| 00:00.11 | [TK]D-Fender | learnITNoob: look in "ps -A|grep asterisk" |
| 00:00.14 | [TK]D-Fender | learnITNoob: If you don't see it, then for sure * is not running |
| 00:00.25 | learnITNoob | ok |
| 00:00.32 | learnITNoob | let me check |
| 00:00.34 | [TK]D-Fender | learnITNoob: then if you are, did you indeed start it as a daemon, or did you do so manually? |
| 00:01.03 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 00:01.05 | learnITNoob | I restart the server early morning today |
| 00:01.16 | learnITNoob | after that i could not start the asterisk |
| 00:01.20 | learnITNoob | I dont know why |
| 00:01.44 | *** join/#asterisk matt_ (n=matt@mattspc.ipv6.mattstone.net) |
| 00:01.52 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 00:02.10 | *** join/#asterisk dogmeat (n=Bob@unaffiliated/dogmeat) |
| 00:02.26 | learnITNoob | This is the messages I get when I did tail |
| 00:02.35 | learnITNoob | Dec 15 23:44:51 WARNING[25701] pbx.c: Context 'app-cf-busy-on' tries includes nonexistent context 'app-cf-busy-on-custom' |
| 00:02.35 | learnITNoob | Dec 15 23:44:51 WARNING[25701] pbx.c: Context 'app-cf-busy-off-any' tries includes nonexistent context 'app-cf-busy-off-any-custom' |
| 00:02.35 | learnITNoob | Dec 15 23:44:51 WARNING[25701] pbx.c: Context 'app-cf-busy-off' tries includes nonexistent context 'app-cf-busy-off-custom' |
| 00:02.35 | learnITNoob | Dec 15 23:44:51 NOTICE[25701] src/chan_h323.c: Unable to load config ooh323.conf, OOH323 disabled |
| 00:02.37 | learnITNoob | Dec 15 23:44:51 WARNING[25701] chan_zap.c: Unable to specify channel 1: No such device or address |
| 00:02.39 | learnITNoob | Dec 15 23:44:51 ERROR[25701] chan_zap.c: Unable to open channel 1: No such device or address |
| 00:02.41 | learnITNoob | here = 0, tmp->channel = 1, channel = 1 |
| 00:02.43 | learnITNoob | Dec 15 23:44:51 ERROR[25701] chan_zap.c: Unable to register channel '1-15' |
| 00:02.43 | *** join/#asterisk joako (n=joako@adsl-074-170-252-213.sip.gnv.bellsouth.net) |
| 00:02.45 | learnITNoob | Dec 15 23:44:51 WARNING[25701] loader.c: chan_zap.so: load_module failed, returning -1 |
| 00:02.47 | learnITNoob | Dec 15 23:44:51 WARNING[25701] loader.c: Loading module chan_zap.so failed! |
| 00:02.52 | [TK]D-Fender | learnITNoob: PASTEBIN |
| 00:02.54 | [TK]D-Fender | ~pb |
| 00:02.55 | 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, or , http://bin.cakephp.org/ |
| 00:02.59 | [TK]D-Fender | learnITNoob: please do not spam in here |
| 00:03.13 | learnITNoob | ok thank you |
| 00:03.16 | learnITNoob | sorry |
| 00:03.18 | [TK]D-Fender | learnITNoob: And yes, Zaptel is clearly not loading and is causeing * to bomb out |
| 00:03.37 | [TK]D-Fender | learnITNoob: run "ztcfg -vvvv" if that doesn't report errors, try starting * manually following it |
| 00:03.46 | learnITNoob | can you please tell me how i can load zaptel as i am trying to figure it out since morning |
| 00:03.53 | learnITNoob | i will be grateful |
| 00:04.19 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 00:04.26 | [TK]D-Fender | learnITNoob: try as I suggested. |
| 00:04.54 | [TK]D-Fender | learnITNoob: if that works, then likely you did not have a startup script to initialize Zaptel on boot prior to trying to start * |
| 00:04.57 | learnITNoob | this is th emessage i get |
| 00:04.58 | learnITNoob | Zaptel Configuration Channel map:0 channels configured. |
| 00:05.16 | [TK]D-Fender | learnITNoob: What hardware cards are you using? |
| 00:05.18 | *** join/#asterisk jayyers (n=jayyers@c-71-59-10-252.hsd1.ga.comcast.net) |
| 00:05.53 | *** join/#asterisk farkus (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 00:06.08 | learnITNoob | i will put in the pastebin |
| 00:06.12 | learnITNoob | 1 sec |
| 00:06.22 | *** join/#asterisk Nasra (n=maxshipp@187stb20.codetel.net.do) |
| 00:06.45 | learnITNoob | http://pastebin.com/m48cfbe23 |
| 00:07.44 | [TK]D-Fender | learnITNoob: I'm just taling telecom cards |
| 00:07.48 | [TK]D-Fender | talking* |
| 00:08.01 | learnITNoob | ok |
| 00:08.15 | learnITNoob | so any suggestions |
| 00:09.55 | *** join/#asterisk terracon (n=greisky@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 00:10.13 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 00:10.43 | [TK]D-Fender | learnITNoob: I suggest you answer the pretty straightforward question I jsut asked :) |
| 00:10.54 | [TK]D-Fender | learnITNoob: What cards requiring ZAPTEL are you using? |
| 00:11.08 | learnITNoob | I am not sure |
| 00:11.19 | learnITNoob | can you tell me how to find out this |
| 00:11.41 | [TK]D-Fender | learnITNoob: How can you not know what telecom specific hardware you have bought & installed if any at all? |
| 00:12.28 | learnITNoob | I didnt installed it I am trying to fix it for my friend |
| 00:12.35 | learnITNoob | he bought it from some where |
| 00:12.39 | [TK]D-Fender | learnITNoob: And you don't know what he's using at all? |
| 00:12.49 | [TK]D-Fender | learnITNoob: "it" is not a great answer... |
| 00:12.54 | learnITNoob | hes trying to use it for home |
| 00:13.14 | learnITNoob | He bought the system of Bootsale |
| 00:13.32 | eric2 | what's with <ZOMBIE> in the cdr logs under the channel? |
| 00:13.32 | [TK]D-Fender | learnITNoob: increasingly non-helpful. |
| 00:13.53 | learnITNoob | ok let me explain |
| 00:14.28 | [TK]D-Fender | learnITNoob: Whats to explain? Find out what model of telecom card(s) he has in his box. |
| 00:14.37 | [TK]D-Fender | learnITNoob: this doesn't take a story. This takes a MODEL # |
| 00:14.46 | learnITNoob | My friend bought this dell server off boot sale and I am trying to help me to create VOIP extensions for his home |
| 00:15.12 | [TK]D-Fender | learnITNoob: Does it even HAVE a zaptel interface card of some kind in it? |
| 00:15.17 | learnITNoob | can I find it out through web interface |
| 00:15.26 | [TK]D-Fender | learnITNoob: "dmesg" |
| 00:15.31 | learnITNoob | yes it has |
| 00:15.33 | learnITNoob | ok |
| 00:15.42 | [TK]D-Fender | learnITNoob: then do share... |
| 00:17.10 | learnITNoob | its a WILD CARD |
| 00:17.23 | learnITNoob | wildcard Te220 |
| 00:17.32 | [TK]D-Fender | THERE... |
| 00:17.57 | [TK]D-Fender | learnITNoob: And has he got that plugged into an E1 interface at HOME? (I'd doubt it personally) |
| 00:19.23 | learnITNoob | this is the report link when i excute dmesg |
| 00:19.25 | learnITNoob | http://pastebin.com/d66a8f20d |
| 00:20.50 | pfn | is there a comparo of tdm400 vs tdm410 anywhere? |
| 00:21.23 | mosty | pfn, it's the same base card, with different modules added |
| 00:21.33 | [TK]D-Fender | learnITNoob: Dear God that is an ANCIENT version |
| 00:21.34 | pfn | tdm410p is a new card |
| 00:21.38 | pfn | tdm400p is the original model |
| 00:21.52 | pfn | has the latter--would like to know what's new in the new model |
| 00:22.05 | learnITNoob | :) bought of boot sale |
| 00:22.08 | [TK]D-Fender | pfn: Not sure if there is anything to directly compare. Its basically a very different PCI design. Same modules IIRC |
| 00:22.25 | [TK]D-Fender | learnITNoob: He has no need of that card I'm betting... |
| 00:22.39 | mosty | pfn, ahh ok, we stopped using digium cards a while back |
| 00:22.40 | pfn | [TK]D-Fender, the different pci design should mean something in terms of contrast |
| 00:22.41 | [TK]D-Fender | learnITNoob: You just want to use it with VoIP for phones & outside connectivity? |
| 00:22.45 | learnITNoob | he was working fine upo till morning when i restart the server |
| 00:22.57 | [TK]D-Fender | pfn: IRQ handling, etc |
| 00:23.14 | learnITNoob | yes [TK]D-Fender |
| 00:23.30 | [TK]D-Fender | learnITNoob: Well right now its trying to initialize 16 E1 (EuroISDN) cannels. I highly doubt this is appropriate or necessary. |
| 00:23.32 | pfn | [TK]D-Fender, that doesn't mean much without elaboration--this isn't documented anywhere? |
| 00:24.02 | [TK]D-Fender | pfn: Go read Digium's press-releases, card spec-sheet, etc |
| 00:24.13 | learnITNoob | so what should you suggest |
| 00:24.13 | pfn | yeah, I've looked, nothing in there of interest |
| 00:24.26 | [TK]D-Fender | pfn: Then phone them up |
| 00:24.35 | pfn | the only thing might be the echo-can module |
| 00:24.39 | [TK]D-Fender | learnITNoob: Go into zapata.conf and comment out those channels |
| 00:24.40 | pfn | I don't think the 400p has that or support for it |
| 00:25.21 | learnITNoob | [TK]D-Fender: zapata.conf file through webinterface? |
| 00:25.22 | [TK]D-Fender | pfn: Yes, that too.. and no the TDM400P doesn't support an EC module |
| 00:25.39 | [TK]D-Fender | learnITNoob: WHAT web interface? you're is 3rd party and not supported here |
| 00:25.56 | *** part/#asterisk korihor (n=korihor@201.210.239.172) |
| 00:25.57 | pfn | [TK]D-Fender, not necessary for fax, fortunately :) |
| 00:26.08 | [TK]D-Fender | pfn: Correct |
| 00:26.12 | pfn | and I don't have a problem with echo on FXS |
| 00:26.26 | [TK]D-Fender | pfn: its more on the PCI latency & IRQ handling. |
| 00:26.43 | learnITNoob | [TK]D-Fender: we have a trixbox plus Intrintech webinterface with it, we have tested 100 extensions so far |
| 00:27.31 | [TK]D-Fender | learnITNoob: Trixbox (and FreePBX) are not supported here, and I've never even heard of Intrintech before |
| 00:27.43 | *** join/#asterisk farkus (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 00:27.53 | [TK]D-Fender | learnITNoob: 1 or 100 makes no difference |
| 00:28.03 | [TK]D-Fender | learnITNoob: Go disable those zapata channels |
| 00:28.04 | learnITNoob | [TK]D-Fender: ok so what should I change in zapata.conf |
| 00:28.08 | learnITNoob | ok |
| 00:28.20 | [TK]D-Fender | learnITNoob: comment out the "channels =>" line |
| 00:28.44 | learnITNoob | [TK]D-Fender: can you please tell me how i can go in to zapata.cnf file thanks |
| 00:28.54 | [TK]D-Fender | learnITNoob: "man vi" |
| 00:28.59 | learnITNoob | thank you |
| 00:29.57 | pfn | The TDM410P, utilizing Digium's patent-pending VoiceBus⢠technology and with a PCI interface found in millions of servers worldwide, provides the same functional capacity as the TDM400P, but improves substantially upon its system compatibility and provides the capability of connecting to Digium's hardware-based echo cancellation modules. |
| 00:30.02 | pfn | not a big deal, it sounds like |
| 00:30.29 | [TK]D-Fender | pfn: If you are looking for answers, go CALL THEM |
| 00:32.06 | pfn | nah, thanks |
| 00:32.57 | [TK]D-Fender | pfn: Indeed your approach to this says you're looking for anything you can find to give you a reason NOT to consider it. |
| 00:33.05 | pfn | [TK]D-Fender, of course |
| 00:33.12 | pfn | why spend money when what I have is good enough |
| 00:33.16 | [TK]D-Fender | pfn: So next time, just don't bother asking. |
| 00:33.27 | pfn | self-reassurance is common :p |
| 00:33.38 | [TK]D-Fender | pfn: And if you knew that to begin with than this entire exercise is indeed a complete waste |
| 00:33.55 | pfn | and why bother wasting digium's time when if I'm not really planning to buy unless there is something particularly motivating |
| 00:34.02 | pfn | s/if// |
| 00:34.06 | [TK]D-Fender | pfn: Since we haven't gotten anything fully conclusive you're actually no further ahead. Good luck with that. |
| 00:34.50 | [TK]D-Fender | pfn: Sometimes the fine points aren't well documented but I've heard specific reference to faxing concerning the new PCI design of their cards |
| 00:35.13 | [TK]D-Fender | pfn: But yeah, clearly not worth thinking about. Move along. These aren't the cards you're looking for. |
| 00:36.05 | pfn | also, considering this is #asterisk, I would expect someone from digium to eventually chime in |
| 00:36.33 | mosty | pfn, you could just call your local digium reseller and ask |
| 00:37.01 | pfn | yeah, but I'd rather use irc :p |
| 00:37.17 | pfn | I can be lazy and not have to pick up the phone |
| 00:38.22 | [TK]D-Fender | pfn: The only person you're kidding or cheating here is yourself. |
| 00:38.30 | pfn | shrugs |
| 00:38.36 | pfn | not a mission-critical application |
| 00:38.38 | learnITNoob | [TK]D-Fender: how can I open the zapata.cnf file in man vi text editor please help thanks |
| 00:38.48 | [TK]D-Fender | learnITNoob: Go ask in ##linux |
| 00:38.58 | learnITNoob | ok |
| 00:39.30 | [TK]D-Fender | learnITNoob: I'm starting to wonder how much "help" you are for this friend of yours.... |
| 00:40.58 | learnITNoob | he repaired my car for free so I am just helping him in return |
| 00:41.19 | pfn | at least you're appropriately "named" |
| 00:41.58 | pfn | Current Balance: $1.57 |
| 00:42.00 | pfn | Last Payment: 2006-08-17 19:42:28 |
| 00:42.06 | pfn | hmm, just another hundredish minutes to go |
| 00:42.08 | *** join/#asterisk SiberAIR (i=SibrPhre@cpe-67-243-43-136.nyc.res.rr.com) |
| 00:42.54 | pfn | hmm, less, nufone is 2c/min |
| 00:44.39 | [TK]D-Fender | Wow... Toshiba 40" 1080P @ $800 |
| 00:45.22 | pfn | still <3 projector |
| 00:46.14 | [TK]D-Fender | pfn: Yes... my * server has the biggest screen on any out there :) |
| 00:46.18 | [TK]D-Fender | of* |
| 00:46.25 | [TK]D-Fender | (probably) |
| 00:46.34 | pfn | why would you do that, unless your * server does more than just * |
| 00:46.38 | [TK]D-Fender | 120" = healthy :) |
| 00:46.45 | [TK]D-Fender | pfn: It does a LOT more. |
| 00:46.48 | pfn | my screen is only 80" wide :( |
| 00:47.01 | [TK]D-Fender | pfn: file, web server + router, etc |
| 00:47.32 | pfn | usually those kinda boxes run headless |
| 00:47.32 | [TK]D-Fender | pfn: 80" WIDE? that still pretty hugs. My 120" diagonal (4:3) is 96" wide |
| 00:47.44 | [TK]D-Fender | pfn: Its also my Media PC :) |
| 00:47.51 | [TK]D-Fender | pfn: uber-all-in-one |
| 00:47.56 | pfn | my box is rinkydink |
| 00:48.03 | pfn | p3-1200, not enough hp to run everything |
| 00:48.10 | pfn | so it just runs asterisk and tomcat and bittorrent |
| 00:48.24 | [TK]D-Fender | pfn: minimal GUI doesn't take much... |
| 00:48.40 | [TK]D-Fender | pfn: Min is an AMD XP2000+ |
| 00:50.41 | learnITNoob | [TK]D-Fender: can you tell me where exactly this file locates zapata.cnf do i have to cd to it or open it directly |
| 00:50.48 | learnITNoob | thanks for your helo |
| 00:50.53 | learnITNoob | *help |
| 00:51.05 | [TK]D-Fender | learnITNoob: /etc/asterisk/zapata.conf |
| 00:52.11 | learnITNoob | thanks TK |
| 00:58.31 | learnITNoob | [TK]D-Fender: shell i remove these channels ;channel => 1-20,21-31 |
| 00:58.31 | learnITNoob | channel => 1-15,17-31 |
| 01:01.12 | [TK]D-Fender | learnITNoob: I've already said it twice... |
| 01:01.18 | learnITNoob | ok sorry |
| 01:01.26 | learnITNoob | after doing it shell i reboot |
| 01:01.59 | rue_mohr | kenn<tab> |
| 01:02.02 | rue_mohr | :/ |
| 01:04.52 | [TK]D-Fender | LEANnO NEED. yOUS HOULD BE ABLE TO JUST RUN * |
| 01:05.12 | pfn | people ruv rebooting |
| 01:05.15 | pfn | its the windows way |
| 01:05.27 | learnITNoob | ok thanks |
| 01:06.15 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 01:06.53 | *** join/#asterisk r0d3nt (i=astrutt@pinky.ratman.org) |
| 01:09.10 | learnITNoob | [TK]D-Fender: its still giving me the same error Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 01:10.28 | *** join/#asterisk Trido (i=trido@ppp178-168.static.internode.on.net) |
| 01:10.53 | [TK]D-Fender | learnITNoob: Go start * manually and see what's failing |
| 01:11.18 | *** part/#asterisk Trido (i=trido@ppp178-168.static.internode.on.net) |
| 01:12.00 | *** join/#asterisk Trido (i=trido@ppp178-168.static.internode.on.net) |
| 01:12.48 | learnITNoob | [TK]D-Fender: this is the pastebin link report when i excute genzaptelconf http://pastebin.com/d5e716b45 |
| 01:13.14 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 01:13.32 | mchou | anyone here ever use a pingtel xpressa? |
| 01:13.48 | *** part/#asterisk `paul (n=admin@122.55.36.3) |
| 01:14.02 | *** join/#asterisk zchaos (n=none@CPE001d7ef0ba9d-CM001ceab63f9a.cpe.net.cable.rogers.com) |
| 01:15.33 | *** join/#asterisk JAMMAN2110 (n=James@unaffiliated/jamman2110) |
| 01:21.43 | learnITNoob | [TK]D-Fender: did you get the chance to read my pastebin? |
| 01:26.18 | *** join/#asterisk prodyan (n=ian@124.104.71.66) |
| 01:26.24 | prodyan | hello all |
| 01:27.33 | *** join/#asterisk rcy (n=rcy@S01060002553240a8.vc.shawcable.net) |
| 01:29.35 | *** join/#asterisk rcy`` (n=rcy@S01060002553240a8.vc.shawcable.net) |
| 01:32.23 | [TK]D-Fender | learnITNoob: Why did you go and run something that might jsut UNDO the change I jsut told you to make? |
| 01:32.50 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:34.49 | *** join/#asterisk unixdawg (n=unixdawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 01:35.47 | rue_mohr | [TK]D-Fender, |
| 01:36.42 | rue_mohr | Dec 14 23:16:37 WARNING[7527] chan_zap.c: Ignoring hanguponpolarityswitch |
| 01:36.42 | rue_mohr | Dec 14 23:16:37 WARNING[7527] chan_zap.c: Ignoring signalling |
| 01:36.56 | rue_mohr | erm |
| 01:38.12 | rue_mohr | is this in regards to the randomly hang up my calls you were talking about? |
| 01:38.16 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 01:40.04 | *** join/#asterisk mog (n=mog@c-68-62-172-152.hsd1.al.comcast.net) |
| 01:40.04 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:40.19 | rue_mohr | I suppose my question is how do I know what signalling its ignoring |
| 01:40.57 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:41.04 | rue_mohr | and why isn't that string in there |
| 01:42.47 | *** join/#asterisk andresmujica (n=andresmu@201.244.110.112) |
| 01:43.55 | rue_mohr | hanguponpolarityswitch=yes |
| 01:44.02 | rue_mohr | Dec 14 23:16:37 WARNING[7527] chan_zap.c: Ignoring hanguponpolarityswitch |
| 01:44.06 | rue_mohr | hmmm |
| 01:44.49 | rue_mohr | does that seem to conflict with anyone but me? |
| 01:45.04 | rue_mohr | er reverse that |
| 01:46.13 | learnITNoob | can any one help me to solve this problem please: DEBUG[3707] chan_sip.c: Auto destroying call |
| 01:46.33 | rue_mohr | your call was terminated |
| 01:46.42 | rue_mohr | are you having any dropped calls? |
| 01:47.02 | learnITNoob | yes |
| 01:47.10 | rue_mohr | I recall the office system I was helping with had those, and they were part of normal operation, I think, then your aren't |
| 01:47.20 | rue_mohr | hmm |
| 01:47.26 | rue_mohr | wonder why its doing that |
| 01:47.31 | rue_mohr | be nice if it said |
| 01:47.39 | learnITNoob | let me reboot my server |
| 01:47.41 | rue_mohr | too many error messages are not fix-oriented |
| 01:47.45 | rue_mohr | why |
| 01:47.45 | rue_mohr | ? |
| 01:47.50 | learnITNoob | are you sure its a normal procedure? |
| 01:47.57 | rue_mohr | its not on yours |
| 01:48.12 | rue_mohr | not if your getting dropped calls |
| 01:48.17 | rue_mohr | what does it say beofre that? |
| 01:48.21 | rue_mohr | too much lag? |
| 01:48.36 | learnITNoob | not lagging just dropped |
| 01:49.04 | rue_mohr | are you using UDP through a firewall? |
| 01:49.29 | learnITNoob | i just run trail /var/log/full command and it gave me this error report |
| 01:49.43 | rue_mohr | I once heard about routers that time out the reverse port map DURING traffic sessions |
| 01:49.44 | learnITNoob | yes I am usin a Sonic firewall |
| 01:50.07 | rue_mohr | that might be it then, hmm |
| 01:50.14 | rue_mohr | what did I read about that |
| 01:50.35 | rue_mohr | something about "there should be a means for a keepalive to get around this" |
| 01:50.36 | learnITNoob | lol you will know it better than me |
| 01:50.51 | learnITNoob | do i have to open a udp port for it |
| 01:50.52 | rue_mohr | its just something I crossed |
| 01:51.11 | rue_mohr | 10000 and 5060, right? |
| 01:51.30 | *** join/#asterisk Bad_Robot- (n=Bad_Robo@cpe-76-173-219-25.socal.res.rr.com) |
| 01:51.43 | learnITNoob | okay |
| 01:51.47 | learnITNoob | i will check |
| 01:52.11 | rue_mohr | is it a polycom phone? |
| 01:52.23 | learnITNoob | no its a snom phone |
| 01:52.31 | learnITNoob | snom 360 n 37 |
| 01:52.36 | learnITNoob | *370 |
| 01:52.50 | rue_mohr | SonicWall and transfer |
| 01:52.50 | rue_mohr | Hold and transfer functions of at least the IP 600 behind certain versions of Sonic Wall routers does not work. A call placed on hold would drop at exactly 5 seconds. Placing and receiving calls works fine. Replacing the Sonic Wall with a later version solved that problem. |
| 01:53.08 | rue_mohr | so you get killed after 5 seconds? |
| 01:53.13 | learnITNoob | yea |
| 01:53.23 | rue_mohr | :) its your firewall, I'm sure of it |
| 01:53.58 | learnITNoob | but it was fine until this morning i restarted my server |
| 01:54.06 | learnITNoob | i could not able to run asterisk |
| 01:54.21 | learnITNoob | TK]D:helped me on running it |
| 01:55.09 | rue_mohr | learnITNoob, what do your firewall logs say |
| 01:55.53 | rue_mohr | For customers using the Junction Networks SIP Hosted PBX service and the Sonic Firewall, the SIP Transformations sections should be DISABLED (unchecked). |
| 01:56.08 | learnITNoob | ok |
| 01:56.08 | rue_mohr | thats not asterisk... |
| 01:56.26 | rue_mohr | http://www.junctionnetworks.com/knowledgebase/onsip/devices/router-configuration/sonic-firewall/sonic-firewall-sip-transformations |
| 01:56.29 | learnITNoob | I am using asterisk with trixbox running on it |
| 01:56.43 | learnITNoob | Thnak you very much rue: |
| 01:56.55 | rue_mohr | never used trixbox, never heard of sonic firewall |
| 01:57.00 | rue_mohr | :) |
| 01:57.24 | learnITNoob | whats wrong with trixbox |
| 01:57.29 | learnITNoob | its free version |
| 01:57.42 | learnITNoob | which one is best for commerical voip system |
| 01:57.50 | rue_mohr | I'm a guy who likes a standard transmission, with a clutch |
| 01:57.57 | rue_mohr | so I just run asterisk raw |
| 01:58.24 | learnITNoob | what about cicso voip system |
| 01:58.36 | rue_mohr | nope, not that wealthy |
| 01:59.18 | learnITNoob | how much would it cost, any ideas? |
| 01:59.27 | rue_mohr | cisco? |
| 01:59.56 | rue_mohr | cost isn't the hard part, its getting permission to own cisco equiptment |
| 02:00.34 | learnITNoob | yea true |
| 02:00.55 | learnITNoob | is it bad idea to run asterisk on PBX |
| 02:02.07 | rue_mohr | say wha? |
| 02:02.32 | rue_mohr | pbx, but you dont mean pbx |
| 02:03.38 | rue_mohr | I heard: is it a bad idea to use a calculator for calculating |
| 02:04.22 | rue_mohr | your lag is going up, I'm gonna go work on my hexapod robot now |
| 02:13.25 | [TK]D-Fender | there is a LOT of crack going around it seems... |
| 02:13.39 | [TK]D-Fender | [21:00]<learnITNoob>is it bad idea to run asterisk on PBX <- WTF |
| 02:14.25 | [TK]D-Fender | rue_mohr: And you mention "sonic firewall", and that links to a SonicWALL (totally different brand) guide |
| 02:14.49 | [TK]D-Fender | And Cisco = $$ |
| 02:15.07 | rue_mohr | [TK]D-Fender, well, you should have been there |
| 02:15.18 | [TK]D-Fender | rue_mohr: Where? |
| 02:15.45 | rue_mohr | I tried to bring back up what I read about a firewall closing a port off based on a timer and ignoring traffic |
| 02:15.56 | [TK]D-Fender | rue_mohr: What model? |
| 02:16.06 | rue_mohr | I dont know where I read it |
| 02:16.14 | rue_mohr | it wasn't what I was looking for |
| 02:16.16 | digime | hi guys ,quick question, when you dial an invalid extension on my ivr, it drops the call. this is the error: [Dec 15 18:26:28] WARNING[29209]: pbx.c:1832 pbx_extension_helper: No application 'SetVar' for extension (101-main-ivr, i, 1) what did i do wrong? |
| 02:16.49 | rue_mohr | hmm |
| 02:16.56 | [TK]D-Fender | digime: You used a dialplan app that was deprecated in **1.2** |
| 02:17.04 | digime | yes i think you're right |
| 02:17.12 | [TK]D-Fender | digime: Let me guess... more blind cut&paste from the WIKI, right? |
| 02:17.15 | digime | i upgraded from 1.2 to 1.4 but I didn't change anything |
| 02:17.36 | [TK]D-Fender | digime: next time read the upgrade.txt, README, etc. |
| 02:17.37 | digime | no but nevertheless, you are right about the problem |
| 02:17.45 | [TK]D-Fender | digime: we call them very fine manuals here... |
| 02:17.52 | digime | yeah |
| 02:17.55 | [TK]D-Fender | digime: Set() <- |
| 02:17.58 | digime | do you know off hand how i can correct it? |
| 02:18.07 | digime | aha so SetVar becomes Set |
| 02:19.05 | digime | works! thanks! |
| 02:20.49 | digime | I have one more but this is trickier: WARNING[29220]: ast_expr2.fl:407 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end; Input: = 2 |
| 02:20.57 | digime | and that is from my queue extensions |
| 02:21.29 | *** join/#asterisk afink (n=afink@ip68-13-89-102.om.om.cox.net) |
| 02:21.35 | digime | i found the =2 entry but it seems OK to me?? |
| 02:23.15 | digime | exten => 8804,1,Set(101-qs8804=${DB(101-qs/8804)}) |
| 02:23.32 | digime | exten => 8804,2,GotoIf,"$[${101-qs8804} = 2]?8804|3:8804|8" |
| 02:23.43 | digime | so something it doesnt like in that code. |
| 02:25.34 | seanbright | you can't have '-' in a variable name |
| 02:25.36 | afink | anyone have any input on what could be causing this? http://pastebin.com/m41de9066 I was hoping that upgrading to DAHDI would fix the problem but that wasn't the case. I have seen lots of similar posts on the internet but haven't seen any solutions. |
| 02:25.37 | seanbright | i don't think... |
| 02:25.48 | digime | aha |
| 02:26.02 | seanbright | but this is asterisk... i may be wrong |
| 02:26.08 | digime | well, the queue works actually |
| 02:26.13 | [TK]D-Fender | I'm also pretty sure you can't start a var with a DIGIT |
| 02:26.18 | digime | but I just get the puzzling error |
| 02:27.23 | [TK]D-Fender | digime>exten => 8804,2,GotoIf,"$[${101-qs8804} = 2]?8804|3:8804|8" <- 50% longer than it needs to be |
| 02:27.33 | digime | ok |
| 02:28.04 | digime | what do you mean 50% longer? |
| 02:28.55 | drmessano | OMG a viagra ad |
| 02:29.29 | [TK]D-Fender | digime: exten => 8804,2,GotoIf($["${myvar}" != "2"]?8" |
| 02:29.35 | [TK]D-Fender | digime: exten => 8804,2,GotoIf($["${myvar}" != "2"]?8) |
| 02:29.41 | digime | wow |
| 02:29.51 | digime | wow |
| 02:29.55 | digime | that's interesting |
| 02:30.19 | [TK]D-Fender | digime: you don't seem to have much of a programming background for this to seem impressive |
| 02:30.27 | digime | true! |
| 02:30.34 | digime | I am not a programmer |
| 02:30.58 | digime | Okay let me try and implement your code |
| 02:31.32 | digime | should i leave my set line the same? |
| 02:31.50 | [TK]D-Fender | digime: use som IQ here and look at what its referencing |
| 02:32.07 | digime | myvar |
| 02:32.09 | digime | ok |
| 02:32.37 | digime | attempting to use IQ |
| 02:33.29 | [TK]D-Fender | smess something burning |
| 02:33.33 | [TK]D-Fender | smells* |
| 02:34.13 | digime | hmm something broke with that code. i used the second line you pasted. but it is now saying the queue is unavailable. |
| 02:34.32 | [TK]D-Fender | digime: Funny... that line has NOTHIGN to do with a queu so far |
| 02:34.48 | [TK]D-Fender | digime: maybe you should pastebin the whole exten. |
| 02:34.51 | digime | ok |
| 02:36.35 | digime | http://pastebin.com/d49c67883 |
| 02:37.46 | [TK]D-Fender | digime: exten => 8804,2,GotoIf($["${myvar}" != "2"]?8) <--- you did not SET this variable name to anything. |
| 02:38.29 | [TK]D-Fender | digime: Holy crap stop cut& pasting everything you see and look at what it's REFERENCING. that var you set above is a ridiculous name hence my suggestion to use something SIMPLE like "myvar" |
| 02:38.41 | digime | yes yes i see |
| 02:39.34 | digime | so myvar becomes 101-qs8804 (can't I use that name for now) |
| 02:40.26 | *** join/#asterisk TrentCreek (n=kvirc@adsl-75-14-31-197.dsl.hrlntx.sbcglobal.net) |
| 02:40.26 | digime | the 101- is there because I have a multi tenant configuraiton |
| 02:40.54 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 02:41.40 | [TK]D-Fender | digime: NO POINT |
| 02:42.11 | [TK]D-Fender | digime: "101-qs8804" is a stupid name. You are only using it in the Gotoif that follows. |
| 02:42.12 | drmessano | You're putting stunt pegs on a tricycle |
| 02:42.34 | digime | true. okay. |
| 02:42.37 | [TK]D-Fender | drmessano: Just wait for the skirt-kit, giant honking wing on the bag and new mags... |
| 02:42.47 | [TK]D-Fender | drmessano: then it'll be a BITCHIN' ASS RIDE! |
| 02:42.54 | digime | well either way, i am hitting voicemail again. |
| 02:42.59 | drmessano | Yep.. a Bitchcyle! |
| 02:43.06 | drmessano | Bitchcycle! |
| 02:43.19 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 02:43.20 | seanbright | digime: pastebin the whole exten again |
| 02:43.30 | [TK]D-Fender | digime: Either way you "fixed it" and didn't show us what you did and considering how bright you were on the last fix, I trust this one even LESS |
| 02:43.33 | digime | ok |
| 02:43.44 | digime | thanks! |
| 02:43.47 | digime | I am trying.. |
| 02:43.55 | digime | pastebinning.. |
| 02:45.13 | [TK]D-Fender | drmessano: KY & Assembly not included |
| 02:45.19 | digime | http://pastebin.com/d4cff3fad |
| 02:45.59 | digime | oh i see |
| 02:46.04 | digime | i didnt put the full extension at the end |
| 02:46.47 | [TK]D-Fender | digime: NO |
| 02:47.07 | [TK]D-Fender | digime: you don't need to specify the extension. read the INSTRUCTS already... |
| 02:47.18 | digime | your're right, it's not the extension... |
| 02:47.47 | [TK]D-Fender | digime: And you're showing code without its EXECUTION. |
| 02:47.47 | digime | before I read the instructions.. can you tell me why your code isn't dialing the queue? |
| 02:47.59 | digime | i can show that |
| 02:48.13 | [TK]D-Fender | digime: and FFS pick a simple var, one without that dash as recommended. |
| 02:48.30 | [TK]D-Fender | digime: My "code" doesn't "DIAL" anything |
| 02:48.48 | seanbright | ~ffs |
| 02:48.49 | jbot | well, ffs is for f**k's sake, or for fine's sake. UCB's Fast File System |
| 02:49.10 | [TK]D-Fender | digime: I showed you a better GotoIf than what you coded. what makes you think I trust the CONTENT of that variable at the point of testing in the first place? |
| 02:49.14 | digime | here is the execution: |
| 02:49.20 | [TK]D-Fender | seanbright: Wanna bet its not the latter? :) |
| 02:49.26 | seanbright | heh |
| 02:49.30 | seanbright | that variable name is HORRIBLE |
| 02:49.42 | digime | I don't think that |
| 02:49.58 | digime | I am simply trying to understand why the queue extension is hitting voicemail |
| 02:50.12 | [TK]D-Fender | digime: its not an F-ING Queue extension! |
| 02:50.19 | [TK]D-Fender | digime: its 1 GOTOIF |
| 02:50.25 | digime | ok |
| 02:50.26 | seanbright | digime: pastebin the CLI output |
| 02:50.35 | seanbright | with 'core set verbose 10' |
| 02:50.36 | *** join/#asterisk Bad_Robot- (n=Bad_Robo@adsl-99-55-118-230.dsl.irvnca.sbcglobal.net) |
| 02:50.36 | [TK]D-Fender | digime: and the variable name is horrid. Go fix it |
| 02:51.40 | [TK]D-Fender | In fact even the Set() is pointless |
| 02:51.49 | [TK]D-Fender | nothing is reused in there at all |
| 02:52.19 | digime | http://pastebin.com/d513e41a7 |
| 02:52.48 | seanbright | digime: the value in the DB is blank |
| 02:53.16 | prodyan | guys, in which conf files do i configure an E1R2 connection? |
| 02:53.22 | seanbright | digime: what are the possible values in the DB and what do they mean? |
| 02:53.24 | [TK]D-Fender | digime: -- Executing [8804@101-queue-extensions:1] Set("SIP/101-8501-b7e107c8", "101-qs8804=") in new stack <--- yup... so... how's that working out for you? |
| 02:53.29 | [TK]D-Fender | </drphil> |
| 02:53.46 | seanbright | prodyan: that depends... what is an E1R2 connection? |
| 02:53.51 | [TK]D-Fender | prodyan: Unicall IIRC |
| 02:54.06 | seanbright | digime: please don't PM |
| 02:54.31 | prodyan | oki thanks, D-Fender |
| 02:54.53 | seanbright | digime: what are the possible values in the DB and what do they mean? |
| 02:55.23 | digime | I am not sure |
| 02:55.52 | seanbright | digime: alright. so who built this system initially? |
| 02:56.04 | digime | another programmer |
| 02:56.11 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 02:56.57 | seanbright | digime: run 'database show' at the CLI and pastebin the output |
| 02:58.08 | digime | shows my extensions, agents, etc. You want to see all? |
| 02:58.18 | digime | I don't actually use the queue service, I may just take it all out |
| 02:58.27 | seanbright | 'database show 101-qs' |
| 02:58.40 | digime | I think I know what's up: there is a queue service to de-activate queues. That is probably the database it is referencing |
| 02:58.51 | [TK]D-Fender | digime: exten => 8804,1,Set(101-qs8804=${DB(101-qs/8804)}) <--- where is this magical DB value being set from? |
| 02:59.09 | digime | 170*CLI> database show 101-qs |
| 02:59.25 | [TK]D-Fender | digime>I think I know what's up: there is a queue service to de-activate queues. That is probably the database it is referencing <-- this is not a "queue service" this is just DIALPLAN. that value had BETTER get set somewhere else... |
| 02:59.40 | seanbright | digime: you pasted the wrong line |
| 02:59.45 | digime | yep sorry /101-qs/8801 1 |
| 02:59.56 | seanbright | ok, so there is no value for 8804 |
| 02:59.57 | *** join/#asterisk outtolunc (n=me@c-67-188-204-139.hsd1.ca.comcast.net) |
| 03:00.05 | digime | ok |
| 03:00.20 | seanbright | if you want the dialplan you posted earlier to actually go to a queue, you need to do this |
| 03:01.07 | seanbright | database put 101-qs 8804 2 |
| 03:01.33 | digime | what does that command do? |
| 03:01.43 | seanbright | sets the values of 101-qs/8804 to "2" |
| 03:01.47 | [TK]D-Fender | digime: How is that value even supposed to have gotten set in the first place? |
| 03:02.15 | seanbright | the other programmer that initially built this probably built a way to set and update those values, however. |
| 03:02.18 | digime | I'm not sure |
| 03:02.29 | digime | yes, that sounds right |
| 03:03.01 | digime | What is this database used for and why is it needed? |
| 03:03.02 | [TK]D-Fender | digime: You are messing with code and you're not even backtracing. You are comparing stuff blind. Stop and go grab a coffee and find out where this junk gets set in the first place. You haven't even looked that far and a re running blind |
| 03:03.11 | [TK]D-Fender | digime: Messing with values like this is pointless |
| 03:03.24 | seanbright | but the first line of the dialplan you pasted pulls that value from the DB and if it is '2' then you go to the queue, if it's not '2' then you go to voicemail |
| 03:03.37 | *** part/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
| 03:03.38 | seanbright | pretty simple stuff |
| 03:03.39 | [TK]D-Fender | digime: this is YOUR DILAPLN. Go tell US what its fdoing. We're only psychic on TUESDAYS |
| 03:03.42 | digime | Actually the queue is working. that is not the issue, I am just trying to fix the error and understand more about what is happening |
| 03:03.57 | seanbright | digime: want to go line by line? |
| 03:04.03 | digime | sure |
| 03:04.11 | seanbright | this pastebin: http://pastebin.com/d4cff3fad |
| 03:04.14 | digime | ok |
| 03:04.28 | [TK]D-Fender | digime: Whats happening is you're testing the value of a DB key you have NO CLUE where or how it might be set for reasons you don't even understand |
| 03:04.50 | seanbright | line 1: assign the value from the database (101-qs/8804) to the (horribly named variable) 101-qs8804 |
| 03:04.57 | digime | ok |
| 03:06.01 | seanbright | line 2: if said value (the one stored in 101-qs8804) is not equal to '2' go to priority 8 in the current extension. if it *is* 2, just go to the next priority (priority 3) |
| 03:06.14 | digime | okay, and I get the rest |
| 03:06.20 | seanbright | ok, cool. |
| 03:06.20 | [TK]D-Fender | digime: http://pastebin.com/m65999ff0 |
| 03:06.26 | digime | okay |
| 03:06.36 | seanbright | heh |
| 03:06.46 | digime | the thing is, I may just rip out the database code and test it like that |
| 03:06.50 | [TK]D-Fender | digime: you're testing a value and you can't even say why... |
| 03:06.54 | seanbright | [TK]D-Fender: i think we've established that he doesn't know the answer to those questions. |
| 03:07.07 | digime | [TK]D-Fender: how are your comments helping? |
| 03:07.30 | drmessano | digime: How are your questions helping? |
| 03:07.34 | [TK]D-Fender | digime: It'll help if you stop looking at this one extension and look ELSEWHERE to see where it should be set. |
| 03:07.43 | [TK]D-Fender | digime: DB values don't pop in out of thin air |
| 03:07.55 | seanbright | digime: he's trying to tell you that you should try to understand the rest of your system, and not just this small part. |
| 03:07.56 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 03:07.58 | [TK]D-Fender | digime: they get set somewhere. |
| 03:08.07 | digime | Yes okay. I am looking now. |
| 03:08.38 | digime | I understand the logic, and I appreciate the guidance. |
| 03:08.48 | digime | I am looking at all the code. I will see what I can find |
| 03:09.29 | digime | drmessano: i thought this was a place where * users could ask questions? |
| 03:09.56 | seanbright | digime: don't go down this path |
| 03:09.57 | digime | i may have found it... |
| 03:10.08 | digime | ok |
| 03:10.30 | seanbright | digime: don't question anyone in here... it does more harm than good :) just friendly advice |
| 03:11.02 | digime | no I am not. I am just looking for help/support and to learn. |
| 03:11.24 | seanbright | digime: well we can only see what you show us |
| 03:11.27 | digime | I did not expect the reaction I got. But no harm. I appreciate it nonetheless. |
| 03:11.35 | digime | I understand. I am trying. |
| 03:12.39 | seanbright | digime: did this other developer build some kind of GUI to manage the queues? |
| 03:12.45 | digime | no |
| 03:13.02 | digime | Rather, I am not aware of one |
| 03:13.26 | [TK]D-Fender | ~aware |
| 03:13.28 | seanbright | digime: fire up a browser, go to http://<ip of phone system>/ |
| 03:13.46 | [TK]D-Fender | ~aware |
| 03:13.50 | [TK]D-Fender | ! rather |
| 03:13.52 | [TK]D-Fender | darnit |
| 03:14.02 | outtolunc | ~giggles |
| 03:14.05 | digime | seanbright: no, there is no GUI interface o the WAN side |
| 03:14.08 | [TK]D-Fender | seanbright: No, I highly doubt this is from a GUI |
| 03:14.17 | digime | correct, no GUI |
| 03:14.43 | seanbright | digime: then you want to check for cron jobs most likely |
| 03:14.48 | [TK]D-Fender | seanbright: The kind of crap you can only build by hand :) |
| 03:14.49 | TrentCreek | while on that...how about somone pass me a link to examples of PHP passing number values to Asterisk to use to dial out with? |
| 03:15.00 | [TK]D-Fender | TrentCreek: |
| 03:15.03 | [TK]D-Fender | ~book |
| 03:15.04 | jbot | extra, extra, read all about it, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 03:15.06 | [TK]D-Fender | ^^^^^^^ |
| 03:15.09 | [TK]D-Fender | TrentCreek: AGI <- |
| 03:15.15 | [TK]D-Fender | TrentCreek: Plenty of samples |
| 03:15.35 | [TK]D-Fender | TrentCreek: And massively Google-able |
| 03:15.46 | TrentCreek | I got the book,. The section covering PHP is only about 1 page. It covers mostly Perl |
| 03:15.50 | digime | seanbright: i have asterisk restart every day, that is the only cron job that i am aware of, related to asterisk |
| 03:16.29 | seanbright | digime: interesting. well there has to be something that changes the values in the asterisk DB. i'll let you search. |
| 03:17.32 | *** join/#asterisk voxter (n=voxter@ip67-88-206-99.z206-88-67.customer.algx.net) |
| 03:17.33 | digime | seanbright: either way, when I pulled the code out, errors are gone and queue works perfectly |
| 03:18.10 | seanbright | digime: well sure... if that is your goal, then you're good to go. |
| 03:18.26 | seanbright | digime: the original author had a way of toggling queue access on and off for some reason |
| 03:18.29 | [TK]D-Fender | TrentCreek: http://www.voip-info.org/wiki/view/Asterisk+AGI+php |
| 03:18.33 | seanbright | digime: if you don't need that anymore, then you're set. |
| 03:18.37 | digime | ultimately, yes. although I will look into the db issue more to find out what's happening. Y'all brought up a good point about it, I never even realized what it was doing |
| 03:18.41 | [TK]D-Fender | TrentCreek: 1st result of my Google Search. Put some effort in here... |
| 03:19.02 | digime | seanbright: yes, I am not looking to disable queues actually. although I found code for that |
| 03:19.14 | [TK]D-Fender | digime: this *IS* code like that |
| 03:19.20 | seanbright | digime: did you? where is that code? |
| 03:19.21 | digime | seanbright: so yes, he did put a "queue service aka qs" in there, that will allow you to turn the queue on or off |
| 03:19.28 | seanbright | ok |
| 03:19.30 | seanbright | err |
| 03:19.47 | seanbright | digime: really? what does that code look like? |
| 03:19.57 | digime | stand by |
| 03:20.49 | digime | http://pastebin.com/d3180c4e6 |
| 03:21.03 | seanbright | heh |
| 03:21.10 | seanbright | yeah |
| 03:21.12 | digime | there is the database reference again |
| 03:21.29 | seanbright | there is a number you can dial |
| 03:21.37 | digime | right |
| 03:21.41 | seanbright | and it will allow you to enable/disable the queue |
| 03:21.51 | seanbright | do you know what number that is? |
| 03:21.51 | digime | yes I found that extension as well |
| 03:22.05 | digime | yes but I went ahead and commented it out |
| 03:22.11 | seanbright | ... |
| 03:22.12 | [TK]D-Fender | WOW, its almost like obvious and stuff! And documented no less! |
| 03:22.12 | digime | I actually do not want to have that functionality |
| 03:22.20 | seanbright | gotcha |
| 03:22.25 | digime | okay cool, then |
| 03:22.33 | [TK]D-Fender | ~whee |
| 03:22.33 | jbot | [~whee] Weeeeeeeee! http://www.albinoblacksheep.com/flash/weeee |
| 03:22.34 | seanbright | but you could have just called the number and activated the queue |
| 03:22.39 | seanbright | which would have involved no code changes |
| 03:22.41 | seanbright | correct? |
| 03:22.46 | digime | let's see |
| 03:23.05 | digime | yes |
| 03:23.07 | digime | in theory, |
| 03:23.15 | digime | if he coded it correctly, it should work |
| 03:23.24 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 03:23.24 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:23.26 | seanbright | it appears that he coded it correctly |
| 03:23.28 | digime | I did not realize the queues would need to be activated in that way. But yes it makes sense. |
| 03:23.36 | digime | Although again, the queues have been working |
| 03:23.56 | digime | but your database command would probably have fixed it |
| 03:24.02 | seanbright | yeah, it would have. |
| 03:24.11 | seanbright | amour du code! je sus en trance |
| 03:24.12 | seanbright | err |
| 03:24.16 | seanbright | exten => 1,1,Set(DB(101-qs/8804)=2) |
| 03:24.32 | seanbright | that is the same as 'database put 101-qs 8804 2' |
| 03:24.35 | digime | aha. even better. okay. |
| 03:24.38 | digime | issue solved. seanbright, you are very patient, helpful and kind. I am very grateful for your time. |
| 03:24.47 | seanbright | digime: no sweat |
| 03:25.19 | digime | You are definately the nicest, kindest and most patient selfless service asterisk user I have met on here to date. Thank you. |
| 03:25.24 | seanbright | haha |
| 03:25.33 | seanbright | thanks :] |
| 03:25.36 | [TK]D-Fender | The code behind this really needs some abstraction... |
| 03:26.14 | outtolunc | ewwww, i think i stepped in som'tin |
| 03:26.19 | seanbright | heh |
| 03:26.22 | [TK]D-Fender | http://pastebin.com/d3180c4e6 <--- could have been made variable to a muti-tennent setup with 2 more lines. I hate to think how many times this stuff gets C&P'd |
| 03:26.52 | [TK]D-Fender | outtolunc: Call the cleaners before it sets in or you'll never get it out |
| 03:27.02 | outtolunc | actually i think it is really nice someone said 'thank you' afterward |
| 03:27.12 | seanbright | i do my multi-tenant stuff with setvar=ENTITY=CompanyA |
| 03:27.18 | seanbright | outtolunc: agreed. |
| 03:27.33 | [TK]D-Fender | seanbright: .... and get caught up to 1.4 at least :p |
| 03:27.33 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 03:28.10 | seanbright | yeah, the commas instead of () in the app calls is a little scary |
| 03:28.35 | [TK]D-Fender | seanbright: And I'm referring to your use of SetVar... the very start of digime's problem here today |
| 03:28.49 | seanbright | in sip.conf |
| 03:28.58 | seanbright | setvar=ENTITY=CompanyA |
| 03:29.08 | [TK]D-Fender | seanbright: BEttER :) |
| 03:29.10 | seanbright | not in dialplan |
| 03:29.22 | seanbright | i use AEL2 anyway |
| 03:29.27 | [TK]D-Fender | ooohh funky caps... |
| 03:29.29 | seanbright | cuz i'm hardcore |
| 03:29.31 | seanbright | heh |
| 03:29.35 | [TK]D-Fender | seanbright: EWWWWWWWW |
| 03:29.45 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 03:29.58 | seanbright | don't hate |
| 03:31.29 | [TK]D-Fender | lets the hate flow through him... |
| 03:31.40 | [TK]D-Fender | fries seanbright with FORCE-LIGHTNING! |
| 03:32.02 | [TK]D-Fender | grabs some marshmallows |
| 03:32.17 | seanbright | i can't wait for AEL3 |
| 03:32.45 | [TK]D-Fender | seanbright: The best trilogies come in 3's.... |
| 03:33.01 | [TK]D-Fender | seanbright: ... 15th time's the charm? |
| 03:33.04 | seanbright | heh |
| 03:33.25 | seanbright | if it requires you to quote your strings... that's good enough for me |
| 03:33.39 | seanbright | i guess i could just use pbx_lua |
| 03:33.41 | [TK]D-Fender | seanbright: Best 97 out of 193? |
| 03:33.59 | seanbright | 193 isn't divisable by 3 |
| 03:34.13 | [TK]D-Fender | seanbright: Separate joke. They are not mutually inclusive :p |
| 03:34.50 | seanbright | ah, i'm slow. |
| 03:34.52 | [TK]D-Fender | seanbright: If I could get away with porting my language's parser to * we would have typed vars & parameters. |
| 03:35.06 | seanbright | what is your language? |
| 03:35.19 | [TK]D-Fender | seanbright: One I wrote about 15 years ago in TP7 |
| 03:35.36 | seanbright | if the P stands for pascal this conversation is over. |
| 03:35.38 | seanbright | heh |
| 03:36.20 | [TK]D-Fender | seanbright: Easy enough to port to a better letter ;) |
| 03:36.32 | [TK]D-Fender | seanbright: Heck.. there is a script for that |
| 03:37.10 | seanbright | i was just kidding |
| 03:37.31 | [TK]D-Fender | seanbright: But I was able to deal with strings, floating, integer, hex & binary all with relative ease. |
| 03:37.54 | seanbright | would probably make more sense to switch to a different language binding altogether instead of updating AEL. something already established. |
| 03:38.31 | [TK]D-Fender | seanbright: Nope... problem with AEL is it gets parsed back to extensions.conf. that is the key problem |
| 03:38.38 | [TK]D-Fender | seanbright: THAT needs to get replaced |
| 03:38.53 | [TK]D-Fender | seanbright: http://pastebin.com/m7f07e79d <-- code sample and some templating for loops when I was working on that. |
| 03:39.49 | seanbright | wow |
| 03:39.51 | seanbright | cool |
| 03:40.02 | seanbright | what did you call it? |
| 03:40.20 | [TK]D-Fender | seanbright: All I would need to do is port a few odd pages of text parsing code I have that separates the parameters by type. Easy enough. |
| 03:40.26 | [TK]D-Fender | seanbright: "Acronym" |
| 03:40.31 | seanbright | catchy |
| 03:40.39 | [TK]D-Fender | seanbright: world's first Mid-Level Language :) |
| 03:40.53 | seanbright | i'd be happy with pbx_perl |
| 03:41.28 | seanbright | could make my entire dialplan a one-liner |
| 03:41.28 | seanbright | mmmm |
| 03:41.28 | seanbright | heh |
| 03:41.28 | [TK]D-Fender | seanbright: You can see I used ASM for my testing, a standard Application format, # for defines and a few other structures |
| 03:41.36 | [TK]D-Fender | ASM style that is |
| 03:41.39 | *** join/#asterisk xpot (n=jim@67.222.236.132) |
| 03:41.57 | [TK]D-Fender | #JE (Jump if Equal), #JNE,#JL, etc |
| 03:42.07 | russellb | or, you know, we could just put work into letting people use an existing language :-) |
| 03:42.16 | russellb | and stop trying to build and maintain our own |
| 03:42.32 | [TK]D-Fender | russellb: Not a bad idea :) |
| 03:42.40 | [TK]D-Fender | russellb: well... maybe :) |
| 03:42.51 | [TK]D-Fender | russellb: lets just say " complementary" |
| 03:42.58 | [TK]D-Fender | would be nice. |
| 03:43.12 | seanbright | russellb: perl pls |
| 03:43.16 | seanbright | kthxbai |
| 03:43.21 | russellb | perl is fine, i don't really care |
| 03:43.25 | [TK]D-Fender | russellb: Tricky part is the linking rights (c), etc. Then the long term support dependency. |
| 03:43.25 | russellb | one of the major ones would be preferable |
| 03:43.40 | russellb | and perhaps provide a few choices ... |
| 03:43.47 | [TK]D-Fender | russellb: Perl does probably have one of the best odds of longer term survivability |
| 03:44.04 | [TK]D-Fender | russellb: and a modicum of respectability. |
| 03:44.16 | [TK]D-Fender | russellb: lets say a layer better than AGI :) |
| 03:44.27 | russellb | nods |
| 03:44.55 | [TK]D-Fender | russellb: I live in a perfect world..... that dies with a buzzing sound at 7:0am |
| 03:45.01 | russellb | heh, yep |
| 03:46.25 | russellb | there are a lot of people that want to get this api 2.0 / pinemango / whatever you want to call it project off the ground and running |
| 03:46.32 | russellb | so it's possible a lot of this stuff could happen in 2009 |
| 03:46.34 | russellb | we'll see |
| 03:47.48 | [TK]D-Fender | russellb: Lets just say a bit more protocol control code would be nice. Like being able to define SIP progress like opting not to report "ringing" or "trying" and leaving those to the dialplan to progress through. Better compatibilit with other SIP solutions. |
| 03:47.58 | [TK]D-Fender | russellb: and then actually implementing SIP-B ;) |
| 03:48.38 | [TK]D-Fender | russellb: OH and with that custom progress option, that allows you to choose to "404" not just on dialplan match... |
| 03:48.42 | [TK]D-Fender | russellb: even BETTER |
| 03:49.02 | [TK]D-Fender | russ this allows a RADICAL new option for "invalid" handling |
| 03:49.48 | [TK]D-Fender | russellb: You know I can see alot of EASY ways to make this possible... |
| 03:49.55 | russellb | heh |
| 03:50.07 | file | if you know the cause code that maps to 404 you could probablyyyyyyy already make the dialplan send a 404 |
| 03:51.01 | [TK]D-Fender | russellb: [contextname&] <- "&" signifying that devices dumping calls into this exten immediately go to exten "s,1" and allow Dialplan apps to provide call progress updates |
| 03:51.49 | [TK]D-Fender | file: This is if it doesn't matcha dialplan pattern it repots 404. this idea is if you want to THINK ABOUT IT first and then later to decide to report 404. |
| 03:51.59 | file | [TK]D-Fender: yes, Hangup(1) |
| 03:52.35 | [TK]D-Fender | file: That doesn't 404 :) You get a "trying" first. You saying that a straight Hangup will follow with a 404? |
| 03:53.40 | file | Hangup(1) will cause chan_sip to send back a 404 |
| 03:54.08 | file | chan_sip probably would send a 100 Trying before it though |
| 03:54.20 | file | shouldn't matter though |
| 03:54.22 | [TK]D-Fender | file: that is in itself interesting. A bonus if you can more tightly control the preceeding messages. Seen a few people come in here trying to do jsut that. |
| 03:54.36 | file | that depends on what you mean by control |
| 03:54.37 | [TK]D-Fender | file: Well all know how assy some UA's can be :) |
| 03:54.54 | [TK]D-Fender | Where RFC meets NFC :) |
| 03:55.08 | file | you can get chan_sip to do such things, Ringing will cause a 180 Ringing to get sent and Progress will cause 183 Session Progress |
| 03:55.57 | [TK]D-Fender | file: I guess this is all fine & dandy... following getting SIP-B *hint* |
| 03:56.12 | [TK]D-Fender | wave's a 100$ bill for grabs if it gets in soon |
| 03:56.24 | file | shrugs |
| 03:56.30 | seanbright | heh |
| 03:56.48 | seanbright | $100 whole dollars!? |
| 03:56.59 | seanbright | make it $100,000 and i'll learn how to program |
| 03:57.06 | *** join/#asterisk k-man (n=jason@unaffiliated/k-man) |
| 03:57.31 | k-man | what is a good ATA to get? |
| 03:57.31 | [TK]D-Fender | seanbright: And when we see that your spaghetti didn't come with meatballs... how much to make you STOP? :p |
| 03:57.39 | [TK]D-Fender | k-man: Linksys |
| 03:57.44 | [TK]D-Fender | k-man: for the most-part |
| 03:57.48 | k-man | [TK]D-Fender: thanks |
| 03:57.54 | [TK]D-Fender | k-man: Long time no see... |
| 03:58.10 | k-man | [TK]D-Fender: yeah |
| 03:58.17 | [TK]D-Fender | k-man: Been about a year, no? |
| 03:58.32 | k-man | i'm a #asterisk floozy - i only come here when i need something ;) |
| 03:58.47 | k-man | [TK]D-Fender: nah, i was here a few months ago asking about SIP phones |
| 03:59.03 | k-man | [TK]D-Fender: unless you are confusing me with someone else? |
| 03:59.22 | [TK]D-Fender | k-man: Few moths may as well be a year for me :) |
| 03:59.27 | [TK]D-Fender | k-man: Aussie, right? |
| 03:59.39 | [TK]D-Fender | months* |
| 04:00.22 | k-man | [TK]D-Fender: yeah, aussie |
| 04:00.28 | k-man | [TK]D-Fender: where are you from? |
| 04:00.42 | [TK]D-Fender | k-man: See... still remember (barely). |
| 04:00.47 | [TK]D-Fender | <-Canuckian |
| 04:01.08 | k-man | [TK]D-Fender: where is that? |
| 04:01.16 | [TK]D-Fender | k-man: Canada :) |
| 04:01.24 | k-man | oh |
| 04:01.32 | k-man | didn't know that slang |
| 04:01.43 | [TK]D-Fender | k-man: Take it in the same vein as "kiwi" |
| 04:01.50 | k-man | i visited canada to go skiing once - loved it |
| 04:02.18 | k-man | went to blackcomb/whistler |
| 04:02.32 | k-man | i hear its even nicer if you go to the less popular resorts |
| 04:03.44 | [TK]D-Fender | k-man: All variable... mind you what you marvel at I despise and am getting buried in now :) |
| 04:04.11 | k-man | [TK]D-Fender: yeah, im sure its very different living there. where abouts do you live? |
| 04:04.14 | [TK]D-Fender | k-man: I need to go for a trip to the outback & NZ.... |
| 04:04.27 | [TK]D-Fender | k-man: Montreal, QC. |
| 04:04.42 | k-man | [TK]D-Fender: yeah, you should - if you like walking, do the Overland Track in Tasmania |
| 04:05.42 | [TK]D-Fender | k-man: Just added to my new list. will see how things shap up next year |
| 04:06.24 | [TK]D-Fender | shape* |
| 04:06.44 | k-man | yeah - i never managed to get asterisk working at home |
| 04:06.48 | [TK]D-Fender | k-man: You got some Linksys SPA phones already, didn't you? |
| 04:06.59 | [TK]D-Fender | k-man: No? After all this time? |
| 04:07.08 | k-man | well - i gave up |
| 04:07.12 | k-man | its a long story |
| 04:07.22 | [TK]D-Fender | k-man: Were you fighting with Telestra for the card's lack of certs / disconnect issues? |
| 04:07.41 | [TK]D-Fender | k-man: Trying to remember back here |
| 04:08.04 | mosty | what is the benefit of using Gosub instead of a macro? does it just handle nesting better? |
| 04:08.06 | k-man | but essentially, i wanted to have my billion voip router continue with the voip service and develop asterisk internaly - once i had asterisk working i was going to switch the billion over to accessing asterisk instead of the voip provider |
| 04:08.12 | k-man | [TK]D-Fender: no, that wasn't me |
| 04:08.29 | sah-work | so do sangoma fsX cards show up in ifconfig like the pri ones? |
| 04:08.46 | k-man | [TK]D-Fender: but i couldn't get asterisk to talk to a voip server from behind my billion modem - then i got busy and gave up |
| 04:09.22 | mosty | sah-work, yes |
| 04:09.22 | [TK]D-Fender | sah-work: I believe you get the same w1d1 style entry IIRC |
| 04:09.41 | sah-work | hum, okay i pulled an analog card from a box |
| 04:09.44 | sah-work | then put it back |
| 04:09.44 | [TK]D-Fender | k-man: Yeah, * behind a SIP routing device usually gets FUBAR'd |
| 04:09.48 | sah-work | lspci shows it |
| 04:10.08 | sah-work | but nothing in ifconfig and put back the old zap* files chokes on start |
| 04:10.10 | [TK]D-Fender | sah-work: "wanrouter hwcheck" |
| 04:10.20 | mosty | k-man, beware that some billion routers have SIP ALG, which tends to hurt more than it helps. disable it if you can |
| 04:10.26 | k-man | [TK]D-Fender: its hard for me to justify bringing down the voip line to work on it when it works ok as it is - and my wife uses it all the time |
| 04:10.33 | [TK]D-Fender | sah-work: And you seem to be ignoring the most important part... wanpipe |
| 04:10.38 | sah-work | yes |
| 04:10.38 | k-man | what is ALG? |
| 04:10.42 | mosty | sah-work, did you try the troubleshooting steps on sangoma's wiki? |
| 04:10.50 | [TK]D-Fender | k-man: yu0p, makes testing a trickiet thing for sure |
| 04:10.55 | sah-work | hum, i removed a card and rebooted |
| 04:10.59 | sah-work | then i put it back |
| 04:11.00 | sah-work | and rebooted |
| 04:11.08 | sah-work | i would like to think it should just work |
| 04:11.16 | mosty | k-man, http://www.voip-info.org/wiki/view/Routers+SIP+ALG |
| 04:11.17 | [TK]D-Fender | sah-work: and make no mention of check wanpipe's status anywhere |
| 04:11.26 | sah-work | 1 . AFT-A200-SH : SLOT=4 : BUS=4 : IRQ=137 : CPU=A : PORT=PRI : HWEC=32 : V=10 |
| 04:11.27 | sah-work | 2 . AFT-A102-SH : SLOT=1 : BUS=1 : IRQ=153 : CPU=A : PORT=1 : HWEC=64 : V=31 |
| 04:11.27 | sah-work | 3 . AFT-A102-SH : SLOT=1 : BUS=1 : IRQ=153 : CPU=A : PORT=2 : HWEC=64 : V=31 |
| 04:11.29 | sah-work | so it is there |
| 04:11.39 | [TK]D-Fender | sah-work: "wanrouter status" |
| 04:11.40 | jql | application level gateway. a firewall that tries to be a transparent sip proxy |
| 04:11.44 | jql | evil bastards |
| 04:12.15 | sah-work | hum, see wanpipe1/2 on the A102 card |
| 04:12.29 | k-man | mosty, thanks |
| 04:12.37 | sah-work | realizes he does not even remember which is what card |
| 04:13.00 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 04:13.22 | sah-work | okay it does not see the card |
| 04:13.30 | sah-work | but lspci does. strange |
| 04:15.12 | [TK]D-Fender | sah-work: Go check your configs. |
| 04:15.18 | sah-work | checking |
| 04:16.48 | sah-work | getting fustrated ; Configuring interfaces: w3g1 w3g1: unknown interface: No such device |
| 04:17.09 | mosty | sah-work, configure wanpipe again |
| 04:17.37 | sah-work | thanks. doing it |
| 04:21.19 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 04:21.29 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 04:22.17 | *** join/#asterisk kerx (n=prepro@adsl-69-105-21-113.dsl.irvnca.pacbell.net) |
| 04:22.48 | sah-work | okay, same thing - Configuring interfaces: w3g1 w3g1: unknown interface: No such device |
| 04:22.56 | sah-work | does this mean the driver did not load? |
| 04:25.01 | *** join/#asterisk harry_v (n=lork@S010600a0c93f6f7e.vs.shawcable.net) |
| 04:28.48 | *** join/#asterisk farkus_ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 04:29.21 | afink | can anyone please help me get to the bottom of why everytime I get a call on my PRI it goes offline? What should I be checking to find the problem? |
| 04:29.49 | k-man | so which linksys ATA should i get if I just plan to use it to connect an analogue phone to asterisk? |
| 04:30.39 | mosty | k-man, the cheapest one with an FXO port |
| 04:30.58 | mosty | pap2 or 2102 (from memory) |
| 04:31.04 | k-man | ok, thanks |
| 04:31.10 | mosty | 3102 if you want to connect it to an analogue phone line also |
| 04:31.20 | k-man | no, we have no analogue line |
| 04:31.27 | k-man | we are free of the telstra tax |
| 04:32.51 | mosty | actually- i was not concentrating before, i meant FXS (not FXO) |
| 04:34.24 | k-man | can you put asterisk on a device that can run openwrt? |
| 04:35.29 | mosty | yes, i believe that they have packaged it. don't try to transcode though |
| 04:35.55 | drmessano | 3102 is FXO |
| 04:36.00 | drmessano | 3102 is FXO/FXS |
| 04:36.14 | drmessano | PAP2 is FXS/FXS and SPA-2102 is FXS/FXS |
| 04:39.31 | *** join/#asterisk Fr0Gs (n=Dean@39.79.96.58.exetel.com.au) |
| 04:39.46 | Fr0Gs | hey guys, what is the limitation on the sip protocol with conferences |
| 04:39.50 | Fr0Gs | what is the max amount of users |
| 04:40.02 | Fr0Gs | allowed in one call |
| 04:40.36 | drmessano | 32,000 or so |
| 04:41.25 | Fr0Gs | lol |
| 04:41.29 | Fr0Gs | serious |
| 04:41.35 | drmessano | I was |
| 04:41.42 | Fr0Gs | wouldent the server die |
| 04:41.45 | Fr0Gs | under the stress |
| 04:41.56 | drmessano | facepalms |
| 04:42.08 | drmessano | Yeah dude, you're limited by your hardware |
| 04:42.13 | Fr0Gs | yea |
| 04:42.15 | drmessano | Isn't that obvious? |
| 04:42.20 | Fr0Gs | of course |
| 04:42.28 | Fr0Gs | but i was just wondering because ive never been on a sip server |
| 04:42.29 | drmessano | You asked about the SIP protocol |
| 04:42.31 | Fr0Gs | that could support more then 8 |
| 04:42.47 | drmessano | Sounds like a crappy server |
| 04:42.55 | drmessano | 8 is pretty low metric |
| 04:43.23 | *** part/#asterisk Fr0Gs (n=Dean@39.79.96.58.exetel.com.au) |
| 04:43.54 | drmessano | um ok |
| 04:46.04 | digime | seanbright: still there? |
| 04:46.13 | seanbright | digime: maybe |
| 04:46.34 | [TK]D-Fender | RUN FORREST RUN!!! |
| 04:47.33 | seanbright | digime: what's up? |
| 04:47.35 | digime | seanbright: do you have experience setting up qos and traffic shaping? |
| 04:47.56 | seanbright | digime: i do not, no. |
| 04:48.11 | digime | seanbright: no harm, it was a technical question, nvrmind |
| 04:48.23 | seanbright | others in here might, though. |
| 04:49.01 | digime | seanbright: i do have another question. i set up an after hours queue so that we could pick up calls and it works, however, at times it will call a cell number and immediately hit the cell's voicemail, even though the cell is on and has a good signal. any ideas? |
| 04:49.04 | [TK]D-Fender | minds |
| 04:49.07 | [TK]D-Fender | :p |
| 04:49.37 | digime | seanbright: note that it is only happening with a particular cell number, other numbers have no issues. I am very puzzled by this. |
| 04:49.38 | [TK]D-Fender | digime: digime cell has an issue obviously and nothing to do with * |
| 04:49.59 | seanbright | digime: yeah, what [TK]D-Fender said. it's a cell/provider thing. |
| 04:50.28 | digime | seanbright: it is not my cell, but either way, what happens is, it doesn't give the cell time to ring, or the cell will ring for half a second and then hit the voicemail. |
| 04:50.45 | [TK]D-Fender | digime: Same answer |
| 04:50.48 | digime | seanbright: but the cell will work perfectly with a normal agent login |
| 04:51.00 | digime | seanbright: that is, if I log the cell into a queue, it works every time |
| 04:51.18 | digime | seanbright: but if I log in my after hours queue, which includes several cells and a SIP phone, it does not |
| 04:51.22 | [TK]D-Fender | digime: and what is a "normal" login? |
| 04:51.49 | digime | okay: I can log in via ACD by dialing an agent number and password and inputting the cell # |
| 04:52.06 | [TK]D-Fender | digime: PSTN based agents require CONFIRMATION otherwise you run into real problems. "core show application dial" <- M() |
| 04:52.55 | digime | okay, does this mean that my after hours extension, which has several cell #'s in it, is not a good idea? |
| 04:53.28 | [TK]D-Fender | digime: Its not a good idea period. but sometimes unavoidable |
| 04:53.33 | digime | hmm |
| 04:53.40 | [TK]D-Fender | digime: aI just gave you the solution you should aim for. Get reading |
| 04:54.00 | digime | okay, my end goal: I want to log in 3 or 4 cell phones into all the queues after hours |
| 04:54.25 | digime | yes i am reading that. lots of options there |
| 04:55.09 | [TK]D-Fender | digime: and I just handed yout he one to use on a PLATTER |
| 04:55.35 | digime | dial command. okay, thanks |
| 04:57.42 | digime | so it's not a provider issue, it's an * issue actually after all |
| 04:58.11 | digime | or rather, the requirements for pstn agents |
| 04:59.03 | [TK]D-Fender | digime: No. Nothing * can do will force the Cell VM to grab the call instantly. |
| 04:59.11 | [TK]D-Fender | digime: You are not the boss of the Cell Company |
| 04:59.36 | [TK]D-Fender | digime: That phone will right as long as the cell company tries to. |
| 04:59.45 | *** join/#asterisk farkus_ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 04:59.55 | seanbright | stabs farkus_ in the face |
| 04:59.56 | [TK]D-Fender | digime: because It is in control of that VM box. |
| 05:00.22 | seanbright | farkus_: where the eff are you? the ISS? |
| 05:00.39 | seanbright | or just stealing a neighbor's wifi? |
| 05:01.02 | digime | but again, why would the cell grab the call and push it to voicemail like that? |
| 05:02.46 | seanbright | digime: it's the provider, not the phone |
| 05:03.22 | seanbright | digime: the call is "ringing" at the telco before your cell starts ringing |
| 05:03.59 | seanbright | if that happens for a long enough time (3-4 rings) it's going to go to VM, regardless of when the cell actually started ringing |
| 05:04.02 | [TK]D-Fender | digime: Ask your cell company. |
| 05:04.59 | digime | seanbright: okay. But it only happens in a certain circumstance with *. With other situations involving this cell and *, no issues |
| 05:05.40 | seanbright | digime: it happens all the time with my cell. asterisk or not. |
| 05:05.46 | [TK]D-Fender | digime: Nothing * does controls how this phone drops to VM |
| 05:05.48 | digime | but I can reproduce the issue |
| 05:06.04 | seanbright | digime: strange. |
| 05:06.10 | seanbright | digime: don't really know then |
| 05:06.10 | digime | If I log the same cell into my queues, it will be fine and perfect |
| 05:06.15 | digime | ok |
| 05:06.41 | digime | what I am doing is this: after hours, I am logging in an extension. |
| 05:06.46 | digime | this extension has the following line: |
| 05:07.20 | digime | exten => 8527,4,Dial(SIP/101-8702&${101-ITSP1}/number1&${101-ITSP1}/number2,30,rt) |
| 05:07.34 | digime | so number1 works but number2 is the cell with the problem |
| 05:07.49 | digime | the SIP extension also works fine, obviously |
| 05:08.37 | digime | is there an inherent problem with this? am I allowed to login a 4 digit extension that then calls SIP and multiple cell #'s when dialed? |
| 05:08.48 | seanbright | if you change the order to number2&number1 does the problem happen with number1? |
| 05:08.52 | *** join/#asterisk Defraz (n=T0tal@24-117-156-21.cpe.cableone.net) |
| 05:09.29 | digime | no |
| 05:09.31 | digime | i tried that |
| 05:09.42 | seanbright | weird |
| 05:09.46 | seanbright | all things point to cell/telco |
| 05:09.47 | digime | yep |
| 05:09.49 | digime | ok |
| 05:09.56 | seanbright | even this --> |
| 05:09.59 | seanbright | see? pointing |
| 05:10.06 | digime | heh |
| 05:10.16 | digime | frustrating though |
| 05:10.21 | seanbright | i hear ya |
| 05:10.29 | seanbright | i guess it could be asterisk, but i don't know how |
| 05:10.45 | seanbright | esp. when switching the numbers around doesn't have the same effect |
| 05:10.54 | digime | but then, why would this number2 work when I use an agent login |
| 05:11.18 | digime | I think it's something about how it is dialing, I still think it's the * side somewhere |
| 05:11.24 | seanbright | shrugs |
| 05:11.27 | seanbright | i'm off to bed |
| 05:11.28 | digime | ok |
| 05:11.30 | seanbright | g'night folks |
| 05:11.32 | digime | thankyou!!! |
| 05:11.34 | digime | good man!! |
| 05:11.45 | seanbright | no problem. |
| 05:11.54 | [TK]D-Fender | digime: What agent login? SHOW US |
| 05:12.01 | digime | ok |
| 05:12.17 | digime | agent => 1,1,Agent 23 ; after hours queue |
| 05:12.55 | [TK]D-Fender | digime: no. the DIALPLAN show us this phone "logging in". WShow us the sucessful call, and the failed call |
| 05:13.02 | digime | stand by |
| 05:18.08 | digime | hmm can't replicate the issue right now, it seems to be working |
| 05:19.43 | digime | although the cli is showing something strange |
| 05:22.12 | digime | http://pastebin.com/d5b2b294c |
| 05:22.27 | digime | it says the agent answered, but actually on my end it just rang |
| 05:24.21 | *** join/#asterisk steerpike (n=Unknown@unaffiliated/steerpike) |
| 05:24.42 | steerpike | hi, where can i get cheap asterisk hosting? |
| 05:27.02 | digime | it is doing some kind of loopback! it is answering on the same line i am calling in on! |
| 05:27.38 | digime | whatever number I call in on, the CLI outputs that it is answering on that extension or IP address! |
| 05:29.31 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 05:30.11 | trnzmeta | guys: I want asterisk to play a beep every x seconds during outgoing phone calls to indicate the convo is being recorded |
| 05:30.14 | trnzmeta | what should I google? |
| 05:30.38 | *** join/#asterisk thansen (n=thansen@7.247.sfcn.org) |
| 05:30.41 | [TK]D-Fender | digime: -- Executing [8527@101-acd-dialplan:1] Answer("Local/8527@101-acd-dialplan-db13,2", "") in new stack <-- YOU issued the ANSWER |
| 05:30.55 | [TK]D-Fender | digime: Behold the glory of line # 28 |
| 05:31.18 | [TK]D-Fender | trnzmeta: "core show application dial" <- L() |
| 05:31.34 | *** join/#asterisk freakazoid0223 (n=mattc@pool-68-238-182-170.phil.east.verizon.net) |
| 05:31.55 | ricko73 | there is no way to send a beep every X seconds...not without rewriting the application |
| 05:32.05 | *** join/#asterisk Pagautas (n=bigman@ns.voip.ktu.lt) |
| 05:32.24 | [TK]D-Fender | ricko73: Dial supports this |
| 05:32.31 | digime | line #28, so its an issue with my dialplan? |
| 05:32.40 | [TK]D-Fender | digime: Look at the "big print" |
| 05:32.42 | trnzmeta | hmm maybe I can send a beep at the beginning of conversation |
| 05:32.57 | [TK]D-Fender | trnzmeta: You can do what you requested. |
| 05:33.44 | ricko73 | [TK]D-Fender: where? I know you can send a courtesy beep that only you hear when recording is started...(that's in features.conf) |
| 05:34.54 | [TK]D-Fender | ricko73: Nobody said it was on a TRIGGERED recording. His statement can be interpreted that the ENTIRE call should be beeping. |
| 05:35.40 | ricko73 | and how do you propose sending a beep every X seconds during a call |
| 05:35.49 | [TK]D-Fender | ricko73: On the premise that all calls are recorded |
| 05:35.55 | ricko73 | ah ok |
| 05:36.08 | [TK]D-Fender | [00:31]<[TK]D-Fender>trnzmeta: "core show application dial" <- L() |
| 05:36.24 | [TK]D-Fender | Does nobody pay attention when I hand answers outright anymore? |
| 05:36.59 | trnzmeta | raises hand slowly... and hides in back of lecture hall |
| 05:37.32 | ricko73 | admits he had his head on the desk... |
| 05:37.34 | digime | [TK]D-Fender: what big print, please |
| 05:37.52 | [TK]D-Fender | digime: Your CLI output clearly shows you calling dialplan that answers the Agent channel |
| 05:37.59 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 05:38.12 | [TK]D-Fender | digime: its the first bloody line of that exten even. |
| 05:38.30 | [TK]D-Fender | [00:30]<[TK]D-Fender>digime: -- Executing [8527@101-acd-dialplan:1] Answer("Local/8527@101-acd-dialplan-db13,2", "") in new stack <-- YOU issued the ANSWER |
| 05:38.32 | [TK]D-Fender | ^^^ |
| 05:38.33 | digime | that's right but I use the same dialplan for other agents and I don't have the same issue |
| 05:38.39 | [TK]D-Fender | digime: Blatantly visible |
| 05:38.58 | [TK]D-Fender | digime: well you wanted to know why it said "answered", well there it is |
| 05:39.09 | digime | okay |
| 05:39.14 | [TK]D-Fender | digime: And you aren't showing me anything to compare against either |
| 05:39.30 | digime | true |
| 05:45.38 | digime | okay it's a dialplan issue. I get that. Now how to proceed? |
| 05:48.37 | [TK]D-Fender | digime: Don't like that Answer there? REMOVE IT |
| 05:48.39 | digime | i think i get it |
| 05:49.13 | digime | My agent is set to dial one number and instead I am telling it to dial an extension which in turn dials more numbers after that. not what it was intended for |
| 05:49.20 | digime | no i love your answers! |
| 05:50.54 | *** part/#asterisk steerpike (n=Unknown@unaffiliated/steerpike) |
| 05:51.14 | [TK]D-Fender | digime: Your queue effectively is not a Queue at all |
| 05:51.42 | [TK]D-Fender | digime: It is a "hey lets just chain a local channel which I could have reached with a GOTO with a lot less effort" |
| 05:51.42 | digime | okay |
| 05:52.03 | digime | how would the GOTO statement be used in this case? |
| 05:52.55 | [TK]D-Fender | digime: Forget that. I don't know what you think this is SUPPOSED to do. |
| 05:53.19 | [TK]D-Fender | digime: Nothing seems to work the way you think and you don't seem to see it until I point it out to you. |
| 05:53.24 | digime | I can tell you what I am aiming at: I want to login one extension that will then in turn dial a set of cell numbers |
| 05:53.45 | [TK]D-Fender | digime: You have never expressed what you you want to have happen EXACTLY so we can't tell that what is happening now is WRONG for that goal. |
| 05:54.06 | digime | how exact do you wish me to be? |
| 05:54.26 | [TK]D-Fender | digime: With your record for pretty weak detail, impress me... |
| 05:56.47 | digime | I want to login and activate an agent that will be associated with multiple queues. That agent will have several pots numbers attached to it. When someone calls into our queues, the agent will pass the call to the various pots lines |
| 05:58.10 | [TK]D-Fender | digime: are these various POTS numbers actually calling the same person via multiple different numbers SIMULTANEOUSLY? |
| 05:58.45 | [TK]D-Fender | digime: Normally a queue calss *1* agent at a time via a single call. And proceeds from one to the other in rotation |
| 05:58.58 | digime | round robin? |
| 05:59.28 | digime | what if I want a queue to call multiple agents simultaneously, and whoever picks up the call, wins |
| 05:59.37 | [TK]D-Fender | digime: Typically. The point of a queue is ensuring a call gets answered. managing calling multiple devices at once gets tricky for reasons like that VM kicking off.. |
| 05:59.50 | digime | aha |
| 05:59.53 | [TK]D-Fender | digime: Doing that while including PSTN #'s = PAIN |
| 05:59.59 | digime | aha |
| 06:00.03 | digime | that's what i am after though |
| 06:00.08 | [TK]D-Fender | digime: This is asking for SEVERE pain. |
| 06:00.15 | digime | hmm |
| 06:00.50 | [TK]D-Fender | digime: You need to dial each of those external #'s using the L() option as I stated long ago so that they are forced to CONFRIM receipt of the call so VM doesn't grab them |
| 06:01.53 | digime | I have never heard of L(). can you explain its usage and provide a context as an example |
| 06:02.13 | [TK]D-Fender | sorry, M() |
| 06:02.21 | [TK]D-Fender | digime: I gave you that answer hours ago |
| 06:02.40 | digime | it was unclear to me, sorry |
| 06:03.19 | [TK]D-Fender | [23:52]<[TK]D-Fender>digime: PSTN based agents require CONFIRMATION otherwise you run into real problems. "core show application dial" <- M() |
| 06:03.31 | [TK]D-Fender | Only a little over an hour ago... God time is passing slow. |
| 06:04.26 | k-man | anyway |
| 06:04.29 | digime | you have explained its usage well, and I understand that. But I am unclear as to how to implement it |
| 06:04.31 | k-man | thanks guys- see you around |
| 06:04.33 | *** part/#asterisk k-man (n=jason@unaffiliated/k-man) |
| 06:04.53 | [TK]D-Fender | digime: Go read the instructions. |
| 06:05.13 | digime | I have. it is unclear to me. |
| 06:05.34 | *** part/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 06:05.48 | *** join/#asterisk Segnale007 (n=Pietro@host92-254-dynamic.35-79-r.retail.telecomitalia.it) |
| 06:06.12 | *** join/#asterisk robbiet480 (n=robbiet4@c-24-4-144-97.hsd1.ca.comcast.net) |
| 06:06.13 | [TK]D-Fender | digime: You poit it to a dialplan macro. In your macro you play a prompt, read input from the user with a time delay (core show application read) and if they press "1" for example you set the valu to tell Dial to have treated the call as "answered" |
| 06:06.32 | digime | aha. I know what you are talking about, actually. |
| 06:06.32 | robbiet480 | hey does anyone have a good callthrough example. the one on voip-info isnt working correctly w/ gizmo out |
| 06:06.47 | digime | You are asking for the cell caller to confirm before answering the call. hmm... |
| 06:06.58 | digime | rather the pstn number to confirm |
| 06:07.25 | [TK]D-Fender | digime: Not quite. The cell answers, but DIAL does not consider it "answered" until the callee confirms it in the macro. |
| 06:07.33 | digime | okay. yes. that makes sense. |
| 06:08.24 | digime | and you want that for all the agents that dial pstn lines. |
| 06:09.03 | digime | what if two pstn lines answer at once, is it whoever confirms the call first gets the call? |
| 06:09.28 | [TK]D-Fender | digime: Should |
| 06:09.59 | [TK]D-Fender | digime: Of course you need EACH person being called simultaneously to have their own M() opportunity. |
| 06:10.13 | digime | so because i am not doing that, the multiple pstn devices are creating some issue |
| 06:10.14 | [TK]D-Fender | digime: Which means you need to dial them as separate Local channels. |
| 06:10.25 | rue_mohr | [TK]D-Fender, what happens if an extension has no answer() before it dial() s? does the call connect? |
| 06:10.30 | [TK]D-Fender | digime: One VM kicks in, you = FUBAR'd |
| 06:10.44 | digime | yes i have seen that already |
| 06:10.49 | digime | all the other callers get shut down |
| 06:10.57 | [TK]D-Fender | rue_mohr: call progress is passed through to the calling channel |
| 06:11.12 | [TK]D-Fender | digime: Which is why you need to confirm it or you're DOA |
| 06:11.36 | digime | yes i am having that exact issue now |
| 06:12.22 | digime | I suppose there is no other way.. and I will say that when I login a single individual pstn line i have 0 issues. really! |
| 06:12.49 | [TK]D-Fender | digime: First... there is NO "login" there. |
| 06:12.57 | [TK]D-Fender | digime: this is just calling a stupid local channel. |
| 06:13.17 | [TK]D-Fender | digime: The fact that the agent # gets pointed to that exten & context doesn't even matter |
| 06:13.26 | [TK]D-Fender | digime: FINE. * calls that local channel. |
| 06:13.31 | [TK]D-Fender | digime: that isn't the issue |
| 06:13.46 | [TK]D-Fender | digime: the nature of your outbound calls is |
| 06:14.12 | digime | okay |
| 06:15.08 | digime | yes it calls a local channel, but that local channel has the various pstn numbers attached to it, that is the problem |
| 06:15.45 | rue_mohr | do you need to make sure caller x dosn't go out at "y" ? |
| 06:16.14 | digime | i shouldnt be doing it that way at all. what you want is to give each pstn number its own local channel |
| 06:16.38 | [TK]D-Fender | [01:09]<[TK]D-Fender>digime: Of course you need EACH person being called simultaneously to have their own M() opportunity. |
| 06:16.39 | [TK]D-Fender | ^^^^^^ |
| 06:16.46 | [TK]D-Fender | Could have sworn I said that already... |
| 06:17.01 | carrar | say it again for extra measure! |
| 06:17.04 | digime | so when i login my agent, would the local channel then dial the other local channels? |
| 06:18.16 | *** join/#asterisk Defraz (n=T0tal@24-117-156-21.cpe.cableone.net) |
| 06:18.25 | [TK]D-Fender | digime: Dial(Local/number1@contextwithmatchandM&(Local/number2@contextwithmatchandM&(Local/number3@contextwithmatchandM&SIP/100,40) |
| 06:18.36 | digime | okay |
| 06:19.04 | digime | now that is an answer! thank you! |
| 06:19.30 | [TK]D-Fender | digime: lose the extra ('s I C&P'd there |
| 06:19.36 | digime | ok |
| 06:19.41 | [TK]D-Fender | digime: Dial(Local/number1@contextwithmatchandM&Local/number2@contextwithmatchandM&Local/number3@contextwithmatchandM&SIP/100,40) |
| 06:19.45 | digime | sure |
| 06:20.05 | rue_mohr | [dest_dan] exten => s,1,Dial,Zap/2&Zap/6&Zap/4|30 |
| 06:20.34 | [TK]D-Fender | digime: Only really requires 1 extra context thatwill have a waide-range pattern macth to dial out the # requested with M() |
| 06:20.52 | digime | great. I am going to give it a try. |
| 06:20.53 | [TK]D-Fender | digime: And you'll need to make the actual Macro that will prompt for confirmation. |
| 06:21.03 | digime | sure |
| 06:21.07 | digime | and record the voice for it |
| 06:21.19 | [TK]D-Fender | rue_mohr: .... and that is? |
| 06:21.34 | harry_v | TK, Is it possible to dump a current two way call into conferance? |
| 06:21.45 | rue_mohr | I was comparing, sorry, yours has something new |
| 06:21.48 | harry_v | Without the caller calling back |
| 06:22.01 | [TK]D-Fender | harry_v: Nope. Every related means bunrs off one end of the call |
| 06:22.08 | digime | [TK]D-Fender: thank you very much. I will let you know how it goes! |
| 06:22.17 | harry_v | okay |
| 06:23.31 | rue_mohr | so you would need to split the call and converence them both? |
| 06:23.52 | [TK]D-Fender | rue_mohr: 2 words : Not. Happening. |
| 06:23.57 | rue_mohr | and accept that all the kings horses and all the kings men cant put your call back togethor again? |
| 06:24.18 | rue_mohr | hmm, the state machine is already done with the call? |
| 06:25.30 | [TK]D-Fender | rue_mohr: Not saying that it is not "possible" in the sense taht you COULD go and write some raw C code to try and ID the connected call to hijack each side. Just that that is exactly what it would take. C code. |
| 06:26.01 | [TK]D-Fender | rue_mohr: Because every other means of trying to toss a channel one way or another drop the other end of what they were doing like a rock. |
| 06:26.02 | robbiet480 | hey does anyone have a good callthrough example. the one on voip-info isnt working correctly w/ gizmo out |
| 06:26.21 | robbiet480 | or even better |
| 06:26.26 | robbiet480 | how do i strip a # off a number |
| 06:26.27 | [TK]D-Fender | robbiet480: Maybe you should be looking at MAKING it work "correctly" and perhaps showing us whats happening. |
| 06:26.39 | [TK]D-Fender | robbiet480: PASTEBIN is your friend |
| 06:26.42 | [TK]D-Fender | ~pb |
| 06:26.43 | 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, or , http://bin.cakephp.org/ |
| 06:26.48 | robbiet480 | yeah i know what a pastebin is |
| 06:26.59 | *** join/#asterisk JohnnyBeGood (n=JohnnyBe@c-98-232-40-217.hsd1.wa.comcast.net) |
| 06:27.00 | [TK]D-Fender | robbiet480: Can't be too safe around here |
| 06:27.03 | robbiet480 | sure |
| 06:27.16 | robbiet480 | all i need to know is how to strip one character off the end |
| 06:27.19 | robbiet480 | and then its all good |
| 06:27.32 | [TK]D-Fender | robbiet480: And stripping digits off a var is 101 stuff... go read channelvariables.txt or the WIKI page on Asterisk Variables |
| 06:27.40 | [TK]D-Fender | robbiet480: thats IT!? |
| 06:27.40 | robbiet480 | ok |
| 06:27.52 | robbiet480 | [TK]D-Fender: yeah. gizmo doesnt play nice with the # |
| 06:27.53 | [TK]D-Fender | robbiet480: which end. Every value has TWO. |
| 06:28.02 | rue_mohr | [TK]D-Fender, it would be like a modified transfer |
| 06:28.12 | robbiet480 | ok so the number is 1925XXXYYYY |
| 06:28.21 | robbiet480 | and when i hit enter to start the call when connected to asterisk |
| 06:28.29 | [TK]D-Fender | robbiet480: What var is the number stored in? |
| 06:28.32 | robbiet480 | asterisk dials 1925XXXYYYY#@proxy01.sipphone.com |
| 06:28.43 | [TK]D-Fender | robbiet480: and which digit in that sample to dyou want stripped off? |
| 06:28.43 | robbiet480 | ${EXTEN:1} |
| 06:28.47 | robbiet480 | just the # |
| 06:28.56 | [TK]D-Fender | robbiet480: is the "#" always present? |
| 06:29.02 | robbiet480 | yeah |
| 06:29.08 | robbiet480 | because thats how asterisk knows to start the dialout |
| 06:29.32 | rue_mohr | do calls consists of two data paths, 1 in and 1 out? |
| 06:29.35 | robbiet480 | heres the full code |
| 06:29.36 | robbiet480 | http://www.voip-info.org/wiki-Asterisk+tips+call+through |
| 06:30.10 | [TK]D-Fender | robbiet480: ${EXTEN:1:$[${LEN(${EXTEN})} - 2]} |
| 06:30.19 | robbiet480 | awesome |
| 06:30.21 | robbiet480 | lemme test |
| 06:30.33 | [TK]D-Fender | rue_mohr: A call is 2 bridged channels |
| 06:30.43 | rue_mohr | ok |
| 06:30.45 | [TK]D-Fender | rue_mohr: the concept of "in" and "out do not exist. |
| 06:31.05 | robbiet480 | [TK]D-Fender: exten => #,1,Dial(SIP/${EXTEN:1:$[${LEN(${EXTEN})} - 2]}@proxy01.sipphone.com,20,r) |
| 06:31.06 | [TK]D-Fender | rue_mohr: every channel is jsut a channel. |
| 06:31.07 | robbiet480 | that look ok? |
| 06:31.14 | rue_mohr | I asked that way because of the way * records calls |
| 06:31.46 | rue_mohr | so, in theroy you could have a ring, where each person can only hear the person to their 'right' |
| 06:32.03 | [TK]D-Fender | rue_mohr: never just say "* records calls". More like how DIAL hooks in to record |
| 06:32.26 | robbiet480 | [TK]D-Fender: didnt output anything |
| 06:32.30 | [TK]D-Fender | rue_mohr: Do not overassociate some specific apps and their functionality to the raw nature of 2 bridged channels |
| 06:32.33 | robbiet480 | To: <sip:proxy01.sipphone.com>; |
| 06:32.36 | [TK]D-Fender | robbiet480: pastebin... |
| 06:32.38 | robbiet480 | k |
| 06:33.06 | [TK]D-Fender | robbiet480>[TK]D-Fender: exten => #,1,Dial(SIP/${EXTEN:1:$[${LEN(${EXTEN})} - 2]}@proxy01.sipphone.com,20,r) <-- robbie... there IS NO NUMBER HERE |
| 06:33.14 | [TK]D-Fender | robbiet480: thats just a # |
| 06:33.19 | robbiet480 | http://pastebin.ca/1286637 |
| 06:33.22 | [TK]D-Fender | robbiet480: where is the NUMBER? |
| 06:33.37 | robbiet480 | [TK]D-Fender: its passed from dialed input |
| 06:33.45 | robbiet480 | ill paste the full config again |
| 06:33.49 | robbiet480 | http://www.voip-info.org/wiki-Asterisk+tips+call+through |
| 06:33.58 | robbiet480 | thats it exactly minus your change |
| 06:34.00 | [TK]D-Fender | robbiet480: Where? You are having us compare ${EXTEN} . that is "#" here! |
| 06:34.22 | robbiet480 | ok look at that code from voip-info |
| 06:34.35 | robbiet480 | it oringinally uses ${TRUNK} but im running on a VPS |
| 06:34.36 | [TK]D-Fender | exten => #,1,Dial(SIP/${EXTEN:1:$[${LEN(${EXTEN})} - 2]}@proxy01.sipphone.com,20,r) <- ${EXTEN} here only holds the "#" symbol |
| 06:34.37 | robbiet480 | so i dont have zap |
| 06:34.50 | robbiet480 | ok so what do i need to do then |
| 06:35.01 | [TK]D-Fender | robbiet480: Make up your mind and substitute the var where apppropriate |
| 06:35.25 | robbiet480 | ok listen |
| 06:35.31 | rue_mohr | <PROTECTED> |
| 06:35.36 | robbiet480 | the # in this case tells asterisk to parse everything BEFORE # |
| 06:35.39 | rue_mohr | <PROTECTED> |
| 06:35.41 | robbiet480 | and then pass to the Dial command |
| 06:35.56 | [TK]D-Fender | robbiet480: I see no DIALPLAN in your CLI output, nor the raw code |
| 06:36.09 | [TK]D-Fender | robbiet480: and your description is broken |
| 06:36.28 | [TK]D-Fender | robbiet480>the # in this case tells asterisk to parse everything BEFORE # <- what tells you this? or us? |
| 06:37.11 | robbiet480 | [TK]D-Fender: the voip-info article i keep posting does |
| 06:37.19 | [TK]D-Fender | robbiet480: show me YOUR CODE. |
| 06:37.28 | robbiet480 | its exactly the same but ok... |
| 06:37.42 | [TK]D-Fender | robbiet480: I don't want some other story I can't trust you copies letter for letter esp after saying you took in changes I gave you. |
| 06:37.50 | [TK]D-Fender | robbiet480: Don't make us run around to help you. |
| 06:38.04 | [TK]D-Fender | awaits pastebin #2 |
| 06:38.09 | robbiet480 | http://pastebin.ca/1286641 |
| 06:38.48 | [TK]D-Fender | robbiet480: You don't need to strip the # |
| 06:38.49 | *** join/#asterisk Astlinux_Fan (n=chatzill@c-76-28-233-22.hsd1.wa.comcast.net) |
| 06:39.07 | [TK]D-Fender | robbiet480: Because you are collecting digits in an IVR into the var NR |
| 06:39.20 | robbiet480 | [TK]D-Fender: when i check sip debug output it is trying to call 1XXXYYYZZZZ#@proxy01.sipphone.com |
| 06:39.44 | [TK]D-Fender | robbiet480: And whats really sad is this is a complete reinvention of the READ COMMAND that would have this whole thing done in *1 line* |
| 06:39.53 | robbiet480 | wtf really |
| 06:39.59 | robbiet480 | i was just using the stuff from the wiki |
| 06:39.59 | robbiet480 | ok |
| 06:40.25 | [TK]D-Fender | robbiet480: exten => #,1,Dial(SIP/${NR}@proxy01.sipphone.com,20,r) |
| 06:40.40 | [TK]D-Fender | robbiet480: there. You wre collecting digits into NR. FFS USE IT :p |
| 06:40.45 | robbiet480 | sorry |
| 06:40.46 | robbiet480 | lol |
| 06:40.48 | robbiet480 | thanks for the help |
| 06:41.09 | robbiet480 | lemme test |
| 06:41.17 | [TK]D-Fender | robbiet480: exten => _X,1,Set(NR=${NR}${EXTEN}) <-- you really should look at what the important bits are doing |
| 06:41.31 | [TK]D-Fender | robbiet480: Every digit = add to the end of NR |
| 06:41.51 | [TK]D-Fender | robbiet480: "#" never gets added. that jsut says "dial what was collected" |
| 06:42.11 | [TK]D-Fender | robbiet480: "core show application read" <-- this is what your IRV basically reinvents |
| 06:42.35 | [TK]D-Fender | robbiet480: Now I have done similar things in cases where my client required "*" as the terminating char, etc. |
| 06:42.55 | [TK]D-Fender | robbiet480: However yours is a complete reinvention. |
| 06:43.05 | robbiet480 | i was just copy pasting |
| 06:43.10 | robbiet480 | like a nub lol |
| 06:43.49 | [TK]D-Fender | robbiet480: Well there were 2-3 things to learn from this. Hopefully you got them all. |
| 06:43.51 | *** join/#asterisk scardinal (n=supreme@90.184.100.170) |
| 06:43.56 | robbiet480 | i think i did |
| 06:44.05 | [TK]D-Fender | robbiet480: I can live with that. |
| 06:47.54 | [TK]D-Fender | Ok.... way late here... checkout time. Back in the morning. |
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| 07:19.22 | yang | I have this dialplan - http://pastebin.ca/1286664 , I would like extension 64 to ring if extension 60 is busy, but not wait 7 seconds for this to happen. |
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| 08:49.14 | Karlitoo | good morning all |
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| 09:10.51 | Karlitoo | I can not make an outgoing call i always get == Everyone is busy/congested at this time (1:0/0/1) |
| 09:10.51 | Karlitoo | <PROTECTED> |
| 09:11.16 | yang | Karlitoo: paste your dialplan |
| 09:11.20 | yang | ~b |
| 09:11.22 | yang | ~pb |
| 09:11.23 | 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, or , http://bin.cakephp.org/ |
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| 09:19.24 | Karlitoo | yang, http://pastebin.com/m3313215 |
| 09:19.45 | Karlitoo | that is the dialplan that has only 5 lines that are added by me all the reas are sample |
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| 09:20.40 | Karlitoo | rest* |
| 09:20.49 | yang | Karlitoo: are you using freepbx? |
| 09:21.28 | Karlitoo | * 1.4 |
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| 09:24.32 | kippi | hey |
| 09:25.06 | kippi | I have a linksys spa942, can I get information from it for example when someone logs in? |
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| 09:53.20 | Qas | Hello every one |
| 09:53.31 | Qas | I have a problem with my asterisk server |
| 09:53.53 | Qas | we cant make any outgoing calls or recevie any incoming calls our internal system is fine |
| 09:54.13 | Qas | can any one help me out with it |
| 09:54.25 | yang | Qas: you (most probably) need to create extensions, do you have any ? |
| 09:55.28 | Qas | yes |
| 09:55.30 | Qas | plenty |
| 09:55.49 | yang | so paste your extensions and errors to pastebin |
| 09:55.55 | Qas | I have around 100 extensions all working internally absoluty fine |
| 09:56.05 | yang | jbot: tell Qas about pb |
| 09:56.06 | Qas | ok |
| 10:01.31 | Qas | yang: pastebin link tail /var/log/asterisk/full |
| 10:02.59 | Qas | yang: http://pastebin.com/m4292a93b http://pastebin.com/d40e05b8d |
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| 10:04.54 | yang | Qas: dial in to your VOIP number, paste the CLI output and extensions |
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| 10:05.16 | *** join/#asterisk stimpie (n=stimpie@84-104-5-227.cable.quicknet.nl) |
| 10:06.59 | Qas | yang: ok let me try |
| 10:08.37 | *** join/#asterisk ido (n=ido@unaffiliated/ido) |
| 10:10.42 | Qas | yang: where can I find CLI and paste what in th extensions |
| 10:11.02 | Qas | can you explain me a bit more, Thank you |
| 10:12.22 | Subdolus | Is it possible to make Asterisk continue down the extension command list after a call picks up? |
| 10:12.34 | Subdolus | like Dial(number) |
| 10:12.46 | Subdolus | and then senddtmf(555) |
| 10:12.52 | h-idrisi | Qas, from command line type asterisk -vr and dial and paste the output |
| 10:13.50 | yang | Qas: you get CLI with "asterisk -vvvvr" for verbose output |
| 10:14.33 | Qas | ok thanx yang and h- |
| 10:15.33 | Qas | MY paste link http://pastebin.com/d57262698 |
| 10:15.50 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 10:16.07 | Subdolus | No? |
| 10:16.44 | yang | Qas: Dial in and paste the output again, don't forget to paste extensions to a separate link |
| 10:16.59 | Qas | One question do I have to have channels to make outgoing calls or receive incoming calls from outside |
| 10:17.00 | yang | extensions.conf |
| 10:17.40 | yang | Qas: you need an incoming and outgoing dialplan (extensions) |
| 10:17.53 | Qas | ok |
| 10:18.14 | Qas | does it take long to create or I can create it quickly? |
| 10:18.30 | yang | We haven't seen your dialplan YET |
| 10:19.18 | Qas | how can I show it to you |
| 10:19.39 | yang | upload it to pastebin |
| 10:19.44 | h-idrisi | Qas, what you mean by ( do I have to have channels ) |
| 10:20.23 | Qas | Yesterday I couldnot load my asterisk manager so i deleted channels that I previously have in zapata.cnf file |
| 10:20.27 | yang | Qas: you wrote that you have 100 extensions and that your internal calls work, so paste those lines |
| 10:20.53 | Qas | can you tell me how to get the lines please |
| 10:21.06 | yang | <PROTECTED> |
| 10:21.07 | Qas | i am new in Voip so dont know much |
| 10:21.09 | kerx | how can asterisk handle bad responses from SIP provider? |
| 10:21.10 | Qas | ok |
| 10:21.14 | Qas | thanks yang: |
| 10:23.10 | Qas | yang: h-isrisi: - patse link is http://pastebin.com/d296edf75 |
| 10:23.42 | yang | Qas: FREEpbx ? |
| 10:23.48 | yang | ~freepbx |
| 10:23.49 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 10:24.50 | Qas | ok thanks your help so far yang: |
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| 10:29.06 | kerx | is there anything that i can use to handle bad sip responses from a provider to re-try the call? |
| 10:31.19 | Qas | hi yang do you know any FreePBX support company |
| 10:33.16 | yang | hm |
| 10:33.29 | yang | Qas: maybe those are mentioned on freepbx.org website? |
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| 10:36.05 | tokozedg | guys what does this means? |
| 10:36.13 | tokozedg | << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/10-ac0dd1b0] |
| 10:38.25 | tokozedg | i get this message when i answer |
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| 10:45.42 | *** join/#asterisk invalidrecord (n=fares@87.113.88.31.plusnet.pte-ag2.dyn.plus.net) |
| 10:46.24 | invalidrecord | hi can anyone help me with a realtime error, it looks like the sql is messed up but thats hard to belive considering its compiled into asterisk |
| 10:46.38 | invalidrecord | ~ask |
| 10:46.38 | jbot | somebody said ask was 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. |
| 10:46.58 | invalidrecord | i keep getting:WARNING: nonstandard use of \\ in a string literal |
| 10:46.59 | invalidrecord | LINE 1: SELECT * FROM extensions WHERE exten LIKE '\\_%' AND context... |
| 10:48.14 | yang | I have this dialplan - http://pastebin.ca/1286664 , I would like extension 64 to ring if extension 60 is busy, but not wait 7 seconds for this to happen. |
| 10:50.54 | Subdolus | No way to continue down the extension after Dial() picks up? |
| 10:51.39 | Subdolus | If not, what's the best method for entering a pin after it picks up. I need to space the DTMFs out because of a semi-dodgey line |
| 10:55.58 | angryuser | Subdolus: try Read() |
| 10:59.08 | *** join/#asterisk henk (n=hank@netwichtig.de) |
| 10:59.09 | henk | moin |
| 10:59.22 | *** join/#asterisk ZX81 (n=matt@202.20.97.211) |
| 11:01.48 | mort_gib | moin |
| 11:01.55 | henk | i'm trying to queue a caller in one queue and if no one answers in another, looks like that atm: http://paste.debian.net/23796/ the timeout value seems to get ignored. the call stays in the queue about 3 minutes afaict and not 10 seconds. what do i do? |
| 11:02.49 | mort_gib | henk: make sure that it times out once and not a number of times |
| 11:03.30 | *** join/#asterisk Silicium (n=marco@mail.2am.ch) |
| 11:03.32 | Silicium | hi there |
| 11:03.46 | ZX81 | anyone know anything about xorcom astribanks? |
| 11:04.00 | invalidrecord | does no one in here use realtime i have been asking for daqys now |
| 11:04.02 | invalidrecord | days |
| 11:04.05 | Silicium | omg |
| 11:04.05 | henk | mort_gib: what are you trying to say? |
| 11:04.09 | invalidrecord | WARNING: nonstandard use of \\ in a string literal |
| 11:04.10 | invalidrecord | LINE 1: SELECT * FROM extensions WHERE exten LIKE '\\_%' AND context... Anyone? |
| 11:04.11 | angryuser | ZX81: hello tzafrir is an expert ;) |
| 11:04.14 | ZX81 | heh |
| 11:04.15 | ZX81 | yeah |
| 11:04.19 | Silicium | first i read are metaquestions |
| 11:04.20 | ZX81 | silent expert :) |
| 11:04.30 | Silicium | zomgthing :D |
| 11:04.37 | angryuser | ZX81: describe your problem |
| 11:04.39 | ZX81 | have a hotel owner on the phone wanting to know why his hotel has no phones :) |
| 11:04.46 | ZX81 | it locks with a firware error |
| 11:04.51 | ZX81 | trying to start zaptel |
| 11:04.58 | mort_gib | henk: I did something like what you are trying to do, Reception (two girls) are in the same queue, 4 more staff members double as reception if queue one is full or not responding |
| 11:04.58 | ZX81 | NOTICE-xpp: XBUS-02(00): FIRMWARE: ERROR_CODE CODE = 0x3 (Premature packet end) (rate_limit=3) |
| 11:05.21 | angryuser | ZX81: have you tryed to get latest firmware and load it manually ? |
| 11:05.24 | invalidrecord | hunt group |
| 11:05.32 | mort_gib | henk: initially the second queue never received the call as the first queue timed out some 5 times.... |
| 11:05.41 | ZX81 | using: /usr/src/zaptel-1.4/kernel/xpp/utils/xpp_fxloader load |
| 11:05.50 | ZX81 | says --------- FIRMWARE LOADING: (load) [0 devices] |
| 11:05.50 | ZX81 | Got all 0 devices |
| 11:05.50 | ZX81 | --------- FIRMWARE IS LOADED |
| 11:06.00 | ZX81 | and yet there are devices |
| 11:06.17 | ZX81 | they show up with that error in "tail -f /var/log/messages" |
| 11:07.16 | angryuser | ZX81: have you tryed to power off astribank to clear the firmware ? |
| 11:07.19 | henk | mort_gib: what do you mean with 'timed out 5 times'? |
| 11:07.44 | ZX81 | angryuser: nah is it volatile? |
| 11:07.45 | mort_gib | henk: Aparently I had accepted the default, which means that it times out a few times. |
| 11:08.08 | henk | mort_gib: could you stop repeating the same phrase i don't understand? |
| 11:08.16 | mort_gib | henk: It's a while back, but it took me a while to figure out, because I didn't set that option in the queue |
| 11:08.23 | henk | mort_gib: 'times out a few times'? how can one call timeout more than once? |
| 11:08.36 | henk | mort_gib: what option? |
| 11:08.40 | Silicium | hmm, when i redirect a ISDN channel from S0 over a BRI Card to another asterisk Server |
| 11:08.41 | mort_gib | henk: retry the queue, if you wish |
| 11:08.55 | ZX81 | is getting him to try |
| 11:08.55 | Silicium | does misdn modify the signal? |
| 11:09.18 | henk | mort_gib: you are not helping by confusing me. and that is what you are doing. wtf are you talking about? |
| 11:09.22 | ZX81 | Silicium, in that it uses an alaw signal yeah |
| 11:09.26 | ZX81 | not clear data |
| 11:09.42 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 11:09.53 | ZX81 | ooh looks like fw is volatile |
| 11:09.56 | mort_gib | henk: okay suit yourself |
| 11:10.06 | henk | mort_gib: sorry, perhaps i'm just too dumb to get what you are trying to say. |
| 11:10.20 | Silicium | ZX81: hmm how you mean that? |
| 11:10.47 | henk | or it's the understanding of english of either of us that's the problem. i just don't understand what you are trying to tell me... |
| 11:11.06 | ZX81 | Silicium: well it depends what you want to do |
| 11:11.19 | ZX81 | in terms of a call from one device to another, no it doesn't change |
| 11:11.23 | Silicium | h324m |
| 11:11.25 | *** join/#asterisk tyuil (n=chatzill@bgl93-3-82-230-208-124.fbx.proxad.net) |
| 11:11.26 | ZX81 | but if say you wanted to do a 324 |
| 11:11.28 | ZX81 | heh |
| 11:11.32 | ZX81 | yeah |
| 11:11.33 | tyuil | hi |
| 11:11.33 | ZX81 | so it does change |
| 11:11.35 | ZX81 | :) |
| 11:11.37 | Silicium | damn |
| 11:11.40 | Silicium | damn damn |
| 11:11.43 | ZX81 | have a look at asterisk-video |
| 11:11.49 | ZX81 | it doesn't have to |
| 11:11.50 | Silicium | iam on the list |
| 11:11.55 | mort_gib | henk: http://pastebin.com/d3aee2e1a |
| 11:12.00 | ZX81 | search for sergio's posts |
| 11:12.01 | Silicium | hmm |
| 11:12.18 | tyuil | <PROTECTED> |
| 11:12.21 | Silicium | i will try using a direct connection |
| 11:12.31 | Silicium | tyuil: fail, OSX is worst |
| 11:12.38 | tyuil | when i try to connect to asterisk it display |
| 11:12.51 | tyuil | yes the worst os of work |
| 11:13.01 | tyuil | yes the worst os of the world |
| 11:13.06 | Silicium | it display "Error, OSX found, checkout OpenBSD" |
| 11:13.27 | Silicium | and with verbose 6 it also displays HUMPPA |
| 11:13.30 | Silicium | :D |
| 11:13.32 | Silicium | sorry |
| 11:13.32 | tyuil | it display |
| 11:14.29 | henk | mort_gib: does 'retry' affect the timout option of the Queue command? o_O |
| 11:14.32 | tyuil | asterisk -c |
| 11:14.33 | tyuil | Asterisk 1.4.20.1, Copyright (C) 1999 - 2008 Digium, Inc. and others. |
| 11:14.34 | tyuil | Created by Mark Spencer <markster@digium.com> |
| 11:14.36 | tyuil | Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. |
| 11:14.38 | tyuil | This is free software, with components licensed under the GNU General Public |
| 11:14.40 | tyuil | License version 2 and other licenses; you are welcome to redistribute it under |
| 11:14.41 | tyuil | certain conditions. Type 'core show license' for details. |
| 11:14.43 | tyuil | ========================================================================= |
| 11:14.43 | Silicium | ohnoez |
| 11:14.44 | henk | 'sigh' |
| 11:14.44 | tyuil | [ Booting... |
| 11:14.46 | tyuil | [ Reading Master Configuration ] |
| 11:14.47 | tyuil | [ Initializing Custom Configuration Options ] |
| 11:14.49 | tyuil | Unable to open pid file '/opt/local/var/run/asterisk.pid': No such file or directory |
| 11:14.50 | tyuil | Unable to bind socket to /opt/local/var/run/asterisk.ctl: No such file or directory |
| 11:14.52 | Silicium | pastebin... |
| 11:14.52 | tyuil | Unable to open logger.conf: No such file or directory; default settings will be used. |
| 11:14.53 | tyuil | [Dec 16 12:07:44] ERROR[15947]: logger.c:615 init_logger: Unable to create event log: No such file or directory |
| 11:14.55 | tyuil | what to do ? |
| 11:14.56 | henk | KICK! |
| 11:15.11 | Silicium | touch /etc/asterisk/logger.conf? |
| 11:15.19 | tyuil | no |
| 11:15.31 | tyuil | i done nothing |
| 11:15.40 | mort_gib | henk: I'm not an expert on queues but until I changed retry to 0 it kept !"£%%$^ retrying never going to the next queue, who will wait for 3 minutes ++ |
| 11:16.00 | tyuil | i don't know why i got the error ? |
| 11:16.14 | tyuil | can anyone help iher ? |
| 11:16.22 | Silicium | tyuil: AFAIK an OSX Problem |
| 11:16.35 | henk | mort_gib: ah ok, thanks :) i'll try that! |
| 11:17.04 | Silicium | or PEBKAC |
| 11:17.44 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 11:17.48 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:19.40 | *** join/#asterisk Subdolus (n=subby@subby.afraid.org) |
| 11:20.19 | kerx | is there anything that i can use to handle bad sip responses from a provider to re-try the call? |
| 11:20.30 | kerx | i am sending call via AMI |
| 11:20.36 | kerx | any suggestions would be appreciated |
| 11:23.13 | *** join/#asterisk tokozedg (n=toka@85.118.98.122) |
| 11:29.05 | henk | mort_gib: it seems timeout AND retry are needed... |
| 11:29.26 | mort_gib | henk: Yes, but you can set retry to 0 |
| 11:29.39 | henk | yes of course... timeout probably as well. |
| 11:32.42 | henk | mort_gib: retry isn't even needed. just timeout. and that can be set to 0 as well... |
| 11:32.51 | henk | oh wait a sec |
| 11:33.44 | henk | mort_gib: yes. that way it works. timeout = 0 is everything that's needed. |
| 11:34.02 | mort_gib | Which Is what I wrote here, retrying |
| 11:34.45 | henk | oh, right. you said retry has to be 0, not that it has to be set at all. sorry |
| 11:35.49 | mort_gib | Yees.... |
| 11:51.03 | mort_gib | henk: I think that my initial problem was that I didn't set it and hence didn't understand why it was retrying, which was a default setting above in the file... |
| 11:54.35 | *** join/#asterisk MindTheGap (n=MindTheG@201.80.82.57) |
| 11:59.30 | yang | I have this dialplan - http://pastebin.ca/1286664 , I would like extension 64 to ring if extension 60 is busy, but not wait 7 seconds for this to happen. |
| 12:03.01 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 12:08.44 | *** join/#asterisk tmjb (n=tmjb@212.200.239.230) |
| 12:11.57 | mort_gib | yang: exten => s,n,Set(CallCount=${SIPPEER(SIP/60|curcalls)}) |
| 12:13.10 | tmjb | http://bugs.digium.com/view.php?id=13488 any solved this? I have the same problem maybe some know third party patch ? |
| 12:14.14 | yang | mort_gib: In which line do i substitute that? |
| 12:14.47 | mort_gib | yang: You set CallCount to the amount of calls that SIP device is in... |
| 12:14.56 | yang | curcalls stays "curcalls" or does it get substituted by a number? |
| 12:15.09 | mort_gib | yang: remember to set type=peer in sip.conf |
| 12:15.39 | yang | could you fix that dialplan and upload a working example, please? |
| 12:15.46 | yang | I kinda lost you there |
| 12:15.55 | mort_gib | yang: currcals is the variable you are asking for google asterisk SIPPEER if in doubt |
| 12:16.05 | mort_gib | I can upload something I use... |
| 12:16.39 | *** join/#asterisk propellerhead (n=yogurt2u@host172.190-138-93.telecom.net.ar) |
| 12:16.43 | yang | ok |
| 12:17.31 | mort_gib | yang: do you understand this: http://pastebin.com/d4f249ff3 |
| 12:18.02 | yang | ok i am gonna try it |
| 12:18.18 | mort_gib | This example "transfers" the second call into vm, but feel free to change that to dialing another extension... |
| 12:18.33 | yang | yes, thanks |
| 12:19.04 | yang | how come you don't have the Answer() section =? |
| 12:19.11 | mort_gib | Not needed |
| 12:19.24 | tyuil | ok thx Silicium |
| 12:20.08 | henk | mort_gib: ok, thanks! |
| 12:20.15 | henk | mort_gib: answer() is not needed? |
| 12:20.16 | mort_gib | np :-) |
| 12:20.27 | henk | mort_gib: when is a call actually established then? |
| 12:20.44 | mort_gib | Eh, not for what yang wanted... For queues I think it is!! |
| 12:20.58 | henk | mort_gib: not regarding queues, but generally. |
| 12:21.25 | mort_gib | Like playing back something, you need answer first |
| 12:21.56 | henk | mort_gib: ah ok, in the example you give the call is established (as in: the phone company will charge you for it) as soon as the Dial-command is answered by a phone. correct? |
| 12:22.24 | mort_gib | Generally I don't use Answer(), but for certain things you need it |
| 12:22.59 | mort_gib | Yeah, when I have an incoming call on say zaptel or Woomera that call is still ringing until it's picked up |
| 12:24.11 | henk | ok, nice ;) |
| 12:24.14 | mort_gib | So if I go exten => 20072036,1,Dial(SIP/110,30,Tt) then the call should not be charged until SIP/110 picks up |
| 12:24.29 | mort_gib | But that would be if you trust the telco's.... |
| 12:24.36 | henk | yes, that's perfect :) |
| 12:24.58 | mort_gib | Which I happen not to.... :-) |
| 12:25.03 | henk | 'g' |
| 12:25.05 | henk | me neither |
| 12:25.40 | mort_gib | Local guys are crap, like they just added 200 in front of all number, so I arranged an update to all dialplans out there |
| 12:26.00 | mort_gib | Only they still send the 72036 down the ISDN lines... |
| 12:26.06 | henk | so if i want to playback a message to every caller i will have to answer first and so the call is being charged even when no one actually answers it. just for playing back some stuff, right? |
| 12:26.22 | mort_gib | Right |
| 12:26.36 | mort_gib | You "pick up the call to playback" |
| 12:26.37 | henk | hm so it's a rather stupid idea... |
| 12:26.50 | henk | not for us but for our callers... |
| 12:26.54 | mort_gib | Answer, no, it VERY useful |
| 12:27.21 | mort_gib | We are after all talking rather low charges |
| 12:27.26 | henk | no, answer is sensible. i mean 'playing back a message when someone calls no matter what time or who can answer the call anyway' |
| 12:27.40 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 12:27.54 | mort_gib | henk: This will be ALL new to you, but NOBODY likes talking machines!!! :-) |
| 12:28.05 | henk | my boss does 'sigh' |
| 12:28.23 | mort_gib | Ah, no rule without exception.... |
| 12:28.34 | henk | but he also thinks fax over voip will work with the right codec... |
| 12:29.12 | henk | and that the ruin right across the street is actually a hotel only because there is a sign saying so... |
| 12:29.21 | mort_gib | Ha ha ha! Glad I'm not in your shoes, although, if that's what he thinks you can do whatever to faxing, he wont know the difference! |
| 12:30.24 | mort_gib | brb |
| 12:30.25 | henk | that was some time back when a colleague and i switched the company's phone system to asterisk. he said 'its definitely a codec problem'. since then i know: he's a moron. |
| 12:31.00 | henk | or rather: since 5 hours after him saying so and us becoming stupid searching the web for a solution. |
| 12:31.40 | *** join/#asterisk Segnale007 (n=Pietro@host92-254-dynamic.35-79-r.retail.telecomitalia.it) |
| 12:34.54 | *** join/#asterisk DarkRift (n=dark@bas1-sthubert21-1096591606.dsl.bell.ca) |
| 12:35.08 | yang | mort_gib: does this looks allright (I still don't know what to put in the 1st Verbose extension - http://pastebin.ca/1286787 |
| 12:35.49 | yang | mort_gib: you can update the paste below |
| 12:43.30 | *** join/#asterisk SibRphrek (i=SibrPhre@cpe-67-243-43-136.nyc.res.rr.com) |
| 12:49.00 | mort_gib | yang: Hmm, what happens if SIP/64 is unavailable ?? |
| 12:49.24 | yang | Actually then it needs to go to SIP/63 for 30 seconds and hangup |
| 12:49.57 | mort_gib | Ever heard of Voicemail, quite popular with my uses :-) |
| 12:50.14 | yang | yes, but my boss doesn't like it |
| 12:50.41 | yang | I took my example from the VoiceMail config, trying to set it to Dial a second extension now |
| 12:50.56 | mort_gib | Ah, so the next logical question is, -Do you like your boss?? |
| 12:51.27 | yang | Oh well, who likes their boss? |
| 12:51.39 | mort_gib | Or more to the point, who will like phone to just go dead while trying to call someone?? |
| 12:51.55 | yang | It won't go dead it will go to SIP/63 |
| 12:51.57 | mort_gib | Mine is quite agreeable actually ;-) |
| 12:52.00 | freckle | hi guys could anyone help me out and let me know what phone this is http://www.flickr.com/photos/viperdude_uk/3112508521/ |
| 12:52.03 | yang | and the third person will pick up |
| 12:52.11 | freckle | a customer has asked |
| 12:52.19 | mort_gib | -And then go dead if 63 is out for lunch |
| 12:52.37 | yang | mort_gib: I tried to apply your rules, but now it doesn't even ring the first extension...however I didn't know what to do with "verbose" extension |
| 12:53.22 | mort_gib | YANG; yOU |
| 12:53.38 | mort_gib | -Sorry, eating my lunch |
| 12:53.46 | yang | bon apetit |
| 12:53.56 | mort_gib | yang: you can take that line out (thanks) |
| 12:54.02 | yang | i took it out |
| 12:54.14 | yang | and it happens what i said - i will pb the cli |
| 12:54.14 | mort_gib | yang:pastebin |
| 12:54.45 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 12:55.30 | yang | http://pastebin.ca/1286793 |
| 12:56.43 | mort_gib | yang: pb sip.conf of that user please |
| 12:57.08 | mort_gib | yang: type=peer << important |
| 12:57.26 | yang | ah you said, its gotta be type=peer , I have friend |
| 12:57.36 | yang | ok I will try again |
| 12:59.47 | yang | still won't work - http://pastebin.ca/1286801 |
| 13:01.22 | Karlitoo | hummm |
| 13:01.27 | Karlitoo | Proxy-Authorization: Digest username="6001",realm="asterisk",nonce="2b25a257",uri="sip:215@10.0.0.223:5060;transport=udp",response="17d28b625b88cc11e8bec53f6a82cc9d",algorithm=MD5 |
| 13:01.54 | Karlitoo | I have this lil problem that the uri that is beeing called has the wrong ip address |
| 13:02.26 | Karlitoo | it's beeing pointed to the asterisk server instead of the the avaya server |
| 13:02.32 | Karlitoo | that is h323 channel on asterisk |
| 13:02.36 | Karlitoo | then to avaya |
| 13:02.56 | mort_gib | yang: cli ?? |
| 13:03.30 | *** join/#asterisk sasargen (n=chatzill@174-152-32-136.pools.spcsdns.net) |
| 13:04.01 | yang | http://pastebin.ca/1286806 |
| 13:08.19 | mort_gib | yang: http://pastebin.ca/1286810 |
| 13:09.03 | mort_gib | yang: your problem is that CallCount is not set, that has to do with the settings in sip.conf |
| 13:09.42 | mort_gib | yang: Try adding qualify=2000 |
| 13:09.55 | yang | ok |
| 13:10.51 | *** join/#asterisk ggiusti (n=giovanni@lpierucci.micc.unifi.it) |
| 13:11.16 | yang | mort_gib: are you guessing ? its not working |
| 13:12.17 | mort_gib | No,not guessing, what I pased in is from a working config, when you call SIPPEER you get nothing, try changing your user 70 into the section I added |
| 13:12.18 | yang | but you have incominglimit=2 |
| 13:12.27 | mort_gib | That's not important here... |
| 13:13.14 | yang | well i changed the qualify=yes to qualify=2000 anything else? |
| 13:13.19 | *** part/#asterisk henk (n=hank@netwichtig.de) |
| 13:13.38 | mort_gib | limitonpeers=yes |
| 13:13.55 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 13:13.55 | *** mode/#asterisk [+o russellb] by ChanServ |
| 13:14.49 | yang | I get the same errors all the time |
| 13:16.19 | yang | i will paste you my extensions once again - http://pastebin.ca/1286813 |
| 13:20.09 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.130.105) |
| 13:20.11 | mort_gib | yang: SIPPEER is not picking up if that device is in a call |
| 13:20.34 | yang | its not being in a call |
| 13:21.13 | mort_gib | Well you should get a number, not CurCalls= (nothing) |
| 13:21.16 | yang | but it acts like that the channel has allready been taken |
| 13:22.07 | mort_gib | yang: That's because the condition (CallCount!=0) |
| 13:22.11 | mort_gib | Y |
| 13:22.16 | yang | (SIP/70|curcalls)} to (SIP/70|1)} for 1 busy channel ? |
| 13:22.17 | mort_gib | yang: It's nothing |
| 13:22.47 | yang | CallCount = 0 and it should be = 1 right? |
| 13:24.19 | *** join/#asterisk freckle (n=chatzill@sokar.enta.net) |
| 13:25.05 | yang | Can you please update that paste, It doesn't seem that we are getting anywhere for the last 20 minutes. |
| 13:26.15 | mort_gib | yang: you get CallCount=null nothing, SIPPEER is not returning any number |
| 13:26.51 | yang | exten => _5863170,n,GotoIf($["${CallCount}" = "0"]?DialExten:VmBusy) turning into exten => _5863170,n,GotoIf($["${CallCount}" = "1"]?DialExten:VmBusy) doesn't work either |
| 13:27.13 | unixdawg | i forget but is there a way to set a inbound vs outbound limit on a exten |
| 13:27.31 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:27.38 | yang | And don't ask me why its not returning,couse I don't know |
| 13:27.49 | unixdawg | pelts [TK]D-Fender with snowballs |
| 13:27.51 | mort_gib | yang: Well I do |
| 13:28.02 | yang | maybe [TK]D-Fender will have a good day today. |
| 13:28.10 | unixdawg | yeah right |
| 13:28.26 | unixdawg | the only good day for [TK]D-Fender is when he is asleep |
| 13:28.29 | unixdawg | ducking |
| 13:28.31 | mort_gib | yang: exten => _5863170,1,Set(CallCount=${SIPPEER(70|curcalls)}) |
| 13:28.35 | [TK]D-Fender | unixdawg: A snowball? Thats it? |
| 13:28.48 | [TK]D-Fender | dumps an entire Montreal winter on unixdawg |
| 13:28.49 | yang | mort_gib: lets see...heh |
| 13:28.55 | unixdawg | the snowball machine gun is out for repair |
| 13:29.53 | mort_gib | yang: Did you get the difference?? |
| 13:30.05 | yang | yes |
| 13:30.19 | mort_gib | yang: :-) |
| 13:31.41 | yang | hands mort_gib a freezing beer |
| 13:32.16 | mort_gib | Thanks :-0 -How did you know I'm Danish?? |
| 13:32.38 | yang | i didn't :) |
| 13:32.56 | mort_gib | :-) |
| 13:33.30 | *** join/#asterisk l8router (n=l8router@122.111.125.175) |
| 13:34.45 | unixdawg | [TK]D-Fender, Happy Hoidays man |
| 13:35.07 | unixdawg | wishes all in the channel Happy Holidays |
| 13:35.31 | unixdawg | now lets gang up on [TK]D-Fender and tickle him till he passes out |
| 13:38.54 | *** join/#asterisk tokozedg (n=root@77.92.234.236) |
| 13:39.32 | anonymouz666 | I'll send a ticket to [TK]D-Fender to enjoy the summer elsewhere |
| 13:39.39 | tokozedg | i have created all tables and columnes but i get this this erro |
| 13:39.40 | tokozedg | Failed to insert into database: |
| 13:39.51 | tokozedg | [Dec 16 17:39:35] ERROR[4996]: cdr_addon_mysql.c:320 mysql_log: Failed to insert into database: (1136) Column count doesn't match value count at row 1 |
| 13:40.11 | yang | mort_gib: what about this - a bit of forwarding http://pastebin.ca/1286830 |
| 13:41.14 | mort_gib | yang: eh a label line n(fuckoffanddie) can/should only be used once |
| 13:41.45 | mort_gib | so having multiple labels line n(vmexten) might help readability but will not work |
| 13:42.01 | yang | *sigh* |
| 13:42.06 | mort_gib | as intended that is.... |
| 13:42.18 | mort_gib | I'm all for random :-) |
| 13:43.12 | yang | so simply removing (DialExten) from the 4. and 5. line? |
| 13:44.12 | mort_gib | yang: http://pastebin.ca/1286835 |
| 13:45.05 | mort_gib | yang: do this kind of things in a simple flowchart first, especially when you are not 100% sure what you are doing |
| 13:46.55 | yang | Well, Its easy - First I want 60 to ring for 7 seconds, then if nobody picks up 64 and trhirdly 63 , but If 60 is busy on the start it continues to 64 without ringing 60 at all |
| 13:47.18 | mort_gib | yang: how do you turn on/off voicemail for that ext |
| 13:47.20 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 13:47.36 | yang | there is no voicemaikl |
| 13:48.21 | mort_gib | so 1. test to see if 60 is in use if inuse goto64 if 64 rings out goto 63 |
| 13:49.21 | yang | yes, previously I had a setup like this http://pastebin.ca/1286839 , which now needs to be applied to the busy extensions setup which you made |
| 13:49.46 | yang | but as you suggested n(DialExten) twice will fail |
| 13:50.59 | yang | What if 60 is not in use, and nobody picks in 7 second, it should go to 64 then |
| 13:52.14 | mort_gib | DIALSTATUS |
| 13:52.43 | mort_gib | Every time (in your case) you dial an extension (SIP device) you check the dial status |
| 13:53.54 | *** join/#asterisk LiNeTuX (n=LiNeTuX@64.132.248.206) |
| 13:54.22 | yang | ah it goes to the section unavail (if nobody picks up the 60) |
| 13:55.19 | yang | puts a lemon into mort_gib's beer |
| 13:56.16 | mort_gib | yang: Yes, and I'm NOT Mexicano :-) Hombre, Lemon y Cerveza no me gusto! |
| 13:56.42 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 13:57.26 | yang | lol |
| 13:57.42 | mort_gib | :-) |
| 13:57.52 | yang | No Tequila para ti |
| 13:58.49 | yang | I still don't get the Dialplan right, becouse I can only have one exten => _5863170,n(unavail),Dial(SIP/64,30,rtk) |
| 13:59.05 | mort_gib | O Cerveza, pero Coronito con limones :-P |
| 13:59.19 | yang | After passing this it should ring SIP/63 for the rest of the seconds |
| 13:59.37 | yang | But with only one n(unavail) possible I cannot make it work |
| 13:59.47 | mort_gib | yang: You go Dial(SIP/60) dialstatus goto dial70 dialstatus goto63 |
| 13:59.52 | yang | only ring SIP/64&SIP/63 at the same time |
| 14:00.17 | mort_gib | unavail is just a label, you can call it, say n(fuckoffanddie) if you want :-) |
| 14:00.29 | yang | i know |
| 14:01.26 | mort_gib | So every time you try dialing a new extension/collection of extensions you just move off to the next dial statement |
| 14:01.45 | mort_gib | Mind you, in real world examples you are wasting your time |
| 14:01.59 | yang | has registered pbx.si |
| 14:02.27 | mort_gib | dial one sip/device and fall over to the rest of the group DIAL(SIP/ALL&SIP/DEVICES) |
| 14:02.44 | mort_gib | You can get really lost with stuff like this |
| 14:03.30 | yang | I think that I would just need an additional line and it would work |
| 14:04.50 | yang | Its simply silly to forward it to the third extension |
| 14:04.57 | mort_gib | YES!!! |
| 14:05.01 | yang | I should just drop it afterwards |
| 14:05.20 | mort_gib | But as an exercise to learn extensions.conf good! |
| 14:05.27 | mort_gib | Try to get it to work |
| 14:05.34 | yang | ok |
| 14:07.15 | yang | its got too complicated for me *sigh* |
| 14:08.56 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:09.40 | eppigy | hello |
| 14:09.42 | eppigy | i am dave |
| 14:10.39 | yang | Hi Dave ! "Don't do this to me Dave" |
| 14:10.47 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 14:15.38 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 14:16.30 | yang | mort_gib: http://paste.debian.net/23804/ |
| 14:16.34 | Subdolus | Completely useless talk, but what's everyones favourite SIP or even IAX client for Linux? |
| 14:17.12 | eppigy | XLITE |
| 14:17.40 | yang | Subdolus: ekiga, has a new IAX support |
| 14:21.00 | Subdolus | yang: I don't know how to use ekiga with asterisk. Is it possible? |
| 14:21.23 | Subdolus | It looks like it only uses SIP addresses |
| 14:21.54 | yang | Subdolus: its possible |
| 14:22.13 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 14:22.27 | yang | Subdolus: I think they support IAX too since version 3 |
| 14:22.51 | *** join/#asterisk psy0nid3 (n=b0red@bookit-dev.com) |
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| 14:24.20 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:27.12 | dominic1 | anybody here with knowledge about asterisk and E1/S2M? |
| 14:27.39 | dominic1 | I want to use 2 two pri-cards on one pri |
| 14:27.57 | dominic1 | Can anybody tell me, if the call will be signaled on both cards |
| 14:27.58 | dominic1 | ? |
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| 14:33.17 | *** join/#asterisk naitram (n=naitram@12.105.199.38) |
| 14:34.58 | naitram | anyone know if asterisk now allows access through the AMI. How about switchvox? |
| 14:35.31 | russellb | AsteriskNOW: yes. Switchvox: No. |
| 14:35.39 | *** join/#asterisk tokozedg (n=root@77.92.234.236) |
| 14:35.51 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 14:36.22 | tokozedg | i want to match all local numbers |
| 14:36.26 | tokozedg | with go to if |
| 14:36.43 | tokozedg | and can i match a range of numbers? |
| 14:36.57 | dominic1 | <PROTECTED> |
| 14:37.01 | naitram | russellb: thanks. anyone know of an asterisk hardware appliance that supports access via AMI |
| 14:37.26 | tokozedg | GotoIf($[${CALLERID(num)} = 20-30]?yes,no) |
| 14:37.29 | tokozedg | is it correct |
| 14:38.01 | *** join/#asterisk codaine (n=Onur@198.64.168.130) |
| 14:38.06 | russellb | naitram: the digium asterisk appliance does |
| 14:38.50 | tokozedg | ? |
| 14:39.36 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:40.30 | naitram | russellb: I called digium yesterday and the only appliance they have with AMI supports only up to 20 calls. The AA50. Any other options for something with say 50 to 100 calls? |
| 14:41.24 | mort_gib | yang: I just had a look at your pb |
| 14:41.29 | mort_gib | not valif |
| 14:41.49 | yang | mort_gib: oh well... |
| 14:41.56 | yang | mort_gib: I am out of ideas |
| 14:42.05 | mort_gib | yang: Hang on in here.... |
| 14:43.26 | yang | I am here |
| 14:44.23 | dominic1 | I want to use 2 two pri-cards on one pri; Can anybody tell me, if the call will be signaled on both cards? |
| 14:44.47 | yang | russellb: The G729 codec its binded to the MAC address, is it possible to make it work on a Virtual server (no MAC) ? |
| 14:44.49 | *** join/#asterisk BlargMaN00 (n=blarg@12.234.16.130) |
| 14:45.13 | mort_gib | yang: how will IPv4 work without a MAC |
| 14:45.38 | florz | dominic1: of course, one will be sending the signalling, the other one will receive it. |
| 14:46.04 | rue_mohr | dominic1, you cant do that in parallel for the same reason you cant do that for ethernet |
| 14:46.06 | yang | I only see HWaddr 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00 |
| 14:46.50 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 14:46.52 | mort_gib | yang: not so, if the IF works, at least on VM then you also have a MAC |
| 14:46.58 | russellb | yang: I think so ... talk to support@digium.com |
| 14:47.26 | yang | russellb: Well I allready have the official answer that it isn't possible, so I thought if you knew a hack around that :) |
| 14:47.42 | mort_gib | yang: what does ifconfig give you |
| 14:51.53 | yang | http://pastebin.ca/1286855 |
| 14:59.47 | *** join/#asterisk invalidrecord (n=fares@87.113.88.31.plusnet.pte-ag2.dyn.plus.net) |
| 15:00.06 | invalidrecord | where can i fine the query realtime runs to look up extensions |
| 15:00.59 | unixdawg | anyone here know where the src for app_conf is |
| 15:01.24 | dominic1 | florz: thank you! |
| 15:01.27 | unixdawg | looking to use it till dahdi is ported to bsd |
| 15:01.36 | dominic1 | rue_mohr: Thank you! |
| 15:03.36 | anonymouz666 | Corydon76-dig: I setup the cdr odbc adaptive in Asterisk 1.4. Very nice, but there is ANY documentation |
| 15:04.07 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 15:06.39 | *** join/#asterisk etfonhomey (n=chatzill@74-143-192-77.static.insightbb.com) |
| 15:08.26 | *** join/#asterisk vi390 (n=fc@unaffiliated/vi390) |
| 15:09.40 | vi390 | hi, is there a way to test agi scripts in the cmdline ? (iam using python and pyst) maybe there is a wrapper to test scripts as the behave. I cant find anything on the net about this stuff |
| 15:11.33 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:14.17 | yang | mort_gib: there is no MAC address http://pastebin.ca/1286855 |
| 15:14.17 | Linuturk | I've got a strange problem. It seems my zhone channel bank is acting up. some devices can fax and such, while others cannot |
| 15:14.41 | mort_gib | yamg; what VM do you use?? |
| 15:15.12 | yang | VM ? |
| 15:15.27 | yang | I don't know what the use, some sort of Vserver software |
| 15:15.42 | mort_gib | ok |
| 15:18.13 | Linuturk | anyone have any ideas on a channel bank allowing some devices to fax, while not allowing others to work correclty? |
| 15:19.14 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 15:20.08 | naitram | how does Specification of max concurrent calls on asterisk appliance relate to sip calls? |
| 15:20.53 | *** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) |
| 15:21.01 | Karlitoo | what does the 't' at the end and the 45 by the end represent in Dial(H323/12345,45,t)?? |
| 15:21.07 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:22.13 | jaytee | Karlitoo, the 45 is ringing timeout and the t is to enable transfer by the called party |
| 15:22.48 | jaytee | Karlitoo, something you could have easily answered yourself if you'd bothered to look in Appendix B of the book. |
| 15:25.05 | codaine | is there an ubuntu repository for asterisk 1.6? |
| 15:26.02 | eppigy | hello jaytee |
| 15:26.03 | *** part/#asterisk psy0nid3 (n=b0red@bookit-dev.com) |
| 15:26.06 | eppigy | my internet pal |
| 15:27.42 | jaytee | hello eppigy a.k.a. I am Dave |
| 15:30.42 | *** join/#asterisk mcab (n=mb@mostly-harmless.ca) |
| 15:31.34 | *** join/#asterisk bbryant (n=Brett_Br@adsl-068-016-200-248.sip.chs.bellsouth.net) |
| 15:32.55 | ride330 | hey does anyone have a good sip softphone for linux that has 729 ablity |
| 15:34.55 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-ef3c751d3b1dadb0) |
| 15:34.55 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:35.29 | mort_gib | yang: ... |
| 15:35.44 | *** join/#asterisk Kobaz (n=kobaz@its.kobaz.net) |
| 15:36.04 | Kobaz | what's the max length of callerid.. it's like 20 or something if i remember |
| 15:36.12 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 15:36.24 | dominic1 | florz: If I have to pri connections, both signaling the same number, one asterisk on every pri. Some number I want to answer on the second pri and some on the first. 100-200 on the first pri 300-400 on the second. Will that be possible? Or will the caller get a busy if he is signaled on the second priv when calling the number 110? |
| 15:36.34 | Karlitoo | thanx jaytee, sry I didn't even open the book, I do everything trough google |
| 15:37.22 | Karlitoo | cause in my case I'm not doing a typical central office -- asterisk --office.... |
| 15:38.02 | Karlitoo | I'm doing hardphone - avaya -h323 - asterisk - sip - softphone |
| 15:38.13 | Karlitoo | and it's a pain |
| 15:38.21 | yang | ride330: ekiga.org |
| 15:38.30 | Kobaz | h323? |
| 15:38.39 | Kobaz | Karlitoo: you got h323 going in asterisk? |
| 15:38.40 | dominic1 | omg avaya on asterisk, that's great :-) |
| 15:38.45 | florz | dominic1: I'm not aware of any way to do that - but it doesn't seem to make much sense, anyhow |
| 15:39.28 | Karlitoo | yeah I got it compiled and working one way thords asterisk but outbound thords avaya I get a hangup before it even goes to avaya |
| 15:39.54 | Kobaz | Karlitoo: mind sharing the details? i've bene fighting h323 forever, cant even get netmeeting to connect |
| 15:40.06 | *** join/#asterisk etfonhomey (n=chatzill@74-143-192-77.static.insightbb.com) |
| 15:40.11 | dominic1 | my problem is, I get two pri's from my carrier with the same numbers signaled. I want the most numbers for my old asterisk and some numbers for my new asterisk MeetMe system |
| 15:40.12 | Kobaz | what's a thords |
| 15:40.49 | Kobaz | dominic1: it's up to your provider as to where the calls go |
| 15:41.17 | Kobaz | dominic1: what your provider is doing is probably round-robin, or something to that effect |
| 15:41.28 | Karlitoo | I have another 20 min here then I'm here from tommor at 9am 1+ GST |
| 15:41.36 | Kobaz | Karlitoo: k |
| 15:41.39 | Karlitoo | grrr damn keyboard |
| 15:41.53 | Karlitoo | did u get h323 to compile |
| 15:41.55 | florz | dominic1: well, either tell your provider to distribute calls differently, or just forward them from one asterisk to the other? |
| 15:41.59 | Kobaz | yeah it compiles |
| 15:42.02 | Kobaz | i cant make any calls |
| 15:42.15 | Kobaz | i found all kinds of sample configs, but no go |
| 15:42.23 | Karlitoo | so how can you now connect netmeeting to it |
| 15:42.26 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:42.30 | Kobaz | i can't |
| 15:42.53 | Karlitoo | I hope it's not the new netmeeting cause I heard that the new 1 does not use h323 (not sure) |
| 15:42.56 | dominic1 | okay, I will ask my carrier to distribute the calls differently |
| 15:43.00 | Karlitoo | did you set up a user for it |
| 15:43.00 | dominic1 | thank you guys |
| 15:43.02 | Kobaz | i was using an old one |
| 15:43.02 | dominic1 | !!! |
| 15:43.09 | Kobaz | yeah |
| 15:43.29 | Kobaz | i dont think i have the configs anymore, lemme see |
| 15:43.49 | Karlitoo | which channal driver did u use |
| 15:43.54 | Kobaz | ooh323 |
| 15:43.56 | Karlitoo | h323 oh323 or ooh323 |
| 15:44.04 | Karlitoo | ah ok I got that 1 as well |
| 15:44.14 | Karlitoo | let me try connecting net meeting to it |
| 15:44.18 | *** join/#asterisk kerframil (n=kerframi@gentoo/user/kerframil) |
| 15:44.22 | Karlitoo | give me a sec |
| 15:44.27 | Kobaz | none of the other ones semed to come even close to working |
| 15:44.36 | Karlitoo | true |
| 15:45.10 | Kobaz | are you using a gatekeeper |
| 15:45.39 | Karlitoo | nope |
| 15:45.44 | Kobaz | i gave up on the base h323 channel driver since it seemed like it required a gatekeeper, and i couldn't get the gnu gatekeeper going |
| 15:45.45 | Karlitoo | should I be?? |
| 15:45.49 | Kobaz | dunno |
| 15:46.02 | Kobaz | does it work? |
| 15:46.20 | Kobaz | if you can place calls and recieve calls and all that, then i wouldnt think you would need one |
| 15:46.20 | Karlitoo | ok well I'll try with the gatekeeper tomorrow right now I need to download netmeeting 1 sec |
| 15:46.33 | Kobaz | what have you gotten to work? |
| 15:46.48 | *** join/#asterisk bijit (n=benji@201.198.72.142) |
| 15:47.03 | Kobaz | i can actually place a call with netmeeting, but it doesn't complete the call.... like, the other phone will ring, i pick it up, and then the call drops dead |
| 15:47.14 | Kobaz | that's as far as i get |
| 15:47.57 | Karlitoo | the call from hardphone connected to avaya trough a h323 trunk thords asterisk and then to a softphone connected via sip |
| 15:47.57 | Kobaz | and that works? |
| 15:47.57 | Kobaz | what about the other way? |
| 15:47.57 | Karlitoo | yeah it works just fine |
| 15:47.57 | Karlitoo | but I cant do reverse |
| 15:48.00 | Kobaz | reverse of what? |
| 15:48.04 | Kobaz | and what's a thords? |
| 15:48.07 | Kobaz | do you mean towards? |
| 15:48.11 | Kobaz | or through? |
| 15:48.13 | Kobaz | heh |
| 15:48.33 | Karlitoo | towards |
| 15:48.49 | Karlitoo | reverse same call just in the other direction |
| 15:49.05 | Kobaz | okay, so from soft phone to avaya is not working |
| 15:49.13 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:49.14 | ride330 | yang: does that windows version support 729? |
| 15:49.16 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 15:49.42 | Kobaz | you're using an avaya pbx right...? |
| 15:49.55 | Kobaz | or connecting an avaya h323 ip phone to asterisk? |
| 15:50.10 | Karlitoo | asterisk 1.4 and awaya media gateway g350 |
| 15:50.15 | Kobaz | k |
| 15:51.33 | yang | ride330: I don't know about windows version |
| 15:51.36 | yang | ~ekiga |
| 15:51.37 | jbot | [~ekiga] Ekiga is an OSS SIP soft-phone for Windows, MacOSX, and Linux (Requires GTK+ which is included with the binary releases) that can be found at http://www.ekiga.org |
| 15:51.51 | BlargMaN00 | has anyone had any experience connecting a CCM with * via h323?? |
| 15:52.12 | yang | ride330: irc.gimp.org #ekiga |
| 15:52.18 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 15:52.57 | ride330 | okay cool thanks |
| 15:53.36 | Karlitoo | BlargMaN00, http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integration |
| 15:54.32 | Karlitoo | ooops that's for sip sry |
| 15:54.36 | *** join/#asterisk bijit (n=benji@201.198.72.142) |
| 15:55.07 | Karlitoo | well it has info for h323 as well |
| 15:55.43 | Karlitoo | [TK]D-Fender, when I did the tcpdump I got only sip output no h323 at all |
| 15:55.57 | BlargMaN00 | Karlitoo: thanks, I think that is the same site I used to do the original SIP integration... |
| 15:56.09 | Karlitoo | lol |
| 15:57.46 | Linuturk | hello, I'm new to asterisk in all it's glory, and I'm trying to figure out a tricky problem. some of our fax and voice devices are not working correctly. For example, our postage meter is unable to dial out for postage, and a fax machine is acting strange. We have a T1 from our ISP, that connects to a Ditech echo canceller. from there, one line goes to our zhone voice channel bank, and the other line goes to our asterisk server. off the |
| 15:58.51 | Karlitoo | hey guys, and girls if any c ya all tomorrow |
| 15:58.53 | Karlitoo | beyz |
| 16:00.50 | Linuturk | also, an analog phone in the main waiting area is getting a dial tone, but the line is not recognizing when a call is being made. the tones are not recognized |
| 16:01.02 | Linuturk | so, the dial tone just stays up |
| 16:01.34 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-3524389fe33865b7) |
| 16:01.34 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 16:01.37 | Linuturk | this all seems to have started after an outage with our ISP prompted me to power cycle the channel banks, along with our asterisk server and echo canceller |
| 16:02.12 | Linuturk | I've also noticed a lot of d-channel errors in the message log of asterisk, but I've been working with our ISP, and I believe that might have been sorted out. |
| 16:02.23 | invalidrecord | i am having difficulty with extensions in realtime postgres, is it possible that the query is wrong as it is returning no rows but when i put same data in normal extension (not realtime) it works, sip peers in realtime work fine |
| 16:03.11 | invalidrecord | wonder if i should use odbc instead |
| 16:05.00 | *** join/#asterisk CunningPike (n=arodgers@204.239.10.119) |
| 16:05.36 | Linuturk | is there anything I can do to try to debug this issue? |
| 16:05.54 | Linuturk | anywhere I should be looking for error messages? |
| 16:08.49 | adr3nalin3 | are these messages anything I need to worry about? chan_dahdi.c:1481 dahdi_enable_ec: Unable to enable echo cancellation on channel 1 (No such device) |
| 16:10.59 | *** join/#asterisk linuxer_igor (n=linuxer_@mlsrj200152096p202.mls.com.br) |
| 16:11.21 | *** join/#asterisk iratik (n=itariki@209.248.216.146.nw.nuvox.net) |
| 16:11.34 | iratik | Why would you recomend a sangoma over a digium |
| 16:11.36 | iratik | ? |
| 16:12.33 | Linuturk | do you think I should try powercycling everything again now that the issue with the ISP seems to be resolved? |
| 16:12.56 | *** join/#asterisk shido6 (n=shido6@209.114.208.111) |
| 16:12.56 | adr3nalin3 | Digium's customer service and tech support is second to none. |
| 16:12.59 | *** join/#asterisk qdk (n=qdk@94.191.241.11.bredband.3.dk) |
| 16:14.08 | ride330 | hey when you put your asterisk server on a public network what do you use to setup a firewall? |
| 16:15.06 | ride330 | i usually put my * server behind a firewall but i have a couple of setups where i need to put it right on the public net what should i use to firewall it |
| 16:15.49 | mog | iptables |
| 16:16.30 | ride330 | i dont have any experaince with that is there a web based gui for it? |
| 16:16.51 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 16:16.51 | mog | several i think |
| 16:16.56 | mog | i dont know any off hand |
| 16:17.07 | ride330 | are they as secure as doing the commands? |
| 16:17.14 | mog | yes |
| 16:17.23 | mog | as they are just translators |
| 16:17.45 | ride330 | when you use that is there a need to open 5060 and rtp ports? |
| 16:17.46 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 16:18.24 | *** join/#asterisk tobias (n=tobias@cpe-076-182-095-181.nc.res.rr.com) |
| 16:19.23 | kerframil | ride330: you may not even need iptables. another approach is to minimise the number of network services running, and configure those that should not be exposed to the net not to bind to the net-connected interface (a lot of services bind to 0.0.0.0 by default, i.e. any available IP interface). |
| 16:20.05 | kerframil | ride330: for example, let's say you had a database server running, if it only needed to be used by the local host, you could bind it to 127.0.0.1 |
| 16:21.14 | ride330 | okay so anyone from outside trying to use it would say it is not running |
| 16:21.34 | kerframil | ride330: right |
| 16:21.47 | ride330 | where do you set those binds, in the config files for each service |
| 16:22.01 | kerframil | ride330: yes |
| 16:22.19 | kerframil | ride330: if you check out asterisk itself, you'll see that many of the chan_* config files facilitate that ;) |
| 16:22.19 | Linuturk | ride330: vuurmuur |
| 16:22.21 | ride330 | thats a pretty good idea but i like the security of blocking ports |
| 16:22.50 | kerframil | ride330: yes, but what are you blocking exactly? if traffic hits a port to which a service isn't bound, that traffic isn't serviced. you DROP with iptables, that doesn't stop the traffic arriving anyway. |
| 16:23.03 | kerframil | ride330: I'm not saying iptables doesn't have other uses but think it through |
| 16:23.52 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 16:23.52 | kerframil | ride330: doing connection tracking on SIP can be tough too, but I'll give you a link for it in the event that you are going to using iptables: http://www.iptel.org/sipalg/ |
| 16:23.56 | ride330 | okay well that is true but what about ports that don't have a service running is there a way for hackers to get into the system and do what they want if you don't block on certian ports? |
| 16:24.30 | kerframil | ride330: no, unless there is a vulnerability in the IP stack itself (in which case, you're potentially screwed anyway). |
| 16:25.02 | kerframil | ride330: about that link, the conntrack_sip module is mainline in recent kernels (the page suggests that it has to come from patch-o-matic but that's not the case now) |
| 16:25.38 | ride330 | another thing i do with my firewalls is a vpn, what can i use to setup a vpn |
| 16:26.01 | kerframil | ride330: there are quite a few options, but my favourite is openvpn - for what it's worth. it's a great product and a piece of cake to set up. |
| 16:26.10 | *** join/#asterisk seanmh (n=seanmh@216.31.101.11) |
| 16:27.29 | Linuturk | ride330: vuurmuur is a nice front end to iptables |
| 16:27.30 | ride330 | yeah its easy, thats what i like to hear |
| 16:28.07 | kerframil | ride330: it's about as easy as it gets. in particular, setting up a point-to-point tunnel is ridiculously easy. but setting up a CA for multiple clients isn't so hard either. it bundles a few scripts that take the hard work out of managing the certificates. |
| 16:28.08 | ride330 | Linuturk: i am looking at the webpage now seems to be really cool have you used that with openvpn? |
| 16:29.55 | ride330 | kerframil: does it allow for road warriors? |
| 16:30.03 | *** join/#asterisk t0rrieri (n=torrieri@nelug/crew/torrieri) |
| 16:30.17 | *** join/#asterisk davidc (n=david@netman1.us.sargasso.net) |
| 16:30.23 | kerframil | ride330: it certainly does (the windows port works great if you need one, and there's a gui that does the job for setting up and tearing down the link) |
| 16:31.58 | ride330 | kerframil: where is the openvpn server download is it 2.0.9 on their website? |
| 16:34.02 | ride330 | kerframil: okay looks like that is the latest version, so it seems as easy as a make make install, are there scripts to configure clients to access it? |
| 16:40.03 | *** join/#asterisk sdaniels (n=chatzill@216.65.195.52) |
| 16:40.08 | sdaniels | exten => _1NXXNXXXXXX,n,Dial(SIP/+${EXTEN}@WHAT GOES HERE) <-- ip address or the name in sip.conf? |
| 16:41.01 | ride330 | the context |
| 16:41.29 | ride330 | so whatever you have in sip.conf [provider] |
| 16:41.37 | *** join/#asterisk reneger (n=reneger@p3EE2EA0A.dip.t-dialin.net) |
| 16:41.46 | sdaniels | cool, thanks |
| 16:43.06 | ride330 | yup |
| 16:44.09 | Corydon76-dig | anonymouz666: There's just the documentation in the config file |
| 16:44.53 | Corydon76-dig | anonymouz666: Trunk might have more explicit documentation |
| 16:45.25 | iratik | Is there such a thing as an analog T1? |
| 16:45.29 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:45.42 | Corydon76-dig | iratik: Yes, it's called E&M |
| 16:46.13 | iratik | Will it work with a Digium T1 card? |
| 16:46.17 | Corydon76-dig | Or, in the modern world, CAS. It's what channel banks use. |
| 16:46.27 | Corydon76-dig | Yes |
| 16:46.46 | iratik | T1s 24 channels... They can be voice or data.... What makes the difference there? |
| 16:46.56 | iratik | Are they the same... just different hardware at the endpoint? |
| 16:47.04 | Corydon76-dig | E&M lines generally use d4/ami framing/coding, while CAS generally uses ESF/B8ZS |
| 16:47.41 | Corydon76-dig | They are the same, signal wise |
| 16:47.46 | iratik | I arrive on site... they point me to the T1. they tell me its a voice T1... and I see a cat5 cable coming from a demarcation point |
| 16:47.48 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 16:48.14 | iratik | That would be E&M? |
| 16:48.29 | Corydon76-dig | That may be E&M |
| 16:48.34 | iratik | or CAS? |
| 16:49.35 | Corydon76-dig | There's at least 3 different variants of what is commonly called "voice T1". You need to figure out which it is |
| 16:49.38 | coppice | CAS just means channel associated signaling. E&M is *a* channel associated signaling scheme |
| 16:49.49 | iratik | is there a wikipedia page on different types of provided T1s? |
| 16:49.59 | iratik | Is there any kind of provided T1 that won't work with the digium T1 card? |
| 16:50.24 | Corydon76-dig | iratik: I'm sure there are, but they aren't common |
| 16:51.02 | Corydon76-dig | Standards are great... there's so many to choose from. |
| 16:51.40 | *** join/#asterisk wino (n=Justin@unaffiliated/wino) |
| 16:52.14 | kerframil | ride330: I use the _rc's myself but yeah ... the HOWTOs in the Documentation section should tell you what you need to know |
| 16:52.19 | Corydon76-dig | coppice: fair enough. I was thinking of FXO/FXS signalling |
| 16:52.28 | eppigy | d4 ami |
| 16:52.29 | iratik | Is there a wikipedia page on integrating a T1 with asterisk? |
| 16:52.57 | eppigy | iratik: do they have a circuit inventory? |
| 16:53.10 | eppigy | call and verify framing and protocol |
| 16:53.47 | Corydon76-dig | iratik: You might as well be asking about whether you need to put fuel in your car. Diesel? Gasoline? Biofuel? Ethanol? |
| 16:54.24 | iratik | Call the T1 provider and ask them about framing and protocol? |
| 16:54.33 | eppigy | yes |
| 16:54.53 | eppigy | that way you are not just playing musical configs |
| 16:55.38 | eppigy | if they say esf b8zs ask what switch type |
| 16:55.51 | eppigy | as well |
| 16:56.41 | iratik | I won't know what they are talking about... is there some document out there i can read so I will be more knowledgable when speaking with them? |
| 16:56.57 | eppigy | dude why are you doing this job then? |
| 16:57.35 | carrar | hahah |
| 16:59.05 | carrar | iratik, start out with this book |
| 16:59.05 | carrar | http://www.amazon.ca/Sonet-T1-Architecture-Transport-Networks/dp/0134475909 |
| 16:59.10 | eppigy | yeah |
| 16:59.14 | eppigy | and come back in a week |
| 16:59.34 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 17:00.36 | sdaniels | how can i show regisered peers in the console? |
| 17:01.04 | eppigy | sip show peers |
| 17:01.11 | sdaniels | thx |
| 17:02.02 | iratik | lol |
| 17:02.03 | iratik | thanks |
| 17:02.37 | Linuturk | how can I isolate the problems I'm having? |
| 17:02.38 | iratik | Can I test the T1 card using the data T1 drop in our office? |
| 17:02.44 | iratik | probably not lol |
| 17:02.46 | Linuturk | between the channel banks and the asterisk server? |
| 17:03.09 | eppigy | iratik: do they have a circuit inventory |
| 17:03.12 | eppigy | they should have one |
| 17:03.30 | eppigy | if you just call the provider and ask those simple questions |
| 17:03.35 | eppigy | you dont have to understand them |
| 17:03.41 | eppigy | you just have to know the answer |
| 17:04.00 | eppigy | then you can set it up |
| 17:04.06 | eppigy | using google |
| 17:04.17 | eppigy | and .conf examples |
| 17:04.52 | iratik | Thanks for your help |
| 17:05.06 | Linuturk | this is incredibly frusterating |
| 17:11.33 | *** join/#asterisk tokozedg (n=toko@62.212.33.96) |
| 17:11.33 | ride330 | wow that Vuurmuur app is pretty easy to use |
| 17:11.57 | ride330 | what are the basic rules i need to drop hackers? |
| 17:12.47 | tokozedg | qartveli aravina??? |
| 17:12.49 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 17:17.51 | ride330 | can someone help me with a little dialplan stuff |
| 17:17.55 | tokozedg | exten => 22,2.GotoIf($[${CALLERID(num) = 20) |
| 17:18.25 | ride330 | I need to set the outbound caller id based on the ext # that is dialing out |
| 17:18.52 | ride330 | so for example 2304 has the outbound ccaler id 212-555-1234 |
| 17:19.20 | ride330 | but ext # 27XX has the outbound caller id 212-321-2234 |
| 17:19.30 | ride330 | how do is set that up? |
| 17:19.42 | tokozedg | and here instead 20 i want to be for range of phone numbers and how cant it be done??? |
| 17:19.47 | ride330 | i have 3 sets of ext number 23XX, 27XX, and 28XX |
| 17:21.58 | codaine | how about GotoIf($["${CALLERID(num):0:2}" = "23"]?........ and same thing for others? |
| 17:22.16 | *** join/#asterisk Bad_Robot- (n=Bad_Robo@cpe-76-173-219-25.socal.res.rr.com) |
| 17:22.16 | codaine | i'm not sure if thats the right syntax for substring |
| 17:22.23 | *** join/#asterisk t0rrieri (n=torrieri@nelug/crew/torrieri) |
| 17:22.42 | ride330 | yeah that makes sense |
| 17:22.56 | Bad_Robot- | good morning all |
| 17:23.32 | ride330 | so when they dial NXXNXXXXXX,s,1(goto,macro,1) or something |
| 17:23.36 | *** join/#asterisk Leddy (n=Leddy@72.54.198.194) |
| 17:24.00 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.17) |
| 17:25.15 | tokozedg | can anyone share me asterisk configs? i want to discuss it and learn from it. ?? |
| 17:26.04 | ride330 | so does anyone know if that is the right way to do substrings? |
| 17:26.11 | ride330 | <PROTECTED> |
| 17:28.28 | fogo | ride330: looks right to me - http://www.voip-info.org/wiki/index.php?page=Asterisk+variables |
| 17:30.18 | ride330 | yeah thanks i was just checking that out |
| 17:30.36 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 17:43.39 | *** join/#asterisk Subdolus (n=subby@subby.afraid.org) |
| 17:47.15 | *** join/#asterisk SiberAIR (n=SibRphre@ip67-93-6-162.z6-93-67.customer.algx.net) |
| 17:59.16 | jasonwoot | what's your best source/price for 501's? |
| 18:00.57 | Qwell | jasonwoot: I know a guy who knows a guy, who got a bunch that fell out of the back of a truck |
| 18:03.02 | *** join/#asterisk segun (n=segun@62.173.48.96) |
| 18:03.07 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 18:03.56 | jasonwoot | mintintheboxneverbeenopened? |
| 18:04.34 | segun | please I am having a serious issue with my portech mv 378 and asterisk....only channels 1 and 2 are actually configured, no matter what I tried, the other channels all get a 401 response from asterisk, what do I do please |
| 18:04.58 | *** join/#asterisk jsolis (n=jimmy@200.121.160.48) |
| 18:05.38 | segun | looking at the asterisk console, I found out that the register message were from a different IP from the one I set on the portech |
| 18:11.27 | codaine | did parameter seperator for realtime applications changed from "|" to ","? |
| 18:13.53 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 18:17.47 | [TK]D-Fender | codaine: Changed for all in 1.6 |
| 18:17.59 | [TK]D-Fender | jasonwoot: Why are you looking for 501's at all these days? |
| 18:18.11 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-70-57.w86-215.abo.wanadoo.fr) |
| 18:18.50 | *** join/#asterisk WHYS (i=lpfm@137.28.94.209) |
| 18:18.54 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-70-57.w86-215.abo.wanadoo.fr) |
| 18:19.23 | *** join/#asterisk chi6IT41 (n=chigital@tmo-096-232.customers.d1-online.com) |
| 18:19.39 | wino | Anyone care to share their asterisk box system specs? I'm looking to build something that handles 8 FXS ports using SIP in a 1u server and I've read the "Asterisk Book" (well, most of it), and it indicates that separate physical processors over dual core... And I'm wondering how to gauge what to get |
| 18:20.37 | *** join/#asterisk bmoraca (n=bmoraca@209.60.253.58) |
| 18:21.05 | bmoraca | is anyone familiar with Polycom 330 phones? |
| 18:21.12 | [TK]D-Fender | wino: You're requirements are pathetically low. |
| 18:21.27 | [TK]D-Fender | wino: WinDon't even think about it.... |
| 18:21.40 | [TK]D-Fender | wino: a P3 would be more than enough |
| 18:21.56 | [TK]D-Fender | bmoraca: ... |
| 18:21.58 | [TK]D-Fender | ~ask |
| 18:21.59 | jbot | extra, extra, read all about it, ask is 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. |
| 18:22.10 | unpaidbill | you could use an alix board with minipci fxs'! |
| 18:22.37 | bmoraca | yeah, yeah...i realized it's not good etiquette |
| 18:22.46 | [TK]D-Fender | unpaidbill: Yup... not sure how the EC would survive mind you... |
| 18:23.01 | [TK]D-Fender | bmoraca: Just an invitation to ask a specific question. |
| 18:23.37 | wino | [TK]D-Fender: That's what I was thinking. I have some dual p3 1.3's with 2GB pc133 and u160 disks and I was hoping that re-purposing one of them would be sufficient |
| 18:23.45 | wino | [TK]D-Fender: And, thank you |
| 18:23.54 | bmoraca | anyway...i actually think it might be related to Asterisk specifically. it's not recognizing 3-digit dials, such as 411, etc. my dial plan on the phone is correct (it works for a 550), but i've got to be overlooking something simple. 411 exists as an extension, but it just plain won't dial from a Polycom 330. |
| 18:24.18 | *** join/#asterisk freckle_home (n=chatzill@84.45.168.57) |
| 18:24.37 | Bad_Robot- | dont' you need to dial 9 to get an outside line? |
| 18:24.42 | bmoraca | no |
| 18:24.53 | bmoraca | and i'd prefer not to have to |
| 18:25.01 | Nugget | dialing 9 is dumb, it's a vestigial relic of older, deprecated technology. |
| 18:25.15 | unpaidbill | do you have your dial patterns set up in the polycom? it may just be waiting for a digit timeout |
| 18:25.18 | Nugget | it's the telphony equivalent of putting "www." in front of the hostname on an url. |
| 18:25.21 | bmoraca | my dial string is set up: [2-9]11T|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT|xxxxxT|[2-9]xxT|1xT|*xxT|[1-9]*x.T|1xxT|1xxxT |
| 18:25.26 | unpaidbill | guess so |
| 18:25.34 | *** join/#asterisk kerframil (n=kerframi@gentoo/user/kerframil) |
| 18:25.37 | *** join/#asterisk pittstains (n=frank@mx1.distributivenetworks.com) |
| 18:25.40 | bmoraca | the first portion of that says if I dial 411, it should work |
| 18:25.42 | bmoraca | but it doesn't |
| 18:25.53 | *** join/#asterisk neurosys (n=neurosys@adsl-067-032-132-060.sip.bct.bellsouth.net) |
| 18:25.56 | bmoraca | it bounces me back to dial tone and says "Enter more digits:" |
| 18:26.30 | pittstains | hello, i'm having a little trouble with AMD (answering machine detection) |
| 18:26.58 | bmoraca | interestingly enough, if I try to pipe 411 to my asterisk box via a SIP trunk from another asterisk box, it (the destination box) bounces back and says that the extension is not long enough |
| 18:27.18 | bmoraca | which leads me to believe that it's an issue with Asterisk and not the phone |
| 18:27.43 | bmoraca | i'll get the exact error in a moment |
| 18:27.46 | pittstains | when the call is answered, there seems to be some kind of lag before AMD can "hear" what is happening on the line |
| 18:28.34 | pittstains | my dialplan has this: |
| 18:28.35 | pittstains | exten => 0,1,Answer() |
| 18:28.35 | pittstains | exten => 0,n,Monitor(wav,test,m) |
| 18:28.35 | pittstains | exten => 0,n,Verbose(Play a bit of silence to help out AMD) |
| 18:28.35 | pittstains | exten => 0,n,Playback(silence/one-tenth) ; this is a hack to allow a buffer so AMD() does not always return NOTSURE with AMDCAUSE=TOOLONG |
| 18:28.36 | pittstains | exten => 0,n,AMD() |
| 18:28.38 | pittstains | (more stuff) |
| 18:29.15 | pittstains | when i listen to test.wav, the first second or so (which is when most people will say "hello?") isn't there |
| 18:29.19 | bmoraca | the relevant portion of the asterisk console for my error: http://pastebin.com/d69c7a57c (this was dialed from a Polycom 550) |
| 18:30.03 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 18:30.04 | *** join/#asterisk chi6IT41 (n=chigital@tmo-096-232.customers.d1-online.com) |
| 18:30.21 | pittstains | if i say hello a second time, then AMD can "hear" me, and then it detects that I am human |
| 18:31.09 | pittstains | otherwise it times me out, because all it's "heard" for the first x seconds of the call is silence |
| 18:31.24 | pittstains | then i get classified as MACHINE |
| 18:35.07 | ride330 | i got that outbound caller id stuff all done |
| 18:36.28 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 18:37.12 | pittstains | i wonder if this just boils down to a lag issue. if so, are there easy ways to correct this? |
| 18:39.00 | [TK]D-Fender | [13:24]<Bad_Robot->dont' you need to dial 9 to get an outside line? <- how 1980's |
| 18:41.06 | Bad_Robot- | :D |
| 18:42.44 | [TK]D-Fender | bmoraca: thats your "411" peer complaining. Not your phone, and not * |
| 18:43.15 | *** join/#asterisk rdgr (n=rich@82-46-0-91.cable.ubr01.aztw.blueyonder.co.uk) |
| 18:44.56 | bmoraca | my 411 peer IS an asterisk box |
| 18:45.30 | bmoraca | and i just figured it out |
| 18:45.33 | bmoraca | bah |
| 18:45.37 | bmoraca | i knew it was something simple |
| 18:46.58 | bmoraca | i'd forgotten that i was bypassing a certain portion of the dialplan with this phone and my sip trunk and didn't create an exception...thus 411 didn't exist in the context it was needed. |
| 18:47.08 | bmoraca | yay for sip set debug peer. |
| 18:57.53 | *** join/#asterisk loather (n=loather@damnit.us) |
| 18:58.09 | jasonwoot | <[TK]D-Fender>: I'm trying to slowly convert the call center to software phones and wonderfully cheap USB headsets, but they just don't want to give up those handsets |
| 18:58.43 | jasonwoot | I bought some of these riparius.com 3.5 - handsets, these are awesome |
| 18:58.47 | [TK]D-Fender | jasonwoot: So you're trying to get them more? |
| 18:59.47 | *** join/#asterisk zoid_99 (n=chris@router.asteriasgi.com) |
| 19:02.56 | *** join/#asterisk kannan (i=kannan@121.246.243.14) |
| 19:03.00 | kannan | hello |
| 19:04.04 | wino | Hi |
| 19:04.35 | ride330 | hey jason what softphone are you using |
| 19:07.41 | ride330 | i am setting up a new room and the agetns need headsets so i figured why get hard phones when softphones and headsets are cheaper |
| 19:08.44 | kannan | in process of trying to do SLA , with sla.conf. I am trying with eyebeam softphone. The docs state that (a) Two line buttons must be configured to subscribe to the state of |
| 19:08.44 | kannan | the following extensions: - station1 line1 - station1 line2 |
| 19:08.44 | kannan | (b) The line appearance buttons should be configured to dial the extensions |
| 19:08.44 | kannan | that they are subscribed to when they are pressed. |
| 19:09.18 | kannan | any ideas if this is possible on the eyebeam sofphone? |
| 19:10.49 | *** join/#asterisk Maliuta_CA (i=biteme@S0106001a927737b1.fm.shawcable.net) |
| 19:10.58 | *** join/#asterisk harry_v (n=lork@S010600a0c93f6f7e.vs.shawcable.net) |
| 19:15.37 | kannan | :) i think it (SLA) is ended in failure with eyebeam |
| 19:15.50 | *** join/#asterisk Arsenick- (n=rpurcell@modemcable026.33-70-69.static.videotron.ca) |
| 19:16.54 | kannan | on a polycom or cisco 9760 for example how would a line button, subscribe to the state of n exnsion. (I can uderstand registering to aother server or ip account) |
| 19:17.09 | kannan | an extension , i meant |
| 19:17.59 | Arsenick- | Hi all, I would like to know if it's possible to listen for digit when the phone is ringing ? |
| 19:18.38 | [TK]D-Fender | Arsenick-: "core show application dial" <- |
| 19:20.07 | *** join/#asterisk mog (n=mog@nat/digium/x-479c3fa192a35841) |
| 19:20.07 | *** mode/#asterisk [+o mog] by ChanServ |
| 19:20.35 | [TK]D-Fender | kannan: those would not be LINE BUTTONS per se. They are merely WATCHED BUDDIES |
| 19:24.21 | harry_v | Any one seen a case where a ISP rejects a mailhop for the purpouses of vm to email transmital? |
| 19:28.28 | WHYS | scratches her head |
| 19:28.31 | WHYS | I don't seem to have any odbc.so files in my usr/src/asterisk/apps directory. I have unixODBC installed, and /etc/odbi.ini configured. |
| 19:28.32 | WHYS | I did a ./configure, cleaned and remade asterisk (menuselect odbc things). The book says check the 'modules' directory, but I have none. |
| 19:28.58 | [TK]D-Fender | WHYS: /usr/lib/modules/asterisk |
| 19:29.04 | WHYS | AH... |
| 19:30.10 | WHYS | using 1.6... no usr/lib/modules |
| 19:31.39 | WHYS | Ah... you mean /usr/lib/asterisk/modules |
| 19:31.43 | [TK]D-Fender | WHYS: /usr/libe/asterisk/modules |
| 19:31.47 | [TK]D-Fender | -e |
| 19:32.19 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 19:32.22 | WHYS | ok, I have the .so files there. Hummmmmm... |
| 19:33.56 | WHYS | I've been following the book and trying to setup ODBC. I am at the pointof doing an odbc show, but it doesn not list anything. all other tests have worked for each step along the way. What's my next step? |
| 19:41.44 | WHYS | sorry, I'm just a little lost, and new to debuging my setup. I've reloaded * and restarted * manually to look for errors, but don't see anything related. |
| 19:42.08 | *** join/#asterisk kannan (i=kannan@121.246.243.14) |
| 19:42.18 | WHYS | I'm also following the book exactly, but I don't know it things have changed in 1.6 |
| 19:42.22 | kannan | any one know a sip softphone that can have multi line subscribes? mapped to the line appearance buttons |
| 19:43.00 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 19:45.21 | jasonwoot | <ride330> We use Zoiper and occasionally Miaphone |
| 19:46.21 | kannan | jasonwwot, can we config each line individiually? |
| 19:46.33 | kannan | i.e line 3 has got to bee sip/station3 |
| 19:46.49 | kannan | not just muliple accounts |
| 19:47.14 | kannan | but the line buton itself, when pressed automatically can be dialng the given extension |
| 19:54.36 | bmoraca | kannan, SLA is not really supported in Asterisk. There are certain things that can simulate it, but they're really no more than a busy lamp field. |
| 19:55.17 | kannan | bmoraca, yes i am getting it, now , |
| 19:55.19 | kannan | slowly |
| 19:56.01 | ride330 | jasonwoot: do either of those do 729? |
| 19:58.29 | kannan | if we register softphones from different locations, all registering to the same account, can we expect problems? |
| 19:58.43 | kannan | or would it work |
| 19:59.11 | kannan | like a ringall strategy of a different kind? |
| 20:00.22 | pittstains | well, in case anyone is wondering, i've figured it out... i was trying to use AMD to determine whether or not the other end of the call was being answered by a human or a machine. the problem i was running into is that the first second of the call was getting snipped, so my first "hello?" wasn't being registered.... |
| 20:00.30 | *** join/#asterisk LND (n=LND@89.193.213.96) |
| 20:00.40 | pittstains | the problem wasn't with asterisk or AMD so much as my testing device |
| 20:01.23 | pittstains | i was having asterisk place calls to my iPhone -- apparently, when you "answer" the call on an iPhone, the call is not connected immediately |
| 20:01.36 | stintel | iPhone is slooooow :) |
| 20:02.00 | pittstains | the phone hesitates for a bit and it's not until the call duration counter starts on your phone that the call is actually connected |
| 20:02.16 | pittstains | the moral of the story is: don't test with an iPhone! |
| 20:02.32 | pittstains | woulda saved me... oh, half a day |
| 20:03.09 | *** join/#asterisk Linuturk (n=linuturk@fluxbuntu/developer/Linuturk) |
| 20:05.13 | freckle_home | iphone is crap |
| 20:05.48 | stintel | well I still prefer it over any other smartphone |
| 20:05.49 | wino | Haha, well, I say give it time, it's still in beta. |
| 20:06.38 | pittstains | mine was free, so I can't complain too much... except when it causes me to work harder |
| 20:06.51 | *** join/#asterisk stimpie (n=stimpie@84-104-5-227.cable.quicknet.nl) |
| 20:07.37 | *** join/#asterisk t0rrieri (n=torrieri@nelug/crew/torrieri) |
| 20:08.35 | unixdawg | on why 1.6.0.3-rc1 nd 1.6.1-rc1 |
| 20:08.39 | unixdawg | whast the diff |
| 20:08.48 | unixdawg | soory beta 1 |
| 20:09.00 | unixdawg | beta3 |
| 20:09.02 | unixdawg | sorry |
| 20:09.09 | unixdawg | whats the diff between the 2 |
| 20:09.33 | russellb | 1.6.1 has more features than 1.6.0. |
| 20:09.55 | ride330 | are there any free windows softphones that do 729? |
| 20:10.59 | Linuturk | what is your opinion on astlinux? |
| 20:11.05 | unixdawg | nope you have to pay for the g729 |
| 20:12.23 | unixdawg | nice needs work but nice |
| 20:13.10 | *** join/#asterisk viperdude_uk (n=jon@84.45.168.57) |
| 20:13.38 | *** join/#asterisk dijungal (n=kdaniel@205.244.151.188) |
| 20:13.38 | ride330 | is there a way to compile 729 on windows to work with any of the softphoens out there? |
| 20:13.52 | unixdawg | g729 is not a free open codec |
| 20:13.57 | unixdawg | yo have to pay for it |
| 20:14.01 | *** part/#asterisk viperdude_uk (n=jon@84.45.168.57) |
| 20:14.18 | ride330 | okay |
| 20:15.02 | unixdawg | the only softphone out there that might hae it is eyebeam and you have to get the full version and request g729 |
| 20:15.50 | dijungal | hello... can someone help me, i have Asterisk 1.4.21.2 with some queues taking calls for about 15 agents...and mixmonitor recording the calls [MixMonitor(${MON_FILENAME}.wav,b)], but the 2 legs audio is mixing out of sink... any reason why this is and how to fix it? |
| 20:16.03 | *** join/#asterisk ziram19 (n=chatzill@41.226.223.51) |
| 20:17.10 | *** join/#asterisk bijit (n=benji@201.198.72.142) |
| 20:17.26 | *** join/#asterisk etfonhomey_ (n=chatzill@74-143-192-77.static.insightbb.com) |
| 20:17.52 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 20:18.15 | *** join/#asterisk kannan (i=kannan@121.246.243.14) |
| 20:18.30 | dijungal | any ideas?\ |
| 20:18.47 | ziram19 | hello can some one tells me why this is does not work? |
| 20:18.49 | ziram19 | |
| 20:19.26 | ziram19 | exten = _XXX,1,Set(DYNAMIC_FEATURES=automon) ; enable One-touch |
| 20:19.26 | ziram19 | exten = _XXX,2,Dial(SIP/${EXTEN},30,wW) ; wW allow one-touch recording |
| 20:19.26 | ziram19 | exten = _XXX,3,Dial(SIP/${EXTEN}) |
| 20:19.27 | ziram19 | when i tape *1 nothing happens? |
| 20:19.28 | ziram19 | i use x-lite |
| 20:19.52 | Leddy | A sip call comes in to our main number and a user dials the extension. At that point we get 1 way audio (lose inbound). If the caller dials the users DID 2 way audio works fine |
| 20:20.14 | unixdawg | so is it 1.6.0 or 1.6.1 that should be gone with |
| 20:21.09 | unixdawg | working on a new port for bsd |
| 20:21.39 | ziram19 | please can some one tells me what i must do to record a call? |
| 20:22.25 | dijungal | hello... can someone help me, i have Asterisk 1.4.21.2 with some queues taking SIP inbound calls for about 15 agents...and mixmonitor recording the calls [MixMonitor(${MON_FILENAME}.wav,b)], but the 2 legs of the audio is mixing out of sync... any reason why this is and how to fix it? |
| 20:32.44 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-197-154-rrdg-esr-2.dynamic.isadsl.co.za) |
| 20:35.00 | codaine | is there a way to execute applications on the cli? |
| 20:42.29 | *** join/#asterisk andresmujica (n=andresmu@190.27.6.10) |
| 20:45.30 | Bananaskin | is back (gone 21:27:03) |
| 20:53.42 | *** join/#asterisk fexy (n=fexy@208.3.217.29) |
| 20:54.06 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 20:54.22 | fexy | Is there anyway to configure skinny.conf to allow all devices to register? |
| 20:54.46 | fexy | and have the automatically assigned an extension? |
| 20:55.50 | dijungal | channel is dead today |
| 20:55.53 | *** join/#asterisk freckle_home (n=jon@84.45.168.57) |
| 20:58.11 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) |
| 21:08.04 | [TK]D-Fender | codaine: Like? |
| 21:08.39 | dijungal | hello... can someone help me, i have Asterisk 1.4.21.2 with some queues taking SIP inbound calls for about 15 agents...and mixmonitor recording the calls [MixMonitor(${MON_FILENAME}.wav,b)], but the 2 legs of the audio is mixing out of sync... any reason why this is and how to fix it? |
| 21:09.50 | Leddy | Are there any documents to help troubleshoot 1 way audio? |
| 21:10.22 | Leddy | A sip call comes in to our main number and a user dials the extension. At that point we get 1 way audio (lose inbound). If the caller dials the users DID 2 way audio works fine. It appears to be when the trunk/invite is used for the phone |
| 21:10.33 | *** join/#asterisk telecos (n=sergio@6.166.219.87.dynamic.jazztel.es) |
| 21:16.02 | *** join/#asterisk fogo (n=Paul@69.169.132.35.provo.static.broadweave.net) |
| 21:18.39 | *** part/#asterisk tshine (n=tshine@ip70-160-111-108.hr.hr.cox.net) |
| 21:20.24 | *** join/#asterisk RobH (n=RobH@130-227.127-70.tampabay.res.rr.com) |
| 21:25.47 | *** join/#asterisk Docd (n=asd@190.19.19.101) |
| 21:26.20 | Docd | somebody alive? |
| 21:26.52 | ReDNeQ | not me, im here for the cocktails! |
| 21:27.28 | Docd | who's here for the pleasure of helping others? |
| 21:34.24 | wino | Well, I suppose no one is, but you can put your question out there and maybe someone will respond. |
| 21:36.40 | Docd | fair enough |
| 21:36.40 | Docd | i want to set up a sip server that can also dial out with a dialup modem. for what i have googled i think this can't be done with asterisk but i'd like to know if anyone knows another way |
| 21:37.25 | codaine | what do you want to do w/ dialup modem? |
| 21:37.46 | *** join/#asterisk HttpErrors (n=seraphim@unaffiliated/httperror/bot/httperrors) |
| 21:39.35 | Docd | call out through the landline |
| 21:39.57 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
| 21:40.12 | codaine | well, you need an FXO card or SIP ATA for that |
| 21:40.33 | Docd | i read something like that |
| 21:40.47 | Docd | but isn't there a way to do it with a standard dialup modem? |
| 21:41.05 | Docd | maybe not with asterisk but with some other software |
| 21:41.14 | mmlj4 | yes, for some small subset of standard modems |
| 21:41.27 | mmlj4 | there's an intel one, for instance |
| 21:41.28 | codaine | some voicemodems used to do that i think |
| 21:43.10 | Docd | i have an integrated modem in my laptop |
| 21:43.20 | Docd | i was hoping to be able to use it for something |
| 21:44.28 | codaine | well, that modem is probably a software dsp modem, which won't work in that scenario |
| 21:45.06 | codaine | you can just get a sip fxo ata, and hook it up to ur network |
| 21:47.25 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
| 21:48.14 | *** join/#asterisk rue_mohr (n=rue@24.207.122.10) |
| 21:48.21 | rue_mohr | iiiits trouble! :) |
| 21:48.27 | *** part/#asterisk rue_mohr (n=rue@24.207.122.10) |
| 21:48.35 | *** join/#asterisk rue_mohr (n=rue@24.207.122.10) |
| 21:48.40 | rue_mohr | erp |
| 21:49.05 | rue_mohr | ok, asterisk dosn't have port 5060 open for sip, where is the switch I missed? |
| 21:49.36 | rue_mohr | (nmap on local machine) |
| 21:52.15 | *** join/#asterisk Trido (i=trido@ppp178-168.static.internode.on.net) |
| 21:53.07 | loather | rue_mohr: sip.conf under [general] : bindport=5060 bindaddr=0.0.0.0 (on separate lines) |
| 21:53.21 | rue_mohr | hmm I have that, I'll recheck |
| 21:53.36 | *** join/#asterisk hfb (n=hfb@pool-96-247-109-136.lsanca.dsl-w.verizon.net) |
| 21:53.59 | rue_mohr | I have that, the port isn't open |
| 21:54.11 | loather | firewall? selinux? |
| 21:54.16 | rue_mohr | local machine |
| 21:54.55 | rue_mohr | nmap localhost -s 5000-10000 |
| 21:55.33 | mmlj4 | rue_mohr: tell nmap to do UDP |
| 21:56.02 | mmlj4 | also netstat |
| 21:56.03 | *** join/#asterisk TrentCreek (n=kvirc@adsl-75-14-31-197.dsl.hrlntx.sbcglobal.net) |
| 21:56.22 | rue_mohr | netstat dosn't show 5060 |
| 21:56.41 | rue_mohr | I dont know anything else that turns of sip |
| 21:56.54 | mmlj4 | it does if you tell it to do UDP, and you actually have asterisk listening for SIP connections |
| 21:56.56 | rue_mohr | wonder if it was missing something when I compiled it? |
| 21:57.26 | rue_mohr | but to have it listening, I just need the two lines you said, yes? |
| 21:57.43 | mmlj4 | netstat -u |
| 21:57.56 | rue_mohr | MT |
| 21:57.57 | loather | udp 0 0 0.0.0.0:5060 0.0.0.0:* 3047/asterisk |
| 21:57.57 | rue_mohr | :/ |
| 21:58.01 | loather | thats what it looks like |
| 21:58.09 | rue_mohr | its empty |
| 21:58.26 | rue_mohr | so something critical is wrong here |
| 21:59.18 | *** join/#asterisk andresmujica (n=andresmu@correo.seaq.com.co) |
| 21:59.56 | andresmujica | hi, anyone knows how can i send a fax that goes to an internal extension at the destination PBX? |
| 21:59.58 | andresmujica | i've got to dial a number, the machine anwsers and i've got to dial de fax extension number |
| 22:00.04 | andresmujica | but i didn't manage to find a way to send it using winprinhylafax |
| 22:00.10 | andresmujica | i'm using asterisk + iaxmodem + hylafax |
| 22:00.59 | rue_mohr | hu? |
| 22:01.44 | *** join/#asterisk SQLDarkly (n=dakendri@192.147.57.6) |
| 22:01.45 | rue_mohr | looks through configures output for "network libaries... no" |
| 22:02.18 | SQLDarkly | I am looking a the CLI commands but I cannot seem to find how to fire an agi script from the console. Is this possible? |
| 22:02.49 | russellb | SQLDarkly: sort of ... |
| 22:03.01 | russellb | *CLI> originate SIP/myphone application AGI myscript |
| 22:03.11 | SQLDarkly | Only way I was able to think is make an extension to fire it and console dial the exten |
| 22:03.18 | russellb | that tells asterisk to make an outbound call to your phone and then connect it to AGI |
| 22:03.24 | SQLDarkly | ah ok |
| 22:03.31 | russellb | console dial would work, too |
| 22:03.53 | russellb | alternatively, you can do an originate to connect your phone to an extension that runs AGI |
| 22:04.03 | russellb | whatever works for you :-) |
| 22:04.27 | SQLDarkly | I am pleased to say also that I got my conference cluster deployed and is now in production :) step two is to get this cluster at our different sites.... Thanks Asterisk :) |
| 22:04.38 | russellb | you're welcome!! |
| 22:04.44 | russellb | It's nice to hear from happy users :-) |
| 22:04.52 | rue_mohr | checking arpa/inet.h presence... yes |
| 22:04.52 | rue_mohr | checking for arpa/inet.h... yes ok, so network support should be there |
| 22:05.12 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 22:06.05 | rue_mohr | [TK]D-Fender, sip.conf has bindport and bindaddr lines, and netstat shows no 5060 or 10000 port(s) open |
| 22:06.07 | rue_mohr | ideas? |
| 22:06.16 | SQLDarkly | Yes VERY pleased. Genesys is now canceled and Asterisk is powering our teleconferences. I now plan on submitting my app back to the community as I told my company I would develop it under the condition I could release it opensource save their private info |
| 22:06.20 | [TK]D-Fender | rue_mohr: the usual. |
| 22:06.33 | rue_mohr | this is usual for me, hint? |
| 22:06.44 | [TK]D-Fender | rue_mohr: PASTEBIN |
| 22:06.49 | rue_mohr | ok |
| 22:07.49 | SQLDarkly | Its an AJAX app that you can connect to an LDAP(open or AD) auth users. Create,Manage,Schedule Conferences. THinking of porting it to Adobe Air :) |
| 22:11.08 | rue_mohr | [TK]D-Fender, just sip.conf? |
| 22:11.23 | [TK]D-Fender | rue_mohr: EVERYHING |
| 22:11.34 | rue_mohr | hmm |
| 22:11.54 | rue_mohr | will it not open sip ports if things are wrong in other files? |
| 22:12.04 | rue_mohr | pretty borring, most of them are default |
| 22:12.21 | [TK]D-Fender | rursip.conf, netstat dump, maybe some backup for an actual PROBLEM, etc |
| 22:13.07 | loather | SQLDarkly: interesting. i have interest in such a beast to administer LDAP |
| 22:13.55 | rue_mohr | hmm which brings up an interesting point, as netstat dosn't show anything for my working sys at home either |
| 22:15.00 | rue_mohr | ah, its listed under netstat -au, try that on this one... |
| 22:15.14 | rue_mohr | aha! |
| 22:15.37 | *** join/#asterisk Trido (i=trido@ppp178-168.static.internode.on.net) |
| 22:15.40 | rue_mohr | :) |
| 22:15.45 | mmlj4 | [jkelly@samson ~]$ netstat -anu | grep 5060 |
| 22:15.46 | mmlj4 | udp 0 0 0.0.0.0:5060 0.0.0.0:* |
| 22:15.54 | mmlj4 | really... how hard is this? |
| 22:16.13 | rue_mohr | I was missing the an |
| 22:16.25 | rue_mohr | udp 0 0 0.0.0.0:5060 0.0.0.0:* |
| 22:16.31 | mmlj4 | <mmlj4> rue_mohr: tell nmap to do UDP |
| 22:16.31 | mmlj4 | <mmlj4> also netstat |
| 22:16.34 | rue_mohr | ok, this is good |
| 22:16.42 | mmlj4 | like 20 minutes ago |
| 22:16.45 | dijungal | hello... can someone help me, i have Asterisk 1.4.21.2 with some queues taking SIP inbound calls for about 15 agents...and mixmonitor recording the calls [MixMonitor(${MON_FILENAME}.wav,b)], but the 2 legs of the audio is mixing out of sync... any reason why this is and how to fix it? |
| 22:16.59 | rue_mohr | I didn't find the switch before I tried netstat |
| 22:17.06 | SQLDarkly | Its not really for LDAP administration its more for connection to a corporate LDAP that exists and to auth users off of that so they can use their same logins for their conferences as they do for their desktops |
| 22:17.27 | SQLDarkly | However with modification and a user that has admin rights to write to the ldap then it would be possible |
| 22:17.29 | loather | oh i see |
| 22:18.05 | SQLDarkly | loather take a look at phpmyldapadmin i think its called. Its a web based ldap admin tool for openldap |
| 22:18.25 | loather | yea, i use that now but it's kind of a beast |
| 22:18.49 | SQLDarkly | ldap administration is ;) |
| 22:19.06 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 22:19.23 | loather | indeed |
| 22:19.53 | rue_mohr | ok! |
| 22:19.59 | rue_mohr | so, 5060 is open, GOOD |
| 22:20.56 | rue_mohr | next... |
| 22:21.11 | rue_mohr | stupid simple dialplan and get the phone to register |
| 22:22.31 | rue_mohr | http://www.pastebin.ca/1287158 |
| 22:22.50 | rue_mohr | so, if I have the phone right I can dial 111 |
| 22:23.06 | [TK]D-Fender | rue_mohr: No |
| 22:23.09 | rue_mohr | damn |
| 22:23.12 | rue_mohr | answer first? |
| 22:23.23 | [TK]D-Fender | rue_mohr: You seem to have no concept of the dialplan at all. |
| 22:23.31 | [TK]D-Fender | rue_mohr: Go read chapter 5 of the BOOK |
| 22:23.55 | rue_mohr | ~book |
| 22:23.56 | jbot | hmm... book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 22:24.07 | rue_mohr | wondered what that link was |
| 22:24.08 | dijungal | ahhh anyone? |
| 22:24.24 | *** join/#asterisk SQLDarkly (n=dakendri@192.147.57.6) |
| 22:24.34 | eppigy | dijungal: Queue() will record calls |
| 22:24.36 | eppigy | on its own |
| 22:25.25 | dijungal | yes but i want the call named a specific way |
| 22:25.26 | SQLDarkly | When using Dial() or Originate to call an external number and that number passes DTMF. How can I store those tones to be passed back to the dialplan? |
| 22:25.31 | dijungal | i don't like how queue() names the calls |
| 22:25.55 | eppigy | well I think you can set that variable |
| 22:26.00 | eppigy | in the dialplan |
| 22:29.45 | SQLDarkly | I dont think that will work for this purpose. I am calling an external IVR with a php script that fires Originate. The external IVR passes DTMF to me. I need to hold that DTMF and pass those to an agi script php or otherwise |
| 22:30.42 | eppigy | sorry that was directed at dijungal |
| 22:31.08 | SQLDarkly | not a problem :) |
| 22:31.09 | dijungal | oooh |
| 22:31.13 | [TK]D-Fender | SQLDarkly: Do your call-out, then dump THEM into an IVR and read what they send |
| 22:31.14 | dijungal | yes.. but that never worked for me |
| 22:31.19 | dijungal | i will have to look into that again |
| 22:32.13 | freckle_home | SQLDarkly: the phpagi classes will allow you to capture the DTMF |
| 22:32.47 | SQLDarkly | Option G on Dial() to dump them into an IVR... hmm it may work however this test will be performed at a few hundred a second. I need accuracy |
| 22:33.00 | *** join/#asterisk saftsack (n=oliver@g230130160.adsl.alicedsl.de) |
| 22:33.04 | [TK]D-Fender | SQLDarkly: when you call out, who is on each end? |
| 22:33.09 | SQLDarkly | freckle_home really? is tehre any documentation out there on this? |
| 22:33.24 | SQLDarkly | Asterisk and the External IVR(AspectM3) |
| 22:33.25 | freckle_home | google phpagi |
| 22:33.36 | SQLDarkly | no human interaction |
| 22:33.40 | freckle_home | its a class file to develop agi with php |
| 22:34.09 | [TK]D-Fender | SQLDarkly: If * is calling out only to pick up DTMF, athen you are really only sending them into the dialplan. So this is nothing special. Just dump them in your own IVR to read the digits. |
| 22:34.09 | SQLDarkly | freckle_home interesting ill have to check it out. Is the code sound? Production ready I mean? |
| 22:34.27 | [TK]D-Fender | SQLDarkly: This does not imply any required AGI, etc |
| 22:34.37 | freckle_home | you should be able to get the agi to dial out wait for a answer then collect the DTMF |
| 22:35.03 | SQLDarkly | Fender I think it may require AGI to pass the DTMF BACK to the orignating machine the flow is like such |
| 22:35.09 | freckle_home | SQLDarkly: i run a ITSP using phpagi, called 1000's of times a day |
| 22:36.17 | malcolmd | it's certainly the wrong time of day to be asking this, but is anyone in the channel using the b410p with the new dahdi drivers wcb4xxp? |
| 22:36.25 | SQLDarkly | MachineA passes a Sequence number(uniqueID) to * via an XMLHTTPRequest. PHP tells * to originate the call to the External IVR. |
| 22:36.28 | [TK]D-Fender | SQLDarkly: Nope |
| 22:37.02 | freckle_home | SQLDarkly: look for the get_data() in phpagi |
| 22:37.25 | SQLDarkly | It has to be done that way as the uniqueID only comes via an AJAX request so I can only GET that var and pass it in an Orignate from PHP |
| 22:38.18 | SQLDarkly | Dumping them into an IVR to capture DTMF will work still I think. I can read the DTMF and pass those back to a second php script. |
| 22:38.25 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 22:38.34 | SQLDarkly | If I am making this complicated smack me with a hachet in the neck. |
| 22:38.51 | freckle_home | SQLDarkly: you sould be able to do it all in one php script |
| 22:38.53 | [TK]D-Fender | me reaches for his hatchet |
| 22:39.21 | rue_mohr | me squeezes a fluffy teadybear infront of [TK]D-Fender |
| 22:39.35 | freckle_home | FAIL |
| 22:39.41 | SQLDarkly | Fender if this can all be done native in the dialplan how can the dialplan handle an XMLHTTPRequest |
| 22:40.04 | rue_mohr | oh click, right |
| 22:40.09 | rue_mohr | duh |
| 22:40.25 | freckle_home | damn |
| 22:40.34 | freckle_home | can't stop sneezing |
| 22:40.37 | [TK]D-Fender | SQLDarkly: the Originate does not listen for input. and you can pass a parm to a system called script as easily as the next thing |
| 22:40.56 | [TK]D-Fender | (input from XML that is) |
| 22:41.50 | [TK]D-Fender | SQLDarkly: (web stuff) -> Originate -> gets answered -> Listens for digits -> call external script passing the DTMF collected |
| 22:42.43 | SQLDarkly | hmmmmm. I think I understand you. |
| 22:42.50 | rue_mohr | ok, my problem here is that I'm used to my analog phones |
| 22:43.10 | rue_mohr | these sip sets have to dial something |
| 22:43.33 | [TK]D-Fender | rue_mohr: You're right, and analog phones don't have to dial anything either. |
| 22:44.04 | SQLDarkly | Thanks Fender I think I got it. |
| 22:44.54 | *** join/#asterisk rjsystems (n=mrdigita@24.229.167.234.res-cmts.sm.ptd.net) |
| 22:44.58 | SQLDarkly | freckle_home are you speaking of the phpagi.sourceforge.net/phpagi2/docs |
| 22:45.12 | rue_mohr | so if the context just has a definition for 111 to get to hte demo audio, I'm ok, cause there is no delay waiting for digits |
| 22:45.15 | freckle_home | SQLDarkly: yes |
| 22:45.37 | rjsystems | is anyone looking to hire a * Coder? |
| 22:45.38 | *** join/#asterisk ariel_ (n=ariel_@c-24-127-219-186.hsd1.fl.comcast.net) |
| 22:45.47 | SQLDarkly | freckle_home it looks interesting. You currently use this for anything? |
| 22:45.51 | *** join/#asterisk bijit (n=benji@201.198.72.142) |
| 22:45.53 | [TK]D-Fender | rue_mohr: Still not making much sense... |
| 22:45.56 | freckle_home | yes i use it a LOT |
| 22:46.32 | rue_mohr | well the sip phone connects with the number to be dialed, I'm used to having to play audio while waiting for the connection to dial something |
| 22:46.53 | [TK]D-Fender | SQLDarkly: AGI is useful if you have to do a bunch of stuff on both sides of a bunch of dialplan level transactions, or read in channel vars, etc |
| 22:46.57 | ariel_ | Evening Everyone |
| 22:47.18 | rjsystems | hi ariel_ |
| 22:47.20 | bijit | a very good rss ? |
| 22:47.26 | [TK]D-Fender | SQLDarkly: So far your request is summed up by "read one continuous DTMF stream. No multiple events. Therefor who needs to sit around in AGI to read DTMF in? |
| 22:47.50 | bijit | sry wrong chan.. :( |
| 22:47.55 | *** join/#asterisk CrazyTux (n=brandon@rrcs-67-52-124-226.west.biz.rr.com) |
| 22:48.02 | SQLDarkly | [TK]D-Fender: Very much understand. Was simply interested in the phpclass nonetheless ;) |
| 22:48.19 | SQLDarkly | For other purposes outside the scope of this task |
| 22:50.15 | SQLDarkly | exit |
| 22:50.17 | [TK]D-Fender | SQLDarkly: Ok, that is another matter. As long as this doesn't become a tunel-vision exploration of overkill |
| 22:50.18 | SQLDarkly | opps ;) |
| 22:50.41 | SQLDarkly | lol no. I thirst for knowledge is all |
| 22:50.54 | SQLDarkly | I am non going to kill a misquito with a cannon |
| 22:51.14 | [TK]D-Fender | SQLDarkly: Glad to hear. |
| 22:51.22 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 22:51.56 | SQLDarkly | You will see I have a minimalistic approach to my code. Once I release my AJAX app :) Its very clean and certainly not bloated. |
| 22:52.31 | SQLDarkly | anywho I must get this working before I leave or I wont be able to rest easy tonight in the capitol wasteland that is fallout3 |
| 22:52.43 | SQLDarkly | Later all. |
| 22:52.58 | rue_mohr | kees trying to get the aastra to connect |
| 22:54.52 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 22:55.55 | *** join/#asterisk RB2 (n=RB2@pool-71-255-89-136.nwrknj.east.verizon.net) |
| 22:57.37 | rjsystems | anyone need a coder |
| 22:58.57 | [TK]D-Fender | rjsystems: You can stop whoring yourself every few minutes now... |
| 23:02.11 | eppigy | i need a coder that works for free |
| 23:02.27 | [TK]D-Fender | eppigy: Oh... SLUTS we do ;) |
| 23:02.36 | rjsystems | eppigy: what do you need done> |
| 23:02.46 | eppigy | SLUTS |
| 23:02.55 | eppigy | rjsystems: just some dumb php stuff |
| 23:03.00 | rjsystems | like? |
| 23:03.10 | eppigy | ok this company has this frontend |
| 23:03.14 | eppigy | that runs reports |
| 23:03.17 | rjsystems | ok |
| 23:03.21 | eppigy | well when you initially click |
| 23:03.33 | eppigy | for some reason it select the entire damn cdr table |
| 23:03.48 | eppigy | I need that shit to be changed |
| 23:03.53 | rjsystems | ok |
| 23:04.18 | eppigy | you realize im just jokingn though |
| 23:04.26 | eppigy | I cant just put shit in production |
| 23:04.35 | rjsystems | ?? |
| 23:04.36 | eppigy | that I had some dude in #asterisk modify |
| 23:04.42 | rjsystems | so test it |
| 23:04.43 | mmlj4 | why not? microsoft does |
| 23:04.48 | eppigy | mmlj4: lol true |
| 23:04.58 | beek | mmlj4: all except the testing part... |
| 23:05.07 | eppigy | oh boy |
| 23:05.19 | eppigy | hair trigger violence |
| 23:05.22 | eppigy | about to ensue |
| 23:08.56 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 23:11.40 | *** join/#asterisk hmmhesays (n=hmmhesay@97-114-162-143.farg.qwest.net) |
| 23:12.59 | xorl | hmm, i have a phone number that calls my system (on purpose) with an automated voice message system, not voice mail, but when the number calls and I try to go through the menu system |
| 23:13.10 | xorl | It does not see/hear/catch the button presses |
| 23:15.09 | rue_mohr | am I not right that the first thing I should work on is getting it so that I can see the sip phone register? |
| 23:15.13 | SwK | anyone know a good high usage provider that does unrestricted flat rate? |
| 23:15.27 | rue_mohr | with debug and verbose at 10 wouldn't I see any registrations that were enven attempted/ |
| 23:15.42 | [TK]D-Fender | rue_mohr: "sip set debug on" <--- |
| 23:15.53 | rue_mohr | hah its seperate, great |
| 23:15.57 | [TK]D-Fender | rue_mohr: Verbose doesn't say anything about SIP convos it ignores |
| 23:16.10 | [TK]D-Fender | AFK for a while |
| 23:16.16 | *** join/#asterisk codaine (n=Onur@198.64.168.130) |
| 23:16.32 | rue_mohr | 1.4 sip set debug |
| 23:16.47 | rue_mohr | 1.6 + on? |
| 23:17.54 | rue_mohr | I'd swear I never set that on 1.2 |
| 23:18.10 | rue_mohr | oh wait, its set in sip.conf |
| 23:18.16 | rue_mohr | thats why I never had to set it |
| 23:18.18 | rue_mohr | ok |
| 23:18.37 | rue_mohr | so, still not seeing it log in |
| 23:19.03 | denon | so, I hate to even ask it .. but any of you have much hands-on with Avaya IP Office? |
| 23:19.06 | rue_mohr | this is evil, you have no way of knowing if my sip phone is set up right |
| 23:20.15 | rue_mohr | maybe I shoudl try the polycom phone first, its much more verbose |
| 23:20.30 | *** join/#asterisk WindBack (i=jorge@201-213-250-41.net.prima.net.ar) |
| 23:20.44 | *** join/#asterisk propellerhead (n=yogurt2u@218-13-16-190.fibertel.com.ar) |
| 23:21.13 | rue_mohr | [TK]D-Fender, is there a way I can break up building this system into smaller atomic pieces? this is kinda a 'get it all working at once' kinda thing |
| 23:23.47 | rue_mohr | ok, now I have to write a polycom 601 (turns out its not a 600) from scratch cause there isn't on I can find on google that has all the entries in it |
| 23:27.22 | xorl | hmmm |
| 23:27.54 | mmlj4 | rue_mohr: you're making all of this way too difficult |
| 23:28.05 | mmlj4 | rue_mohr: get a phone, ANY phone, to register |
| 23:28.09 | rue_mohr | well PLEASE tell me how to complify |
| 23:28.19 | rue_mohr | er simplty |
| 23:28.31 | mmlj4 | set up a trivial extension, then call it |
| 23:28.35 | rue_mohr | look, I'v been on a 0 sleep plan, and will be the rest of the week |
| 23:28.41 | rue_mohr | I'm trying :) |
| 23:28.50 | mmlj4 | fair enough :-) |
| 23:29.01 | rue_mohr | the aastra wont register, I dont know why |
| 23:29.08 | *** join/#asterisk cellofellow (n=josh@209-193-111-14.mammothnetworks.com) |
| 23:29.12 | mmlj4 | the book, did you see the book? the book gives step-by-step almost |
| 23:29.43 | rue_mohr | k i'll look, but the port, ip address username loginname thisname thatname and password are all correct |
| 23:29.54 | cellofellow | Using my little home file/print/dns/web server, can I use Asterisk to make calls through SIP on my home telephone line? |
| 23:30.10 | mmlj4 | can you ping the phone, or load its IP into a browser? |
| 23:30.27 | rue_mohr | the panic is setting in, I'm hoping I'm gonna not go stupid and not listen to the help I get |
| 23:30.30 | mmlj4 | cellofellow: you need an ATA |
| 23:30.36 | rue_mohr | ok, ping it, yes, good idea |
| 23:31.06 | cellofellow | mmlj4: what's that? |
| 23:31.41 | *** join/#asterisk snapper14 (n=snapper1@cpc3-reig1-0-0-cust301.hers.cable.ntl.com) |
| 23:31.45 | mmlj4 | cellofellow: http://www.voip-info.org/wiki-ATA |
| 23:31.49 | *** join/#asterisk jeffspeff2 (n=Administ@c-98-211-62-9.hsd1.ky.comcast.net) |
| 23:32.15 | codaine | cellofellow: its one of those boxes that can bridge between ur asterisk box and the phone line. like the vonage boxes, if that makes sense to you |
| 23:32.45 | cellofellow | codaine: ok, the Wiki page makes it look like it's a stand alone box that doesn't need Asterisk. |
| 23:33.01 | mmlj4 | it doesn't |
| 23:33.06 | cellofellow | (I suppose using just an old modem to interface with the telephone line isn't going to work) |
| 23:33.23 | mmlj4 | you suppose correct, for 99% of modems |
| 23:34.06 | cellofellow | does that 1% exist? |
| 23:34.07 | codaine | cellofellow: ATAs are actually only adapters (as the name implies) between the voice line and the network |
| 23:34.13 | cellofellow | ok |
| 23:34.46 | cellofellow | any guesses on how much they cost? |
| 23:34.57 | codaine | mostof them are very cheap on ebay |
| 23:34.58 | mmlj4 | yes, they exist, but they're problematic and you still have to purchase hardware... go get a linksys or whatever ATA... the correct kind of ATA, there are two |
| 23:35.14 | *** join/#asterisk rue_mohr (n=rue@24.207.122.10) |
| 23:35.35 | *** join/#asterisk etfonhomey (n=chatzill@74-143-196-254.static.insightbb.com) |
| 23:35.37 | cellofellow | http://www.google.com/products/catalog?q=VoIP+Gateway&btnG=Search+Products&cid=16698171681693510686#ps-sellerslike this? |
| 23:35.39 | rue_mohr | oopsed my network cable... |
| 23:35.40 | codaine | cellofellow: if you like to play w/ hardware you can get one provider specific box for a discount and unlock it |
| 23:35.54 | rue_mohr | ok I can ping the phone |
| 23:35.58 | rue_mohr | its a polycom 601 |
| 23:36.06 | rue_mohr | its not configured at all |
| 23:36.23 | Madkiss | hi all. |
| 23:36.23 | cellofellow | Provider specific like VoIP company specific or telephone company specific? |
| 23:36.58 | Madkiss | i have a setup where two asterisks communicate with each other. what i want to reach is that a call that comes in via iax2 and is prefixed with 5 gets redirected into the "default-out" context. how do I do this? |
| 23:37.02 | mmlj4 | so configure it |
| 23:37.08 | rue_mohr | hold up |
| 23:37.10 | codaine | cellofellow: voip company |
| 23:37.17 | cellofellow | k |
| 23:37.47 | rue_mohr | I think I can only conigfutre it from tftp, gonna try manual... |
| 23:38.03 | codaine | Madkiss: something like Goto(default-out,s,1) in your dialplan ? |
| 23:38.03 | cellofellow | Finding them on Ebay, already unlocked, from $40-70, ok thanks. |
| 23:38.17 | etfonhomey | Madkiss, in the context that you have for your iax2 connection, use a GoTo() |
| 23:38.29 | codaine | cellofellow: yep, should be pretty cheap. make sure u get a fxo box, not a fxs |
| 23:38.47 | cellofellow | What might the difference be? |
| 23:39.25 | Madkiss | codaine, etfonhomey: thank you very much |
| 23:39.28 | codaine | cellofellow: FXS provides the dialtone (energy) to the line, FXO uses the dialtone |
| 23:39.30 | codaine | http://www.digium.com/en/docs/misc/fxs_fxo_desc.php |
| 23:39.49 | cellofellow | ok |
| 23:40.20 | rue_mohr | ok, lets see if the polycom will register |
| 23:40.50 | cellofellow | Ok, so FXO is like the client and FXS is like the server? |
| 23:41.24 | rue_mohr | FX O for office, like the service provider FX S for set like the set on your desk |
| 23:42.07 | codaine | cellofellow: yes, you can assume that. basically if you want to use a line with dialtone provided (like the line coming to ur place) use a FXO. |
| 23:42.25 | Madkiss | codaine: okay, and what if default-out does not have an "s"-extension but only suchstarting with _? |
| 23:42.32 | *** join/#asterisk Leddy (n=Leddy@polar.artica.net) |
| 23:42.58 | codaine | Madkiss: anything matching _? should go through |
| 23:43.27 | cellofellow | ok, so I connect the FXO ATA to my network and telephone line, and point Asterisk on my server to the ATA, and then I can point my SIP client at Asterisk to make (and maybe receive?) calls? |
| 23:44.20 | codaine | cellofellow: yes |
| 23:44.25 | mmlj4 | rue_mohr: you want to do it manually... tftp or ftp is for larger installations |
| 23:44.36 | Madkiss | codaine: erm. default-out actually is a context that only includes lots of other contexts |
| 23:44.48 | WHYS | Seen this? : Limit should be a number, not a boolean: '0'. Disabling ODBC class 'asterisk'. |
| 23:44.57 | rue_mohr | I found it, I thought I recalled that it didn't have a manual set for that |
| 23:45.11 | cellofellow | Seems a bit more technical than I expected. Well, if I do get an ATA it'll be fun to set up. Thanks guys. |
| 23:45.16 | rue_mohr | it didn't register |
| 23:45.35 | Madkiss | "sent into invalid extension default-out". wtf. |
| 23:45.56 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 23:47.03 | codaine | Madkiss: so add a Goto() to whatever context/extension/priority combination you want to. |
| 23:47.56 | rue_mohr | witht eh tfpt server set up I can see its log though |
| 23:48.39 | Leddy | Any ideas on what would cause 1 way audio when dialing main number and having the IVR dial the extension vs dialing a users did and having 2 way audio? |
| 23:49.05 | Madkiss | codaine: is it a problem if the extension that is actually called is defined _after_ the default-out context in extensions.con? |
| 23:49.57 | *** part/#asterisk snapper14 (n=snapper1@cpc3-reig1-0-0-cust301.hers.cable.ntl.com) |
| 23:51.16 | rue_mohr | it still didn't register |
| 23:52.10 | rue_mohr | there is NOTHING in the asterisk console |
| 23:52.20 | rue_mohr | even whne I try to make a call |
| 23:52.24 | *** join/#asterisk JAMMAN2110 (i=James@unaffiliated/jamman2110) |
| 23:53.32 | rue_mohr | question, how come I can get this to work through 2 firewall to my home * server and not to a local one? |
| 23:54.14 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:54.33 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:54.56 | edoceo | Hey all, I've got an odd issue - when an extension is set to forward only internal calls to that extension are forwared |
| 23:55.00 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 23:55.04 | edoceo | External calls coming in are not forwareded |
| 23:55.40 | *** join/#asterisk JAMMAN2110 (n=James@unaffiliated/jamman2110) |
| 23:57.05 | rue_mohr | when the phone is finished booting, I should be able to see a few events in the asterisk console, right? sip debug is on, core verbose is 10 core debug is 10 |
| 23:57.53 | edoceo | rue_mohr: yes - should see something |
| 23:58.10 | rue_mohr | I dont |
| 23:58.16 | rue_mohr | I'm confused |