ringing, on the extension and when you pick up they pick up |
| 20:44.49 | pabelanger | jaytee: Is that your box? |
| 20:45.27 | jaytee | @pabelanger, my box? huh? |
| 20:45.28 | pabelanger | jaytee: wrong person |
| 20:45.32 | jaytee | ok |
| 20:45.35 | pabelanger | Kraln: is that your box? |
| 20:45.52 | Kraln | logs from my machine, yeah |
| 20:45.56 | Kraln | someone wardialing from michigan state |
| 20:46.03 | pabelanger | you deserve to get hacked |
| 20:46.16 | pabelanger | you AMI is publicly exposed |
| 20:46.39 | pabelanger | telnet 69.169.91.15 5038 |
| 20:46.39 | MJCS | lulz |
| 20:46.56 | fubada | Connected to 69.169.91.15. |
| 20:46.56 | fubada | Escape character is '^]'. |
| 20:46.56 | fubada | Asterisk Call Manager/1.1 |
| 20:46.57 | fubada | total |
| 20:47.52 | MJCS | is there a better way to do this or is Answer() then dial() the only way |
| 20:48.06 | Kraln | anything else I should block? :P |
| 20:50.47 | MJCS | it also seems like I can only answer after the 2nd ring |
| 20:51.15 | gusto | what is this asterisk call manager? is it enabled by default? |
| 20:51.32 | MJCS | brb my colon is about to go boom |
| 20:51.40 | jpsharp | <PROTECTED> |
| 20:51.55 | fubada | UGH dude Polycom XML documentation is the worst |
| 20:52.07 | fubada | how the hell am i supposed to configure these phones for provisioning |
| 20:52.15 | pabelanger | Kraln: atleast you are running the latest and greatest: User-Agent: FPBX-2.10.0(1.8.13.0) |
| 20:52.24 | pabelanger | but you need to take that box off the public web |
| 20:52.24 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 20:52.47 | jpsharp | But it is XML, therefore cool! |
| 20:53.09 | Kraln | pabelanger: firewalled that port off, unloaded skinny and mgcp and dundi. anything else I need to do to secure this? |
| 20:53.14 | fubada | maybe someones done this in here, how the hell do i disable mwi stutter and chrip on polycoms |
| 20:53.18 | pabelanger | fubada: I wrote a puppet module to do it, and it works like a champ |
| 20:53.23 | fubada | dude i use puppet |
| 20:53.24 | fubada | share |
| 20:53.37 | fubada | whats the purpose of your module |
| 20:54.00 | pabelanger | fubada: https://github.com/kickstandproject/puppet-modules |
| 20:54.18 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 20:54.18 | pabelanger | purpose, to fully provision and manage an asterisk box using puppet |
| 20:54.23 | fubada | oh, im using phoneprov |
| 20:54.29 | fubada | oh nice |
| 20:54.30 | pabelanger | don't |
| 20:54.33 | pabelanger | :D |
| 20:54.45 | fubada | well phoneprov is working :) heh but puppet would be nicer |
| 20:55.09 | fubada | pabelanger: do you know how to disable vm stutter and chirp on polys |
| 20:55.11 | pabelanger | Yup, it is pretty slick once you get it working |
| 20:55.37 | pabelanger | stutter? Sounds like a bandwidth or CPU issue on your asterisk box |
| 20:55.42 | pabelanger | are you transcoding audio? |
| 20:55.50 | jaytee | he means stutter dial tone |
| 20:55.55 | fubada | nono i man when you pickup the phone and theres a voicemail |
| 20:55.58 | fubada | the tone jumps |
| 20:56.03 | fubada | aka stutter tone |
| 20:56.09 | pabelanger | Oh |
| 20:56.11 | jaytee | default setting on polycoms |
| 20:56.11 | fubada | s/man/mean |
| 20:56.13 | pabelanger | what firmware you using? |
| 20:56.14 | Kraln | pabelanger: any other advice, though? |
| 20:56.16 | fubada | latest |
| 20:56.18 | jaytee | when there's voicemail |
| 20:56.20 | fubada | 4.0.4 |
| 20:56.35 | pabelanger | Kraln: only that I would not leave that box on the public web |
| 20:56.47 | Kraln | define 'leave box on public web' |
| 20:57.09 | fubada | your asterisk mdule is sick |
| 20:57.11 | pabelanger | Is it behind a firewall |
| 20:57.17 | fubada | but itd be difficult to implement |
| 20:57.21 | Kraln | pabelanger: yes, it is behind a firewall |
| 20:57.43 | pabelanger | fubada: I'm using bootrom 4.1.4 and 3.3.4 release |
| 20:58.00 | fubada | oh, i sycnd to latest |
| 20:58.02 | pabelanger | fubada: well, it is there for people to use |
| 20:58.29 | Kraln | pabelanger: what needs to be blocked, etc. |
| 20:58.34 | fubada | i like your puppet skills, i can use this to improve mine |
| 20:58.43 | pabelanger | Had an issue with another site, the phone would not generate ringing for incoming calls, upgrade to the latest, seems to have fixed the problem |
| 21:00.14 | fubada | how long have you been doing puppet? |
| 21:00.16 | fubada | really nice layout |
| 21:00.35 | fubada | you dont do chaining but i guess its not needed |
| 21:00.43 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 21:01.32 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:01.44 | pabelanger | Kraln: you want to fix this: http://pastebin.com/6TYXEUgH |
| 21:01.56 | pabelanger | fubada: about 6months |
| 21:02.12 | pabelanger | Ya, I have not run into chaining issue... yet |
| 21:02.23 | fubada | impressive...whats this "::monitor" stuff |
| 21:02.24 | pabelanger | I've consider implementing it but haven't needed to |
| 21:02.33 | fubada | every module has 'monitor' |
| 21:02.34 | Kraln | pabelanger: I only see port 80 open? |
| 21:02.52 | pabelanger | fubada: yup, for setting up nagios for each service |
| 21:02.57 | fubada | ah ok |
| 21:03.00 | anonymouz666 | what's the most common name in north america? "contact center" or "call center"? for a system name |
| 21:03.28 | pabelanger | Kraln: look again, you have netbios exposed |
| 21:03.37 | Kraln | it's a linux machine |
| 21:03.38 | pabelanger | unless that is your ISP |
| 21:03.57 | fubada | anonymouz666: call center |
| 21:04.05 | Kraln | pabelanger: that's not even running |
| 21:04.20 | pabelanger | so must be your ISP |
| 21:04.43 | Kraln | my isp is transparent |
| 21:04.51 | Kraln | that's nmap telling you ports aren't doing anything |
| 21:04.53 | Kraln | http://pastebin.com/e0ZT9AFt |
| 21:07.07 | pabelanger | Kraln: well, that fact you still have freepbx publicly exposed is still an issue |
| 21:07.26 | Kraln | about to firewall that off to my vpn |
| 21:07.29 | Kraln | anything else? |
| 21:09.08 | pabelanger | Kraln: http://69.169.91.15/recordings/misc/callme_page.php?callmenum=0038095&action=c |
| 21:09.20 | pabelanger | a simple exploit that could happen |
| 21:09.39 | Kraln | alright, point taken |
| 21:10.52 | pabelanger | Ya, I don't want to be a dink. But, people can do some damage to your asterisk box if it is exposed. Especially if you start adding freepbx and other pre-bundled software that some distros use |
| 21:11.04 | Kraln | no, I appreciate it |
| 21:11.05 | *** join/#asterisk GeoGeek (~steve-o_@12.71.122.227) |
| 21:11.13 | Kraln | I'm just curious what I need to filter and what needs to be not filtered. |
| 21:12.13 | *** join/#asterisk JamKo (~JamKo@173-160-6-202-naples.hfc.comcastbusiness.net) |
| 21:12.47 | Kraln | seems the answer is "the system is swiss cheese, don't let anyone touch anything" |
| 21:15.37 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 21:18.55 | *** join/#asterisk tfgtech (~tfgtech@232.Red-80-39-240.dynamicIP.rima-tde.net) |
| 21:23.00 | *** join/#asterisk tfgtech (~tfgtech@232.Red-80-39-240.dynamicIP.rima-tde.net) |
| 21:23.53 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 21:25.20 | tfgtech | i am using 1.8.13 on Centos 5.5 , and i have a sever issue with attended transfers creating ZOMBIE channels, leading to the caller and the person receiving the call to think the call has been dropped, so they both disconnect |
| 21:25.48 | tfgtech | this issue seems to be for older versions of asterisk, but somehow i have it in my install |
| 21:25.52 | Kraln | pabelanger: alright, I believe I have firewalled it so that the web interface is only available via vpn |
| 21:25.59 | Kraln | which means there are now no exposed tcp ports |
| 21:26.12 | Kraln | so, should be good? |
| 21:26.20 | pabelanger | yup |
| 21:26.21 | tfgtech | can anyone point me in the right direction_ no idea how to fix this, and i can´´t fina patch for my version |
| 21:26.39 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 21:30.01 | *** join/#asterisk uskerine (~uske@58.Red-88-1-239.dynamicIP.rima-tde.net) |
| 21:30.17 | uskerine | hi, i am looking for recommendations on softphone clients (light ones), both windows and linux |
| 21:30.37 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-yqdfqbggdkyfxqsp) |
| 21:31.01 | tfgtech | linuxÑ linphone, ekiga |
| 21:31.07 | Kraln | uskerine: I'm happy with both zoiper and 3cx |
| 21:31.30 | uskerine | let's have a look on those ones |
| 21:31.53 | tfgtech | zoiper is good, x-lite a bit more cumbersome |
| 21:31.55 | WIMPy | likes Zoiper for the small visual footprint, but the Windows version has issues with packetization. |
| 21:32.12 | uskerine | i downloaded x-lite and did not like that it wants to install a lot of c++ stuff |
| 21:38.39 | agisamentum | likes zoiper for it's simplicity |
| 21:39.29 | agisamentum | sometimes the gui craps up, but you can just text edit or sed the config file under .zoiper/ and voila |
| 21:39.46 | uskerine | is there any softphone for linux i can configure so it uses a given extension (sip username) from a config file so I can fix the configuration for each user? |
| 21:40.13 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 21:40.15 | MJCS | anyways back to dialing extensions. Is the only way to forward a call to an extension by "Answer()" then "Dial()"? or is there another way? |
| 21:40.28 | WIMPy | uskerine: I'd assume, they all do. |
| 21:40.50 | WIMPy | MJCS: Yes. skip the Answer(). |
| 21:40.57 | jpsharp | You shouldn't have to "answer" explicitly before a Dial. The channel gets answered if the Dial()ed channel answers. |
| 21:43.31 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 21:46.33 | MJCS | something is strange here because I have to wait for the 2nd ring to answer or else the person on the other line just keeps ringing |
| 21:50.02 | uskerine | what "exten => 10,1,Verbose(1|Echo test application) |
| 21:50.04 | uskerine | actually means? |
| 21:50.38 | [TK]D-Fender | Means it'll print a line in the console... if you're at verbose 1 or higher |
| 21:50.42 | WIMPy | MJCS: You shouldn't Answer() at all. And what do you answer and what "other line"? |
| 21:51.02 | WIMPy | uskerine: It means your Dialplan was for Asterisk 1.2 or older. |
| 21:51.13 | [TK]D-Fender | That too |
| 21:51.56 | uskerine | <PROTECTED> |
| 21:52.05 | uskerine | i guess that is what you mean |
| 21:52.28 | WIMPy | indeed |
| 21:52.47 | uskerine | this command, (then there is Echo and Hangup) |
| 21:52.55 | uskerine | under the [internal] |
| 21:53.14 | uskerine | means that the command sequence will be applied to incoming calls for this extension or for outgoing calls from this extension? |
| 21:53.30 | WIMPy | TO |
| 21:53.40 | WIMPy | Always and only to. |
| 21:53.43 | uskerine | ok |
| 21:53.59 | [TK]D-Fender | uskerine: that is STEP ONE in processing "10" in that context |
| 21:54.06 | WIMPy | If you want do take different routes, depending on where a call comes from, use contexts. |
| 21:54.42 | uskerine | i am just trying to understandhow this works (i am reading asterisk, the future of the telephony |
| 21:54.45 | uskerine | and doing some tests |
| 21:56.42 | *** join/#asterisk sekil (~sekil@78.187.94.151) |
| 21:58.11 | *** join/#asterisk flyingbull (~Adium@cpe-065-190-148-230.nc.res.rr.com) |
| 21:58.55 | flyingbull | Hi everyone:) I hope everyone is doing well today. |
| 22:00.36 | flyingbull | I got a question, I'm working with VoiceMailMain() and basically I want the user to type in *97, and then Voicemail main should prompt them for their password. I can't seem to get it to work, it prompts for both their Extension and Password. Which brings up another problem, I suspect that that it isn't going into the correct context. |
| 22:01.03 | flyingbull | I think it is going to default. |
| 22:01.06 | WIMPy | Well, you have to supply the mailbox number. |
| 22:01.39 | uskerine | so exten=> 10,1,Echo() |
| 22:01.47 | uskerine | exten => 10,n,Hangup() |
| 22:01.57 | uskerine | should echo my voice when i call extension 10? |
| 22:02.33 | WIMPy | yes |
| 22:02.33 | [TK]D-Fender | flyingbull: You should probably show us the dialing you made to go toVMM |
| 22:02.51 | uskerine | it does not work |
| 22:02.56 | [TK]D-Fender | It shouldn't |
| 22:02.58 | [TK]D-Fender | ANSWER FIRST |
| 22:03.00 | [TK]D-Fender | ^ |
| 22:03.30 | flyingbull | Fender, currently I have it going straight to VoiceMailMain() |
| 22:03.48 | flyingbull | exten => *97,1,VoiceMailMain(${) |
| 22:03.54 | WIMPy | Echo() doesn't answer? |
| 22:04.00 | flyingbull | oh sorry, that is typo LOL |
| 22:04.06 | flyingbull | exten => *97,1,VoiceMailMain() |
| 22:04.35 | uskerine | it works with Answer, thjanks [TK]D-Fender |
| 22:04.37 | uskerine | :) |
| 22:04.40 | WIMPy | flyingbull: I don't see a mailbox number in those brackets. |
| 22:05.06 | WIMPy | If you don't give it a mailbox to work on it will have to ask the user.s |
| 22:05.11 | [TK]D-Fender | flyingbull: yYou have to tell it what mailbox on the app line |
| 22:05.48 | flyingbull | I understand that, my point is how can I grab from the sip channel that information? Or do I for each person have to create a seprate line for each extenstion? |
| 22:06.19 | [TK]D-Fender | flyingbull: There are all sorts of nifty Cchannel variables you should read up on... |
| 22:06.23 | WIMPy | How do you assign mailbox numbers? |
| 22:06.33 | [TK]D-Fender | flyingbull: including ${CHANNEL} |
| 22:06.47 | [TK]D-Fender | flyingbull: And there are other identifying possibilities.. like the CALLERID() |
| 22:07.05 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-imghhyoefifxyqer) |
| 22:07.09 | [TK]D-Fender | flyingbull: Or other means... like setting a channel var in your sip peer with SetVar= |
| 22:07.54 | WIMPy | SetVar is a nice thing, but unfortunatly limited to certain channeltypes. |
| 22:09.05 | flyingbull | Interesting. I never noticed that before. |
| 22:09.08 | *** join/#asterisk d_preston215 (~chatzilla@50-73-214-237-philadelpia.hfc.comcastbusiness.net) |
| 22:09.22 | flyingbull | I think I'll play with the SetVar, it makes more sense for me to do it there…. |
| 22:09.26 | uskerine | i have answer/playback/hangup sequence in [default] with "s" as extension number |
| 22:09.30 | d_preston215 | Can I use a call file to enter into a conference? |
| 22:09.49 | uskerine | is that default sequence for unknown extensions? |
| 22:10.11 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-ykdnuecmqlhhwskv) |
| 22:10.16 | WIMPy | uskerine: No it's for calls without destination. |
| 22:10.21 | WIMPy | d_preston215: Sure |
| 22:10.48 | uskerine | what should you do to playback something when calling unknown extensions from internal phones? |
| 22:11.24 | uskerine | and Wimpy how a call can have no destionation? |
| 22:11.28 | WIMPy | There is the i extensions for invalid. |
| 22:11.31 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 22:11.36 | [TK]D-Fender | ~stdextens |
| 22:11.37 | infobot | [~stdextens] The "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), or from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf. "s" is also used to make IVRs & macros. |
| 22:11.39 | [TK]D-Fender | ^^^ |
| 22:11.59 | WIMPy | Ha, I think I've never seen that one. |
| 22:12.22 | [TK]D-Fender | I got tired of hand typing it years ago |
| 22:12.40 | WIMPy | uskerine: There are many ways, e.g. callig sip:@your.host or for classic phones, just lifting the handset. |
| 22:12.54 | [TK]D-Fender | All of the [~....] botlets were originally my creation to standardise the info |
| 22:13.36 | uskerine | ~iextens |
| 22:13.55 | uskerine | ~invextens |
| 22:13.57 | [TK]D-Fender | uskerine: I only made that one |
| 22:13.59 | WIMPy | But that's only the s extension. A mention of all special extensions might be nifty. |
| 22:14.04 | uskerine | :) |
| 22:14.19 | [TK]D-Fender | uskerine: "i" will not match an incoming call that doesn't match anything else |
| 22:14.30 | [TK]D-Fender | 'i" only works in IVR's |
| 22:14.41 | [TK]D-Fender | a non-matching SIP call will error out with a 4040 |
| 22:14.44 | [TK]D-Fender | 404* |
| 22:14.48 | WIMPy | No, i also works for calls. |
| 22:15.06 | uskerine | so if i want to playback something when an extension dials an extension which does not exist? |
| 22:15.11 | uskerine | what do i have to do? |
| 22:15.13 | WIMPy | Not for SIP, but for other channeltypes. |
| 22:15.14 | [TK]D-Fender | WIMPy: Nope.... |
| 22:15.26 | [TK]D-Fender | Other channels? Like? not PRI... |
| 22:15.32 | [TK]D-Fender | shocertainly shouldn't |
| 22:15.37 | [TK]D-Fender | I'd love to see otehrwise... |
| 22:16.06 | WIMPy | PRI should do it, yes. |
| 22:16.23 | WIMPy | uskerine: You can do a default extension like "_X.". |
| 22:17.12 | fubada | does anyone know kbps rates for ulaw? |
| 22:17.32 | uskerine | exten => 1000,n,Dial(SIP/1000,30) |
| 22:17.33 | WIMPy | 64kbps + overhead |
| 22:17.39 | d_preston215 | Local/### or SIP/### to do so? |
| 22:17.41 | uskerine | what does that mean? |
| 22:17.49 | WIMPy | For more info ask google for a voip bandwidth calculator. |
| 22:18.17 | fubada | thnaks man |
| 22:30.12 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 22:34.25 | *** join/#asterisk blitzrage (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:34.25 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 22:41.24 | *** join/#asterisk sawgood (~sawgood@173-13-158-29-sfba.hfc.comcastbusiness.net) |
| 22:54.23 | sawgood | $100 dollar instant bounty paid for an acceptable solution to this: http://filebin.ca/4PUkvBiqrpo |
| 22:56.21 | WIMPy | sawgood: You should tell your webserver that this is a pdf file. |
| 22:57.01 | flyingbull | Damn, I can't believe I just clicked on that link. |
| 22:57.43 | sawgood | I created the .PDF (I'll fix that) |
| 22:59.55 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 23:25.53 | *** part/#asterisk flyingbull (~Adium@cpe-065-190-148-230.nc.res.rr.com) |
| 23:27.36 | *** join/#asterisk luka12345 (~luka12345@unaffiliated/luka12345) |
| 23:28.12 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-vecgchmvzxoxaibn) |
| 23:29.53 | luka12345 | hi all |
| 23:30.16 | luka12345 | can anybody tell me how to turn debug messages only for specific part of the code - ie asterisk channel? |
| 23:30.46 | WIMPy | Wait for a version that can do it. |
| 23:31.58 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-132-58.ks.ks.cox.net) |
| 23:32.02 | luka12345 | ok |
| 23:32.49 | luka12345 | i need it because i'm writing new channel driver, and i would like only messages i'm interested in to get printed on screen |
| 23:33.14 | luka12345 | here is the code http://git.nanl.de/?p=asterisk_channel_lantiq.git;a=summary |
| 23:33.27 | luka12345 | any tips for this? |
| 23:34.15 | WIMPy | What is lantiq? |
| 23:35.34 | luka12345 | it's company that makes chips for routers |
| 23:36.00 | luka12345 | http://www.lantiq.com/ |
| 23:36.06 | WIMPy | So what does that channel connect to? |
| 23:36.31 | luka12345 | to kernel module that does all the heavy lifting |
| 23:36.36 | luka12345 | it's called tapi |
| 23:37.18 | luka12345 | we only use exposed functions for stuff like ringing, tones, etc... |
| 23:37.35 | luka12345 | btw. i run openwrt on it ;) |
| 23:38.06 | WIMPy | Err. I get to iis.net. That doesn't seem to fit routers. |
| 23:38.51 | WIMPy | Oops. accidentally removed the l when adding the www... |
| 23:47.20 | *** join/#asterisk nsgn (~nsgn@rrcs-108-178-100-46.sw.biz.rr.com) |
| 23:47.59 | nsgn | howdy. is there any way when an agent dials to log into a queue dynamically that we can have asterisk bypass asking for their extension and juse use the extension they dialed from? |
| 23:48.34 | nsgn | the boss here is asking why we can't just have it be 200* and 200** to log in and out, instead of the process of being asked what phone you're at when everyone is always at the same phone we assigned them |
| 23:55.30 | jpsharp | You can map SIP phones directly as Queue members and then pause/unpause agents from the dialplan with 200* and 200** |
| 23:56.00 | nsgn | as static agents? |
| 23:59.19 | nsgn | jpsharp, mind expanding on that a bit? i tried adding the extensions as static agents and it still asks the same question when i do 200* |
| 23:59.33 | nsgn | is mapping queue members somehow different than making them static agents? |
