| 00:03.44 | Janos | Chainsaw: getting the same behavior on a different channel so i'm guessing there is a problem with the line |
| 00:04.00 | Janos | Chainsaw: ty for your help |
| 00:04.05 | Chainsaw | Janos: Agreed, time to contact your telco. |
| 00:04.46 | Chainsaw | Janos: Tell them you see no battery on their analog channel. |
| 00:04.50 | Chainsaw | Janos: You're welcome. |
| 00:05.10 | Janos | Chainsaw: ahh ok that's good to know, will do |
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| 01:40.30 | maxagaz | leifmadsen: thanks! |
| 01:40.37 | leifmadsen | :) |
| 01:40.43 | leifmadsen | that was some high latency :) |
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| 02:13.20 | ghostmediapro | help setting static ip in PIAF |
| 02:13.52 | ghostmediapro | using pbx-menu netconfig, doesn't seem to hold ip |
| 02:13.53 | maxagaz | leifmadsen: yes, I went to sleep in between :) |
| 02:14.01 | leifmadsen | oic :) |
| 02:14.16 | maxagaz | leifmadsen: the doc you gave me is the more complete I've read |
| 02:14.26 | leifmadsen | which one? |
| 02:14.37 | leifmadsen | the ofps link or the one on the wiki? |
| 02:14.53 | leifmadsen | (both are quite new, one being about 3 weeks old, the other about 3 hours :)) |
| 02:15.07 | maxagaz | leifmadsen: this one https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial |
| 02:15.25 | maxagaz | leifmadsen: did you do it ? |
| 02:15.37 | leifmadsen | that was Malcolm Davenport |
| 02:15.48 | leifmadsen | there is some documentation on setting it up at OFPS as well |
| 02:15.55 | leifmadsen | infobot: tell maxagaz about newbook |
| 02:15.58 | leifmadsen | see msg |
| 02:17.11 | ghostmediapro | tring to configure static ip in PIAF but using pbx-menu netconfig option doesn't seem to hold ip |
| 02:17.16 | maxagaz | leifmadsen: just one point remains unclear, where should I put the user pem file (using zoiper biz on a windows machine), the ca.crt has been automatically handled by windows |
| 02:17.33 | leifmadsen | I can't tell you as I haven't set either SRTP or TLS up |
| 02:21.09 | maxagaz | leifmadsen: thanks very much for the book too, that will help me *a lot* :-) |
| 02:21.24 | leifmadsen | no worries! let us know if the book description is missing significant |
| 02:21.50 | leifmadsen | then we can get it fixed prior to print (which it is going for in the middle of February. Feb 14th is the last possible day to make changes) |
| 02:23.52 | *** join/#asterisk shapr (~shapr@nat/digium/x-ynafsjdnxichlced) |
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| 02:24.47 | p3nguin | ghostmediapro: It is recommended to change the IP address using webmin rather than netconfig. |
| 02:26.01 | p3nguin | ghostmediapro: It is also said that if you choose to use netconfig anyway, you'll have to change the IP address more than just once. |
| 02:27.16 | p3nguin | The reason for this is because PiaF is a flaming pile of crap and most of the tools they want to use are not implemented well. |
| 02:28.18 | p3nguin | If you want to get a PBX running quickly and reliably, use AsteriskNOW rather than PiaF. |
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| 02:39.10 | drmessano | pbxpilaf |
| 02:39.21 | drmessano | Need to add that one |
| 02:39.23 | drmessano | ~pbxpilaf |
| 02:39.51 | drmessano | ~pbxpilaf |
| 02:39.51 | infobot | pbxpilaf asterisk + freepbx + x11 + gnome/kde + mythtv + bittorrent + mp3 jukebox + other various programs, season to taste |
| 02:40.05 | p3nguin | Not as nutritious as rice pilaf, that's for sure. |
| 02:40.13 | rizwank | Hi there. It's a bit off-topic, but I've just been elected to become the GSM/wireless expert in our calling card company. I need a somewhat technical understanding of GSM - single and dual IMSI; roaming agreements; what sort of applications and call forwarding can be done on a SIM card; etc. Can anyone recommend a solid resource that covers the bases - a good book (preferable) or website? |
| 02:40.36 | *** join/#asterisk chopp (~chopp@unaffiliated/chopp) |
| 02:41.06 | drmessano | rizwank, we're all just gonna google for something and post links |
| 02:41.48 | rizwank | I've done so also, most of what I've found on amazon were books around 2005 --- might still be useful, but I thought and ask just in case someone had a preferred source. |
| 02:44.21 | rizwank | Any thoughts on IRC channels that might have references? |
| 02:45.19 | WIMPy | rizwank: The standard one is "Global System for Mobile Communication" byy Mouly and Pautet. |
| 02:45.58 | drmessano | rizwank, perhaps something connected with the OpenBTS project |
| 02:45.59 | WIMPy | Or you take a look at projects like OpenBSC, OpenBTS and osmocom-bb. |
| 02:46.24 | rizwank | thanks - great starting points. |
| 02:46.43 | rizwank | SIM card applications are technologically distinct from GSM; so I'd probably have to look elsewhere for information on those -- right? |
| 02:46.49 | WIMPy | And the ITU, off course. |
| 02:47.01 | drmessano | rizwank, maybe, maybe not.. |
| 02:47.06 | WIMPy | There is the GSM association as well. |
| 02:47.24 | drmessano | rizwank, I would think anyone implementing GSM in software would have knowledge of the whole stack |
| 02:47.37 | WIMPy | They should be a good starting point for SIP apps. |
| 02:47.58 | WIMPy | SIM |
| 02:49.26 | rizwank | Thanks. |
| 02:50.06 | rizwank | Headed out to GSMA in Feb to discuss possible new businesses. We had glazed over eyes in our meetings at PTC about dual-IMSI SIMs and realized some schooling was necessary. |
| 02:51.18 | WIMPy | Maybe you should look at the sim changers that are available? |
| 02:51.44 | rizwank | SIM changers? |
| 02:52.22 | WIMPy | Devices that you put in to you phone instead of the SIM and that itself takes 2 or 3 SIMs. |
| 02:53.14 | WIMPy | But what is your target application for a dual IMSI SIM? |
| 02:54.01 | rizwank | Global roaming reach. |
| 02:54.38 | WIMPy | How? |
| 02:55.22 | rizwank | that's what we're trying to understand, but best I could extract - we'd have a US provider associated with the primary IMSI and the second would be with a partner (for instance, Belgacom) - that has it's own roaming agreements. |
| 02:55.40 | rizwank | Application : Either making (yet another) World SIM, or providing dual-national based card for our primary customers - work in your home country as well as US. |
| 02:56.39 | WIMPy | But the SIM is given to the customer by the operator, so gettinge two different operators to hand out one card together seems unlikely to happen. |
| 02:57.15 | WIMPy | And if they'd collaboate a 2nd IMSI wouldn't even be neccessary. |
| 03:00.17 | rizwank | We'd be partnering with an operator in the states (an MVNO) and belgacom offers this as a service ... |
| 03:00.25 | rizwank | but yeah, as I said, there's a lot I don't understand. |
| 03:01.03 | WIMPy | It's surely a lot less flexible than a changer that works with any combination of SIMs. |
| 03:01.32 | rizwank | changer = phone that can handle multiple sims? |
| 03:02.07 | WIMPy | Those are rare. I was thinking of the adaptors as described above. |
| 03:02.31 | WIMPy | But they won't fit all phones, either off course. |
| 03:02.38 | rizwank | its something new to me .. "SIM Changers" should be enough to check for on google? |
| 03:02.57 | WIMPy | Guess so. |
| 03:04.11 | WIMPy | Better try "dual sim adapter" |
| 03:04.21 | rizwank | ok, thanks. |
| 03:05.06 | WIMPy | stands corrected. They seem to be availabe in SIM format by now. |
| 03:05.15 | WIMPy | That should make them a one fits all. |
| 03:08.45 | rizwank | interesting tech. |
| 03:08.56 | rizwank | not really something we can sell retail, but still quite interesting. |
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| 03:13.57 | leifmadsen | nite nite |
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| 03:37.26 | ghostmediapro | Please enable manager access. |
| 03:37.26 | ghostmediapro | Most often you should be able to do that by setting 'enable = yes' in manager.conf under the [general] context, and reloading asterisk. |
| 03:37.40 | ghostmediapro | error |
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| 03:56.39 | shapr | Has anyone tried to do TDM over Ethernet? |
| 04:09.35 | drmessano | I am sure some have.. You need it for those redfone boxes |
| 04:09.38 | drmessano | ~ask |
| 04:09.38 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 04:10.02 | ghostmediapro | asterisk-gui alert Please enable manager access. |
| 04:16.32 | maxagaz | I have a strange problem, with zoiper, I have the sound in the earphone about 25 seconds after the call begin, does someone know how to fix it ? |
| 04:17.31 | *** join/#asterisk Faiz (~mr.s4g3@c-68-44-94-177.hsd1.nj.comcast.net) |
| 04:17.40 | Faiz | hi |
| 04:27.39 | ghostmediapro | help |
| 04:48.12 | ketema | does anyone have a cisco 7970 actually working via SIP with asterisk ? and if so would they share their xml config file ? |
| 04:48.29 | shapr | ghostmediapro: What? |
| 04:50.21 | ghostmediapro | i get the error when i got the the http://localhost:8088 alert please enable manager access, which is has been enabled already |
| 04:50.24 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 04:51.21 | ghostmediapro | everything is enable but i still get please enable manager access when i visit asterisk-gui web portal |
| 04:51.31 | shapr | ghostmediapro: Did you check that manager.conf has been changed, saved, and that asterisk has been reloaded? |
| 04:51.46 | ghostmediapro | yes |
| 04:51.58 | shapr | Do you see any errors in the asterisk logs? |
| 04:52.54 | ghostmediapro | just getting back into asterisk, don't remeber command to view log |
| 04:53.51 | shapr | look in /var/log/asterisk/ |
| 04:56.16 | ghostmediapro | ok how to open file to view |
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| 05:36.53 | atan | ghostmediapro, nano, vi, cat... tail... pick your poison |
| 05:45.26 | p3nguin | less, more, most |
| 05:48.38 | *** part/#asterisk Faiz (~mr.s4g3@c-68-44-94-177.hsd1.nj.comcast.net) |
| 05:48.39 | drmessano | notepad.exe works too |
| 05:48.43 | drmessano | so does winword.exe |
| 05:49.03 | drmessano | lol |
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| 05:58.22 | shapr | g'day ChrisInSydney |
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| 06:33.01 | dandate2 | my pbx has 70% disk space usage, im not recording anything, i believe this might have to do with not upgrading queues with the latest patch, as per the update logs it said something about ever-increasing file, anyway to confirm? |
| 06:33.46 | carrar | check your logs and filesystem |
| 06:36.04 | dandate2 | im not that keen on linux, any simple command to scan the hd for large files like windows search feature |
| 06:36.20 | carrar | start in root |
| 06:36.21 | carrar | cd / |
| 06:36.23 | carrar | du -d 1 `ls -1` |
| 06:38.08 | dandate2 | is that a one or L |
| 06:38.28 | dandate2 | 1 l |
| 06:38.31 | dandate2 | k |
| 06:39.25 | dandate2 | says invalid option -d |
| 06:40.43 | carrar | du --max-depth=1 `ls -1` |
| 06:42.04 | dandate2 | du 'cannot access directory: ls-1 no such directory |
| 06:42.51 | ghostmediapro | Asterisk Manager Connection Failure Failed to connect to the Asterisk manager through port: 5038 |
| 06:51.06 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-eqproiklvzhbvoam) |
| 07:06.01 | ghostmediapro | Asterisk Manager Connection Failure Failed to connect to the Asterisk manager through port: 5038 |
| 07:06.28 | dandate2 | ok i found the 11 gig file with filezilla file search, its /var/log/asterisk/full , should u just delete this or will that wreck my sstem? |
| 07:07.48 | shapr | ghostmediapro: Dude, spamming your error messages won't help. |
| 07:08.05 | shapr | dandate2: deleting that file will not wreck your system |
| 07:13.52 | ghostmediapro | sharpr: not spamming, it was atleast 30 min, but srry just tring to get a little help as i can't find a fix using google |
| 07:18.07 | dandate2 | well i deleted it, but cannot access GUI anymore =( |
| 07:18.41 | dandate2 | freepbx GUI is tming out on load, back on old systems i would do restart httpd service or something like that but cant remember the command |
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| 07:22.57 | dandate2 | well i gues its all fixed now |
| 07:23.46 | ghostmediapro | has anyone encounter this error Failed to connect to the Asterisk manager through port: 5038 please shed some lite |
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| 07:28.51 | maxagaz | why do I have to wait about 25 sec with zoiper (on linux and windows) to have sound in the earphones ? |
| 07:29.04 | maxagaz | (it works fine with other sip softwares) |
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| 07:29.43 | Daejeo | is there any FREE sip/iax client for iphone/ipod touch? |
| 07:32.22 | Daejeo | :) |
| 07:41.17 | wdoekes2 | ghostmediapro: it is not asterisk that is printing that error, is it? the tool that tries to connect to asterisk fails. there can be a variety of reasons for that: wrong IP, wrong port, asterisk is not listening (/etc/asterisk/manager.conf), the connecting IP is blacklisted, and so on.. |
| 07:45.17 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 07:49.39 | ghostmediapro | wdoekes2: would you like me to paste bi my manager.conf |
| 07:52.03 | wdoekes2 | begin with doing "netstat -apn | grep 5038" |
| 07:53.15 | ChannelZ | your manager.conf needs to be enabled = yes and you have some sort of user setup the same that your GUI is trying to login with |
| 07:54.00 | ChannelZ | and then make sure manager is even being loaded, easy way is to try 'help manager' on the console to see |
| 07:57.38 | ghostmediapro | manager.conf http://pastebin.com/0e45gRVk |
| 08:00.05 | ChannelZ | and the rest? what is your GUI trying to login into/as? Is manager running? |
| 08:00.11 | ghostmediapro | help manager http://pastebin.com/4j1p6WY0 |
| 08:01.04 | ghostmediapro | pbx*CLI> manager show connected |
| 08:01.05 | ghostmediapro | <PROTECTED> |
| 08:02.37 | wdoekes2 | There are no manager users. |
| 08:03.01 | ghostmediapro | then that most mean the the manager.conf isn't loading |
| 08:03.14 | ghostmediapro | correct me if i'm wrong |
| 08:04.13 | wdoekes2 | possibly because it chokes on some lines |
| 08:04.20 | wdoekes2 | like permit=192.168.1.1-255/255.255.255.0 |
| 08:04.33 | wdoekes2 | I don't believe that's valid |
| 08:04.47 | wdoekes2 | replace 1-255 with 0 |
| 08:06.10 | ghostmediapro | done |
| 08:06.37 | ChannelZ | reload |
| 08:07.25 | ghostmediapro | reload done |
| 08:07.33 | ghostmediapro | still no user are shown |
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| 08:09.43 | wdoekes2 | start asterisk with asterisk -c and look at the errors and warnings.. you will probably see some around the manager.conf loading |
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| 08:11.20 | ChannelZ | look at manager_additional.conf and manager_custom.conf too, if there's something busted in those it might be breaking the parse |
| 08:11.45 | ChannelZ | (and you replaced all of your multiple permit lines yes?) |
| 08:11.46 | ghostmediapro | this was a fresh install how could so much go wrong |
| 08:11.59 | ghostmediapro | yes |
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| 08:12.08 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 08:13.31 | ChannelZ | remove 'displaysystemname' and 'enable' |
| 08:14.22 | ChannelZ | those aren't valid. It's "enabled" (which you also have) and I'm not sure what displaysystemname is supposed to do (old version option?) |
| 08:15.04 | ChannelZ | I'm assuming this being a 'fresh' install it's probably Asterisk 1.6 or 1.8? |
| 08:15.30 | ghostmediapro | no errors on asterisk -c |
| 08:16.35 | ghostmediapro | Asterisk Source Version = 1.4.21.2 |
| 08:16.55 | Corydon76-home | Old |
| 08:17.03 | ChannelZ | not-so-fresh |
| 08:17.11 | ghostmediapro | piaf |
| 08:17.15 | ghostmediapro | centos |
| 08:17.23 | ChannelZ | core set verbose 3 |
| 08:17.25 | ChannelZ | manager reload |
| 08:17.47 | ChannelZ | does it complain of anything? |
| 08:18.03 | ghostmediapro | lost are those entered in the cli as well |
| 08:18.25 | ChannelZ | yes |
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| 08:21.44 | ghostmediapro | Verbosity is at least 3 and manager reload does nothing |
| 08:22.11 | ChannelZ | hmm. well do be honest I'm not sure what it should say under 1.4 anymore |
| 08:22.49 | ghostmediapro | would everything else work if i just ran a update on asterisk |
| 08:22.59 | ghostmediapro | to latest version |
| 08:23.51 | ChannelZ | ? the GUI on top might not know what's going on unless it's updated too, but I don't use them so I couldn't really say for sure |
| 08:24.23 | ChannelZ | And some config options have changed in the meantime so some of your existing configs might not work, depending |
| 08:26.12 | ghostmediapro | another fresh and still, what would you suggest with a gui |
| 08:26.39 | ChannelZ | well I don't personally suggest a GUI at all, but I guess FreePBX is about it. |
| 08:27.39 | *** join/#asterisk gerhard7 (~gerhard7@212-123-146-122.ip.telfort.nl) |
| 08:28.12 | ghostmediapro | i'm new so gui helps me alot, i can to some commands from cli or terminal |
| 08:28.40 | ghostmediapro | does lates version have asterisk 1.6 |
| 08:31.18 | ghostmediapro | whats distro piaf or asterisk now |
| 08:32.28 | ChannelZ | hmm I'm not sure if AsteriskNOW was updated to 1.8 or if it's still in 1.6 |
| 08:33.34 | ghostmediapro | so asterisk now is your recommendation |
| 08:34.52 | ChannelZ | if you want a whole Linux distro with everything in it, I suppose yes |
| 08:35.37 | ChannelZ | if you just want to fart around with a PBX, it's probably fine. But if you want to really learn Asterisk, as I said I prefer no GUI |
| 08:38.22 | ghostmediapro | its been about two years, i did get one up and running with irc help using google voice, i loved it |
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| 09:09.37 | saxa | hi I see in sip.conf example that grandstream has dtmfmode=info but in the grandstream manual I see it uses rfc2833, so, which setting is correct ? |
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| 09:23.47 | Dovid | saxa: What ever your phone has |
| 09:23.57 | Dovid | you can use either or. I like INFO better since it will show in a sip debug |
| 09:26.20 | saxa | Dovid: thx, I just read that GXP2120 uses rfc2833 but I saw that dtmfmode=info is in sip.conf so it was confusing, on which one is correct |
| 09:26.32 | saxa | Dovid: if both are equal then it should be no problem |
| 09:27.40 | *** join/#asterisk diemos (~root@173-13-138-49-sfba.hfc.comcastbusiness.net) |
| 09:27.55 | diemos | just like to point out, compiling asterisk on a 300MHz machine blows. |
| 09:27.57 | diemos | that is all. |
| 09:33.09 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 09:45.30 | *** join/#asterisk Tim_Toady (~moi@178.128.134.192.dsl.dyn.forthnet.gr) |
| 09:54.39 | maxagaz | how to have the maximum of logs in /var/log/asterisk/messages ? |
| 09:55.44 | diemos | maxagaz: can you better rephrase your question? I understand English might not be your first language, its hard to understand you. |
| 09:58.27 | maxagaz | diemos: I have some problems with my server (no sound with zoipper in my earphones for the first 25 seconds of a call), but I can't see anything in asterisk logs about this, so I think I need a higher level's logs |
| 10:00.33 | *** join/#asterisk joobie (~joobie@CPE-121-214-141-146.lnse3.lon.bigpond.net.au) |
| 10:01.01 | diemos | maxagaz: you can read about Asterisk and it's different log levels here: http://www.automated.it/asterisk/lah-3-6-05_5.html |
| 10:07.00 | tzafrir_laptop | maxagaz, look into logrotate |
| 10:24.29 | *** join/#asterisk ccesario (~ccesario@linux.unialco.com.br) |
| 10:30.28 | *** part/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 10:33.00 | *** join/#asterisk tasca (~tasca@189.4.104.162) |
| 10:38.42 | stix | Can I have two dial patterns on the same line? Separate them somehow. |
| 10:38.43 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 10:47.10 | mzb | anyone on who's familiar with the AT-command sequence processed by chan_mobile? |
| 10:47.23 | *** join/#asterisk ruyo (~psantos@a83-132-152-91.cpe.netcabo.pt) |
| 10:47.45 | mzb | I'm trying to write a script that rings and sends CLID to a headset, but I can't seem to get the CLID to work |
| 10:48.37 | *** join/#asterisk maxagaz (dce7256a@gateway/web/freenode/ip.220.231.37.106) |
| 10:51.23 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85cc.bcn.adamo.es) |
| 10:52.19 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 10:54.34 | maxagaz | nobody has an idea why I don't have sound with zoiper the first 25 seconds of each call ? I can't see anything in the log, even in debug mode. |
| 10:55.01 | maxagaz | the microphone works, but not the earphone |
| 11:05.18 | *** join/#asterisk braunr (~rbraun@dalaran.sceen.net) |
| 11:05.22 | braunr | hello |
| 11:05.38 | diemos | hi |
| 11:05.43 | braunr | does someone know why asterisk is linked with the --version-script option ? |
| 11:06.00 | braunr | AIUI, it's an option intended for shared objects, not executables, right ? |
| 11:06.25 | braunr | (this is when building under Linux, with GNU ld, ofc) |
| 11:06.49 | *** join/#asterisk E-bola (~bola@188.120.76.228) |
| 11:10.34 | *** join/#asterisk m_tadeu (~quassel@89-180-176-52.net.novis.pt) |
| 11:23.38 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 11:25.29 | *** join/#asterisk iulhk (~iulhk@117.102.32.200) |
| 11:25.34 | iulhk | hi all |
| 11:35.13 | E-bola | hi |
| 11:46.27 | iulhk | i am trying to test video call, in extensions.conf, do i need to mention more with this line .. "exten => _X.,1,Dial(SIP/${EXTEN}@${EXTEN})"? |
| 11:48.49 | Tozz_ | no |
| 11:48.54 | Tozz_ | but you have to enable video support in sip.conf |
| 11:50.37 | wdoekes2 | braunr: does it break anything? |
| 11:57.54 | *** join/#asterisk Dovid (~Dovid@213.8.118.62) |
| 12:02.58 | *** join/#asterisk ccesario (~ccesario@189.20.219.10) |
| 12:05.56 | iulhk | <Tozz_>: i hv enabled videosupport=yes .. and then hv made two peers 100 and 200 ,, i hv allowed h261, h263 , but still not getting video, ? and idea ? |
| 12:20.54 | *** join/#asterisk jkroon (~jkroon@dsl-240-212-27.telkomadsl.co.za) |
| 12:25.09 | braunr | wdoekes2: on mips, it creates a dynamic section MIPS_RLD_MAP of size 0 |
| 12:25.50 | braunr | wdoekes2: and, because the linker doesn't consistently check the size, it doesn't allocate such a section, but still references it later |
| 12:26.06 | braunr | wdoekes2: which makes the process segfault on startup |
| 12:26.24 | braunr | i have fixed our linker, and found a patch in Debian doing just the same thing |
| 12:26.45 | braunr | but still, an empty RLD_MAP section doesn't look appropriate, so I wondered |
| 12:28.45 | *** join/#asterisk beebeeep (~miga@85.94.32.98) |
| 12:29.01 | *** join/#asterisk Elwinf (d4f13482@gateway/web/freenode/ip.212.241.52.130) |
| 12:29.43 | *** part/#asterisk tasca (~tasca@189.4.104.162) |
| 12:30.05 | Elwinf | Hello. Ive got a problem with IAX2 not giving a ringbacktone to internal calls, running 1.4.31. Anyone got any tips for me? |
| 12:33.38 | beebeeep | hi folks |
| 12:33.38 | beebeeep | can anybody help me to debug prolems with xmldoc? when i starting my 1.6.2.16.1 i get warnings like: |
| 12:33.38 | beebeeep | xmldoc.c:1720 xmldoc_build_field: Couldn't find function VOLUME in XML documentation |
| 12:33.38 | beebeeep | and i cannot use core show application <whatever> |
| 12:33.38 | beebeeep | looks like all xml files are in their places |
| 12:34.32 | tzafrir_laptop | beebeeep, I suppose the docs are not loaded. Do you have the .xml docs file in place? |
| 12:34.36 | tzafrir_laptop | (installed) |
| 12:36.31 | *** join/#asterisk gerhard7 (~gerhard7@212-123-146-122.ip.telfort.nl) |
| 12:36.39 | beebeeep | tzafrir_laptop: i have many text file in /usr/share/doc/asterisk-1.6.2.16.1/ and core-en_US.xml too |
| 12:39.33 | *** join/#asterisk mzahariev (adminimini@piem-skrishno-v.unixsol.org) |
| 12:39.48 | beebeeep | btw this asterisk is installed from RPM, when i'm installing it from tarball, everything is ok. I'm just trying to understand what is wrong with my RPM and its .SPEC-file :) |
| 12:45.51 | *** join/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 12:59.45 | Daejeo | is there any FREE sip/iax client for iphone/ipod touch? |
| 13:00.54 | diemos | uh |
| 13:00.58 | diemos | i should know this |
| 13:01.09 | diemos | but I jailbroke my phone and get all my commercial apps for free :/ |
| 13:01.36 | tzafrir_laptop | beebeeep, it should be under /var/lib/asterisk/docs or somehting similar |
| 13:01.53 | tzafrir_laptop | What rpm package? By who? |
| 13:02.50 | E-bola | Can you disable loading the pbx_dundi.so if you do not use dundi? |
| 13:02.56 | E-bola | Or is some other stuff dependant on it? |
| 13:03.06 | beebeeep | tzafrir_laptop: from Axel Thimm |
| 13:03.26 | tzafrir_laptop | This warants a bug report, I guess. |
| 13:03.46 | diemos | Daejeo: try 3cx phone? |
| 13:04.15 | Daejeo | diemos: thax |
| 13:04.35 | diemos | np |
| 13:05.30 | Elwinf | Hello. Ive got a problem with IAX2 not giving a ringbacktone to internal calls, running 1.4.31. Anyone got any tips for me? |
| 13:06.21 | tzafrir_laptop | beebeeep, try: |
| 13:06.31 | tzafrir_laptop | mkdir /var/lib/asterisk/documentation |
| 13:08.16 | tzafrir_laptop | ln -s /usr/share/doc/asterisk-WHATEVERVERSION/core-en_us.xml /var/lib/asterisk/documentation/ |
| 13:08.35 | tzafrir_laptop | # fill in command to reload xmldocs. Not sure what |
| 13:09.48 | wdoekes2 | braunr: I'm not as familiar with dynamic linking as you seem to be, but I think you should file a bug report at issues.asterisk.org |
| 13:10.21 | beebeeep | yes, it helps me |
| 13:10.21 | beebeeep | looks like there is some parameter in asterisk.conf :) |
| 13:12.33 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 13:13.34 | braunr | wdoekes2: i'm submitting it to uclibc first (although glibc/eglibc could suffer from this too), then i'll report it to asterisk if it makes sense |
| 13:14.02 | braunr | wdoekes2: and to be honest, i'm not that familiar with dynamic linking either, which is why I ask this question |
| 13:14.12 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 13:15.09 | braunr | wdoekes2: I guess there aren't many users running Asterisk on MIPS |
| 13:16.47 | *** join/#asterisk zvpunry (~ml@p579DCD81.dip.t-dialin.net) |
| 13:16.53 | zvpunry | hi |
| 13:18.21 | *** join/#asterisk MmixX (~mmixx@unaffiliated/mmixx) |
| 13:18.53 | *** join/#asterisk i_heart_asterisk (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 13:20.17 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:20.17 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:20.41 | *** join/#asterisk Fruchthoernschen (~Fruchthoe@trir-5d8004de.pool.mediaWays.net) |
| 13:22.05 | i_heart_asterisk | i have question about asterisk's payload type |
| 13:22.12 | i_heart_asterisk | does it use 101 or 96 ? |
| 13:25.14 | *** join/#asterisk Fruchthoernschen (~Fruchthoe@trir-5d8004de.pool.mediaWays.net) |
| 13:25.50 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 13:36.23 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 13:38.52 | i_heart_asterisk | can you change rfc2833 payload type in asterisk? |
| 13:40.06 | *** join/#asterisk scalex000 (~chatzilla@186.6.182.79) |
| 13:41.02 | i_heart_asterisk | guess everyone is still asleep |
| 13:47.25 | *** join/#asterisk n1x0n (nixon@dft-labs.eu) |
| 13:47.49 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 13:48.30 | pabelanger | i_heart_asterisk: Why? |
| 13:49.06 | pabelanger | There are no configuration options, that I know of. So you'd have to modify the source |
| 13:59.48 | i_heart_asterisk | pabelanger, do you know what it is set to be default in asterisk 1.6? |
| 13:59.53 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
| 14:08.01 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 14:08.45 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 14:09.16 | Katty | zombies in |
| 14:09.24 | Katty | caaaaaaaaaffffffeeeeineeeeeeeeeeeeeeeeeee |
| 14:09.54 | tzanger | nrrrgh |
| 14:10.03 | tzanger | another hour and I can have brunch |
| 14:10.12 | tzanger | there will be caffeine then. oh yes. |
| 14:11.07 | Katty | take me with you :> |
| 14:11.36 | tzanger | love to, but (I think) OK is a long way from MI |
| 14:12.27 | pabelanger | i_heart_asterisk: 101, if I remember right. Checking the source would be your best answer |
| 14:12.35 | Katty | tzanger: well, missouri is a long way from MI |
| 14:12.36 | i_heart_asterisk | thanks |
| 14:12.39 | Katty | tzanger: but close enough ;) |
| 14:12.59 | tzanger | why the hell do I keep thinking you're in OK? |
| 14:13.10 | tzanger | MO->MI is only one letter apart |
| 14:13.18 | Katty | who knows. |
| 14:13.40 | tzanger | we'll have to virtually brunch together |
| 14:14.15 | Katty | kay |
| 14:14.42 | tzanger | how goes things? you never write, never call, I'm starting to think you don't care anymore |
| 14:17.15 | Katty | oh well, ya know. |
| 14:17.18 | Katty | i'm teaching a new stripper class. |
| 14:17.23 | Katty | it keeps me pretty busy |
| 14:17.49 | Katty | tzanger: how're you dear? |
| 14:17.51 | mzahariev | :) |
| 14:18.49 | E-bola | Do stripper classes need audiences? :) |
| 14:20.23 | Katty | E-bola: mmmm, no |
| 14:20.40 | E-bola | What a shame, that'd be a nice time killer :) |
| 14:20.59 | Katty | ha |
| 14:21.41 | tzanger | oh, I'm working away, you know how it is |
| 14:21.48 | tzanger | all work and no play |
| 14:21.57 | tzanger | better than starving though ;-) |
| 14:22.00 | tzanger | er :-) |
| 14:22.01 | *** join/#asterisk gerhard7 (~gerhard7@212-123-146-122.ip.telfort.nl) |
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| 14:28.09 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
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| 14:35.03 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 14:37.02 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 14:37.42 | omani | does anyone know how to include the caller picture from openldap to snom phones? |
| 14:38.01 | *** join/#asterisk theHub (~karl@69.177.93.21) |
| 14:41.02 | tzanger | Katty: I didnt' know you taught classes |
| 14:41.10 | Katty | i don't. |
| 14:42.18 | Daejeo | Katty :) |
| 14:43.53 | *** join/#asterisk SeTTleR (~bernd@p5DDEF2D7.dip.t-dialin.net) |
| 14:44.36 | tzanger | Katty: well jeez. in that case I'm the sugar gum princess. |
| 14:45.05 | Katty | *hee* |
| 14:45.17 | Katty | tzanger: i have /taken/ classes tho |
| 14:45.27 | Elwinf | Hello. Ive got a problem with IAX2 not giving a ringbacktone to internal calls, running 1.4.31. Anyone got any tips for me? |
| 14:45.27 | Katty | tzanger: not for professional reasons tho |
| 14:45.52 | tzanger | if I ever took a class like that the video would win me $millions on america's funniest home videos |
| 14:45.58 | Katty | lol |
| 14:46.05 | Katty | ya it takes a certain ammount of grace |
| 14:46.07 | *** join/#asterisk garymc (~chatzilla@host81-148-2-229.in-addr.btopenworld.com) |
| 14:46.35 | *** join/#asterisk m_tadeu (~quassel@89.180.176.52) |
| 14:46.39 | tzanger | actually more than that, because I would negotiate a reality tv series beforehand, with the penultimate episode being the application to america's funniest home videos |
| 14:46.57 | tzanger | Katty: indeed. I have some grace, but none of it involves moving my body parts |
| 14:48.57 | Katty | tzanger: well you get better with practice, ya know :P |
| 14:50.01 | tzanger | Katty: the amount of practice I'd have to perform (and thus my teacher would have to endure) would make such an undertaking pointless, though amusing for them. |
| 14:50.15 | Katty | yeah, but what if you got really good at it |
| 14:51.03 | tzanger | I once -- ONCE -- took a ballroom dancing course with a girlfriend long, long ago.... the teacher singled me out in the group during some kind of exercise and exclaimed "Ladies and gentlemen, this is a tragedy... this man was born with no hips!" |
| 14:51.32 | Katty | LOL |
| 14:51.55 | Katty | you poor dear. |
| 14:52.37 | tzanger | I took it in stride but that was the last time I ever attempted to do anything like that. I can dance as well as the average self-conscious white guy, but that's obviously not saying much |
| 14:53.16 | tzanger | if it's normal slow dancing kind of stuff I can do that without much trouble, but if there's any kind of beat I am sure I start resembling a chicken caught on an electric fence |
| 14:53.45 | Katty | hehehe |
| 14:53.56 | Katty | i bet i could teach ya a thing or two ;) |
| 14:54.20 | tzanger | with enough whiskey I become a regular Fred Astaire... at least in my own mind |
| 14:55.49 | tzanger | ohmigod... t minus 5 minutes to brunch time |
| 14:55.51 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 14:55.56 | Katty | well there ya go! |
| 14:56.04 | Katty | that's the whole point anywya, to have a good time! |
| 14:56.34 | tzanger | Katty: I appreciate the offer but I will counter with one I think might be more enjoyable... I will build and install the stage and pole or whatever you need, and you can dance |
| 14:56.52 | Katty | ha! |
| 14:56.58 | tzanger | we can drink, and when the critical point is hit, I can join you onstage |
| 14:57.10 | tzanger | it's best to have paramedics standing by though |
| 14:58.39 | tzanger | Katty: http://hackaday.com/2011/01/25/refurbishing-a-refrigerator-for-fermenting |
| 14:58.43 | tzanger | that is a *fantastic* idea |
| 14:59.06 | tzanger | sourdough, homemade yogurt... and it doesn't look like ass either |
| 14:59.59 | Linuturk | is there a way for me to have asterisk ask my polycom phones what BootROM they are running? |
| 15:00.20 | Naikrovek | asterisk doesn't care what bootrom a phone is using - it doesn't affect SIP at all |
| 15:00.37 | Naikrovek | you can however tell the phones to log to an FTP server, and you can grep through that |
| 15:00.38 | Linuturk | yeah, I know. I was hoping there was a way to query that though |
| 15:00.51 | Linuturk | yeah |
| 15:00.57 | *** join/#asterisk ks3 (~ks3@74.203.195.1) |
| 15:01.06 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 15:01.08 | tzanger | Linuturk: I don't think there is any way to do that... Asterisk only knows the phones through SIP. Unless there is a vendor-specific SIP extension for such things I think you're out of luck |
| 15:02.05 | *** join/#asterisk coppice (~chatzilla@220.229.255.5) |
| 15:03.05 | Katty | hi ariel_ |
| 15:03.06 | Katty | hi coppice |
| 15:03.16 | ks3 | How do I clear a hash? I have a context that may be called more than once to pull info from a database, and it appears that if the query returns no results, the old hash data is still there after Set(HASH(name)=ODBC_xxxx). I've tried Set(HASH(name)=) to no avail. |
| 15:03.20 | ariel_ | hello Katty |
| 15:03.28 | ariel_ | hello everyone |
| 15:03.31 | *** part/#asterisk Fruchthoernschen (~Fruchthoe@trir-5d8004de.pool.mediaWays.net) |
| 15:05.17 | ariel_ | tzafrir_laptop: you around for some Xorcom/Twinstart questions? |
| 15:05.31 | tzafrir_laptop | ariel_, yes |
| 15:09.04 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:11.07 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
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| 15:17.29 | *** part/#asterisk ks3 (~ks3@74.203.195.1) |
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| 15:22.33 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:24.51 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
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| 15:31.54 | pecenipicek | Good day. I have quite an interesting problem at my hands now. i've got 3 asterisk boxes running, 2 are in my local net, a third is at a remote location, connected to a much larger company net. Communication between boxes A and B work, and also work between B and C (as the B box is the one that is also connected to the before mentioned companies net via VPN) |
| 15:32.15 | pecenipicek | dundi lookups work on all three boxes for all three servers and can find peers on each. |
| 15:32.27 | pecenipicek | calls from a to b and from b to c also work. |
| 15:32.29 | *** join/#asterisk ks3 (~ks3@74.203.195.1) |
| 15:32.39 | pecenipicek | the problem however is when i try to call from box a to box c. |
| 15:33.39 | pecenipicek | since the dundi lookup returns the local IP address of the box in the companies network, and since box A isnt connected to the bigger net via vpn, it cant make an outgoing call. |
| 15:34.03 | pecenipicek | i'd like to set up the whole deal to transfer all traffic via IAX2 trunks. |
| 15:34.50 | pecenipicek | the problem is that instead of using box B as a stepping stone, boxes A and C try to communicate directly. which doesnt work, for obvious reasons. |
| 15:35.09 | Katty | Qwell: can i use scp to copy an entire directory recursively? |
| 15:35.28 | E-bola | scp -r |
| 15:35.55 | pecenipicek | Also, since i forgot to mention, Box B is used as a dundi lookup server for lookups among other things. |
| 15:36.31 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 15:37.00 | *** join/#asterisk ccesario (~ccesario@189.20.219.10) |
| 15:37.34 | Katty | E-bola: something like scp -r user@ip:/path/to/directory /path/to/here |
| 15:38.26 | E-bola | Yup |
| 15:38.44 | Katty | cheers. ty sir |
| 15:38.51 | Naikrovek | Katty: what did you think of that music |
| 15:38.52 | *** join/#asterisk Fruchthoernschen (~Fruchthoe@trir-5d8004de.pool.mediaWays.net) |
| 15:38.58 | Katty | Naikrovek: i liked it! |
| 15:39.06 | Naikrovek | nice |
| 15:39.07 | Katty | Naikrovek: duck hunt, zelda, and the marios were awesome |
| 15:39.15 | Katty | Naikrovek: but those are the ones i know the best from nes. |
| 15:39.32 | Naikrovek | i played the heck out of dragon warrior so i liked that one as well as the ones you mentioned |
| 15:43.14 | Katty | E-bola: woot. it works (= |
| 15:43.24 | tzanger | Brunch commencing. I was late. ;-) |
| 15:44.55 | Katty | ok here's a good question for ya'll. |
| 15:45.21 | Katty | i have a production server, and then a backup server. the backup server pulls config files and such from the production box. |
| 15:45.33 | Katty | how often do you think they should do this file copy? |
| 15:45.54 | tzanger | How often does your config change? |
| 15:45.57 | _Corey_ | Katty: Depends on how often the files change... |
| 15:46.21 | tzanger | First coffee... Aaaaaaaahhhhhhh |
| 15:47.32 | Katty | _Corey_: hmm, well not very often tbh |
| 15:47.42 | Katty | _Corey_: i'll occasionally change an IVR if the weather is bad. |
| 15:47.50 | tzanger | Katty: I would expect a production system to not change often. An hourly rsync would be plenty |
| 15:48.14 | Katty | hourly? i was thinking daily |
| 15:48.21 | Katty | we have maybe 20 people in the building |
| 15:48.28 | tzanger | You could also play with inotify/dnotify and sync only when the due changes |
| 15:48.30 | _Corey_ | We do it a couple ways usually |
| 15:48.35 | tzanger | Er dir |
| 15:48.38 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 15:48.44 | _Corey_ | we sync /var/spool/asterisk/voicemail like every 15 mins |
| 15:48.54 | _Corey_ | and /etc/asterisk maybe hourly |
| 15:49.11 | tzanger | A daily change would not do you good for ivr change due to storm |
| 15:49.11 | Katty | yeah, i'm not opposed to syncing some things more frequently than others |
| 15:49.35 | Katty | yeah well an IVR change is nothing. |
| 15:49.38 | Katty | that's not what i'm concerned about |
| 15:49.49 | Katty | i'm concerned about hardware failure, and needing another server /functional/ el pronto |
| 15:49.56 | Katty | i can always make an IVR update in 30 seconds |
| 15:50.32 | _Corey_ | you'll want to sync /var/lib/asterisk/astdb (and MySQL if you're running FreePBX or CDR SQL) |
| 15:50.37 | E-bola | Katty: Join the group of ppl needing a simple asterisk HA solution :) |
| 15:50.51 | leifmadsen | Simple + HA don't really go together ;) |
| 15:50.58 | Katty | freepbx is dirty. |
| 15:51.04 | E-bola | leifmadsen: sometimes it can :) SMTP ftw |
| 15:51.14 | Katty | i could use some help backing up pgsql and restoring it |
| 15:51.14 | leifmadsen | email? |
| 15:51.16 | _Corey_ | lol |
| 15:51.17 | *** join/#asterisk chuhnk (~asim@88-98-162-230.ll.zen.net.uk) |
| 15:51.24 | Katty | i've used pgdump and pgrestore, but i can't seem to get it to work right |
| 15:51.24 | tzanger | <PROTECTED> |
| 15:51.28 | Katty | and recreate the tables and what not |
| 15:51.49 | E-bola | My smtp antispam/virus cluster is very simple, and rock solid |
| 15:51.50 | chuhnk | hey all |
| 15:52.29 | Katty | OH! |
| 15:52.30 | Katty | that reminds me. |
| 15:52.37 | Katty | is there a way to back up the asterisk blacklist? |
| 15:52.40 | tzanger | Katty: it takes a bit of playing |
| 15:53.02 | Katty | i'd be better off just keep a txt file of the numbers, and then readd them? |
| 15:53.12 | leifmadsen | isn't that all just kept in astdb? |
| 15:53.30 | chuhnk | I'm looking to upgrade from 1.6.0.1 to 1.6.2. Are there any incompatibilities through the lifetime of the minor revisions in 1.6? |
| 15:53.47 | tzanger | blacklist |
| 15:53.56 | Katty | leifmadsen: can i safely just copy and paste /var/lib/asterisk/astdb back and forth between machines? |
| 15:54.13 | Katty | or is there more to it than that |
| 15:54.29 | leifmadsen | never tried it |
| 15:54.39 | Katty | let's use Qwell as a guinea pig |
| 15:55.12 | Naikrovek | chuhnk: 1.6.0 is a different release than 1.6.2, just like 1.6.2. and 1.8 are different |
| 15:55.23 | Naikrovek | don't think that .0 or .2 at the end are minor revisions |
| 15:55.35 | pecenipicek | joyful |
| 15:56.09 | pecenipicek | googling about interlinking 3 or more asterisk boxes via iax2 trunks is nowhere to be found -.- |
| 15:56.28 | tzanger | dont listen to leifmadsen, he has to look everything up in the book |
| 15:56.35 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 15:56.46 | _Corey_ | haha |
| 15:57.03 | _Corey_ | in our experience, you can sync astdb safely, though i wouldn't copy over a live one |
| 15:57.41 | _Corey_ | normally we use this in conjunction with a floating IP so that when you bring up your secondary asterisk box, all the phones are "registered" |
| 15:57.57 | _Corey_ | if you're using realtime, this becomes unnecessary |
| 15:57.57 | tzanger | can you cheat with database dump and database put. itd be kind of nasty |
| 15:58.36 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:59.18 | tzanger | wow theres a name i havent seen in forever |
| 15:59.27 | Katty | hi tony |
| 15:59.31 | tzanger | how goes anthm? |
| 15:59.39 | *** join/#asterisk SaiSoma (~chatzilla@client105.jdcc.edu) |
| 16:00.55 | pecenipicek | this channel continues to prove over and over again how useless it is when you come to anything above "I use only one Asterisk box for a home PBX"... Not that the overall documentation is in a much better scope... |
| 16:00.59 | pecenipicek | Later all. |
| 16:01.31 | *** join/#asterisk admin0 (~admin0@cm180.omega140.maxonline.com.sg) |
| 16:01.35 | tzanger | another satisfied customer. NEXT!!! |
| 16:05.21 | tzanger | gah. i hate when wait staff "warm up" your perfectly configured coffee |
| 16:14.56 | leifmadsen | wtf does he want us to say? |
| 16:15.18 | leifmadsen | there is no, "Push the BLUE button for a distributed, many level redundant HA system" |
| 16:16.08 | *** join/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 16:18.05 | asteriskATmarmuD | does anyone know how I can call a number to check wether it is valid while avoiding the phone on the other end to ring? |
| 16:18.32 | WIMPy | asteriskATmarmuD: What technology? |
| 16:19.02 | asteriskATmarmuD | <PROTECTED> |
| 16:19.13 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 16:19.25 | asteriskATmarmuD | <PROTECTED> |
| 16:19.39 | WIMPy | So which part are you interested in? SIP or ISDN? |
| 16:20.32 | asteriskATmarmuD | <PROTECTED> |
| 16:21.42 | WIMPy | So you are talking SIP to an ISDN gateway, right? But do you want to check for presence of extensions on that gateway or do you want to check for a PSTN number that way? |
| 16:22.18 | asteriskATmarmuD | <PROTECTED> |
| 16:23.31 | *** join/#asterisk ccesario (~ccesario@linux.unialco.com.br) |
| 16:23.50 | *** join/#asterisk ks3 (~ksandy@74.203.195.1) |
| 16:23.59 | WIMPy | Change the Bearer Capability to something else. But that might be an interesting task via a SIP gateway. |
| 16:25.20 | asteriskATmarmuD | <PROTECTED> |
| 16:25.40 | WIMPy | Either the manual or ask, yes. |
| 16:25.41 | asteriskATmarmuD | <PROTECTED> |
| 16:26.01 | asteriskATmarmuD | <PROTECTED> |
| 16:27.24 | tzanger | That's an interesting problem, I don't think it can be done cleanly, much less via SIP |
| 16:28.22 | *** part/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 16:28.39 | WIMPy | Off course you never know if there is something on the other end that supports whatever you selected by chance. |
| 16:28.59 | tzanger | How would you do it from asterisk if you were directly connected to the isdn network? |
| 16:29.42 | tzanger | If you went low enough into q.931 there might be a facility for that but I don't know for sure |
| 16:30.08 | WIMPy | AFAIK no |
| 16:30.10 | *** join/#asterisk ks3 (~ksandy@74.203.195.1) |
| 16:31.18 | WIMPy | I'm actually not sure. I have seen something for BC with dahdi, but it might be a read-only thing. |
| 16:31.48 | tzanger | Maybe if you did a half-setup kind of thing |
| 16:32.14 | tzanger | Id have to review isdn call setup |
| 16:32.37 | WIMPy | That would be down to timing, so you will have the other end ringing even if only for a fraction of a second. |
| 16:33.59 | WIMPy | Ah, there it is: CHANNEL(transfercapability) |
| 16:34.08 | tzanger | Better to do it via some out of band mechanism |
| 16:34.36 | tzanger | Or build up a cache of valid numbers or something |
| 16:34.42 | WIMPy | I guess that would be possible with SS7. |
| 16:35.10 | tzanger | World peace is possible through SS7 |
| 16:35.20 | WIMPy | :-) |
| 16:35.48 | tzanger | One of the many reasons why it's so difficult to write a good implementation. :-) |
| 16:36.37 | WIMPy | I'd just set up a video call. Chances are pretty high that it's not supported. |
| 16:37.08 | WIMPy | Or UDI with some weird LLC. That should be the safest. |
| 16:38.18 | WIMPy | However you can't set LLC with Asterisk. |
| 16:38.32 | tzanger | But will the other end respond correctly for "service not supported" vs "unallocated number"? I know both responses are there but if the switch itself doesn't support video it may test for that first |
| 16:39.06 | WIMPy | The switch doesn't care. It's an end-to-end thing. |
| 16:40.12 | tzanger | What I'm saying though is that whether your switch will attempt a connection or the far one will reject it |
| 16:40.23 | WIMPy | But that can be a catch with the UDI aproach. In north America you might find a switch only supporting RDI, but I'm not sure if that's still likely. |
| 16:41.01 | tzanger | The far end may reject the call because it's video without trying to see if the number's valid first |
| 16:41.01 | WIMPy | Other than the last mentioned exception it's always the user. |
| 16:41.18 | tzanger | There's no telling how the check is made |
| 16:41.25 | *** join/#asterisk jkroon (~jkroon@dsl-241-250-86.telkomadsl.co.za) |
| 16:41.27 | WIMPy | At least for the fixed network. GSM operators sometimes filter. |
| 16:41.57 | tzanger | If the dest is a pots call the switch is the one making that check. I don't think you can guarantee the far end is digitally terminated |
| 16:42.18 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 16:42.27 | WIMPy | They can tell you anything, including the called number is unallocated if the call comes from an unwanted caller-id. But that's not the normal behaviour. |
| 16:43.22 | WIMPy | That's not a bad thing. The switch most certainly won't have been hacked into doing fancy stuff noone expects. |
| 16:43.50 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.178.126.dsl.dyn.forthnet.gr) |
| 16:44.30 | WIMPy | But there's always the location bits you can check. |
| 16:44.54 | tzanger | I don't know the spec well enough to say if the number checking must be done before capabilities |
| 16:45.32 | WIMPy | Not sure if that's defined at all, but it works in practice. |
| 16:46.32 | *** join/#asterisk JoseBravo (~Jose@190.144.124.194) |
| 16:46.34 | JoseBravo | Hello |
| 16:47.34 | JoseBravo | Someone can recomend me a SIP, AIX provider to do international calls from my PBX? |
| 16:48.20 | WIMPy | Anyway: If the check isn't done the way you'd expect it could only prevent you from scanning extensions behing a PBX, but it surely works for allocated numbers. |
| 16:49.01 | WIMPy | JoseBravo: One in the country you want to call? |
| 16:49.17 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 16:49.39 | JoseBravo | I used to use voipjet but they just stop giving us support, we have sent a lot of emails but none have responded. |
| 16:49.48 | JerJer | typical |
| 16:50.00 | JoseBravo | WIMPy, Im in Colombia but most calls will go to USA. |
| 16:50.20 | WIMPy | ~itsplis-us |
| 16:50.30 | WIMPy | ~itsplist-us |
| 16:50.30 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 16:50.47 | WIMPy | JoseBravo: Try one of those. |
| 16:53.00 | JoseBravo | Thanks WIMPy |
| 17:00.14 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 17:00.22 | *** join/#asterisk espiceland (~erin@nat/digium/x-tknffcaguortuzyd) |
| 17:04.29 | omani | is it possible to get a caller picture to the phone? |
| 17:04.52 | omani | I use snom 821 phones and it has the functionality of displaying caller pictures through ldap |
| 17:05.12 | omani | but it just doenst work |
| 17:05.14 | _Corey_ | omani: Didn't you ask this yesterday? |
| 17:08.27 | *** join/#asterisk breardo (~no@234-200-29-134.hcc.mnscu.edu) |
| 17:08.30 | breardo | hello all.. |
| 17:08.50 | zvpunry | i want queuemembers to send a dtmf-digit to answer a call and playback a message like "press 1 to answer this call or any other key to let someone else handle this call", but i have no idea how to do this with app_queue or with the dialplan. Is there a way to do this |
| 17:08.57 | breardo | sorry for the dumb question.. but this is the first time i've run Asterisk on Linux with a Digium Card... im curious as to how I can ensure the onboard echo canceller is functioning? |
| 17:09.13 | breardo | I have some users complaining of echo... this card has the onboard hwec, so im not sure if its enabled or not.. |
| 17:09.49 | _Corey_ | breardo: run 'dahdi show channel X' where X is a live channel |
| 17:10.03 | _Corey_ | it will tell you what echo can is being used |
| 17:10.12 | breardo | thanks.. i'll try in a sec |
| 17:11.47 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 17:12.52 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 17:13.30 | *** part/#asterisk tclark (~Administr@S0106001310ead738.du.shawcable.net) |
| 17:16.31 | omani | _Corey_: yes I did |
| 17:17.18 | _Corey_ | omani: Since Asterisk has nothing to do with causing the Snom to interact with LDAP, I would suggest contacting Snom support or a Snom-specific forum |
| 17:17.30 | _Corey_ | I don't get the impression that anyone here has tried what you're attempting |
| 17:17.55 | omani | ok |
| 17:19.25 | breardo | hey _Corey_ .. so.. when it says which echo canceller is in use... what should it say if its using the onboard hwec? |
| 17:19.43 | breardo | I figured if I saw something like MG2 it'd mean its doing inline EC in the dahdi drive, not on the card.. correct? |
| 17:19.50 | breardo | dahdi driver, that is |
| 17:20.21 | breardo | sorry to say I dont have the access to the PBX like I should at the moment, but I'll be seeing it on-site in a couple of days is all.. |
| 17:21.29 | saxa | hello, anybody good in debugging sip ? Why when I call from one extension to another, as soon as I pick up it drops the line ? |
| 17:21.43 | saxa | here is the sip set debug on |
| 17:21.46 | saxa | http://pastebin.com/YXTZdK8P |
| 17:21.59 | saxa | any ideas ? |
| 17:22.14 | _Corey_ | breardo: I think you're right but I'd have to look at one |
| 17:22.38 | breardo | saxa.. have you done a rtp debug? |
| 17:22.54 | breardo | Corey.. ok.. I'll check when im on-site, or if I can get ssh access reestablished.. |
| 17:22.58 | breardo | thanks for the help |
| 17:23.51 | breardo | saxa.. looks like the SIP stuff is transmitting OK, but RTP might be failing.. |
| 17:24.08 | breardo | i had that problem once with a misconfigured (aka, not configured at all) default gateway on the server.. |
| 17:24.23 | *** join/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 17:26.04 | saxa | breardo: hmm, will check that, when I call from the dahdi station to the sip I can pick up and seem it works, the problem is calling from a sip to dahdi |
| 17:26.43 | breardo | yeah.. look into that |
| 17:27.10 | breardo | it'll be obvious with an rtp debug.. |
| 17:27.17 | breardo | it was for me, anyways |
| 17:27.32 | breardo | and this WAS back in 1.4 |
| 17:27.37 | breardo | so.. take my advice with a grain of salt :) |
| 17:27.59 | saxa | http://pastebin.com/uEFe3V0X <<--- here is the rtp debug on |
| 17:28.25 | saxa | i dont see anything strange in that |
| 17:28.37 | saxa | maybe I should upgrade to 1.8.x |
| 17:36.39 | breardo | thats showing one-way RTP |
| 17:36.45 | breardo | you should see both sent and received.. |
| 17:37.07 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 17:42.06 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 17:42.39 | zvpunry | bye |
| 17:43.28 | leifmadsen | o/ |
| 17:43.36 | *** join/#asterisk JonnyD_work (~Jon@12.222.63.34) |
| 17:44.03 | JonnyD_work | is there an option or a way in asterisk queues to limit how many calls an agent can answer? |
| 17:45.03 | leifmadsen | JonnyD_work: what do you mean? |
| 17:45.12 | leifmadsen | agents should typically only answer 1 call at a time |
| 17:45.26 | leifmadsen | in which case, you want 'ringinuse=no' in queues.conf and 'callcounter=yes' in sip.conf |
| 17:45.41 | leifmadsen | (presuming your end points are SIP and you're calling them via SIP in the queue) |
| 17:45.51 | JonnyD_work | right i mean in any given day, is there a way to limit an agent from answering more than (x) calls for the whole day? |
| 17:45.59 | leifmadsen | sure, use AstDB |
| 17:46.04 | leifmadsen | track the number of calls |
| 17:46.11 | JonnyD_work | so if i only want an agent to be able to take 3 calls |
| 17:46.17 | leifmadsen | then use Local channels to check prior to calling the agent |
| 17:46.31 | leifmadsen | if they have taken more than X calls, then remove them from the queue |
| 17:46.43 | leifmadsen | then call Congestion() to tell the Queue() to find somenoe else |
| 17:48.08 | JonnyD_work | when do i check to see if they have taken x calls? before the call enters the queue? |
| 17:48.25 | JonnyD_work | or is here a way i could do that when calling agents? |
| 17:48.45 | leifmadsen | Using Local Channels: http://ofps.oreilly.com/titles/9780596517342/ch13.html#ACD_id288989 |
| 17:49.01 | leifmadsen | JonnyD_work: no, just prior to calling the agent via the Local channel |
| 17:49.09 | leifmadsen | read the documentation I just linked |
| 17:49.12 | leifmadsen | it explains what I mean |
| 17:52.57 | JonnyD_work | cool thanks |
| 17:54.06 | *** join/#asterisk tuxx- (tuxx@vps460.directvps.nl) |
| 17:54.15 | tuxx- | oontz. |
| 17:58.03 | *** join/#asterisk ghostmediapro (~Trupsalms@71.23.182.164) |
| 17:58.39 | ghostmediapro | could some one show me a complete sample host.conf, as mines is empty |
| 17:59.21 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 17:59.53 | *** join/#asterisk ccesario (~ccesario@linux.unialco.com.br) |
| 18:01.36 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v012-182.mobile.uci.edu) |
| 18:02.07 | Corydon76-home | ghostmediapro: type: man host.conf |
| 18:02.29 | Katty | has homemade chicken nuggets and homemade fries! |
| 18:02.52 | ghostmediapro | i'm trying to ping google from asterisk now, with no luck, then check host.conf which only contians: order hosts,bind |
| 18:02.59 | carrar | homemade chickens? |
| 18:03.13 | Katty | no but it's organic chicken breast |
| 18:03.21 | Katty | which kind of helps, i guess |
| 18:03.44 | coppice | all living things are organic |
| 18:04.00 | Katty | your face is organic. |
| 18:04.14 | coppice | I sure hope so |
| 18:05.10 | Corydon76-home | coppice: only if they're based around carbon |
| 18:05.49 | coppice | OK. all known living things are organic |
| 18:06.28 | Qwell | ghostmediapro: host.conf is not what you're looking for. you want /etc/hosts |
| 18:06.47 | Qwell | and, really, if it's a DNS issue, you'll want /etc/resolv |
| 18:06.56 | Qwell | err, resolv.conf |
| 18:07.27 | Qwell | ghostmediapro: Just type `setup`. It doesn't appear that you have a DHCP server (or it's functioning very poorly). |
| 18:07.59 | ghostmediapro | static assigned, dns works on other machine on the network |
| 18:08.38 | Qwell | run setup and add DNS servers |
| 18:09.13 | drift- | i got freepbx 3.0 setup is there anyway to get google voice working with this? |
| 18:09.16 | drift- | freepbx gui n all |
| 18:11.04 | Katty | gives Qwell a chicken nugget. |
| 18:12.17 | ghostmediapro | [root@phone ~]# ping google.com |
| 18:12.17 | ghostmediapro | ping: unknown host google.com |
| 18:13.03 | Qwell | Did you do what I suggested? |
| 18:13.40 | ghostmediapro | run setup added dns |
| 18:14.16 | ghostmediapro | pointed to my first dns 192.168.1.2 and second 8.8.8.8 |
| 18:14.28 | Qwell | Can you ping both of those addresses? |
| 18:15.07 | carrar | I can't ping 192.168.1.2, must be down!!! :) |
| 18:15.08 | ghostmediapro | yes |
| 18:15.23 | Qwell | ghostmediapro: and is 192.168.1.2 a DNS server? |
| 18:15.31 | ghostmediapro | yes |
| 18:16.31 | *** join/#asterisk wdbl (~not@ool-44c0619d.dyn.optonline.net) |
| 18:17.54 | Qwell | ghostmediapro: dig google.com @192.168.1.2 |
| 18:19.30 | ghostmediapro | the results http://pastebin.com/WyJveqj8 |
| 18:20.06 | Qwell | Did you change other files (such as /etc/host.conf..)? |
| 18:20.50 | ghostmediapro | host.conf only contains order hosts,bind |
| 18:21.08 | Qwell | That didn't answer my question. ;) |
| 18:21.13 | Qwell | s/;/:/ |
| 18:21.18 | ghostmediapro | not |
| 18:21.20 | ghostmediapro | no |
| 18:22.36 | ghostmediapro | resolv.conf http://pastebin.com/QJjPiQ1s |
| 18:23.01 | Qwell | It doesn't look like you added the DNS servers in `setup` |
| 18:23.04 | carrar | empty |
| 18:23.11 | p3nguin | If you ran dig @192.168.1.2 ... then it wouldn't matter what is in resolv.conf anyway. |
| 18:23.39 | ghostmediapro | but i did add dns servers |
| 18:23.47 | Faustov | the SPA922 phones got a setting like "moh server" - can I make asterisk act as a moh server? |
| 18:23.55 | Qwell | wait... did you install X? |
| 18:24.19 | drift- | heh anyone wanna make some $ setup my freepbx 3.0 to work with google voice |
| 18:24.30 | ghostmediapro | no x: as in gui |
| 18:24.32 | carrar | ghostmediapro, use http://pastebin.com/nak7tztY |
| 18:24.39 | drift- | i just did http://www.linode.com/stackscripts/view/?StackScriptID=514 < fresh install on linode |
| 18:24.55 | drift- | dont know wtf i gotta do next... it installed debain + freepbx 3.0 |
| 18:25.21 | ghostmediapro | carrar: thats for resolv.conf correct |
| 18:25.22 | carrar | ghostmediapro, if you don't know your domain remove the two top lines |
| 18:25.35 | p3nguin | In case you didn't notice, this is an ASTERISK channel, and the majority of us don't care about FreePBX. |
| 18:25.40 | carrar | ghostmediapro, add it |
| 18:25.45 | drift- | jah |
| 18:25.46 | Qwell | drift-: and FreePBX v3 is dead. |
| 18:25.53 | drift- | okay |
| 18:26.00 | leifmadsen | Qwell: o/ |
| 18:26.02 | drift- | well i cant get gui asterisk installed |
| 18:26.09 | drift- | because its only in .iso |
| 18:26.11 | p3nguin | YOu don't need it anyway. |
| 18:26.19 | carrar | ghostmediapro, REPLACE what is in your resolv.conf with that |
| 18:26.32 | p3nguin | GUIs just get in the way of the real configuration. |
| 18:26.42 | leifmadsen | drift-: the asterisk GUI is available via subversion, not just ISO |
| 18:27.07 | drift- | hrm |
| 18:27.08 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 18:27.22 | espiceland | drift-: http://svn.digium.com/svn/asterisk-gui/branches/2.0/ |
| 18:27.26 | Faustov | nvm google magistra vitae |
| 18:28.03 | drift- | i just wanna get google voice working dam it heh i dont care how lol |
| 18:28.08 | drift- | i cant seem to get it to work |
| 18:28.23 | p3nguin | Read the wiki. Get to work. |
| 18:28.28 | Qwell | drift-: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 18:28.54 | drift- | that doesnt do shit tho |
| 18:28.56 | drift- | i tired that |
| 18:28.59 | carrar | people still use google? |
| 18:28.59 | drift- | i need a phone to work with it |
| 18:29.02 | ghostmediapro | carrar: http://pastebin.com/h5UdAiLX |
| 18:29.04 | drift- | heh |
| 18:29.17 | carrar | REPLACE |
| 18:29.19 | carrar | not ADD TO |
| 18:29.23 | Qwell | and not change. |
| 18:29.25 | drift- | i need a polycom 501 sip phone to work with a linode / virtual server |
| 18:29.29 | drift- | + google voice |
| 18:29.31 | carrar | and thats not what I sent you |
| 18:29.41 | p3nguin | Why such a specific phone model? |
| 18:30.04 | drift- | just the phones i have |
| 18:30.12 | p3nguin | oh |
| 18:30.15 | Qwell | and how do those instructions preclude you doing that? |
| 18:30.23 | carrar | ghostmediapro, it needs to look EXACTLY like this: http://pastebin.com/dMr2rBAQ |
| 18:30.26 | drift- | now i know how to config the phone |
| 18:30.38 | p3nguin | You're half way there! |
| 18:30.41 | drift- | heh |
| 18:30.46 | drift- | i know how to install asterisk... |
| 18:30.59 | carrar | click GO |
| 18:31.03 | carrar | err START |
| 18:31.19 | p3nguin | NetworkMangler is a problem. |
| 18:31.32 | Qwell | p3nguin: I dunno wtf it's using that |
| 18:32.05 | p3nguin | Probably his exceptional administrative skills helped it along. |
| 18:32.31 | ghostmediapro | carrar: http://pastebin.com/tBS2t86d |
| 18:32.51 | carrar | ghostmediapro, AGAIN thats not what I posted |
| 18:32.58 | carrar | however you are getting closer |
| 18:33.09 | carrar | ghostmediapro, http://pastebin.com/dMr2rBAQ |
| 18:33.11 | p3nguin | Lines 4 and 5 are the only ones needed. |
| 18:33.54 | carrar | why only do the bare minimum |
| 18:34.13 | p3nguin | It's the only thing needed. |
| 18:34.34 | carrar | needed is objectionaly |
| 18:34.34 | p3nguin | The other lines aren't really going to help anything. |
| 18:34.34 | ghostmediapro | carrar: done |
| 18:34.44 | carrar | and? |
| 18:34.54 | drift- | p3nguin: http://www.jcarle.com/2010/12/17/installing-freepbx-2-8-with-asterisk-1-8-on-centos-5-5/ |
| 18:34.58 | carrar | ping google.com |
| 18:35.14 | drift- | so you gotta install asterisk and freepbx? |
| 18:35.16 | p3nguin | drift-: You'll be better off with official documentation in most cases. |
| 18:35.27 | Qwell | eww |
| 18:35.32 | p3nguin | You don't HAVE to install FreePBX. |
| 18:35.39 | ghostmediapro | carrar: life saver thanks, off to school |
| 18:35.57 | carrar | yeah |
| 18:36.03 | carrar | thats cause I have a HAM RADIO License |
| 18:36.07 | carrar | HAM RDAIO SAVES LIVES |
| 18:36.10 | p3nguin | hahahahaha |
| 18:36.49 | Katty | ham radio? |
| 18:36.51 | Katty | carrar: 66 |
| 18:36.57 | carrar | YES |
| 18:37.08 | p3nguin | pork communicator |
| 18:37.17 | tzanger | my dad and my brother are both amateur radio licensed. I always wanted to get it but never had the time |
| 18:37.19 | drift- | heh |
| 18:37.29 | drift- | what is freepbx its just a gui for asterisk? |
| 18:37.41 | p3nguin | It's a GUI, for sure. |
| 18:37.44 | carrar | tzanger, it's easyer then installing Asterisk |
| 18:37.50 | carrar | or setting DNS |
| 18:37.59 | p3nguin | It's also easier THAN installing Asterisk. |
| 18:38.14 | Katty | i'll pork your communicator in a minute |
| 18:38.16 | carrar | for some it is |
| 18:38.20 | carrar | heh |
| 18:38.25 | carrar | I would think |
| 18:38.26 | tzanger | carrar: haha yes |
| 18:38.28 | Qwell | Katty: language! |
| 18:38.41 | tzanger | haha |
| 18:38.42 | Katty | Qwell: :< |
| 18:40.02 | Qwell | Katty: <3 |
| 18:40.07 | Katty | Qwell: :> |
| 18:40.14 | carrar | You two get a room! |
| 18:43.29 | Katty | okay |
| 18:43.32 | Katty | can it be your room? |
| 18:43.35 | Qwell | blinks |
| 18:43.36 | p3nguin | I wonder how much the exams have changed since '95 when I decided to get licensed. |
| 18:43.39 | carrar | sure |
| 18:43.41 | Katty | k |
| 18:43.45 | carrar | But I'll video tape it |
| 18:43.47 | Katty | :< |
| 18:43.58 | Katty | k but i want a copy |
| 18:44.04 | *** part/#asterisk ghostmediapro (~Trupsalms@71.23.182.164) |
| 18:44.15 | carrar | np, you can download it from youtube! |
| 18:44.18 | Katty | k |
| 18:44.47 | p3nguin | Must be fairly mild if they don't delete it. |
| 18:45.10 | Qwell | it's highly downrated |
| 18:45.10 | carrar | Well this is a family channel |
| 18:45.19 | Katty | mhmm |
| 18:45.30 | Katty | but that'd be an awesome movie of us playing Uno |
| 18:45.54 | carrar | Wild Card Katty! |
| 18:46.02 | Katty | oh yeah! break out the vodka |
| 18:46.30 | tzanger | Katty: oh so you are trying to get him to dance, I see... :-) |
| 18:46.49 | Katty | tzanger: no that's just you, dear. |
| 18:46.58 | *** join/#asterisk y7n (~mrg0xt@188-223-79-144.zone14.bethere.co.uk) |
| 18:47.02 | tzanger | oic |
| 18:47.07 | Katty | tzanger: there's room for plenty of people at the Uno drinking game tho |
| 18:47.10 | *** join/#asterisk blee (~blee@99-117-188-231.lightspeed.dybhfl.sbcglobal.net) |
| 18:47.20 | *** join/#asterisk i_heart_asterisk (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 18:47.22 | tzanger | haha |
| 18:47.40 | Katty | which is good thing, cause after about 3 shots i will need help playing the game ;> |
| 18:48.23 | i_heart_asterisk | is anybody familiar w/ the cisco as5400? |
| 18:48.27 | y7n | I have a Atcom IP01 running Asterisk 1.4. When I dial out via PSTN from a SIP account I get the occasional repeating blips. Is this something to do with the Caller ID? |
| 18:49.22 | i_heart_asterisk | I cannot get DTMF to relay from Cisco AS5400 to asterisk |
| 18:50.44 | tzanger | i_heart_asterisk: I've done a lot of work with the as5248 boxes but that was a LONG time ago |
| 18:50.58 | tzanger | i_heart_asterisk: how do you have the dtmf signalled |
| 18:51.00 | blee | Anyone familiar with using wireshark to do packet captures? Everytime I run a capture it says that the all the packets are out of sequence |
| 18:51.05 | blee | so I can't listen to any of the calls |
| 18:51.11 | i_heart_asterisk | tzanger, rfc2833 |
| 18:51.21 | *** join/#asterisk errr_ (~errr@fedora/errr) |
| 18:51.22 | *** join/#asterisk m_tadeu (~quassel@89-180-176-52.net.novis.pt) |
| 18:51.22 | blee | in the rtp strea, |
| 18:51.24 | blee | stream |
| 18:51.29 | i_heart_asterisk | yes |
| 18:51.45 | WIMPy | i_heart_asterisk: I once used a stack of them when there was a chair missing. Does that count? |
| 18:52.43 | i_heart_asterisk | WIMPy, no it does not |
| 18:52.50 | tzanger | i_heart_asterisk: and do you see this with the sip debugging? |
| 18:52.53 | tzanger | or with wireshark |
| 18:53.53 | i_heart_asterisk | tzanger, i believe so, |
| 18:53.56 | i_heart_asterisk | m=audio 12398 RTP/AVP 0 101 |
| 18:53.58 | i_heart_asterisk | a=rtpmap:0 PCMU/8000 |
| 18:54.00 | i_heart_asterisk | a=rtpmap:101 telephone-event/8000 |
| 18:54.01 | i_heart_asterisk | a=fmtp:101 0-16 |
| 18:54.03 | i_heart_asterisk | is that what i'm looking for ? |
| 18:54.36 | tzanger | I think that looks pretty good... I'd check with wireshark too to make sure the RTP events are actually being emitted by teh as5400 |
| 18:55.53 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 18:56.21 | i_heart_asterisk | how to do tcpdump to record RTP ? |
| 18:57.26 | tzanger | tcpdump -ni eth0 -s0 -w blah.bin host 1.2.3.4 |
| 18:57.30 | tzanger | (or at least that's what I use) |
| 18:57.35 | tzanger | then you have a capture of everything |
| 18:57.52 | tzanger | although really I just end up lodaing that stuff into wireshark or using wireshark to capture it since the GUI's nicer |
| 18:58.27 | i_heart_asterisk | yeah im going to use wireshark, thanks |
| 19:05.30 | Katty | so quiet. |
| 19:05.48 | y7n | i know |
| 19:06.19 | y7n | are you able to answer my question above? |
| 19:07.05 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 19:07.07 | p3nguin | I don't know what repeating blips are. |
| 19:07.16 | y7n | popping sounds |
| 19:07.24 | Katty | like...popcorn? |
| 19:07.24 | Qwell | sounds like a terrible phone line |
| 19:07.26 | y7n | www.simplygenius.co.uk/file/misc/Atcom-IP01-PSTN-Sample-Recording.mp3 |
| 19:07.35 | y7n | i made a recording |
| 19:07.56 | y7n | it doesn't happen when I use my normal landline phone |
| 19:08.11 | y7n | only via the FXO module |
| 19:08.26 | Qwell | Are you using a headset with that? O.o |
| 19:08.28 | saxa | ok, got it working, it was the codec thing, seems that I need to eanble specific codecs with allow= instead of allow=all to work with grandstream |
| 19:08.44 | saxa | thx breardo |
| 19:09.05 | breardo | np.. glad you're good to go now :) |
| 19:09.06 | *** join/#asterisk luckman212_phone (~luckman21@mobile-166-137-136-138.mycingular.net) |
| 19:09.06 | Qwell | y7n: I'd say that's electrical. |
| 19:09.10 | p3nguin | saxa: That's how it should be done every time, regardless of what phone you have. |
| 19:09.45 | saxa | p3nguin: yes, but I saw somewhere that it should work allow=all to automatically select the best codec |
| 19:09.57 | saxa | p3nguin: but it seems its not true :) |
| 19:10.09 | y7n | really.. Can it have anything to do with the caller id? |
| 19:10.28 | _Corey_ | caller id shouldn't create noise :) |
| 19:10.28 | saxa | anyway, now also my MESSAGE button works. |
| 19:10.29 | Qwell | y7n: no |
| 19:10.48 | y7n | ok, ill try moving it further from the mains sockets |
| 19:10.53 | y7n | thank you! |
| 19:11.01 | Qwell | y7n: that's like direct electrical interference... |
| 19:11.08 | Qwell | like, a headset with a short or something |
| 19:12.06 | Qwell | How is Asterisk connected to the PSTN (brand, model, etc)? |
| 19:13.55 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-74-74-245-250.rochester.res.rr.com) |
| 19:14.33 | saxa | btw, any suggestion which are best codecs to use on a lan ? |
| 19:14.50 | breardo | g711 |
| 19:14.54 | Qwell | depends on your definition of "best", but probably g.722 or siren14 |
| 19:15.02 | breardo | g.722 is really nice i hear |
| 19:15.14 | breardo | pun intended |
| 19:15.20 | breardo | *sniff* |
| 19:15.32 | Qwell | there was a pun there? |
| 19:15.37 | breardo | :| |
| 19:15.49 | breardo | g.722 is really nice "i hear" |
| 19:15.58 | Qwell | I see |
| 19:19.04 | carrar | I HEAR YOU |
| 19:20.01 | blee | breardo: yea 722 is pretty awesome |
| 19:20.04 | luckman212_phone | g729 is not nice |
| 19:21.14 | p3nguin | I haven't had any issue with it. |
| 19:21.15 | breardo | whats wrong with g729? |
| 19:21.21 | breardo | aside from its licencing. |
| 19:22.39 | luckman212_phone | well it sounds like garbage, is one hehe |
| 19:23.17 | n3hxs-wk | 711 sounds like coffee or big gulps. |
| 19:23.44 | breardo | I hadnt noticed any issues with g.711 or g.729 quality.. |
| 19:23.53 | *** join/#asterisk Defraz (~Defraz@63.226.95.152) |
| 19:23.58 | luckman212_phone | I hear CELT is good. but I haven't used it. isn't that what Skype uses? |
| 19:24.52 | luckman212_phone | maybe I haven't given 729 a fair shake but I can say that a g729 call over a 3G iPhone sounds like 2 rusty tin cans |
| 19:26.10 | *** join/#asterisk nny (~Scott@174.107.201.103) |
| 19:26.24 | nny | does 1.6 or 1.8 natively support custom cdr fields? |
| 19:26.51 | Qwell | 1.0 supported a user field |
| 19:31.36 | *** join/#asterisk `paul (~kutimoy@119.93.15.74) |
| 19:32.54 | `paul | im installing asterisk 1.4.21.2 and zaptel with it. I get compile errors for zaptel and tried dahdi but on make menuselect for asterisk meetme is not enabled (saying zaptel is required) |
| 19:33.07 | `paul | pls help i need meetme to work |
| 19:33.14 | Qwell | upgrade |
| 19:33.22 | Qwell | that version doesn't support dahdi... |
| 19:36.14 | `paul | will vicidial work on 1.4.29.1? |
| 19:36.18 | `paul | 39 i mean |
| 19:36.21 | Qwell | I don't see why not |
| 19:36.28 | `paul | ok thanks |
| 19:36.44 | `paul | just following the scratch_install.txt on vicidial docs :( |
| 19:37.39 | *** join/#asterisk cesar_CR (~cesar@190.10.50.20) |
| 19:39.57 | *** join/#asterisk chazzam (~chazz@173-24-239-247.client.mchsi.com) |
| 19:42.18 | nny | Qwell: had issues with 1.4.36 and cdr_custom last time. Testing 1.8 on my vm to see if it's ported |
| 19:43.08 | p3nguin | I've used custom cdr throughout all of the 1.4 versions. |
| 19:43.11 | *** join/#asterisk ghostmediapro (~Trupsalms@71.23.182.164) |
| 19:43.35 | nny | p3nguin: one sec need to find the source file emailed to me from Digium during the last build |
| 19:43.56 | ghostmediapro | carrar: retrieve_conf encountered an error: 1, no fix found on google |
| 19:44.11 | Qwell | ghostmediapro: #freepbx |
| 19:44.31 | nny | ghostmediapro: chmod +x /var/www/html/admin/modules/framework/bin/* |
| 19:44.33 | p3nguin | I'll also let you know that I have never had to ask Digium for stuff to be emailed to me to get custom cdr to work correctly. |
| 19:44.50 | nny | p3nguin: that's all good and gravy, may be an issue i am stating wrong |
| 19:44.54 | carrar | ghostmediapro, we don't support FreeBPX here |
| 19:45.02 | carrar | Please just download Asterisk Source |
| 19:45.13 | p3nguin | I just looked at a document that showed how to set it up, followed it, and used it ever since. |
| 19:45.27 | nny | p3nguin: see my bad, (it's been months) it's cdr_adaptive_odbc |
| 19:45.50 | nny | confused the two |
| 19:45.51 | p3nguin | I see. I don't think I use that. Let me see what I do have going. |
| 19:46.04 | carrar | ghostmediapro, try joining #freepbx |
| 19:46.13 | nny | wondering if that is availble in 1.8, have a setup here for testing |
| 19:46.22 | ghostmediapro | thanks |
| 19:46.40 | nny | ghostmediapro: join #freepbx and i'll help you, i dealt with that issue yesterday |
| 19:47.04 | p3nguin | It looks like I use cdr_csv, cdr_custom, and cdr_pgsql. |
| 19:48.01 | p3nguin | I'm not sure what I would need odbc for. |
| 19:50.21 | nny | p3nguin: it's used for multiple mysql databases usually |
| 19:50.43 | nny | p3nguin: honestly i may try to find a better way, this setup has too much custom garbage lol |
| 19:58.57 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.80) |
| 19:59.17 | p3nguin | I try my best to keep things as simple as possible. |
| 20:01.55 | saxa | breardo: thx |
| 20:03.39 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 20:10.04 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 20:10.10 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 20:16.07 | p3nguin | Anyone here use VMware ESXi 3? |
| 20:20.35 | *** join/#asterisk b2 (~ion@pdpc/supporter/active/beckb) |
| 20:27.04 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 20:27.16 | *** part/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 20:28.05 | _Corey_ | p3nguin: Yeah, 3 and 4 now I guess |
| 20:28.06 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 20:28.39 | p3nguin | _corey_: Do you remember the installation procedure? If so, I have a technical question about it. |
| 20:28.39 | _Corey_ | p3nguin: We use a lot of ESXi these days |
| 20:29.06 | _Corey_ | p3nguin: Go ahead, if I don't know I will ask someone here |
| 20:29.07 | retentiveboy | Can I add custom key=value entries in users.conf and access those values in configs generated by phoneprov? |
| 20:29.09 | p3nguin | I've only installed it once, and I'm wanting to do it again soon with a different configuration. |
| 20:29.43 | _Corey_ | ESXi itself or a guest OS? |
| 20:29.56 | *** join/#asterisk enzz (~lenz@188.129.224.123) |
| 20:30.05 | enzz | Halo folks!!!1 |
| 20:30.26 | enzz | maay I ask something regaarding digium hardware? |
| 20:31.02 | WIMPy | ~ask |
| 20:31.02 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 20:31.33 | p3nguin | _corey_: When the installer launches and it goes to the page where it shows storage devices, I remember it listed two disks, but I think it could only be installed to one disk. I'm wondering if there is any way to install to both disks using a software RAID configuration, or even if there is a way to install to one disk but use the other for the datastore. I don't know a lot about ESXi, so forgive my ignorance. |
| 20:32.20 | p3nguin | _corey_: If either configuration is possible, that could be useful for what I would like to do. |
| 20:32.53 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 20:33.21 | p3nguin | _corey_: I got the idea from how I would configure a NAS: install to a CF card and put storage on different disks. |
| 20:33.43 | _Corey_ | p3nguin: I don't think you can do software RAID |
| 20:34.02 | enzz | so, am I right that digiums ss7 4 port ci boards working with nly one point code? |
| 20:34.36 | _Corey_ | p3nguin: I would handle all that on a NAS like you're suggesting |
| 20:35.20 | WIMPy | enzz: YOu might want to take a look at chan_ss7. |
| 20:35.21 | p3nguin | _corey_: So you think it's a better idea to install ESXi to one machine and put the data store on a separate appliance? |
| 20:36.15 | _Corey_ | p3nguin: Better for flexbility |
| 20:36.19 | *** part/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 20:36.23 | _Corey_ | you could go vmotion that way and move machines if needed |
| 20:37.09 | p3nguin | It's not free, though, as far as I know. ESXi won't cost me anything. |
| 20:37.15 | drift- | p3nguin: http://www.jcarle.com/2010/12/17/installing-freepbx-2-8-with-asterisk-1-8-on-centos-5-5/ < i got alll that successfully installed freepbx + asterisk on centos 5.5 fresh install... so what do i do next to get google voice going for in and out calls? any suggestions? |
| 20:37.53 | p3nguin | drift-: 1) Forget FreePBX. 2) Follow the wiki on using Google for making phone calls. |
| 20:38.17 | drift- | i need to setup a phone |
| 20:38.26 | p3nguin | Hop to it, then. |
| 20:38.27 | drift- | so that means make an extension and dial plan ... correct? |
| 20:38.38 | *** join/#asterisk pecenipicek (~pecenipic@cpe-109-60-68-222.zg3.cable.xnet.hr) |
| 20:38.59 | pecenipicek | Quick question, do register => style statements work in dundi.conf? |
| 20:39.14 | p3nguin | There are essentially two parts to setting up a phone. a: configure a SIP peer in sip.conf b: create extensions so the phone has a way to be called and can call other things |
| 20:39.18 | p3nguin | drift-: ^ |
| 20:39.26 | enzz | WIMPy, actualy i have no expirience with this boards. only i want to check before buy, is possible to config boaRD FOR USING MAny point codes for all ports thaT AvailaBLE ON THE e1 BORD |
| 20:39.47 | drift- | p3nguin okay |
| 20:39.48 | p3nguin | <drift-> so that means make an extension and dial plan ... correct? <--- extensions MAKE up the dial plan |
| 20:39.56 | enzz | sorry for caps lock, keyboArd stuck |
| 20:40.04 | drift- | so whats that wikki thing for? |
| 20:40.09 | drift- | sip.conf? i lost link i'm sorry |
| 20:40.19 | drift- | https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 20:40.20 | drift- | got it |
| 20:40.32 | p3nguin | drift-: sip.conf is where your phones (devices/peers) are configured. |
| 20:40.41 | drift- | right |
| 20:40.49 | WIMPy | enzz: That's not a hardware question, but a software one. Hence the pointer to chan_ss7. I have no experience either. Getting Asterisk to work with normal ISDN is interesting anough for me. |
| 20:41.01 | drift- | so i edit this gtalk.conf thing |
| 20:41.04 | p3nguin | drift-: Configuring phones for use on the PBX is outside of the scope of setting up Google calling. |
| 20:41.20 | drift- | but... question is doesnt show anything about username or password for google |
| 20:41.37 | drift- | whoops lol did not scroll down |
| 20:41.48 | enzz | WIMPy, thaNKS FOR INFORMAtion |
| 20:41.56 | drift- | okay let me fallow this thing hopefully i wont get stuck :) |
| 20:42.08 | enzz | I will check that |
| 20:42.58 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 20:42.59 | pecenipicek | grr. damn you dundi. |
| 20:43.06 | *** join/#asterisk engaged (~engaged@173.160.43.6) |
| 20:43.23 | engaged | Where would be the cheapest place to get local numbers that I just fwd to an 800 number? |
| 20:43.40 | p3nguin | Any good ITSP could do that. |
| 20:44.49 | p3nguin | I like VoIP.ms for that, actually, but I bet many of the others allow forwarding configurations. |
| 20:45.51 | p3nguin | Purchase a DID, manage the DID, configure forwarding numbers. |
| 20:46.11 | pecenipicek | fellas, anyone know anything about dundi stuff? |
| 20:47.01 | WIMPy | pecenipicek: Ask |
| 20:47.21 | pecenipicek | concretely, do statements of this style work in dundi.conf : |
| 20:47.22 | pecenipicek | register => <username>:<secret>@<hostname> |
| 20:47.32 | pecenipicek | since i'm having some problems getting them through properly |
| 20:48.55 | pecenipicek | or at all, rather. |
| 20:49.31 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 20:50.09 | leifmadsen | pecenipicek: let me look |
| 20:50.20 | leifmadsen | as long as you promise not to storm off this time |
| 20:50.23 | WIMPy | pecenipicek: I've never seen anything the like. Where did you get that from? |
| 20:50.53 | pecenipicek | the same sort of statement works for iax trunk registering. |
| 20:51.05 | pecenipicek | pretty much the exact same damn statement |
| 20:51.22 | leifmadsen | huh? |
| 20:51.24 | drift- | p3nguin: just did the wikki thing https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google < verbatim .... how do i configure my phone now? just where says "malcom" put my userid: on phone malcom? |
| 20:51.27 | pecenipicek | like this: register => iaxuser:${SECRET}@TPAP-T-B1 |
| 20:51.28 | WIMPy | So it's your own invention? |
| 20:51.32 | pecenipicek | no. |
| 20:51.37 | leifmadsen | there is no registering for DUNDi I don't think |
| 20:51.45 | pecenipicek | hrrrm. |
| 20:51.58 | pecenipicek | how does dundi behave when hosts with dynamic ip's try to connect? |
| 20:52.08 | p3nguin | drift-: Perhaps now you need a different wiki page or maybe you could refer to the asterisk book for instructions on configuring phones. |
| 20:52.35 | leifmadsen | there is no 'register' statements in the sample configuration file |
| 20:52.46 | pecenipicek | i know. |
| 20:52.57 | pecenipicek | i'm asking if anyone knows if it works :p |
| 20:53.43 | p3nguin | drift-: After you've read the book, or at least the relevant section(s), if there is a part which you do not understand, I will gladly do my best to answer any questions regarding the configuration of the SIP peer definition. |
| 20:54.01 | drift- | li182-68*CLI> sip show peers No such command 'sip show peers' (type 'core show help sip show' for other possible commands) |
| 20:54.31 | Kobaz | drift-: module load chan_sip.so |
| 20:54.32 | leifmadsen | drift-: chan_sip.so is then not loaded |
| 20:54.33 | p3nguin | drift-: Perhaps you haven't created a sip.conf and loaded the sip module. |
| 20:54.37 | pecenipicek | but from what i noticed with asterisk so far is that if any lines are where they shouldnt be, there's some complaining noted on the CLI after a reload command. |
| 20:54.48 | pecenipicek | and there's none in this case. |
| 20:56.40 | pecenipicek | different kind of question then, how do you make sure that a dundi server thats on a dynamic ip registers to a dundi server with a fixed ip? |
| 20:56.40 | drift- | i fallowed that wiki page |
| 20:56.41 | leifmadsen | I've never registered a DUNDi peer |
| 20:56.42 | drift- | i made a sip.conf |
| 20:56.46 | *** part/#asterisk enzz (~lenz@188.129.224.123) |
| 20:56.58 | pecenipicek | ah, borderline cases, how i love you... |
| 20:56.58 | p3nguin | drift-: correctly? |
| 20:57.12 | drift- | verbatim |
| 20:57.14 | leifmadsen | it's always been static I guess. I'm looking through svn log to see if I can see where it was added. |
| 20:57.32 | drift- | [malcolm] type=peer secret=my_secure_password host=dynamic context=local |
| 20:57.35 | pecenipicek | even tho the host option can be dynamic when defining trusted peers? |
| 20:57.47 | drift- | all i have in my sip.conf is like [mike] type=peer secret=mypass host=dynamic context=local |
| 20:58.01 | p3nguin | drift-: There's more required for a sip.conf. That just adds the phone. |
| 20:58.31 | pecenipicek | points to the samples provided dundi.conf |
| 20:58.39 | pecenipicek | this requires some moar testing on my end. |
| 20:58.39 | drift- | hrm |
| 20:58.43 | drift- | my sip.conf is completly empty |
| 20:59.21 | drift- | other then that heh |
| 20:59.38 | p3nguin | Better get back to reading the book, then. |
| 21:00.45 | engaged | any place cheaper than voip.ms for just forwarding a local DID? |
| 21:00.57 | p3nguin | Not really. |
| 21:01.04 | drift- | my http://pastebin.com/ageyfh1V < modules reload |
| 21:01.22 | p3nguin | $1 per phone number per month is pretty reasonable. |
| 21:02.20 | drift- | li182-68*CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status mike (Unspecified) D 0 Unmonitored |
| 21:02.24 | drift- | ok wee are getting some where now heh |
| 21:02.33 | drift- | Name/username Host Dyn Forcerport ACL Port Stat |
| 21:02.35 | drift- | it registered! |
| 21:02.35 | drift- | :)_ |
| 21:02.41 | drift- | mike/mike 172.16.1.4 D 5060 Unmonitored |
| 21:02.41 | engaged | p3nguin: there is a per minute fee too though |
| 21:02.53 | engaged | i guess their per minute fee is pretty reasonable though |
| 21:03.06 | *** join/#asterisk pecenipicek (~pecenipic@cpe-109-60-73-149.zg3.cable.xnet.hr) |
| 21:03.08 | p3nguin | In that case, I have no idea where to look for just forwarding numbers. |
| 21:03.20 | p3nguin | I send my DIDs to my PBXs and deal with them there. |
| 21:03.22 | drift- | i got busy signal :( when i try to dial out |
| 21:03.25 | pecenipicek | sorry, my net switched ip's for a moment and locked me outta everywhere |
| 21:03.44 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
| 21:03.46 | pecenipicek | leifmadsen , if you wrote anything i might have missed it. |
| 21:03.49 | p3nguin | drift-: You have to create at least one useful extension. |
| 21:04.02 | drift- | okay |
| 21:04.45 | leifmadsen | pecenipicek: I didn't write anything. My guess is that host=dynamic is used for inbound requests from that peer where the IP address is dynamic. So the lookups always have to be to an IP address (static) but the other end can by dynamic, and we'll just match on EID and key. |
| 21:05.05 | p3nguin | drift-: Those go in extensions.conf, in case you didn't read the friggin' book. |
| 21:07.06 | WIMPy | Thinking about it, it has to be that way. Dundi requests may be forwarded so it's no use to register with one peer. |
| 21:07.37 | drift- | i did not read book but i do learn quick |
| 21:07.48 | drift- | i'm sure i'm real close to making this thing work now |
| 21:07.52 | p3nguin | drift-: Do you have an aversion to reading? |
| 21:07.58 | pecenipicek | leifmadsen , ah. that might be a problem. |
| 21:08.05 | drift- | lol well i'm reading here arent i? |
| 21:08.24 | drift- | i have super ADD |
| 21:08.27 | p3nguin | drift-: I'm just trying to figure out if you're more special than everyone else who has to read. |
| 21:08.28 | drift- | lol |
| 21:08.33 | WIMPy | drift-: It reads a little like tweeting. |
| 21:08.34 | drift- | not at all |
| 21:08.46 | pecenipicek | the problem on my end is that i must have the lookups working through all the peers in the net. |
| 21:08.55 | drift- | i been working on this for 2 days now heh |
| 21:08.56 | pecenipicek | and on the remote peer, well, it just dont work |
| 21:09.01 | drift- | its should of been 2 hour project |
| 21:09.01 | leifmadsen | pecenipicek: looking at the code, there may be a register=yes line |
| 21:09.18 | pecenipicek | doesnt hurt to try now. |
| 21:09.20 | drift- | just getting irratated as hell , and id offer to pay for help just to end this |
| 21:09.21 | WIMPy | pecenipicek: Yes, so it can't work, just like explained. |
| 21:09.25 | p3nguin | drift-: If you would have spent the first day reading the book, the second day would have allowed you to complete the configuration. |
| 21:09.35 | leifmadsen | although I'm not sure how it would be used |
| 21:09.40 | drift- | what book? |
| 21:09.44 | p3nguin | ~book |
| 21:09.44 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 21:09.48 | p3nguin | ~newbook |
| 21:09.48 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/ Released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 21:10.06 | pecenipicek | WIMPy, the sheer amount of "it cant work do it this way" stuff i heard in the last few days from this channel and other sources begs me to think otherwise. |
| 21:10.35 | carrar | You can't say that |
| 21:10.55 | leifmadsen | pecenipicek: you're welcome to look at the code and figure it out on your own then. I'm done looking for you now. |
| 21:10.56 | carrar | say it this way, "This channel is great" |
| 21:11.26 | pecenipicek | leifmadsen , i appreciate it. |
| 21:11.38 | pecenipicek | coding however isnt my strong point. |
| 21:11.42 | pecenipicek | nor is reading code :p |
| 21:11.52 | pecenipicek | but i might take a peek for the dundi module. |
| 21:11.58 | leifmadsen | either is mine, and yet I'm still trying to help you, but your tone makes me wish to go help someone else |
| 21:12.01 | carrar | no time like the present to start learning!! |
| 21:12.03 | pecenipicek | and the register=yes statement at least does something. |
| 21:12.43 | leifmadsen | This didn't impress me earlier: <pecenipicek> this channel continues to prove over and over again how useless it is when you come to anything above "I use only one Asterisk box for a home PBX"... Not that the overall documentation is in a much better scope... |
| 21:12.59 | pecenipicek | i dont doubt that. |
| 21:13.03 | *** join/#asterisk Gugge (~gugge@vlan2.dlxhosting.dk) |
| 21:13.08 | pecenipicek | cant say it isnt true in many cases however. |
| 21:13.15 | pecenipicek | especially on the documentation front. |
| 21:13.36 | p3nguin | You're upset that people use an Asterisk-based PBX at home? |
| 21:13.39 | leifmadsen | so because you're using an edge case solution that hasn't been documented, this channel is useless. Nice. |
| 21:14.20 | *** join/#asterisk CraigW76 (~techcaw@cl-427.dal-01.us.sixxs.net) |
| 21:14.26 | carrar | feels so useless |
| 21:14.32 | p3nguin | needs more jalapeno poppers |
| 21:14.46 | pecenipicek | p3nguin , nope. i'm upset that the documentation is pretty much limited to that use with limited forays outwards. |
| 21:15.28 | pecenipicek | but i will apologise for the previous outburst, and it was out of line. |
| 21:15.30 | leifmadsen | I'll try not to suck at writing documentation so much then. |
| 21:15.33 | carrar | pecenipicek, please contribute to the documentation |
| 21:15.37 | p3nguin | It isn't the fault, nor the responsibility, of the channel and people in it that there is not a lot of documentation. |
| 21:15.48 | leifmadsen | p3nguin: it's totally my fault though :) |
| 21:15.55 | p3nguin | I agree! |
| 21:15.56 | leifmadsen | but mostly russellb's |
| 21:15.57 | p3nguin | ;) |
| 21:16.10 | carrar | ALL leifmadsen & russellb's fault!! |
| 21:16.13 | carrar | ALL!!! |
| 21:16.19 | pecenipicek | i will. once i develop this fully. after all, i wouldnt be touching this if i wasnt paid to develop a solution for a company here in croatia to replace their CCM solution. |
| 21:16.21 | p3nguin | Documentation is kind of hard to write. |
| 21:16.53 | carrar | In a Charlton Heston voice, "Damn! You" |
| 21:17.01 | p3nguin | I've tried to write docs about how to do things, but to be quite honest, once I get the project done I really don't care to take the time to write docs. |
| 21:17.11 | pecenipicek | i'll have to. |
| 21:17.12 | russellb | ~newbook |
| 21:17.12 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/ Released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 21:17.16 | russellb | that's all I have to say about that. |
| 21:17.27 | carrar | no 4th? |
| 21:17.29 | carrar | wtf |
| 21:17.49 | pecenipicek | since we'll be paid for maintenance of the whole deal after the basic setup. and there are some things here that i truly do not want to forget. |
| 21:20.22 | pecenipicek | and as a side note, leifmadsen , the register=yes statement helped. |
| 21:21.33 | WIMPy | pecenipicek: If your host is dynamic, what des it hand out as URIs? |
| 21:22.08 | pecenipicek | WIMPy , umm, what? |
| 21:22.52 | WIMPy | pecenipicek: You configure URIs in dundi.conf that get handed out in response to maching requests. |
| 21:23.13 | pecenipicek | as in mappings or? |
| 21:23.38 | WIMPy | yes |
| 21:24.23 | pecenipicek | <PROTECTED> |
| 21:24.40 | carrar | russellb, is there a pdf of the 3rd edition? |
| 21:24.52 | russellb | we do not plan to distribute a PDF. |
| 21:25.01 | russellb | We plan to make it freely available in web form. |
| 21:25.05 | carrar | thats the best way to read it |
| 21:25.09 | WIMPy | pecenipicek: It that IP is dynamic do you disable caching? |
| 21:25.27 | russellb | If you purchase it as an ebook from O'Reilly, you can download it in a number of electronic formats, including PDF. |
| 21:25.33 | pecenipicek | its static actually. |
| 21:25.47 | pecenipicek | they're all written on the respective servers. |
| 21:26.03 | carrar | purchase things over the internet? |
| 21:26.04 | russellb | but that will not be available until it is shipping (March). |
| 21:26.06 | carrar | I've heard of tht |
| 21:26.08 | carrar | that |
| 21:26.11 | pecenipicek | i just didnt feel like giving out any ip addresses in public so i just replace the address with the ${IPADDR} var. |
| 21:26.35 | pecenipicek | why do you ask tho? |
| 21:26.45 | WIMPy | pecenipicek: You ARE giving out the address. |
| 21:26.56 | pecenipicek | i meant in the irc. |
| 21:26.59 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 21:27.02 | *** join/#asterisk ccesario (~ccesario@201-42-148-53.dsl.telesp.net.br) |
| 21:27.15 | pecenipicek | and i'm aware of the whois command on irc giving out my public ip :p |
| 21:27.26 | devmod | when using mysql cdr can I do Set(CDR(MyOwnField)=blah) ?? |
| 21:27.31 | carrar | I'm logged into your keyboard right now |
| 21:27.40 | p3nguin | haha |
| 21:27.41 | pecenipicek | yeah yeah |
| 21:27.43 | WIMPy | pecenipicek: And why do you ask about a register you obviousely don't need? |
| 21:27.51 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 21:27.52 | leifmadsen | devmod: yes, if you use cdr_adaptive_odbc |
| 21:27.55 | pecenipicek | i need it actually. |
| 21:28.13 | leifmadsen | if the register=yes uses mappings to send the register command, then that is very strange indeed |
| 21:28.15 | pecenipicek | there's a third remote box that i need to connect to my local boxes and we connect to it via VPN. |
| 21:28.17 | p3nguin | static addresses...and you need it why? |
| 21:28.40 | pecenipicek | but, the problem is that the vpn address we get when connected to the network is not static and changes every 90 minutes. |
| 21:28.54 | citywok | hmmm. what's the dect phone of choice? |
| 21:29.01 | carrar | hahah |
| 21:29.09 | carrar | ip changes every 90 minutes |
| 21:29.13 | carrar | thats funny |
| 21:29.51 | pecenipicek | and the admins of said other network were quite... unhelpful when we requested that they make the address we get static when connecting to their vpn |
| 21:29.53 | devmod | leifmadsen, got it thanks |
| 21:30.05 | pecenipicek | guess they didnt hear of fixed dhcp leases, but whatever. |
| 21:30.27 | pecenipicek | i'm not that familiar with various VPN stuff and on top of that they use Cisco VPN concentrators for the VPN. |
| 21:31.27 | pecenipicek | if it will clarify i'll share the three dundi.conf files i'm using atm. |
| 21:32.24 | WIMPy | pecenipicek: Can both other boxes reach the one on the VPN with the same address at all? |
| 21:33.20 | pecenipicek | box 1 dundi conf http://pastebin.com/Hjri3bvA |
| 21:33.20 | pecenipicek | box 2 dundi conf http://pastebin.com/PTn32Sxv |
| 21:33.20 | pecenipicek | box 3 dundi conf http://pastebin.com/35mB6dTm |
| 21:33.56 | pecenipicek | all three boxes have static ip addresses in their respective networks. |
| 21:34.10 | pecenipicek | box 1 connects via vpnc to the remote network where box 3 is situated. |
| 21:34.17 | WIMPy | That does not answer my question. |
| 21:34.32 | pecenipicek | the ip address box 1 gets when connected to the vpn changes every 90 minutes. |
| 21:34.46 | p3nguin | That's ridiculous configuration. |
| 21:34.48 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 21:34.54 | pecenipicek | and yes, at this point it all works. |
| 21:35.01 | p3nguin | There can't be any good purpose for doing that. |
| 21:35.18 | pecenipicek | p3nguin, not really my fault. i blame it on their sysadmins not being able to configure their stuff correctly. |
| 21:35.27 | p3nguin | yeah |
| 21:35.39 | p3nguin | I'd demand an explanation. |
| 21:35.51 | pecenipicek | we did. |
| 21:36.24 | pecenipicek | "We cannot do it as its not supported in that particular subnet." |
| 21:36.33 | p3nguin | If they don't give a satisfactory response, I'd send Guido over to get to the bottom of it. |
| 21:36.38 | pecenipicek | :p |
| 21:36.39 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 21:36.46 | pecenipicek | eh, i dont care. |
| 21:36.55 | carrar | Please esculate to your CEO |
| 21:37.01 | p3nguin | I bet they change port numbers, too, and call it "security." |
| 21:37.01 | pecenipicek | what ceo? |
| 21:37.05 | carrar | thanks |
| 21:37.14 | pecenipicek | me and my father are doing this as freelancers for them :p |
| 21:37.15 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 21:37.26 | pecenipicek | p3nguin, yup. |
| 21:37.53 | p3nguin | How did I know that? |
| 21:38.13 | pecenipicek | amusingly enough, our ast server thats in their server room is probably more secure than all their servers combined. |
| 21:38.22 | pecenipicek | simply due to the virtue of being configured properly. |
| 21:38.40 | p3nguin | It's not hard when you know what you're doing. :) |
| 21:38.42 | carrar | secure systems are so unusable!! |
| 21:38.56 | pecenipicek | and not using a pass in the likes of <insert username>_<insert number from 1 to 4> |
| 21:39.15 | carrar | open root |
| 21:39.34 | pecenipicek | define open root. |
| 21:39.44 | pecenipicek | as in root that someone can ssh into remotely? |
| 21:40.06 | carrar | no passwords |
| 21:40.11 | carrar | open system |
| 21:40.13 | carrar | no security |
| 21:40.19 | p3nguin | I used to run a server with no root password. |
| 21:40.21 | carrar | trust the internet |
| 21:40.33 | carrar | we're here 4U |
| 21:40.45 | p3nguin | It was actually more secure (outside of having physical access). |
| 21:41.42 | p3nguin | Not having a root password prevented many mechanisms from working, so root was pretty difficult to get. |
| 21:43.17 | p3nguin | Of course it had nothing to do with obtaining root privs through unconventional means. |
| 21:45.36 | pecenipicek | carrar nope. but they do all they work in root. |
| 21:48.22 | *** join/#asterisk Tekati (~robert@cpe-75-85-246-201.bak.res.rr.com) |
| 21:50.01 | Tekati | Can anyone tell me if the Cisco SPA8800 is a good unit to work with Asterisk with 4 FXO and 4 FXS ports? Or would I be better off putting in a AEX844B card? |
| 21:52.26 | carrar | Go Digium cards |
| 21:52.44 | p3nguin | or maybe Sangoma? |
| 21:53.39 | carrar | or Audio Codes |
| 21:55.16 | p3nguin | Seriously? I thought those were kind of crappy. |
| 21:55.29 | carrar | nope |
| 21:55.38 | carrar | if anything, the LinkSys (SPA) are crappy |
| 21:55.42 | p3nguin | yeah |
| 21:55.44 | _Corey_ | Digium cards do a decent job :) |
| 21:57.15 | pecenipicek | done with this for today. thanks for the help. |
| 21:57.17 | pecenipicek | later |
| 22:14.39 | *** join/#asterisk bjhaid (~IceChat7@41.206.12.5.vgccl.net) |
| 22:22.15 | *** join/#asterisk bjhaid (~IceChat7@41.206.12.6.vgccl.net) |
| 22:41.27 | *** join/#asterisk m_tadeu (~quassel@89-180-176-52.net.novis.pt) |
| 22:47.14 | paulc | Having a brain fart... I've got source for Asterisk 1.6.2.16 - make menuselect shows res-jabber as [*], but after make; make install and running Asterisk, I can't use JabberSend in my dialplan.. any ideas why? I get "... No application 'JabberSend' for extension (ono-call-agent, s, 28)" |
| 22:49.55 | Qwell | It's probably not loading because there is no config, or the config is invalid, |
| 22:53.25 | paulc | Hmm.. I thought my config was ok.. but that said, it's a brand new box.. trying to set it up + write the install guide for the ops guys, maybe something fell through the cracks.. I'll compare with my "known good working box" :) |
| 22:53.30 | paulc | Thanks @Qwell |
| 22:53.53 | Qwell | module load res_jabber.so |
| 23:00.33 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 23:08.18 | paulc | turns out it didn't like where iksemel was installed from source.. all fixed, all good, I'm happy - thanks again :) |
| 23:12.42 | *** join/#asterisk joobie (~joobie@CPE-121-214-141-146.lnse3.lon.bigpond.net.au) |
| 23:23.49 | *** join/#asterisk CapitanA (~A@adsl-71-130-211-79.dsl.irvnca.pacbell.net) |
| 23:23.53 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.111) |
| 23:24.43 | CunningPike | Has anyone had any luck getting the Polycom setting softkey.feature.basicCallManagement.redundant="0" to have an effect? This is SIP 3.1.4 on IP501 phones |
| 23:25.01 | CunningPike | And yes, I have mapped all the hardkeys |
| 23:36.38 | CapitanA | my asterisk server is in the internet... my phones are at the office (different networks)... do I need to open the RTP port range on my server's firewall? |
| 23:37.54 | p3nguin | If you don't have some other way of passing the media ports into the PBX, yes. |
| 23:39.06 | CapitanA | p3nguin, thanks for the answer... i thought the media was gonna be directly between the phones and the trunk... |
| 23:39.38 | p3nguin | What, exactly, is this "trunk" you speak of? |
| 23:40.34 | CapitanA | p3nguin, a little more info would be helpful huh? :P sorry |
| 23:41.09 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 23:46.10 | CapitanA | ok i'm back... p3nguin, my phones are on one network... my asterisk box is on another... and the voip provider is on a whole other network |
| 23:46.24 | p3nguin | not uncommon |
| 23:46.25 | CapitanA | sorry for leaving out details |
| 23:47.40 | CapitanA | so my (probably flawed) understanding was as follows: i was under the impression that after signaling between the provider and my server, that the "conversation" goes directly between the voip provider and the phones |
| 23:50.37 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 23:59.11 | paulc | It can do.. but doesn't have to.. |