| 00:07.48 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 00:36.24 | *** join/#asterisk DaPrivateer (~matt7229@71-9-155-174.static.oxfr.ma.charter.com) |
| 00:37.42 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 01:15.06 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 01:27.48 | *** join/#asterisk luckman212_ (~irc@2001:470:8abb:0:4cb5:ce24:38d4:da24) |
| 01:28.09 | *** join/#asterisk luckman212__ (~irc@2001:470:8abb:0:dc71:d10e:5b7e:4865) |
| 01:29.08 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 01:59.26 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 02:14.03 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 02:34.13 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 02:46.00 | raden | anyone know how to charge ringer on polycom ? |
| 02:47.48 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 02:49.30 | *** join/#asterisk YoMomma (~YoMomma@cpe-98-149-120-227.socal.res.rr.com) |
| 02:58.02 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 03:13.37 | *** join/#asterisk Nemus (~Nemus@c-76-23-30-50.hsd1.ut.comcast.net) |
| 03:18.59 | ChannelZ | insert coin to continue |
| 03:32.10 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 03:41.21 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 03:46.20 | *** join/#asterisk DarthExpeditor (~IceChat9@173.46.183.185) |
| 03:49.20 | raden | inserts coin |
| 04:15.01 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 04:26.24 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 04:36.27 | *** join/#asterisk slav3_kitten (~kitten@unaffiliated/slav3-kitten/x-0866809) |
| 04:51.43 | *** join/#asterisk amigamia (~sambagirl@unaffiliated/sambagirl) |
| 05:21.17 | *** join/#asterisk resist0r (~resist0r@69.31.131.51) |
| 05:26.44 | *** join/#asterisk Nemus (~Nemus@c-76-23-30-50.hsd1.ut.comcast.net) |
| 05:31.59 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 05:37.10 | *** join/#asterisk resist0r (~resist0r@69.31.131.51) |
| 06:17.32 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 06:26.40 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 06:42.04 | *** join/#asterisk jmls1 (~julian@host217-36-208-155.in-addr.btopenworld.com) |
| 06:42.10 | jmls1 | morning all |
| 06:42.22 | jmls1 | <PROTECTED> |
| 06:42.28 | jmls1 | is this an attack ? |
| 06:42.37 | jmls1 | what confuses me is |
| 06:42.58 | jmls1 | 1) I have a firewall in place to only allow udp 5060 access from a single ip (my provider) |
| 06:43.25 | jmls1 | 2) there is only the 5550000 attempt recorded in the logs. |
| 06:44.02 | jmls1 | 3) why does it have the external address of my router ? Is it the router trying to register something ? |
| 06:54.38 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 07:25.59 | kaldemar | jmls1: that's most likely an incoming call. you don't have a matching peer that would allow calls from the provider with no authentication. |
| 07:29.08 | kaldemar | your address is in the request because that's the address your provider has after you register to them. define a peer in sip.conf with type=peer, host=<their_ip>, insecure=port,invite. |
| 07:31.47 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 07:37.35 | ChannelZ | is 55500000 what you replaced or is that literally the number it says? |
| 07:49.36 | *** join/#asterisk Nemus (~Nemus@c-76-23-30-50.hsd1.ut.comcast.net) |
| 08:13.59 | jmls1 | ChannelZ: that's the actual entry |
| 08:14.33 | jmls1 | kaldemar: I do have a peer with host=<their_ip> |
| 08:33.37 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 08:40.02 | ChannelZ | does it have a secret? |
| 09:03.37 | *** join/#asterisk libryder (~libryder@libryder.com) |
| 09:06.09 | ChannelZ | actually it's just not matching your peer I guess, else you'd get an outright rejection |
| 09:19.13 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 09:25.24 | Starvirus | could someone help me, i'm looking for some extension where my clients ring while the caller can enter a pin - but invisible...if the pin is right i want the pbx to send back a dtmf code |
| 09:26.24 | ectospasm | Starvirus: what do you mean by "invisible?" |
| 09:26.53 | Starvirus | there shouldn'T be some text "please enter your pin" |
| 09:27.02 | ectospasm | ...you want your clients to ring *while* the caller is entering a DTMF pin? |
| 09:27.09 | Starvirus | for the caller it should just be as would the phones ring |
| 09:27.44 | ectospasm | You can use the Read() app to receive the PIN |
| 09:27.46 | Starvirus | yeah, the caller should be able to enter a dtmf pin, but it's not necessary and the clients should ring |
| 09:28.06 | Starvirus | yeah but the client's wouldn't ring as far as i understood read |
| 09:28.51 | ectospasm | so you want the PIN read to happen at the same time the clients are ringing... |
| 09:28.59 | Starvirus | yes |
| 09:29.20 | Starvirus | and only if the pin is right jump out of the "normal" calling process and do something else |
| 09:30.00 | ectospasm | well, you could dump your clients into a queue, and configure the queue to have the caller jump out to an extension if they dial it |
| 09:30.26 | ectospasm | but afaik you'd have to have only one such PIN per queue. |
| 09:31.01 | Starvirus | well that wouldn't be the problem, i need only one pin, so that would be fine |
| 09:31.18 | Starvirus | hm...ok, guess i have to look into queues than |
| 09:32.27 | Starvirus | thanks |
| 09:32.35 | ectospasm | Starvirus: looks like it can only be a single digit extension... |
| 09:32.59 | Starvirus | hm...ok that would be unsecure |
| 09:34.25 | ectospasm | ...well, in the dialplan context jumped to you could do a silent Read() for the rest of the digits, without playing an announcement or tone. |
| 09:36.28 | ectospasm | you could even have Background() play a ringback tone, so the caller wouldn't know they're entering a PIN |
| 09:37.01 | ectospasm | I can see at least two ways to solve this |
| 09:37.04 | Starvirus | the question is would the clients keep ringing and could they catch the call |
| 09:37.30 | ectospasm | no, because they would have jumped out of the queue |
| 09:37.40 | ectospasm | what do you intend to happen when they dial the PIN? |
| 09:38.17 | ectospasm | ...if they get the PIN wrong they can be dumped back in the queue, but beware, you'd have to use some extra logic to ensure they don't lose their place. |
| 09:38.19 | Starvirus | it's a door opener, i want to be able to open the door myself at the door, but i don't want anyone else to do it, so if they press a key - on accident it has to keep ringing because the mail guy and ups guy are stupid and press to much keys ^^ |
| 09:40.06 | ectospasm | so have the first digit execute within the dialplan context, use the Background app to play a ringback tone, and set up another extension (multiple digits) that opens the door (or whatever) |
| 09:43.29 | Starvirus | hm...i will try, thanks |
| 09:44.37 | ectospasm | Starvirus: note that there isn't a ringback sound file included with the core sounds distribution, so you'll have to find or generate your own. |
| 09:45.04 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 09:45.43 | ectospasm | if you're lucky the ringback tone may be imperceptibly broken as the caller is transitioned from the queue to the PIN read |
| 09:47.02 | ectospasm | you may also want to set an invalid handler that jumps back to the ringback Background() playback app, rather than playing an invalid message, or hanging up on the caller. |
| 09:47.23 | ectospasm | ...this is just one way to handle this problem |
| 09:47.36 | ectospasm | there are probably several other just as valid ways to do it |
| 09:48.45 | Starvirus | yeah think so too, and yeah this jumping back and forth without messages is clear :) |
| 09:49.32 | *** join/#asterisk Neptu (~Neptu@c-af90e255.113-1-64736c14.cust.bredbandsbolaget.se) |
| 09:51.01 | *** join/#asterisk imox (~imox@91-64-185-199-dynip.superkabel.de) |
| 10:10.36 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 10:14.24 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 10:17.52 | *** join/#asterisk wonderworld (~ww@dsdf-4db5fe13.pool.mediaWays.net) |
| 10:22.45 | *** join/#asterisk ectospasm (~ectospasm@unaffiliated/ectospasm) |
| 10:32.47 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 10:34.28 | *** part/#asterisk libryder (~libryder@libryder.com) |
| 10:39.52 | *** join/#asterisk Neptu (~Neptu@mail.avtech.aero) |
| 11:32.53 | *** join/#asterisk brian98 (~brian98@79.97.24.23) |
| 11:33.20 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 11:43.37 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 11:50.02 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
| 12:06.51 | *** join/#asterisk Neptu (~Neptu@c-af90e255.113-1-64736c14.cust.bredbandsbolaget.se) |
| 12:27.40 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 12:56.16 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 12:58.01 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 13:07.48 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 13:10.55 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 13:16.19 | *** join/#asterisk Bullmoose (~Bullmoose@63-155-51-139.bois.qwest.net) |
| 13:41.30 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 13:42.30 | *** join/#asterisk hehol (~Adium@ip-178-202-243-147.unitymediagroup.de) |
| 13:45.46 | *** join/#asterisk Goldwing (~Goldwing@84.245.47.128) |
| 13:47.19 | Goldwing | Q: im trying to replace my asterisk server, but for some reason i get "Unrecognized Trunk" in the status of Asterisk-GUI. |
| 13:47.54 | Goldwing | i moved the complete /etc/asterisk folder from a working system to the new system |
| 13:50.23 | Goldwing | Only difference in the 2 asterisk server is the build of asterisk (old is 10.0.1 new is 10.7.0) and the version of the gui (old is SVN--r5209 new is SVN--rexported) |
| 13:52.32 | ectospasm | AsteriskGUI? You mean the (old) one from Digium? |
| 13:54.08 | *** join/#asterisk Salah100 (~Sal@cpc1-bmly1-0-0-cust135.2-3.cable.virginmedia.com) |
| 13:55.52 | Salah100 | hi pls can someone help me for the last whole week i been trying to setup sipgate.co.uk to use on my asterisk server with no luck.. i can make test call to digum opertor |
| 13:56.05 | Salah100 | i get time out error |
| 13:56.11 | Salah100 | n also dns host error |
| 13:56.11 | WIMPy | Goldwing: Try in #asterisk-gui |
| 13:56.15 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 13:57.16 | Salah100 | am using asterisk win32 |
| 13:57.17 | ectospasm | Salah100: can you ping the ITSP from the Linux shell? |
| 13:57.25 | ectospasm | er... |
| 13:57.27 | Salah100 | puurely for hobbie use |
| 13:58.22 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 14:00.00 | ectospasm | I dunno if the Win32 port is supported here... I doubt it |
| 14:00.17 | Salah100 | oh |
| 14:00.23 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 14:01.07 | ectospasm | Salah100: but first check if you get name resolution in cmd shell... ping sipgate.co.uk |
| 14:01.22 | ectospasm | ...if that isn't happening then Asterisk will never be able to connect to it |
| 14:01.56 | Salah100 | ok i test now ectospasm thnk u |
| 14:02.28 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 14:02.30 | Salah100 | yeh its receving |
| 14:02.47 | WIMPy | I didn;t even know it still works unter Windows. Or ist that an old version? |
| 14:03.20 | Salah100 | i jst download asteriskwin32 from web |
| 14:03.25 | Salah100 | n worked |
| 14:03.32 | Salah100 | let me call digum server |
| 14:03.40 | WIMPy | What version is it? |
| 14:03.42 | ectospasm | the version on the download page says 0.66, which doesn't sound like any Asterisk version I remember ever hearing about |
| 14:04.09 | ectospasm | I'd stay very far away from AsteriskWin32 |
| 14:04.30 | WIMPy | Have you tried to call a sipgate (test) number? |
| 14:04.37 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 14:04.45 | WIMPy | They might just silently ignore incorrect registration attempts. |
| 14:04.46 | ectospasm | ...better still, use VirtualBox or VMWare, load your favorite Linux server distro, and load a proper version of Asterisk (I recommend 1.8.15.0 |
| 14:04.48 | ectospasm | ) |
| 14:05.31 | ectospasm | that's pretty much what I do with the Asterisk machine I use at home. |
| 14:06.04 | Salah100 | i tried to use vm ware |
| 14:06.13 | Salah100 | but i get "x" error |
| 14:06.41 | Salah100 | n i can connect to sipgate using x lite |
| 14:06.43 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 14:07.04 | WIMPy | What about another machine to run it on? An old PC or mayve your router. |
| 14:07.12 | Salah100 | i tried to in trixbox and asterisk now |
| 14:07.35 | Salah100 | i tried different internet connection |
| 14:07.49 | Salah100 | opened 5060 port on router |
| 14:07.57 | Salah100 | which sipgate uses |
| 14:08.46 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 14:08.56 | ectospasm | Salah100: you'd need to pastebin the Asterisk CLI when you try to register with the provider |
| 14:10.34 | Salah100 | ectospasm can u kindly explain dis pls.. am just a hobbiest |
| 14:10.51 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 14:12.06 | ectospasm | ~pb |
| 14:12.07 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 14:12.14 | ectospasm | ~thebook |
| 14:12.15 | 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 |
| 14:13.01 | ectospasm | rather than potentially confusing you, you should read The Definitive Guide. It's available for free online, possibly even PDF version |
| 14:13.46 | Salah100 | ok thnk u sirs |
| 14:18.57 | *** join/#asterisk rethus (~suther@p54BBC95D.dip.t-dialin.net) |
| 14:22.19 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 14:28.11 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 14:44.14 | Goldwing | What is tje CLI command to check the status of a SIP trunk? |
| 14:45.36 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 14:46.00 | rethus | is there a howto to install asterisk11 on debian squeeze |
| 14:47.41 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 14:49.05 | ectospasm | Goldwing: sip show peers |
| 14:49.20 | ectospasm | rethus: are you installing from source? |
| 14:49.34 | rethus | i got it, thanks |
| 14:49.38 | rethus | an yes |
| 14:49.40 | rethus | :) |
| 14:49.50 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 14:51.53 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 14:53.57 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 14:56.04 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 14:58.07 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 15:18.46 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 15:22.15 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 15:24.45 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 15:26.52 | ujjain | is IAX used to communicate between VOIP servers or SIP too? |
| 15:27.29 | ujjain | I want to ip-restrict SIP to specific IP's, I wonder if this is possible if the SIP protocol is also used by VOIP providers |
| 15:28.29 | WIMPy | SIp and IAX are two different tings. |
| 15:28.53 | WIMPy | But can be used for the same thing. |
| 15:29.01 | WIMPy | SIP+RTP that is. |
| 15:32.05 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 15:33.15 | ujjain | SIP can be used for communication, so I am unable to ip-restrict the protocol to the used phones , right? |
| 15:40.15 | WIMPy | I'm not sure what you intend to do. |
| 15:40.22 | Salah100 | manged to get rid of dns error now am gettin time out |
| 15:40.24 | Salah100 | ? |
| 15:46.28 | *** join/#asterisk sawgood (~sawgood@173-13-158-29-sfba.hfc.comcastbusiness.net) |
| 15:46.50 | ujjain | I don't want any IP to be able to use the VOIP server except the whitelisted IP. |
| 15:47.03 | ujjain | It seems impossible as communication to Intervoip also uses the same protocol and port. |
| 16:08.14 | *** join/#asterisk gringo (~gringo@unaffiliated/gringo) |
| 16:08.31 | WIMPy | You need to open the SIP port to anyone you want to receive calls from. |
| 16:36.30 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 16:55.56 | *** join/#asterisk navaismo (~navaismo@189.144.212.238) |
| 17:02.04 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 17:17.08 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 17:18.05 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 17:24.15 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 17:24.15 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 17:26.01 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 17:36.16 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 17:36.40 | *** join/#asterisk jakent (~jakent@ip-64-134-71-104.public.wayport.net) |
| 17:41.38 | Goldwing | ectospasm : primary*CLI> sip show peers |
| 17:41.38 | Goldwing | No such command 'sip show peers' (type 'core show help sip show' for other possible commands) |
| 17:43.22 | [TK]D-Fender | then chan_sip isn't even loaded |
| 17:43.48 | [TK]D-Fender | either a bad or missing modules.conf, asterisk.conf, or your config has an error preventing it from loading |
| 17:43.59 | [TK]D-Fender | so "module load chan_sip.so" |
| 17:44.11 | [TK]D-Fender | if that doesnt seem t complain run "sip show peers" after |
| 17:45.16 | Goldwing | [TK]D-Fender : that would explain the error i'm getting in the GUI also.."Unrecognized Trunk" with all my sip trunks |
| 17:45.36 | Goldwing | Command 'module load chan_sip.so' failed. |
| 17:45.36 | Goldwing | [Aug 25 19:44:26] WARNING[8097]: loader.c:402 load_dynamic_module: Error loading module 'chan_sip.so': /usr/lib/asterisk/modules/chan_sip.so: cannot open shared object file: No such file or directory |
| 17:45.36 | Goldwing | [Aug 25 19:44:26] WARNING[8097]: loader.c:854 load_resource: Module 'chan_sip.so' could not be loaded. |
| 17:49.47 | [TK]D-Fender | check that you have a modules.conf. then check the folder it points to |
| 17:50.43 | [TK]D-Fender | actully the path is clear |
| 17:50.54 | [TK]D-Fender | see if OTHER modules are even there |
| 17:51.14 | [TK]D-Fender | you either have a botched package install or failed a pre-req installing * from source, etc |
| 17:51.19 | Goldwing | [modules] |
| 17:51.20 | Goldwing | autoload=yes |
| 17:51.21 | Goldwing | noload => pbx_gtkconsole.so |
| 17:51.28 | Goldwing | noload => chan_alsa.so |
| 17:51.34 | Goldwing | noload => chan_console.so |
| 17:51.42 | [TK]D-Fender | that should have caught it. make sure the path is where the modules actually got put |
| 17:51.45 | [TK]D-Fender | and PASTEBIN... |
| 17:51.46 | [TK]D-Fender | ~pb |
| 17:51.46 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 17:51.49 | [TK]D-Fender | ^^^ |
| 17:51.52 | [TK]D-Fender | do nt flood in here |
| 17:51.54 | [TK]D-Fender | not* |
| 17:52.12 | Goldwing | those are the only lines that are being used and don't have a ; in front of them |
| 17:52.14 | Goldwing | ok.. sorry |
| 17:52.55 | [TK]D-Fender | how did you install * in the first place? |
| 17:53.06 | Goldwing | compiled it from source |
| 17:53.37 | [TK]D-Fender | Ok, check if your modules folder has other stuff in there. if it has a pile then chan_sip alone didn't compile (well maybe other stuff too) |
| 17:54.03 | [TK]D-Fender | go back into "make menuconfig" and check if it had XXX and look at the pre-req's for it |
| 17:54.57 | [TK]D-Fender | And on that note .... I'm out the door... back in a while... |
| 17:55.01 | [TK]D-Fender | heads off |
| 17:55.02 | Goldwing | ok |
| 17:55.04 | Goldwing | thanks alot |
| 18:10.04 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 18:10.53 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 18:13.59 | *** join/#asterisk WolWid (WolWid@pD9ED4866.dip.t-dialin.net) |
| 18:19.03 | Goldwing | [TK]D-Fender : fyi, i was missing libssl-dev on my system, therefore chan_sip.so wouldn't compile |
| 18:19.22 | *** join/#asterisk jakent (~jakent@ip-64-134-71-104.public.wayport.net) |
| 18:19.36 | Goldwing | it's up and running now, and even the GUI shows "registered" with my sip trunks |
| 18:27.53 | *** join/#asterisk DarthExpeditor (~IceChat9@173.46.183.185) |
| 18:40.39 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 18:56.57 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 19:09.09 | sawgood | thinking "Tacos or a Hamburger"? |
| 19:10.47 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 19:18.35 | *** join/#asterisk jakent_ (~jakent@149.sub-70-192-130.myvzw.com) |
| 19:24.12 | ChannelZ | Tacos |
| 19:24.31 | ChannelZ | Ironically I've been sitting here looking for good taco seasoning recipes |
| 19:25.02 | jpsharp | Mmmmm tacos. |
| 19:27.37 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 19:31.05 | learath | any java guys willing to give me a few minutes of help? |
| 19:34.07 | *** join/#asterisk Bullmoose (~Bullmoose@65-129-39-102.bois.qwest.net) |
| 19:35.38 | sawgood | 2 tacos from the mexican store it is then |
| 19:35.42 | sawgood | $1.83 each |
| 19:41.31 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 19:42.27 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 19:47.13 | *** join/#asterisk Sicelo (sicelo@unaffiliated/sicelo) |
| 19:48.01 | sawgood | two carne asada tacos with verde (sp) sauce ... |
| 19:51.57 | navaismo | uh too expensive |
| 19:57.19 | jpsharp | carne guisada tacos from Taco Cabana in San Antonio. Mmmmmm. |
| 19:58.53 | ChannelZ | they closed all those here |
| 19:59.25 | ChannelZ | "Number 41, your order is hot, fresh, and ready!" |
| 20:05.07 | *** join/#asterisk zombu2 (~zombu2@c-98-208-163-240.hsd1.fl.comcast.net) |
| 20:05.37 | zombu2 | afternoon |
| 20:06.52 | zombu2 | any way to force a payload size for video ? |
| 20:10.53 | sawgood | These tacos were good (normally I do not care for verde green sauce), but I'm starting to like it |
| 20:17.55 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 20:18.29 | *** join/#asterisk wonderworld (~ww@dsdf-4db5fe13.pool.mediaWays.net) |
| 20:20.43 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 20:28.14 | *** part/#asterisk WolWid (WolWid@pD9ED4866.dip.t-dialin.net) |
| 20:34.48 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 20:51.01 | *** join/#asterisk kriger (~norge@68-190-90-80.dhcp.mdsn.wi.charter.com) |
| 20:56.35 | *** join/#asterisk Bullmoose (~Bullmoose@65-129-43-29.bois.qwest.net) |
| 21:06.12 | *** join/#asterisk oneadvent (~oneadvent@ip68-105-63-31.pn.at.cox.net) |
| 21:06.19 | oneadvent | hi is anyone familiar with hylafax here? |
| 21:07.26 | WIMPy | ~ask |
| 21:07.26 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 21:07.45 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 21:07.47 | oneadvent | ah alrighty then, sorry about that WIMPy and infobot |
| 21:07.57 | sawgood | infobot can be so un-forgiving ... |
| 21:08.23 | oneadvent | my question: I am trying to setup failed fax delivery and currently it is sending a ps on failed faxes (to the correct email) and I would like it to be a pdf. How can I set that up in hylafax? |
| 21:09.01 | navaismo | Hello, I cant send the email voicemail alert in html template bigger than 530 bytes, any hints? I want to send a better template not an 1998's html template |
| 21:10.53 | sawgood | actually, 1998 was a decent year for me |
| 21:11.21 | jpsharp | oneadvent: Wash it through ps2pdf? |
| 21:16.01 | *** part/#asterisk amigamia (~sambagirl@unaffiliated/sambagirl) |
| 21:19.52 | *** join/#asterisk danfromuk (~IceChat77@2.30.230.189) |
| 21:19.57 | oneadvent | well jpsharp i'd rather send it correctly the first time. I've done a lot of reading on FaxNotify but i'm not familiar enough with it to know if I set it up right |
| 21:21.07 | danfromuk | I'm having trouble with this message. Does anyone know the cause? Correct auth, but based on stale nonce received from |
| 21:33.51 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 21:34.18 | *** join/#asterisk Greek-Boy (~Greek-Boy@196.46.120.119) |
| 21:41.08 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 21:44.11 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 21:51.48 | *** part/#asterisk Goldwing (~Goldwing@84.245.47.128) |
| 21:52.48 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 22:16.32 | *** join/#asterisk Nemus (~Nemus@c-76-23-30-50.hsd1.ut.comcast.net) |
| 22:20.23 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 23:26.14 | *** join/#asterisk YoMomma (~YoMomma@mail.jkrsi.com) |
| 23:28.32 | *** join/#asterisk metabsd (~metabsd@cable-17.246.173-101.electronicbox.net) |
| 23:43.37 | metabsd | Anyone can help me with asterisk and Aruba S2500 layer switching |