| 00:00.48 | *** join/#asterisk dennis- (n=dennis@cipherpunk.org) |
| 00:00.51 | dennis- | hey guys! |
| 00:07.42 | *** join/#asterisk Benny_132 (n=benmarti@144.131.132.107) |
| 00:08.40 | Benny_132 | hi guys need some advice on how to record call on an asterisk machine |
| 00:08.51 | dug | is /var/lib/asterisk/sounds set in any asterisk config file? |
| 00:09.16 | _ShrikE | Benny_132: http://www.voip-info.org/wiki-Asterisk+cmd+monitor |
| 00:14.16 | *** join/#asterisk jsuter (n=jsuter@openwrt.us) |
| 00:18.27 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-2ab7f6430094827e) |
| 00:19.12 | *** join/#asterisk Tene (n=tene@poipu/supporter/slacker/tene) |
| 00:19.25 | Tene | Is there a preferred source for asterisk packages for Fedora? |
| 00:20.34 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 00:23.38 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 00:24.00 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-2e4cfbfdef42faea) |
| 00:25.58 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 00:25.58 | *** mode/#asterisk [+o mog] by ChanServ |
| 00:29.41 | riddlebox | [TK]D-Fender, you around yet? |
| 00:33.02 | *** part/#asterisk Tene (n=tene@poipu/supporter/slacker/tene) |
| 00:37.16 | *** join/#asterisk rvhi (n=chatzill@66.135.230.96) |
| 00:37.39 | rvhi | is there an app to list all variables of a channel? mainly for testing? |
| 00:37.45 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 00:40.56 | *** join/#asterisk sergey (n=sergey@gw4-130.iks.ru) |
| 00:45.42 | *** join/#asterisk lowlevel (n=Stuart@74.103.188.10) |
| 00:49.51 | *** join/#asterisk nDuff (n=ccd@fw2.isgenesis.com) |
| 00:55.43 | Qwell | mog: How is your intarweb? |
| 00:57.11 | *** join/#asterisk Krurst (n=me@eth244.wa.adsl.internode.on.net) |
| 00:58.00 | Krurst | anyone have any ideas on how to stop chan_mobile segfaulting? |
| 00:58.19 | hmmhesays | what is chan_mobile? |
| 00:58.33 | Krurst | it used to be called chan_cellphone |
| 00:58.39 | Qwell | hmmhesays: bluetooth channel drivers for phones/headsets |
| 00:58.46 | Qwell | s/rs/r/ |
| 00:59.19 | Krurst | it's awesome when its working |
| 00:59.33 | Qwell | Krurst: check the bug tracker. bugs.digium.com |
| 01:03.09 | mog | Qwell: its wonderous |
| 01:03.13 | Qwell | meh |
| 01:03.17 | mog | i transfered big file to my sister in law today |
| 01:03.23 | mog | no problems |
| 01:03.33 | Qwell | it's good right this second? |
| 01:05.41 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 01:08.47 | Benny_132 | _ShrikE: that works a treat thankyou |
| 01:09.17 | *** join/#asterisk davixx (n=davixx@193.189.125.231) |
| 01:09.45 | *** join/#asterisk blinky42 (n=me@c-71-230-47-244.hsd1.pa.comcast.net) |
| 01:12.13 | _ShrikE | np |
| 01:13.09 | Krurst | bugger. the fix for a problem I was having seems to be "get a different bluetooth adapter". |
| 01:18.10 | Qwell | Krurst: I'm inclined to say that that's wrong...but you never know |
| 01:18.26 | Qwell | who was it that said that? |
| 01:18.38 | Krurst | http://bugs.digium.com/view.php?id=10443&nbn=5 |
| 01:18.47 | Qwell | I can't access the web right now |
| 01:19.02 | Krurst | it seems some bluetooth adapters might have a race condition or something |
| 01:19.09 | Qwell | oh, hmm |
| 01:19.17 | Qwell | if dbowerman said it, I'd believe it |
| 01:19.52 | Qwell | if it was somebody else, I might take it with a grain of salt |
| 01:20.04 | Krurst | I tried updating to get rid of the problem, and thats when my segfaults started. |
| 01:20.20 | Krurst | But it looks like it might not get rid of the problem anyway |
| 01:20.57 | Krurst | well dbowerman hasn't said anything one way or the other |
| 01:21.26 | Krurst | It looks like he needs a faulty dongle to replicate the problem. |
| 01:24.48 | dug | does asterisk use the language conf section, I set http://pastebin.com/m6babe937 in my asterisk.conf and asterisk still cannot find my sound files unless I put a full path in the playback |
| 01:25.29 | Qwell | dug: 1.4? there's another option for language paths in asterisk.conf.. I forget what the option is though, lemme see if I can find it |
| 01:25.38 | dug | 1.2 |
| 01:26.08 | Qwell | oh, no idea then |
| 01:26.25 | Qwell | make sure your prompts are in the right place - run strace, and see where it's looking for them at |
| 01:27.19 | *** join/#asterisk |dennis| (n=dennis@200.32.236.18) |
| 01:27.54 | dug | Qwell: not sure how to run strace |
| 01:29.02 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583591.dsl.bell.ca) |
| 01:30.00 | dug | Qwell how can I run strace to see where it is looking |
| 01:30.15 | Qwell | strace asterisk -c |
| 01:30.16 | Qwell | it's gonna be spammy |
| 01:31.40 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 01:32.22 | *** join/#asterisk chendy (n=chendy@218.242.110.26) |
| 01:33.18 | CCFL_Man2 | anyone here do voip dial peers in cisco? |
| 01:35.43 | Kwakwa | I have an * server (a) installed which is connected to another one (b) over IAX2, where b has zaptel installed. I now want to trunk them over IAX2 but as a doesn't have zaptel it won't support trunking without it (for the timing). I've installed ztdummy and turned on trunking but it gives the same error. Do I need to recompile it? :/ |
| 01:36.58 | Krurst | Kwakwa: does zttest work on the server with ztdummy? |
| 01:38.14 | nDuff | I'm trying to do a Dial() which bridges two local IAX devices with a S() parameter causing the call to be terminated after a given amount of time. However, Asterisk appears to be bridging the two devices directly, thus getting itself out of the path and removing its ability to enforce a maximum call length. How do I prevent this? |
| 01:38.38 | dug | Qwell: still no luck ... just says custom/mainmenu |
| 01:38.46 | Kwakwa | Krurst: I'll check, ta |
| 01:38.52 | bintut | is anyone here uses the fax feature of asterisk ? how is it? does it provides good quality? |
| 01:39.00 | nDuff | bintut: what fax feature? |
| 01:39.22 | nDuff | bintut: I use Asterisk+iaxmodem+hylafax and it works very well, but I don't know what this "fax feature" you speak of is. |
| 01:39.50 | nDuff | bintut: I certainly don't trust rxfax/txfax, if that's what you're referring to. |
| 01:40.45 | Krurst | anyone know what "Incoming call: Got SIP response 400 "Out Of Order" back from 10.0.0.18" means? The phone at 10.0.0.18 is working fine. |
| 01:40.47 | bintut | nDuff: first, sending fax from a traditional fax machine inside the asterisk box to a pstn fax machine through the zap channel |
| 01:41.35 | bintut | nDuff: second, two asterisk boxes peered with each other can send/receive faxes.. |
| 01:41.40 | nDuff | bintut: sure, that works fine so long as you don't have timing issues on your zap device. Lose too many interrupts or fail to be sync'd to the PSTN's clocking and you have all kinds of problems. |
| 01:41.41 | Kwakwa | Krurst: Yeah it works on both sides, more accurate on A than B |
| 01:42.25 | bintut | nDuff: basically, the users don't need to know that there is an asterisk box.. they don't need to learn new things. |
| 01:42.31 | nDuff | bintut: as for the second scenario, peered with each other *how*? You're not going to get reliable faxing over any kind of VoIP protocol other than T.37/T.38.. |
| 01:42.42 | Krurst | Kwakwa: ok, is chan_zap compiled in the the B server? |
| 01:43.41 | bintut | nDuff: that's what i was about to ask.. honestly, i don't know much about faxing but i was able to hear about this t.38.. |
| 01:44.02 | nDuff | bintut: if I were you, I'd just stick to using POTS for faxing. |
| 01:44.05 | Kwakwa | Krurst: That's what I was unsure of, I've not recompiled it since adding ztdummy. I guess I need to recompile it --with-zap |
| 01:44.06 | Kwakwa | ? |
| 01:44.23 | bintut | nDuff: so, based on your experience, is asterisk able to handle/provide quality fax messages? |
| 01:44.26 | nDuff | bintut: faxing over zap channels is rock-solid, if your configuration and hardware is good. |
| 01:44.34 | nDuff | bintut: faxing any other way, I'd avoid. |
| 01:44.40 | nDuff | s/is good/are good/ |
| 01:45.22 | nDuff | bintut: I'm in the process of setting up a 92-line (4-PRI) iaxmodem-based softfax server, so I'd damned well better trust asterisk for reliable faxing. :) |
| 01:45.25 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 01:45.26 | bintut | nDuff: what do you mean by avoiding faxing any other way? you mean for peered asterisk? |
| 01:46.10 | nDuff | bintut: faxing over SIP or IAX should be avoided, yes. |
| 01:46.16 | Krurst | Kwakwa: do a "modules show" and see if chan_zap is listed |
| 01:46.34 | nDuff | bintut: "peered asterisk" doesn't mean much -- you could peer two servers over TDM, and then it would be fine. |
| 01:46.36 | Krurst | module show sorry |
| 01:46.48 | nDuff | bintut: please be more specific about what you're asking. |
| 01:47.35 | bintut | nDuff: sorry, i'm a newbie.. i'm asking because i want to know based on your experience.. |
| 01:47.38 | nDuff | bintut: TDMoE is peering every bit as much as an IAX trunk is. |
| 01:47.58 | nDuff | bintut: so when you ask me if you can do reliable faxing between two peered asterisk servers, I can't answer you because it depends on how you peer them. |
| 01:48.50 | bintut | nDuff: so what do you suggest for a good (if not the best) quality faxing service for an end to end over IP line? |
| 01:48.50 | Kwakwa | nDuff: What server are you running that 92-line softfax server? |
| 01:50.00 | nDuff | Kwakwa: what do you mean, hardware? it's a PowerEdge 2950 with a 4-span Sangoma card. |
| 01:50.41 | Kwakwa | ahh nice |
| 01:50.51 | Kwakwa | Krurst: Just checking now |
| 01:51.48 | Kwakwa | nDuff: I'm actually trying to set up a fax service on one of my boxes, had all kinds of trouble with app_tx and app_rx |
| 01:52.03 | nDuff | bintut: I don't have a recommendation for that, because personally I avoid doing it. If I *were* going to do it, I'd try to peer them via t.37 or t.38. |
| 01:52.16 | Kwakwa | nDuff: U recommend iaxmodem then? (U got a good source on how to set it up?) |
| 01:52.23 | nDuff | Kwakwa: I'm a HylaFAX bigot, so I wouldn't touch {tx,fx}fax with a 10-foot pole. |
| 01:52.30 | Kwakwa | :) |
| 01:53.11 | nDuff | Kwakwa: no good source other than the README and the mailing list archive. |
| 01:53.49 | Kwakwa | damn, I like it wen guides hold my hand |
| 01:55.01 | aptura | nduff, I may be in a position to use fax back in my line of work. Dial in get IVR for work order number then fax number to send to. Hangup then get the fax from the server. I think in a case like this it has to be PSTN. Now can * transmit and recieve fax from the server to pstn or is that not a option? |
| 01:55.30 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 01:55.33 | Kwakwa | nDuff: I've used hylaxfax before over analog, couldn't get enough information out of it though to be useful |
| 01:55.42 | aptura | I have not looked into it but it is nice because it eliminates the human factor. |
| 01:55.49 | Kwakwa | nDuff: does it use spansp still for it? |
| 01:56.12 | nDuff | Kwakwa: yes, it uses spandsp, but it has a much better class 1 implementation than {tx,rx}fax. |
| 01:56.15 | aptura | Kwakwa was the text missing or jiberish? |
| 01:56.52 | nDuff | aptura: that seems pretty reasonable. |
| 01:57.08 | Kwakwa | aptura: It only passed the caller id, I needed to DDI too but couldn't get that. As we have multiple lines I need to know where to send it based on the DDI, so I had to hack it :/ |
| 01:57.45 | nDuff | Kwakwa: it does pass the DID in using iaxmodem. |
| 01:58.32 | nDuff | Kwakwa: you can also pass arbitrary other data in from asterisk to iaxmodem as part of the call string and have that come out accessible to the notify info as one of the caller ID strings. |
| 01:58.43 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 01:59.07 | aptura | nduff, yes it may be. It would be nice to have thougt of this weeks ago so I would not need to chace down a web enables PC in the field. Not always practicle. This project is winding down so the next project is not going to be like this one. |
| 02:00.41 | Kwakwa | nDuff: sounds good |
| 02:01.21 | aptura | well im going to relax since its been a long day. |
| 02:01.50 | Kwakwa | nDuff: I've tried to get it working in the past with app_rx but it kept crashing *, would you say it was app_rx rather than spandsp being screwed? |
| 02:02.35 | *** join/#asterisk ST-3 (i=ser@dipsy.tch.org) |
| 02:02.46 | bintut | nDuff: peering with t.37 or t.38? example for that one? |
| 02:02.48 | Kwakwa | Wish I could help nDuff, when do they bridge? |
| 02:03.44 | bintut | nDuff: i was thinking on what's the best solution to cut cost on faxing documents over 2 different locations |
| 02:06.37 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 02:06.44 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 02:07.32 | tengulre | my asterisk box cut fax when it receive half of one, why? (FAX--->PSTN--->FXO--->Asterisk-->FXO-->PBX-->FAX) |
| 02:07.50 | *** join/#asterisk |dennis| (n=dennis@200.32.236.18) |
| 02:08.05 | nDuff | Kwakwa: yes, I'd absolutely say that; iaxmodem has been rock-solid for me, while I had stability issues with app_rx when playing with it myself. |
| 02:08.22 | riddlebox | does anyone use the Grandstream HT488? |
| 02:08.45 | nDuff | Kwakwa: best I can tell, they're bridging pretty much immediately after a Dial() between them, even though that Dial has an S() argument. |
| 02:12.26 | *** join/#asterisk litage_ (n=nick@70.55.220.203.static.comindico.com.au) |
| 02:12.39 | *** join/#asterisk Snake-eyes (n=hgjg@70.55.220.203.static.comindico.com.au) |
| 02:13.51 | Echinos | So, has anyone been able to realize $$ savings with their cell phone by setting up a callback function and using a cheap/unlimited incoming calls plan on their cell? |
| 02:13.54 | nDuff | tengulre: faxing is pretty sensitive to timing. Does zttest show a minimum value lower than 99.98? Is your internal FXO clocked to synchronize with the PSTN one, and the PBX configured to take its timing from there? |
| 02:14.09 | Echinos | I'm planning on trying something like that. |
| 02:17.10 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 02:18.30 | sergey | Hi. What different in connection via IAX with trunk=yes and trunk=no? |
| 02:19.08 | fujin | multiple calls are trunked down a single connection |
| 02:19.08 | fujin | vs. not |
| 02:19.09 | Kwakwa | http://www.voip-info.org/wiki-Asterisk+bandwidth+iax2 |
| 02:19.12 | fujin | I assume, anyway :) |
| 02:21.13 | sergey | TKS a lot |
| 02:23.07 | sergey | well and how limit outgoing calls via IAX2 (ex. 3 calls only)? |
| 02:23.19 | bintut | nDuff: thanks.. i have to leave now.. thanks.. :) |
| 02:27.42 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:32.07 | *** join/#asterisk Yourname`` (n=IM@unaffiliated/yourname/x-837320) |
| 02:33.25 | Yourname`` | This afternoon, I asked two questions. 1) How can I log a phone number of a call that fails to go through? (call failed to go through, reason 0) |
| 02:33.52 | Yourname`` | 2) How can I create a custom log of every step in a dialplan that was made by the called party. |
| 02:34.32 | Yourname`` | I got an answer to the first one, saying use DIALSTATUS and HANGUPCAUSE in the dialplan. However, these two vars don't seem to work when used in conjunction with call files. |
| 02:35.59 | fujin_ | that's because they're call files dude :) |
| 02:36.08 | fujin_ | what are you using call filse for, anyway? |
| 02:36.09 | *** join/#asterisk beek (n=klinebl@static-71-240-222-16.alt.east.verizon.net) |
| 02:36.19 | Yourname`` | Then came across http://tinyurl.com/2r5o6e where it says funny things. |
| 02:36.25 | Yourname`` | For dialing out, fujin_ |
| 02:37.22 | fujin_ | did you read the followups to that? |
| 02:37.31 | Yourname`` | Yup, nothing concrete. |
| 02:38.06 | Yourname`` | Just one, and that answers an age old question of mine! :D |
| 02:39.11 | Kwakwa | If you write your calls to a database you can get calls which don't go through with a basic `select * from cdr where disposition='NO ANSWER'` and obviously add more criteria to wittle down to what you're after |
| 02:39.44 | Yourname`` | What do you mean write your calls to a db? |
| 02:39.47 | Kwakwa | The `src` will hold the caller id |
| 02:40.02 | Kwakwa | Well, if you have cdr logging enabled |
| 02:40.48 | Yourname`` | All I really want to do is log all the phone numbers that seem disconnected. |
| 02:40.55 | Yourname`` | That's phase 1. |
| 02:41.05 | Kwakwa | http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf |
| 02:41.41 | Kwakwa | I'm assuming your using the 'h' for hangup? (Checking the hangup cause etc..?) |
| 02:41.44 | Yourname`` | Phase 2 is logging all dialplan stuff. Like for example, a call is made, a msg is played. Press 1 and it goes to play another msg (log that press1), and then the other msg requires you to press 9 or something (log that press9), phne number, etc. |
| 02:42.04 | Kwakwa | Thats kind of written to log files anyway |
| 02:42.16 | Kwakwa | although a bit messy :) |
| 02:43.09 | Kwakwa | . |
| 02:43.13 | Kwakwa | \/var/log/asterisk/* (assuming you have logger.conf set up accordingly) |
| 02:44.26 | Yourname`` | Yeah, that's the thing. |
| 02:44.37 | Yourname`` | Don't wanna have the whole cdr spiel going on. |
| 02:44.52 | Kwakwa | Well it would be nice :) |
| 02:45.20 | Kwakwa | If you use a database which support views you can make some funky things happen |
| 02:45.20 | Yourname`` | But you got what I'm looking for though, right? |
| 02:45.28 | Kwakwa | yeah |
| 02:45.35 | Kwakwa | in the log directory you'll find that info |
| 02:45.59 | Kwakwa | are you familiar with the asterisk CLI? |
| 02:46.05 | Yourname`` | Yes.. |
| 02:46.23 | Kwakwa | It basically writes what you see there (with a certain amount of verbosity) to the log files |
| 02:46.36 | Yourname`` | Everything? |
| 02:46.50 | Kwakwa | it'll write whatever you want it to write |
| 02:46.55 | Kwakwa | vim /etc/asterisk/logger.conf |
| 02:47.07 | Kwakwa | it explains everything in the config file |
| 02:49.02 | Yourname`` | So if I want something like 41962525555|Pressed1|Pressed3|spoke to agent|hung up |
| 02:49.15 | Yourname`` | Or 4194645447|DISCONNECTED |
| 02:49.22 | Yourname`` | Logger.conf will take care of it? |
| 02:49.30 | Kwakwa | hmm, not really |
| 02:49.45 | Yourname`` | That's what I meant by custom log. |
| 02:49.48 | Kwakwa | u will need to do a bit of bash scripting to be honest |
| 02:50.01 | Kwakwa | You could be hacky and use the NoOp command |
| 02:50.10 | Yourname`` | That's what I'm doing right now, lol |
| 02:50.17 | Kwakwa | haha |
| 02:50.20 | Kwakwa | then there you go :) |
| 02:50.24 | Yourname`` | Using NoOp everywhere. Saving CLI output. And searching for NoOps. |
| 02:50.30 | Yourname`` | BUT, there has to be an easier way! |
| 02:50.54 | Kwakwa | if you use the NoOp commands and they're written to the logs, u can do some simple bash to get them out |
| 02:50.55 | Yourname`` | I mean it only makes sense I'd like reporting on what happened to a number when it was called. |
| 02:51.29 | CCFL_Man2 | oh noes!!!! |
| 02:51.47 | CCFL_Man2 | the adit 600 fxs 4g card doesn't support pulse dialing |
| 02:51.47 | Kwakwa | like if you set all the NoOp's with NoOp(--YY--|${UNIQUEID}|dialled) |
| 02:52.17 | Kwakwa | you can `grep --YY-- /var/log/asterisk/full` |
| 02:52.24 | Krurst | some places still need pulse dialing?!? |
| 02:52.38 | Kwakwa | and then pipe that to something else, hacky but I don't believe something you're after exists |
| 02:52.44 | Kwakwa | unless you use an AGI() |
| 02:53.04 | Yourname`` | Hmm. |
| 02:53.08 | CCFL_Man2 | Krurst: yes, western electric phone geeks |
| 02:53.51 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 02:53.52 | Krurst | wow. The 50's called and they want their signalling back. |
| 02:54.12 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 02:54.12 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:54.19 | CCFL_Man2 | was used up until the 80s |
| 02:55.21 | pkunkra | its really amazing how G.711 eats 200mbps of bandwidth |
| 02:55.24 | Krurst | well I suppose I could still use it today, thanks to backwards comapatability. |
| 02:55.32 | pkunkra | my god that's a lot of data |
| 02:55.56 | CCFL_Man2 | you can use it on almost every POTS connection |
| 02:56.42 | CCFL_Man2 | plus, new phones suck balls |
| 02:56.59 | Yourname`` | Totally sucks. :( |
| 02:57.08 | CCFL_Man2 | nothing works or sounds better than an old western electric phone |
| 02:57.47 | CCFL_Man2 | Yourname``: exactly |
| 02:58.19 | CCFL_Man2 | the phones of the 500 model era, they work farking perfectly |
| 02:58.47 | tengulre | nDuff: Average: 99.987793 |
| 02:59.35 | Krurst | I'd suggest getting some sort of adaptor, but you'd probably get an interdigit timeout whenever you dial 0 anyway. |
| 03:00.23 | CCFL_Man2 | you mean 1? |
| 03:00.43 | Krurst | isn't 0 10 disconnects? |
| 03:00.49 | CCFL_Man2 | right |
| 03:01.44 | Echinos | Has anyone else been having trouble with IAX and FWD? |
| 03:02.08 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 03:02.36 | Echinos | I made ana ccount recently, but on the forums it says there's some problem on the server |
| 03:02.44 | *** join/#asterisk CrashSys (n=kumba@158-211.187-72.tampabay.res.rr.com) |
| 03:02.54 | Echinos | I'm not sure I want to forward all those rtp ports thru my firewall for sip :/ |
| 03:03.00 | Krurst | I just realised the only analogue phone I have access to is a fax machiene :( |
| 03:03.12 | Echinos | but I suppose I don't have much choice... |
| 03:04.17 | CrashSys | I'm trying to put Asterisk between the PSTN and an old Comdial FXII... Both systems using a T1 PRI... I know I can route from Asterisk to the comdial's pri, but i'm trying to figure out if I can have Asterisk features.conf once the pri has routed the channel... |
| 03:05.11 | CCFL_Man2 | Krurst: i advise you get some western electric phones |
| 03:05.34 | Krurst | I'm in australia. They're not so common here. |
| 03:05.53 | CCFL_Man2 | oh, you are in the land of non bellcore |
| 03:06.54 | Krurst | yeah, the land of Telstra |
| 03:08.02 | CrashSys | Will features.conf's #1 transfer mechanism still work if I have Zap/1 call Zap/25? (1 pri to another) |
| 03:08.31 | CrashSys | assuming it will do a native bridge |
| 03:09.45 | russellb | CrashSys: yes, it will work as long as the call is going through asterisk |
| 03:09.47 | *** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
| 03:10.03 | CCFL_Man2 | wtf the fuck the fluff |
| 03:10.06 | russellb | it will not work if the call gets completely transferred away with 2BCT (b-channel transfer) |
| 03:10.19 | CCFL_Man2 | dammit, pulse dialing does work |
| 03:10.33 | russellb | heh, of course pulse works :) |
| 03:10.34 | CrashSys | Yes... I am going to do a migration... The call path will be PSTN --> Asterisk Span1 ---> comdial on Asterisk Span2 |
| 03:10.51 | russellb | then yeah, it will work just fine |
| 03:11.06 | CCFL_Man2 | russellb: this is on an adit 600 |
| 03:11.13 | CrashSys | Russel: You are the man |
| 03:11.31 | CCFL_Man2 | my dial must not be working right |
| 03:12.49 | *** join/#asterisk obnauticus (n=obnautic@c-76-115-29-47.hsd1.wa.comcast.net) |
| 03:12.57 | obnauticus | hey umm how do i configure *67 stuff on asterisk |
| 03:13.09 | obnauticus | like do an NAI lookup when I dial the extension *67 |
| 03:14.10 | *** join/#asterisk watchy (n=watchy@h120.184.255.206.cable.cmdn.cablelynx.com) |
| 03:15.04 | fujin_ | what the hell is an NAI? |
| 03:15.52 | CCFL_Man2 | fukin a |
| 03:16.19 | *** part/#asterisk beek (n=klinebl@static-71-240-222-16.alt.east.verizon.net) |
| 03:16.39 | *** join/#asterisk Juggie (n=Juggie@CPE00c0f02a7bf2-CM001371886eee.cpe.net.cable.rogers.com) |
| 03:17.01 | obnauticus | ANI funjin_ |
| 03:17.03 | obnauticus | my bad |
| 03:17.17 | CCFL_Man2 | my tone phone set to pulse works on the adit 600 |
| 03:19.20 | watchy | i gotta sell a * box with 12 incoming lines 31 extensions what kinda hardware should i go with |
| 03:19.41 | obnauticus | What I mean is, can you get another person's phone number even if they call you with *67 |
| 03:20.51 | CCFL_Man2 | my rotary dials won't work |
| 03:23.27 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 03:24.47 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:25.03 | [TK]D-Fender | watchy, Here |
| 03:25.17 | [TK]D-Fender | Yourname``, You're not using the right kind of channel. Think on this... |
| 03:25.31 | [TK]D-Fender | Yourname``, (for your call-file question) |
| 03:25.38 | Yourname`` | k |
| 03:25.52 | *** join/#asterisk map7 (n=map7@teksup41.lnk.telstra.net) |
| 03:26.45 | *** join/#asterisk Juggie (n=Juggie@CPE00c0f02a7bf2-CM001371886eee.cpe.net.cable.rogers.com) |
| 03:27.05 | *** part/#asterisk Juggie (n=Juggie@CPE00c0f02a7bf2-CM001371886eee.cpe.net.cable.rogers.com) |
| 03:27.21 | *** join/#asterisk t3rror (n=t3rror@adsl-065-005-255-180.sip.owb.bellsouth.net) |
| 03:27.47 | t3rror | what is the best way to check call audio quality once you have a working pbx setup? |
| 03:27.56 | t3rror | i want to check my audio quality going out teliax |
| 03:28.05 | Yourname`` | You call your cell phone. |
| 03:29.16 | J4k3 | haha |
| 03:29.21 | t3rror | its kinda hard to guage if i am talking into one handset and it is coming out another in that close proximity |
| 03:29.38 | J4k3 | got AMPS? otherwise, don't bother... digital cellular sounds like shit. |
| 03:29.52 | *** join/#asterisk Juggie (n=Juggie@CPE00c0f02a7bf2-CM001371886eee.cpe.net.cable.rogers.com) |
| 03:30.18 | CCFL_Man2 | making a liar out of me |
| 03:31.29 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 03:31.57 | Yourname`` | [TK]D-Fender: I still don't get it.. :S |
| 03:32.15 | *** join/#asterisk Tili (n=tili@cm48.gamma244.maxonline.com.sg) |
| 03:32.44 | CCFL_Man2 | what does T mean in a destination-pattern? |
| 03:32.50 | CCFL_Man2 | with cisco |
| 03:36.30 | watchy | tk: here what? |
| 03:36.31 | CCFL_Man2 | i think i know why this isn't working |
| 03:36.42 | CCFL_Man2 | my we302 is pulsing too fast |
| 03:36.49 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 03:41.23 | fujin_ | anyone know if I can delay the execution of a macro? |
| 03:41.27 | fujin_ | I'm using a local channel for call delivery |
| 03:41.36 | fujin_ | and it tries to run the macro multiple times simultaneously |
| 03:41.39 | fujin_ | and causes a cpu spike |
| 03:41.52 | fujin_ | Wait(x); doesn't work |
| 03:43.21 | watchy | i gotta sell a * box with 12 incoming lines 31 extensions what kinda hardware should i go with. motherboard/cpu/how much ram |
| 03:44.07 | bkruse_home | watchy: what kind of work will the box be doing? |
| 03:44.14 | bkruse_home | 12 lines, are you using POTS or a PRI? |
| 03:44.33 | bkruse_home | you could get a tdm2400p and then have room to grow... |
| 03:45.03 | bkruse_home | 31 extensions, probably a maximum of 10 calls at the same time, you should be fine with anything with decent cache size and > 512megs of ram though its one of the hardest things to guess... |
| 03:45.06 | *** join/#asterisk famicon (i=pastry@c51447ddc.cable.wanadoo.nl) |
| 03:45.37 | watchy | PRI using a sangoma PCI echo cancellation card |
| 03:45.46 | bkruse_home | eww |
| 03:46.01 | watchy | man i aint had nothing but issues from digium |
| 03:46.08 | bkruse_home | Explain :D |
| 03:46.13 | watchy | i had 2 TDM400ps and they were pieces of shit |
| 03:46.17 | bkruse_home | you know digium does a 100% gaurentee now, cutting out the crap. |
| 03:46.20 | watchy | they got hit by lightning thank god |
| 03:46.36 | watchy | they take lightning damage cards back? |
| 03:46.43 | bkruse_home | RMA no questions now, they are really taking a turn with hardware..... |
| 03:47.03 | watchy | i had so many problems with echo my client wanted to RAPE me |
| 03:47.05 | bkruse_home | watchy: honestly, they might if you were throgholy disappointed with the card before the fact, worth a shot. |
| 03:47.30 | bkruse_home | lol, did you ever even run fxotune and mess with your module rx/tx gain settings and your zapata rx/tx gain settings |
| 03:48.02 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:48.09 | bkruse_home | watchy: so I would give it a shot |
| 03:48.10 | watchy | yea |
| 03:48.36 | watchy | well the cards are dead now. probably outta warranty to |
| 03:48.38 | bkruse_home | watchy: they really took and stand and said enough is enough, its very cool if you ask me....always making improvements also |
| 03:48.54 | watchy | well if they are improving thats great. they needed to |
| 03:49.03 | bkruse_home | watchy: I dont think matters....I would do that, and also, next time, mess with the gain settings |
| 03:49.16 | watchy | they did help me one time, but when the box rebooted it was back to echoing like crazy |
| 03:49.19 | bkruse_home | i might make a tutorial on cancelling echo that way, and now they offer software based echo can, and hardware based.... |
| 03:49.37 | bkruse_home | thats probably from the module settings.....that got overriden when you restarted... :/ |
| 03:49.51 | CCFL_Man2 | watchy: 1u 1ghz box with 512mb ram and two port t1 card |
| 03:49.54 | bkruse_home | and they did need changes, and they got them, i agree... |
| 03:49.58 | bkruse_home | thats fine, thatll work fine. |
| 03:50.23 | watchy | ccfl: someone told me using intel 965 chipsets caused problems is that true |
| 03:51.15 | CCFL_Man2 | watchy: honestly i'm not sure, if in doubt then use something non via and non sis |
| 03:51.21 | watchy | unfortunately the * box is going to be trixbox |
| 03:52.02 | CCFL_Man2 | then specify that |
| 03:52.10 | watchy | these guys i work with at this new company sure love trixbox |
| 03:52.14 | watchy | i think its a bad idea |
| 03:52.22 | CCFL_Man2 | i'd use a server motherboard |
| 03:52.32 | watchy | i think eventually trixbox is going to not be free anymore and fuck every idiot using it |
| 03:52.37 | *** join/#asterisk bmg505 (n=leon@196.209.180.200) |
| 03:52.46 | watchy | plus if you rely on a gui for everything thats a bad idea |
| 03:52.58 | bkruse_home | watchy: http://asterisknow.org |
| 03:53.15 | bkruse_home | watchy: thats not 100% true, its mainly, if you rely on a GUI but know how to do it at the command line...its bad |
| 03:53.30 | bkruse_home | but starting from the gui....is not all bad. the asterisk gui has made substantial growth. |
| 03:53.32 | *** join/#asterisk Teln1100A (i=hello123@bas2-toronto12-1128738098.dsl.bell.ca) |
| 03:54.18 | watchy | i just don't think trixbox is a good idea for a huge install |
| 03:54.28 | Teln1100A | How can I use asterisk to make 2 calls and bridge them together? Not connect them to any extensions or phones just bridge the 2 outgoing calls |
| 03:54.32 | bkruse_home | watchy: Have you used the asterisk gui? |
| 03:54.41 | watchy | no |
| 03:55.25 | watchy | this * now looks intresting |
| 03:55.40 | fujin_ | it's shit |
| 03:55.42 | fujin_ | why bother |
| 03:55.44 | fujin_ | vi is all the gui you need |
| 03:55.45 | fujin_ | tbh |
| 03:55.51 | Teln1100A | hi fujin |
| 03:55.51 | fujin_ | vi+screen |
| 03:55.53 | fujin_ | hi |
| 03:55.58 | Teln1100A | what bout keyboard |
| 03:56.08 | watchy | haaha |
| 03:56.15 | fujin_ | keyboard doesn't really count for gui now does it |
| 03:56.21 | fujin_ | input, yes, gui, no |
| 03:56.34 | Teln1100A | I guess you are right |
| 03:56.40 | Teln1100A | now about my question |
| 03:56.42 | bkruse_home | fujin_: vim is my hero, correct. |
| 03:56.44 | Teln1100A | How can I use asterisk to make 2 calls and bridge them together? Not connect them to any extensions or phones just bridge the 2 outgoing calls |
| 03:56.53 | fujin_ | die in a fire |
| 03:57.14 | bkruse_home | fujin_: I would like to see you use the GUI without a keyboard. |
| 03:57.20 | bkruse_home | onscreen keyboard? your lame. :P |
| 03:57.34 | fujin_ | yes, but a keyboard != gui |
| 03:57.55 | bkruse_home | I think you mean keyboard != functional use |
| 03:58.07 | fujin_ | no, I mean keyboard != gui |
| 03:58.10 | fujin_ | thanks for your input though |
| 03:58.19 | Teln1100A | I say use whats best for the job |
| 03:58.22 | bkruse_home | You are awesome. |
| 03:58.25 | bkruse_home | s/awesome/lame/g |
| 03:58.33 | watchy | haha |
| 03:58.38 | Teln1100A | if you can do something in gui which would take much longer on command line go for it |
| 03:58.40 | bkruse_home | ~lart fujin_ |
| 03:58.40 | jbot | changes fujin_'s permissions to 0777 and tells the world |
| 03:58.51 | fujin_ | I'm on a private address. |
| 03:59.01 | fujin_ | changing my permissions would also require access |
| 03:59.06 | fujin_ | ugh, why even bother? |
| 03:59.08 | Teln1100A | on the other hand if you need to say ping something fire up command line cos gui would be longer / not equivalent |
| 03:59.10 | bkruse_home | fujin_: wtf are you talking about? |
| 03:59.13 | bkruse_home | its a bot... |
| 03:59.21 | fujin_ | You're a dumbass |
| 03:59.23 | bkruse_home | it says funny stuff. |
| 03:59.30 | fujin_ | seriously, why try to correct someone if you're going to be a dumbshit? |
| 03:59.36 | fujin_ | <bkruse_home> I think you mean keyboard != functional use |
| 03:59.39 | fujin_ | I see what you did there |
| 03:59.39 | bkruse_home | "i have a private adress behind my windows firewall!" |
| 03:59.47 | fujin_ | lol @ windows |
| 03:59.51 | bkruse_home | lol @ you |
| 03:59.51 | fujin_ | way to assume too much |
| 03:59.57 | Teln1100A | the address is private |
| 04:00.04 | Teln1100A | but the firewall is not strong |
| 04:00.11 | bkruse_home | "my address starts with 192.!!!" |
| 04:00.13 | Teln1100A | ie you get pwned |
| 04:00.19 | fujin_ | lol |
| 04:00.21 | fujin_ | yeah, sure; |
| 04:00.35 | Teln1100A | not all 192 are private |
| 04:00.39 | Teln1100A | only 192.168 |
| 04:00.49 | bkruse_home | fujin_: Can I give you my public address and see what you can do with it? I have sshd running, along with apache |
| 04:01.05 | fujin_ | now why would I do that? |
| 04:01.06 | Teln1100A | how bout the root password while you are at it? |
| 04:01.08 | fujin_ | You're not paying me to pentest you. |
| 04:01.12 | *** join/#asterisk osiris (n=osiris@c-71-205-35-230.hsd1.mi.comcast.net) |
| 04:01.12 | fujin_ | are you, now? |
| 04:01.24 | bkruse_home | are you worth more than $0.10? |
| 04:01.38 | Teln1100A | a second sure |
| 04:01.51 | fujin_ | That was a hoot |
| 04:01.53 | fujin_ | a real knee slapper |
| 04:02.04 | fujin_ | I enjoyed that thoroughly. |
| 04:02.05 | bkruse_home | ok country boy, save the talk for #nolife |
| 04:02.11 | Teln1100A | $0.10 a second = $360 an hour |
| 04:02.22 | fujin_ | top of the class |
| 04:02.30 | Teln1100A | back to voip ppl |
| 04:02.32 | bkruse_home | "I believe that chmod 0777 applies to IP address'!" |
| 04:02.46 | fujin_ | That's not what I said |
| 04:02.56 | bkruse_home | ok... |
| 04:02.57 | fujin_ | 'telling the world' would imply that the world could connect |
| 04:03.03 | fujin_ | to whatever device I was chmod 0777'd on. |
| 04:03.16 | fujin_ | are you like an Australian or something? |
| 04:03.23 | fujin_ | I'm not quite sure you understand the english language |
| 04:03.27 | bkruse_home | telling the world would imply connectivity between the world...and/or people of it |
| 04:03.28 | watchy | hahaha |
| 04:03.31 | bkruse_home | aka......irc. |
| 04:03.42 | fujin_ | are you ircing as root? |
| 04:03.43 | bkruse_home | take note of the "relay" in the acro |
| 04:03.50 | bkruse_home | yes, hack me now please. |
| 04:03.57 | fujin_ | that's pretty classy |
| 04:04.02 | bkruse_home | seriously, try it |
| 04:04.06 | bkruse_home | wow, you can do an irc whois |
| 04:04.28 | fujin_ | are you an internet professional? |
| 04:04.40 | jwh | certified? :P |
| 04:04.53 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 04:04.56 | bkruse_home | whos assuming now? |
| 04:04.59 | Damin | Morning.. |
| 04:05.04 | bkruse_home | assume = ass-u-me |
| 04:05.07 | Damin | Anyone going to Ohio Linuxfest? |
| 04:05.18 | Damin | WE're having an Asterisk Meet and Greet there.. |
| 04:05.31 | Damin | http://digg.com/linux_unix/Bradley_Kuhn_and_Max_Spevak_to_speak_at_Ohio_Linux_Fest/blog |
| 04:05.31 | bkruse_home | Damin: where in? |
| 04:06.01 | Damin | Ohio Linuxfest in Columbus, Ohio... Digium is sponsoring the conference bags this year.. |
| 04:06.33 | bkruse_home | haha, sweet |
| 04:06.39 | bkruse_home | I will have to grab one before they ship :P |
| 04:06.50 | bkruse_home | thats very exciting, however. Whats the details on the "Meet and Greet" ? |
| 04:07.36 | Damin | bkruse: We meet, we greet, we talk about Asterisk.. and I'll probably hand out Digium swchag that I've got leftover from the Digium ISPCon booth.. |
| 04:07.55 | bkruse_home | Damin: Thats very sweet, what day now? |
| 04:08.09 | bkruse_home | I might be up there for family type issues, taking a break from huntsville...ugh |
| 04:08.09 | Damin | bkruse: Sat, Sept 29th.. |
| 04:08.36 | Damin | bkruse: I'd be very appreciative if you could pass that Digg link around.. we're trying a little social experiment now.. |
| 04:08.40 | *** join/#asterisk linagee_ (n=linagee@about/linux/staff/linagee) |
| 04:09.20 | bkruse_home | Damin: np, will do |
| 04:09.40 | *** join/#asterisk linagee_ (n=linagee@about/linux/staff/linagee) |
| 04:13.24 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 04:17.51 | watchy | any problems with Dual Core proccessors and *? |
| 04:19.34 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 04:19.54 | Yourname`` | http://www.binarytag.com/EXMAIL/mailinglists/binarytag_asterisk-users/msg03212.html |
| 04:20.02 | Yourname`` | This rocks, seriously. |
| 04:20.16 | Yourname`` | I've asked that question so many times, googled it .. and I never got any answers. |
| 04:20.22 | Yourname`` | And today, it shows me! |
| 04:20.22 | Yourname`` | :D |
| 04:27.39 | *** join/#asterisk kiscokid (n=xxx@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 04:28.35 | kiscokid | anyone understand how followme.conf works? The sample file doesn't seem to have any per user settings |
| 04:32.07 | watchy | any problems with Dual Core proccessors and *? |
| 05:00.53 | *** join/#asterisk linagee (n=linagee@about/linux/staff/linagee) |
| 05:17.26 | *** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
| 05:28.28 | *** join/#asterisk linagee (n=linagee@about/linux/staff/linagee) |
| 05:36.02 | cavediver | I coulnd't find any documentation on the GUI? Anyone have a link ? |
| 05:38.38 | *** join/#asterisk Nuitari (n=Nuitari@melchior.nuitari.net) |
| 05:39.05 | hmmhesays | what up folks |
| 05:39.31 | Nuitari | I just got my hopes shattered by the LOCAL/ channel |
| 05:39.34 | kiscokid | cavediver: I think all there is is in the installation tarball |
| 05:39.46 | cavediver | Hm ok |
| 05:40.38 | Nuitari | I'm trying to use the call out features of Asterisk, but I need the flexibility of LOCAL/ with the error handling of SIP/ |
| 05:41.13 | Nuitari | hum |
| 05:43.23 | Nuitari | w00t found what I needed |
| 05:43.51 | hmmhesays | way to be fantastically vague |
| 05:44.33 | Nuitari | When using Action: Originate with a LOCAL/ channel, it's very important not to send it to Voicemail |
| 05:46.11 | hmmhesays | What do you guys think of "The Lost Packet" as a networking company name |
| 05:47.41 | Teln1100A | Nuitari, do you know how I can initiate 2 calls simultanously then bridge them together? Both will be PSTN outbound calls ie no one speaking from Asterisk side |
| 05:47.50 | Nuitari | it invokes images of unreliability |
| 05:48.01 | Nuitari | Teln1100A: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate |
| 05:48.03 | Nuitari | that's one way |
| 05:48.43 | Teln1100A | but I dont want it to link to a sip channel at all internally |
| 05:48.46 | Teln1100A | just 2 external calls |
| 05:49.02 | Nuitari | yeah, you can use LOCAL as the channel |
| 05:49.08 | Nuitari | http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out <-- other ways |
| 05:49.25 | Nuitari | Teln1100A: http://www.voip-info.org/wiki/view/Asterisk+local+channels |
| 05:50.11 | Nuitari | Teln1100A: that is what I am using |
| 05:50.35 | Teln1100A | thanks |
| 05:50.56 | Nuitari | Action: Originate |
| 05:50.56 | Nuitari | Channel: LOCAL/4000@inside |
| 05:50.56 | Nuitari | Context: inside |
| 05:50.56 | Nuitari | Exten: 95555551212 |
| 05:50.56 | Nuitari | Priority: 1 |
| 05:50.57 | Nuitari | ActionID: testAAAAAAAAAAAA |
| 05:51.11 | Nuitari | the Context is the context for Exten |
| 05:51.22 | Nuitari | Channel, being Local, includes the dialplan context in it |
| 05:53.42 | *** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 05:54.33 | Teln1100A | hmmm |
| 05:54.43 | Teln1100A | this is in extensions.conf |
| 05:54.44 | Teln1100A | ? |
| 05:55.15 | kiscokid | Anyone know of a web-based way to control MeetMe conference rooms? |
| 05:55.53 | Nuitari | Teln1100A: that's using the manager api to originate calls |
| 05:56.49 | Teln1100A | I guess that would involve custom programming of some sort? |
| 05:56.55 | Nuitari | yeah |
| 05:57.34 | Teln1100A | is there an easy way say perhaps using teh CLI |
| 05:58.40 | Teln1100A | I am trying to use Asterisk for cheap international calling where one channel dials my home phone the other dials international |
| 05:58.53 | Teln1100A | the asterisk is running on a VPS with fast link so barely any lag |
| 05:59.14 | kiscokid | what is a VPS? |
| 05:59.20 | Teln1100A | Virtual Private Server |
| 05:59.36 | Teln1100A | you have Root on it but there are more than one on the same Server |
| 05:59.40 | Teln1100A | isolated |
| 06:00.06 | kiscokid | so, its like running under Vmware? |
| 06:00.12 | Teln1100A | something like that |
| 06:00.17 | Teln1100A | virtuozo actually |
| 06:00.53 | Teln1100A | but its like a quad xeon with 8+gigs of ram |
| 06:01.46 | *** join/#asterisk linagee (n=linagee@about/linux/staff/linagee) |
| 06:02.22 | kiscokid | how is the voice quality? I tried running * on Microsoft Virtual PC and sound quality was awful. |
| 06:02.32 | watchy | any problems with Dual Core proccessors and *? |
| 06:02.45 | Teln1100A | its excellent |
| 06:02.59 | Teln1100A | Microsoft Virtual Pc is not a very mature product |
| 06:03.03 | Teln1100A | try vmware server |
| 06:03.04 | Teln1100A | its free |
| 06:03.13 | Teln1100A | and better for linux / trixbox |
| 06:03.17 | Teln1100A | or asterisk |
| 06:03.18 | kiscokid | yeah, gonna try that next |
| 06:03.25 | Teln1100A | plus way more features |
| 06:03.29 | Teln1100A | snapshots |
| 06:03.32 | Teln1100A | isolated networks |
| 06:03.39 | Nuitari | vmware is better then virtuozzo |
| 06:03.40 | Teln1100A | cloning |
| 06:03.54 | Teln1100A | well virtuozo is used in hosting environments |
| 06:03.58 | watchy | ninja turtless |
| 06:04.02 | Nuitari | Teln1100A: not that I know |
| 06:04.14 | kiscokid | is virtuozo free? |
| 06:04.22 | Teln1100A | common in VPS market |
| 06:04.26 | Teln1100A | no virtuozo is not free |
| 06:04.36 | Teln1100A | its more a commercial product then for end users |
| 06:05.06 | Teln1100A | Thats the company http://futurehosting.biz/vps.htm |
| 06:05.20 | kiscokid | well, my client is looking for a virtualization solution to combine a bunch of servers in a co-lo |
| 06:05.39 | Teln1100A | you need to own the servers? |
| 06:05.44 | kiscokid | yeah |
| 06:05.55 | Teln1100A | rent has its benefits |
| 06:06.01 | Teln1100A | free hardware replacements |
| 06:06.08 | Teln1100A | and you can move boxes freely |
| 06:06.14 | kiscokid | so far they don't want to rent |
| 06:06.40 | Teln1100A | well check out theplanet.com |
| 06:06.43 | kiscokid | can you add machines for big loads for say, two weeks at a time? |
| 06:07.04 | Teln1100A | well if you own the box then yea |
| 06:07.15 | Teln1100A | run whatever virtual machine whenever |
| 06:07.21 | kiscokid | no, I mean in the VPS world |
| 06:07.28 | Nuitari | there is vmware esx and qemu that can move vms around |
| 06:07.44 | Nuitari | qemu is open source and supports Kernel Virtual Machines, which makes VMs way faster |
| 06:07.53 | Nuitari | I heard that very recent vmwares also support it |
| 06:07.58 | Teln1100A | virtuozo does that as well |
| 06:08.11 | Teln1100A | shared stuff is only stored once in memory accross virtual machines |
| 06:08.18 | Nuitari | I run a double dual-core server that uses qemu extensively |
| 06:08.20 | Teln1100A | if it changes, it is copied |
| 06:08.39 | Teln1100A | Vmware ESX does that |
| 06:08.54 | Nuitari | Vmware Server doesn't |
| 06:09.01 | Teln1100A | not server |
| 06:09.05 | Teln1100A | but vmware ESX |
| 06:09.11 | Teln1100A | which is the baremetal version |
| 06:09.18 | Nuitari | yeah |
| 06:09.22 | Teln1100A | which doesnt use a host OS |
| 06:09.29 | Nuitari | and costs a fortune and a half |
| 06:09.41 | Teln1100A | true, but the benefits are there |
| 06:09.43 | CCFL_Man2 | isdn service over T1 access lines, can it be done with a cisco providing the T1? |
| 06:09.45 | Teln1100A | banks use it |
| 06:09.59 | Nuitari | banks also use asterisk |
| 06:10.13 | CCFL_Man2 | channel banks use it, yes |
| 06:10.33 | Teln1100A | do calling card providers use asterisk too? |
| 06:11.01 | Nuitari | no idea |
| 06:11.15 | Nuitari | my banks call come from "Asterisk <9000>" |
| 06:11.46 | *** join/#asterisk af_ (n=getsmart@81-174-47-128.dynamic.ngi.it) |
| 06:12.36 | kiscokid | CCFL_Man2: are you talking about a PRI? |
| 06:12.53 | Teln1100A | do you know how .call files are used? |
| 06:13.00 | CCFL_Man2 | kiscokid: bri |
| 06:13.55 | CCFL_Man2 | kiscokid: my channel bank gives me an isdn bri u interface |
| 06:14.14 | kiscokid | CCFL_Man2: is it connected to a telco? |
| 06:14.30 | CCFL_Man2 | and the modes it uses is 3 ds0 or 4:1 tdm mode |
| 06:14.40 | CCFL_Man2 | it's connected to an mc3810 |
| 06:15.06 | CCFL_Man2 | mc3810 provides the clocking |
| 06:17.57 | kiscokid | CCFL_Man2: are you going to attach the BRI to a digium or Sangoma card? |
| 06:18.19 | *** join/#asterisk klaus3000 (n=ramon@nat.labs.nic.at) |
| 06:18.23 | CCFL_Man2 | no, i wanted to get an nt1 and attach pots to it |
| 06:20.03 | *** join/#asterisk ryans0921 (n=ryan@ip70-162-10-201.ph.ph.cox.net) |
| 06:20.32 | CCFL_Man2 | but i'm not sure what to configure in the cisco |
| 06:21.14 | kiscokid | CCFL_Man2: that's way beyond what I know about. |
| 06:22.04 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:22.35 | ryans0921 | does anyone have a cisco 7970 working correctly? |
| 06:22.46 | Mavvie | hundreds. |
| 06:22.48 | Mavvie | oh, with asterisk. |
| 06:22.48 | jql | just have a 7961 |
| 06:22.48 | Mavvie | no |
| 06:23.01 | Nuitari | I stopped buying Cisco after 2 major failures |
| 06:23.48 | kiscokid | Phone failures? |
| 06:23.50 | ryans0921 | its driving me nuts... I have it up and running... registered... i can call between extensions... call voicemail... but when I try to make an outbound call it does a fast busy and says reorder... :( |
| 06:24.04 | jql | what? |
| 06:24.06 | jql | that's nuts |
| 06:24.27 | ryans0921 | ya.. doesnt make any sense... and if i connect a linksys and dial out it works fine... |
| 06:24.28 | jql | your ciscoo should be failing WAYYY earlier than that. :) |
| 06:24.34 | kiscokid | ryans: can you make outside calls on anything? |
| 06:24.34 | ryans0921 | lol |
| 06:24.51 | CCFL_Man2 | kiscokid: what about this, how do i do incomming sip calls to the cisco? just call the cisco's ip with a number that matches one of it's destination patterns? |
| 06:25.20 | Nuitari | kiscokid: Major networking components, including a 4 T1 dialup bank and the core router (I think 7000 series) this was 7 years back so my memory is failing a bit |
| 06:25.20 | CCFL_Man2 | ryans0921: which firmware you using? |
| 06:25.25 | ryans0921 | ya... on the linksys it goes out fine... i can get incoming calls from my analog pots lines as well |
| 06:25.31 | ryans0921 | 8.0.3 |
| 06:25.40 | jql | you sure the cisco isn't appending a # or something? |
| 06:26.01 | CCFL_Man2 | ryans0921: 8.0.3 is the right version to use |
| 06:26.11 | ryans0921 | hmmm... not sure.. how would i check that? |
| 06:26.20 | CCFL_Man2 | possibly the dial plan isn't set right |
| 06:26.35 | ryans0921 | in the phone itself or the server? |
| 06:26.52 | CCFL_Man2 | see what the server gets |
| 06:26.56 | jql | if you can't exclude one, it could be either. :) |
| 06:27.00 | kiscokid | CCFL_Man2: I guess I would try that but I really have no experience in what you're doing |
| 06:27.23 | CCFL_Man2 | kiscokid: ahh, ok, honestyly neither do i :P |
| 06:27.38 | kiscokid | ryans: you should be able to see the resulting outgoing number on the * console |
| 06:27.42 | Teln1100A | can a .call file be used to make 2 outgoing calls and conference them? |
| 06:27.47 | CCFL_Man2 | ryans0921: i would check the dial plan too on the phone |
| 06:28.41 | jql | Teln1100A: yes, you can do that |
| 06:29.22 | ryans0921 | kiscokid: when i watch the console i see nothing after i dial... |
| 06:29.43 | Nuitari | Got to go, bye |
| 06:29.56 | CCFL_Man2 | ryans0921: you see the number it's getting from the phone? |
| 06:30.06 | Teln1100A | heres my syntax for 1, but how would I add the other? http://pastebin.com/m71d9cb4 |
| 06:30.07 | kiscokid | ryan: are you connecting to the console with asterisk -vvvvr ? |
| 06:30.10 | *** part/#asterisk Nuitari (n=Nuitari@melchior.nuitari.net) |
| 06:30.36 | ryans0921 | nope.. nothing... im actually logged in remotely... is there something i have to do for it to show me everything? |
| 06:31.09 | ryans0921 | kiscokid: ya... with asterisk -vvvvr |
| 06:31.22 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 06:31.40 | kiscokid | I thought the v's made you see more |
| 06:31.57 | ryans0921 | i think i did like 5... should i do more?? haha |
| 06:32.07 | kiscokid | I think 5 is enough |
| 06:32.26 | kiscokid | do you ever see anything on the console when you make an outside call? |
| 06:32.43 | ryans0921 | hmm.. let me go try from the linksys.. one sec |
| 06:33.08 | CCFL_Man2 | anyone here use an adit 600 channelbank? |
| 06:33.16 | Teln1100A | jql: do you know how I could modify http://pastebin.com/m71d9cb4 .call file to conference 2 outgoing calls |
| 06:33.51 | ryans0921 | kiscokid: ya when i call from the linksys i get something |
| 06:34.18 | kiscokid | ryan: sounds like the Cisco phone is dial an extension that is not in the dial plan |
| 06:34.29 | kiscokid | *dialing |
| 06:35.41 | ryans0921 | hmmm.. when i call an extension number i get this... and then it does ring the extension... [Sep 6 23:34:10] WARNING[4458]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3 - No route to destination) |
| 06:35.46 | ryans0921 | does that make any sense? |
| 06:36.16 | CCFL_Man2 | ryans0921: i farking told you to check the dial plan and you ignored me |
| 06:36.20 | kiscokid | how is IAX involved? |
| 06:36.54 | ryans0921 | ccfl_man2: sorry about that... ya i did... what am i looking for though?? |
| 06:37.01 | klaus3000 | Hi! I have problems with the asterisk-dev mailing list. If I reply to a thread, the email gets to the list immediately. But if I send a new email (start a new thread), the email does not appear on the mailing list. Does somebody else have such problems with asterisk-dev? |
| 06:37.23 | kiscokid | ryan: the phone must be adding or subtracting digits |
| 06:37.44 | CCFL_Man2 | ryans0921: for the number being sent from the phone, is it the same as the one being sent from the linksys phone |
| 06:38.13 | citats | Teln1100A: if you set the 'channel' to one destination, and then set the 'application' to dial and the 'data' to the other destination those 2 will get bridged together |
| 06:40.07 | ryans0921 | ccfl_man2: bare with me.. haha im a little new at this... are you talking about the dialplan on the phone? |
| 06:40.17 | kiscokid | anyone gotten followme to work? |
| 06:40.27 | Teln1100A | hmm, that seems really simple |
| 06:40.33 | Teln1100A | now only if I could try it |
| 06:40.44 | Teln1100A | perhaps with a toll free number |
| 06:40.51 | Teln1100A | thanks citats |
| 06:41.17 | citats | Teln1100A: your destinations can be any type of device, so if you have 2 sip phones you can test a proof of concept and see that it works |
| 06:41.37 | *** join/#asterisk litage|w (n=nick@70.55.220.203.static.comindico.com.au) |
| 06:43.49 | Teln1100A | seems to work |
| 06:43.58 | Teln1100A | except that my dialplan is messed up and hangs up |
| 06:44.30 | citats | Teln1100A: if you use 2 sip devices how i mentioned it really has nothing to do with the dialplan |
| 06:44.48 | Teln1100A | I did it with 2 external numbers |
| 06:44.55 | Teln1100A | my cell phone and a toll free number |
| 06:45.08 | Teln1100A | as thats my goal |
| 06:45.36 | Teln1100A | my cell rang, then it dialed the toll free number and then hung up |
| 06:45.58 | citats | i'm not sure what type of technology your using (sip, iax, zap, etc) so i cant really say how it should work. but regardless of the tech it wont touch the dialplan unless you use context/extension/priority instead of application/data |
| 06:47.48 | Teln1100A | hers the log |
| 06:47.49 | Teln1100A | http://pastebin.com/m693340f8 |
| 06:48.23 | citats | Teln1100A: why are you using Local/CELLPHONE@phone1? |
| 06:48.38 | Teln1100A | CELLPHONE is my number |
| 06:48.45 | Teln1100A | which part are you refering to |
| 06:48.49 | citats | Teln1100A: the local part |
| 06:49.07 | Teln1100A | you mean it should just be cellphone/phone1 |
| 06:49.16 | Teln1100A | ? |
| 06:49.23 | citats | Teln1100A: it looks like your using iax, so just use that |
| 06:50.15 | Teln1100A | so iax/cellphone/phone1 ? |
| 06:50.29 | citats | Teln1100A: Channel: IAX2/USERNAME@teliax/CELLPHONE |
| 06:51.34 | citats | then, Application: Dial\n Data: IAX2/USERNAME@teliax/TOLLFREE |
| 06:52.14 | pkunkra | wish i could make my cisco ip phone talk gsm. |
| 06:52.25 | Teln1100A | k, let me try that |
| 06:53.22 | pkunkra | my mrtg graphs are fun |
| 06:53.52 | *** part/#asterisk cavediver (i=jonas@trimix.eklof.eu) |
| 06:54.10 | citats | pkunkra: IIRC cisco ip phones support g726 which is less bandwidth than g711. not as little as gsm, but much better |
| 06:57.16 | pkunkra | my model (7912G) supports ulaw, alaw and G.729a. |
| 06:57.33 | pkunkra | i don't want to pay 5k for the license. |
| 06:58.13 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:58.57 | citats | pkunkra: oh i guess not all of them support g726. for like 10 bucks per channel you can buy g729 for asterisk |
| 06:59.17 | pkunkra | 10 bucks? |
| 06:59.22 | pkunkra | that's not bad at all. |
| 06:59.37 | citats | pkunkra: thats how much it used to be a few years ago, cant imagine its changed much |
| 06:59.49 | pkunkra | there's a site i looked at that seemed to sell it for 5k. wasn't very specific. |
| 07:00.11 | pkunkra | i'd pay 20-30 bucks for a few channels. |
| 07:00.28 | citats | pkunkra: http://www.digium.com/en/products/voice/g729codec.php |
| 07:01.15 | pkunkra | nice! |
| 07:02.08 | pkunkra | Hmm.... "The G.729 key must be re-registered if certain hardware changes are made to your Asterisk server." |
| 07:02.11 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:02.21 | pkunkra | more hardware registration? |
| 07:02.31 | pkunkra | i've had enough of that with windows. |
| 07:03.37 | citats | pkunkra: there has to be some way to keep you from buying 1 set of licenses and using them on a hundred servers |
| 07:04.06 | pkunkra | oh. that's true. |
| 07:05.00 | Teln1100A | citats : heres what I get now |
| 07:05.01 | Teln1100A | http://pastebin.com/m2878dfd |
| 07:05.21 | Teln1100A | my phone rings, I pick up and in 4 seconds it hangs up |
| 07:05.45 | citats | Teln1100A: from your output it looks like its kicking back busy. IAX2/teliax-3 is circuit-busy |
| 07:06.48 | Teln1100A | can you tell me anything about the error : WARNING[11967]: pbx.c:5138 ast_pbx_outgoing_app: IAX2/teliax-2 already has a call record |
| 07:07.34 | citats | Teln1100A: if you want to get tricky, then you can use the local channel driver and use your dialplan to process it. for example, first play a sound file "Please wait while I connect you" then dial, and if it fails play another message |
| 07:07.43 | citats | Teln1100A: what error? thats a WARNING |
| 07:09.47 | Teln1100A | k |
| 07:10.12 | Teln1100A | is there any place that could be restricting it to 1 call per trunk? |
| 07:10.17 | Teln1100A | in config files |
| 07:10.45 | *** join/#asterisk tc3driver-nii (n=huh@rrcs-67-52-113-254.west.biz.rr.com) |
| 07:10.47 | citats | Teln1100A: since its making it out of your system its not your problem. its possibly that teliax limits you to one call |
| 07:11.07 | Teln1100A | shouldnt be as I can do 2 from xlite softphone and bridge |
| 07:11.35 | Teln1100A | its pay as you go so no restriction on number of channels |
| 07:11.52 | citats | Teln1100A: are you sure the number you are dialing is valid? |
| 07:12.55 | Teln1100A | I will try another number |
| 07:14.17 | *** join/#asterisk craigk (n=craigk@58.174.113.53) |
| 07:15.11 | Teln1100A | ok, works now |
| 07:15.16 | Teln1100A | thank you very much citats |
| 07:15.31 | Teln1100A | I have been trying to do this for over a week now |
| 07:17.21 | Teln1100A | how hard would it be to dial a number and allow that person to choose who they would like to call and bridge that way? |
| 07:17.31 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 07:19.00 | *** part/#asterisk kiscokid (n=xxx@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 07:19.23 | citats | Teln1100A: just send the call to the DISA application. though it will require you to setup the dialplan to do the outgoing calls. or you could find some generic calling card type script and use that |
| 07:20.55 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 07:22.20 | *** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 07:25.01 | Teln1100A | Say I wanted to use the call file again how would I setup the second part so it reaches the DISA |
| 07:25.44 | citats | Teln1100A: Application: DISA and Data: whatever variables you want to pass to it |
| 07:26.12 | Teln1100A | are there any that are required Data Variables? |
| 07:26.17 | *** join/#asterisk dikdust (n=dikdust@2001:1418:1c7:0:211:11ff:fed3:ecc7) |
| 07:27.28 | citats | Teln1100A: probably not. i think if you dont specify the context it goes to it defaults to one named 'disa'. if you look at the application info it should say |
| 07:27.58 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:28.20 | Teln1100A | WARNING[11278]: app_disa.c:135 disa_exec: DISA requires an argument (passcode/passcode file) |
| 07:28.58 | citats | Teln1100A: if you dont want a passcode you can use the phrase 'no-password' as the passcode |
| 07:29.43 | Teln1100A | application: disa/no-password |
| 07:29.44 | Teln1100A | like that? |
| 07:30.30 | citats | look how you used dial before and duplicate that |
| 07:30.45 | citats | since dial was the application |
| 07:31.24 | Teln1100A | application: disa , dada: no-password ? |
| 07:31.35 | Teln1100A | *data |
| 07:35.21 | Teln1100A | ok I got the dialtone, but I guess my dialplan needs tweaking |
| 07:36.28 | citats | Teln1100A: now you just need to make it so that context can dialout |
| 07:38.42 | Teln1100A | this would be in extensions.conf ? |
| 07:39.10 | Teln1100A | would the context be called disa ? |
| 07:39.22 | citats | Teln1100A: yes. i recommend you take a look at the asterisk book, the wiki, or something else online |
| 07:39.48 | citats | Teln1100A: type 'show application disa' and read that. it says what the default is, i believe it to be 'disa' but its been a while since i looked at it |
| 07:40.07 | citats | Teln1100A: you can also pass an argument to tell it to use whatever context you want to use |
| 07:40.18 | Teln1100A | ok |
| 07:40.48 | *** part/#asterisk craigk (n=craigk@58.174.113.53) |
| 07:43.16 | *** join/#asterisk Linex (n=kvirc@hqfirewall.cmsb.com.my) |
| 07:43.41 | *** join/#asterisk saftsack (n=oliver@p54A7EBD3.dip.t-dialin.net) |
| 07:43.46 | Linex | Is there anything I need to do to make a three way call or its just a client side thing ? |
| 07:45.43 | *** join/#asterisk watchy2 (n=watchy@h120.184.255.206.cable.cmdn.cablelynx.com) |
| 07:48.23 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:50.45 | *** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
| 08:03.49 | Teln1100A | Thank you once again for your help citats |
| 08:08.23 | *** join/#asterisk saftsack (n=oliver@p54A7EA4F.dip.t-dialin.net) |
| 08:09.40 | litage|w | is it possible to adjust rxgain and txgain in zapata.conf during a call? |
| 08:10.34 | WildPikachu | yep |
| 08:10.39 | WildPikachu | asterisk -rx reload |
| 08:10.45 | WildPikachu | change in the config file |
| 08:10.49 | WildPikachu | then reload |
| 08:10.56 | WildPikachu | i did it yesterday |
| 08:15.29 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:16.17 | litage|w | sweet, thanks |
| 08:16.32 | litage|w | is zaptel v1.4 compatible with asterisk v1.2 ? |
| 08:16.46 | litage|w | i'm guessing not, but though i'd ask just in case |
| 08:19.17 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 08:20.38 | *** join/#asterisk giasai68 (n=giasai@ip-240-130.sn2.eutelia.it) |
| 08:20.43 | giasai68 | hello |
| 08:21.15 | giasai68 | I have cofigured asterisk with linksys 3102 and tdm400p card |
| 08:21.42 | giasai68 | I have connected fax machine to linksys 3102 |
| 08:22.16 | giasai68 | I need send fax using fax machine trougth fxo module and PSTN line |
| 08:22.19 | creativx | wtf.. restarted asterisk and now suddenly iax2 throws "No authority found |
| 08:22.21 | giasai68 | any suggest? |
| 08:22.23 | creativx | and denies incoming calls. |
| 08:23.38 | giasai68 | I'm not interested about incoming call at moment |
| 08:24.40 | *** join/#asterisk Tili (n=tili@cm48.gamma244.maxonline.com.sg) |
| 08:28.21 | *** join/#asterisk bird_of_Luck (n=melifaro@secured.by.ipfw.ru) |
| 08:31.15 | *** join/#asterisk mattboll (n=mattboll@br137-1-82-228-156-113.fbx.proxad.net) |
| 08:31.38 | mattboll | Hi |
| 08:31.40 | *** join/#asterisk saftsack (n=oliver@p54A7EC51.dip.t-dialin.net) |
| 08:32.16 | mattboll | I've configured asterisk with destar but I can't call |
| 08:32.35 | mattboll | I think it's a problem with extensions.conf |
| 08:32.50 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 08:34.06 | mattboll | is normal that I don't have any",1,Dial(XXX)" ? |
| 08:35.06 | mattboll | actually what this should do : |
| 08:35.09 | mattboll | exten=1111,1,Set(CDR(pbx)=defaultpbx,CDR(userfield)=bmatthieu) ? |
| 08:40.08 | giasai68 | hello |
| 08:40.12 | giasai68 | I have cofigured asterisk with linksys 3102 and tdm400p card |
| 08:40.17 | giasai68 | I have connected fax machine to linksys 3102 |
| 08:40.20 | giasai68 | I need send fax using fax machine trougth fxo module and PSTN line |
| 08:40.23 | giasai68 | any suggest? |
| 08:40.29 | giasai68 | I'm not interested about incoming call at moment |
| 08:40.35 | giasai68 | thank you... :) |
| 08:44.57 | mattboll | <PROTECTED> |
| 08:45.09 | mattboll | does anyone know sth about that ? |
| 08:46.33 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 08:57.03 | *** join/#asterisk Hrox (n=support@84.12.108.34) |
| 08:57.56 | *** join/#asterisk DarkFlib (n=mike@host90-152-23-30.ipv4.regusnet.com) |
| 08:58.16 | Hrox | hi people, after someone i can talk twould anyone be able to help?o about support for asterisk now. |
| 09:03.08 | DarkFlib | what do you need? most asterisk people are stateside... it should start waking up about midday .eu time... |
| 09:04.10 | Hrox | :D i have installed asterix and apears to be working ok but on the asterix now i get the console but no ip i can interface with. i dont think its loading the network card drivers. only thing is im a windows guru not a linux person |
| 09:04.34 | Hrox | the network card is Broadcom Netxtream gigabit |
| 09:04.54 | Hrox | any ideas |
| 09:07.57 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 09:09.27 | creativx | Hrox: eth0 up |
| 09:10.21 | Hrox | sorry you want me to put that in to the console? |
| 09:11.31 | Linex | Is there anything I need to do to make a three way call or its just a client side thing ? |
| 09:12.30 | creativx | no Hrox.. have you configured eth0 ?? |
| 09:12.47 | creativx | what does ifconfig give you ? not in the asterisk console |
| 09:12.48 | Hrox | erm... no but when i created a virtual pc it already gave me a ip |
| 09:12.48 | puzzled | hi |
| 09:13.51 | Hrox | you will need to bear with me on this. what i have done it downloaded asterisknow and burned the image to cd. i have then booted from the cd and installed the application. im unsure how i would find access to the linux console. |
| 09:14.10 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:22.32 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 09:31.02 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-230-95.dsl.irvnca.pacbell.net) |
| 09:34.54 | *** join/#asterisk casix (n=casix@edifici-pub.adam.es) |
| 09:34.57 | casix | hello |
| 09:35.47 | casix | how can I call to sip user but using the sip uri in the dial cmd? |
| 09:36.09 | casix | like dial(SIP:user:pass@server/exten) |
| 09:41.22 | penguinFunk | you mean like: Dial(SIP/${EXTEN}@192.168.1.100) ? |
| 09:44.09 | casix | but I don't have the peer into the sip.conf and I want put the user and the pass |
| 09:44.09 | *** join/#asterisk zeeesh (i=zeeesh@202.166.161.45) |
| 09:44.40 | penguinFunk | why haven't you put the peer into sip.conf? |
| 09:44.47 | penguinFunk | that's what your suppose to do |
| 09:45.21 | casix | yes but I have to do it without using the sip.conf |
| 09:45.24 | casix | it is possible?? |
| 09:45.29 | penguinFunk | howcome? |
| 09:45.40 | casix | can I put the user and pass in the dial cmd? |
| 09:46.42 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 09:46.43 | *** join/#asterisk SA007 (n=sa007@ip565e0006.direct-adsl.nl) |
| 09:47.08 | penguinFunk | how does your sip user register itself without an entry in sip.conf? |
| 09:48.15 | casix | ok I've my users in sip.conf |
| 09:49.14 | casix | if my user is 111 I want to call to another asterisk with username 44111 |
| 09:49.20 | casix | and if my user is 112 |
| 09:49.36 | casix | I want to call to another asterisk with username 44112 |
| 09:50.32 | casix | and I would like to do it like Dial(SIP/${EXTEN}@44${username}:pass@newasterisk) |
| 09:51.51 | DarkFlib | why not omit the pass and just use an ip based match on the far end? |
| 09:52.42 | casix | maybe I can do it but I don't know if the ip is static o dynamic... |
| 09:52.55 | casix | but without pass how can I construct the uri? |
| 09:54.41 | *** join/#asterisk yannj_fr (n=yannj_fr@chilli.esiee.fr) |
| 09:55.33 | *** join/#asterisk knarfly (n=knarfly@c-98-203-55-196.hsd1.fl.comcast.net) |
| 09:59.19 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-230-95.dsl.irvnca.pacbell.net) |
| 10:03.55 | *** join/#asterisk ming_zym (n=ming_zym@124.254.55.140) |
| 10:15.57 | mattboll | does anyone know destar ? |
| 10:20.29 | *** join/#asterisk syn (i=syn@kenobi.sceen.net) |
| 10:20.34 | syn | hello |
| 10:21.40 | syn | I have a series of patch for a module in the trunk. I was wondering about the best way to submit them, taking into account the type of changes and the order of applying |
| 10:36.34 | *** join/#asterisk hrox (n=support@84.12.108.34) |
| 10:37.16 | *** join/#asterisk hrox (n=support@84.12.108.34) |
| 10:41.46 | *** join/#asterisk cheGGo (n=snafu__@gate.goobernetworks.com) |
| 10:41.49 | cheGGo | hi there |
| 10:43.07 | syn | hello |
| 10:43.13 | awk | grrr |
| 10:43.21 | awk | in asterisk how can I see how many channels im allowed for g729 |
| 10:43.21 | awk | ? |
| 10:43.56 | awk | there was a cli command to see how many channels u have |
| 10:47.54 | cheGGo | awk, show channels? |
| 10:48.43 | snuff-work | awk: show g729 |
| 10:50.37 | cheGGo | hm, anybody has worked with an outboundproxy for sip as NON peer? |
| 10:50.44 | awk | show g**** doesn't go anything |
| 10:52.13 | *** join/#asterisk geoff_k (n=geoff_k_@host81-152-90-217.range81-152.btcentralplus.com) |
| 10:54.49 | penguinFunk | bloody digium downloads only giving me 1.9KB/sec :( |
| 10:54.54 | penguinFunk | stingy bastards |
| 10:55.56 | awk | u have to pay for faster d/l from them |
| 10:56.30 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 10:57.17 | penguinFunk | pfff |
| 10:57.29 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 10:58.10 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 10:58.27 | penguinFunk | we pay enough for digium hardware and g729 licenses. trying to charge us more for a better download rate to get the g729 codec is scandalous |
| 11:00.23 | awk | grr, somebody from digium |
| 11:00.24 | awk | tell me please |
| 11:00.39 | awk | how do I check how many channels I have for my g29 codec i bought |
| 11:00.42 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:03.11 | penguinFunk | show g729 |
| 11:03.17 | penguinFunk | if that doesnt work you have not installed them properly |
| 11:03.44 | penguinFunk | http://ftp.digium.com/pub/telephony/codec_g729/README |
| 11:22.17 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 11:22.23 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128666903.dsl.bell.ca) |
| 11:25.14 | *** join/#asterisk Juggie (n=Juggie@CPE00c0f02a7bf2-CM001371886eee.cpe.net.cable.rogers.com) |
| 11:29.59 | *** join/#asterisk blinky42 (n=me@67.106.175.130.ptr.us.xo.net) |
| 11:35.03 | juuva | how can I unpause queue member? |
| 11:37.50 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 11:45.00 | *** join/#asterisk ManxPower (n=manxpowe@247.sub-75-202-158.myvzw.com) |
| 11:45.28 | *** join/#asterisk coppice (n=chatzill@140.196.17.210.dyn.pacific.net.hk) |
| 11:47.21 | *** join/#asterisk yannj_fr (n=yannj_fr@chilli.esiee.fr) |
| 11:49.05 | Kwakwa | dunno, remove and then readd the member? |
| 11:50.01 | Renacor | anybody here use grandstream analog converters? |
| 11:51.25 | hrox | hey all. appologies to those who replied earlier was waiting for us to come online. I have downloaded Asterisk Now and burned image to cd and then booted + followed install instructions. im at the console and it says point your browser to http:/// <- no ip address, i dont think its loaded the network card and ideas as to how i can install the drivers. im a windows guru not linux so pls bear with me. i not sure on what version of linux the asterix now is |
| 11:51.38 | hrox | anyone got any ideas |
| 11:53.10 | *** join/#asterisk DaveCanoe (n=Dave@74.93.16.81) |
| 11:54.12 | Sci_05 | Renacor: what do you mean analog converters? |
| 11:54.17 | Sci_05 | ~gs |
| 11:54.18 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 11:54.41 | Renacor | Sci_05: SIP -> Analog phone line converter |
| 11:55.07 | Sci_05 | you mean an ATA right? |
| 11:57.10 | Renacor | yeah |
| 11:57.22 | Renacor | http://www.voip-info.org/wiki/index.php?page=HT-286 |
| 11:57.25 | Renacor | that one specifically |
| 11:57.29 | Sci_05 | what about it? |
| 11:58.04 | Renacor | entered all the info for it and it's not registering itself |
| 12:04.54 | Kwakwa | Which version of asterisk do people find most stable? I'm running 1.4.6 atm and want to upgrade to a newer one |
| 12:11.13 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:13.00 | jwh | i'm running 1.4.11, it periodicly segfaults though :( |
| 12:13.02 | jwh | for me anyway |
| 12:13.10 | jwh | 1.2.24 was fine |
| 12:16.13 | *** join/#asterisk hellop (n=hellop@cpe-66-91-197-100.hawaii.res.rr.com) |
| 12:21.58 | bkruse_home | jwh: compile with debug flags, get a core dump, and submit it to the bug tracker and get it fixed |
| 12:22.07 | bkruse_home | its probably just 1 issue that will never get taken care of unless you help :] |
| 12:23.34 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-4e9f63fcbc4ab216) |
| 12:24.21 | jwh | hehe, I have been meaning to |
| 12:24.44 | jwh | its kinda a production install as well, which isn't good, fallen back to 1.2 for now |
| 12:26.15 | ManxPower | bkruse_home: the issue will never be taken care of even if he does that. He said he is running a production system. He can't test fixes and he can't take the system down for a few weeks for testing. |
| 12:27.18 | *** join/#asterisk mog (i=mog@nat/digium/x-94523b1a5fccc423) |
| 12:27.18 | *** mode/#asterisk [+o mog] by ChanServ |
| 12:28.44 | bkruse_home | mog: !! |
| 12:28.54 | ManxPower | "The last time I visited your pit bull ripped my stomach out." "Don't worry, I trained him not to do that anymore." "I think I'll stay home anyway." |
| 12:28.55 | bkruse_home | ManxPower: It would get fixed, especially if he can easily recreate the issue |
| 12:29.08 | bkruse_home | you can often do this very easily by just watching the call flow |
| 12:29.20 | ManxPower | jwh: can you make 1.4.11 segfault on demand? |
| 12:29.46 | bkruse_home | ManxPower: Watch your logs/errors/warnings and you can see what its coming from.... |
| 12:29.49 | bkruse_home | mog: !! |
| 12:29.53 | mog | bkruse why are you up |
| 12:29.58 | mog | its f-in early |
| 12:30.35 | *** join/#asterisk akirch_ (n=akirch@umcc-pdc.ultimatemedium.com) |
| 12:30.53 | ManxPower | bkruse_home: It has been a long time since I've been able to make Asterisk segfault on demand. |
| 12:31.07 | ManxPower | It has not been a long time since I've had Asterisk segfault. |
| 12:31.39 | ManxPower | Digium needs to accept that there is a culture of fear regarding upgrading Asterisk. |
| 12:32.14 | ManxPower | Too many people have been burned by it. |
| 12:32.20 | *** part/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 12:33.57 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 12:34.04 | Renacor | Got SIP response 486 "Busy" back from 10.1.142.7 <--- I keep getting that from my grandstream ATA |
| 12:34.05 | lirakis | morning |
| 12:35.00 | *** join/#asterisk s0ck (n=m@unaffiliated/s0ck) |
| 12:35.07 | lirakis | is it off hook? .. or properly registered? |
| 12:35.29 | s0ck | im getting "all lines are busy" |
| 12:35.40 | s0ck | phones direct into the voip line, works fine :/ |
| 12:35.42 | s0ck | any clues? |
| 12:36.00 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkgroup.com.au) |
| 12:36.21 | lirakis | s0ck: .. you have not given any where near a clear enough picture to answer your question |
| 12:36.42 | s0ck | ok, what do you need to know? |
| 12:37.00 | Renacor | lirakis: yep it's registered, can make outgoing calls fine, not off the hook |
| 12:37.18 | jwh | ManxPower: no |
| 12:37.27 | jwh | I haven't actually been able to catch it erroring yet |
| 12:37.34 | lirakis | s0ck: .. seriously? ... look at your question.. "i get all lines busy .. why?" .. you get busy dialing in... dialing out.. from a sip phone ...etc... you have given us ZERO info to help you out |
| 12:38.36 | lirakis | Renacor: .. so .. its working when you pick up the phone its attached to.. but when you try to dial into * and route a call it gives you 486-busy |
| 12:38.41 | s0ck | im using Asterisk 1.2.19, using some SNOM voip phones, attempting to dial out via a SIP trunk and recieving the error above. The lines are shown as registered. |
| 12:38.49 | *** part/#asterisk SA007 (n=sa007@ip565e0006.direct-adsl.nl) |
| 12:39.16 | lirakis | s0ck: your sip "trunk" ... wich is just a sip peers.. its not a trunk... does not have any channels available for you |
| 12:39.25 | lirakis | s0ck: .. or it is not properly set up |
| 12:39.25 | Renacor | lirakis: no when I try to call it I get a 486 busy |
| 12:39.53 | lirakis | Renacor: right.. from another ext.. you try to dial(sip/1234) where 1234 is the ext of the ata |
| 12:40.03 | Renacor | yeah |
| 12:40.06 | Renacor | like sip/fax |
| 12:40.12 | s0ck | hey, sorry about the lack of information, i am completely new to this setup and my colleage who normally deals with this, is off this week and we are having problems |
| 12:40.26 | s0ck | could you help me or suggest somewhere to look? |
| 12:40.35 | s0ck | sometimes we can dial out, some other times we can not. |
| 12:41.04 | lirakis | s0ck: it is like when you try and dial from your cell ... or land line and you get "all circuits are busy.. please try again later" |
| 12:41.36 | s0ck | its while dialing a landline outside of the asterisk system, from within the asterisk system |
| 12:41.43 | penguinFunk | zap show channels |
| 12:41.46 | lirakis | s0ck: you either dont have enough channels available (I.e. you can only make 1 simultaneous call) .. or you have a crappy sip provider that does not have enough availability |
| 12:41.48 | penguinFunk | check how many are in use |
| 12:42.06 | lirakis | s0ck: .. so .. you arent dialing out a sip peer... |
| 12:43.16 | s0ck | what does dialing out a sip peer mean? |
| 12:43.18 | lirakis | Renacor: sip/fax ? |
| 12:43.22 | lirakis | Renacor: are you faxing? |
| 12:43.28 | Renacor | nod |
| 12:43.33 | Renacor | eventually |
| 12:43.39 | s0ck | i have set up a trunk/route with the sip details from our provider |
| 12:43.41 | Renacor | i know i know |
| 12:43.52 | lirakis | s0ck: sending a call to a sip provider.. or other sip entity registered on your system |
| 12:44.25 | lirakis | s0ck: but you just said that this happens when you dial out the tdm card you have hooked to a land line |
| 12:44.31 | lirakis | s0ck: right.??? |
| 12:44.54 | lirakis | Renacor: .. so is the ata hooked to a phone or a fax machine now |
| 12:45.04 | Renacor | lirakis: a phone right now to test |
| 12:45.21 | [TK]D-Fender | s0ck: Pastebin a failed call at verbose 10 and SIP debug enabled. |
| 12:46.40 | [TK]D-Fender | ~pb |
| 12:46.41 | jbot | A Pastebin is a place to paste your stuff 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 |
| 12:46.42 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 12:46.53 | *** join/#asterisk ShakaGoldSaint (n=eleazar@190.38.75.1) |
| 12:49.35 | Renacor | anybody know the onhook voltage for german analog phones? |
| 12:50.58 | penguinFunk | 48v DC |
| 12:51.42 | Renacor | are u sure? |
| 12:51.45 | cheGGo | nope |
| 12:51.50 | Renacor | i think i just burned out a phone with 36volts |
| 12:51.52 | cheGGo | between 60-40v |
| 12:51.58 | cheGGo | during ringing |
| 12:52.18 | cheGGo | on a phone call it will be reduced to 10-12v |
| 12:52.50 | *** join/#asterisk minkus (n=minkus@pool-72-84-53-31.clrkwv.east.verizon.net) |
| 12:54.39 | Renacor | so onhook voltage i got 18-24-36-48-51 |
| 12:54.56 | Renacor | so 48 or 51 would be good? |
| 12:56.17 | s0ck | s0ck: but you just said that this happens when you dial out the tdm card you have hooked to a land line |
| 12:56.44 | *** join/#asterisk Shido6 (n=shido6@204.126.120.132) |
| 12:56.54 | s0ck | we have a bb router with voip cabalities, has accounts registered to the broadband line |
| 12:57.27 | s0ck | im unsure 100% what we have |
| 13:01.43 | *** join/#asterisk guillote_GNU (n=bancaria@host32.190-136-186.telecom.net.ar) |
| 13:04.46 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:05.25 | *** join/#asterisk Daejeo1 (n=chatzill@211.177.189.25) |
| 13:09.52 | Daejeo1 | any have sip firmware for CP 7961G? |
| 13:09.54 | tzanger | coppice: omg you're here |
| 13:09.59 | tzanger | question for you regarding sliptest.c |
| 13:10.15 | tzanger | I've looped back a digital interface driver I'm working on |
| 13:10.22 | tzanger | basically zaptel is seeing it as tdmoe |
| 13:10.35 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:10.37 | tzanger | so ethernet in, elastic buffers, looped back, ethernet back out |
| 13:11.10 | tzanger | ssliptest is reporting a very consistent 512, which would make sense to me, given the buffer lengths and whatnot, but does that sound correct to you? |
| 13:11.25 | tzanger | not so much the number, but that I should be able to loopback a digital interface and sliptest still work correctly |
| 13:12.10 | coppice | that rather depends on the buffers you have in your driver, and the latency of the channel |
| 13:12.41 | tzanger | coppice: right, that is understood |
| 13:20.31 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:28.55 | syn | anyone using res_config_sqlite here ? |
| 13:29.41 | syn | or any developer able to tell me if it's a good thing that there are both cdr_sqlite and res_config_sqlite that provide CDR support with SQLite 2 ? |
| 13:33.24 | *** join/#asterisk etfonhomey (n=chatzill@mail.advancmed.org) |
| 13:34.02 | *** part/#asterisk syn (i=syn@kenobi.sceen.net) |
| 13:38.54 | *** join/#asterisk Ebola (n=Ebola@host86-143-7-120.range86-143.btcentralplus.com) |
| 13:39.34 | *** join/#asterisk pks (n=pks@elit.pl) |
| 13:39.59 | pks | hi |
| 13:40.35 | pks | i've setup asterisk, so that i can loggin with softphon and i can even call the voice mail (sip:8500) |
| 13:40.47 | pks | but when i try to connect to ther user it fails |
| 13:40.50 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 13:41.30 | pks | do i need to install something in addition to make phonecalls? |
| 13:42.16 | javar | www.asteriskdocs.org is down, someone know something? |
| 13:43.56 | etfonhomey | I know that it was down a few days ago as well. |
| 13:44.13 | *** join/#asterisk michael-i (n=michael-@141.41.40.55) |
| 13:44.58 | javar | yeah |
| 13:46.48 | cheGGo | hm |
| 13:46.49 | cheGGo | guys |
| 13:46.57 | cheGGo | anybody has worked with an outboundproxy for sip as NON peer? |
| 13:47.17 | cheGGo | for direct calls through an SIP URI |
| 13:47.44 | cheGGo | the options of sip.conf 'outboundproxy=proxy.domain.tld' didnt work |
| 13:51.00 | cheGGo | *sigh* |
| 13:51.15 | cheGGo | it not real that nobody used this function before :D |
| 13:51.18 | cheGGo | +s |
| 13:52.21 | etfonhomey | When provisioning a PRI, is there anything in particular I need to look out for in order to make sure DID's will work correctly with *? |
| 13:53.42 | Strom_M | etfonhomey: you just need to know how the telco is delivering DNIS to you |
| 13:53.43 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 13:55.24 | [TK]D-Fender | etfonhomey: I would highly recommend demanding 10-digit DNIS |
| 13:55.25 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 13:57.01 | etfonhomey | Just to show my newbie-ness. What does DNIS stand for? |
| 13:57.04 | Strom_M | etfonhomey: and if you ask your telco about it, and they say something like "what's genus?" then I would find a new telco fast. |
| 13:57.18 | Strom_M | etfonhomey: Dialed Number Identification Service |
| 13:57.33 | etfonhomey | Thanks |
| 13:57.35 | *** join/#asterisk tsurko (n=tsurko@213.91.216.130) |
| 13:57.50 | Strom_M | pronounced "DEE-nis" |
| 14:00.38 | Shido6 | geenius |
| 14:01.40 | *** join/#asterisk jfitzgibbon (n=chatzill@64.72.237.130) |
| 14:09.48 | *** join/#asterisk CM_3_1_2_634 (n=CM_3_1_2@cm222-166-6-33.hkcable.com.hk) |
| 14:12.30 | *** join/#asterisk nephfl (n=none@wsip-68-110-130-57.ga.at.cox.net) |
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| 14:15.21 | *** join/#asterisk coolbeans (n=null@adsl-070-148-053-190.sip.bhm.bellsouth.net) |
| 14:15.55 | coolbeans | Hi all. We just got a bunch of Cisco 7940's and they are working fine with 8.2 SIP firmware... My question is, do we have to provision multiple "lines" to get the adhoc conferecing to work? |
| 14:17.33 | *** part/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 14:19.24 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 14:20.23 | *** join/#asterisk limbeaux (n=IceChat7@69.17.126.244) |
| 14:23.00 | Daejeo1 | any have sip firmware for CP 7961G? i do not have a smartnet account |
| 14:23.22 | Qwell | get a smartnet account |
| 14:23.26 | Qwell | That's the only way you can get it |
| 14:23.28 | Nugget | Yes, Cisco has a SIP firmware for the 79x1 phones. |
| 14:24.00 | Daejeo1 | Qwell: 60$ |
| 14:24.13 | Daejeo1 | trouble |
| 14:24.15 | Qwell | plus $100 for the license to use it |
| 14:24.16 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:24.28 | Daejeo1 | more trouble |
| 14:24.41 | Qwell | next time, buy polycom |
| 14:24.47 | Daejeo1 | yes sir |
| 14:24.58 | limbeaux | what should be the proper settings so that when i call someone with ivr, their ivr will recognize my selections? |
| 14:26.13 | coolbeans | Hi all. We just got a bunch of Cisco 7940's and they are working fine with 8.2 SIP firmware... My question is, do we have to provision multiple "lines" to get the conferecing softkey to work? |
| 14:26.33 | limbeaux | i use linksys SPA962 phones |
| 14:28.00 | Shido6 | hehe |
| 14:28.22 | Shido6 | you should be able to handle 2 calls on a single line |
| 14:28.48 | Zeeek | who uses ENUM? |
| 14:29.15 | Nugget | I have ENUM configured but I don't know if I'd say I "use" it. |
| 14:29.43 | Zeeek | I had it configged but there was a change since then, the adding of a '+' in front |
| 14:30.02 | Zeeek | seems to still work though once that change was made :) |
| 14:30.16 | *** join/#asterisk ManxPower (n=manxpowe@71-8-60-51.dhcp.leds.al.charter.com) |
| 14:33.09 | *** join/#asterisk MdeP (n=mdep@112-86-22-190.adsl.tie.cl) |
| 14:34.25 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 14:34.47 | *** join/#asterisk RsaMan (n=aa@196.210.154.3) |
| 14:35.20 | RsaMan | hi |
| 14:35.57 | RsaMan | I want to connect 2 asterisk servers via iax |
| 14:36.11 | RsaMan | by using this i have a question : register => REC_SERVER:mysecret@my.receiving.server.ca |
| 14:36.24 | RsaMan | erm this : register => REC_SERVER:mysecret@my.receiving.server.ca |
| 14:36.34 | RsaMan | with rec server being my remote server |
| 14:36.42 | RsaMan | and that line being on my client server |
| 14:36.55 | RsaMan | will i be able to dial iax2/rec_server/1002? |
| 14:38.03 | Strom_M | RsaMan: registration is only used if one server is on a dynamic IP |
| 14:38.26 | Strom_M | if both servers are on static IPs, then you don't need registration lines |
| 14:46.24 | *** join/#asterisk km- (n=pgrace@aeneas.fierymoon.com) |
| 14:48.44 | *** join/#asterisk eldon (i=eldon@nat/digium/x-7d0ba23c272caaba) |
| 14:49.12 | eldon | is there any way to check the kernel stack size of a running kernel? the best I got is the kernel config which may or may not refer to the currently running kernel... |
| 14:49.46 | eldon | oops, wrong channel... I think... |
| 14:53.48 | *** join/#asterisk slakware (n=attila@201.19.152.12) |
| 14:55.05 | slakware | I'me trying to setup Asterisk realtime with an odbc connection. I've got unixODBC and unixODBC-devel installed via rpm. When I configure and make Asterisk the res_config_odbc and res_odbc .so modules are not compiled... I'm running Asterisk 1.4.11. Any ideas? |
| 14:56.28 | slakware | I've also tried configure with the --wisth-odbc=/usr option as unixODBC has been installed into /usr/lib and /usr/include |
| 15:03.54 | km- | anyone here ever have problems getting sip to work right through ASA's? |
| 15:04.12 | ManxPower | ASA? |
| 15:04.28 | *** join/#asterisk elyograg (n=elyograg@saidar.elyograg.org) |
| 15:04.59 | *** part/#asterisk elyograg (n=elyograg@saidar.elyograg.org) |
| 15:05.40 | *** join/#asterisk phillipk (n=pkey@216.248.143.87) |
| 15:08.19 | Zeeek | ManxPower! |
| 15:08.44 | km- | cisco ASA |
| 15:08.51 | km- | adaptive security appliance |
| 15:09.08 | km- | it's a pretty crazy cisco pix with application gateway filtering and trend micro stuff built into it |
| 15:09.17 | km- | I have a sneaking suspicion it's mangling my sip packets for some hosts |
| 15:09.43 | km- | I really just wanna drop a damn line to the asterisk box avoiding the firewall altogether |
| 15:09.51 | km- | but I'm pretty darn sure I'll get a ton of resistance to that |
| 15:10.00 | ManxPower | Cisco has a history of messing up SIP. Turn off sip fixup |
| 15:10.15 | km- | do you know if fixup is on by default? |
| 15:10.41 | km- | I checked the config and there's no fixup lines at all |
| 15:10.53 | km- | trying to get console on it to fiddle with the fixup line |
| 15:11.16 | ManxPower | km-: On IOS it is something like no nat fixup sip or something like that |
| 15:11.31 | km- | no fixup protocol sip |
| 15:11.34 | km- | I believe |
| 15:11.54 | km- | the asa is being a prick, I have 15 on the router but it wont let me get on the console, it just wants to insist on ASDM |
| 15:13.52 | *** join/#asterisk yannj_fr (n=yannj_fr@ALagny-152-1-45-179.w83-112.abo.wanadoo.fr) |
| 15:16.47 | *** join/#asterisk t3rror (n=t3rror@gateway.sscgp.com) |
| 15:28.31 | *** join/#asterisk Ebola (n=Ebola@host86-143-7-120.range86-143.btcentralplus.com) |
| 15:30.13 | *** join/#asterisk Dossy_ (i=nobody@aolserver/dossy) |
| 15:32.05 | *** join/#asterisk Tech3030 (n=j@207.210.204.55) |
| 15:33.28 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 15:33.58 | *** join/#asterisk Dossy_ (i=nobody@aolserver/dossy) |
| 15:34.44 | _x86_ | hmm |
| 15:35.13 | _x86_ | i dont want to use music on hold anymore... so in modules.conf, i put in noload => res_musiconhold.so |
| 15:35.37 | km- | yup, it was the damn ASA's fixup rules |
| 15:35.39 | _x86_ | but, i want musiconhold to be disabled immediately, and I have 7 production calls active |
| 15:36.00 | km- | so add that to your databanks, the cisco ASA, despite all it's glory, still mangles the hell out of SIP packets. Turn it off asap. |
| 15:36.09 | _x86_ | so from the CLI, I did unload res_musiconhold.so, and it told me Unable to unload resource res_musiconhold.so |
| 15:36.27 | _x86_ | km-: has that changed since the PIX? |
| 15:36.42 | _x86_ | km-: i used to run a production asterisk server behind a PIX 501 just fine |
| 15:36.57 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 15:38.04 | wunderkin | _x86_: remove the classes from musiconhold.conf and reload res_musiconhold.so |
| 15:38.30 | _x86_ | ah righto |
| 15:39.00 | _x86_ | great idea :) |
| 15:39.39 | Dossy_ | can asterisk receive faxes yet? :) |
| 15:41.01 | ManxPower | Dossy_: It does so every single day in my systems |
| 15:41.35 | ManxPower | km-: you can send your thankyou via paypal to eric@fnords.org |
| 15:42.02 | Zeeek | funny, I just emptyed that account |
| 15:42.19 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 15:42.29 | Zeeek | ManxPower going to Astricon? Speaking there? |
| 15:42.59 | Dossy_ | ManxPower: software fax? Tell me more. |
| 15:42.59 | ManxPower | Zeeek: no. |
| 15:43.13 | Dossy_ | I don't want to use a SIP ATA w/ T.38 to a real fax machine ... |
| 15:43.36 | ManxPower | Dossy_: rxfax and spandsp of course. The resulting tiff file is passed to a script I wrote to convert the file to PDF and e-mail it to where it needs to go. |
| 15:43.51 | Dossy_ | spandsp? |
| 15:43.56 | ManxPower | of course, I have real telco lines |
| 15:44.04 | ManxPower | Dossy_: google is your friend. |
| 15:44.09 | Dossy_ | yeah, googling now |
| 15:44.13 | Dossy_ | is the quality reasonably good? |
| 15:44.31 | ManxPower | same as a hardware fax. |
| 15:44.40 | ManxPower | we had some issues with older versions os spandsp |
| 15:49.49 | _x86_ | Dossy_: a lot of people generally prefer iaxmodem + hylafax over spandsp |
| 15:50.23 | ManxPower | _x86_: does that work on PRIs? |
| 15:50.39 | [TK]D-Fender | ManxPower: Any channel type actually |
| 15:50.50 | Dossy_ | yeah, I like hylafax |
| 15:50.53 | Dossy_ | let me look at iaxmodem :) |
| 15:51.15 | Tech3030 | NEED to install Asterisk on a Dell Vostro 200. Promblem is it only came with USB ports. Don't want to install Astrisk ontop of linux. Need the USB drivers though. Thoughts?? |
| 15:52.08 | etfonhomey | "Don't want to install Astrisk ontop of linux"??? |
| 15:52.31 | Tech3030 | just want to install Asterisk now |
| 15:52.41 | Damin | Anyone check this out? |
| 15:52.52 | Damin | http://digg.com/linux_unix/Bradley_Kuhn_and_Max_Spevak_to_speak_at_Ohio_Linux_Fest |
| 15:53.12 | Damin | Digium's sponsoring the bags.. and there will be an Asterisk users meetup there.. |
| 15:53.52 | etfonhomey | Where in Ohio? |
| 15:55.18 | ManxPower | Tech3030: We have no idea what you are talking about.l |
| 15:55.40 | ManxPower | Do you, perhaps, mean AsteriskNOW and were too lazy to spell it correctly? |
| 15:55.55 | ManxPower | If that is the case, then as the /topic says, TRY #ASTERISKNOW! |
| 15:56.35 | etfonhomey | And doesn't AsteriskNOW install a flavor of Linux? |
| 15:56.45 | Dossy_ | oh yuck, iaxmodem needs hardware? ugh |
| 15:56.53 | Tech3030 | appologies |
| 15:57.49 | *** join/#asterisk mrbond82 (n=bondo@d221-91-164.commercial.cgocable.net) |
| 15:58.21 | mrbond82 | Is there any easy way to add dialing rules to change the caller ID based on whether you are dialing from extension a or extension b ? |
| 15:58.39 | Dossy_ | it looks like all the fax/asterisk solutions require hardware above and beyond just a PC. sigh--oh well |
| 15:58.54 | ManxPower | Dossy_: Well, except for the one I use. |
| 15:59.26 | Dossy_ | ManxPower: doesn't spandsp require a card? |
| 15:59.38 | ManxPower | No. |
| 15:59.44 | ManxPower | What ever gave you that idea. |
| 15:59.49 | Dossy_ | http://www.voip-info.org/wiki-Asterisk+fax |
| 15:59.56 | ManxPower | Results 1 - 10 of about 118,000 English pages for spandsp AND "too lazy to look it up on google". (0.15 seconds)Â |
| 16:00.11 | Dossy_ | oh, asterisk will "simulate a fax machine" |
| 16:00.18 | ManxPower | Dossy_: the wiki is the LAST place to look for information. It is full of misinformation |
| 16:00.22 | Dossy_ | woah?! |
| 16:00.37 | Dossy_ | that's ironic. typically a wiki is the /best/ place to go as should be the most up to date |
| 16:00.58 | Zeeek | TMI: never up to date |
| 16:01.01 | Dossy_ | that's pretty bad since the wiki is usually one of the first results for "asterisk" and anything |
| 16:01.05 | ManxPower | the first places are "show applications" and "show application X" in the Asterisk CLI, then /path/to/src/asterisk/docs, then the mailing list archives, then the Wiki. |
| 16:01.11 | ManxPower | Dossy_: no it isn't. |
| 16:01.25 | ManxPower | it is full of wrong information, out dated information, and just plain wrong information. |
| 16:01.29 | Dossy_ | http://www.google.com/search?q=asterisk%20spandsp |
| 16:01.34 | *** join/#asterisk ming_zym (n=ming_zym@124.254.55.140) |
| 16:01.35 | Dossy_ | yields the wiki as the first two links. |
| 16:01.40 | ManxPower | It is a decent place to look for examples of stuff you already know about. |
| 16:02.42 | ManxPower | Is there a way in the dialplan to change the verbosity level? |
| 16:03.41 | Dossy_ | spandsp-0.0.4pre4 or spandsp-0.0.3? (wow, is this really usable?) |
| 16:04.16 | coppice | no. leave it alone |
| 16:04.49 | ManxPower | coppice: Dossy_ is horrible gullible, don't tease him/her |
| 16:04.52 | Zeeek | Dossy_ a good start might be fax site:lists.digium.com |
| 16:05.04 | ManxPower | coppice: he thought the wiki was an accurate source of info. HAHAHAHAHA! |
| 16:05.05 | Zeeek | errr |
| 16:05.24 | Zeeek | Google fax site:lists.digium.com |
| 16:05.27 | coppice | leave iaxmodem alone too. that's just spandsp with spandsp crossed out, and iaxmodem written in crayon |
| 16:07.17 | Dossy_ | oh, does spandsp still have problems receiving from HP faxes? (reading list archives from 2005) |
| 16:07.40 | [TK]D-Fender | Dossy_: No, thats been corrected. It now has problems receiving for all faxes equally... |
| 16:07.58 | coppice | of course. nothing ever gets fixed. use the oldest version you can possibly find, and it will be the best |
| 16:08.16 | ManxPower | Dossy_: No. |
| 16:08.26 | ManxPower | I have not found a single issue with the recent spandsps |
| 16:08.31 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 16:08.32 | Dossy_ | http://soft-switch.org/spandsp-known-issues.html ... doesn't list anything for 0.0.3 or 0.0.4pre releases |
| 16:08.54 | *** join/#asterisk CVirus (n=GoD@196.205.191.19) |
| 16:09.52 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 16:10.59 | coppice | then it is surely perfect |
| 16:11.34 | CVirus | I need to integrate 10 phone lines ... What's the best FXO choice for that number ? |
| 16:12.41 | [TK]D-Fender | CVirus: What kind of case? |
| 16:14.05 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583591.dsl.bell.ca) |
| 16:14.26 | CVirus | [TK]D-Fender: 10 lines connected to the asterisk server and asterisk handles the calls and if all the lines are busy ... asterisk will put the agent on hold |
| 16:14.33 | codejunky | Hello, I am phoning out with asterisk+misdn, it works quite well, except an echo of my voice when talking. Any tips how I can get rid of it? I am using asterisk 1.2.x |
| 16:15.06 | CVirus | [TK]D-Fender: I mean it's better to use a PCI card or external device or what ? |
| 16:15.45 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:16.00 | CVirus | [TK]D-Fender: what's the best choice with respect to your experience I mean |
| 16:17.37 | *** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM000a73a1559d.cpe.net.cable.rogers.com) |
| 16:18.36 | ManxPower | CVirus: the best choice is to get a PRI and a Digium or Sangoma card |
| 16:19.41 | coppice | codejunky: add the oslec echo canceler to misdn |
| 16:21.10 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 16:21.36 | CVirus | ManxPower: what if I can't get PRI ? |
| 16:21.37 | codejunky | coppice: I added echocancel=yes to my asterisk/misdn.conf now it is better. |
| 16:22.10 | ManxPower | CVirus: then you won't have the best choice. |
| 16:22.11 | coppice | codejunky: make sure you use the oslec canceler, though. the others are junk |
| 16:22.44 | CVirus | ManxPower: why ? |
| 16:23.01 | Sweeper | hey, someone name a provider that takes instant CC payments and offers instant DID setup? |
| 16:23.12 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583591.dsl.bell.ca) |
| 16:23.41 | codejunky | coppice: How can I make sure that I am using it? |
| 16:23.44 | *** join/#asterisk elriah (n=null@adsl-070-148-053-190.sip.bhm.bellsouth.net) |
| 16:23.48 | ManxPower | because the best was to get calls into/out of Asterisk is a PRI |
| 16:23.54 | elriah | Hi all. Anyone have the latest polycom firmware that works with the 330? |
| 16:24.07 | ManxPower | elriah: your polycom reseller |
| 16:24.25 | elriah | Yea, I know that. It's voipsupply, kind of a pain, I was looking for the quick fix ... |
| 16:24.42 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 16:24.43 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 16:25.00 | ManxPower | freedomphones doesn't have it? |
| 16:25.32 | elriah | No, it would be a 2007 revision. |
| 16:25.39 | elriah | Thanks, though. |
| 16:25.47 | ManxPower | polycom's web site does not have it? |
| 16:25.56 | elriah | Yea, but you have to be a reseller to get it. |
| 16:26.17 | ManxPower | no, polycom has the 2nd to the latest releases freely available |
| 16:28.40 | elriah | hmm.. it looks like all of it is there downloadable now... |
| 16:29.04 | ManxPower | everything except for the latest |
| 16:29.36 | elriah | Ahh.. This may do, hey thanks ManxPower!!! |
| 16:30.11 | ManxPower | elriah: you are welcome to paypal your thanks to eric@fnords.org (that goes for everyone I've helped in the past 5 years) |
| 16:31.20 | *** join/#asterisk Netgeeks (n=chris@204.11.231.198.static.etheric.net) |
| 16:31.37 | [TK]D-Fender | CVirus: 2nd to latest isn't so great. That leaves you on 2.1.X family and 2.2.X is out. |
| 16:31.47 | [TK]D-Fender | elriah: rather |
| 16:33.52 | hmmhesays | I wish there was an equivalent to exchange in linux |
| 16:34.07 | [TK]D-Fender | hmmhesays: OpenExchange <- |
| 16:34.08 | Qwell | hmmhesays: postfix |
| 16:34.31 | cpm | OX is hardly an equivalent. |
| 16:34.52 | cpm | postfix is a fine, if not the best MTA out there, but it is still not an exchange replacement |
| 16:35.05 | Qwell | okay, postfix with a script that takes 80% CPU usage |
| 16:35.09 | cpm | heh |
| 16:35.13 | Qwell | that'll get you close enough |
| 16:35.17 | ManxPower | exchange usually means "network/group scheduling using outlook" |
| 16:35.27 | ManxPower | at least that is what people are looking for. |
| 16:35.31 | cpm | ManxPower, yup. |
| 16:35.33 | *** join/#asterisk shinao1 (n=shinao1@77.220.1.216) |
| 16:35.52 | [TK]D-Fender | cpm: Screw them all.. phpGroupWare ;) |
| 16:36.09 | ManxPower | phpgrouware is a piece of shit. |
| 16:36.14 | cpm | [TK]D-Fender, I think eGroupware just might be the best of breed at this time. |
| 16:36.27 | ManxPower | Their release cycle and massive bugs make Digium's developement process look perfect. |
| 16:37.06 | cpm | eGw 1.4 is pretty good stuff. The email client is seriously ugly, but works quite well. And I like their shared calendar a lot |
| 16:37.47 | hmmhesays | openexchange eh? |
| 16:37.48 | *** join/#asterisk Juggie (n=Juggie@CPE00c0f02a7bf2-CM001371886eee.cpe.net.cable.rogers.com) |
| 16:37.56 | *** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com) |
| 16:38.00 | hmmhesays | I see exchange 07 is x64 only |
| 16:38.07 | Qwell | ha, seriously? |
| 16:38.11 | *** join/#asterisk zydrunas (n=Miranda@216.49.228-ip-184.ckt.net) |
| 16:38.13 | coppice | yep |
| 16:38.23 | Qwell | stupid |
| 16:38.24 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-08b10e906082fa7a) |
| 16:38.24 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 16:38.32 | cpm | hmmhesays, OpenXchange = OX, it worth looking at, but I had it for a year, and found it quite lacking. Exchange it ain't. |
| 16:38.38 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 16:38.58 | coppice | is that a compliment? |
| 16:39.08 | Qwell | yeah, really.. |
| 16:39.09 | hmmhesays | say what you will but exchange is pretty damn nice |
| 16:39.18 | hmmhesays | 03 was anyway |
| 16:39.19 | coppice | huh? |
| 16:39.20 | Qwell | hmmhesays: ever managed exchange? |
| 16:39.25 | Qwell | or used it? |
| 16:39.25 | hmmhesays | Qwell: yes |
| 16:39.44 | hmmhesays | name me a good replacement that is easier to manager |
| 16:39.46 | hmmhesays | *manage |
| 16:39.49 | coppice | the usually opinion of exchange from MS fans is its ()*&(*&^#$(R*&W$^(*&#^(*&^ awful |
| 16:40.32 | hmmhesays | sure they complain, but name a good replacement |
| 16:40.39 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 16:40.45 | Zeeek | see #asterisk-users-conference - conference is on now about ENUM |
| 16:40.48 | hmmhesays | i bet 99% of them can't and they are just bitching just to bitch |
| 16:40.50 | cpm | there is no 'drop-in' replacement for exchange. One can get some of the features from various pieces of software, but nothing replaces all of it. |
| 16:41.09 | Qwell | russellb: are we connected? |
| 16:41.27 | russellb | Qwell: no but we can be |
| 16:41.43 | Qwell | You'd be pretty awesome if we were :D |
| 16:43.39 | cpm | coppice, that's only the ones who are aware of how much of their information has been lost. |
| 16:43.44 | russellb | Qwell: done |
| 16:45.36 | CCFL_Man2 | i have no idea how to configure these terminal servers |
| 16:45.51 | *** join/#asterisk Yourname`` (n=IM@unaffiliated/yourname/x-837320) |
| 16:45.52 | Sweeper | CCFL_Man2: user manual maybe? |
| 16:45.57 | Yourname`` | http://bugs.digium.com/view.php?id=10603 |
| 16:46.13 | Yourname`` | According to this, I don't think I'd be able to use HANGUPCAUSE or DIALSTATUS for callfiles. |
| 16:46.22 | Sweeper | I mean, ffs, it's usually just ip/port -> serial, and set the serial connection params |
| 16:46.39 | Qwell | russellb: You rock |
| 16:46.44 | [TK]D-Fender | Yourname``: Again You need to think about what kind of CHANNEL you are originating.... |
| 16:46.58 | Yourname`` | What do you mean? |
| 16:47.16 | [TK]D-Fender | Yourname``: Show me a sample of the channel ayou are creating with your call-file. |
| 16:47.17 | Qwell | Zeeek: what's the channel? |
| 16:47.26 | CCFL_Man2 | Sweeper: it's a lantronix ets8p |
| 16:47.28 | Yourname`` | And the codes that HANGUPCAUSE always return 0. |
| 16:47.39 | Yourname`` | [TK]D-Fender : one sec. |
| 16:47.56 | CCFL_Man2 | i can't figure out how to get into the mode to change parameters |
| 16:47.59 | Sweeper | CCFL_Man2: first, I shall say "should have bought cyclades" |
| 16:48.23 | CCFL_Man2 | Sweeper: we use lantronix at work :P |
| 16:48.28 | russellb | Qwell: #asterisk-users-conference |
| 16:48.49 | Sweeper | CCFL_Man2: http://las.perkinelmer.com/Content/TechnicalInfo/TCH_LantronixETS8PTerminalServer.pdf <-- pretty basic guide there... |
| 16:48.59 | CCFL_Man2 | ok |
| 16:51.12 | CCFL_Man2 | Sweeper: this is exactly what i needed! |
| 16:52.06 | Yourname`` | [TK]D-Fender: It's a SIP channel |
| 16:52.10 | CCFL_Man2 | think i should connect DCE to port 1 or no? |
| 16:52.26 | [TK]D-Fender | Yourname``: So use a channel type where you can issue the DIAL yourself. |
| 16:52.42 | [TK]D-Fender | Yourname``: AND check the result |
| 16:53.00 | Yourname`` | [TK]D-Fender: And which one would that be? |
| 16:53.32 | [TK]D-Fender | Yourname``: Ask yourself which CHANNEL type allows you to execute DIALPLAN..... |
| 16:53.54 | Yourname`` | SIP |
| 16:54.01 | *** join/#asterisk slakware2 (n=attila@201.19.152.12) |
| 16:54.03 | [TK]D-Fender | Yourname``: this is a baited question you really had better not need any more leading to reach.... |
| 16:54.19 | [TK]D-Fender | Yourname``: No, SIP just dials OUT. has nothing to do with DIALPLAN. |
| 16:54.24 | slakware2 | I'm not sure if anyone answered my question as I lost connection to irc. I will repeat. |
| 16:54.26 | slakware2 | Is it posible to add a peer to multiple sip contexts? context=incoming&outgoing ? |
| 16:54.28 | Yourname`` | [TK]D-Fender : LOL.. exten? |
| 16:54.35 | [TK]D-Fender | Yourname``: You're not thinking, and clearly not reading the list of channel types in * |
| 16:55.04 | [TK]D-Fender | Yourname``: "exten" is a part of the dialplan. |
| 16:55.26 | Yourname`` | [TK]D-Fender : Local? |
| 16:55.33 | [TK]D-Fender | Yourname``: Go to the damn WIKI and read the list of * channel types! If you don't even know this, you should head back to sqaure one. |
| 16:55.40 | [TK]D-Fender | Yourname``: **YES** |
| 16:55.57 | wunderkin | slakware2, no you have to use include, in extensions.conf |
| 16:56.02 | slakware2 | ok |
| 16:56.03 | slakware2 | thanks |
| 16:56.49 | Yourname`` | [TK]D-Fender : lol.. but the reason why I couldn't come up with local is because I don't know how it could help.. |
| 16:59.04 | [TK]D-Fender | Yourname``: Local exectutes DIALPLAN code to start your call. Hence YOU are the one calling DIAL and you can check whatever you want after that too. |
| 16:59.19 | [TK]D-Fender | Yourname``: And have full access to all the usual variables. |
| 17:00.25 | *** join/#asterisk ixx (i=foobar@cpe-70-113-209-33.austin.res.rr.com) |
| 17:00.42 | Yourname`` | [TK]D-Fender : Reading it now.. sounds like just what I need! |
| 17:01.02 | NOT_guru | ~wiki |
| 17:02.20 | [TK]D-Fender | ~wikis |
| 17:02.21 | jbot | from memory, wikis is http://www.voip-info.org |
| 17:02.49 | NOT_guru | heh thanks fender |
| 17:03.40 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 17:03.42 | NOT_guru | fender.. do you use alternative OS's ( BSD MAC Linux ) and if so have you personally used a softphone on any/ |
| 17:04.00 | NOT_guru | its just a flat question I am curious about |
| 17:05.05 | [TK]D-Fender | NOT_guru: I use Slackware, CentOS, and Ubuntu personally in addition to MS OS's On Linux I think I tried Linphone once, and Ekiga a few times. |
| 17:05.19 | Yourname`` | How can I logoff a manager from CLI? |
| 17:05.34 | NOT_guru | yah I didn't like ekiga |
| 17:05.41 | NOT_guru | thank you for your input its appreciated |
| 17:05.53 | NOT_guru | and as ussual thanks for helping in this channel |
| 17:06.10 | NOT_guru | I know you have helped me in past ..."thank you" |
| 17:06.11 | Echinos | ekiga locks up on me consistently |
| 17:06.21 | [TK]D-Fender | NOT_guru: np |
| 17:06.27 | Echinos | ntm it doesn't do IAX |
| 17:06.35 | NOT_guru | on windows I have really liked counterpath xlite best |
| 17:06.41 | NOT_guru | thinking of buying a lic for it even |
| 17:07.01 | [TK]D-Fender | NOT_guru: there aren't a lot of FOSS softphone out there, especially decent ones. Give Twinkle a shot. No experience with it, but I guess its ont he list. |
| 17:07.16 | Yourname`` | Great, there doesn't seem to be a way.. |
| 17:07.27 | Echinos | I'm using kiax and iaxcomm.. kiax is simple but works |
| 17:07.53 | [TK]D-Fender | NOT_guru: on Win I prefer Zoiper as it has native transfer, multiple call support, and IAX/SIP in 1 program. X-Lite I like for simple video conferencing. |
| 17:08.26 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 17:09.32 | *** join/#asterisk dug (n=chatzill@sfnsfw01.pillsburywinthrop.com) |
| 17:12.38 | neverblue | can you pipe asterisk -r into a file ? |
| 17:12.52 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 17:14.07 | hellop | anyone play with asterisk on a router? I have a couple. How's the quality with 3 phones? |
| 17:15.52 | NOT_guru | thank you for the input fender I will look into those |
| 17:16.00 | NOT_guru | zoiper and twinkle |
| 17:17.36 | hellop | I figured every asterisk nerd would have one. |
| 17:18.35 | [TK]D-Fender | hellop: Many have tried on OpenWRT. It works as long as you keep everything in the same codec without transcoding and don't ask too much. VM storage is an issue. |
| 17:19.01 | [TK]D-Fender | hellop: And most experienced users know * deserves a real PC to run on, not jsut some shitbox |
| 17:20.46 | hmmhesays | whats the difference between the sangoma a200 and a400? |
| 17:20.59 | Qwell | 200 |
| 17:21.41 | hmmhesays | well yeah, but other than the card length I don't see any difference |
| 17:21.47 | coppice | an a400 is bigger |
| 17:22.02 | coppice | its bigger so it can take more channels |
| 17:22.29 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 17:22.30 | [TK]D-Fender | hmmhesays: mor ports per backplane. |
| 17:22.40 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 17:23.41 | hmmhesays | is the software echo can decent, this is just going to be a for port machine |
| 17:24.05 | coppice | as opposed to an indecent echo can? |
| 17:24.09 | Corydon76-dig | hmmhesays: yes, it's very good... it's just a little processor intensive... |
| 17:24.23 | hmmhesays | well as opposed to hardware echo can |
| 17:24.48 | Corydon76-dig | Hence why Digium doesn't sell it for more than a few concurrent channels |
| 17:25.07 | coppice | most echo cans are software. its just a matter of where that software runs |
| 17:26.41 | *** join/#asterisk Op3r (n=Op3r@121.97.200.135) |
| 17:31.01 | hellop | [TK]D-Fender, that was the answer I was looking for. hmm.. maybe a faster/overclocked router would be more usable. |
| 17:31.23 | [TK]D-Fender | hellop: Far to wimpy to sconsider. |
| 17:31.25 | hellop | [TK]D-Fender, I was just thinking that it would save power at home where I only need one phone |
| 17:31.49 | hellop | [TK]D-Fender, so like noticible lag, echo, etc... |
| 17:32.05 | [TK]D-Fender | hellop: leave it all G.711 and life is good. |
| 17:32.23 | [TK]D-Fender | hellop: Go try it out and see, jsut forget conferencing (MeetMe), etc |
| 17:35.24 | Cresl1n | hmmhesays: the nice thing about the HPEC though is you can set the number of taps you are using |
| 17:35.32 | Cresl1n | so you can make it less processor intensive |
| 17:35.48 | coppice | aren't they called faucets in the US? :-) |
| 17:36.44 | *** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM000a73a1559d.cpe.net.cable.rogers.com) |
| 17:37.38 | hellop | [TK]D-Fender, there is a USB timer so MeetMe is possible on a USB router |
| 17:37.51 | hmmhesays | Cresl1n: that would be nice |
| 17:37.51 | [TK]D-Fender | coppice: Kernel leak flooded my bandwidth! |
| 17:37.58 | coppice | A CPE canceler doesn't generally need to be longer than about 16ms. If there is a longer echo path for a long distance call there will usually be a canceler in the network dealing with that |
| 17:38.07 | hmmhesays | I also need a good cheap ata |
| 17:38.14 | hmmhesays | well, a good deal |
| 17:38.31 | hmmhesays | mediatrix has given me hell on the faxing side |
| 17:38.34 | [TK]D-Fender | hellop: if you've confirmed that its been coded and tested sure... But I doubt there's enough CPU to think about this even if it were possible. |
| 17:38.34 | hmmhesays | I need something better |
| 17:38.57 | coppice | mediatrix has the world's suckiest T.38 implementation |
| 17:38.58 | Echinos | Anyone have any experience with unlimitel (in canada)? |
| 17:39.06 | hmmhesays | oh I know it |
| 17:39.21 | hmmhesays | coppice: who has a good one? |
| 17:39.38 | coppice | its very software revision dependent. |
| 17:39.55 | coppice | even people like cisco have some stinkers |
| 17:40.41 | hmmhesays | heh, how about a pap2 |
| 17:41.04 | [TK]D-Fender | Echinos: I have several clients who are very pleased with them. |
| 17:41.54 | hmmhesays | are pap2's reliable? |
| 17:42.21 | Damin | Yes.. |
| 17:42.26 | *** join/#asterisk saftsack (n=saftsack@pD9E07B5E.dip.t-dialin.net) |
| 17:43.58 | hmmhesays | now I need some good ip phones, haha |
| 17:44.04 | hmmhesays | i like these spa-942s |
| 17:48.56 | Echinos | [TK]D-Fender: I just want to start small and ramp up the * features as I can integrate them.. I just called them, and they say it's ok for residential so long as you are using it for both in & outbound |
| 17:50.04 | [TK]D-Fender | Echinos: Its good for either IMO |
| 17:50.19 | [TK]D-Fender | Echinos: If I even have the need for local termination I'd probably go with them. |
| 17:50.48 | Echinos | Well, they said on the phone that they would'nt like you to useit for just outbound |
| 17:50.55 | Echinos | IIRC |
| 17:51.03 | Echinos | But that's no problem |
| 17:51.24 | Echinos | now if I can just get sip to iax working, I'll behome free |
| 17:52.04 | Echinos | (for some reason, I can't hear the PC using sip from the IAX phone on the other PC) |
| 17:52.04 | [TK]D-Fender | Echinos: I WOULDN'T use them just for outbound. Their LD rates aren't the "best", but very well valued overall. |
| 17:52.35 | Echinos | Well, it's 2cents/mi if it's not in their local net |
| 17:52.39 | [TK]D-Fender | Echinos: Where are each of these phones relative to your * server? |
| 17:52.45 | Echinos | both on local lan |
| 17:52.55 | Echinos | I must have a setting messed up somewhere |
| 17:53.03 | [TK]D-Fender | Echinos: go play around |
| 17:53.38 | Echinos | I tried 2 different sip softphones, they can hear me, but I can't hear them, and they can hear themselves on their speaker, so I know the mic works |
| 17:53.53 | Echinos | I wanna just install an IAX phone |
| 17:53.55 | Echinos | :) |
| 17:54.48 | *** join/#asterisk reallost1 (n=reallost@dpc6745150081.direcpc.com) |
| 17:55.14 | CCFL_Man2 | anyone here use an adit 600? |
| 17:55.23 | Echinos | I also saw during the NFLpregame show yesterday - Sprint has a mobile broadband service |
| 17:55.38 | Qwell | sprint sucks though |
| 17:55.44 | Qwell | I fully expect anything they do to fail |
| 17:55.50 | Qwell | at least on the mobile side |
| 17:55.55 | Echinos | I wonder how $$$ it would be |
| 17:56.08 | Echinos | I can use my cell for data, but it's lke 10$ a meg |
| 17:56.11 | Qwell | most carriers have something like that |
| 17:56.14 | CCFL_Man2 | it's $70 a month, like verizon |
| 17:56.21 | Echinos | unlimited!? |
| 17:56.22 | Qwell | it's just a pcmcia card they give you for laptops |
| 17:56.28 | CCFL_Man2 | yeah |
| 17:56.40 | CCFL_Man2 | yeah, they give you an avdo pcmcia card |
| 17:56.42 | Echinos | that's actually not too bad, solong as the service doesn't suck |
| 17:56.43 | CCFL_Man2 | evdo |
| 17:56.53 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 17:57.01 | Sci_05 | I have the verion evdo card, its not bad but its nothing great |
| 17:57.09 | Sci_05 | ping times are 300+ms |
| 17:57.14 | CCFL_Man2 | Echinos: verizon says you can use it only for browsing the net, "typical" residential use |
| 17:57.19 | Echinos | what kind of bandwidth? |
| 17:57.35 | Echinos | ie. kps... |
| 17:57.43 | reallost1 | Are there any know problems with func_odbc doing inserts? I can do reads all day, but inserts are dying. |
| 17:57.44 | Sci_05 | I had the broadband service at $70 a month and I would get like 30-40Lbps |
| 17:57.52 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 17:57.54 | Echinos | hrm |
| 17:58.06 | CCFL_Man2 | vzn evdo gives you 700kbit |
| 17:58.09 | Echinos | well, it would have to be for someone on the road alot |
| 17:58.36 | Echinos | and no laptop -> * phone calls |
| 17:58.45 | Sci_05 | if you have verison I would recommand getting a blackberry and then you can pay $40 a month for intenet access and $15 a month of "tethering" (sp?) so you can use your phone like a modem |
| 17:59.18 | CCFL_Man2 | Sci_05: yeah, though the ping times are ass |
| 17:59.30 | Sci_05 | ya 300+ms |
| 17:59.30 | Echinos | Well, I'm not going to get into that anyway, I just want a way to reduce my cell bill |
| 17:59.34 | CCFL_Man2 | but vzn wants it that way |
| 17:59.53 | CCFL_Man2 | Echinos: easy, get a cheap voip provider |
| 18:00.06 | CCFL_Man2 | cancel cellphone |
| 18:00.29 | Echinos | I need to talk while in the carquite a bit |
| 18:01.20 | etfonhomey | Anyone used XO Communications in the NJ/NYC area? |
| 18:01.40 | Echinos | I'm planning to get a free/cheap incoming call plan, and do some kind of callback fuction with * |
| 18:03.29 | Echinos | I would love to be able to send sms for free from *, but alas, not possible |
| 18:07.02 | Linex | Is there anything I need to do to make a three way call or its just a client side thing ? |
| 18:07.04 | Echinos | CCFL_Man2: is the 'CFL' part of your nick anything to do with the canadian football league? ;) |
| 18:08.07 | *** join/#asterisk Zion800 (n=None@cpe-76-167-156-224.socal.res.rr.com) |
| 18:09.05 | *** join/#asterisk potsboy (n=chrisg@vc-196-207-32-227.3g.vodacom.co.za) |
| 18:09.12 | *** join/#asterisk Libila (n=irssi@ip68-101-243-103.sd.sd.cox.net) |
| 18:10.44 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:13.51 | *** join/#asterisk alancio (n=alan@201.208.167.107) |
| 18:14.15 | alancio | quick question: how many regular phones can I connect to an fxs port of a TDM? |
| 18:14.55 | CCFL_Man2 | Echinos: no, Cold Cathode Flourescent Lamp |
| 18:15.05 | Strom_M | alancio: up to a REN of 3.0 IIRC |
| 18:15.26 | alancio | what does that mean? |
| 18:15.30 | CCFL_Man2 | 5 is the common standard |
| 18:16.05 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 18:16.27 | Corydon76-dig | REN==Ring Equivalence Number |
| 18:16.41 | Qwell | ~ren |
| 18:16.42 | jbot | methinks ren is Ringer Equivalence Number - a telephone line can normally supply upto 4 REN, where a standard telephone/answering machine etc would equal 1 REN |
| 18:16.47 | Corydon76-dig | Traditionally, a very old mechanical ringer used a REN of 1.0 |
| 18:17.07 | Corydon76-dig | Most phones today use under .1 REN apiece |
| 18:17.17 | Corydon76-dig | It's basically the amperage output |
| 18:17.35 | Corydon76-dig | but in a slightly more useful number |
| 18:17.58 | alancio | ok, so that means I can connect up to 30 modern regular phones? |
| 18:18.05 | Corydon76-dig | Or more |
| 18:18.16 | Qwell | They will all have the same extension though |
| 18:18.22 | Corydon76-dig | Or less, depending upon the model |
| 18:18.23 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 18:18.36 | alancio | yes, I'm aware of the extension thing |
| 18:18.43 | Qwell | and they'll all ring at once, and if two people pick up two phones, they'll be in the same conversation |
| 18:19.08 | alancio | ok, and both will be able to talk and hear the other party and each other |
| 18:19.11 | alancio | right? |
| 18:19.14 | Qwell | yes |
| 18:19.57 | alancio | thats ok, I am using this for a big house, with 2 floors and 8 phones in each floor |
| 18:20.15 | alancio | I can't buy 16 ip phones, so this is the way I'll go, one fxs for each floor |
| 18:20.30 | jfitzgibbon | alancio: you won't be able to do anything to an individual phone though, so you lose a lot of the flexibility that * provides |
| 18:20.53 | alancio | anything like what? |
| 18:20.56 | jfitzgibbon | alancio: i.e. don't make phones in bedrooms ring after 10pm - forget it if they're all on a single FXS port off your * server |
| 18:21.07 | CCFL_Man2 | Corydon76-dig: western electric bell ringers were rated at 1 REN |
| 18:21.19 | jfitzgibbon | alancio: you need to be able to address phones individually to do stuff like that |
| 18:21.25 | alancio | I could disable incoming calls after 10pm for a complete floor, right? |
| 18:21.31 | Qwell | sure |
| 18:21.45 | jfitzgibbon | alancio: only if that entire floor was on a single port and the other floors were on a different port |
| 18:21.51 | CCFL_Man2 | plus no matter the REN, there is a limit on how many phones than a loop can drive |
| 18:21.52 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:22.21 | jfitzgibbon | alancio: nvm, I see that's what you're planning to do |
| 18:22.27 | alancio | ok, I'll go this way until hardware sip phones become more affordable |
| 18:23.15 | alancio | this might be a silly question, but is it ok to share a regular computer network with ip phones? |
| 18:23.31 | Qwell | sure |
| 18:23.40 | Qwell | having separate vlans with qos would be ideal, but... |
| 18:24.41 | alancio | ok, thanks |
| 18:24.45 | *** join/#asterisk Lucky7 (n=Adam@207.200.28.175) |
| 18:24.58 | alancio | I'll buy the TDM with 2 fxs and 2 fxo |
| 18:25.10 | hmmhesays | this is crazy i have a subscribe 'looking for 200 in presence' and in [presence] exten => 200,hint,SIP/100 |
| 18:25.17 | Lucky7 | Hm. |
| 18:25.19 | Lucky7 | i've got a Polycom 330 |
| 18:25.25 | Lucky7 | for some reason (I assume somehow its got the wrong sip.ld) but anytime it gets a voicemail, my phone reboots. |
| 18:26.15 | Lucky7 | anyone got a good sip.ld / bootrom.ld that I can use? (or tell me one thats downloadable, 2.2.0 isn't because polyocm is lame) |
| 18:26.21 | hmmhesays | that seems to indicate 200 exists in context presence |
| 18:27.06 | alancio | hey, any of you people can advice me where to order the TDM? |
| 18:27.22 | *** join/#asterisk maxyz (n=maxy@190.55.86.61) |
| 18:27.22 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 18:29.05 | maxyz | Hi, I have two sip FXO devices, I would like to use them in a group (that is, to be able to get the first unused), is there a trivial way to do it? |
| 18:29.45 | jfitzgibbon | maxyz: you need to define the groups in zapata.conf and then dial using Zap/[GgRr]GROUPNUM/phonenumber |
| 18:30.03 | hmmhesays | so I'm stumped |
| 18:30.09 | jfitzgibbon | maxyz: whoops, just saw that you're using SIP |
| 18:31.12 | maxyz | jfitzgibbon: Yeap, in sip there are no groups, as far a I could find. |
| 18:31.49 | maxyz | But, there it /should/ be some other way around |
| 18:32.05 | jfitzgibbon | maxyz: a macro that calls Dial on the first device, then re-calls dial on the second if $DIALSTATUS is 'CHANUNAVAIL' would work. There are probably more elegant solutions others know of - all my outbound SIP is trunked, so I haven't tackled this before |
| 18:33.15 | *** join/#asterisk |NexT| (n=next@netcache1.altecom.net) |
| 18:34.04 | hmmhesays | what the heck am I missing |
| 18:34.30 | maxyz | jfitzgibbon: Ok, that sounds, like it should work. Mmh, I may add the FXO lines in a "circular" database array, and get the index of the last used. |
| 18:35.23 | jfitzgibbon | maxyz: you might also look at the group count dialplan functions |
| 18:35.37 | maxyz | I'm still a bit shy with the asterisk MACRO languague, though |
| 18:36.24 | jfitzgibbon | maxyz: assign outgoing calls to a named group based on the device you're attempting to dial out on. Then check the count of said named groups before dialing the next call |
| 18:37.04 | jfitzgibbon | maxyz: so if ${GROUP_COUNT(sip_fxo_1)} is 1, then you know you should not try dialing the first device, but should go to the second immediately |
| 18:37.31 | jfitzgibbon | maxyz: saves the round trip of doing an INVITE and getting a 480 back from the first device every time you have one call in progress |
| 18:38.17 | jfitzgibbon | maxyz: * doesn't really have a macro language per se. It's all diaplan, macros are just an alternate way to jump around in the dialplan and avoid repeating yourself |
| 18:38.58 | maxyz | jfitzgibbon: Asterisk does always know if a line is available? |
| 18:39.58 | jfitzgibbon | maxyz: there is the ChanIsAvail application, but as I understand it, the return values cannot be absolutely counted upon for stuff like this. Especially for SIP. I think it's a bit more accurate for Zaptel channels |
| 18:40.04 | maxyz | jfitzgibbon: Or should I enable something to rely on group counts? |
| 18:40.28 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 18:41.15 | jfitzgibbon | maxyz: you just have to add dialplan logic that uses the GROUP and GROUP_COUNT functions to implement what I suggested above. It would make sense if that logic were in a macro (i.e. macro-dialout-pstn) that you could call from various other places in your dialplan |
| 18:42.02 | *** join/#asterisk martinvandiest (n=chatzill@140.239.173.226) |
| 18:42.55 | CCFL_Man2 | Sweeper: i guess there is no ssh with the lantronix? |
| 18:44.11 | maxyz | jfitzgibbon: Ok, I'll work on it. |
| 18:44.34 | maxyz | thanks |
| 18:47.00 | martinvandiest | Hello, I having trouble getting my fxs extensions to pass cid or stutter tone over rhino channel bank. |
| 18:47.28 | Lucky7 | [TK]D-Fender > You around? |
| 18:48.52 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 18:49.02 | hmmhesays | this is driving me nuts |
| 18:49.03 | *** part/#asterisk potsboy (n=chrisg@vc-196-207-32-227.3g.vodacom.co.za) |
| 18:49.35 | *** join/#asterisk CVirus (n=GoD@212.12.250.74) |
| 18:49.37 | hmmhesays | I have a hint extension in the proper context, but asterisk is still returning a 404 |
| 18:50.28 | martinvandiest | We tried various mailbox= combinations (FreePBX) to no avail. |
| 18:51.35 | *** join/#asterisk Shido6 (n=shido6@204.126.120.132) |
| 18:52.55 | martinvandiest | We don't understand why when the fxs channel goes off-hook asterisk never shows ext. #. |
| 18:57.00 | hmmhesays | is the hint extension in 1.4 broken? |
| 18:57.06 | hmmhesays | well 1.4.4 |
| 18:57.25 | Lucky7 | anyone here have a polycom IP330 |
| 18:57.34 | *** join/#asterisk EricL (n=eric@clydesdale.linkexperts.com) |
| 18:58.12 | EricL | Is there another (non-Java) to have faxes come in over SIP and then get emailed to a user based on whichever phone number (DID) the fax came in on? |
| 18:58.25 | EricL | Another way meaning besides AsterFax. |
| 18:59.03 | hmmhesays | rxfax and some scripts |
| 18:59.21 | Yourname`` | <PROTECTED> |
| 18:59.31 | Yourname`` | LOL sorry, couldn't help but share. |
| 18:59.32 | [TK]D-Fender | Lucky7: yup |
| 18:59.58 | Netgeeks | Hey folks, does anyone know off the top of thier head of a 'is numeric' type function for asterisk dialplan? |
| 19:00.12 | hmmhesays | bah [TK]D-Fender help me |
| 19:00.24 | hmmhesays | i'm going insane |
| 19:00.55 | [TK]D-Fender | hmmhesays: it SHOULD see the exten for presence. That looks NORMAL. Whats the problem? |
| 19:01.06 | hmmhesays | check out this debug |
| 19:01.07 | EricL | hmmhesays:It says that I need a fax machine for that. |
| 19:02.11 | hmmhesays | http://www.pastebin.ca/686034 |
| 19:02.14 | EricL | Is there any way that this can be done without an ATA device and a fax machine (software based)? |
| 19:02.37 | hmmhesays | whatever you read is wrong |
| 19:02.52 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:02.58 | hmmhesays | rxfax is an app for receiving faxes via wherever |
| 19:03.38 | EricL | So you don't need it to be processed through a fax machine? |
| 19:04.08 | hmmhesays | we obviously the fax has to come from somewhere |
| 19:05.00 | hmmhesays | well I guess I'll try this with 1.2 and see if it is a 1.4 problem |
| 19:05.09 | EricL | I meant where I don't need a fax machine and the other end will have one. |
| 19:05.22 | hmmhesays | yes |
| 19:05.49 | linagee | what if i put a T1 loopback on a PRI? would it loopback all the channels, or fsck things up, or sparks? |
| 19:06.22 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 19:06.22 | *** mode/#asterisk [+o anthm] by ChanServ |
| 19:07.30 | outtolunc | linagee: no it will not hurt a pri, it is a T1 after all |
| 19:08.01 | [TK]D-Fender | hmmhesays: pastebin your sip.conf |
| 19:08.18 | linagee | outtolunc: right... but what is the physical different in a T1 for data and a T1 for voice? (er, technically an ISDN T1 i guess) |
| 19:08.29 | *** part/#asterisk martinvandiest (n=chatzill@140.239.173.226) |
| 19:08.31 | [TK]D-Fender | linagee: PRI is a SIGNALLING over TA. So a loopback is a loopback |
| 19:08.40 | [TK]D-Fender | T1 |
| 19:08.47 | linagee | hmm |
| 19:08.53 | [TK]D-Fender | linagee: All jsut signalling. |
| 19:09.00 | linagee | [TK]D-Fender: you just can't find books on the subject. lol |
| 19:09.06 | [TK]D-Fender | ~101 |
| 19:09.06 | jbot | i guess 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 19:09.08 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 19:09.10 | [TK]D-Fender | There |
| 19:09.16 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 19:09.17 | [TK]D-Fender | I found them nice and fast... |
| 19:09.23 | linagee | [TK]D-Fender: cool. thanks. :) |
| 19:09.36 | akirch_ | [TK]D-Fender, yeah but you cheat, you've been here for ages! :-p |
| 19:09.57 | [TK]D-Fender | linagee: thank Strom_M , its his hosting, I'm just the messenger |
| 19:09.58 | linagee | [TK]D-Fender: s/you/i/ s/can't/now can/ |
| 19:10.20 | linagee | strom bad. :) |
| 19:10.23 | hmmhesays | http://www.pastebin.ca/686047 <-- |
| 19:10.23 | linagee | er, strong bad. hehe |
| 19:10.30 | Strom_M | i'm bad? |
| 19:10.37 | linagee | Strom_M: just trying a play on names |
| 19:10.42 | J4k3 | my voip provider is several orders of magitude more reliable than the local ILEC. |
| 19:10.55 | Strom_M | J4k3: who's your ILEC? GTE? :) |
| 19:10.59 | linagee | Strom_M: you've never read strong bad's emails? |
| 19:11.05 | EricL | hmmhesays: Thanks for the help. Looks like this isn't as straightforward as I was hoping. |
| 19:11.06 | Strom_M | linagee: forever ago |
| 19:11.16 | J4k3 | Strom_M: kinda... we're contel -> GTE -> Valor -> Windstream turf, and the local switch is a DMS-10 |
| 19:11.19 | J4k3 | :P |
| 19:11.24 | Strom_M | J4k3: close enough |
| 19:11.31 | Strom_M | there's GTE in there |
| 19:11.53 | J4k3 | Valor = mexican-owned shit telco, windstream = the merger of valor, alltel (awhell) and KMC (voted by the PUC worst telco in texas...) |
| 19:11.56 | linagee | Strom_M: do you know about my teleco type question? what happens if you put a loopback plug on a T1 voice PRI? (can it be used to troubleshoot, or will it just sit there like, duhhh.) |
| 19:11.58 | [TK]D-Fender | hmmhesays: 1.4? |
| 19:12.03 | *** part/#asterisk EricL (n=eric@clydesdale.linkexperts.com) |
| 19:12.06 | Strom_M | linagee: first off, it's telco, not teleco |
| 19:12.23 | Strom_M | second, looping back a PRI will work just like looping back any other T1 circuit |
| 19:12.35 | linagee | hm |
| 19:12.36 | Strom_M | third, read the fscking telephony 101 document already |
| 19:12.42 | J4k3 | so, yeah, as long as windstream can bother to keep my T1 lit, my voip works... hauling calls through that DMS-10 leads to "this call cannot be completed as dialed" at least 20% of the time. |
| 19:12.43 | linagee | Strom_M: reading! :) |
| 19:12.43 | hmmhesays | [TK]D-Fender yeah |
| 19:12.46 | [TK]D-Fender | hmmhesays: change from type=friend to type=peer <----------------- |
| 19:13.15 | hmmhesays | yeah I have limitonpeers though |
| 19:13.30 | linagee | hmmhesays: you only have a small close group of friends? |
| 19:13.35 | [TK]D-Fender | hmmhesays: Do it. |
| 19:13.36 | cpm | I love the fact that a decent voip call actually eats bigger holes in a T1 than a channelized call would have on a voice DS1 |
| 19:13.49 | J4k3 | cpm: welcome to overhead. |
| 19:13.59 | cpm | welcome to idiocy, like TVoIP |
| 19:14.03 | cpm | what crap |
| 19:14.05 | linagee | cpm: welcome to protocol overheads |
| 19:14.10 | J4k3 | cpm: and thats not *totally* true, I think you can cram 24 IAX2 trunk g711 calls in a T1, barely. |
| 19:14.11 | hmmhesays | now I'm getting 489 "bad event" |
| 19:14.12 | linagee | J4k3: jinx |
| 19:14.13 | J4k3 | assuming nothing hiccups. |
| 19:14.21 | cpm | J4k3, on a good day, maybe |
| 19:14.35 | cpm | I can get 24 calls on a T1, no worries, no lag, no stuttering |
| 19:14.42 | cpm | with a channel bank. |
| 19:14.45 | *** join/#asterisk dalfry (n=Dalfry@70.89.177.109) |
| 19:14.49 | linagee | J4k3: just use some crappy G729 and the customer will never know. :) |
| 19:14.49 | J4k3 | I can get a lot more than 24 g729 calls through a T1. |
| 19:14.50 | J4k3 | ;) |
| 19:14.57 | linagee | jinx again |
| 19:15.11 | [TK]D-Fender | hmmhesays: reboot the phone & * |
| 19:15.34 | cpm | not nearly as reliably |
| 19:15.39 | J4k3 | hey, at least its g729 and not gsm... euros seem to think that GSM crap is acceptable.... I won't even carry that crap on my hip :P |
| 19:15.47 | cpm | heh |
| 19:15.56 | linagee | J4k3: gsm protocol over a hard line? eww |
| 19:16.20 | J4k3 | gsm compression over anything should be illegal, including cellular :) |
| 19:16.23 | cpm | J4k3, what are you doing slumming here? |
| 19:16.28 | hmmhesays | yeah same |
| 19:16.50 | J4k3 | cpm: I'm doing some things with *... obviously not much since its running on a P3-700.... ;) |
| 19:18.21 | cpm | free at last! |
| 19:18.24 | linagee | we use some sort of avaya thing to route around the SIP at work. (anyone heard of a solacom box?) |
| 19:18.48 | hmmhesays | i'm going to try 1.2 |
| 19:19.12 | many | ohwell |
| 19:19.23 | cpm | try callweaver |
| 19:19.46 | CCFL_Man2 | i get double text on this terminal server |
| 19:20.00 | CCFL_Man2 | and how the hell can you chose a different port |
| 19:20.06 | CCFL_Man2 | damn lantronix |
| 19:20.09 | [TK]D-Fender | hmmhesays: Check out "show hints" |
| 19:21.10 | hmmhesays | for? |
| 19:21.42 | *** join/#asterisk Remowylliams (n=Mare@71.16.217.178) |
| 19:22.42 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 19:23.02 | Remowylliams | Hello everyone Does anyone have any comments on Zapmicro's boards like the zma400p or even the zma100p? |
| 19:23.34 | Qwell | are those motherboards or something? |
| 19:24.47 | Remowylliams | no they are ATA interface cards. the zma400p is like Digium's TDM400P |
| 19:25.28 | Daejeo1 | can you point me to any good documentation "How to upgrade SIP firmware on 7961g"? I want to use this phone with asterisk server. |
| 19:26.19 | Qwell | Remowylliams: I've never heard of them, so I'm guessing junk |
| 19:26.23 | Qwell | ~cheap |
| 19:26.24 | jbot | cheap is probably a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 19:26.27 | *** join/#asterisk Trionnis (n=blah@65-117-172-195.dia.static.qwest.net) |
| 19:27.03 | reallost1 | anyone familar with func_odbc? |
| 19:27.38 | Trionnis | anyone have any experience with * not allowing an outbound sip call from a device even though it's set as "insecure=very", and registered? |
| 19:27.41 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 19:27.55 | J4k3 | junk PC + grandsuck budgetpwned 101's + no pstn connectivity (well, there is an unused x101p in there, that *could* connect to our one last POTS line) |
| 19:28.09 | Remowylliams | Qwell they aren't much cheaper than Digium's stuff. |
| 19:28.20 | J4k3 | total investment for a 3-man office = $110ish |
| 19:28.21 | Remowylliams | Qwell: but thanks for th einput. |
| 19:28.21 | Qwell | so it's an expensive chinese clone |
| 19:29.13 | Trionnis | version 1.2.24, btw |
| 19:29.45 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 19:30.36 | Trionnis | and for reference, the same settings work fine in 1.4.8, I just have all kinds of rtp problems in that version |
| 19:31.10 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:31.28 | *** join/#asterisk limbje (n=root@limbique.xs4all.nl) |
| 19:32.08 | hmmhesays | same problem in 1.2 |
| 19:32.11 | hmmhesays | 404 not found |
| 19:32.38 | limbje | how to get a list of users in ircII |
| 19:32.39 | Daejeo1 | Qwell: sir any help |
| 19:35.17 | hmmhesays | well spa-942's with asterisk subscribes seem to be broken |
| 19:36.02 | *** join/#asterisk guillote_GNU (n=bancaria@host128.190-136-202.telecom.net.ar) |
| 19:36.12 | Remowylliams | when it comes to echo cancellation on a card, that's mostly important if you're running copper back some distance to the switch office right? |
| 19:38.16 | limbje | puzzled here? |
| 19:38.46 | Remowylliams | limbje use /who #channel name |
| 19:39.57 | limbje | tyvm |
| 19:40.07 | Remowylliams | you're welcome |
| 19:40.32 | limbje | i'm spoiled by graphical clients :) |
| 19:40.54 | limbje | now i'm using ircII here |
| 19:40.58 | limbje | brb |
| 19:43.19 | sheppard | irssi++ |
| 19:45.31 | hmmhesays | yes this seems very broken |
| 19:45.59 | Strom_M | what's broken, hmmhesays? |
| 19:46.45 | hmmhesays | Strom_M: my linksys spa-942 trying subscribe/ hint support |
| 19:47.04 | Echinos | irsii-- |
| 19:47.11 | Echinos | I'm using it, but not well :/ |
| 19:47.13 | Echinos | ;) |
| 19:47.50 | Echinos | can't get my windows sorted out |
| 19:49.49 | Shido6 | then use linux |
| 19:49.51 | Shido6 | or osx |
| 19:50.05 | Echinos | me? |
| 19:50.28 | Shido6 | you're the one using windows :) |
| 19:50.37 | Echinos | no, not hose widows, silly |
| 19:50.41 | Echinos | er |
| 19:50.45 | Echinos | not those windows |
| 19:50.50 | Echinos | irssi windows |
| 19:50.50 | Shido6 | :) |
| 19:50.55 | Shido6 | just messing with you |
| 19:50.59 | Echinos | GAH! |
| 19:51.10 | Echinos | tricked! I feel ashamed. :( |
| 19:51.52 | hmmhesays | yes very broken |
| 19:52.14 | Echinos | maybe it's just the implication that I would use windows on any of my PCs |
| 19:52.24 | Echinos | gets me all riled up :) |
| 20:02.24 | *** part/#asterisk Cresl1n (i=matt@nat/digium/x-08b10e906082fa7a) |
| 20:07.51 | HCEvan | anyone had success in resolving NMI problems on an IBM machine with an X100P? |
| 20:10.24 | HCEvan | So far I have tried disabling all ethernet cards, upgraded bios and firmware, changed the irq on the x100p, seated the card in another pci slot, compiled a new kernel, turned off acpi, and a bunch of other stuff.. |
| 20:11.25 | ManxPower | usually NMI indicates a hardware failure |
| 20:11.33 | HCEvan | true. |
| 20:11.44 | HCEvan | but it is only caused by the X100P |
| 20:11.46 | ManxPower | and there have been no actual X100P or X101Ps made for several years |
| 20:11.55 | ManxPower | well, maybe the X100P is broken |
| 20:12.00 | HCEvan | and I have it working perfectly in a Dell Poweredge. |
| 20:12.08 | ManxPower | in any case, contact the vendor you purchased it from. |
| 20:12.09 | HCEvan | yeah I tested in the above ^ |
| 20:13.05 | HCEvan | I have tried 2 separate X100Ps in the machine, same results. |
| 20:13.22 | HCEvan | I have also checked the memory and run diagnostics on the IBM machine. |
| 20:13.24 | HCEvan | no errors. |
| 20:13.35 | HCEvan | the error only comes when I load up the wcfxo kernel module. |
| 20:15.23 | HCEvan | http://www.pastebin.ca/686148 |
| 20:15.23 | HCEvan | relevant dmesg output. |
| 20:15.23 | hmmhesays | has anyone gotten subscribe to work with spa-942's? |
| 20:15.36 | Daejeo1 | anyone can help me upgrade sip firmware - 7961g |
| 20:15.47 | Remowylliams | HCEvan: I'm going to be tweaking my system tonight. After reading the troubleshooting info for analog pci cards. I may have found some fixes to problems I've been having. Seems my clone x100p is on irq 11, which is the same as the vga, and my network interface is using irq 9 |
| 20:16.48 | ManxPower | HCEvan: I have never solved an NMI problem when installing a card in a system. |
| 20:16.59 | ManxPower | I have always had to try a different card or a different system model |
| 20:18.36 | HCEvan | Remowylliams: my X100P is on IRQ 15 while my network interface is on IRQ 10. |
| 20:18.59 | ManxPower | HCEvan: I assume you tried different slots |
| 20:19.01 | Remowylliams | HCEvan: you have only one IDE controller? |
| 20:19.09 | HCEvan | yes I have. |
| 20:19.39 | Remowylliams | HCEvan: normally the second ide controller found on most mother boards is hardwired to IRQ 15. is why I ask |
| 20:19.49 | HCEvan | well |
| 20:19.58 | J4k3 | x100p on IRQ 15... isn't IRQ 15 used for the secondary IDE channel? |
| 20:19.59 | J4k3 | yeah |
| 20:20.31 | HCEvan | there is one PCI slot that is 32-bit for the IBM Remote Supervisor Adapter, then there are 4 more at 64-bit. |
| 20:20.38 | Remowylliams | Normally to fit things in as best you can. You disable irq 3 or 4 or even 3 and 4 which are for he serial ports... |
| 20:20.45 | Remowylliams | and use those. |
| 20:20.47 | HCEvan | I have tried that too. |
| 20:21.02 | J4k3 | HCEvan: you can put a 32 bit card in a 64 bit slot |
| 20:21.11 | HCEvan | yes that is how it is set right now. |
| 20:21.11 | J4k3 | (you can USUALLY put 64 bit pci devices in 32 bit slots, too) |
| 20:21.24 | HCEvan | the only 32 bit I have has an IBM RSA in it. |
| 20:21.36 | HCEvan | so that only leaves 64-bit slots for anything else. |
| 20:21.37 | Remowylliams | And of course there's IRQ 7 for the printer and 6 is used for the mouse usually. |
| 20:22.07 | ManxPower | actually 12 is ps/2 mouse |
| 20:22.11 | ManxPower | 6 is floppy, I think |
| 20:22.24 | Remowylliams | You never want something important on irq 9 and 2. |
| 20:22.38 | Remowylliams | Ah yeah 12 I forgots it. :) |
| 20:22.44 | Remowylliams | ManxPower: Thanks |
| 20:22.44 | *** part/#asterisk Dossy_ (i=nobody@aolserver/dossy) |
| 20:23.16 | J4k3 | well, the real question is... |
| 20:23.31 | J4k3 | why would a * box need a floppy, a ps2 mouse, or PATA controllers? :) |
| 20:24.01 | J4k3 | floppies suck, mouse tends to mean theres graphics involved, and PATA eats cycles like a fat kid with the munchees |
| 20:24.03 | Remowylliams | I've had curious breakups even using the echo test on the local network and I'm betting it's a problem with my network IRQ being on 9. (hope hope hope) |
| 20:24.29 | J4k3 | Remowylliams: I had a lot of funky breakup problems here, ended up being an ethernet switch |
| 20:24.33 | J4k3 | and a couple bad cables |
| 20:24.41 | J4k3 | basically I had to 'overhaul' the network |
| 20:24.53 | Remowylliams | J4k3: I've changed switches even. same deal |
| 20:25.00 | Remowylliams | And cables for that matter. |
| 20:25.11 | J4k3 | ack |
| 20:25.16 | *** part/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 20:25.19 | J4k3 | then yeah, irq or bad ethernet card. |
| 20:25.19 | HCEvan | well on the Dell Poweredge here I have the X100P on the same IRQ as the Ethernet interface and haven't had any problems at all... so they can work together I suppose. |
| 20:25.24 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 20:25.30 | Remowylliams | I've got no ethernet errors either. |
| 20:25.38 | J4k3 | HCEvan: different pci implementations are made differently |
| 20:25.44 | HCEvan | right |
| 20:25.47 | HCEvan | case by case. :/ |
| 20:25.53 | J4k3 | what works on a 440BX won't work on say, a Via KT133 |
| 20:25.59 | Remowylliams | HCEvan: My last Asterisk box used a Dell. (shaking head sadly) |
| 20:26.23 | HCEvan | heh |
| 20:26.31 | J4k3 | my * box is using an anus BX board, and a processor and ram out of a dell |
| 20:26.48 | J4k3 | err asus |
| 20:26.54 | Remowylliams | First the Drive died.. then the powersupply. We hatesesssss Dells. |
| 20:26.54 | J4k3 | its so easy to call asus anus... and its usually more fitting :) |
| 20:27.07 | J4k3 | the PSU killed the drive most likely. |
| 20:27.34 | J4k3 | the dell I stole the ram and cpu out of was one of those godawful ones with the funny ATX-like plug but non-standard pinout |
| 20:27.43 | J4k3 | I enjoyed throwing that thing in the dumpster. |
| 20:27.44 | HCEvan | yeah Dell does not have a really good track record, and I wouldn't use them in production, but this Poweredge 2450 has been good as an Asterisk box. |
| 20:28.10 | J4k3 | trying to compare a [anybody] desktop to an [anybody] server is hard. |
| 20:28.28 | J4k3 | unless its one of those companies that sells "servers" that are nothing but a shitbox peecee with 'server' software installed |
| 20:28.43 | Remowylliams | J4k3: I'm not a big fan of Asus either. 5 years ago I bought a Asus board replacing a board with cap death. And 3 months later I'm crashing left and right.. eventually, snooping around I found it was caused by the memory bus. The board though it has 3 memory slots will only run smoothly with 1 Memory stick. |
| 20:29.08 | J4k3 | Dell's *real* servers aren't cheap, and don't share parts with their consumer desktop machines... they do sell some "shitbox pc servers" too, which are to be avoided. |
| 20:29.35 | J4k3 | Remowylliams: yep... Asus burned me on a few boxes, too. |
| 20:29.56 | J4k3 | literally burned me on one... assholes didn't bother to put the 4-pin secondary power connector on an nforce2 board... smoked the ATX header after a couple months |
| 20:30.08 | HCEvan | heh ouch. |
| 20:30.20 | J4k3 | and it wasn't that the assembler forgot to put the cable on... ASUS DIDN'T BOTHER TO PUT THE PLUG ON THE BOARD AT ALL |
| 20:30.23 | J4k3 | ;) |
| 20:30.32 | Remowylliams | My current Asterisk box is a little hp Pizza box with a 300 Mhz Celleron and over all I can't complain.. it's been mostly a test and try since it's all cludged together in the first place. |
| 20:30.45 | J4k3 | oh well... I could start ranting about Abit and be here all afternoon... my Asus complaints list is much smaller. |
| 20:30.58 | J4k3 | oddly, I'll still buy parts from both... I'm just *Really* careful what I buy, and from where I buy it. |
| 20:31.11 | J4k3 | Remowylliams: wow, thats sloooow. I thought my P3-700 was slow. |
| 20:31.23 | J4k3 | and is that a celeron 300, or 300A? |
| 20:31.33 | HCEvan | heh reminds me of my previous Asterisk box. |
| 20:31.38 | HCEvan | also a 300Mhz celly. |
| 20:31.49 | Remowylliams | Celeron 300a I think. no chance to overclock it though |
| 20:32.20 | J4k3 | it gets beat up by P5-166's on good board :) |
| 20:33.07 | J4k3 | can you even transcode one call with that? |
| 20:33.12 | Remowylliams | It works quite fine handles my current single line and passes on faxes etc. But I'm wanting to make it as solid as I can. the PSTN comes from a pipeline P75 router I use to interface the ISDN |
| 20:33.27 | J4k3 | oh man, oldschool |
| 20:33.31 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 20:33.47 | J4k3 | I haven't heard anyone talk about Ascend Pipelines since... sheeit... 2001? |
| 20:33.57 | J4k3 | awesome :) |
| 20:34.53 | Remowylliams | My first and only pipeline lasted me almost 7 years but 3 weeks ago had a lightning strike about 10 yards from the house and the magic smoke lefteth the box. |
| 20:35.06 | J4k3 | bummer :( |
| 20:35.22 | Remowylliams | I picked up 2 more (one for a spare) off ebay for 30 bucks, the second I got for 18. |
| 20:35.27 | J4k3 | I've got a lot of old random ISDN stuff in the closet, but most of it has died from similar causes over the years. |
| 20:35.41 | J4k3 | my favorite BRI toy is the livingston pm2-ei-5 |
| 20:35.50 | J4k3 | it'll bond 5 BRIs happily. |
| 20:35.58 | Remowylliams | LOL yes very fun. |
| 20:36.12 | J4k3 | (technically it can bond 15 but it runs out of CPU first, and requires expansion cards that haven't been manufactured in like 8-9 years) |
| 20:36.43 | Remowylliams | it costs me $14.00/month for my isdn line and 60 for my analog. |
| 20:37.09 | J4k3 | I had two BRIs connecting my house to the net for about 2 years... 224k (56k per call thanks to a retarded telco) |
| 20:37.16 | *** join/#asterisk Dossy_ (i=nobody@aolserver/dossy) |
| 20:37.26 | Dossy_ | what happened to newmantelecom.com? |
| 20:37.32 | J4k3 | wow, $65ish after taxes per BRI here. |
| 20:37.36 | J4k3 | and $150+ install |
| 20:38.07 | Strom_M | Dossy_: perhaps they've been temporarily disconnected |
| 20:39.06 | HCEvan | hmm well I disabled serial, ide, floppy, and mouse this time... and now the X100P is on IRQ3 but still giving out NMI errors. :/ |
| 20:39.07 | Remowylliams | I'm trying to dig up a cheep TDM400P with 3 FXO's my box only has 1 pci slot |
| 20:39.11 | J4k3 | blah, I gotta go make some money |
| 20:39.12 | J4k3 | I'll be back |
| 20:39.18 | Remowylliams | kk |
| 20:39.56 | Dossy_ | woo, thank god for the Wayback Machine ... |
| 20:40.57 | Remowylliams | LOL yeah archive.org rocks when you have to become an internet archaeologist |
| 20:41.14 | Qwell | meh, I j ust cache the internet locally |
| 20:41.15 | *** join/#asterisk crycos (n=crycos@72.54.46.18) |
| 20:41.38 | Qwell | what do you do when archive.org is down? (like it was earlier this week) |
| 20:41.57 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-d9d2acfd39a13df4) |
| 20:42.10 | Remowylliams | Eat crackers and drink Woodchuck Amber. |
| 20:51.01 | watchy2 | any problems with Dual Core proccessors and *? |
| 20:51.12 | nDuff | watchy2: nope. |
| 20:51.34 | nDuff | watchy2: well, unless you consider missing less interrupts a problem. |
| 20:52.12 | watchy2 | missing less interuppts? |
| 20:53.21 | _x86_ | what do you guys normally do for T1 failover? |
| 20:53.21 | Dossy_ | Qwell: google cache. or, I cry. |
| 20:53.55 | _x86_ | for example, say you have 6 T1's (2 to the PSTN, 4 to stations), and you have 2 servers, each with 6 T1 interfaces |
| 20:54.23 | _x86_ | and you want to failover to the secondary server should the primary die |
| 20:56.44 | *** join/#asterisk Blackthorn (n=support@76-77-161-226.smyth.net) |
| 20:57.25 | Blackthorn | I am running Asterisk 1.4.4. and it keeps failing on me with the message repeated over and over: |
| 20:57.26 | Blackthorn | WARNING[18647]: app_dial.c:674 wait_for_answer: Unable to forward voice frame |
| 20:57.28 | nDuff | _x86_: well, we use IP going out to our stations, so failover there is trivial; for the telco side of things, we haven't implemented it yet but are planning on making it the telco's problem (we get more trunks than we'd need and they route calls to our other trunk if the primary one isn't able to take them; I don't remember what our staff member who used to do telco work called the configuration they do on their side to support it). |
| 20:57.35 | nDuff | _x86_: not the answer you're looking for, I'm sure. |
| 20:57.44 | Blackthorn | know what causes that? I can do stop now, and restart the server and it works fine again for a day or too |
| 20:58.09 | Trevor_b | _x86_: you looked at http://www.failsafevoip.com/index.php?main_page=product_info&products_id=1&zenid=3cf143f34c73388c269b076ee47c558b?? |
| 20:58.12 | Trevor_b | oops |
| 20:58.15 | Trevor_b | _x86_: you looked at http://www.failsafevoip.com/index.php?main_page=product_info&products_id=1&zenid=3cf143f34c73388c269b076ee47c558b ?? |
| 21:00.55 | Trevor_b | _x86_: also when you say you have 4 pri to the stations, are they on another PBX? Or are you just equating the channels in the PRI to number of stations in your environment? |
| 21:02.31 | *** join/#asterisk m0t3jl (n=m0t3jl@ip103.galance.net) |
| 21:04.05 | _x86_ | Trevor_b: no, did not say anything about PRI... this is all entirely CAS |
| 21:04.57 | _x86_ | Trevor_b: we have a huge Toshiba phone system currently... i'm replacing it with (2) HP DL380 G5's with (1) Sangoma A102DX per server and (1) Sangoma A104DX per server |
| 21:05.12 | _x86_ | Trevor_b: we have (2) CAS T1's going out to the PSTN |
| 21:05.19 | _x86_ | Trevor_b: the other (4) T1's are local stations |
| 21:05.28 | m0t3jl | Hi. I have a little problem. I've set up one testing asterisk dedicated PC, I am running FreePBX as well on that machine (along with apache,php, and mysql) ... I have one real phone (a machine) and one sw phone. I am experiencing this: I can call to the *99,*65 and other Asterisk numbers, but I cant call between those two phones. Is there something I omitted? (Those two phones have extension accounts created) |
| 21:05.28 | *** join/#asterisk smultron (n=lukas@cpe-67-9-146-21.austin.res.rr.com) |
| 21:05.45 | _x86_ | m0t3jl: #freepbx |
| 21:05.55 | *** join/#asterisk HCEvan (i=edude@pool-72-77-226-91.tampfl.fios.verizon.net) |
| 21:06.03 | m0t3jl | _x86_, ok ... :-) |
| 21:06.31 | m0t3jl | _x86_, thx :-) |
| 21:07.02 | *** join/#asterisk tuxd00d (i=tuxinato@nat/novell/x-b16e65856b358bfe) |
| 21:08.13 | _x86_ | Trevor_b: so what I need is a failover switch that can failover all (6) T1's at layer 1 |
| 21:08.29 | smultron | is the Digium AIXy only used to connect a regular POTS telephone to an IP PBX? |
| 21:08.36 | _x86_ | (6) T1 in, (12) T1 out |
| 21:09.25 | smultron | or can it be used to connect an incoming FXO line to an IP PBX? |
| 21:09.42 | m0t3jl | Which is the cheapest module I can use to connect my Asterisk PC to the normal telephone network? (We have currently two lines and we would like them to be accesible through asterisk as well) |
| 21:10.24 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 21:10.37 | JerJer | anyone see what's wrong with this: |
| 21:10.42 | JerJer | exten => _X.,n,Set(intended_context=${IF($[ ${SIP_HEADER(X-Intended-Context)} != ""]?${SIP_HEADER(X-Intended-Context)}:0)}) |
| 21:10.55 | JerJer | parser doesn't like the != |
| 21:11.37 | m0t3jl | JerJer, try not to use the spaces aroung the != |
| 21:12.07 | JerJer | same same |
| 21:12.15 | JerJer | i have spaces around other expressions that work |
| 21:12.22 | m0t3jl | JerJer, well, I dont even halfly understand what you command line should do ... I just started with asterisk today :D |
| 21:12.38 | m0t3jl | JerJer, :-) |
| 21:12.56 | JerJer | set the value of intended_context to the value of a custom sip header if it exists, if not set it to 0 |
| 21:13.10 | outtolunc | i always thought IF was eval only to IF (1/true) this : orthat |
| 21:14.38 | JerJer | Syntax IF(<expr>?[<true>][:<false>] |
| 21:14.45 | JerJer | i hope any valid expr can be used |
| 21:14.50 | m0t3jl | JerJer, what does the $ before that [ ${SIP_BLABLA states for? |
| 21:15.25 | JerJer | here is a working line |
| 21:15.26 | JerJer | exten => _X.,n,Set(CDR(on_net)=${IF($[ ${SIP_HEADER(X-On-Net)} = 1]?1:0)}) |
| 21:15.37 | JerJer | that works 100% |
| 21:15.58 | m0t3jl | JerJer, are you sure that its not always true? |
| 21:16.17 | JerJer | m0t3jl: i really have no clue - i am coding this via example here....without fully understanding the concepts |
| 21:16.23 | *** part/#asterisk Remowylliams (n=Mare@71.16.217.178) |
| 21:16.26 | *** join/#asterisk Op3r (n=Op3r@125.212.122.175) |
| 21:16.33 | JerJer | ie i am in problem solving mode |
| 21:16.49 | m0t3jl | ^^ |
| 21:16.55 | JerJer | no that on_net code works perfectly - as expected |
| 21:17.28 | JerJer | we get both 0 and 1 results based on the value of the header |
| 21:18.05 | JerJer | i think the difference with the intended-context crap is we are simply looking for a valid (or invalid) string - just to test if its defined or not |
| 21:18.09 | JerJer | which may be the problem |
| 21:18.32 | m0t3jl | JerJer, what about using $[ ! ${SIP_HEADER(...blabla...)} |
| 21:19.04 | JerJer | i can try but i don't think the parser is currently that powerful |
| 21:19.33 | Dossy_ | hey, anyone who's set up RxFAX-- ever seen this error message? |
| 21:19.35 | Dossy_ | Sep 7 17:19:10 NOTICE[13198]: channel.c:1956 ast_read: Dropping incompatible voice frame on IAX2/jnctn_out-2 of format slin since our native format has changed to ulaw |
| 21:19.35 | JerJer | nope doesn't like it |
| 21:19.36 | m0t3jl | lets try |
| 21:19.46 | m0t3jl | what exactly does the parser spit out? |
| 21:19.50 | Op3r | is allianzinfo any good? |
| 21:20.44 | Damin | JerJer: Moo... |
| 21:20.50 | JerJer | meep meeep |
| 21:21.20 | ManxPower | Dossy_: what format is the call? |
| 21:21.50 | ManxPower | you really should not expect fax to work over IP |
| 21:22.23 | Damin | ManxPower: I've been having great success at faxing over LPC10 lately.. |
| 21:22.27 | outtolunc | what about this? |
| 21:22.30 | outtolunc | _X.,n,Set(intended_context=${IF(${ISNULL(${SIP_HEADER(X-Intended-Context)})}?${SIP_HEADER(X-Intended-Context)}:1)}) |
| 21:23.05 | m0t3jl | outtolunc, do you think that theres such a function? |
| 21:23.14 | outtolunc | which? |
| 21:23.21 | ManxPower | Damin: please don't joke. Dossy_ is very gullible. |
| 21:23.31 | m0t3jl | outtolunc, I dismissed this about the time I saw that it cant handle ! SOMETHING :) |
| 21:23.35 | m0t3jl | outtolunc, the ISNULL |
| 21:23.41 | Daejeo1 | WomanxPower :) |
| 21:23.48 | outtolunc | yes it exists (or did in 1.2) |
| 21:24.01 | outtolunc | ISNULL ISNULL(<data>) NULL Test: Returns 1 if NULL or 0 otherwise |
| 21:24.12 | outtolunc | do a 'show functions' from the CLI |
| 21:24.20 | outtolunc | i use it all the time |
| 21:24.28 | Daejeo1 | ManxPower alias WomaxPower |
| 21:24.37 | ManxPower | Daejeo1: who are you? |
| 21:25.03 | Daejeo1 | I am living human being |
| 21:25.18 | m0t3jl | outtolunc, nice |
| 21:25.43 | m0t3jl | well, as I said some messages ago: I started with asterisk today :-) |
| 21:25.48 | Daejeo1 | did i bother you my friend? |
| 21:28.24 | JerJer | ISNULL - gottcha |
| 21:28.26 | m0t3jl | Anyway ... as the #freepbx is dead like a graveyard .. Can somebody please please take a look at this: I've set up one testing asterisk dedicated PC, I am running FreePBX as well on that machine (along with apache,php, and mysql) ... I have one real phone (a machine) and one sw phone. I am experiencing this: I can call to the *99,*65 and other Asterisk numbers, but I cant call between those two phones. Is there something I omitted? (Those two phones |
| 21:28.26 | m0t3jl | <PROTECTED> |
| 21:28.28 | JerJer | now where is that documented ? |
| 21:28.39 | m0t3jl | JerJer, congrats :-) |
| 21:31.01 | JerJer | outtolunc: ok i don't get a parse error but i don't get the intended 0 (or in your case 1) |
| 21:31.17 | outtolunc | oops <G> :0 |
| 21:31.20 | Trevor_b | m0t3jl: Sounds like errors in extensions.conf, but thats just my first guess. Have an example of your exten => xxx line? |
| 21:31.22 | JerJer | in the ${intended_context} |
| 21:31.32 | outtolunc | change ending :1 to :0 |
| 21:31.48 | m0t3jl | Trevor_b, I can paste the whole file, if you want |
| 21:31.59 | JerJer | i did |
| 21:31.59 | Trevor_b | ~pastebin |
| 21:32.00 | jbot | well, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
| 21:32.07 | JerJer | i get nothing in the CLI |
| 21:32.14 | JerJer | <PROTECTED> |
| 21:32.31 | Trevor_b | stick just your 2 extension lines into pastebin |
| 21:32.37 | Trevor_b | lets start small ;) |
| 21:32.48 | outtolunc | then X-Intened-Context is a var, just an empty one an ISNULL would return 1 IF the var 'X...' didn't exist |
| 21:32.52 | m0t3jl | Trevor_b, thats very kind of you :-) ... However, I did not intend to paste it into the channel :-) |
| 21:33.16 | JerJer | there is no X-intended-Context being sent on this call |
| 21:33.17 | outtolunc | give me a sec.. i'll redo |
| 21:33.36 | Trevor_b | not really, more of a CYA, as i forget im in IRC sometimes and paste stupidly large things myself ;) |
| 21:33.48 | Dossy_ | so, fax over IP is impossible? |
| 21:34.05 | Dossy_ | .. or at least, not reasonable to expect it to actually work, huh |
| 21:34.07 | Trevor_b | Dossy_: no, just messy and unreliable |
| 21:34.24 | JerJer | Dossy_: using T.38 its nearly 100% rock solid |
| 21:34.40 | Trevor_b | Although makes me wonder about TDMoE, but that requires local systems or ethernet bridging. |
| 21:34.55 | outtolunc | haah oops |
| 21:34.57 | JerJer | outtolunc: don't we want the result the other way - itf it is null make it a zero |
| 21:35.02 | JerJer | if not use the sip header |
| 21:35.16 | outtolunc | _X.,n,Set(intended_context=${IF(${ISNULL(${SIP_HEADER(X-Intended-Context)})}?SIP_HEADER(X-Intended-Context):0)}) |
| 21:35.16 | *** join/#asterisk khronos (n=khronos@c-66-229-159-201.hsd1.fl.comcast.net) |
| 21:35.30 | outtolunc | i had a function wrapped in ${} |
| 21:35.42 | _x86_ | found some cool stuff here |
| 21:35.43 | _x86_ | http://dataprobe.com/products/switch/sw_rack/index.html |
| 21:35.46 | _x86_ | Trevor_b: check that out |
| 21:35.49 | outtolunc | actually that is still wrong |
| 21:36.04 | Dossy_ | JerJer: so, what do I need to do to implement T.38? What's involved? |
| 21:36.09 | _x86_ | Trevor_b: these crazy guys make rack shelves the size of a giant DAX that can failover entire DS-3's! |
| 21:36.25 | _x86_ | like, multiple DS3's at a time |
| 21:36.52 | JerJer | Dossy_: T.38 support |
| 21:37.07 | Dossy_ | t38modem? hmm |
| 21:37.21 | Trevor_b | _x86_: what they charge> |
| 21:37.21 | JerJer | thats h323 crap |
| 21:37.39 | JerJer | Your ATA needs T.38 and the provider you send calls to needs to support T.38 |
| 21:37.55 | lirakis | later all |
| 21:37.57 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 21:38.02 | m0t3jl | Trevor_b, http://www.pastebin.ca/686267 < theres the extensions.conf, but it includes two other files (from which one does not exist) and the other one is here: http://www.pastebin.ca/686268 |
| 21:38.05 | _x86_ | Trevor_b: i told them I want 6 T1's in, 12 T1's out, they told me a K-16 setup would run me ~$2,200 USD |
| 21:38.26 | _x86_ | Trevor_b: they also have a K-3 model that maxes out at 6in/12out for ~$2,000 USD |
| 21:39.15 | Trevor_b | thats really not that bad as long as its reliable. |
| 21:39.26 | *** join/#asterisk luke-jr (n=luke-jr@2002:48ce:72ec:0:20e:a6ff:fec4:4e5d) |
| 21:39.30 | _x86_ | i know man, i was expecting it to cost much, much more |
| 21:39.47 | luke-jr | I am considering converting my network to only IPv6 |
| 21:39.51 | Dossy_ | JerJer: ok, so then T.38 doesn't help me. sigh |
| 21:39.55 | Trevor_b | m0t3jl: which exten's are the 2? |
| 21:39.56 | luke-jr | any ideas on getting the PAP2 to do it? |
| 21:39.58 | khronos | What you guys think of those Redfone devices that do t1 to ehternet bridging? |
| 21:40.11 | _x86_ | and it's telco-grade... all that other crap I was finding was made, sold and supported out of the country... |
| 21:40.20 | JerJer | Dossy_: it can provided you meet those requirements |
| 21:40.25 | _x86_ | and most of them only did 1 or 2 ports! |
| 21:40.36 | m0t3jl | Trevor_b, should be 001 and 002 |
| 21:40.41 | JerJer | we support T.38 and there are ATAs out there that support T.38 |
| 21:40.52 | JerJer | like the SPA-300X support T.38 |
| 21:40.52 | Dossy_ | JerJer: which, I can't meet, so ... oh well |
| 21:41.16 | JerJer | another less reliable option is iaxmodem and hylafax |
| 21:42.59 | JerJer | its not too bad if you have local PRI - say 90% success rate |
| 21:43.06 | luke-jr | ... |
| 21:43.22 | JerJer | but i wouldn't count on relying on faxing if you have a VoIP provider - without T.38 |
| 21:43.34 | Trevor_b | m0t3jl: I see the from-did-direct portions, but I dont actually see anything that just has exten => 001,1,Dial,sip/001|20|to, and then exten => 001,2,Hangup |
| 21:43.52 | khronos | What kind of bandwidth does t.38 use? |
| 21:44.10 | Trevor_b | it seems to forward between contexts, but then i didnt look through every include file. Are you using a GUI configure system or did you get a custom dialplan file? |
| 21:44.11 | _x86_ | heh... people talk about PRI like it's the only option ;) |
| 21:44.23 | Qwell | _x86_: If you want reliable telephony - it is |
| 21:44.30 | *** join/#asterisk dug (n=chatzill@c-76-102-23-25.hsd1.ca.comcast.net) |
| 21:44.55 | Trevor_b | Qwell: How reliable is TDMoE compared to PRI, given no issues in the LAN switching etc. |
| 21:45.02 | luke-jr | JerJer: VoipJet works fine for ulaw faxing |
| 21:45.03 | Qwell | When was the last time you saw a voip provider that could claim they had even 3 9's of uptime? |
| 21:45.10 | Dossy_ | JerJer, yeah, I don't know if my VoIP provider does T.38 |
| 21:45.25 | JerJer | luke-jr: i would not bet my life on it |
| 21:45.30 | Qwell | JerJer: correct me if I'm wrong here, please... |
| 21:45.37 | luke-jr | JerJer: I would not bet my life on faxing period |
| 21:45.37 | m0t3jl | Trevor_b, there would have to be a bug or something in freepbx if those lines were not there |
| 21:45.53 | JerJer | or the fate of a important legal document via ulaw faxing over the public internet |
| 21:45.54 | Qwell | but I doubt that you would give customers any sort of guarantees over the public internet :) |
| 21:46.01 | Trevor_b | ok let me compare to my freepbx box and see how extens on that are written |
| 21:46.19 | JerJer | we will give people 0 9s of uptime using the public internet |
| 21:46.24 | Qwell | heh |
| 21:46.29 | JerJer | all day long |
| 21:46.30 | Qwell | 88% guarantee |
| 21:46.37 | Qwell | or 9 5's? |
| 21:46.38 | Trevor_b | m0t3jl: check your extensions_additional.conf for that line i mentioned. |
| 21:46.40 | Qwell | 55.5555555% |
| 21:46.44 | JerJer | more like 9 5s :) |
| 21:46.46 | Qwell | ;) |
| 21:46.54 | *** join/#asterisk dlynes (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 21:46.57 | JerJer | or 9 .5s |
| 21:47.00 | Qwell | JerJer: You should put out a press release next April |
| 21:47.11 | Qwell | I would give you so much props, it isn't even funny. |
| 21:47.19 | luke-jr | lol |
| 21:47.31 | Trevor_b | As long as you have something there it should be working from inherits, although mine looks a little weird, seems to do a Macro to VM before dial... |
| 21:47.33 | luke-jr | so about getting my PAP2 to talk IPv6? |
| 21:47.41 | m0t3jl | Trevor_b, youre right, theres nothing like that ... |
| 21:47.52 | Qwell | luke-jr: well, asterisk doesn't do ipv6, so it wouldn't do you much good |
| 21:47.58 | luke-jr | Qwell: ?!?! really |
| 21:48.02 | luke-jr | that sucks |
| 21:48.13 | Trevor_b | m0t3jl: you have 001,1,Marcro(exten-vm) or you not enable the vm for those? |
| 21:48.34 | JerJer | VoIP Provider Announces 9 5's availability |
| 21:48.40 | Qwell | JerJer: yes, totally |
| 21:48.55 | Qwell | have quotes from analysts and such... it would be awesome |
| 21:49.08 | m0t3jl | Trevor_b, I dont wanna use voicemails |
| 21:49.09 | Trevor_b | Qwell: Any comment on reliability for TDMoE compared to PRI, given no issues in the LAN switching etc. |
| 21:49.19 | Qwell | Trevor_b: no, I've never done TDMoE |
| 21:49.20 | JerJer | "we are proud to offer our 9 5's of service.", said the CTO of VoIP Provider :) |
| 21:49.22 | Trevor_b | m0t3jl: no problem, just using it for refference. |
| 21:49.23 | CoaxD | JerJer: Is that before or after they take their 800# DID's offline for 3 weeks? |
| 21:49.27 | luke-jr | :( |
| 21:49.33 | Trevor_b | m0t3jl: grep 001 extensions_additional.conf |
| 21:49.37 | Qwell | JerJer: If you need a good customer praise thing, I'd be all over it |
| 21:49.38 | Trevor_b | what do you get? |
| 21:49.52 | Qwell | "Now I can be *SURE* that my call might get through." |
| 21:50.18 | m0t3jl | Trevor_b, thats bad ... |
| 21:50.38 | luke-jr | JerJer: you're too late |
| 21:50.39 | Trevor_b | HAHAHAH |
| 21:50.41 | Trevor_b | m0t3jl: |
| 21:50.45 | luke-jr | Sellvoip already has 9 5's |
| 21:50.52 | Qwell | luke-jr: but do they GUARANTEE it? |
| 21:50.53 | Qwell | :) |
| 21:50.55 | luke-jr | or was it 9 1's? |
| 21:50.55 | m0t3jl | Trevor_b, nothing? .-) |
| 21:50.59 | luke-jr | Qwell: no |
| 21:51.08 | m0t3jl | Trevor_b, just some lines that doesnt look like thos we are looking for |
| 21:51.15 | WilliamK | where's a good place to get more hold music for Asterisk? |
| 21:51.18 | Trevor_b | m0t3jl: I think you might have stepped on international dialing or something. try changing your extens to 100 and 200 instead of 001 and 002, this is a wild guess at freepbx current dialplan conflicts ;) |
| 21:51.21 | Qwell | WilliamK: freeplaymusic.com |
| 21:51.27 | *** join/#asterisk slakware (n=attila@201.19.212.99) |
| 21:51.28 | Qwell | I think it's .com...might be .org |
| 21:51.33 | luke-jr | speaking of which, has anyone ever managed to get a refund from sellvoip? |
| 21:51.37 | WilliamK | thanks |
| 21:51.43 | Trevor_b | notice the 001-outside when you grep 001 from extens? |
| 21:51.45 | WilliamK | I'm just getting tired of listening to the same music tracks |
| 21:51.47 | WilliamK | :) |
| 21:51.52 | slakware | With 1.4.11 is it posible to use include within realtime extensions? |
| 21:51.57 | JerJer | luke-jr: where u see this at? |
| 21:51.57 | m0t3jl | Trevor_b, hm ... I tried those two combinations earlier today :-) |
| 21:52.02 | luke-jr | JerJer: ? |
| 21:52.09 | Qwell | WilliamK: what, you don't like fpm-world-mix? |
| 21:52.10 | JerJer | sellvoip 5 9's |
| 21:52.14 | JerJer | er 9 5s |
| 21:52.18 | Trevor_b | test phones were connected on the dialing into system and then still wouldnt dial between? Hmm, what freepbx you running? |
| 21:52.20 | luke-jr | JerJer: my Asterisk box |
| 21:52.26 | luke-jr | JerJer: it's always spamming me with |
| 21:52.35 | WilliamK | I've listened to it over over an hr prior :) |
| 21:52.40 | luke-jr | Sep 7 21:49:44 NOTICE[6023]: chan_sip.c:5495 sip_reg_timeout: -- Registration for '8682010000000000@sellvoip-newyork' timed out, trying again (Attempt #663) |
| 21:52.46 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 21:52.49 | JerJer | lol - Qwell is talking about a April 1st Press Release |
| 21:53.06 | luke-jr | JerJer: yeah, I was just extending it to mock sellvoip's suckiness |
| 21:53.14 | JerJer | nice |
| 21:53.22 | JerJer | that's the word of the day kids |
| 21:53.24 | JerJer | suckiness |
| 21:53.33 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 21:53.33 | *** mode/#asterisk [+o anthm] by ChanServ |
| 21:54.08 | Trevor_b | Well my first suggestion is to get off those numbers. Makes it easier to grep out extensions if they dont include components of dialplan in them. But again just my prefference. When using freepbx i NEVER start under 100, and usually around 200 for extensions. But thats just me and the few guys I know doing it, mainly dealing with older weird issues with freepbx, probably fixed now, but i never bothered to change. |
| 21:54.14 | Dossy_ | is there a way to find out what codecs are available on an IAX2? |
| 21:54.21 | Dossy_ | there's no "iax2 show codecs" :) |
| 21:54.32 | russellb | core show codecs |
| 21:54.35 | Qwell | Dossy_: codecs aren't really channel type specific |
| 21:54.44 | Qwell | except for like zap |
| 21:54.53 | Qwell | but even then, it could potentially use something else |
| 21:56.57 | Dossy_ | well, on a channel you can specify what codecs are allowed/disallowed |
| 21:57.02 | Trevor_b | also just temporarily enable VM for them, that way we can see the extens VM Macro, and see if were missing things when comparing. |
| 21:57.07 | Dossy_ | i just want to check what codecs actually did end up being allowed based on the config |
| 21:57.46 | luke-jr | iax2 debug |
| 21:57.47 | luke-jr | ? |
| 21:57.56 | Qwell | show channel should show that |
| 21:58.13 | Dossy_ | Sep 7 17:57:46 NOTICE[29386]: channel.c:1956 ast_read: Dropping incompatible voice frame on IAX2/jnctn-peer-1 of format slin since our native format has changed to g726 |
| 21:58.15 | Dossy_ | argh. wtf. |
| 21:58.25 | m0t3jl | Trevor_b, okay |
| 21:58.29 | Dossy_ | I even have "transcode_via_sln => no" in asterisk.conf |
| 21:58.33 | Dossy_ | this is so broken. |
| 21:59.03 | CoaxD | okay, rotting away at LAX sucks. |
| 21:59.58 | Nugget | Track your flight at flightaware.com! :) |
| 22:00.49 | hmmhesays | so anyone want to help me fix my linksys subscribe problem? |
| 22:02.28 | JerJer | hmmhesays: perhaps if you ask a specific question |
| 22:03.05 | m0t3jl | Trevor_b, I am gonna go to bed ..... I will see what I will think of in the morning, btw: when I deleted those 2 extensions and added one with the number 400, then grep 400 extensions.conf outputs nothing and grep 400 extensions_additional.conf outputs: http://www.pastebin.ca/686306 |
| 22:05.59 | Trevor_B|Away | m0t3jl: thats what I would expect from FreePBX with voicemail enabled. |
| 22:06.17 | Trevor_B|Away | So after that it would be to see the console output when a call is transfered to the extension. |
| 22:06.23 | Trevor_B|Away | we can try that tomorrow ;) |
| 22:07.05 | m0t3jl | Trevor_B|Away, okay ... I know that my ISP has this asterisk solution made up in his own home, so I will try to ask him as well tomorrow for help |
| 22:07.16 | m0t3jl | Trevor_B|Away, see you tomorrow :-) |
| 22:07.55 | Yourname`` | How can I logoff a manager from the CLI? |
| 22:08.37 | hmmhesays | JerJer: been all day, I get a 489 bad request when I try and make my spa 942 subscribe to a hint extension |
| 22:09.06 | *** join/#asterisk Tha_Duck (n=miep@ip50-97-173-82.adsl2.versatel.nl) |
| 22:15.14 | outtolunc | do a 'show manager connected' and get the ip and add a line to iptable to drop that traffic <G> |
| 22:15.30 | outtolunc | or |
| 22:15.50 | outtolunc | hmm i wonder if they fixed this |
| 22:15.58 | outtolunc | nevermind |
| 22:17.41 | WilliamK | are we still using mpg123 to play mp3s? |
| 22:18.09 | JerJer | WilliamK: its not the preferred method, no |
| 22:18.51 | WilliamK | what's the best way nowdays? |
| 22:19.02 | WilliamK | doesn't appear to be picking up the files I dropped into the directory |
| 22:19.38 | Yourname`` | Change the mode to files, WilliamK |
| 22:20.24 | WilliamK | already set that way |
| 22:20.31 | WilliamK | mode=files |
| 22:20.31 | WilliamK | directory=/var/lib/asterisk/moh |
| 22:20.47 | Yourname`` | moh or mohmp3? |
| 22:20.48 | *** join/#asterisk riddlebox (n=victoria@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:21.07 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 22:21.40 | WilliamK | don't have a directory mohmp3 |
| 22:21.41 | WilliamK | just moh |
| 22:21.49 | *** part/#asterisk Dossy_ (i=nobody@aolserver/dossy) |
| 22:21.56 | Yourname`` | Using sox I converted all the mp3s in mohmp3 to ulaw instead of mp3, and used the files mode.. and it worked for me. |
| 22:22.07 | WilliamK | so I tossed them into the moh directory |
| 22:22.54 | Yourname`` | I think the ones in moh are .wav? |
| 22:23.00 | WilliamK | yeah |
| 22:23.09 | Yourname`` | Try converting them. |
| 22:23.15 | Yourname`` | To ulaw if you use ulaw.. |
| 22:23.59 | Yourname`` | I use ulaw in those dirs to save bandwidth (someone named fujin showed me that). |
| 22:24.58 | Yourname`` | Either way, I think it should play mp3s. |
| 22:25.51 | WilliamK | sox a windows based converter or ? |
| 22:25.53 | WilliamK | never used it |
| 22:26.45 | Yourname`` | Not windows, no |
| 22:27.23 | Yourname`` | Try: sox -V *.wav -t ul -r 8000 -b -c 1 *.ulaw |
| 22:29.35 | Teln1100A | Hi, I am having trouble with DISA, using a call file I am calling a sip extension and passing it to DISA, I get the dial tone but am unable to dial out where could something be misconfigured |
| 22:30.08 | *** join/#asterisk GlobeTrotter (n=eric@196.40.26.98) |
| 22:30.30 | GlobeTrotter | hello,, has anyone one hear seen this error in 1.4.1.1 ?? translate.c: no samples for g729tolin |
| 22:30.41 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 22:41.40 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 22:43.10 | *** join/#asterisk kiscokid (n=ron@208.106.35.66) |
| 22:47.03 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 22:47.46 | ManxPower | Teln1100A: most of the time DISA problems are DTMF provlems |
| 22:48.04 | ManxPower | GlobeTrotter: do you have a G729 license from Digium? |
| 22:48.52 | ManxPower | I'm not sure why it would be "g729tolin" rather than "g729toslin" |
| 22:49.14 | ManxPower | Of course you would not be stupid enough to try to retype the message than copy/paste it like you should, are you? |
| 22:50.16 | ManxPower | Yourname``: Um, using ulaw will save NO bandwidth |
| 22:50.30 | ManxPower | perhaps you mean some other word like "CPU usage" |
| 22:51.11 | ManxPower | But I'm sure you didn't. Using the wrong words confuses people makes it harder for them to learn Asterisk. |
| 22:51.24 | ManxPower | And you would never do that. |
| 22:51.31 | Yourname`` | Yes, sorry I meant to say less CPU usage, heh |
| 22:52.03 | ManxPower | In a week there will be a cult of people converting all their sound files to ulaw to save bandwidth, in a weeks a detailed howto on using ulaw to save bandwidth will be on the wiki. |
| 22:52.08 | riddlebox | anyone using the grandstream HT488 with asterisk? |
| 22:52.13 | ManxPower | you have destroyed the asterisk community all by yourself. |
| 22:52.25 | ManxPower | riddlebox: not anyone sane. |
| 22:52.28 | Teln1100A | ManxPower: I have dtmfmode=rfc2833 is there something else I can check for the DISA issue? |
| 22:52.41 | ManxPower | Teln1100A: what is the dtmf mode of your provider? |
| 22:52.46 | ManxPower | they MUST MATCH |
| 22:52.49 | Teln1100A | teliax |
| 22:52.56 | Teln1100A | not sure what they use |
| 22:52.57 | ManxPower | that was not my question |
| 22:53.10 | ManxPower | they use Level3 which has chronic and unresolved DTMF issues. |
| 22:53.13 | Teln1100A | but the dtmf is being used to communicate with asterisk no? |
| 22:53.19 | riddlebox | ManxPower, you serious? |
| 22:53.25 | ManxPower | ~grandstream |
| 22:53.26 | jbot | rumour has it, grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 22:53.27 | ManxPower | ~gs |
| 22:53.28 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 22:53.35 | ManxPower | jbot knows all, riddlebox |
| 22:53.41 | ManxPower | ~phones |
| 22:53.42 | jbot | i heard phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 22:53.47 | Teln1100A | ~disa |
| 22:53.47 | jbot | i heard disa is direct inward system access. show application disa |
| 22:53.49 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 22:53.55 | riddlebox | hrmm I bought the grandstream phones and they work fine |
| 22:54.13 | Teln1100A | The 'show application' command is deprecated and will be removed in a future release. |
| 22:54.15 | Teln1100A | hmm |
| 22:54.25 | ManxPower | Teln1100A: it was replaced with another command |
| 22:54.35 | riddlebox | ManxPower, I dont use those phones in a production environment, just at home, to play with asterisk |
| 22:54.42 | kiscokid | grandstream phones just seem crummier than any other phone |
| 22:54.46 | ManxPower | riddlebox: and that is all they might be good for. |
| 22:55.08 | ManxPower | they don't even make a good door stop, they are not heavy enough. Maybe a paperweight in a light breeze. |
| 22:55.14 | riddlebox | ManxPower, for sure, the polycom phones and aastra phones look the best |
| 22:55.15 | Yourname`` | Yes, all it takes is me to destroy the world! |
| 22:55.15 | Yourname`` | :D |
| 22:55.24 | Yourname`` | Also, Teliax uses 3-4 upstream carriers. |
| 22:55.35 | Yourname`` | Not just L3. Maybe for personal users. |
| 22:56.00 | Teln1100A | well is DISA the same as dialing by picking up a phone from within the system? |
| 22:56.05 | Teln1100A | or does it need to be configured to allow that |
| 22:56.16 | Teln1100A | once you get a dialtone I mean |
| 22:57.21 | GlobeTrotter | yes i have a licence |
| 22:57.28 | GlobeTrotter | for 40 channels |
| 22:58.29 | ManxPower | Teln1100A: it is more like picking up an analog phone connected to a zap channel |
| 22:58.46 | Teln1100A | and would connect using the same route too? |
| 22:58.52 | Teln1100A | for outgoing calls |
| 22:58.55 | ManxPower | GlobeTrotter: then you really should contact Digium |
| 22:59.01 | ManxPower | Teln1100A: no. SIP changes EVERYTHING |
| 22:59.26 | Teln1100A | how so? |
| 23:00.35 | ManxPower | well, you have codecs and dtmf modes and packet loss and carrier issues |
| 23:00.47 | riddlebox | ManxPower, I wish I made the money to but a TDM card ;) I will have to save for awhile |
| 23:00.48 | ManxPower | when you are going to a provider you have no control over anything on their side. |
| 23:00.53 | Tha_Duck | Does anybody knows a free SIP provider which I can use within AsteriskNow? I only want to test some stuff at home, I don't need any freaking stuff |
| 23:01.13 | Yourname`` | lol what's freaking stuff? |
| 23:01.18 | ManxPower | Tha_Duck: you are a cheap bastard. voip accounts for most providers start at $10 and no contract |
| 23:01.27 | ManxPower | so spend the damn money |
| 23:01.36 | Teln1100A | make toll free calls if you have to |
| 23:01.46 | Teln1100A | they only charge for connection 2 cents |
| 23:01.56 | ManxPower | Yourname``: he's using AsteriskNow and asking for help here. That already tells you everything you need to know. |
| 23:02.24 | Yourname`` | LOL |
| 23:02.28 | Tha_Duck | I don't even want to dial to land lines or something. I just want to connect the stuff |
| 23:02.39 | ManxPower | Teln1100A: have you used the same Teliax connection to call into an IVR and confirm that DTMF works that way? |
| 23:02.47 | Yourname`` | Tha_Duck: http://fwd.pulver.com |
| 23:03.23 | Yourname`` | I used *now when it was in Beta3, couldn't get it to work.. and came to this channel. Oh, my life got changed after I came here. |
| 23:03.52 | Tha_Duck | Ah, then it's worth trying I think :) |
| 23:04.12 | Tha_Duck | The problem is, we are using @Com at our company since last week. i want to learn more about the Asterisk system |
| 23:04.14 | ManxPower | Yourname``: We told you that you are a moron for using AsteriskNow and told to get your ass out of this channel and over to #asterisknow. |
| 23:04.15 | ManxPower | right? |
| 23:04.27 | Tha_Duck | that is why I don't need any freaking stuff like dialing for free to land lines |
| 23:04.34 | ManxPower | never heard of @Com |
| 23:04.34 | Teln1100A | ManxPower: heres the call file I am trying to use : http://pastebin.com/m4d2a1df3 |
| 23:04.45 | Yourname`` | Nope. I got welcomed because I came here AFTER I ditched asterisknow! See, I'm smart. |
| 23:04.48 | ManxPower | Teln1100A: stop using call files. make it work just using a phone on the system to dialout |
| 23:04.49 | Tha_Duck | @Com is just a GUI. Developed by Trends Telematics in the Netherlands |
| 23:05.00 | ManxPower | we don't help with guis here. |
| 23:05.04 | Teln1100A | well I can dial out |
| 23:05.09 | Teln1100A | using phone |
| 23:05.15 | ManxPower | Teln1100A: and the far end recognizes your dtmf? |
| 23:05.19 | Teln1100A | the DISA is not working |
| 23:05.32 | ManxPower | Teln1100A: disa is working just fine. |
| 23:05.35 | ManxPower | DTMF is not working., |
| 23:05.44 | Teln1100A | how can I fix DTMF then |
| 23:05.54 | ManxPower | now stop obsessing on .call files and DISA and start obsessing about DTMF. |
| 23:06.02 | ManxPower | Teln1100A: you still have not answered my question. |
| 23:06.33 | Teln1100A | well if I call an automated system I can use my touch pad so yes DTMF is being recognized |
| 23:06.43 | ManxPower | My question is: when you call out your teliax conneciton to a phone number with a voice menu, does the far side recognise the tones you send to select options? |
| 23:06.50 | ManxPower | Teln1100A: consistantly? |
| 23:06.50 | Teln1100A | yes |
| 23:06.51 | Teln1100A | it does |
| 23:06.51 | Yourname`` | Teln1100A: I found using auto or inband with Teliax works fine. |
| 23:07.10 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 23:07.13 | ManxPower | Yourname``: as you know inband DTMF only works if the codec for the call is ulaw or alaw |
| 23:07.24 | Yourname`` | ManxPower: Teliax uses ulaw. |
| 23:07.28 | Teln1100A | codec is ulaw |
| 23:07.39 | ManxPower | Teln1100A: the just set it to inband and be done with it |
| 23:08.06 | Teln1100A | this would be in sip.conf? |
| 23:08.23 | ManxPower | of course |
| 23:08.30 | ManxPower | in the section for teliax |
| 23:08.33 | ManxPower | either friend or peer |
| 23:09.33 | Teln1100A | I am using IAX though |
| 23:09.39 | Teln1100A | same thing? |
| 23:09.45 | Teln1100A | for the trunk |
| 23:09.55 | *** join/#asterisk Lawbringer (n=Lawbring@212.183.134.208) |
| 23:10.33 | ManxPower | no, there is no such thing as a dtmf mode in iax |
| 23:10.57 | ManxPower | if you are using IAX then why did your call file say SIP/USER |
| 23:11.04 | ManxPower | and why is your teliax username USER? |
| 23:11.19 | Teln1100A | I changed that for pasting |
| 23:11.19 | ManxPower | if you don't give complete information we can't give complete answers. |
| 23:11.27 | Teln1100A | sip is my internal phone |
| 23:11.31 | Teln1100A | teliax is trunk |
| 23:11.34 | Teln1100A | trunk uses iax |
| 23:11.35 | ManxPower | Teln1100A: the only thing you should remove is passwords |
| 23:11.41 | Teln1100A | k |
| 23:11.59 | ManxPower | Now I have NO idea what you are doing. |
| 23:12.20 | ManxPower | I thought you generated a call file to initate a call to a phone somewhere out there then run DISA. |
| 23:12.33 | ManxPower | now you are saying that the destination is your SIP phone? |
| 23:12.33 | Teln1100A | for testing I am using sip internal phone |
| 23:12.38 | ManxPower | What SIP phone are using? |
| 23:12.44 | ManxPower | BRAND and MODEL |
| 23:12.46 | Teln1100A | that will be changed to external pstn phone |
| 23:12.50 | Teln1100A | softphone xlite |
| 23:13.13 | ManxPower | then your DTMF mode should be rfc2833 for the xlite user/friend/peer |
| 23:13.21 | Teln1100A | it is |
| 23:13.41 | Teln1100A | the softphone can dial like a voicemail and pass through touch tones |
| 23:14.33 | ManxPower | switch to using SIP for Teliax, set the dtmfmode to inband for the teliax entry in sip.conf |
| 23:14.33 | Teln1100A | but when listening to dialtone of DISA press any key and it switches to buy and hangs up |
| 23:14.39 | ManxPower | Oh! |
| 23:14.49 | ManxPower | well, it is recognizing the DTMF then, isn't it. |
| 23:14.53 | Teln1100A | yea it is |
| 23:14.56 | ManxPower | what exactly are you trying to dial? |
| 23:15.03 | Teln1100A | a pstn number |
| 23:15.09 | ManxPower | and I mean the number, not some fake hidden stuff. |
| 23:15.24 | ManxPower | You can leave off the last 4 digits if you are that aparnoid |
| 23:15.35 | Teln1100A | 416655XXXX |
| 23:15.51 | Teln1100A | which works fine if I call directly from the softphone |
| 23:15.51 | ManxPower | and do you have a matching exten line in [local] in extensions.conf? |
| 23:16.42 | Teln1100A | exten => _X.,1,DIAL(IAX2/USER@teliax/${EXTEN},30,tr) |
| 23:16.45 | Teln1100A | yea |
| 23:16.56 | Teln1100A | that should match everything right? |
| 23:17.05 | ManxPower | other than your excessive caps, that should work. |
| 23:17.15 | ManxPower | is that a PASTE or a retype? |
| 23:17.25 | Teln1100A | paste other than USER for my username |
| 23:17.39 | ManxPower | and what messages appear on the console when DISA fails |
| 23:17.58 | ManxPower | btw, remove the ,tr |
| 23:18.36 | Teln1100A | http://pastebin.com/m3d7197af |
| 23:18.52 | Teln1100A | what does the tr mean |
| 23:19.00 | Teln1100A | I copied it from the teliax sample config |
| 23:22.47 | *** join/#asterisk Op3r (n=Op3r@125.212.122.175) |
| 23:26.23 | ManxPower | tr means "I rode the short bus to the Asterisk school and I use options I don't understand" |
| 23:26.34 | ManxPower | I suggested you remove it to keep people from laughing at you. |
| 23:26.44 | Teln1100A | very well |
| 23:33.50 | Trevor_B|Away | ManxPower: Hmm thought it was T teliax said to use, not t. But then its been long since i looked through their support site. |
| 23:34.03 | Trevor_B|Away | wouldnt T be viable where t is liable? |
| 23:34.16 | Teln1100A | say what |
| 23:34.17 | Teln1100A | ? |
| 23:34.56 | Trevor_B|Away | T being calling party (you) can still transfer, where t is called party can transfer/ |
| 23:35.51 | Teln1100A | can both be used like Tt |
| 23:35.55 | Teln1100A | so both can transfer? |
| 23:35.56 | Trevor_B|Away | i beleive |
| 23:35.59 | ManxPower | so if you call someone with Dial(whatever,,t) then the person you are calling can use * to transfer themselves to 900 number sex lines on your dime. |
| 23:36.14 | ManxPower | If that is what you want then more power to you -- but don't use an option you do not understand |
| 23:36.16 | Trevor_B|Away | right t is a liability. |
| 23:36.25 | Teln1100A | now I get it |
| 23:36.28 | ManxPower | both of them are unless you are VERY careful. |
| 23:36.52 | ManxPower | since virtually all IP phones and all hardphone are able to transfer calls using a transfer or flash button, t/T is not needed for most people |
| 23:36.53 | Trevor_B|Away | Wouldn;t you assume your employees are honorable, or is there something im not thinking of? |
| 23:37.04 | ManxPower | unless you are using something like xlite that does not support transfers |
| 23:37.21 | Teln1100A | it does to conferencing though |
| 23:37.23 | Teln1100A | 2 ppl |
| 23:37.46 | ManxPower | Trevor_B|Away: the employee might be nice, but the person you are calling might not be. |
| 23:38.24 | Trevor_B|Away | But I mean with JUST T not Tt, wouldnt only the issue be if the employee does bad transfers? |
| 23:38.40 | Trevor_B|Away | s/wouldnt only the issue/wouldnt the only issue/ |
| 23:38.50 | ManxPower | For example what is the different between these two things: You pick up your phone and dial a pstn number Dial(SIP/${EXTEN}@provider,,t) and this: Call comes in and goes to your sip phone via Dial(SIP/user,,t) |
| 23:39.45 | ManxPower | the first one is a security risk, the 2nd one is not |
| 23:39.52 | Trevor_B|Away | right but thats lower t, or am i just missing your point? Wouldnt leaving t out and using T for local side, or am I missing that you loop back with T would still do the same. |
| 23:39.53 | ManxPower | Trevor_B|Away: my point is that you almost never need t or T. |
| 23:40.10 | ManxPower | So replace the t with a T |
| 23:40.15 | Trevor_B|Away | true, my poly's handle it, i dont use it, just thinking in theoretical :) |
| 23:40.30 | ManxPower | now the 2nd one is a security risk and the first is not. |
| 23:40.38 | Trevor_B|Away | Ahh, i see your point. |
| 23:40.53 | ManxPower | since you don't need them for 90% of cases, don't use them |
| 23:41.18 | Trevor_B|Away | true, good point to double check i didnt leave them in there on clients systems ;) |
| 23:41.20 | ManxPower | Also in the first example you cannot dial * when calling IVRs like, your bank, for example |
| 23:41.36 | ManxPower | since * will start a transfer |
| 23:42.40 | ManxPower | or # or whatever your activation tone is |
| 23:43.07 | ManxPower | It's tough to accidently configure things so the person you called can press the TRANSFER button on your phone. |
| 23:43.24 | Trevor_B|Away | Yeah. Already have enough issues with ITSP's and codecs for IVR, dont need to introduce more. |
| 23:43.42 | Teln1100A | is it possible I am not passing the right arguments when using DISA in my call file? Application: disa <cr> data: no-password |
| 23:44.23 | ManxPower | Teln1100A: that is not what is in your call file. |
| 23:44.33 | ManxPower | data: no-password,local is what is in your call file |
| 23:44.47 | Teln1100A | I tried with and without same result |
| 23:45.08 | ManxPower | Teln1100A: I'm still waiting for info on what error is generated on the console when you try using DISA |
| 23:45.15 | Teln1100A | I pasted that |
| 23:45.38 | Teln1100A | http://pastebin.com/m3d7197af |
| 23:45.58 | Teln1100A | no errors really just goes to busy tone when a digit is dialed |
| 23:46.44 | ManxPower | Teln1100A: it looks like you are going to have to learn some advanced troubleshooting stuff |
| 23:47.03 | *** join/#asterisk blinky42 (n=me@c-71-230-47-244.hsd1.pa.comcast.net) |
| 23:48.05 | Teln1100A | is it possible my Disa module is not loaded? |
| 23:49.19 | WilliamK | okie, who's familiar with zaptel? I apparently have forgotten a few things... |
| 23:50.58 | Strom_M | WilliamK: what's your specific problem? |
| 23:52.09 | ManxPower | Teln1100A: if it was not loaded it would not run |
| 23:52.31 | *** join/#asterisk mog (n=mog@216-83-246-98.wan.networktel.net) |
| 23:52.31 | *** mode/#asterisk [+o mog] by ChanServ |
| 23:52.57 | Teln1100A | could it be it is restricted to allow only dialing after password authentication |
| 23:53.02 | WilliamK | zaptel can't find the TE120P |
| 23:53.12 | WilliamK | Notice: Configuration file is /etc/zaptel.conf |
| 23:53.12 | WilliamK | line 0: Unable to open master device '/dev/zap/ctl' |
| 23:53.12 | WilliamK | 1 error(s) detected |
| 23:53.16 | Strom_M | WilliamK: are you loading the right drivers? |
| 23:53.19 | Strom_M | is the card properly seated? |
| 23:53.24 | Strom_M | is the PCI bus turned on? |
| 23:53.31 | ManxPower | WilliamK: lsmod shows the kernel module loaded for that card |
| 23:53.40 | WilliamK | prior install, I just updated |
| 23:53.45 | WilliamK | even tried going backwards to stable |
| 23:53.58 | Strom_M | "backwards to stable"? |
| 23:54.29 | ManxPower | what happens when you modprobe wcte12xp |
| 23:55.26 | WilliamK | ZT_SPANCONFIG failed on span 1: Invalid argument (22) |
| 23:55.27 | WilliamK | FATAL: Error running install command for wcte12xp |
| 23:56.21 | ManxPower | Well it sucks to be you |
| 23:56.31 | WilliamK | Manx, thanks :) |
| 23:56.44 | Strom_M | WilliamK: does the card show up when you run lspci -bv ? |
| 23:57.12 | WilliamK | yep |
| 23:57.39 | *** join/#asterisk hal14450 (n=john@cpe-72-226-225-229.rochester.res.rr.com) |
| 23:58.07 | Strom_M | pastebin your zaptel.conf |
| 23:58.20 | WilliamK | only 3 lines |
| 23:58.44 | Strom_M | ok, fine, paste it here |
| 23:58.49 | WilliamK | span=1,1,1,d4,ami |
| 23:58.49 | WilliamK | e&m=1-24 |
| 23:58.49 | WilliamK | loadzone = us |
| 23:59.31 | Strom_M | type lsmod | grep zap |
| 23:59.52 | WilliamK | 2 lines |